From db600b2cf7bfebf44dd67b5d3103336c8aba0dea Mon Sep 17 00:00:00 2001 From: openvela-robot Date: Sat, 24 May 2014 11:15:15 +0800 Subject: [PATCH] vf_format: rework format list parsing Avoid using AV_PIX_FMT_NB, since that hardcodes the number of pixel formats into lavfi and will break when a shared lavu is updated, adding new pixel formats. --- .gitignore | 38 +- CREDITS | 59 +- Changelog | 575 +- INSTALL | 9 +- LICENSE | 70 +- MAINTAINERS | 566 - Makefile | 135 +- README | 12 +- RELEASE | 2 +- arch.mak | 5 - avconv.c | 2633 +++++ ffmpeg.h => avconv.h | 161 +- ffmpeg_dxva2.c => avconv_dxva2.c | 13 +- ffmpeg_filter.c => avconv_filter.c | 447 +- ffmpeg_opt.c => avconv_opt.c | 980 +- ffmpeg_vda.c => avconv_vda.c | 10 +- ffmpeg_vdpau.c => avconv_vdpau.c | 10 +- avplay.c | 3001 +++++ avprobe.c | 969 ++ ffserver.c => avserver.c | 720 +- cmdutils.c | 721 +- cmdutils.h | 119 +- cmdutils_common_opts.h | 16 +- cmdutils_opencl.c | 274 - common.mak | 93 +- compat/aix/math.h | 14 +- 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libavcodec/cook_parser.c | 15 +- libavcodec/cookdata.h | 8 +- libavcodec/copy_block.h | 19 +- libavcodec/cos_tablegen.c | 10 +- libavcodec/cpia.c | 233 - libavcodec/crystalhd.c | 1223 -- libavcodec/cscd.c | 142 +- libavcodec/cyuv.c | 30 +- libavcodec/dca.c | 12 +- libavcodec/dca.h | 8 +- libavcodec/dca_parser.c | 18 +- libavcodec/dcadata.h | 8 +- libavcodec/dcadec.c | 775 +- libavcodec/dcadsp.c | 8 +- libavcodec/dcadsp.h | 10 +- libavcodec/dcaenc.c | 975 -- libavcodec/dcaenc.h | 113 - libavcodec/dcahuff.h | 8 +- libavcodec/dct-test.c | 115 +- libavcodec/dct.c | 10 +- libavcodec/dct.h | 11 +- libavcodec/dct32.h | 8 +- libavcodec/dct32_fixed.c | 8 +- libavcodec/dct32_float.c | 8 +- libavcodec/dct32_template.c | 8 +- libavcodec/dctref.c | 8 +- libavcodec/dctref.h | 8 +- libavcodec/dfa.c | 42 +- libavcodec/dirac.c | 48 +- libavcodec/dirac.h | 11 +- libavcodec/dirac_arith.c | 116 - libavcodec/dirac_arith.h | 191 - libavcodec/dirac_dwt.c | 572 - libavcodec/dirac_dwt.h | 123 - libavcodec/dirac_parser.c | 12 +- libavcodec/diracdec.c | 2001 ---- libavcodec/diracdsp.c | 201 - libavcodec/diracdsp.h | 67 - libavcodec/dnxhd_parser.c | 50 +- libavcodec/dnxhddata.c | 696 +- libavcodec/dnxhddata.h | 20 +- libavcodec/dnxhddec.c | 80 +- libavcodec/dnxhdenc.c | 115 +- libavcodec/dnxhdenc.h | 10 +- libavcodec/dpcm.c | 19 +- libavcodec/dpx.c | 304 +- libavcodec/dpx_parser.c | 108 - libavcodec/dpxenc.c | 164 +- libavcodec/dsd_tablegen.c | 38 - libavcodec/dsd_tablegen.h | 95 - libavcodec/dsddec.c | 167 - libavcodec/dsicinav.c | 71 +- libavcodec/dsputil.c | 263 +- libavcodec/dsputil.h | 24 +- libavcodec/dsputil_template.c | 8 +- libavcodec/dsputilenc_template.c | 10 +- libavcodec/dump_extradata_bsf.c | 13 +- libavcodec/dv.c | 20 +- libavcodec/dv.h | 11 +- libavcodec/dv_profile.c | 46 +- libavcodec/dv_profile.h | 14 +- libavcodec/dv_tablegen.c | 8 +- libavcodec/dv_tablegen.h | 8 +- libavcodec/dvbsub.c | 118 +- libavcodec/dvbsub_parser.c | 10 +- libavcodec/dvbsubdec.c | 238 +- libavcodec/dvd_nav_parser.c | 116 - libavcodec/dvdata.c | 8 +- libavcodec/dvdata.h | 8 +- libavcodec/dvdec.c | 47 +- libavcodec/dvdsub_parser.c | 8 +- libavcodec/dvdsubdec.c | 261 +- libavcodec/dvdsubenc.c | 400 +- libavcodec/dvenc.c | 35 +- libavcodec/dxa.c | 95 +- libavcodec/dxtory.c | 22 +- libavcodec/dxva2.c | 8 +- libavcodec/dxva2.h | 19 +- libavcodec/dxva2_h264.c | 18 +- libavcodec/dxva2_internal.h | 9 +- libavcodec/dxva2_mpeg2.c | 21 +- libavcodec/dxva2_vc1.c | 54 +- libavcodec/eac3_data.c | 8 +- libavcodec/eac3_data.h | 8 +- libavcodec/eac3dec.c | 8 +- libavcodec/eac3enc.c | 16 +- libavcodec/eac3enc.h | 8 +- libavcodec/eacmv.c | 20 +- libavcodec/eaidct.c | 8 +- libavcodec/eaidct.h | 8 +- libavcodec/eamad.c | 55 +- libavcodec/eatgq.c | 24 +- libavcodec/eatgv.c | 34 +- libavcodec/eatqi.c | 15 +- libavcodec/elbg.c | 55 +- libavcodec/elbg.h | 16 +- libavcodec/error_resilience.c | 283 +- libavcodec/error_resilience.h | 13 +- libavcodec/escape124.c | 39 +- libavcodec/escape130.c | 18 +- libavcodec/evrcdata.h | 1499 --- libavcodec/evrcdec.c | 917 -- libavcodec/exif.c | 137 - libavcodec/exif.h | 170 - libavcodec/exr.c | 132 +- libavcodec/faandct.h | 8 +- libavcodec/faanidct.c | 8 +- libavcodec/faanidct.h | 8 +- libavcodec/faxcompr.c | 34 +- libavcodec/faxcompr.h | 8 +- libavcodec/fft-fixed-test.c | 8 +- libavcodec/fft-fixed32-test.c | 21 - libavcodec/fft-internal.h | 29 +- libavcodec/fft-test.c | 50 +- libavcodec/fft.h | 28 +- libavcodec/fft_fixed.c | 9 +- libavcodec/fft_fixed_32.c | 52 - libavcodec/fft_float.c | 9 +- libavcodec/fft_init_table.c | 200 - libavcodec/fft_table.h | 66 - libavcodec/fft_template.c | 192 +- libavcodec/ffv1.c | 164 +- libavcodec/ffv1.h | 65 +- libavcodec/ffv1dec.c | 677 +- libavcodec/ffv1enc.c | 749 +- libavcodec/ffwavesynth.c | 481 - libavcodec/fic.c | 21 +- libavcodec/flac.c | 11 +- libavcodec/flac.h | 8 +- libavcodec/flac_parser.c | 109 +- libavcodec/flacdata.c | 10 +- libavcodec/flacdata.h | 10 +- libavcodec/flacdec.c | 71 +- libavcodec/flacdsp.c | 10 +- libavcodec/flacdsp.h | 9 +- libavcodec/flacdsp_lpc_template.c | 26 +- libavcodec/flacdsp_template.c | 8 +- libavcodec/flacenc.c | 75 +- libavcodec/flashsv.c | 50 +- libavcodec/flashsv2enc.c | 920 -- libavcodec/flashsvenc.c | 14 +- libavcodec/flicvideo.c | 140 +- libavcodec/flv.h | 8 +- libavcodec/flvdec.c | 23 +- libavcodec/flvenc.c | 8 +- libavcodec/fmtconvert.c | 40 +- libavcodec/fmtconvert.h | 13 +- libavcodec/frame_thread_encoder.c | 295 - libavcodec/frame_thread_encoder.h | 26 - libavcodec/fraps.c | 224 +- libavcodec/frwu.c | 43 +- libavcodec/g2meet.c | 51 +- libavcodec/g722.c | 8 +- libavcodec/g722.h | 8 +- libavcodec/g722dec.c | 12 +- libavcodec/g722enc.c | 15 +- libavcodec/g723_1.c | 1304 +-- libavcodec/g723_1_data.h | 171 +- libavcodec/g726.c | 49 +- libavcodec/g729.h | 29 - libavcodec/g729data.h | 382 - libavcodec/g729dec.c | 726 -- libavcodec/g729postfilter.c | 610 - libavcodec/g729postfilter.h | 116 - libavcodec/get_bits.h | 136 +- libavcodec/gif.c | 264 +- libavcodec/gif.h | 49 - libavcodec/gifdec.c | 444 +- libavcodec/golomb-test.c | 12 +- libavcodec/golomb.c | 8 +- libavcodec/golomb.h | 42 +- libavcodec/gsm.h | 8 +- libavcodec/gsm_parser.c | 13 +- libavcodec/gsmdec.c | 17 +- libavcodec/gsmdec_data.c | 8 +- libavcodec/gsmdec_data.h | 8 +- libavcodec/gsmdec_template.c | 8 +- libavcodec/h261.c | 8 +- libavcodec/h261.h | 9 +- libavcodec/h261_parser.c | 22 +- libavcodec/h261data.c | 8 +- libavcodec/h261dec.c | 36 +- libavcodec/h261enc.c | 111 +- libavcodec/h263.c | 8 +- libavcodec/h263.h | 13 +- libavcodec/h263_parser.c | 22 +- libavcodec/h263_parser.h | 8 +- libavcodec/h263data.h | 12 +- libavcodec/h263dec.c | 173 +- libavcodec/h263dsp.c | 8 +- libavcodec/h263dsp.h | 8 +- libavcodec/h264.c | 520 +- libavcodec/h264.h | 113 +- libavcodec/h264_cabac.c | 68 +- libavcodec/h264_cavlc.c | 124 +- libavcodec/h264_direct.c | 17 +- libavcodec/h264_loopfilter.c | 20 +- libavcodec/h264_mb.c | 41 +- libavcodec/h264_mb_template.c | 25 +- libavcodec/h264_mc_template.c | 14 +- libavcodec/h264_mp4toannexb_bsf.c | 59 +- libavcodec/h264_mvpred.h | 18 +- libavcodec/h264_parser.c | 110 +- libavcodec/h264_picture.c | 43 +- libavcodec/h264_ps.c | 146 +- libavcodec/h264_refs.c | 99 +- libavcodec/h264_sei.c | 171 +- libavcodec/h264_slice.c | 455 +- libavcodec/h264addpx_template.c | 8 +- libavcodec/h264chroma.c | 10 +- libavcodec/h264chroma.h | 12 +- libavcodec/h264chroma_template.c | 56 +- libavcodec/h264data.h | 10 +- libavcodec/h264dsp.c | 56 +- libavcodec/h264dsp.h | 8 +- libavcodec/h264dsp_template.c | 46 +- libavcodec/h264idct.c | 16 +- libavcodec/h264idct.h | 10 +- libavcodec/h264idct_template.c | 12 +- libavcodec/h264pred.c | 28 +- libavcodec/h264pred.h | 10 +- libavcodec/h264pred_template.c | 170 +- libavcodec/h264qpel.c | 26 +- libavcodec/h264qpel.h | 8 +- libavcodec/h264qpel_template.c | 116 +- libavcodec/hevc.c | 1587 +-- libavcodec/hevc.h | 128 +- libavcodec/hevc_cabac.c | 679 +- libavcodec/hevc_filter.c | 371 +- libavcodec/hevc_mvs.c | 132 +- libavcodec/hevc_parser.c | 242 +- libavcodec/hevc_ps.c | 41 +- libavcodec/hevc_refs.c | 43 +- libavcodec/hevc_sei.c | 103 +- libavcodec/hevcdsp.c | 138 +- libavcodec/hevcdsp.h | 60 +- libavcodec/hevcdsp_template.c | 1085 +- libavcodec/hevcpred.c | 12 +- libavcodec/hevcpred.h | 45 - libavcodec/hevcpred_template.c | 271 +- libavcodec/hnm4video.c | 97 +- libavcodec/hpel_template.c | 8 +- libavcodec/hpeldsp.c | 10 +- libavcodec/hpeldsp.h | 13 +- libavcodec/huffman.c | 59 +- libavcodec/huffman.h | 13 +- libavcodec/huffyuv.c | 34 +- libavcodec/huffyuv.h | 35 +- libavcodec/huffyuvdec.c | 543 +- libavcodec/huffyuvenc.c | 476 +- libavcodec/idcinvideo.c | 23 +- libavcodec/iff.c | 722 +- libavcodec/iirfilter.c | 15 +- libavcodec/iirfilter.h | 31 +- libavcodec/imc.c | 43 +- libavcodec/imcdata.h | 8 +- libavcodec/imgconvert.c | 191 +- libavcodec/imgconvert.h | 8 +- libavcodec/imx_dump_header_bsf.c | 13 +- libavcodec/indeo2.c | 12 +- libavcodec/indeo2data.h | 8 +- libavcodec/indeo3.c | 99 +- libavcodec/indeo3data.h | 8 +- libavcodec/indeo4.c | 89 +- libavcodec/indeo4data.h | 8 +- libavcodec/indeo5.c | 53 +- libavcodec/indeo5data.h | 8 +- libavcodec/intelh263dec.c | 19 +- libavcodec/internal.h | 76 +- libavcodec/interplayvideo.c | 62 +- libavcodec/intrax8.c | 45 +- libavcodec/intrax8.h | 11 +- libavcodec/intrax8dsp.c | 8 +- libavcodec/intrax8dsp.h | 8 +- libavcodec/intrax8huf.h | 8 +- libavcodec/ituh263dec.c | 66 +- libavcodec/ituh263enc.c | 20 +- libavcodec/ivi_common.c | 89 +- libavcodec/ivi_common.h | 10 +- libavcodec/ivi_dsp.c | 68 +- libavcodec/ivi_dsp.h | 12 +- libavcodec/j2kenc.c | 1055 -- libavcodec/jacosub.h | 45 - libavcodec/jacosubdec.c | 205 - libavcodec/jfdctint.c | 8 +- libavcodec/jfdctint_template.c | 4 +- libavcodec/jpeg2000.c | 125 +- libavcodec/jpeg2000.h | 22 +- libavcodec/jpeg2000dec.c | 474 +- libavcodec/jpeg2000dwt.c | 237 +- libavcodec/jpeg2000dwt.h | 9 +- libavcodec/jpegls.c | 14 +- libavcodec/jpegls.h | 15 +- libavcodec/jpeglsdec.c | 149 +- libavcodec/jpeglsdec.h | 8 +- libavcodec/jpeglsenc.c | 25 +- libavcodec/jrevdct.c | 249 +- libavcodec/jvdec.c | 43 +- libavcodec/kbdwin.c | 22 +- libavcodec/kbdwin.h | 11 +- libavcodec/kgv1dec.c | 43 +- libavcodec/kmvc.c | 37 +- libavcodec/lagarith.c | 151 +- libavcodec/lagarithrac.c | 12 +- libavcodec/lagarithrac.h | 8 +- libavcodec/latm_parser.c | 9 +- libavcodec/lcl.h | 8 +- libavcodec/lcldec.c | 42 +- libavcodec/lclenc.c | 31 +- libavcodec/libaacplus.c | 141 - libavcodec/libavcodec.v | 27 - libavcodec/libcelt_dec.c | 140 - libavcodec/libfaac.c | 30 +- libavcodec/libfdk-aacdec.c | 10 +- libavcodec/libfdk-aacenc.c | 6 +- libavcodec/libgsm.c | 46 +- libavcodec/libilbc.c | 26 +- libavcodec/libmp3lame.c | 23 +- libavcodec/libopencore-amr.c | 40 +- libavcodec/libopenjpegdec.c | 147 +- libavcodec/libopenjpegenc.c | 484 +- libavcodec/libopus.c | 8 +- libavcodec/libopus.h | 8 +- libavcodec/libopusdec.c | 51 +- libavcodec/libopusenc.c | 49 +- libavcodec/libschroedinger.c | 13 +- libavcodec/libschroedinger.h | 8 +- libavcodec/libschroedingerdec.c | 24 +- libavcodec/libschroedingerenc.c | 13 +- libavcodec/libshine.c | 149 - libavcodec/libspeexdec.c | 33 +- libavcodec/libspeexenc.c | 17 +- libavcodec/libstagefright.cpp | 590 - libavcodec/libtheoraenc.c | 72 +- libavcodec/libtwolame.c | 10 +- libavcodec/libutvideo.h | 70 - libavcodec/libutvideodec.cpp | 212 - libavcodec/libutvideoenc.cpp | 246 - libavcodec/libvo-aacenc.c | 20 +- libavcodec/libvo-amrwbenc.c | 18 +- libavcodec/{libvorbisenc.c => libvorbis.c} | 104 +- libavcodec/libvorbisdec.c | 199 - libavcodec/libvpx.c | 21 +- libavcodec/libvpx.h | 10 +- libavcodec/libvpxdec.c | 16 +- libavcodec/libvpxenc.c | 380 +- libavcodec/libwavpackenc.c | 8 +- libavcodec/libwebpenc.c | 8 +- libavcodec/libx264.c | 226 +- libavcodec/libx265.c | 10 +- libavcodec/libxavs.c | 26 +- libavcodec/libxvid.c | 160 +- libavcodec/libxvid.h | 8 +- libavcodec/libxvid_rc.c | 59 +- libavcodec/libzvbi-teletextdec.c | 570 - libavcodec/ljpegenc.c | 70 +- libavcodec/loco.c | 107 +- libavcodec/lossless_videodsp.c | 128 - libavcodec/lossless_videodsp.h | 40 - libavcodec/lpc.c | 26 +- libavcodec/lpc.h | 13 +- libavcodec/lsp.c | 20 +- libavcodec/lsp.h | 8 +- libavcodec/lzw.c | 14 +- libavcodec/lzw.h | 9 +- libavcodec/lzwenc.c | 17 +- libavcodec/mace.c | 19 +- libavcodec/mathops.h | 8 +- libavcodec/mathtables.c | 10 +- libavcodec/mdct_fixed.c | 9 +- libavcodec/mdct_fixed_32.c | 52 - libavcodec/mdct_float.c | 9 +- libavcodec/mdct_template.c | 14 +- libavcodec/mdec.c | 26 +- libavcodec/metasound.c | 10 +- libavcodec/metasound_data.c | 16 +- libavcodec/metasound_data.h | 8 +- libavcodec/microdvddec.c | 377 - libavcodec/mimic.c | 23 +- libavcodec/mips/Makefile | 20 - libavcodec/mips/aaccoder_mips.c | 2498 ----- libavcodec/mips/aacdec_mips.c | 831 -- libavcodec/mips/aacdec_mips.h | 249 - libavcodec/mips/aacpsdsp_mips.c | 459 - libavcodec/mips/aacpsy_mips.h | 230 - libavcodec/mips/aacsbr_mips.c | 618 - libavcodec/mips/aacsbr_mips.h | 493 - libavcodec/mips/ac3dsp_mips.c | 412 - libavcodec/mips/acelp_filters_mips.c | 216 - libavcodec/mips/acelp_vectors_mips.c | 101 - libavcodec/mips/amrwbdec_mips.c | 187 - libavcodec/mips/amrwbdec_mips.h | 62 - libavcodec/mips/celp_filters_mips.c | 288 - libavcodec/mips/celp_math_mips.c | 89 - libavcodec/mips/compute_antialias_fixed.h | 250 - libavcodec/mips/compute_antialias_float.h | 185 - libavcodec/mips/fft_mips.c | 534 - libavcodec/mips/fmtconvert_mips.c | 342 - libavcodec/mips/iirfilter_mips.c | 204 - libavcodec/mips/lsp_mips.h | 109 - libavcodec/mips/mathops.h | 8 +- libavcodec/mips/mpegaudiodsp_mips_fixed.c | 907 -- libavcodec/mips/mpegaudiodsp_mips_float.c | 1250 --- libavcodec/mips/sbrdsp_mips.c | 940 -- libavcodec/mjpeg.c | 12 +- libavcodec/mjpeg.h | 9 +- libavcodec/mjpeg2jpeg_bsf.c | 8 +- libavcodec/mjpeg_parser.c | 68 +- libavcodec/mjpega_dump_header_bsf.c | 13 +- libavcodec/mjpegbdec.c | 18 +- libavcodec/mjpegdec.c | 1097 +- libavcodec/mjpegdec.h | 25 +- libavcodec/mjpegenc.c | 228 +- libavcodec/mjpegenc.h | 17 +- libavcodec/mlp.c | 8 +- libavcodec/mlp.h | 8 +- libavcodec/mlp_parser.c | 55 +- libavcodec/mlp_parser.h | 13 +- libavcodec/mlpdec.c | 74 +- libavcodec/mlpdsp.c | 8 +- libavcodec/mlpdsp.h | 8 +- libavcodec/mmvideo.c | 25 +- libavcodec/motion-test.c | 151 - libavcodec/motion_est.c | 96 +- libavcodec/motion_est_template.c | 64 +- libavcodec/motionpixels.c | 60 +- libavcodec/motionpixels_tablegen.c | 8 +- libavcodec/motionpixels_tablegen.h | 10 +- libavcodec/movsub_bsf.c | 18 +- libavcodec/movtextdec.c | 116 - libavcodec/movtextenc.c | 165 - libavcodec/mp3_header_decompress_bsf.c | 97 - libavcodec/mpc.c | 14 +- libavcodec/mpc.h | 8 +- libavcodec/mpc7.c | 28 +- libavcodec/mpc7data.h | 8 +- libavcodec/mpc8.c | 24 +- libavcodec/mpc8data.h | 8 +- libavcodec/mpc8huff.h | 8 +- libavcodec/mpcdata.h | 46 +- libavcodec/mpeg12.c | 17 +- libavcodec/mpeg12.h | 8 +- libavcodec/mpeg12data.c | 76 +- libavcodec/mpeg12data.h | 11 +- libavcodec/mpeg12dec.c | 502 +- libavcodec/mpeg12enc.c | 141 +- libavcodec/mpeg4audio.c | 18 +- libavcodec/mpeg4audio.h | 8 +- libavcodec/mpeg4data.h | 8 +- libavcodec/mpeg4video.c | 8 +- libavcodec/mpeg4video.h | 28 +- libavcodec/mpeg4video_parser.c | 21 +- libavcodec/mpeg4video_parser.h | 8 +- libavcodec/mpeg4videodec.c | 491 +- libavcodec/mpeg4videoenc.c | 87 +- libavcodec/mpegaudio.c | 8 +- libavcodec/mpegaudio.h | 16 +- libavcodec/mpegaudio_parser.c | 15 +- libavcodec/mpegaudio_tablegen.c | 8 +- libavcodec/mpegaudio_tablegen.h | 15 +- libavcodec/mpegaudiodata.c | 18 +- libavcodec/mpegaudiodata.h | 8 +- libavcodec/mpegaudiodec_fixed.c | 10 +- libavcodec/mpegaudiodec_float.c | 10 +- libavcodec/mpegaudiodec_template.c | 109 +- libavcodec/mpegaudiodecheader.c | 21 +- libavcodec/mpegaudiodecheader.h | 10 +- libavcodec/mpegaudiodectab.h | 8 +- libavcodec/mpegaudiodsp.c | 10 +- libavcodec/mpegaudiodsp.h | 11 +- libavcodec/mpegaudiodsp_data.c | 8 +- libavcodec/mpegaudiodsp_fixed.c | 10 +- libavcodec/mpegaudiodsp_float.c | 10 +- libavcodec/mpegaudiodsp_template.c | 17 +- ...mpegaudioenc_template.c => mpegaudioenc.c} | 90 +- libavcodec/mpegaudioenc_fixed.c | 41 - libavcodec/mpegaudioenc_float.c | 42 - libavcodec/mpegaudiotab.h | 8 +- libavcodec/mpegutils.c | 8 +- libavcodec/mpegutils.h | 8 +- libavcodec/mpegvideo.c | 1199 +- libavcodec/mpegvideo.h | 69 +- libavcodec/mpegvideo_enc.c | 515 +- libavcodec/mpegvideo_motion.c | 38 +- libavcodec/mpegvideo_parser.c | 34 +- libavcodec/mpegvideo_xvmc.c | 61 +- libavcodec/mpl2dec.c | 96 - libavcodec/mqc.c | 8 +- libavcodec/mqc.h | 22 +- libavcodec/mqcdec.c | 8 +- libavcodec/mqcenc.c | 119 - libavcodec/msgsmdec.c | 8 +- libavcodec/msgsmdec.h | 8 +- libavcodec/msmpeg4.c | 41 +- libavcodec/msmpeg4.h | 8 +- libavcodec/msmpeg4data.c | 8 +- libavcodec/msmpeg4data.h | 8 +- libavcodec/msmpeg4dec.c | 39 +- libavcodec/msmpeg4enc.c | 20 +- libavcodec/msrle.c | 29 +- libavcodec/msrledec.c | 81 +- libavcodec/msrledec.h | 8 +- libavcodec/mss1.c | 26 +- libavcodec/mss12.c | 17 +- libavcodec/mss12.h | 22 +- libavcodec/mss2.c | 41 +- libavcodec/mss2dsp.c | 11 +- libavcodec/mss2dsp.h | 8 +- libavcodec/mss3.c | 15 +- libavcodec/mss34dsp.c | 12 +- libavcodec/mss34dsp.h | 8 +- libavcodec/mss4.c | 18 +- libavcodec/msvideo1.c | 14 +- libavcodec/msvideo1enc.c | 305 - libavcodec/mvcdec.c | 8 +- libavcodec/mxpegdec.c | 31 +- libavcodec/nellymoserdec.c | 9 +- libavcodec/nellymoserenc.c | 24 +- libavcodec/noise_bsf.c | 19 +- libavcodec/nuv.c | 67 +- libavcodec/old_codec_ids.h | 399 - libavcodec/on2avc.c | 12 +- libavcodec/on2avcdata.c | 28 +- libavcodec/on2avcdata.h | 28 +- libavcodec/options.c | 91 +- libavcodec/options_table.h | 105 +- libavcodec/opus.c | 8 +- libavcodec/opus.h | 38 +- libavcodec/opus_celt.c | 12 +- libavcodec/opus_imdct.c | 8 +- libavcodec/opus_imdct.h | 8 +- libavcodec/opus_parser.c | 8 +- libavcodec/opus_silk.c | 8 +- libavcodec/opusdec.c | 66 +- libavcodec/paf.c | 44 +- libavcodec/paf.h | 28 - libavcodec/pamenc.c | 75 +- libavcodec/parser.c | 61 +- libavcodec/parser.h | 8 +- libavcodec/pcm-bluray.c | 19 +- libavcodec/pcm-dvd.c | 46 +- libavcodec/pcm.c | 183 +- libavcodec/pcm_tablegen.c | 8 +- libavcodec/pcm_tablegen.h | 8 +- libavcodec/pcx.c | 129 +- libavcodec/pcxenc.c | 30 +- libavcodec/pgssubdec.c | 245 +- libavcodec/pictordec.c | 50 +- libavcodec/pixels.h | 8 +- libavcodec/png.c | 17 +- libavcodec/png.h | 15 +- libavcodec/png_parser.c | 27 +- libavcodec/pngdec.c | 582 +- libavcodec/pngdsp.c | 8 +- libavcodec/pngdsp.h | 8 +- libavcodec/pngenc.c | 158 +- libavcodec/pnm.c | 68 +- libavcodec/pnm.h | 8 +- libavcodec/pnm_parser.c | 8 +- libavcodec/pnmdec.c | 113 +- libavcodec/pnmenc.c | 22 +- libavcodec/ppc/asm.S | 16 +- libavcodec/ppc/dsputil_altivec.c | 8 +- libavcodec/ppc/dsputil_altivec.h | 8 +- libavcodec/ppc/dsputil_ppc.c | 13 +- libavcodec/ppc/fdct_altivec.c | 8 +- libavcodec/ppc/fft_altivec.c | 8 +- libavcodec/ppc/fft_altivec_s.S | 8 +- libavcodec/ppc/fmtconvert_altivec.c | 8 +- libavcodec/ppc/gmc_altivec.c | 12 +- libavcodec/ppc/h264chroma_init.c | 8 +- libavcodec/ppc/h264chroma_template.c | 8 +- libavcodec/ppc/h264dsp.c | 8 +- libavcodec/ppc/h264qpel.c | 8 +- libavcodec/ppc/h264qpel_template.c | 8 +- libavcodec/ppc/hpeldsp_altivec.c | 8 +- libavcodec/ppc/hpeldsp_altivec.h | 8 +- libavcodec/ppc/idct_altivec.c | 12 +- libavcodec/ppc/int_altivec.c | 8 +- libavcodec/ppc/mathops.h | 8 +- libavcodec/ppc/mpegaudiodsp_altivec.c | 8 +- libavcodec/ppc/mpegvideo_altivec.c | 8 +- libavcodec/ppc/vc1dsp_altivec.c | 8 +- libavcodec/ppc/videodsp_ppc.c | 8 +- libavcodec/ppc/vorbisdsp_altivec.c | 8 +- libavcodec/ppc/vp3dsp_altivec.c | 8 +- libavcodec/ppc/vp8dsp_altivec.c | 8 +- libavcodec/proresdata.c | 8 +- libavcodec/proresdata.h | 8 +- libavcodec/{proresdec_lgpl.c => proresdec.c} | 20 +- libavcodec/proresdec.h | 55 - libavcodec/proresdec2.c | 688 -- libavcodec/proresdsp.c | 16 +- libavcodec/proresdsp.h | 13 +- .../{proresenc_kostya.c => proresenc.c} | 24 +- libavcodec/proresenc_anatoliy.c | 626 -- libavcodec/psymodel.c | 34 +- libavcodec/psymodel.h | 10 +- libavcodec/pthread.c | 8 +- libavcodec/pthread_frame.c | 158 +- libavcodec/pthread_internal.h | 8 +- libavcodec/pthread_slice.c | 78 +- libavcodec/ptx.c | 20 +- libavcodec/put_bits.h | 19 +- libavcodec/qcelpdata.h | 40 +- libavcodec/qcelpdec.c | 46 +- libavcodec/qdm2.c | 135 +- libavcodec/qdm2_tablegen.c | 8 +- libavcodec/qdm2_tablegen.h | 29 +- libavcodec/qdm2data.h | 8 +- libavcodec/qdrw.c | 14 +- libavcodec/qpeg.c | 59 +- libavcodec/qtrle.c | 76 +- libavcodec/qtrleenc.c | 156 +- libavcodec/r210dec.c | 29 +- libavcodec/r210enc.c | 123 - libavcodec/ra144.c | 37 +- libavcodec/ra144.h | 20 +- libavcodec/ra144dec.c | 23 +- libavcodec/ra144enc.c | 30 +- libavcodec/ra288.c | 27 +- libavcodec/ra288.h | 8 +- libavcodec/ralf.c | 12 +- libavcodec/ralfdata.h | 8 +- libavcodec/rangecoder.c | 15 +- libavcodec/rangecoder.h | 16 +- libavcodec/ratecontrol.c | 71 +- libavcodec/ratecontrol.h | 16 +- libavcodec/raw.c | 91 +- libavcodec/raw.h | 18 +- libavcodec/rawdec.c | 219 +- libavcodec/rawenc.c | 12 +- libavcodec/rdft.c | 8 +- libavcodec/rdft.h | 8 +- libavcodec/realtextdec.c | 83 - libavcodec/rectangle.h | 21 +- libavcodec/remove_extradata_bsf.c | 13 +- libavcodec/resample.c | 435 - libavcodec/resample2.c | 319 - libavcodec/rl.h | 8 +- libavcodec/rl2.c | 16 +- libavcodec/rle.c | 8 +- libavcodec/rle.h | 8 +- libavcodec/rnd_avg.h | 11 +- libavcodec/roqaudioenc.c | 16 +- libavcodec/roqvideo.c | 8 +- libavcodec/roqvideo.h | 12 +- libavcodec/roqvideodec.c | 33 +- libavcodec/roqvideoenc.c | 79 +- libavcodec/rpza.c | 20 +- libavcodec/rtjpeg.c | 8 +- libavcodec/rtjpeg.h | 8 +- libavcodec/rv10.c | 77 +- libavcodec/rv10enc.c | 8 +- libavcodec/rv20enc.c | 20 +- libavcodec/rv30.c | 25 +- libavcodec/rv30data.h | 10 +- libavcodec/rv30dsp.c | 8 +- libavcodec/rv34.c | 60 +- libavcodec/rv34.h | 10 +- libavcodec/rv34_parser.c | 8 +- libavcodec/rv34data.h | 8 +- libavcodec/rv34dsp.c | 8 +- libavcodec/rv34dsp.h | 8 +- libavcodec/rv34vlc.h | 8 +- libavcodec/rv40.c | 15 +- libavcodec/rv40data.h | 8 +- libavcodec/rv40dsp.c | 13 +- libavcodec/rv40vlc2.h | 8 +- libavcodec/s302m.c | 37 +- libavcodec/s302menc.c | 178 - libavcodec/s3tc.c | 8 +- libavcodec/s3tc.h | 8 +- libavcodec/samidec.c | 158 - libavcodec/sanm.c | 267 +- libavcodec/sbr.h | 38 +- libavcodec/sbrdsp.c | 11 +- libavcodec/sbrdsp.h | 9 +- libavcodec/sgi.h | 8 +- libavcodec/sgidec.c | 53 +- libavcodec/sgienc.c | 23 +- libavcodec/sgirledec.c | 18 +- libavcodec/sh4/README | 6 - libavcodec/shorten.c | 90 +- libavcodec/simple_idct.c | 16 +- libavcodec/simple_idct.h | 13 +- libavcodec/simple_idct_template.c | 58 +- libavcodec/sinewin.c | 8 +- libavcodec/sinewin.h | 8 +- libavcodec/sinewin_tablegen.c | 8 +- libavcodec/sinewin_tablegen.h | 8 +- libavcodec/sipr.c | 12 +- libavcodec/sipr.h | 8 +- libavcodec/sipr16k.c | 9 +- libavcodec/sipr16kdata.h | 8 +- libavcodec/siprdata.h | 8 +- libavcodec/smacker.c | 169 +- libavcodec/smc.c | 14 +- libavcodec/smvjpegdec.c | 209 - libavcodec/snow.c | 709 -- libavcodec/snow.h | 706 -- libavcodec/snow_dwt.c | 865 -- libavcodec/snow_dwt.h | 127 - libavcodec/snowdata.h | 132 - libavcodec/snowdec.c | 607 - libavcodec/snowenc.c | 2005 ---- libavcodec/sonic.c | 1094 -- libavcodec/sp5x.h | 8 +- libavcodec/sp5xdec.c | 19 +- libavcodec/sparc/README | 6 - libavcodec/srtdec.c | 91 +- libavcodec/srtenc.c | 332 - libavcodec/startcode.c | 57 - libavcodec/startcode.h | 35 - libavcodec/subviewerdec.c | 83 - libavcodec/sunrast.c | 63 +- libavcodec/sunrast.h | 8 +- libavcodec/sunrastenc.c | 35 +- libavcodec/svq1.c | 8 +- libavcodec/svq1.h | 8 +- libavcodec/svq13.c | 8 +- libavcodec/svq1_cb.h | 8 +- libavcodec/svq1_vlc.h | 8 +- libavcodec/svq1dec.c | 64 +- libavcodec/svq1enc.c | 63 +- libavcodec/svq1enc_cb.h | 8 +- libavcodec/svq3.c | 127 +- libavcodec/svq3.h | 8 +- libavcodec/synth_filter.c | 8 +- libavcodec/synth_filter.h | 8 +- libavcodec/tableprint.h | 8 +- libavcodec/tak.c | 58 +- libavcodec/tak.h | 13 +- libavcodec/tak_parser.c | 32 +- libavcodec/takdec.c | 345 +- libavcodec/targa.c | 161 +- libavcodec/targa.h | 15 +- libavcodec/targa_y216dec.c | 83 - libavcodec/targaenc.c | 48 +- libavcodec/textdec.c | 149 - libavcodec/thread.h | 23 +- libavcodec/tiertexseqv.c | 14 +- libavcodec/tiff.c | 804 +- libavcodec/tiff.h | 111 +- libavcodec/tiff_common.c | 313 - libavcodec/tiff_common.h | 152 - libavcodec/tiff_data.c | 1870 --- libavcodec/tiff_data.h | 92 - libavcodec/tiffenc.c | 230 +- libavcodec/tmv.c | 16 +- libavcodec/tpel_template.c | 8 +- libavcodec/tpeldsp.c | 8 +- libavcodec/tpeldsp.h | 8 +- libavcodec/truemotion1.c | 18 +- libavcodec/truemotion1data.h | 8 +- libavcodec/truemotion2.c | 108 +- libavcodec/truespeech.c | 12 +- libavcodec/truespeech_data.h | 8 +- libavcodec/tscc.c | 24 +- libavcodec/tscc2.c | 12 +- libavcodec/tscc2data.h | 8 +- libavcodec/tta.c | 367 +- libavcodec/ttadata.c | 52 - libavcodec/ttadata.h | 50 - libavcodec/ttadsp.c | 57 - libavcodec/ttadsp.h | 34 - libavcodec/ttaenc.c | 232 - libavcodec/twinvq.c | 12 +- libavcodec/twinvq.h | 8 +- libavcodec/twinvq_data.h | 8 +- libavcodec/twinvqdec.c | 10 +- libavcodec/txd.c | 22 +- libavcodec/ulti.c | 12 +- libavcodec/ulti_cb.h | 8 +- libavcodec/unary.h | 8 +- libavcodec/utils.c | 1803 +-- libavcodec/utvideo.c | 8 +- libavcodec/utvideo.h | 8 +- libavcodec/utvideodec.c | 16 +- libavcodec/utvideoenc.c | 29 +- libavcodec/v210dec.c | 128 +- libavcodec/v210dec.h | 36 - libavcodec/v210enc.c | 24 +- libavcodec/v210x.c | 8 +- libavcodec/v308dec.c | 83 - libavcodec/v308enc.c | 94 - libavcodec/v408dec.c | 103 - libavcodec/v408enc.c | 113 - libavcodec/v410dec.c | 16 +- libavcodec/v410enc.c | 14 +- libavcodec/vaapi.c | 33 +- libavcodec/vaapi.h | 12 +- libavcodec/vaapi_h264.c | 23 +- libavcodec/vaapi_internal.h | 8 +- libavcodec/vaapi_mpeg.c | 48 - libavcodec/vaapi_mpeg2.c | 12 +- libavcodec/vaapi_mpeg4.c | 21 +- libavcodec/vaapi_vc1.c | 12 +- libavcodec/vb.c | 18 +- libavcodec/vble.c | 107 +- libavcodec/vc1.c | 134 +- libavcodec/vc1.h | 31 +- libavcodec/vc1_parser.c | 289 +- libavcodec/vc1acdata.h | 8 +- libavcodec/vc1data.c | 22 +- libavcodec/vc1data.h | 26 +- libavcodec/vc1dec.c | 388 +- libavcodec/vc1dsp.c | 180 +- libavcodec/vc1dsp.h | 22 +- libavcodec/vcr1.c | 45 +- libavcodec/vda.c | 8 +- libavcodec/vda.h | 30 +- libavcodec/vda_h264.c | 75 +- libavcodec/vda_h264_dec.c | 273 - libavcodec/vda_internal.h | 8 +- libavcodec/vdpau.c | 376 +- libavcodec/vdpau.h | 44 +- libavcodec/vdpau_h264.c | 32 +- libavcodec/vdpau_internal.h | 33 +- libavcodec/vdpau_mpeg12.c | 8 +- libavcodec/vdpau_mpeg4.c | 8 +- libavcodec/vdpau_vc1.c | 12 +- libavcodec/version.h | 30 +- libavcodec/videodsp.c | 9 +- libavcodec/videodsp.h | 27 +- libavcodec/videodsp_template.c | 34 +- libavcodec/vima.c | 18 +- libavcodec/vmdav.c | 108 +- libavcodec/vmnc.c | 33 +- libavcodec/vorbis.c | 27 +- libavcodec/vorbis.h | 8 +- libavcodec/vorbis_data.c | 8 +- libavcodec/vorbis_enc_data.h | 20 +- libavcodec/vorbis_parser.c | 44 +- libavcodec/vorbis_parser.h | 26 +- libavcodec/vorbisdec.c | 191 +- libavcodec/vorbisdsp.c | 8 +- libavcodec/vorbisdsp.h | 8 +- libavcodec/vorbisenc.c | 33 +- libavcodec/vp3.c | 104 +- libavcodec/vp3_parser.c | 8 +- libavcodec/vp3data.h | 8 +- libavcodec/vp3dsp.c | 9 +- libavcodec/vp3dsp.h | 8 +- libavcodec/vp5.c | 37 +- libavcodec/vp56.c | 341 +- libavcodec/vp56.h | 44 +- libavcodec/vp56data.c | 8 +- libavcodec/vp56data.h | 8 +- libavcodec/vp56dsp.c | 8 +- libavcodec/vp56dsp.h | 8 +- libavcodec/vp56rac.c | 8 +- libavcodec/vp5data.h | 8 +- libavcodec/vp6.c | 92 +- libavcodec/vp6data.h | 8 +- libavcodec/vp6dsp.c | 8 +- libavcodec/vp8.c | 144 +- libavcodec/vp8.h | 15 +- libavcodec/vp8_parser.c | 8 +- libavcodec/vp8data.h | 8 +- libavcodec/vp8dsp.c | 24 +- libavcodec/vp8dsp.h | 8 +- libavcodec/vp9.c | 3772 +------ libavcodec/vp9.h | 355 +- libavcodec/vp9_parser.c | 118 - libavcodec/vp9block.c | 1684 +++ libavcodec/vp9data.c | 2133 ++++ libavcodec/vp9data.h | 2173 +--- libavcodec/vp9dsp.c | 1460 +-- libavcodec/vp9dsp.h | 118 - libavcodec/vp9mvs.c | 352 + libavcodec/vp9prob.c | 274 + libavcodec/vqavideo.c | 102 +- libavcodec/wavpack.c | 216 +- libavcodec/wavpack.h | 194 - libavcodec/wavpackenc.c | 2989 ----- libavcodec/wavpackenc.h | 664 -- libavcodec/webp.c | 56 +- libavcodec/webvttdec.c | 100 - libavcodec/webvttenc.c | 219 - libavcodec/wma.c | 28 +- libavcodec/wma.h | 10 +- libavcodec/wma_common.c | 8 +- libavcodec/wma_common.h | 8 +- libavcodec/wmadata.h | 10 +- libavcodec/wmadec.c | 49 +- libavcodec/wmaenc.c | 86 +- libavcodec/wmalosslessdec.c | 75 +- libavcodec/wmaprodata.h | 8 +- libavcodec/wmaprodec.c | 54 +- libavcodec/wmavoice.c | 32 +- libavcodec/wmavoice_data.h | 8 +- libavcodec/wmv2.c | 14 +- libavcodec/wmv2.h | 10 +- libavcodec/wmv2dec.c | 12 +- libavcodec/wmv2dsp.c | 8 +- libavcodec/wmv2dsp.h | 8 +- libavcodec/wmv2enc.c | 25 +- libavcodec/wnv1.c | 18 +- libavcodec/ws-snd1.c | 14 +- libavcodec/x86/Makefile | 29 +- libavcodec/x86/ac3dsp.asm | 8 +- libavcodec/x86/ac3dsp_init.c | 13 +- libavcodec/x86/cabac.h | 58 +- libavcodec/x86/cavsdsp.c | 141 +- libavcodec/x86/constants.c | 8 +- libavcodec/x86/constants.h | 10 +- libavcodec/x86/dca.h | 8 +- libavcodec/x86/dcadsp.asm | 19 +- libavcodec/x86/dcadsp_init.c | 13 +- libavcodec/x86/dct32.asm | 10 +- libavcodec/x86/dct_init.c | 8 +- libavcodec/x86/deinterlace.asm | 8 +- libavcodec/x86/dirac_dwt.c | 202 - libavcodec/x86/dirac_dwt.h | 30 - libavcodec/x86/diracdsp_mmx.c | 157 - libavcodec/x86/diracdsp_mmx.h | 47 - libavcodec/x86/diracdsp_yasm.asm | 264 - libavcodec/x86/dnxhdenc.asm | 8 +- libavcodec/x86/dnxhdenc_init.c | 8 +- libavcodec/x86/dsputil.asm | 230 +- libavcodec/x86/dsputil_init.c | 86 +- libavcodec/x86/dsputil_mmx.c | 214 +- libavcodec/x86/dsputil_qns_template.c | 10 +- libavcodec/x86/dsputil_x86.c | 8 +- libavcodec/x86/dsputil_x86.h | 19 +- libavcodec/x86/dsputilenc.asm | 44 +- libavcodec/x86/dsputilenc_mmx.c | 161 +- libavcodec/x86/dwt_yasm.asm | 306 - libavcodec/x86/fdct.c | 12 +- libavcodec/x86/fft.asm | 23 +- libavcodec/x86/fft.h | 8 +- libavcodec/x86/fft_init.c | 8 +- libavcodec/x86/flacdsp.asm | 74 - libavcodec/x86/flacdsp_init.c | 45 - libavcodec/x86/fmtconvert.asm | 8 +- libavcodec/x86/fmtconvert_init.c | 8 +- libavcodec/x86/fpel.asm | 138 +- libavcodec/x86/fpel.h | 10 +- libavcodec/x86/fpel_mmx.c | 140 + libavcodec/x86/h263_loopfilter.asm | 10 +- libavcodec/x86/h263dsp_init.c | 8 +- libavcodec/x86/h264_chromamc.asm | 8 +- libavcodec/x86/h264_chromamc_10bit.asm | 12 +- libavcodec/x86/h264_deblock.asm | 20 +- libavcodec/x86/h264_deblock_10bit.asm | 16 +- libavcodec/x86/h264_i386.h | 14 +- libavcodec/x86/h264_idct.asm | 8 +- libavcodec/x86/h264_idct_10bit.asm | 24 +- libavcodec/x86/h264_intrapred.asm | 13 +- libavcodec/x86/h264_intrapred_10bit.asm | 50 +- libavcodec/x86/h264_intrapred_init.c | 8 +- libavcodec/x86/h264_qpel.c | 26 +- libavcodec/x86/h264_qpel_10bit.asm | 8 +- libavcodec/x86/h264_qpel_8bit.asm | 8 +- libavcodec/x86/h264_weight.asm | 15 +- libavcodec/x86/h264_weight_10bit.asm | 8 +- libavcodec/x86/h264chroma_init.c | 8 +- libavcodec/x86/h264dsp_init.c | 10 +- libavcodec/x86/hevc_deblock.asm | 831 -- libavcodec/x86/hevc_mc.asm | 1261 --- libavcodec/x86/hevcdsp.h | 123 - libavcodec/x86/hevcdsp_init.c | 414 - libavcodec/x86/hpeldsp.asm | 329 +- libavcodec/x86/hpeldsp.h | 23 +- libavcodec/x86/hpeldsp_init.c | 80 +- libavcodec/x86/hpeldsp_mmx.c | 53 + libavcodec/x86/hpeldsp_rnd_template.c | 8 +- libavcodec/x86/idct_mmx_xvid.c | 8 +- libavcodec/x86/idct_sse2_xvid.c | 29 +- libavcodec/x86/idct_xvid.h | 8 +- libavcodec/x86/imdct36.asm | 35 +- libavcodec/x86/inline_asm.h | 8 +- libavcodec/x86/lossless_videodsp.asm | 292 - libavcodec/x86/lossless_videodsp_init.c | 62 - libavcodec/x86/lpc.c | 14 +- libavcodec/x86/mathops.h | 13 +- libavcodec/x86/mlpdsp.c | 13 +- libavcodec/x86/motion_est.c | 33 +- libavcodec/x86/mpeg4qpel.asm | 10 +- libavcodec/x86/mpegaudiodsp.c | 47 +- libavcodec/x86/mpegvideo.c | 28 +- libavcodec/x86/mpegvideoenc.c | 22 +- libavcodec/x86/mpegvideoenc_template.c | 71 +- libavcodec/x86/pngdsp.asm | 8 +- libavcodec/x86/pngdsp_init.c | 8 +- libavcodec/x86/proresdsp.asm | 134 +- libavcodec/x86/proresdsp_init.c | 10 +- libavcodec/x86/qpel.asm | 8 +- libavcodec/x86/rnd_template.c | 8 +- libavcodec/x86/rv34dsp.asm | 8 +- libavcodec/x86/rv34dsp_init.c | 8 +- libavcodec/x86/rv40dsp.asm | 8 +- libavcodec/x86/rv40dsp_init.c | 58 +- libavcodec/x86/sbrdsp.asm | 214 +- libavcodec/x86/sbrdsp_init.c | 30 +- libavcodec/x86/simple_idct.c | 12 +- libavcodec/x86/snowdsp.c | 909 -- libavcodec/x86/ttadsp.asm | 119 - libavcodec/x86/ttadsp_init.c | 42 - libavcodec/x86/v210-init.c | 48 - libavcodec/x86/v210.asm | 88 - libavcodec/x86/vc1dsp.asm | 8 +- libavcodec/x86/vc1dsp.h | 8 +- libavcodec/x86/vc1dsp_init.c | 15 +- libavcodec/x86/vc1dsp_mmx.c | 120 +- libavcodec/x86/videodsp.asm | 60 +- libavcodec/x86/videodsp_init.c | 45 +- libavcodec/x86/vorbisdsp.asm | 8 +- libavcodec/x86/vorbisdsp_init.c | 8 +- libavcodec/x86/vp3dsp.asm | 8 +- libavcodec/x86/vp3dsp_init.c | 71 +- libavcodec/x86/vp56_arith.h | 23 +- libavcodec/x86/vp6dsp.asm | 8 +- libavcodec/x86/vp6dsp_init.c | 8 +- libavcodec/x86/vp8dsp.asm | 8 +- libavcodec/x86/vp8dsp_init.c | 16 +- libavcodec/x86/vp8dsp_loopfilter.asm | 8 +- libavcodec/x86/{vp9mc.asm => vp9dsp.asm} | 155 +- libavcodec/x86/vp9dsp_init.c | 389 +- libavcodec/x86/vp9intrapred.asm | 1405 --- libavcodec/x86/vp9itxfm.asm | 1669 --- libavcodec/x86/vp9lpf.asm | 817 -- libavcodec/x86/w64xmmtest.c | 15 +- libavcodec/xan.c | 102 +- libavcodec/xbmdec.c | 124 +- libavcodec/xbmenc.c | 33 +- libavcodec/xface.c | 381 - libavcodec/xface.h | 95 - libavcodec/xfacedec.c | 188 - libavcodec/xfaceenc.c | 238 - libavcodec/xiph.c | 10 +- libavcodec/xiph.h | 10 +- libavcodec/xl.c | 15 +- libavcodec/xsubdec.c | 17 +- libavcodec/xsubenc.c | 20 +- libavcodec/xvmc.h | 12 +- libavcodec/xvmc_internal.h | 15 +- libavcodec/xwd.h | 8 +- libavcodec/xwddec.c | 21 +- libavcodec/xwdenc.c | 45 +- libavcodec/xxan.c | 33 +- libavcodec/y41pdec.c | 92 - libavcodec/y41penc.c | 102 - libavcodec/yop.c | 48 +- libavcodec/yuv4dec.c | 84 - libavcodec/yuv4enc.c | 91 - libavcodec/zerocodec.c | 6 +- libavcodec/zmbv.c | 123 +- libavcodec/zmbvenc.c | 12 +- libavdevice/Makefile | 42 +- libavdevice/alldevices.c | 28 +- libavdevice/alsa-audio-common.c | 85 +- libavdevice/alsa-audio-dec.c | 63 +- libavdevice/alsa-audio-enc.c | 63 +- libavdevice/alsa-audio.h | 22 +- libavdevice/avdevice.c | 205 +- libavdevice/avdevice.h | 428 +- libavdevice/avdeviceres.rc | 55 - libavdevice/avfoundation.m | 374 - libavdevice/bktr.c | 30 +- libavdevice/caca.c | 241 - libavdevice/decklink_enc.cpp | 588 - libavdevice/decklink_enc.h | 43 - libavdevice/decklink_enc_c.c | 56 - libavdevice/dshow.c | 1098 -- libavdevice/dshow_capture.h | 279 - libavdevice/dshow_common.c | 190 - libavdevice/dshow_enummediatypes.c | 103 - libavdevice/dshow_enumpins.c | 105 - libavdevice/dshow_filter.c | 202 - libavdevice/dshow_pin.c | 362 - libavdevice/dv1394.c | 13 +- libavdevice/dv1394.h | 8 +- libavdevice/{fbdev_dec.c => fbdev.c} | 94 +- libavdevice/fbdev_common.c | 129 - libavdevice/fbdev_common.h | 38 - libavdevice/fbdev_enc.c | 220 - libavdevice/gdigrab.c | 630 -- libavdevice/iec61883.c | 498 - libavdevice/jack_audio.c | 33 +- libavdevice/lavfi.c | 440 - libavdevice/libcdio.c | 11 +- libavdevice/libdc1394.c | 9 +- libavdevice/openal-dec.c | 253 - libavdevice/opengl_enc.c | 1306 --- libavdevice/opengl_enc_shaders.h | 188 - libavdevice/oss_audio.c | 26 +- libavdevice/{pulse_audio_dec.c => pulse.c} | 80 +- libavdevice/pulse_audio_common.c | 249 - libavdevice/pulse_audio_common.h | 38 - libavdevice/pulse_audio_enc.c | 796 -- libavdevice/qtkit.m | 358 - libavdevice/sdl.c | 375 - libavdevice/sndio_common.c | 10 +- libavdevice/sndio_common.h | 10 +- libavdevice/sndio_dec.c | 9 +- libavdevice/sndio_enc.c | 19 +- libavdevice/timefilter.c | 52 +- libavdevice/timefilter.h | 27 +- libavdevice/v4l.c | 363 - libavdevice/v4l2-common.c | 102 - libavdevice/v4l2-common.h | 62 - libavdevice/v4l2.c | 817 +- libavdevice/v4l2enc.c | 118 - libavdevice/version.h | 16 +- libavdevice/vfwcap.c | 54 +- libavdevice/x11grab.c | 106 +- libavdevice/xv.c | 367 - libavfilter/Makefile | 164 +- libavfilter/aeval.c | 466 - libavfilter/af_aconvert.c | 196 - libavfilter/af_adelay.c | 282 - libavfilter/af_aecho.c | 359 - libavfilter/af_afade.c | 300 - libavfilter/af_aformat.c | 29 +- libavfilter/af_amerge.c | 350 - libavfilter/af_amix.c | 50 +- libavfilter/af_anull.c | 27 +- libavfilter/af_apad.c | 155 - libavfilter/af_aphaser.c | 358 - libavfilter/af_aresample.c | 318 - libavfilter/af_asetnsamples.c | 196 - libavfilter/af_asetrate.c | 119 - libavfilter/af_ashowinfo.c | 55 +- libavfilter/af_astats.c | 274 - libavfilter/af_astreamsync.c | 240 - libavfilter/af_asyncts.c | 39 +- libavfilter/af_atempo.c | 1196 -- libavfilter/af_biquads.c | 620 - libavfilter/af_bs2b.c | 16 +- libavfilter/af_channelmap.c | 28 +- libavfilter/af_channelsplit.c | 40 +- libavfilter/af_compand.c | 139 +- libavfilter/af_earwax.c | 172 - libavfilter/af_join.c | 53 +- libavfilter/af_ladspa.c | 705 -- libavfilter/af_pan.c | 427 - libavfilter/af_replaygain.c | 613 - libavfilter/af_resample.c | 27 +- libavfilter/af_silencedetect.c | 212 - libavfilter/af_volume.c | 204 +- libavfilter/af_volume.h | 36 +- libavfilter/af_volumedetect.c | 159 - libavfilter/all_channel_layouts.inc | 68 - libavfilter/allfilters.c | 137 +- libavfilter/asink_anullsink.c | 10 +- libavfilter/asrc_abuffer.h | 91 - libavfilter/asrc_anullsrc.c | 117 +- libavfilter/asrc_flite.c | 283 - libavfilter/asrc_sine.c | 223 - libavfilter/audio.c | 58 +- libavfilter/audio.h | 50 +- libavfilter/avcodec.c | 157 - libavfilter/avcodec.h | 110 - libavfilter/avf_avectorscope.c | 274 - libavfilter/avf_concat.c | 426 - libavfilter/avf_showspectrum.c | 527 - libavfilter/avf_showwaves.c | 256 - libavfilter/avfilter.c | 606 +- libavfilter/avfilter.h | 462 +- libavfilter/avfiltergraph.c | 598 +- libavfilter/avfiltergraph.h | 9 +- libavfilter/avfilterres.rc | 55 - libavfilter/bbox.c | 75 - libavfilter/bbox.h | 44 - libavfilter/buffer.c | 106 +- libavfilter/bufferqueue.h | 121 - libavfilter/buffersink.c | 426 +- libavfilter/buffersink.h | 102 +- libavfilter/buffersrc.c | 198 +- libavfilter/buffersrc.h | 83 +- libavfilter/deshake.h | 103 - libavfilter/deshake_opencl.c | 200 - libavfilter/deshake_opencl.h | 45 - libavfilter/deshake_opencl_kernel.h | 225 - libavfilter/drawutils.c | 489 +- libavfilter/drawutils.h | 120 +- libavfilter/dualinput.c | 83 - libavfilter/dualinput.h | 46 - libavfilter/f_ebur128.c | 941 -- libavfilter/f_interleave.c | 259 - libavfilter/f_perms.c | 178 - libavfilter/f_select.c | 533 - libavfilter/f_sendcmd.c | 576 - libavfilter/f_zmq.c | 275 - libavfilter/fifo.c | 12 +- libavfilter/filtfmts.c | 77 +- libavfilter/formats.c | 332 +- libavfilter/formats.h | 65 +- libavfilter/framesync.c | 329 - libavfilter/framesync.h | 296 - libavfilter/gradfun.h | 16 +- libavfilter/graphdump.c | 165 - libavfilter/graphparser.c | 133 +- libavfilter/internal.h | 163 +- libavfilter/lavfutils.c | 103 - libavfilter/lavfutils.h | 43 - libavfilter/libavfilter.v | 1 - libavfilter/libmpcodecs/av_helpers.h | 27 - libavfilter/libmpcodecs/cpudetect.h | 60 - libavfilter/libmpcodecs/img_format.c | 244 - libavfilter/libmpcodecs/img_format.h | 309 - libavfilter/libmpcodecs/libvo/fastmemcpy.h | 99 - libavfilter/libmpcodecs/libvo/video_out.h | 300 - libavfilter/libmpcodecs/mp_image.c | 257 - libavfilter/libmpcodecs/mp_image.h | 159 - libavfilter/libmpcodecs/mp_msg.h | 166 - libavfilter/libmpcodecs/mpc_info.h | 43 - libavfilter/libmpcodecs/vf.h | 169 - libavfilter/libmpcodecs/vf_eq.c | 240 - libavfilter/libmpcodecs/vf_eq2.c | 519 - libavfilter/libmpcodecs/vf_fspp.c | 2125 ---- libavfilter/libmpcodecs/vf_ilpack.c | 458 - libavfilter/libmpcodecs/vf_pp7.c | 491 - libavfilter/libmpcodecs/vf_softpulldown.c | 163 - libavfilter/libmpcodecs/vf_uspp.c | 394 - libavfilter/libmpcodecs/vfcap.h | 56 - libavfilter/lswsutils.c | 50 - libavfilter/lswsutils.h | 38 - libavfilter/opencl_allkernels.c | 41 - libavfilter/opencl_allkernels.h | 29 - libavfilter/pthread.c | 13 +- libavfilter/setpts.c | 163 +- libavfilter/settb.c | 73 +- libavfilter/split.c | 88 +- libavfilter/src_movie.c | 613 - libavfilter/thread.h | 8 +- libavfilter/transform.c | 201 - libavfilter/transform.h | 127 - libavfilter/trim.c | 121 +- libavfilter/unsharp.h | 81 - libavfilter/unsharp_opencl.c | 389 - libavfilter/unsharp_opencl.h | 34 - libavfilter/unsharp_opencl_kernel.h | 286 - libavfilter/version.h | 31 +- libavfilter/vf_alphamerge.c | 207 - libavfilter/vf_aspect.c | 184 +- libavfilter/vf_bbox.c | 134 - libavfilter/vf_blackdetect.c | 209 - libavfilter/vf_blackframe.c | 59 +- libavfilter/vf_blend.c | 465 - libavfilter/vf_boxblur.c | 99 +- libavfilter/vf_colorbalance.c | 213 - libavfilter/vf_colorchannelmixer.c | 360 - libavfilter/vf_colormatrix.c | 411 - libavfilter/vf_copy.c | 22 +- libavfilter/vf_crop.c | 158 +- libavfilter/vf_cropdetect.c | 74 +- libavfilter/vf_curves.c | 570 - libavfilter/vf_dctdnoiz.c | 433 - libavfilter/vf_decimate.c | 403 - libavfilter/vf_dejudder.c | 187 - libavfilter/vf_delogo.c | 125 +- libavfilter/vf_deshake.c | 580 - libavfilter/vf_drawbox.c | 333 +- libavfilter/vf_drawtext.c | 1240 +- libavfilter/vf_edgedetect.c | 394 - libavfilter/vf_elbg.c | 212 - libavfilter/vf_extractplanes.c | 335 - libavfilter/vf_fade.c | 305 +- libavfilter/vf_field.c | 111 - libavfilter/vf_fieldmatch.c | 984 -- libavfilter/vf_fieldorder.c | 97 +- libavfilter/vf_format.c | 188 +- libavfilter/vf_fps.c | 87 +- libavfilter/vf_framepack.c | 8 +- libavfilter/vf_framestep.c | 101 - libavfilter/vf_frei0r.c | 175 +- libavfilter/vf_geq.c | 280 - libavfilter/vf_gradfun.c | 54 +- libavfilter/vf_hflip.c | 139 +- libavfilter/vf_histeq.c | 281 - libavfilter/vf_histogram.c | 376 - libavfilter/vf_hqdn3d.c | 74 +- libavfilter/vf_hqdn3d.h | 12 +- libavfilter/vf_hue.c | 453 - libavfilter/vf_idet.c | 308 - libavfilter/vf_il.c | 212 - libavfilter/vf_interlace.c | 58 +- libavfilter/vf_kerndeint.c | 318 - libavfilter/vf_libopencv.c | 62 +- libavfilter/vf_lut.c | 220 +- libavfilter/vf_lut3d.c | 815 -- libavfilter/vf_mcdeint.c | 315 - libavfilter/vf_mergeplanes.c | 313 - libavfilter/vf_mp.c | 794 -- libavfilter/vf_mpdecimate.c | 257 - libavfilter/vf_noise.c | 490 - libavfilter/vf_null.c | 23 +- libavfilter/vf_overlay.c | 683 +- libavfilter/vf_owdenoise.c | 342 - libavfilter/vf_pad.c | 209 +- libavfilter/vf_perspective.c | 403 - libavfilter/vf_phase.c | 329 - libavfilter/vf_pixdesctest.c | 38 +- libavfilter/vf_pp.c | 186 - libavfilter/vf_psnr.c | 386 - libavfilter/vf_pullup.c | 780 -- libavfilter/vf_pullup.h | 71 - libavfilter/vf_removelogo.c | 585 - libavfilter/vf_rotate.c | 567 - libavfilter/vf_sab.c | 339 - libavfilter/vf_scale.c | 400 +- libavfilter/vf_select.c | 362 + libavfilter/vf_separatefields.c | 146 - libavfilter/vf_setfield.c | 94 - libavfilter/vf_showinfo.c | 75 +- libavfilter/vf_shuffleplanes.c | 10 +- libavfilter/vf_smartblur.c | 304 - libavfilter/vf_spp.c | 437 - libavfilter/vf_spp.h | 59 - libavfilter/vf_stereo3d.c | 664 -- libavfilter/vf_subtitles.c | 436 - libavfilter/vf_super2xsai.c | 352 - libavfilter/vf_swapuv.c | 110 - libavfilter/vf_telecine.c | 285 - libavfilter/vf_thumbnail.c | 239 - libavfilter/vf_tile.c | 245 - libavfilter/vf_tinterlace.c | 398 - libavfilter/vf_transpose.c | 221 +- libavfilter/vf_unsharp.c | 222 +- libavfilter/vf_vflip.c | 24 +- libavfilter/vf_vidstabdetect.c | 217 - libavfilter/vf_vidstabtransform.c | 319 - libavfilter/vf_vignette.c | 346 - libavfilter/vf_w3fdif.c | 394 - libavfilter/vf_yadif.c | 253 +- libavfilter/video.c | 17 +- libavfilter/video.h | 10 +- libavfilter/vidstabutils.c | 85 - libavfilter/vidstabutils.h | 36 - libavfilter/vsink_nullsink.c | 8 +- libavfilter/vsrc_cellauto.c | 337 - libavfilter/vsrc_color.c | 202 + libavfilter/vsrc_life.c | 450 - libavfilter/vsrc_mandelbrot.c | 430 - libavfilter/vsrc_movie.c | 286 + libavfilter/vsrc_mptestsrc.c | 361 - libavfilter/vsrc_nullsrc.c | 136 + libavfilter/vsrc_testsrc.c | 772 +- libavfilter/x86/Makefile | 5 +- libavfilter/x86/af_volume.asm | 10 +- libavfilter/x86/af_volume_init.c | 8 +- libavfilter/x86/vf_gradfun.asm | 8 +- libavfilter/x86/vf_gradfun_init.c | 61 +- libavfilter/x86/vf_hqdn3d.asm | 8 +- libavfilter/x86/vf_hqdn3d_init.c | 10 +- libavfilter/x86/vf_pullup.asm | 178 - libavfilter/x86/vf_pullup_init.c | 41 - libavfilter/x86/vf_spp.c | 233 - libavfilter/x86/vf_yadif.asm | 38 +- libavfilter/x86/vf_yadif_init.c | 68 +- libavfilter/x86/yadif-10.asm | 282 - libavfilter/x86/yadif-16.asm | 347 - libavfilter/yadif.h | 50 +- libavformat/4xm.c | 29 +- libavformat/Makefile | 125 +- libavformat/a64.c | 9 +- libavformat/aacdec.c | 26 +- libavformat/ac3dec.c | 77 +- libavformat/act.c | 207 - libavformat/adp.c | 98 - libavformat/adtsenc.c | 38 +- libavformat/adxdec.c | 16 +- libavformat/aea.c | 8 +- libavformat/afc.c | 79 - libavformat/aiff.h | 14 +- libavformat/aiffdec.c | 97 +- libavformat/aiffenc.c | 198 +- libavformat/allformats.c | 87 +- libavformat/amr.c | 28 +- libavformat/anm.c | 10 +- libavformat/apc.c | 19 +- libavformat/ape.c | 52 +- libavformat/apetag.c | 100 +- libavformat/apetag.h | 12 +- libavformat/aqtitledec.c | 148 - libavformat/asf.c | 9 +- libavformat/asf.h | 19 +- libavformat/asfcrypt.c | 8 +- libavformat/asfcrypt.h | 8 +- libavformat/asfdec.c | 305 +- libavformat/asfenc.c | 238 +- libavformat/assdec.c | 224 +- libavformat/assenc.c | 44 +- libavformat/ast.c | 29 - libavformat/ast.h | 30 - libavformat/astdec.c | 122 - libavformat/astenc.c | 214 - libavformat/au.c | 103 +- libavformat/audiointerleave.c | 41 +- libavformat/audiointerleave.h | 8 +- libavformat/avc.c | 8 +- libavformat/avc.h | 8 +- libavformat/avformat.h | 772 +- libavformat/avformatres.rc | 55 - libavformat/avi.h | 8 +- libavformat/avidec.c | 495 +- libavformat/avienc.c | 133 +- libavformat/avio.c | 122 +- libavformat/avio.h | 94 +- libavformat/avio_internal.h | 25 +- libavformat/aviobuf.c | 204 +- libavformat/avisynth.c | 48 +- libavformat/avlanguage.c | 8 +- libavformat/avlanguage.h | 8 +- libavformat/avr.c | 99 - libavformat/avs.c | 13 +- libavformat/bethsoftvid.c | 18 +- libavformat/bfi.c | 18 +- libavformat/bink.c | 40 +- libavformat/bintext.c | 388 - libavformat/bit.c | 156 - libavformat/bluray.c | 235 - libavformat/bmv.c | 8 +- libavformat/boadec.c | 78 - libavformat/brstm.c | 297 - libavformat/c93.c | 10 +- libavformat/cache.c | 140 - libavformat/caf.c | 56 +- libavformat/caf.h | 8 +- libavformat/cafdec.c | 53 +- libavformat/cafenc.c | 286 - libavformat/cavsvideodec.c | 8 +- libavformat/cdg.c | 15 +- libavformat/cdxl.c | 53 +- libavformat/cinedec.c | 301 - libavformat/concat.c | 13 +- libavformat/concatdec.c | 631 -- libavformat/crcenc.c | 9 +- libavformat/crypto.c | 14 +- libavformat/cutils.c | 29 +- libavformat/data_uri.c | 118 - libavformat/daud.c | 12 +- libavformat/dfa.c | 21 +- libavformat/diracdec.c | 23 +- libavformat/dnxhddec.c | 10 +- libavformat/dsfdec.c | 159 - libavformat/dsicin.c | 13 +- libavformat/dtsdec.c | 15 +- libavformat/dtshddec.c | 139 - libavformat/dv.c | 131 +- libavformat/dv.h | 10 +- libavformat/dvenc.c | 64 +- libavformat/dxa.c | 36 +- libavformat/eacdata.c | 17 +- libavformat/electronicarts.c | 111 +- libavformat/epafdec.c | 104 - libavformat/ffm.h | 11 +- libavformat/ffmdec.c | 188 +- libavformat/ffmenc.c | 68 +- libavformat/ffmeta.h | 8 +- libavformat/ffmetadec.c | 16 +- libavformat/ffmetaenc.c | 8 +- libavformat/file.c | 70 +- libavformat/filmstripdec.c | 12 +- libavformat/filmstripenc.c | 10 +- libavformat/flac_picture.c | 26 +- libavformat/flac_picture.h | 11 +- libavformat/flacdec.c | 86 +- libavformat/flacenc.c | 39 +- libavformat/flacenc.h | 8 +- libavformat/flacenc_header.c | 8 +- libavformat/flic.c | 23 +- libavformat/flv.h | 32 +- libavformat/flvdec.c | 281 +- libavformat/flvenc.c | 109 +- libavformat/format.c | 36 +- libavformat/framecrcenc.c | 35 +- libavformat/framehash.c | 11 +- libavformat/frmdec.c | 110 - libavformat/ftp.c | 766 -- libavformat/g722.c | 11 +- libavformat/g723_1.c | 10 +- libavformat/g729dec.c | 103 - libavformat/gif.c | 388 +- libavformat/gifdec.c | 334 - libavformat/golomb_tab.c | 1 - libavformat/gopher.c | 8 +- libavformat/gsmdec.c | 13 +- libavformat/gxf.c | 85 +- libavformat/gxf.h | 8 +- libavformat/gxfenc.c | 182 +- libavformat/h261dec.c | 45 +- libavformat/h263dec.c | 31 +- libavformat/h264dec.c | 8 +- libavformat/hdsenc.c | 34 +- libavformat/hevc.c | 8 +- libavformat/hevc.h | 10 +- libavformat/hevcdec.c | 8 +- libavformat/hls.c | 1527 +-- libavformat/hlsenc.c | 46 +- libavformat/hlsproto.c | 10 +- libavformat/hnm.c | 12 +- libavformat/http.c | 234 +- libavformat/http.h | 10 +- libavformat/httpauth.c | 13 +- libavformat/httpauth.h | 8 +- libavformat/icodec.c | 182 - libavformat/icoenc.c | 202 - libavformat/id3v1.c | 11 +- libavformat/id3v1.h | 8 +- libavformat/id3v2.c | 355 +- libavformat/id3v2.h | 36 +- libavformat/id3v2enc.c | 158 +- libavformat/idcin.c | 58 +- libavformat/idroqdec.c | 13 +- libavformat/idroqenc.c | 38 +- libavformat/iff.c | 585 +- libavformat/ilbc.c | 8 +- libavformat/img2.c | 14 +- libavformat/img2.h | 61 - libavformat/img2_alias_pix.c | 65 - libavformat/img2_brender_pix.c | 49 - libavformat/img2dec.c | 314 +- libavformat/img2enc.c | 132 +- libavformat/ingenientdec.c | 21 +- libavformat/internal.h | 79 +- libavformat/ipmovie.c | 50 +- libavformat/ircam.c | 47 - libavformat/ircam.h | 30 - libavformat/ircamdec.c | 115 - libavformat/ircamenc.c | 62 - libavformat/isom.c | 141 +- libavformat/isom.h | 28 +- libavformat/iss.c | 28 +- libavformat/iv8.c | 8 +- libavformat/ivfdec.c | 8 +- libavformat/ivfenc.c | 8 +- libavformat/jacosubdec.c | 271 - libavformat/jacosubenc.c | 42 - libavformat/jvdec.c | 22 +- libavformat/latmenc.c | 91 +- libavformat/libavformat.v | 33 - libavformat/libgme.c | 201 - libavformat/libmodplug.c | 382 - libavformat/libnut.c | 324 - libavformat/libquvi.c | 146 - libavformat/librtmp.c | 11 +- libavformat/libssh.c | 310 - libavformat/lmlm4.c | 14 +- libavformat/loasdec.c | 94 - libavformat/lvfdec.c | 152 - libavformat/lxfdec.c | 16 +- libavformat/m4vdec.c | 10 +- libavformat/matroska.c | 52 +- libavformat/matroska.h | 43 +- libavformat/matroskadec.c | 787 +- libavformat/matroskaenc.c | 634 +- libavformat/md5enc.c | 85 +- libavformat/md5proto.c | 8 +- libavformat/metadata.c | 10 +- libavformat/metadata.h | 8 +- libavformat/mgsts.c | 106 - libavformat/microdvddec.c | 203 - libavformat/microdvdenc.c | 67 - libavformat/mkvtimestamp_v2.c | 50 - libavformat/mlvdec.c | 471 - libavformat/mm.c | 9 +- libavformat/mmf.c | 92 +- libavformat/mms.c | 8 +- libavformat/mms.h | 8 +- libavformat/mmsh.c | 77 +- libavformat/mmst.c | 29 +- libavformat/mov.c | 844 +- libavformat/mov_chan.c | 11 +- libavformat/mov_chan.h | 8 +- libavformat/movenc.c | 1014 +- libavformat/movenc.h | 26 +- libavformat/movenchint.c | 17 +- libavformat/mp3dec.c | 214 +- libavformat/mp3enc.c | 159 +- libavformat/mpc.c | 17 +- libavformat/mpc8.c | 38 +- libavformat/mpeg.c | 584 +- libavformat/mpeg.h | 11 +- libavformat/mpegenc.c | 76 +- libavformat/mpegts.c | 825 +- libavformat/mpegts.h | 18 +- libavformat/mpegtsenc.c | 398 +- libavformat/mpegvideodec.c | 55 +- libavformat/mpjpeg.c | 19 +- libavformat/mpl2dec.c | 146 - libavformat/mpsubdec.c | 144 - libavformat/msnwc_tcp.c | 14 +- libavformat/mtv.c | 69 +- libavformat/mux.c | 547 +- libavformat/mvdec.c | 10 +- libavformat/mvi.c | 12 +- libavformat/mxf.c | 10 +- libavformat/mxf.h | 20 +- libavformat/mxfdec.c | 398 +- libavformat/mxfenc.c | 458 +- libavformat/mxg.c | 24 +- libavformat/ncdec.c | 10 +- libavformat/network.c | 56 +- libavformat/network.h | 23 +- libavformat/nistspheredec.c | 131 - libavformat/noproxy-test.c | 8 +- libavformat/nsvdec.c | 65 +- libavformat/nullenc.c | 10 +- libavformat/nut.c | 193 +- libavformat/nut.h | 16 +- libavformat/nutdec.c | 264 +- libavformat/nutenc.c | 433 +- libavformat/nuv.c | 101 +- libavformat/oggdec.c | 400 +- libavformat/oggdec.h | 9 +- libavformat/oggenc.c | 149 +- libavformat/oggparsecelt.c | 27 +- libavformat/oggparsedirac.c | 8 +- libavformat/oggparseflac.c | 38 +- libavformat/oggparseogm.c | 28 +- libavformat/oggparseopus.c | 100 +- libavformat/oggparseskeleton.c | 25 +- libavformat/oggparsespeex.c | 10 +- libavformat/oggparsetheora.c | 40 - libavformat/oggparsevorbis.c | 104 +- libavformat/oggparsevp8.c | 137 - libavformat/oma.c | 8 +- libavformat/oma.h | 8 +- libavformat/omadec.c | 28 +- libavformat/omaenc.c | 13 +- libavformat/options.c | 22 +- libavformat/options_table.h | 39 +- libavformat/os_support.c | 8 +- libavformat/os_support.h | 19 +- libavformat/paf.c | 21 +- libavformat/pcm.c | 26 +- libavformat/pcm.h | 9 +- libavformat/pcmdec.c | 68 +- libavformat/pcmenc.c | 8 +- libavformat/pjsdec.c | 138 - libavformat/pmpdec.c | 92 +- libavformat/psxstr.c | 69 +- libavformat/pva.c | 44 +- libavformat/pvfdec.c | 75 - libavformat/qcp.c | 16 +- libavformat/qtpalette.h | 8 +- libavformat/r3d.c | 19 +- libavformat/rawdec.c | 105 +- libavformat/rawdec.h | 10 +- libavformat/rawenc.c | 85 +- libavformat/rawenc.h | 8 +- libavformat/rawvideodec.c | 40 +- libavformat/rdt.c | 12 +- libavformat/rdt.h | 8 +- libavformat/realtextdec.c | 153 - libavformat/redspark.c | 170 - libavformat/replaygain.c | 11 +- libavformat/replaygain.h | 8 +- libavformat/riff.c | 47 +- libavformat/riff.h | 37 +- libavformat/riffdec.c | 65 +- libavformat/riffenc.c | 89 +- libavformat/rl2.c | 27 +- libavformat/rm.c | 8 +- libavformat/rm.h | 10 +- libavformat/rmdec.c | 236 +- libavformat/rmenc.c | 17 +- libavformat/rmsipr.c | 8 +- libavformat/rmsipr.h | 8 +- libavformat/rpl.c | 33 +- libavformat/rsd.c | 168 - libavformat/rso.c | 8 +- libavformat/rso.h | 8 +- libavformat/rsodec.c | 29 +- libavformat/rsoenc.c | 8 +- libavformat/rtmp.h | 8 +- libavformat/rtmpcrypt.c | 8 +- libavformat/rtmpcrypt.h | 8 +- libavformat/rtmpdh.c | 8 +- libavformat/rtmpdh.h | 8 +- libavformat/rtmphttp.c | 8 +- libavformat/rtmppkt.c | 9 +- libavformat/rtmppkt.h | 8 +- libavformat/rtmpproto.c | 49 +- libavformat/rtp.c | 11 +- libavformat/rtp.h | 8 +- libavformat/rtpdec.c | 17 +- libavformat/rtpdec.h | 12 +- libavformat/rtpdec_amr.c | 8 +- libavformat/rtpdec_asf.c | 8 +- libavformat/rtpdec_formats.h | 8 +- libavformat/rtpdec_g726.c | 8 +- libavformat/rtpdec_h263.c | 8 +- libavformat/rtpdec_h263_rfc2190.c | 8 +- libavformat/rtpdec_h264.c | 8 +- libavformat/rtpdec_ilbc.c | 8 +- libavformat/rtpdec_jpeg.c | 8 +- libavformat/rtpdec_latm.c | 13 +- libavformat/rtpdec_mpeg12.c | 8 +- libavformat/rtpdec_mpeg4.c | 14 +- libavformat/rtpdec_mpegts.c | 8 +- libavformat/rtpdec_qcelp.c | 8 +- libavformat/rtpdec_qdm2.c | 16 +- libavformat/rtpdec_qt.c | 8 +- libavformat/rtpdec_svq3.c | 13 +- libavformat/rtpdec_vp8.c | 8 +- libavformat/rtpdec_xiph.c | 22 +- libavformat/rtpenc.c | 25 +- libavformat/rtpenc.h | 12 +- libavformat/rtpenc_aac.c | 8 +- libavformat/rtpenc_amr.c | 8 +- libavformat/rtpenc_chain.c | 12 +- libavformat/rtpenc_chain.h | 8 +- libavformat/rtpenc_h263.c | 12 +- libavformat/rtpenc_h263_rfc2190.c | 8 +- libavformat/rtpenc_h264.c | 8 +- libavformat/rtpenc_jpeg.c | 8 +- libavformat/rtpenc_latm.c | 8 +- libavformat/rtpenc_mpv.c | 8 +- libavformat/rtpenc_vp8.c | 8 +- libavformat/rtpenc_xiph.c | 11 +- libavformat/rtpproto.c | 52 +- libavformat/rtpproto.h | 8 +- libavformat/rtsp.c | 64 +- libavformat/rtsp.h | 21 +- libavformat/rtspcodes.h | 118 +- libavformat/rtspdec.c | 14 +- libavformat/rtspenc.c | 11 +- libavformat/samidec.c | 137 - libavformat/sapdec.c | 8 +- libavformat/sapenc.c | 13 +- libavformat/sauce.c | 10 +- libavformat/sauce.h | 8 +- libavformat/sbgdec.c | 1511 --- libavformat/sctp.c | 16 +- libavformat/sdp.c | 26 +- libavformat/sdr2.c | 121 - libavformat/seek-test.c | 43 +- libavformat/seek.c | 8 +- libavformat/seek.h | 8 +- libavformat/segafilm.c | 85 +- libavformat/segment.c | 762 +- libavformat/sierravmd.c | 89 +- libavformat/siff.c | 62 +- libavformat/smacker.c | 67 +- libavformat/smjpeg.c | 8 +- libavformat/smjpeg.h | 8 +- libavformat/smjpegdec.c | 14 +- libavformat/smjpegenc.c | 10 +- libavformat/smoothstreamingenc.c | 24 +- libavformat/smush.c | 18 +- libavformat/sol.c | 15 +- libavformat/sox.h | 8 +- libavformat/soxdec.c | 38 +- libavformat/soxenc.c | 23 +- libavformat/spdif.c | 8 +- libavformat/spdif.h | 11 +- libavformat/spdifdec.c | 35 +- libavformat/spdifenc.c | 30 +- libavformat/srtdec.c | 145 +- libavformat/srtenc.c | 115 - libavformat/srtp.c | 8 +- libavformat/srtp.h | 8 +- libavformat/srtpproto.c | 8 +- libavformat/subfile.c | 147 - libavformat/subtitles.c | 283 - libavformat/subtitles.h | 124 - libavformat/subviewer1dec.c | 124 - libavformat/subviewerdec.c | 194 - libavformat/swf.c | 8 +- libavformat/swf.h | 105 +- libavformat/swfdec.c | 333 +- libavformat/swfenc.c | 26 +- libavformat/takdec.c | 50 +- libavformat/tcp.c | 69 +- libavformat/tedcaptionsdec.c | 366 - libavformat/tee.c | 497 - libavformat/thp.c | 49 +- libavformat/tiertexseq.c | 8 +- libavformat/tls.c | 63 +- libavformat/tmv.c | 10 +- libavformat/tta.c | 91 +- libavformat/tty.c | 43 +- libavformat/txd.c | 33 +- libavformat/udp.c | 299 +- libavformat/uncodedframecrcenc.c | 172 - libavformat/unix.c | 11 +- libavformat/url-test.c | 8 +- libavformat/url.c | 8 +- libavformat/url.h | 18 +- libavformat/urldecode.c | 8 +- libavformat/urldecode.h | 8 +- libavformat/utils.c | 2221 +--- libavformat/vc1test.c | 15 +- libavformat/vc1testenc.c | 11 +- libavformat/version.h | 36 +- libavformat/vivo.c | 313 - libavformat/voc.c | 8 +- libavformat/voc.h | 8 +- libavformat/vocdec.c | 12 +- libavformat/vocenc.c | 22 +- libavformat/vorbiscomment.c | 9 +- libavformat/vorbiscomment.h | 8 +- libavformat/vplayerdec.c | 128 - libavformat/vqf.c | 23 +- libavformat/w64.c | 50 - libavformat/w64.h | 31 - libavformat/wavdec.c | 357 +- libavformat/wavenc.c | 206 +- libavformat/wc3movie.c | 19 +- libavformat/webvttdec.c | 223 - libavformat/webvttenc.c | 100 - libavformat/westwood_aud.c | 10 +- libavformat/westwood_vqa.c | 51 +- libavformat/{wtvdec.c => wtv.c} | 295 +- libavformat/wtv.h | 60 - libavformat/wtv_common.c | 84 - libavformat/wtvenc.c | 845 -- libavformat/wv.c | 8 +- libavformat/wv.h | 8 +- libavformat/wvdec.c | 20 +- libavformat/wvenc.c | 12 +- libavformat/xa.c | 11 +- libavformat/xmv.c | 190 +- libavformat/xwma.c | 33 +- libavformat/yop.c | 25 +- libavformat/yuv4mpeg.h | 8 +- libavformat/yuv4mpegdec.c | 83 +- libavformat/yuv4mpegenc.c | 161 +- libavresample/Makefile | 3 - libavresample/aarch64/asm-offsets.h | 8 +- libavresample/aarch64/audio_convert_init.c | 8 +- libavresample/aarch64/audio_convert_neon.S | 8 +- libavresample/aarch64/neontest.c | 8 +- libavresample/aarch64/resample_init.c | 8 +- libavresample/aarch64/resample_neon.S | 8 +- libavresample/arm/audio_convert_init.c | 8 +- libavresample/arm/audio_convert_neon.S | 8 +- libavresample/arm/neontest.c | 8 +- libavresample/audio_convert.c | 8 +- libavresample/audio_convert.h | 8 +- libavresample/audio_data.c | 8 +- libavresample/audio_data.h | 8 +- libavresample/audio_mix.c | 8 +- libavresample/audio_mix.h | 8 +- libavresample/audio_mix_matrix.c | 8 +- libavresample/avresample-test.c | 8 +- libavresample/avresample.h | 10 +- libavresample/avresampleres.rc | 55 - libavresample/dither.c | 8 +- libavresample/dither.h | 8 +- libavresample/internal.h | 8 +- libavresample/options.c | 8 +- libavresample/resample.c | 8 +- libavresample/resample.h | 8 +- libavresample/resample_template.c | 8 +- libavresample/utils.c | 14 +- libavresample/version.h | 8 +- libavresample/x86/audio_convert.asm | 42 +- libavresample/x86/audio_convert_init.c | 8 +- libavresample/x86/audio_mix.asm | 16 +- libavresample/x86/audio_mix_init.c | 8 +- libavresample/x86/dither.asm | 8 +- libavresample/x86/dither_init.c | 8 +- libavresample/x86/util.asm | 8 +- libavresample/x86/w64xmmtest.c | 8 +- libavutil/Makefile | 57 +- libavutil/aarch64/asm.S | 8 +- libavutil/aarch64/bswap.h | 8 +- libavutil/aarch64/cpu.c | 8 +- libavutil/aarch64/cpu.h | 8 +- libavutil/aarch64/float_dsp_init.c | 8 +- libavutil/aarch64/float_dsp_neon.S | 8 +- libavutil/aarch64/neontest.h | 8 +- libavutil/adler32.c | 46 +- libavutil/adler32.h | 8 +- libavutil/aes.c | 12 +- libavutil/aes.h | 12 +- libavutil/arm/asm.S | 8 +- libavutil/arm/bswap.h | 8 +- libavutil/arm/cpu.c | 8 +- libavutil/arm/cpu.h | 8 +- libavutil/arm/float_dsp_arm.h | 8 +- libavutil/arm/float_dsp_init_arm.c | 8 +- libavutil/arm/float_dsp_init_neon.c | 8 +- libavutil/arm/float_dsp_init_vfp.c | 8 +- libavutil/arm/float_dsp_neon.S | 8 +- libavutil/arm/float_dsp_vfp.S | 8 +- libavutil/arm/intmath.h | 8 +- libavutil/arm/intreadwrite.h | 8 +- libavutil/arm/neontest.h | 8 +- libavutil/arm/timer.h | 8 +- libavutil/atomic.c | 14 +- libavutil/atomic.h | 8 +- libavutil/atomic_gcc.h | 8 +- libavutil/atomic_suncc.h | 8 +- libavutil/atomic_win32.h | 8 +- libavutil/attributes.h | 44 +- libavutil/audio_fifo.c | 10 +- libavutil/audio_fifo.h | 15 +- libavutil/avassert.h | 10 +- libavutil/avr32/bswap.h | 8 +- libavutil/avr32/intreadwrite.h | 8 +- libavutil/avstring.c | 155 +- libavutil/avstring.h | 140 +- libavutil/avutil.h | 84 +- libavutil/avutilres.rc | 55 - libavutil/base64.c | 172 +- libavutil/base64.h | 20 +- libavutil/bfin/asm.h | 8 +- libavutil/bfin/attributes.h | 8 +- libavutil/bfin/bswap.h | 8 +- libavutil/bfin/timer.h | 8 +- libavutil/blowfish.c | 52 +- libavutil/blowfish.h | 9 +- libavutil/bprint.c | 381 - libavutil/bprint.h | 216 - libavutil/bswap.h | 8 +- libavutil/buffer.c | 41 +- libavutil/buffer.h | 19 +- libavutil/buffer_internal.h | 10 +- libavutil/channel_layout.c | 198 +- libavutil/channel_layout.h | 51 +- libavutil/colorspace.h | 8 +- libavutil/common.h | 79 +- libavutil/cpu.c | 120 +- libavutil/cpu.h | 42 +- libavutil/cpu_internal.h | 8 +- libavutil/crc.c | 60 +- libavutil/crc.h | 19 +- libavutil/des.c | 18 +- libavutil/des.h | 8 +- libavutil/dict.c | 25 +- libavutil/dict.h | 45 +- libavutil/display.c | 14 +- libavutil/display.h | 8 +- libavutil/downmix_info.c | 8 +- libavutil/downmix_info.h | 8 +- libavutil/dynarray.h | 70 - libavutil/error.c | 93 +- libavutil/error.h | 73 +- libavutil/eval.c | 226 +- libavutil/eval.h | 12 +- libavutil/fifo.c | 92 +- libavutil/fifo.h | 39 +- libavutil/file.c | 74 +- libavutil/file.h | 20 +- libavutil/file_open.c | 52 +- libavutil/fixed_dsp.c | 98 - libavutil/fixed_dsp.h | 144 - libavutil/float_dsp.c | 15 +- libavutil/float_dsp.h | 11 +- libavutil/frame.c | 166 +- libavutil/frame.h | 180 +- libavutil/hash.c | 239 - libavutil/hash.h | 112 - libavutil/hmac.c | 128 +- libavutil/hmac.h | 12 +- libavutil/imgutils.c | 145 +- libavutil/imgutils.h | 70 +- libavutil/integer.c | 196 - libavutil/integer.h | 86 - libavutil/internal.h | 55 +- libavutil/intfloat.h | 8 +- libavutil/intfloat_readwrite.c | 101 - libavutil/intfloat_readwrite.h | 44 - libavutil/intmath.c | 8 +- libavutil/intmath.h | 12 +- libavutil/intreadwrite.h | 94 +- libavutil/lfg.c | 12 +- libavutil/lfg.h | 8 +- libavutil/libavutil.v | 2 +- libavutil/libm.h | 19 +- libavutil/{lls1.c => lls.c} | 49 +- libavutil/{lls2.h => lls.h} | 31 +- libavutil/lls1.h | 54 - libavutil/lls2.c | 160 - libavutil/log.c | 320 +- libavutil/log.h | 80 +- libavutil/log2_tab.c | 10 +- libavutil/lzo.c | 60 +- libavutil/lzo.h | 8 +- libavutil/macros.h | 8 +- libavutil/mathematics.c | 62 +- libavutil/mathematics.h | 63 +- libavutil/md5.c | 103 +- libavutil/md5.h | 46 +- libavutil/mem.c | 175 +- libavutil/mem.h | 125 +- libavutil/mips/Makefile | 1 - libavutil/mips/float_dsp_mips.c | 383 - libavutil/mips/intreadwrite.h | 8 +- libavutil/mips/libm_mips.h | 73 - libavutil/murmur3.c | 192 - libavutil/murmur3.h | 32 - libavutil/old_pix_fmts.h | 55 +- libavutil/opencl.c | 837 -- libavutil/opencl.h | 330 - libavutil/opencl_internal.c | 59 - libavutil/opencl_internal.h | 33 - libavutil/opt.c | 1068 +- libavutil/opt.h | 340 +- libavutil/parseutils.c | 292 +- libavutil/parseutils.h | 77 +- libavutil/pca.c | 248 - libavutil/pca.h | 35 - libavutil/pixdesc.c | 845 +- libavutil/pixdesc.h | 87 +- libavutil/pixfmt.h | 160 +- libavutil/ppc/cpu.c | 10 +- libavutil/ppc/cpu.h | 8 +- libavutil/ppc/float_dsp_altivec.c | 8 +- libavutil/ppc/float_dsp_altivec.h | 8 +- libavutil/ppc/float_dsp_init.c | 8 +- libavutil/ppc/intreadwrite.h | 8 +- libavutil/ppc/timer.h | 15 +- libavutil/ppc/types_altivec.h | 8 +- libavutil/ppc/util_altivec.h | 8 +- libavutil/qsort.h | 117 - libavutil/random_seed.c | 73 +- libavutil/random_seed.h | 19 +- libavutil/rational.c | 73 +- libavutil/rational.h | 21 +- libavutil/rc4.c | 8 +- libavutil/rc4.h | 8 +- 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tests/ref/fate/gif-gray | 37 - tests/ref/fate/gifenc-bgr4_byte | 174 - tests/ref/fate/gifenc-bgr8 | 174 - tests/ref/fate/gifenc-gray | 174 - tests/ref/fate/gifenc-pal8 | 174 - tests/ref/fate/gifenc-rgb4_byte | 174 - tests/ref/fate/gifenc-rgb8 | 174 - .../h264-conformance-frext-hi422fr10_sony_b | 6 - .../h264-conformance-frext-hi422fr13_sony_b | 6 - .../h264-conformance-frext-hi422fr1_sony_a | 6 - .../h264-conformance-frext-hi422fr6_sony_a | 6 - ...264-conformance-frext-pph422i1_panasonic_a | 11 - ...264-conformance-frext-pph422i2_panasonic_a | 11 - ...264-conformance-frext-pph422i3_panasonic_a | 11 - ...264-conformance-frext-pph422i4_panasonic_a | 11 - ...264-conformance-frext-pph422i5_panasonic_a | 11 - ...264-conformance-frext-pph422i6_panasonic_a | 11 - ...264-conformance-frext-pph422i7_panasonic_a | 11 - tests/ref/fate/h264-crop-to-container | 8 +- tests/ref/fate/h264-extreme-plane-pred | 4 - tests/ref/fate/h264-lossless | 20 +- ... => hevc-conformance-DBLK_A_MAIN10_VIXS_2} | 0 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6 +- tests/ref/fate/prores-422_hq | 6 +- tests/ref/fate/prores-422_lt | 6 +- tests/ref/fate/prores-422_proxy | 6 +- tests/ref/fate/prores-alpha | 6 +- tests/ref/fate/prores-alpha_skip | 3 - tests/ref/fate/prores-transparency | 5 - tests/ref/fate/prores-transparency_skip | 5 - tests/ref/fate/ptx | 2 +- tests/ref/fate/pva-demux | 53 +- tests/ref/fate/qtrle-16bit | 168 +- tests/ref/fate/qtrle-1bit | 78 +- tests/ref/fate/qtrle-24bit | 68 +- tests/ref/fate/qtrle-2bit | 78 +- tests/ref/fate/qtrle-32bit | 54 +- tests/ref/fate/qtrle-4bit | 78 +- tests/ref/fate/qtrle-8bit | 334 +- tests/ref/fate/quickdraw | 4 +- tests/ref/fate/random_seed | 1 - tests/ref/fate/redcode-demux | 2 +- tests/ref/fate/redspark-demux | 1 - tests/ref/fate/ripemd | 28 - tests/ref/fate/rpza | 62 +- tests/ref/fate/rsd-demux | 1 - tests/ref/fate/rv30 | 220 +- tests/ref/fate/rv40 | 481 +- tests/ref/fate/sanm | 10 +- tests/ref/fate/sgi-gray | 2 +- tests/ref/fate/sgi-gray16 | 2 +- tests/ref/fate/sgi-rgb24 | 2 +- tests/ref/fate/sgi-rgb48 | 2 +- tests/ref/fate/sgi-rgba | 2 +- tests/ref/fate/sgi-rgba64 | 2 +- tests/ref/fate/sha512 | 28 - tests/ref/fate/sierra-vmd-video | 234 +- tests/ref/fate/smc | 240 +- tests/ref/fate/smjpeg | 425 - tests/ref/fate/smjpeg-demux | 2 +- tests/ref/fate/smvjpeg | 13 - tests/ref/fate/sub-aqtitle | 1 - tests/ref/fate/sub-charenc | 1 - tests/ref/fate/sub-jacosub | 1 - tests/ref/fate/sub-microdvd | 1 - tests/ref/fate/sub-microdvd-remux | 1 - tests/ref/fate/sub-movtext | 1 - tests/ref/fate/sub-movtextenc | 1 - tests/ref/fate/sub-mpl2 | 1 - tests/ref/fate/sub-mpsub | 1 - tests/ref/fate/sub-mpsub-frames | 1 - tests/ref/fate/sub-pjs | 1 - tests/ref/fate/sub-realtext | 1 - tests/ref/fate/sub-sami | 1 - tests/ref/fate/sub-srt | 2 +- tests/ref/fate/sub-subripenc | 1 - tests/ref/fate/sub-subviewer | 1 - tests/ref/fate/sub-subviewer1 | 1 - tests/ref/fate/sub-vplayer | 1 - tests/ref/fate/sub-webvtt | 1 - tests/ref/fate/sub-webvttenc | 1 - tests/ref/fate/sub2video | 95 - tests/ref/fate/svq1 | 300 +- tests/ref/fate/svq3 | 360 +- tests/ref/fate/theora-coeff-level64 | 18 +- tests/ref/fate/timefilter | 12 +- tests/ref/fate/truemotion1-15 | 210 +- tests/ref/fate/truemotion1-24 | 1 - tests/ref/fate/tscc-15bit | 450 +- tests/ref/fate/unknown_layout-ac3 | 1 - tests/ref/fate/unknown_layout-pcm | 1 - tests/ref/fate/utvideoenc_rgb_left | 4 - tests/ref/fate/utvideoenc_rgb_median | 4 - tests/ref/fate/utvideoenc_rgb_none | 4 - tests/ref/fate/utvideoenc_rgba_left | 4 - tests/ref/fate/utvideoenc_rgba_median | 4 - tests/ref/fate/utvideoenc_rgba_none | 4 - tests/ref/fate/utvideoenc_yuv420_left | 4 - tests/ref/fate/utvideoenc_yuv420_median | 4 - tests/ref/fate/utvideoenc_yuv420_none | 4 - tests/ref/fate/utvideoenc_yuv422_left | 4 - tests/ref/fate/utvideoenc_yuv422_median | 4 - tests/ref/fate/utvideoenc_yuv422_none | 4 - tests/ref/fate/v210 | 2 +- tests/ref/fate/v410enc | 2 +- tests/ref/fate/vc1-ism | 242 +- tests/ref/fate/vc1_sa10143 | 28 +- tests/ref/fate/vcr2 | 158 - tests/ref/fate/vmnc-16bit | 384 +- tests/ref/fate/vp5 | 1 - tests/ref/fate/vp60 | 192 +- tests/ref/fate/vp6a | 188 +- tests/ref/fate/vp6a-skip_alpha | 94 - tests/ref/fate/vp6f | 350 +- tests/ref/fate/vp8-alpha | 121 - tests/ref/fate/vp8-sign-bias | 7 +- tests/ref/fate/vp8-size-change | 66 +- tests/ref/fate/vp8-test-vector-001 | 4 - tests/ref/fate/vp8-test-vector-002 | 4 - tests/ref/fate/vp8-test-vector-003 | 4 - tests/ref/fate/vp8-test-vector-004 | 4 - tests/ref/fate/vp8-test-vector-005 | 4 - tests/ref/fate/vp8-test-vector-006 | 4 - tests/ref/fate/vp8-test-vector-007 | 4 - tests/ref/fate/vp8-test-vector-008 | 4 - tests/ref/fate/vp8-test-vector-009 | 4 - tests/ref/fate/vp8-test-vector-010 | 4 - tests/ref/fate/vp8-test-vector-011 | 4 - tests/ref/fate/vp8-test-vector-012 | 4 - tests/ref/fate/vp8-test-vector-013 | 4 - tests/ref/fate/vp8-test-vector-014 | 4 - tests/ref/fate/vp8-test-vector-015 | 4 - tests/ref/fate/vp8-test-vector-016 | 4 - tests/ref/fate/vp8-test-vector-017 | 4 - tests/ref/fate/vp9-00-quantizer-00 | 10 +- tests/ref/fate/vp9-00-quantizer-01 | 10 +- tests/ref/fate/vp9-00-quantizer-02 | 10 +- tests/ref/fate/vp9-00-quantizer-03 | 10 +- tests/ref/fate/vp9-00-quantizer-04 | 10 +- tests/ref/fate/vp9-00-quantizer-05 | 10 +- tests/ref/fate/vp9-00-quantizer-06 | 10 +- tests/ref/fate/vp9-00-quantizer-07 | 10 +- tests/ref/fate/vp9-00-quantizer-08 | 10 +- tests/ref/fate/vp9-00-quantizer-09 | 10 +- tests/ref/fate/vp9-00-quantizer-10 | 10 +- tests/ref/fate/vp9-00-quantizer-11 | 10 +- tests/ref/fate/vp9-00-quantizer-12 | 10 +- tests/ref/fate/vp9-00-quantizer-13 | 10 +- tests/ref/fate/vp9-00-quantizer-14 | 10 +- tests/ref/fate/vp9-00-quantizer-15 | 10 +- tests/ref/fate/vp9-00-quantizer-16 | 10 +- tests/ref/fate/vp9-00-quantizer-17 | 10 +- tests/ref/fate/vp9-00-quantizer-18 | 10 +- tests/ref/fate/vp9-00-quantizer-19 | 10 +- tests/ref/fate/vp9-00-quantizer-20 | 10 +- tests/ref/fate/vp9-00-quantizer-21 | 10 +- tests/ref/fate/vp9-00-quantizer-22 | 10 +- tests/ref/fate/vp9-00-quantizer-23 | 10 +- tests/ref/fate/vp9-00-quantizer-24 | 10 +- tests/ref/fate/vp9-00-quantizer-25 | 10 +- tests/ref/fate/vp9-00-quantizer-26 | 10 +- tests/ref/fate/vp9-00-quantizer-27 | 10 +- tests/ref/fate/vp9-00-quantizer-28 | 10 +- tests/ref/fate/vp9-00-quantizer-29 | 10 +- tests/ref/fate/vp9-00-quantizer-30 | 10 +- tests/ref/fate/vp9-00-quantizer-31 | 10 +- tests/ref/fate/vp9-00-quantizer-32 | 10 +- tests/ref/fate/vp9-00-quantizer-33 | 10 +- tests/ref/fate/vp9-00-quantizer-34 | 10 +- tests/ref/fate/vp9-00-quantizer-35 | 10 +- tests/ref/fate/vp9-00-quantizer-36 | 10 +- tests/ref/fate/vp9-00-quantizer-37 | 10 +- tests/ref/fate/vp9-00-quantizer-38 | 10 +- tests/ref/fate/vp9-00-quantizer-39 | 10 +- tests/ref/fate/vp9-00-quantizer-40 | 10 +- tests/ref/fate/vp9-00-quantizer-41 | 10 +- tests/ref/fate/vp9-00-quantizer-42 | 10 +- tests/ref/fate/vp9-00-quantizer-43 | 10 +- tests/ref/fate/vp9-00-quantizer-44 | 10 +- tests/ref/fate/vp9-00-quantizer-45 | 10 +- 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tests/ref/fate/vp9-01-sharpness-6 | 26 +- tests/ref/fate/vp9-01-sharpness-7 | 26 +- tests/ref/fate/vp9-02-size-08x08 | 26 +- tests/ref/fate/vp9-02-size-08x10 | 26 +- tests/ref/fate/vp9-02-size-08x16 | 26 +- tests/ref/fate/vp9-02-size-08x18 | 26 +- tests/ref/fate/vp9-02-size-08x32 | 26 +- tests/ref/fate/vp9-02-size-08x34 | 26 +- tests/ref/fate/vp9-02-size-08x64 | 26 +- tests/ref/fate/vp9-02-size-08x66 | 26 +- tests/ref/fate/vp9-02-size-10x08 | 26 +- tests/ref/fate/vp9-02-size-10x10 | 26 +- tests/ref/fate/vp9-02-size-10x16 | 26 +- tests/ref/fate/vp9-02-size-10x18 | 26 +- tests/ref/fate/vp9-02-size-10x32 | 26 +- tests/ref/fate/vp9-02-size-10x34 | 26 +- tests/ref/fate/vp9-02-size-10x64 | 26 +- tests/ref/fate/vp9-02-size-10x66 | 26 +- tests/ref/fate/vp9-02-size-16x08 | 26 +- tests/ref/fate/vp9-02-size-16x10 | 26 +- tests/ref/fate/vp9-02-size-16x16 | 26 +- tests/ref/fate/vp9-02-size-16x18 | 26 +- tests/ref/fate/vp9-02-size-16x32 | 26 +- tests/ref/fate/vp9-02-size-16x34 | 26 +- tests/ref/fate/vp9-02-size-16x64 | 26 +- tests/ref/fate/vp9-02-size-16x66 | 26 +- tests/ref/fate/vp9-02-size-18x08 | 26 +- tests/ref/fate/vp9-02-size-18x10 | 26 +- tests/ref/fate/vp9-02-size-18x16 | 26 +- tests/ref/fate/vp9-02-size-18x18 | 26 +- tests/ref/fate/vp9-02-size-18x32 | 26 +- tests/ref/fate/vp9-02-size-18x34 | 26 +- tests/ref/fate/vp9-02-size-18x64 | 26 +- tests/ref/fate/vp9-02-size-18x66 | 26 +- tests/ref/fate/vp9-02-size-32x08 | 26 +- tests/ref/fate/vp9-02-size-32x10 | 26 +- tests/ref/fate/vp9-02-size-32x16 | 26 +- tests/ref/fate/vp9-02-size-32x18 | 26 +- tests/ref/fate/vp9-02-size-32x32 | 26 +- tests/ref/fate/vp9-02-size-32x34 | 26 +- tests/ref/fate/vp9-02-size-32x64 | 26 +- tests/ref/fate/vp9-02-size-32x66 | 26 +- tests/ref/fate/vp9-02-size-34x08 | 26 +- tests/ref/fate/vp9-02-size-34x10 | 26 +- tests/ref/fate/vp9-02-size-34x16 | 26 +- tests/ref/fate/vp9-02-size-34x18 | 26 +- tests/ref/fate/vp9-02-size-34x32 | 26 +- tests/ref/fate/vp9-02-size-34x34 | 26 +- tests/ref/fate/vp9-02-size-34x64 | 26 +- tests/ref/fate/vp9-02-size-34x66 | 26 +- tests/ref/fate/vp9-02-size-64x08 | 26 +- tests/ref/fate/vp9-02-size-64x10 | 26 +- tests/ref/fate/vp9-02-size-64x16 | 26 +- tests/ref/fate/vp9-02-size-64x18 | 26 +- tests/ref/fate/vp9-02-size-64x32 | 26 +- tests/ref/fate/vp9-02-size-64x34 | 26 +- tests/ref/fate/vp9-02-size-64x64 | 26 +- tests/ref/fate/vp9-02-size-64x66 | 26 +- tests/ref/fate/vp9-02-size-66x08 | 26 +- tests/ref/fate/vp9-02-size-66x10 | 26 +- tests/ref/fate/vp9-02-size-66x16 | 26 +- tests/ref/fate/vp9-02-size-66x18 | 26 +- tests/ref/fate/vp9-02-size-66x32 | 26 +- tests/ref/fate/vp9-02-size-66x34 | 26 +- tests/ref/fate/vp9-02-size-66x64 | 26 +- tests/ref/fate/vp9-02-size-66x66 | 26 +- tests/ref/fate/vp9-03-deltaq | 10 +- tests/ref/fate/vp9-03-size-196x196 | 26 +- tests/ref/fate/vp9-03-size-196x198 | 26 +- tests/ref/fate/vp9-03-size-196x200 | 26 +- tests/ref/fate/vp9-03-size-196x202 | 26 +- tests/ref/fate/vp9-03-size-196x208 | 26 +- tests/ref/fate/vp9-03-size-196x210 | 26 +- tests/ref/fate/vp9-03-size-196x224 | 26 +- tests/ref/fate/vp9-03-size-196x226 | 26 +- tests/ref/fate/vp9-03-size-198x196 | 26 +- tests/ref/fate/vp9-03-size-198x198 | 26 +- tests/ref/fate/vp9-03-size-198x200 | 26 +- tests/ref/fate/vp9-03-size-198x202 | 26 +- tests/ref/fate/vp9-03-size-198x208 | 26 +- tests/ref/fate/vp9-03-size-198x210 | 26 +- tests/ref/fate/vp9-03-size-198x224 | 26 +- tests/ref/fate/vp9-03-size-198x226 | 26 +- tests/ref/fate/vp9-03-size-200x196 | 26 +- tests/ref/fate/vp9-03-size-200x198 | 26 +- tests/ref/fate/vp9-03-size-200x200 | 26 +- tests/ref/fate/vp9-03-size-200x202 | 26 +- tests/ref/fate/vp9-03-size-200x208 | 26 +- tests/ref/fate/vp9-03-size-200x210 | 26 +- tests/ref/fate/vp9-03-size-200x224 | 26 +- tests/ref/fate/vp9-03-size-200x226 | 26 +- tests/ref/fate/vp9-03-size-202x196 | 26 +- tests/ref/fate/vp9-03-size-202x198 | 26 +- tests/ref/fate/vp9-03-size-202x200 | 26 +- tests/ref/fate/vp9-03-size-202x202 | 26 +- tests/ref/fate/vp9-03-size-202x208 | 26 +- tests/ref/fate/vp9-03-size-202x210 | 26 +- tests/ref/fate/vp9-03-size-202x224 | 26 +- tests/ref/fate/vp9-03-size-202x226 | 26 +- tests/ref/fate/vp9-03-size-208x196 | 26 +- tests/ref/fate/vp9-03-size-208x198 | 26 +- tests/ref/fate/vp9-03-size-208x200 | 26 +- tests/ref/fate/vp9-03-size-208x202 | 26 +- tests/ref/fate/vp9-03-size-208x208 | 26 +- tests/ref/fate/vp9-03-size-208x210 | 26 +- tests/ref/fate/vp9-03-size-208x224 | 26 +- tests/ref/fate/vp9-03-size-208x226 | 26 +- tests/ref/fate/vp9-03-size-210x196 | 26 +- tests/ref/fate/vp9-03-size-210x198 | 26 +- tests/ref/fate/vp9-03-size-210x200 | 26 +- tests/ref/fate/vp9-03-size-210x202 | 26 +- tests/ref/fate/vp9-03-size-210x208 | 26 +- tests/ref/fate/vp9-03-size-210x210 | 26 +- tests/ref/fate/vp9-03-size-210x224 | 26 +- tests/ref/fate/vp9-03-size-210x226 | 26 +- tests/ref/fate/vp9-03-size-224x196 | 26 +- tests/ref/fate/vp9-03-size-224x198 | 26 +- tests/ref/fate/vp9-03-size-224x200 | 26 +- tests/ref/fate/vp9-03-size-224x202 | 26 +- tests/ref/fate/vp9-03-size-224x208 | 26 +- tests/ref/fate/vp9-03-size-224x210 | 26 +- tests/ref/fate/vp9-03-size-224x224 | 26 +- tests/ref/fate/vp9-03-size-224x226 | 26 +- tests/ref/fate/vp9-03-size-226x196 | 26 +- tests/ref/fate/vp9-03-size-226x198 | 26 +- tests/ref/fate/vp9-03-size-226x200 | 26 +- tests/ref/fate/vp9-03-size-226x202 | 26 +- tests/ref/fate/vp9-03-size-226x208 | 26 +- tests/ref/fate/vp9-03-size-226x210 | 26 +- tests/ref/fate/vp9-03-size-226x224 | 26 +- tests/ref/fate/vp9-03-size-226x226 | 26 +- tests/ref/fate/vp9-2pass-akiyo | 106 +- tests/ref/fate/vp9-parallelmode-akiyo | 56 +- tests/ref/fate/vp9-segmentation-aq-akiyo | 56 +- tests/ref/fate/vp9-segmentation-sf-akiyo | 56 +- tests/ref/fate/vp9-tiling-pedestrian | 4 - tests/ref/fate/vqa-cc | 68 +- tests/ref/fate/wmv8-drm | 260 +- tests/ref/fate/wmv8-drm-nodec | 256 +- tests/ref/fate/wmv8-x8intra | 474 - tests/ref/fate/wtv-demux | 86 +- tests/ref/fate/xbm11 | 2 +- tests/ref/fate/xface | 2 - tests/ref/fate/xmv-demux | 176 +- tests/ref/fate/xvid-custom-matrix | 44 +- tests/ref/fate/xvid-idctmmx | 44 +- tests/ref/fate/yop | 12 +- tests/ref/fate/zmbv-15bit | 318 +- tests/ref/fate/zmbv-16bit | 238 +- tests/ref/fate/zmbv-8bit | 1 - tests/ref/lavf-fate/latm | 3 - tests/ref/lavf-fate/mp3 | 3 - tests/ref/lavf-fate/ogg_vp3 | 3 - tests/ref/lavf/aiff | 4 +- tests/ref/lavf/asf | 4 +- tests/ref/lavf/ast | 3 - tests/ref/lavf/au | 4 +- tests/ref/lavf/avi | 6 +- tests/ref/lavf/bmp | 2 +- tests/ref/lavf/caf | 3 - tests/ref/lavf/dpx | 19 +- tests/ref/lavf/dv_fmt | 8 +- tests/ref/lavf/ffm | 4 +- tests/ref/lavf/flm | 3 - tests/ref/lavf/flv_fmt | 6 +- tests/ref/lavf/gif | 6 +- tests/ref/lavf/gxf | 12 +- tests/ref/lavf/ircam | 3 - tests/ref/lavf/ismv | 9 - tests/ref/lavf/jpg | 2 +- tests/ref/lavf/mkv | 9 +- tests/ref/lavf/mmf | 4 +- tests/ref/lavf/mov | 15 +- tests/ref/lavf/mpg | 10 +- tests/ref/lavf/mxf | 12 +- tests/ref/lavf/mxf_d10 | 4 +- tests/ref/lavf/nut | 6 +- tests/ref/lavf/ogg | 4 +- tests/ref/lavf/pam | 17 +- tests/ref/lavf/pcx | 4 +- tests/ref/lavf/pgm | 4 +- tests/ref/lavf/pgmpipe | 4 +- tests/ref/lavf/pixfmt | 16 +- tests/ref/lavf/png | 12 +- tests/ref/lavf/ppm | 2 +- tests/ref/lavf/rm | 4 +- tests/ref/lavf/sgi | 2 +- tests/ref/lavf/smjpeg | 3 - tests/ref/lavf/sunrast | 2 +- tests/ref/lavf/swf | 6 +- tests/ref/lavf/tga | 2 +- tests/ref/lavf/tiff | 2 +- tests/ref/lavf/ts | 6 +- tests/ref/lavf/voc | 2 +- tests/ref/lavf/voc_s16 | 4 +- tests/ref/lavf/w64 | 3 - tests/ref/lavf/wav | 4 +- tests/ref/lavf/wtv | 3 - tests/ref/lavf/xbm | 3 - tests/ref/lavf/xwd | 23 +- tests/ref/seek/acodec-flac | 58 +- tests/ref/seek/acodec-mp2 | 8 +- tests/ref/seek/acodec-pcm-f32be | 54 +- tests/ref/seek/acodec-pcm-f64be | 54 +- tests/ref/seek/acodec-pcm-s16le | 54 +- tests/ref/seek/acodec-pcm-u8 | 54 +- tests/ref/seek/lavf-aiff | 30 +- tests/ref/seek/lavf-asf | 54 +- tests/ref/seek/lavf-au | 30 +- tests/ref/seek/lavf-avi | 36 +- tests/ref/seek/lavf-bmp | 9 +- 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tests/ref/seek/vsynth2-msmpeg4v2 | 8 +- tests/ref/seek/vsynth2-rv10 | 17 +- tests/ref/seek/vsynth2-rv20 | 14 +- tests/ref/seek/vsynth2-snow | 46 - tests/ref/seek/vsynth2-snow-ll | 46 - tests/ref/seek/vsynth2-svq1 | 24 +- tests/ref/seek/vsynth2-wmv1 | 8 +- tests/ref/seek/vsynth2-wmv2 | 16 +- tests/ref/vsynth/vsynth1-amv | 4 - tests/ref/vsynth/vsynth1-asv1 | 2 +- tests/ref/vsynth/vsynth1-asv2 | 2 +- tests/ref/vsynth/vsynth1-avui | 4 - tests/ref/vsynth/vsynth1-cljr | 6 +- tests/ref/vsynth/vsynth1-dnxhd-1080i | 6 +- tests/ref/vsynth/vsynth1-dnxhd-720p | 6 +- tests/ref/vsynth/vsynth1-dnxhd-720p-10bit | 4 +- tests/ref/vsynth/vsynth1-dnxhd-720p-rd | 4 +- tests/ref/vsynth/vsynth1-dnxhd_1080i | 4 - tests/ref/vsynth/vsynth1-dv | 2 +- tests/ref/vsynth/vsynth1-dv-411 | 2 +- tests/ref/vsynth/vsynth1-dv-50 | 2 +- tests/ref/vsynth/vsynth1-dv_411 | 4 - tests/ref/vsynth/vsynth1-ffv1 | 4 +- tests/ref/vsynth/vsynth1-ffv1.0 | 4 - tests/ref/vsynth/vsynth1-ffvhuff | 2 +- 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tests/ref/vsynth/vsynth1-mpeg2-idct-int | 2 +- tests/ref/vsynth/vsynth1-mpeg2-ilace | 6 +- tests/ref/vsynth/vsynth1-mpeg2-ivlc-qprd | 8 +- tests/ref/vsynth/vsynth1-mpeg2-thread | 6 +- tests/ref/vsynth/vsynth1-mpeg2-thread-ivlc | 6 +- tests/ref/vsynth/vsynth1-mpeg4 | 6 +- tests/ref/vsynth/vsynth1-mpeg4-adap | 6 +- tests/ref/vsynth/vsynth1-mpeg4-adv | 4 +- tests/ref/vsynth/vsynth1-mpeg4-error | 8 +- tests/ref/vsynth/vsynth1-mpeg4-nr | 8 +- tests/ref/vsynth/vsynth1-mpeg4-qpel | 8 +- tests/ref/vsynth/vsynth1-mpeg4-qprd | 6 +- tests/ref/vsynth/vsynth1-mpeg4-rc | 6 +- tests/ref/vsynth/vsynth1-mpeg4-thread | 6 +- tests/ref/vsynth/vsynth1-mpng | 4 - tests/ref/vsynth/vsynth1-msmpeg4 | 6 +- tests/ref/vsynth/vsynth1-msmpeg4v2 | 6 +- tests/ref/vsynth/vsynth1-msvideo1 | 4 - tests/ref/vsynth/vsynth1-prores | 8 +- tests/ref/vsynth/vsynth1-prores_ks | 4 - tests/ref/vsynth/vsynth1-qtrle | 8 +- tests/ref/vsynth/vsynth1-qtrlegray | 4 - tests/ref/vsynth/vsynth1-r210 | 4 - tests/ref/vsynth/vsynth1-rgb | 6 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| 4 - tests/regression-funcs.sh | 26 +- tests/rotozoom.c | 8 +- tests/test.ffmeta | 9 - tests/tiny_psnr.c | 171 +- tests/tiny_ssim.c | 246 - tests/utils.c | 10 +- tests/videogen.c | 8 +- tools/aviocat.c | 12 +- tools/bisect-create | 46 - tools/bookmarklets.html | 55 - tools/build_libstagefright | 58 - tools/clean-diff | 11 - tools/crypto_bench.c | 358 - tools/cws2fws.c | 14 +- tools/dvd2concat | 127 - tools/enum_options.c | 143 - tools/ffescape.c | 180 - tools/ffeval.c | 139 - tools/ffhash.c | 151 - tools/fourcc2pixfmt.c | 123 - tools/gen-rc | 121 - tools/graph2dot.c | 33 +- tools/ismindex.c | 24 +- tools/make_chlayout_test | 114 - tools/missing_codec_desc | 36 - tools/normalize.py | 33 - tools/patcheck | 12 +- tools/pktdumper.c | 8 +- tools/plotframes | 164 - tools/probetest.c | 70 +- tools/qt-faststart.c | 18 +- tools/seek_print.c | 105 - tools/trasher.c | 13 +- tools/uncoded_frame.c | 279 - tools/unwrap-diff | 2 - tools/yuvcmp.c | 182 - tools/zmqsend.c | 167 - tools/zmqshell.py | 26 - version.sh | 42 +- 3692 files changed, 72748 insertions(+), 326051 deletions(-) delete mode 100644 MAINTAINERS create mode 100644 avconv.c rename ffmpeg.h => avconv.h (71%) rename ffmpeg_dxva2.c => avconv_dxva2.c (98%) rename ffmpeg_filter.c => avconv_filter.c (57%) rename ffmpeg_opt.c => avconv_opt.c (69%) rename ffmpeg_vda.c => avconv_vda.c (93%) rename ffmpeg_vdpau.c => avconv_vdpau.c (97%) create mode 100644 avplay.c create mode 100644 avprobe.c rename ffserver.c => avserver.c (88%) delete mode 100644 cmdutils_opencl.c delete mode 100644 compat/avisynth/avisynth_c.h delete mode 100644 compat/avisynth/avisynth_c_25.h delete mode 100644 compat/avisynth/avxsynth_c.h delete mode 100644 compat/avisynth/windowsPorts/basicDataTypeConversions.h delete mode 100644 compat/avisynth/windowsPorts/windows2linux.h delete mode 100644 compat/msvcrt/snprintf.h delete mode 100644 compat/os2threads.h delete mode 100644 compat/va_copy.h delete mode 100644 doc/authors.texi rename doc/{ffmpeg.texi => 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-/doc/examples/resampling_audio -/doc/examples/scaling_video +/doc/examples/output /doc/examples/transcode_aac -/doc/fate.txt /doc/print_options /lcov/ /libavcodec/*_tablegen /libavcodec/*_tables.c /libavcodec/*_tables.h /libavutil/avconfig.h -/libavutil/ffversion.h /tests/audiogen /tests/base64 /tests/data/ /tests/rotozoom /tests/tiny_psnr -/tests/tiny_ssim /tests/videogen /tests/vsynth1/ /tools/aviocat -/tools/ffbisect -/tools/bisect.need -/tools/crypto_bench /tools/cws2fws -/tools/fourcc2pixfmt -/tools/ffescape -/tools/ffeval -/tools/ffhash /tools/graph2dot /tools/ismindex /tools/pktdumper /tools/probetest /tools/qt-faststart /tools/trasher -/tools/seek_print -/tools/uncoded_frame -/tools/zmqsend diff --git a/CREDITS b/CREDITS index e29f0b853c..4a537786f0 100644 --- a/CREDITS +++ b/CREDITS @@ -1,6 +1,55 @@ -See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to -get the names of people who have contributed to FFmpeg. +This file contains the names of some of the people who have contributed to +Libav/FFmpeg. The names are sorted alphabetically by last name. As this file is +currently quite outdated and git serves as a much better tool for determining +authorship, it remains here for historical reasons only. -To check the log, you can type the command "git log" in the FFmpeg -source directory, or browse the online repository at -http://source.ffmpeg.org. +DĂ©nes Balatoni +Michel Bardiaux +Fabrice Bellard +Patrice Bensoussan +Alex Beregszaszi +BERO +Thilo Borgmann +Mario Brito +Ronald Bultje +Alex Converse +Maarten Daniels +Reimar Doeffinger +Tim Ferguson +Brian Foley +Arpad Gereoffy +Philip Gladstone +Vladimir Gneushev +Roine Gustafsson +David Hammerton +Wolfgang Hesseler +Marc Hoffman +Falk Hueffner +AurĂ©lien Jacobs +Steven Johnson +Zdenek Kabelac +Robin Kay +Todd Kirby +Nick Kurshev +Benjamin Larsson +LoĂ¯c Le Loarer +Daniel Maas +Mike Melanson +Loren Merritt +Jeff Muizelaar +Michael Niedermayer +François Revol +Peter Ross +MĂ¥ns RullgĂ¥rd +Stefano Sabatini +Roman Shaposhnik +Oded Shimon +Dieter Shirley +Konstantin Shishkov +Juan J. Sierralta +Ewald Snel +Sascha Sommer +Leon van Stuivenberg +Roberto Togni +Lionel Ulmer +Reynaldo Verdejo diff --git a/Changelog b/Changelog index 3d416c4d2f..f157f7d55d 100644 --- a/Changelog +++ b/Changelog @@ -2,276 +2,141 @@ Entries are sorted chronologically from oldest to youngest within each release, releases are sorted from youngest to oldest. version : -- AC3 fixed-point decoding +- libx265 encoder - shuffleplanes filter -- subfile protocol -- Phantom Cine demuxer - replaygain data export -- VP7 video decoder - Alias PIX image encoder and decoder -- Improvments to the BRender PIX image decoder -- Improvments to the XBM decoder -- QTKit input device -- improvments to OpenEXR image decoder +- BRender PIX image decoder +- Amazing Studio PAF playback support +- XBM decoder +- bmp standalone parser +- OpenEXR image decoder +- support encoding and decoding 4-channel SGI images - support decoding 16-bit RLE SGI images -- GDI screen grabbing for Windows -- alternative rendition support for HTTP Live Streaming -- AVFoundation input device -- Direct Stream Digital (DSD) decoder -- Magic Lantern Video (MLV) demuxer +- VP7 video decoder +- LucasArts SMUSH SANM video decoder +- LucasArts SMUSH VIMA audio decoder (ADPCM) +- LucasArts SMUSH demuxer +- MP2 encoding via TwoLAME +- asettb filter +- Silicon Graphics RLE 8-bit video decoder +- Silicon Graphics Motion Video Compressor 1 & 2 decoder +- Silicon Graphics Movie demuxer - On2 AVC (Audio for Video) decoder -- support for decoding through DXVA2 in ffmpeg +- support for decoding through DXVA2 in avconv - libbs2b-based stereo-to-binaural audio filter -- libx264 reference frames count limiting depending on level - native Opus decoder - display matrix export and rotation api -- WebVTT encoder -version 2.2: - -- HNM version 4 demuxer and video decoder -- Live HDS muxer -- setsar/setdar filters now support variables in ratio expressions -- elbg filter -- string validation in ffprobe -- support for decoding through VDPAU in ffmpeg (the -hwaccel option) -- complete Voxware MetaSound decoder -- remove mp3_header_compress bitstream filter -- Windows resource files for shared libraries -- aeval filter -- stereoscopic 3d metadata handling -- WebP encoding via libwebp -- ATRAC3+ decoder -- VP8 in Ogg demuxing -- side & metadata support in NUT -- framepack filter -- XYZ12 rawvideo support in NUT -- Exif metadata support in WebP decoder -- OpenGL device -- Use metadata_header_padding to control padding in ID3 tags (currently used in - MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block. -- Mirillis FIC video decoder -- Support DNx444 -- libx265 encoder -- dejudder filter -- Autodetect VDA like all other hardware accelerations - - -version 2.1: - -- aecho filter -- perspective filter ported from libmpcodecs -- ffprobe -show_programs option -- compand filter +version 10: +- av_strnstr +- support ID3v2 tags in ASF files +- reference-counting for AVFrame and AVPacket data +- avconv now fails when input options are used for output file + or vice versa +- avconv options -filter_script and -filter_complex_script, which allow a + filtergraph description to be read from a file +- uniform options syntax across all filters +- interlace filter +- JPEG 2000 decoder +- asetpts filter (same as setpts, but for audio) +- trim and atrim filters +- avconv -t and -ss (output-only) options are now sample-accurate when + transcoding audio +- Matroska muxer can now put the index at the beginning of the file. +- avconv -deinterlace option removed, the yadif filter should be used instead +- Apple Intermediate Codec decoder +- Escape 130 video decoder +- support for slice multithreading in libavfilter +- VC-1 interlaced B-frame support +- support for WavPack muxing (raw and in Matroska) +- Go2Webinar decoder +- WavPack encoding through libwavpack +- Added the -n parameter to avconv - RTMP seek support -- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate +- when transcoding with avconv (i.e. not streamcopying), -ss is now accurate even when used as an input option. Previous behavior can be restored with the -noaccurate_seek option. -- ffmpeg -t option can now be used for inputs, to limit the duration of +- avconv -t option can now be used for inputs, to limit the duration of data read from an input file -- incomplete Voxware MetaSound decoder -- read EXIF metadata from JPEG -- DVB teletext decoder -- phase filter ported from libmpcodecs -- w3fdif filter -- Opus support in Matroska -- FFV1 version 1.3 is stable and no longer experimental -- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support -- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be - more consistent with other muxers. -- adelay filter -- pullup filter ported from libmpcodecs -- ffprobe -read_intervals option -- Lossless and alpha support for WebP decoder +- Voxware MetaSound decoder +- WebP decoder - Error Resilient AAC syntax (ER AAC LC) decoding - Low Delay AAC (ER AAC LD) decoding - mux chapters in ASF files -- SFTP protocol (via libssh) -- libx264: add ability to encode in YUVJ422P and YUVJ444P -- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does -- make decoding alpha optional for prores, ffv1 and vp6 by setting - the skip_alpha flag. -- ladspa wrapper filter -- native VP9 decoder -- dpx parser -- max_error_rate parameter in ffmpeg -- PulseAudio output device -- ReplayGain scanner +- Opus in Ogg demuxing - Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support) -- Linux framebuffer output device +- F4V muxer +- HNM version 4 demuxer and video decoder - HEVC decoder - raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing -- mergeplanes filter +- remove avplay -vismv option, which has not worked for a long time +- Live HDS muxer +- setsar/setdar filters now support variables in ratio expressions +- dar variable in the scale filter now returns the actual DAR (i.e. a * sar) +- VP9 decoder +- support for decoding through VDPAU in avconv (the -hwaccel option) +- remove mp3_header_(de)compress bitstream filters +- stereoscopic 3d metadata handling +- png standalone parser +- WebP encoding via libwebp +- ATRAC3+ decoder +- framepack filter +- Mirillis FIC video decoder +- Support DNx444 +- compand audio filter -version 2.0: - -- curves filter -- reference-counting for AVFrame and AVPacket data -- ffmpeg now fails when input options are used for output file - or vice versa -- support for Monkey's Audio versions from 3.93 -- perms and aperms filters -- audio filtering support in ffplay -- 10% faster aac encoding on x86 and MIPS -- sine audio filter source -- WebP demuxing and decoding support -- ffmpeg options -filter_script and -filter_complex_script, which allow a - filtergraph description to be read from a file -- OpenCL support -- audio phaser filter -- separatefields filter -- libquvi demuxer -- uniform options syntax across all filters -- telecine filter -- interlace filter -- smptehdbars source -- inverse telecine filters (fieldmatch and decimate) -- colorbalance filter -- colorchannelmixer filter -- The matroska demuxer can now output proper verbatim ASS packets. It will - become the default at the next libavformat major bump. -- decent native animated GIF encoding -- asetrate filter -- interleave filter -- timeline editing with filters -- vidstabdetect and vidstabtransform filters for video stabilization using - the vid.stab library -- astats filter -- trim and atrim filters -- ffmpeg -t and -ss (output-only) options are now sample-accurate when - transcoding audio -- Matroska muxer can now put the index at the beginning of the file. -- extractplanes filter -- avectorscope filter -- ADPCM DTK decoder -- ADP demuxer -- RSD demuxer -- RedSpark demuxer -- ADPCM IMA Radical decoder -- zmq filters -- DCT denoiser filter (dctdnoiz) -- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow") -- Apple Intermediate Codec decoder -- Escape 130 video decoder -- FTP protocol support -- V4L2 output device -- 3D LUT filter (lut3d) -- SMPTE 302M audio encoder -- support for slice multithreading in libavfilter -- Hald CLUT support (generation and filtering) -- VC-1 interlaced B-frame support -- support for WavPack muxing (raw and in Matroska) -- XVideo output device -- vignette filter -- True Audio (TTA) encoder -- Go2Webinar decoder -- mcdeint filter ported from libmpcodecs -- sab filter ported from libmpcodecs -- ffprobe -show_chapters option -- WavPack encoding through libwavpack -- rotate filter -- spp filter ported from libmpcodecs -- libgme support -- psnr filter - - -version 1.2: - +version 9: +- av_basename and av_dirname +- adobe and limelight publisher authentication in RTMP - VDPAU hardware acceleration through normal hwaccel - SRTP support -- Error diffusion dither in Swscale -- Chained Ogg support -- Theora Midstream reconfiguration support -- EVRC decoder -- audio fade filter -- filtering audio with unknown channel layout -- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass - and treble audio filter -- improved showspectrum filter, with multichannel support and sox-like colors -- histogram filter -- tee muxer -- il filter ported from libmpcodecs -- support ID3v2 tags in ASF files -- encrypted TTA stream decoding support -- RF64 support in WAV muxer -- noise filter ported from libmpcodecs -- Subtitles character encoding conversion -- blend filter -- stereo3d filter ported from libmpcodecs -version 1.1: - -- stream disposition information printing in ffprobe -- filter for loudness analysis following EBU R128 -- Opus encoder using libopus -- ffprobe -select_streams option -- Pinnacle TARGA CineWave YUV16 decoder -- TAK demuxer, decoder and parser -- DTS-HD demuxer -- remove -same_quant, it hasn't worked for years -- FFM2 support -- X-Face image encoder and decoder +version 9_beta3: +- ashowinfo audio filter - 24-bit FLAC encoding +- audio volume filter +- deprecated the avconv -vol option. the volume filter is to be used instead. - multi-channel ALAC encoding up to 7.1 +- TAK demuxer, parser, and decoder +- adaptive frame-level multithreading for H.264 + + +version 9_beta2: - metadata (INFO tag) support in WAV muxer -- subtitles raw text decoder - support for building DLLs using MSVC -- LVF demuxer -- ffescape tool -- metadata (info chunk) support in CAF muxer -- field filter ported from libmpcodecs -- AVR demuxer -- geq filter ported from libmpcodecs -- remove ffserver daemon mode -- AST muxer/demuxer -- new expansion syntax for drawtext -- BRender PIX image decoder -- ffprobe -show_entries option -- ffprobe -sections option -- ADPCM IMA Dialogic decoder -- BRSTM demuxer -- animated GIF decoder and demuxer -- PVF demuxer -- subtitles filter -- IRCAM muxer/demuxer -- Paris Audio File demuxer -- Virtual concatenation demuxer -- VobSub demuxer -- JSON captions for TED talks decoding support -- SOX Resampler support in libswresample -- aselect filter -- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder -- Silicon Graphics Motion Video Compressor 1 & 2 decoder -- Silicon Graphics Movie demuxer -- apad filter -- Resolution & pixel format change support with multithreading for H.264 -- documentation split into per-component manuals -- pp (postproc) filter ported from MPlayer -- NIST Sphere demuxer -- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders -- Sony Wave64 muxer -- adobe and limelight publisher authentication in RTMP -- data: URI scheme +- remove avserver daemon mode - support building on the Plan 9 operating system -- kerndeint filter ported from MPlayer -- histeq filter ported from VirtualDub -- Megalux Frame demuxer -- 012v decoder -- Improved AVC Intra decoding support +- ffv1: support version 1.3 -version 1.0: +version 9_beta1: -- INI and flat output in ffprobe -- Scene detection in libavfilter +- XWD encoder and decoder +- Support for fragmentation in the mov/mp4 muxer +- ISMV (Smooth Streaming) muxer +- CDXL demuxer and decoder +- Apple ProRes encoder +- Sun Rasterfile Encoder +- remove libpostproc +- ID3v2 attached pictures reading and writing +- WMA Lossless decoder +- XBM encoder +- RealAudio Lossless decoder +- ZeroCodec decoder +- drop support for avconv without libavfilter +- add libavresample audio conversion library +- audio filters support in libavfilter and avconv +- add fps filter +- audio split filter +- audio mix filter +- avprobe output is now standard INI or JSON. The old format can still + be used with -of old. - Indeo Audio decoder - channelsplit audio filter -- setnsamples audio filter -- atempo filter -- ffprobe -show_data option - RTMPT protocol support - iLBC encoding/decoding via libilbc - Microsoft Screen 1 decoder @@ -284,146 +149,44 @@ version 1.0: - Microsoft Expression Encoder Screen decoder - RTMPS protocol support - RTMPTS protocol support +- JPEG 2000 encoding support through OpenJPEG +- G.723.1 demuxer and decoder - RTMPE protocol support - RTMPTE protocol support -- showwaves and showspectrum filter -- LucasArts SMUSH SANM playback support -- LucasArts SMUSH VIMA audio decoder (ADPCM) -- LucasArts SMUSH demuxer -- SAMI, RealText and SubViewer demuxers and decoders -- Heart Of Darkness PAF playback support -- iec61883 device -- asettb filter -- new option: -progress -- 3GPP Timed Text encoder/decoder -- GeoTIFF decoder support -- ffmpeg -(no)stdin option -- Opus decoder using libopus -- caca output device using libcaca -- alphaextract and alphamerge filters -- concat filter -- flite filter - Canopus Lossless Codec decoder -- bitmap subtitles in filters (experimental and temporary) -- MP2 encoding via TwoLAME -- bmp parser -- smptebars source -- asetpts filter -- hue filter -- ICO muxer -- SubRip encoder and decoder without embedded timing -- edge detection filter -- framestep filter -- ffmpeg -shortest option is now per-output file +- avconv -shortest option is now per-output file, -pass and -passlogfile are now per-output stream -- volume measurement filter - Ut Video encoder - Microsoft Screen 2 decoder -- smartblur filter ported from MPlayer -- CPiA decoder -- decimate filter ported from MPlayer - RTP depacketization of JPEG - Smooth Streaming live segmenter muxer -- F4V muxer -- sendcmd and asendcmd filters -- WebVTT demuxer and decoder (simple tags supported) - RTP packetization of JPEG -- faststart option in the MOV/MP4 muxer +- Opus decoder and encoder using libopus +- remove -same_quant, it hasn't worked for years - support for building with MSVC -version 0.11: +version 0.8: -- Fixes: CVE-2012-2772, CVE-2012-2774, CVE-2012-2775, CVE-2012-2776, CVE-2012-2777, - CVE-2012-2779, CVE-2012-2782, CVE-2012-2783, CVE-2012-2784, CVE-2012-2785, - CVE-2012-2786, CVE-2012-2787, CVE-2012-2788, CVE-2012-2789, CVE-2012-2790, - CVE-2012-2791, CVE-2012-2792, CVE-2012-2793, CVE-2012-2794, CVE-2012-2795, - CVE-2012-2796, CVE-2012-2797, CVE-2012-2798, CVE-2012-2799, CVE-2012-2800, - CVE-2012-2801, CVE-2012-2802, CVE-2012-2803, CVE-2012-2804, -- v408 Quicktime and Microsoft AYUV Uncompressed 4:4:4:4 encoder and decoder -- setfield filter -- CDXL demuxer and decoder -- Apple ProRes encoder -- ffprobe -count_packets and -count_frames options -- Sun Rasterfile Encoder -- ID3v2 attached pictures reading and writing -- WMA Lossless decoder -- bluray protocol -- blackdetect filter -- libutvideo encoder wrapper (--enable-libutvideo) -- swapuv filter -- bbox filter -- XBM encoder and decoder -- RealAudio Lossless decoder -- ZeroCodec decoder -- tile video filter -- Metal Gear Solid: The Twin Snakes demuxer -- OpenEXR image decoder -- removelogo filter -- drop support for ffmpeg without libavfilter -- drawtext video filter: fontconfig support -- ffmpeg -benchmark_all option -- super2xsai filter ported from libmpcodecs -- add libavresample audio conversion library for compatibility -- MicroDVD decoder -- Avid Meridien (AVUI) encoder and decoder -- accept + prefix to -pix_fmt option to disable automatic conversions. -- complete audio filtering in libavfilter and ffmpeg -- add fps filter -- vorbis parser -- png parser -- audio mix filter -- ffv1: support (draft) version 1.3 - - -version 0.10: - -- Fixes: CVE-2011-3929, CVE-2011-3934, CVE-2011-3935, CVE-2011-3936, - CVE-2011-3937, CVE-2011-3940, CVE-2011-3941, CVE-2011-3944, - CVE-2011-3945, CVE-2011-3946, CVE-2011-3947, CVE-2011-3949, - CVE-2011-3950, CVE-2011-3951, CVE-2011-3952 -- v410 Quicktime Uncompressed 4:4:4 10-bit encoder and decoder -- SBaGen (SBG) binaural beats script demuxer -- OpenMG Audio muxer -- Timecode extraction in DV and MOV -- thumbnail video filter -- XML output in ffprobe -- asplit audio filter -- tinterlace video filter -- astreamsync audio filter -- amerge audio filter -- ISMV (Smooth Streaming) muxer - GSM audio parser - SMJPEG muxer -- XWD encoder and decoder + + +version 0.8_beta2: + - Automatic thread count based on detection number of (available) CPU cores -- y41p Brooktree Uncompressed 4:1:1 12-bit encoder and decoder -- ffprobe -show_error option -- Avid 1:1 10-bit RGB Packer codec -- v308 Quicktime Uncompressed 4:4:4 encoder and decoder -- yuv4 libquicktime packed 4:2:0 encoder and decoder -- ffprobe -show_frames option -- silencedetect audio filter -- ffprobe -show_program_version, -show_library_versions, -show_versions options +- Deprecate libpostproc. If desired, the switch --enable-postproc will + enable it but it may be removed in a later Libav release. - rv34: frame-level multi-threading - optimized iMDCT transform on x86 using SSE for for mpegaudiodec -- Improved PGS subtitle decoder -- dumpgraph option to lavfi device -- r210 and r10k encoders -- ffwavesynth decoder -- aviocat tool -- ffeval tool -- support encoding and decoding 4-channel SGI images -version 0.9: +version 0.8_beta1: -- openal input device added -- boxblur filter added - BWF muxer - Flash Screen Video 2 decoder -- lavfi input device added -- added avconv, which is almost the same for now, except +- ffplay/ffprobe/ffserver renamed to avplay/avprobe/avserver +- ffmpeg deprecated, added avconv, which is almost the same for now, except for a few incompatible changes in the options, which will hopefully make them easier to use. The changes are: * The options placement is now strictly enforced! While in theory the @@ -493,40 +256,23 @@ easier to use. The changes are: * -vframes/-aframes/-dframes options are now aliases to the new -frames option. * -vtag/-atag/-stag options are now aliases to the new -tag option. - XMV demuxer -- LOAS demuxer -- ashowinfo filter added - Windows Media Image decoder -- amovie source added - LATM muxer/demuxer -- Speex encoder via libspeex -- JSON output in ffprobe -- WTV muxer -- Optional C++ Support (needed for libstagefright) -- H.264 Decoding on Android via Stagefright -- Prores decoder -- BIN/XBIN/ADF/IDF text file decoder -- aconvert audio filter added -- audio support to lavfi input device added +- showinfo filter +- split filter - libcdio-paranoia input device for audio CD grabbing +- select filter - Apple ProRes decoder - CELT in Ogg demuxing -- G.723.1 demuxer and decoder -- libmodplug support (--enable-libmodplug) - VC-1 interlaced decoding -- libutvideo wrapper (--enable-libutvideo) -- aevalsrc audio source added +- lut, lutrgb, and lutyuv filters +- boxblur filter - Ut Video decoder - Speex encoding via libspeex - 4:2:2 H.264 decoding support - 4:2:2 and 4:4:4 H.264 encoding with libx264 - Pulseaudio input device -- Prores encoder -- Video Decoder Acceleration (VDA) HWAccel module. - replacement Indeo 3 decoder -- new ffmpeg option: -map_channel -- volume audio filter added -- earwax audio filter added -- libv4l2 support (--enable-libv4l2) - TLS/SSL and HTTPS protocol support - AVOptions API rewritten and documented - most of CODEC_FLAG2_*, some CODEC_FLAG_* and many codec-specific fields in @@ -536,27 +282,44 @@ easier to use. The changes are: - Discworld II BMV decoding support - VBLE Decoder - OS X Video Decoder Acceleration (VDA) support -- compact and csv output in ffprobe -- pan audio filter -- IFF Amiga Continuous Bitmap (ACBM) decoder -- ass filter - CRI ADX audio format muxer and demuxer - Playstation Portable PMP format demuxer -- Microsoft Windows ICO demuxer -- life source - PCM format support in OMA demuxer - CLJR encoder -- new option: -report - Dxtory capture format decoder -- cellauto source +- v410 QuickTime uncompressed 4:4:4 10-bit encoder and decoder +- OpenMG Audio muxer - Simple segmenting muxer - Indeo 4 decoder - SMJPEG demuxer -version 0.8: +version 0.7: + +- E-AC-3 audio encoder +- ac3enc: add channel coupling support +- floating-point sample format support for (E-)AC-3, DCA, AAC, Vorbis decoders +- H.264/MPEG frame-level multithreading +- av_metadata_* functions renamed to av_dict_* and moved to libavutil +- 4:4:4 H.264 decoding support +- 10-bit H.264 optimizations for x86 +- bump libswscale for recently reported ABI break + + +version 0.7_beta2: + +- VP8 frame-level multithreading +- NEON optimizations for VP8 +- removed a lot of deprecated API cruft +- FFT and IMDCT optimizations for AVX (Sandy Bridge) processors +- DPX image encoder +- SMPTE 302M AES3 audio decoder +- ffmpeg no longer quits after the 'q' key is pressed; use 'ctrl+c' instead +- 9bit and 10bit per sample support in the H.264 decoder + + +version 0.7_beta1: -- many many things we forgot because we rather write code than changelogs - WebM support in Matroska de/muxer - low overhead Ogg muxing - MMS-TCP support @@ -564,7 +327,6 @@ version 0.8: - Demuxer for On2's IVF format - Pictor/PC Paint decoder - HE-AAC v2 decoder -- HE-AAC v2 encoding with libaacplus - libfaad2 wrapper removed - DTS-ES extension (XCh) decoding support - native VP8 decoder @@ -576,7 +338,6 @@ version 0.8: - RTP depacketization of QDM2 - ANSI/ASCII art playback system - Lego Mindstorms RSO de/muxer -- libavcore added (and subsequently removed) - SubRip subtitle file muxer and demuxer - Chinese AVS encoding via libxavs - ffprobe -show_packets option added @@ -623,7 +384,7 @@ version 0.8: - replace the ocv_smooth filter with a more generic ocv filter - Windows Televison (WTV) demuxer - FFmpeg metadata format muxer and demuxer -- SubRip (srt) subtitle encoder and decoder +- SubRip (srt) subtitle decoder - floating-point AC-3 encoder added - Lagarith decoder - ffmpeg -copytb option added @@ -636,46 +397,11 @@ version 0.8: - sndio support for playback and record - Linux framebuffer input device added - Chronomaster DFA decoder -- DPX image encoder -- MicroDVD subtitle file muxer and demuxer -- Playstation Portable PMP format demuxer -- fieldorder video filter added +- Mobotix MxPEG decoder - AAC encoding via libvo-aacenc - AMR-WB encoding via libvo-amrwbenc - xWMA demuxer -- Mobotix MxPEG decoder -- VP8 frame-multithreading -- NEON optimizations for VP8 -- Lots of deprecated API cruft removed -- fft and imdct optimizations for AVX (Sandy Bridge) processors -- showinfo filter added -- SMPTE 302M AES3 audio decoder -- Apple Core Audio Format muxer -- 9bit and 10bit per sample support in the H.264 decoder -- 9bit and 10bit FFV1 encoding / decoding -- split filter added -- select filter added -- sdl output device added -- libmpcodecs video filter support (3 times as many filters than before) -- mpeg2 aspect ratio dection fixed -- libxvid aspect pickiness fixed -- Frame multithreaded decoding -- E-AC-3 audio encoder -- ac3enc: add channel coupling support -- floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders. -- H264/MPEG frame-level multi-threading -- All av_metadata_* functions renamed to av_dict_* and moved to libavutil -- 4:4:4 H.264 decoding support -- 10-bit H.264 optimizations for x86 -- lut, lutrgb, and lutyuv filters added -- buffersink libavfilter sink added -- Bump libswscale for recently reported ABI break -- New J2K encoder (via OpenJPEG) - - -version 0.7: - -- all the changes for 0.8, but keeping API/ABI compatibility with the 0.6 release +- fieldorder video filter added version 0.6: @@ -916,7 +642,6 @@ version 0.5: - Gopher client support - MXF D-10 muxer - generic metadata API -- flash ScreenVideo2 encoder version 0.4.9-pre1: @@ -1119,7 +844,7 @@ version 0.4.5: - MPEG-4 vol header fixes (Jonathan Marsden ) - ARM optimizations (Lionel Ulmer ). - Windows porting of file converter -- added MJPEG raw format (input/output) +- added MJPEG raw format (input/ouput) - added JPEG image format support (input/output) diff --git a/INSTALL b/INSTALL index 9549346302..ac5dc5dbb6 100644 --- a/INSTALL +++ b/INSTALL @@ -2,14 +2,17 @@ 1) Type './configure' to create the configuration. A list of configure options is printed by running 'configure --help'. -'configure' can be launched from a directory different from the FFmpeg +'configure' can be launched from a directory different from the Libav sources to build the objects out of tree. To do this, use an absolute -path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'. +path when launching 'configure', e.g. '/libavdir/libav/configure'. -2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required. +2) Then type 'make' to build Libav. GNU Make 3.81 or later is required. 3) Type 'make install' to install all binaries and libraries you built. NOTICE - Non system dependencies (e.g. libx264, libvpx) are disabled by default. + + - The default cflags include -g, if you want lean libraries you can either + pass --disable-debug or strip the debug symbols at a later time. diff --git a/LICENSE b/LICENSE index 1f757aa782..fb2917830c 100644 --- a/LICENSE +++ b/LICENSE @@ -1,53 +1,26 @@ -FFmpeg: +Libav: +====== -Most files in FFmpeg are under the GNU Lesser General Public License version 2.1 +Most files in Libav are under the GNU Lesser General Public License version 2.1 or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to -FFmpeg. +Libav. -Some optional parts of FFmpeg are licensed under the GNU General Public License +Some optional parts of Libav are licensed under the GNU General Public License version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of these parts are used by default, you have to explicitly pass --enable-gpl to -configure to activate them. In this case, FFmpeg's license changes to GPL v2+. +configure to activate them. In this case, Libav's license changes to GPL v2+. -Specifically, the GPL parts of FFmpeg are +Specifically, the GPL parts of Libav are -- libpostproc -- libmpcodecs -- optional x86 optimizations in the files - libavcodec/x86/idct_mmx.c -- libutvideo encoding/decoding wrappers in - libavcodec/libutvideo*.cpp - the X11 grabber in libavdevice/x11grab.c -- the swresample test app in - libswresample/swresample-test.c - the texi2pod.pl tool - the following filters in libavfilter: - - f_ebur128.c - vf_blackframe.c - vf_boxblur.c - - vf_colormatrix.c - vf_cropdetect.c - - vf_decimate.c - vf_delogo.c - - vf_geq.c - - vf_histeq.c - vf_hqdn3d.c - - vf_kerndeint.c - - vf_mcdeint.c - - vf_mp.c - - vf_owdenoise.c - - vf_perspective.c - - vf_phase.c - - vf_pp.c - - vf_pullup.c - - vf_sab.c - - vf_smartblur.c - - vf_spp.c - - vf_stereo3d.c - - vf_super2xsai.c - - vf_tinterlace.c - - vsrc_mptestsrc.c Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then the configure parameter --enable-version3 will activate this licensing option @@ -67,37 +40,26 @@ There are a handful of files under other licensing terms, namely: external libraries ================== -FFmpeg can be combined with a number of external libraries, which sometimes +Libav can be combined with a number of external libraries, which sometimes affect the licensing of binaries resulting from the combination. compatible libraries -------------------- -The following libraries are under GPL: - - frei0r - - libcdio - - libutvideo - - libvidstab - - libx264 - - libx265 - - libxavs - - libxvid -When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by +The libcdio, libx264, libx265, libxavs and libxvid libraries are under GPL. When +combining them with Libav, Libav needs to be licensed as GPL as well by passing --enable-gpl to configure. The OpenCORE and VisualOn libraries are under the Apache License 2.0. That license is incompatible with the LGPL v2.1 and the GPL v2, but not with -version 3 of those licenses. So to combine these libraries with FFmpeg, the +version 3 of those licenses. So to combine these libraries with Libav, the license version needs to be upgraded by passing --enable-version3 to configure. incompatible libraries ---------------------- -The Fraunhofer AAC library, FAAC and aacplus are under licenses which -are incompatible with the GPLv2 and v3. We do not know for certain if their -licenses are compatible with the LGPL. -If you wish to enable these libraries, pass --enable-nonfree to configure. -But note that if you enable any of these libraries the resulting binary will -be under a complex license mix that is more restrictive than the LGPL and that -may result in additional obligations. It is possible that these -restrictions cause the resulting binary to be unredistributeable. +The Fraunhofer AAC library, FAAC and OpenSSL are under licenses incompatible +with all (L)GPL versions. Thus, unfortunately, since both licenses cannot be +satisfied simultaneously, binaries resulting from the combination of Libav +with these libraries are nonfree und unredistributable. If you wish to enable +any of these libraries nonetheless, pass --enable-nonfree to configure. diff --git a/MAINTAINERS b/MAINTAINERS deleted file mode 100644 index edfc24a0e8..0000000000 --- a/MAINTAINERS +++ /dev/null @@ -1,566 +0,0 @@ -FFmpeg maintainers -================== - -Below is a list of the people maintaining different parts of the -FFmpeg code. - -Please try to keep entries where you are the maintainer up to date! - -Names in () mean that the maintainer currently has no time to maintain the code. -A (CC
) after the name means that the maintainer prefers to be CC-ed on -patches and related discussions. - - -Project Leader -============== - -Michael Niedermayer - final design decisions - - -Applications -============ - -ffmpeg: - ffmpeg.c Michael Niedermayer - -ffplay: - ffplay.c Marton Balint - -ffprobe: - ffprobe.c Stefano Sabatini - -ffserver: - ffserver.c Reynaldo H. Verdejo Pinochet - -Commandline utility code: - cmdutils.c, cmdutils.h Michael Niedermayer - -QuickTime faststart: - tools/qt-faststart.c Baptiste Coudurier - - -Miscellaneous Areas -=================== - -documentation Stefano Sabatini, Mike Melanson, Timothy Gu -build system (configure,Makefiles) Diego Biurrun, Mans Rullgard -project server ĂrpĂ¡d Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser -presets Robert Swain -metadata subsystem Aurelien Jacobs -release management Michael Niedermayer - - -Communication -============= - -website Robert Swain -Trac bug tracker Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan -mailinglists Michael Niedermayer, Baptiste Coudurier, Lou Logan -Google+ Paul B Mahol, Michael Niedermayer, Alexander Strasser -Twitter Lou Logan -Launchpad Timothy Gu - - -libavutil -========= - -External Interfaces: - libavutil/avutil.h Michael Niedermayer -Internal Interfaces: - libavutil/common.h Michael Niedermayer - -Other: - bprint Nicolas George - bswap.h - des Reimar Doeffinger - dynarray.h Nicolas George - eval.c, eval.h Michael Niedermayer - float_dsp Loren Merritt - hash Reimar Doeffinger - intfloat* Michael Niedermayer - integer.c, integer.h Michael Niedermayer - lzo Reimar Doeffinger - mathematics.c, mathematics.h Michael Niedermayer - mem.c, mem.h Michael Niedermayer - opencl.c, opencl.h Wei Gao - opt.c, opt.h Michael Niedermayer - rational.c, rational.h Michael Niedermayer - rc4 Reimar Doeffinger - ripemd.c, ripemd.h James Almer - timecode ClĂ©ment BÅ“sch - - -libavcodec -========== - -Generic Parts: - External Interfaces: - avcodec.h Michael Niedermayer - utility code: - utils.c Michael Niedermayer - audio and video frame extraction: - parser.c Michael Niedermayer - bitstream reading: - bitstream.c, bitstream.h Michael Niedermayer - CABAC: - cabac.h, cabac.c Michael Niedermayer - codec names: - codec_names.sh Nicolas George - DSP utilities: - dsputils.c, dsputils.h Michael Niedermayer - entropy coding: - rangecoder.c, rangecoder.h Michael Niedermayer - lzw.* Michael Niedermayer - floating point AAN DCT: - faandct.c, faandct.h Michael Niedermayer - Golomb coding: - golomb.c, golomb.h Michael Niedermayer - LPC: - lpc.c, lpc.h Justin Ruggles - motion estimation: - motion* Michael Niedermayer - rate control: - ratecontrol.c Michael Niedermayer - libxvid_rc.c Michael Niedermayer - simple IDCT: - simple_idct.c, simple_idct.h Michael Niedermayer - postprocessing: - libpostproc/* Michael Niedermayer - table generation: - tableprint.c, tableprint.h Reimar Doeffinger - fixed point FFT: - fft* Zeljko Lukac - -Codecs: - 4xm.c Michael Niedermayer - 8bps.c Roberto Togni - 8svx.c Jaikrishnan Menon - aasc.c Kostya Shishkov - ac3* Justin Ruggles - alacenc.c Jaikrishnan Menon - alsdec.c Thilo Borgmann - apedec.c Kostya Shishkov - ass* Aurelien Jacobs - asv* Michael Niedermayer - atrac3* Benjamin Larsson - atrac3plus* Maxim Poliakovski - bgmc.c, bgmc.h Thilo Borgmann - bink.c Kostya Shishkov - binkaudio.c Peter Ross - bmp.c Mans Rullgard, Kostya Shishkov - cavs* Stefan Gehrer - cdxl.c Paul B Mahol - celp_filters.* Vitor Sessak - cinepak.c Roberto Togni - cinepakenc.c Rl / Aetey G.T. AB - cljr Alex Beregszaszi - cllc.c Derek Buitenhuis - cook.c, cookdata.h Benjamin Larsson - cpia.c Stephan Hilb - crystalhd.c Philip Langdale - cscd.c Reimar Doeffinger - dca.c Kostya Shishkov, Benjamin Larsson - dnxhd* Baptiste Coudurier - dpcm.c Mike Melanson - dv.c Roman Shaposhnik - dxa.c Kostya Shishkov - eacmv*, eaidct*, eat* Peter Ross - exif.c, exif.h Thilo Borgmann - ffv1* Michael Niedermayer - ffwavesynth.c Nicolas George - fic.c Derek Buitenhuis - flac* Justin Ruggles - flashsv* Benjamin Larsson - flicvideo.c Mike Melanson - g722.c Martin Storsjo - g726.c Roman Shaposhnik - gifdec.c Baptiste Coudurier - h261* Michael Niedermayer - h263* Michael Niedermayer - h264* Loren Merritt, Michael Niedermayer - huffyuv.c Michael Niedermayer - idcinvideo.c Mike Melanson - imc* Benjamin Larsson - indeo2* Kostya Shishkov - indeo5* Kostya Shishkov - interplayvideo.c Mike Melanson - ivi* Kostya Shishkov - jacosub* ClĂ©ment BÅ“sch - jpeg2000* Nicolas Bertrand - jpeg_ls.c Kostya Shishkov - jvdec.c Peter Ross - kmvc.c Kostya Shishkov - lcl*.c Roberto Togni, Reimar Doeffinger - libcelt_dec.c Nicolas George - libdirac* David Conrad - libgsm.c Michel Bardiaux - libopenjpeg.c Jaikrishnan Menon - libopenjpegenc.c Michael Bradshaw - libschroedinger* David Conrad - libspeexdec.c Justin Ruggles - libtheoraenc.c David Conrad - libutvideo* Derek Buitenhuis - libvorbis.c David Conrad - libvpx* James Zern - libx264.c Mans Rullgard, Jason Garrett-Glaser - libx265.c Derek Buitenhuis - libxavs.c Stefan Gehrer - libzvbi-teletextdec.c Marton Balint - loco.c Kostya Shishkov - lzo.h, lzo.c Reimar Doeffinger - mdec.c Michael Niedermayer - mimic.c Ramiro Polla - mjpeg*.c Michael Niedermayer - mlp* Ramiro Polla - mmvideo.c Peter Ross - mpc* Kostya Shishkov - mpeg12.c, mpeg12data.h Michael Niedermayer - mpegvideo.c, mpegvideo.h Michael Niedermayer - mqc* Nicolas Bertrand - msmpeg4.c, msmpeg4data.h Michael Niedermayer - msrle.c Mike Melanson - msvideo1.c Mike Melanson - nellymoserdec.c Benjamin Larsson - nuv.c Reimar Doeffinger - paf.* Paul B Mahol - pcx.c Ivo van Poorten - pgssubdec.c Reimar Doeffinger - ptx.c Ivo van Poorten - qcelp* Reynaldo H. Verdejo Pinochet - qdm2.c, qdm2data.h Roberto Togni, Benjamin Larsson - qdrw.c Kostya Shishkov - qpeg.c Kostya Shishkov - qtrle.c Mike Melanson - ra144.c, ra144.h, ra288.c, ra288.h Roberto Togni - resample2.c Michael Niedermayer - rl2.c Sascha Sommer - rpza.c Roberto Togni - rtjpeg.c, rtjpeg.h Reimar Doeffinger - rv10.c Michael Niedermayer - rv3* Kostya Shishkov - rv4* Kostya Shishkov - s3tc* Ivo van Poorten - smacker.c Kostya Shishkov - smc.c Mike Melanson - smvjpegdec.c Ash Hughes - snow* Michael Niedermayer, Loren Merritt - sonic.c Alex Beregszaszi - srt* Aurelien Jacobs - sunrast.c Ivo van Poorten - svq3.c Michael Niedermayer - tak* Paul B Mahol - targa.c Kostya Shishkov - tiff.c Kostya Shishkov - truemotion1* Mike Melanson - truemotion2* Kostya Shishkov - truespeech.c Kostya Shishkov - tscc.c Kostya Shishkov - tta.c Alex Beregszaszi, Jaikrishnan Menon - ttaenc.c Paul B Mahol - txd.c Ivo van Poorten - ulti* Kostya Shishkov - v410*.c Derek Buitenhuis - vb.c Kostya Shishkov - vble.c Derek Buitenhuis - vc1* Kostya Shishkov - vcr1.c Michael Niedermayer - vda_h264_dec.c Xidorn Quan - vima.c Paul B Mahol - vmnc.c Kostya Shishkov - vorbis_dec.c Denes Balatoni, David Conrad - vorbis_enc.c Oded Shimon - vp3* Mike Melanson - vp5 Aurelien Jacobs - vp6 Aurelien Jacobs - vp8 David Conrad, Jason Garrett-Glaser, Ronald Bultje - vp9 Ronald Bultje, ClĂ©ment BÅ“sch - vqavideo.c Mike Melanson - wavpack.c Kostya Shishkov - wmaprodec.c Sascha Sommer - wmavoice.c Ronald S. Bultje - wmv2.c Michael Niedermayer - wnv1.c Kostya Shishkov - xan.c Mike Melanson - xbm* Paul B Mahol - xface Stefano Sabatini - xl.c Kostya Shishkov - xvmc.c Ivan Kalvachev - xwd* Paul B Mahol - zerocodec.c Derek Buitenhuis - zmbv* Kostya Shishkov - -Hardware acceleration: - crystalhd.c Philip Langdale - dxva2* Laurent Aimar - libstagefright.cpp Mohamed Naufal - vaapi* Gwenole Beauchesne - vda* Sebastien Zwickert - vdpau* Carl Eugen Hoyos - - -libavdevice -=========== - External Interface: - libavdevice/avdevice.h - - - avfoundation.m Thilo Borgmann - dshow.c Roger Pack (CC rogerdpack@gmail.com) - fbdev_enc.c Lukasz Marek - gdigrab.c Roger Pack (CC rogerdpack@gmail.com) - iec61883.c Georg Lippitsch - lavfi Stefano Sabatini - libdc1394.c Roman Shaposhnik - opengl_enc.c Lukasz Marek - pulse_audio_enc.c Lukasz Marek - qtkit.m Thilo Borgmann - sdl Stefano Sabatini - v4l2.c Luca Abeni - vfwcap.c Ramiro Polla - xv.c Lukasz Marek - -libavfilter -=========== - -Generic parts: - graphdump.c Nicolas George - -Filters: - af_adelay.c Paul B Mahol - af_aecho.c Paul B Mahol - af_afade.c Paul B Mahol - af_amerge.c Nicolas George - af_aphaser.c Paul B Mahol - af_aresample.c Michael Niedermayer - af_astats.c Paul B Mahol - af_astreamsync.c Nicolas George - af_atempo.c Pavel Koshevoy - af_biquads.c Paul B Mahol - af_compand.c Paul B Mahol - af_ladspa.c Paul B Mahol - af_pan.c Nicolas George - avf_avectorscope.c Paul B Mahol - vf_blend.c Paul B Mahol - vf_colorbalance.c Paul B Mahol - vf_dejudder.c Nicholas Robbins - vf_delogo.c Jean Delvare (CC ) - vf_drawbox.c/drawgrid Andrey Utkin - vf_extractplanes.c Paul B Mahol - vf_histogram.c Paul B Mahol - vf_il.c Paul B Mahol - vf_mergeplanes.c Paul B Mahol - vf_psnr.c Paul B Mahol - vf_scale.c Michael Niedermayer - vf_separatefields.c Paul B Mahol - vf_stereo3d.c Paul B Mahol - vf_telecine.c Paul B Mahol - vf_yadif.c Michael Niedermayer - -Sources: - vsrc_mandelbrot.c Michael Niedermayer - -libavformat -=========== - -Generic parts: - External Interface: - libavformat/avformat.h Michael Niedermayer - Utility Code: - libavformat/utils.c Michael Niedermayer - - -Muxers/Demuxers: - 4xm.c Mike Melanson - adtsenc.c Robert Swain - afc.c Paul B Mahol - aiffdec.c Baptiste Coudurier, Matthieu Bouron - aiffenc.c Baptiste Coudurier, Matthieu Bouron - ape.c Kostya Shishkov - ass* Aurelien Jacobs - astdec.c Paul B Mahol - astenc.c James Almer - avi* Michael Niedermayer - avisynth.c AvxSynth Team (avxsynth.testing at gmail dot com) - avr.c Paul B Mahol - bink.c Peter Ross - brstm.c Paul B Mahol - caf* Peter Ross - cdxl.c Paul B Mahol - crc.c Michael Niedermayer - daud.c Reimar Doeffinger - dtshddec.c Paul B Mahol - dv.c Roman Shaposhnik - dxa.c Kostya Shishkov - electronicarts.c Peter Ross - epafdec.c Paul B Mahol - ffm* Baptiste Coudurier - flac* Justin Ruggles - flic.c Mike Melanson - flvdec.c, flvenc.c Michael Niedermayer - gxf.c Reimar Doeffinger - gxfenc.c Baptiste Coudurier - hls.c Anssi Hannula - idcin.c Mike Melanson - idroqdec.c Mike Melanson - iff.c Jaikrishnan Menon - img2*.c Michael Niedermayer - ipmovie.c Mike Melanson - ircam* Paul B Mahol - iss.c Stefan Gehrer - jacosub* ClĂ©ment BÅ“sch - jvdec.c Peter Ross - libmodplug.c ClĂ©ment BÅ“sch - libnut.c Oded Shimon - lmlm4.c Ivo van Poorten - lvfdec.c Paul B Mahol - lxfdec.c Tomas Härdin - matroska.c Aurelien Jacobs - matroskadec.c Aurelien Jacobs - matroskaenc.c David Conrad - metadata* Aurelien Jacobs - mgsts.c Paul B Mahol - microdvd* Aurelien Jacobs - mm.c Peter Ross - mov.c Michael Niedermayer, Baptiste Coudurier - movenc.c Baptiste Coudurier, Matthieu Bouron - mpc.c Kostya Shishkov - mpeg.c Michael Niedermayer - mpegenc.c Michael Niedermayer - mpegts.c Marton Balint - mpegtsenc.c Baptiste Coudurier - msnwc_tcp.c Ramiro Polla - mtv.c Reynaldo H. Verdejo Pinochet - mxf* Baptiste Coudurier - mxfdec.c Tomas Härdin - nistspheredec.c Paul B Mahol - nsvdec.c Francois Revol - nut* Michael Niedermayer - nuv.c Reimar Doeffinger - oggdec.c, oggdec.h David Conrad - oggenc.c Baptiste Coudurier - oggparse*.c David Conrad - oma.c Maxim Poliakovski - paf.c Paul B Mahol - psxstr.c Mike Melanson - pva.c Ivo van Poorten - pvfdec.c Paul B Mahol - r3d.c Baptiste Coudurier - raw.c Michael Niedermayer - rdt.c Ronald S. Bultje - rl2.c Sascha Sommer - rmdec.c, rmenc.c Ronald S. Bultje, Kostya Shishkov - rtmp* Kostya Shishkov - rtp.c, rtpenc.c Martin Storsjo - rtpdec_asf.* Ronald S. Bultje - rtpenc_mpv.*, rtpenc_aac.* Martin Storsjo - rtsp.c Luca Barbato - sbgdec.c Nicolas George - sdp.c Martin Storsjo - segafilm.c Mike Melanson - siff.c Kostya Shishkov - smacker.c Kostya Shishkov - smjpeg* Paul B Mahol - spdif* Anssi Hannula - srtdec.c Aurelien Jacobs - swf.c Baptiste Coudurier - takdec.c Paul B Mahol - tta.c Alex Beregszaszi - txd.c Ivo van Poorten - voc.c Aurelien Jacobs - wav.c Michael Niedermayer - wc3movie.c Mike Melanson - webm dash (matroskaenc.c) Vignesh Venkatasubramanian - webvtt* Matthew J Heaney - westwood.c Mike Melanson - wtv.c Peter Ross - wv.c Kostya Shishkov - wvenc.c Paul B Mahol - -Protocols: - bluray.c Petri Hintukainen - ftp.c Lukasz Marek - http.c Ronald S. Bultje - libssh.c Lukasz Marek - mms*.c Ronald S. Bultje - udp.c Luca Abeni - - -libswresample -============= - -Generic parts: - audioconvert.c Michael Niedermayer - dither.c Michael Niedermayer - rematrix*.c Michael Niedermayer - swresample*.c Michael Niedermayer - -Resamplers: - resample*.c Michael Niedermayer - soxr_resample.c Rob Sykes - - -Operating systems / CPU architectures -===================================== - -Alpha Mans Rullgard, Falk Hueffner -ARM Mans Rullgard -AVR32 Mans Rullgard -MIPS Mans Rullgard, Nedeljko Babic -Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier -Amiga / PowerPC Colin Ward -Linux / PowerPC Luca Barbato -Windows MinGW Alex Beregszaszi, Ramiro Polla -Windows Cygwin Victor Paesa -Windows MSVC Matthew Oliver -Windows ICL Matthew Oliver -ADI/Blackfin DSP Marc Hoffman -Sparc Roman Shaposhnik -x86 Michael Niedermayer - - -Releases -======== - -2.2 Michael Niedermayer -2.1 Michael Niedermayer -1.2 Michael Niedermayer - -If you want to maintain an older release, please contact us - - -GnuPG Fingerprints of maintainers and contributors -================================================== - -Alexander Strasser 1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F -Anssi Hannula 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB -Anton Khirnov 6D0C 6625 56F8 65D1 E5F5 814B B50A 1241 C067 07AB -Ash Hughes 694D 43D2 D180 C7C7 6421 ABD3 A641 D0B7 623D 6029 -Attila Kinali 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765 -Baptiste Coudurier 8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA -Ben Littler 3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C -Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8 -BÅ“sch ClĂ©ment 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9 -Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7 -Diego Biurrun 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5 -FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8 -Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4 -Jaikrishnan Menon 61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368 -Jean Delvare 7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A -Justin Ruggles 3136 ECC0 C10D 6C04 5F43 CA29 FCBE CD2A 3787 1EBF -Loren Merritt ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE -Lou Logan 7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A -Luca Barbato 6677 4209 213C 8843 5B67 29E7 E84C 78C2 84E9 0E34 -Michael Niedermayer 9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB -Nicolas George 24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93 -Panagiotis Issaris 6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029 -Peter Ross A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B -Reimar Döffinger C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7 -Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4 -Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A -Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71 -Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C -Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F -Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863 -Tomas Härdin A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551 -Wei Gao 4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9 diff --git a/Makefile b/Makefile index 86b3508425..71c068ca0e 100644 --- a/Makefile +++ b/Makefile @@ -1,41 +1,86 @@ -MAIN_MAKEFILE=1 include config.mak vpath %.c $(SRC_PATH) -vpath %.cpp $(SRC_PATH) vpath %.h $(SRC_PATH) -vpath %.m $(SRC_PATH) vpath %.S $(SRC_PATH) vpath %.asm $(SRC_PATH) -vpath %.rc $(SRC_PATH) vpath %.v $(SRC_PATH) vpath %.texi $(SRC_PATH) -vpath %/fate_config.sh.template $(SRC_PATH) -AVPROGS-$(CONFIG_FFMPEG) += ffmpeg -AVPROGS-$(CONFIG_FFPLAY) += ffplay -AVPROGS-$(CONFIG_FFPROBE) += ffprobe -AVPROGS-$(CONFIG_FFSERVER) += ffserver +ifndef V +Q = @ +ECHO = printf "$(1)\t%s\n" $(2) +BRIEF = CC HOSTCC HOSTLD AS YASM AR LD +SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM STRIP +MSG = $@ +M = @$(call ECHO,$(TAG),$@); +$(foreach VAR,$(BRIEF), \ + $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR)))) +$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR)))) +$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_PATH)/%=%)); $(INSTALL)) +endif -AVPROGS := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF)) -INSTPROGS = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF)) +ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil swscale + +IFLAGS := -I. -I$(SRC_PATH) +CPPFLAGS := $(IFLAGS) $(CPPFLAGS) +CFLAGS += $(ECFLAGS) +CCFLAGS = $(CPPFLAGS) $(CFLAGS) +ASFLAGS := $(CPPFLAGS) $(ASFLAGS) +YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm +HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS) +LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS) + +define COMPILE + $(call $(1)DEP,$(1)) + $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $< +endef + +COMPILE_C = $(call COMPILE,CC) +COMPILE_S = $(call COMPILE,AS) +COMPILE_HOSTC = $(call COMPILE,HOSTCC) + +%.o: %.c + $(COMPILE_C) + +%.o: %.S + $(COMPILE_S) + +%_host.o: %.c + $(COMPILE_HOSTC) + +%.i: %.c + $(CC) $(CCFLAGS) $(CC_E) $< + +%.h.c: + $(Q)echo '#include "$*.h"' >$@ + +%.ver: %.v + $(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@ + +%.c %.h: TAG = GEN + +AVPROGS-$(CONFIG_AVCONV) += avconv +AVPROGS-$(CONFIG_AVPLAY) += avplay +AVPROGS-$(CONFIG_AVPROBE) += avprobe +AVPROGS-$(CONFIG_AVSERVER) += avserver + +AVPROGS := $(AVPROGS-yes:%=%$(EXESUF)) PROGS += $(AVPROGS) -AVBASENAMES = ffmpeg ffplay ffprobe ffserver -ALLAVPROGS = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF)) -ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF)) +AVBASENAMES = avconv avplay avprobe avserver +ALLAVPROGS = $(AVBASENAMES:%=%$(EXESUF)) $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o)) -$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o)) -OBJS-ffmpeg += ffmpeg_opt.o ffmpeg_filter.o -OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o -OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o -OBJS-ffmpeg-$(CONFIG_VDA) += ffmpeg_vda.o +OBJS-avconv += avconv_opt.o avconv_filter.o +OBJS-avconv-$(HAVE_VDPAU_X11) += avconv_vdpau.o +OBJS-avconv-$(HAVE_DXVA2_LIB) += avconv_dxva2.o +OBJS-avconv-$(CONFIG_VDA) += avconv_vda.o -TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 +TESTTOOLS = audiogen videogen rotozoom tiny_psnr base64 HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options -TOOLS = qt-faststart trasher uncoded_frame +TOOLS = qt-faststart trasher TOOLS-$(CONFIG_ZLIB) += cws2fws # $(FFLIBS-yes) needs to be in linking order @@ -44,14 +89,11 @@ FFLIBS-$(CONFIG_AVFILTER) += avfilter FFLIBS-$(CONFIG_AVFORMAT) += avformat FFLIBS-$(CONFIG_AVCODEC) += avcodec FFLIBS-$(CONFIG_AVRESAMPLE) += avresample -FFLIBS-$(CONFIG_POSTPROC) += postproc -FFLIBS-$(CONFIG_SWRESAMPLE) += swresample FFLIBS-$(CONFIG_SWSCALE) += swscale FFLIBS := avutil -DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.ffpreset) $(SRC_PATH)/doc/ffprobe.xsd -EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/Makefile $(SRC_PATH)/doc/examples/README +DATA_FILES := $(wildcard $(SRC_PATH)/presets/*.avpreset) SKIPHEADERS = cmdutils_common_opts.h compat/w32pthreads.h @@ -66,8 +108,6 @@ $(TOOLS): %$(EXESUF): %.o $(EXEOBJS) $(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS) tools/cws2fws$(EXESUF): ELIBS = $(ZLIB) -tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS) -tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS) config.h: .config .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c)) @@ -79,8 +119,7 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS \ HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS \ ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS \ ALTIVEC-OBJS MMX-OBJS YASM-OBJS \ - MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS \ - OBJS SLIBOBJS HOSTOBJS TESTOBJS + OBJS HOSTOBJS TESTOBJS define RESET $(1) := @@ -101,22 +140,16 @@ include $(SRC_PATH)/doc/Makefile define DOPROG OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes) -$(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1)) +$(1)$(EXESUF): $$(OBJS-$(1)) $$(OBJS-$(1)): CFLAGS += $(CFLAGS-$(1)) -$(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1)) -$(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1)) +$(1)$(EXESUF): LDFLAGS += $(LDFLAGS-$(1)) +$(1)$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1)) -include $$(OBJS-$(1):.o=.d) endef -$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=)))) +$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(EXESUF)=)))) -ffprobe.o cmdutils.o : libavutil/ffversion.h - -$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF) - $(CP) $< $@ - $(STRIP) $@ - -%$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS) +$(PROGS): %$(EXESUF): %.o $(FF_DEP_LIBS) $(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS) OBJDIRS += tools @@ -129,8 +162,8 @@ GIT_LOG = $(SRC_PATH)/.git/logs/HEAD .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak .version: M=@ -libavutil/ffversion.h .version: - $(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION) +version.h .version: + $(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION) $(Q)touch .version # force version.sh to run whenever version might have changed @@ -149,12 +182,11 @@ install-progs-$(CONFIG_SHARED): install-libs install-progs: install-progs-yes $(AVPROGS) $(Q)mkdir -p "$(BINDIR)" - $(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)" + $(INSTALL) -c -m 755 $(AVPROGS) "$(BINDIR)" -install-data: $(DATA_FILES) $(EXAMPLES_FILES) - $(Q)mkdir -p "$(DATADIR)/examples" +install-data: $(DATA_FILES) + $(Q)mkdir -p "$(DATADIR)" $(INSTALL) -m 644 $(DATA_FILES) "$(DATADIR)" - $(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples" uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data @@ -165,20 +197,19 @@ uninstall-data: $(RM) -r "$(DATADIR)" clean:: - $(RM) $(ALLAVPROGS) $(ALLAVPROGS_G) + $(RM) $(ALLAVPROGS) $(RM) $(CLEANSUFFIXES) $(RM) $(CLEANSUFFIXES:%=tools/%) - $(RM) -r coverage-html $(RM) -rf coverage.info lcov distclean:: $(RM) $(DISTCLEANSUFFIXES) - $(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h + $(RM) config.* .config libavutil/avconfig.h .version version.h config: - $(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION) + $(SRC_PATH)/configure $(value LIBAV_CONFIGURATION) -check: all alltools examples testprogs fate +check: all alltools checkheaders examples testprogs fate include $(SRC_PATH)/tests/Makefile @@ -193,5 +224,5 @@ $(sort $(OBJDIRS)): # so this saves some time on slow systems. .SUFFIXES: -.PHONY: all all-yes alltools check *clean config install* +.PHONY: all all-yes alltools check *clean config examples install* .PHONY: testprogs uninstall* diff --git a/README b/README index ae007bcccf..afef671611 100644 --- a/README +++ b/README @@ -1,18 +1,12 @@ -FFmpeg README -------------- +Libav README +------------ 1) Documentation ---------------- -* Read the documentation in the doc/ directory in git. - You can also view it online at http://ffmpeg.org/documentation.html +* Read the documentation in the doc/ directory. 2) Licensing ------------ * See the LICENSE file. - -3) Build and Install --------------------- - -* See the INSTALL file. diff --git a/RELEASE b/RELEASE index bd5ee42fcc..086e7dc11f 100644 --- a/RELEASE +++ b/RELEASE @@ -1 +1 @@ -2.2.git +11_alpha1 diff --git a/arch.mak b/arch.mak index 0e866d8628..4bfc883ffa 100644 --- a/arch.mak +++ b/arch.mak @@ -4,11 +4,6 @@ OBJS-$(HAVE_ARMV8) += $(ARMV8-OBJS) $(ARMV8-OBJS-yes) OBJS-$(HAVE_VFP) += $(VFP-OBJS) $(VFP-OBJS-yes) OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes) -OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes) -OBJS-$(HAVE_MIPS32R2) += $(MIPS32R2-OBJS) $(MIPS32R2-OBJS-yes) -OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS) $(MIPSDSPR1-OBJS-yes) -OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS) $(MIPSDSPR2-OBJS-yes) - OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes) OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes) diff --git a/avconv.c b/avconv.c new file mode 100644 index 0000000000..2cee09bc22 --- /dev/null +++ b/avconv.c @@ -0,0 +1,2633 @@ +/* + * avconv main + * Copyright (c) 2000-2011 The libav developers. + * + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "config.h" +#include +#include +#include +#include +#include +#include +#include +#include + +#include "libavformat/avformat.h" +#include "libavdevice/avdevice.h" +#include "libswscale/swscale.h" +#include "libavresample/avresample.h" +#include "libavutil/opt.h" +#include "libavutil/channel_layout.h" +#include "libavutil/parseutils.h" +#include "libavutil/samplefmt.h" +#include "libavutil/fifo.h" +#include "libavutil/intreadwrite.h" +#include "libavutil/dict.h" +#include "libavutil/mathematics.h" +#include "libavutil/pixdesc.h" +#include "libavutil/avstring.h" +#include "libavutil/libm.h" +#include "libavutil/imgutils.h" +#include "libavutil/time.h" +#include "libavformat/os_support.h" + +# include "libavfilter/avfilter.h" +# include "libavfilter/buffersrc.h" +# include "libavfilter/buffersink.h" + +#if HAVE_SYS_RESOURCE_H +#include +#include +#include +#elif HAVE_GETPROCESSTIMES +#include +#endif +#if HAVE_GETPROCESSMEMORYINFO +#include +#include +#endif + +#if HAVE_SYS_SELECT_H +#include +#endif + +#if HAVE_PTHREADS +#include +#endif + +#include + +#include "avconv.h" +#include "cmdutils.h" + +#include "libavutil/avassert.h" + +const char program_name[] = "avconv"; +const int program_birth_year = 2000; + +static FILE *vstats_file; + +static int nb_frames_drop = 0; + + + +#if HAVE_PTHREADS +/* signal to input threads that they should exit; set by the main thread */ +static int transcoding_finished; +#endif + +#define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass" + +InputStream **input_streams = NULL; +int nb_input_streams = 0; +InputFile **input_files = NULL; +int nb_input_files = 0; + +OutputStream **output_streams = NULL; +int nb_output_streams = 0; +OutputFile **output_files = NULL; +int nb_output_files = 0; + +FilterGraph **filtergraphs; +int nb_filtergraphs; + +static void term_exit(void) +{ + av_log(NULL, AV_LOG_QUIET, ""); +} + +static volatile int received_sigterm = 0; +static volatile int received_nb_signals = 0; + +static void +sigterm_handler(int sig) +{ + received_sigterm = sig; + received_nb_signals++; + term_exit(); +} + +static void term_init(void) +{ + signal(SIGINT , sigterm_handler); /* Interrupt (ANSI). */ + signal(SIGTERM, sigterm_handler); /* Termination (ANSI). */ +#ifdef SIGXCPU + signal(SIGXCPU, sigterm_handler); +#endif +} + +static int decode_interrupt_cb(void *ctx) +{ + return received_nb_signals > 1; +} + +const AVIOInterruptCB int_cb = { decode_interrupt_cb, NULL }; + +static void avconv_cleanup(int ret) +{ + int i, j; + + for (i = 0; i < nb_filtergraphs; i++) { + FilterGraph *fg = filtergraphs[i]; + avfilter_graph_free(&fg->graph); + for (j = 0; j < fg->nb_inputs; j++) { + av_freep(&fg->inputs[j]->name); + av_freep(&fg->inputs[j]); + } + av_freep(&fg->inputs); + for (j = 0; j < fg->nb_outputs; j++) { + av_freep(&fg->outputs[j]->name); + av_freep(&fg->outputs[j]); + } + av_freep(&fg->outputs); + av_freep(&fg->graph_desc); + + av_freep(&filtergraphs[i]); + } + av_freep(&filtergraphs); + + /* close files */ + for (i = 0; i < nb_output_files; i++) { + OutputFile *of = output_files[i]; + AVFormatContext *s = of->ctx; + if (s && s->oformat && !(s->oformat->flags & AVFMT_NOFILE) && s->pb) + avio_close(s->pb); + avformat_free_context(s); + av_dict_free(&of->opts); + + av_freep(&output_files[i]); + } + for (i = 0; i < nb_output_streams; i++) { + OutputStream *ost = output_streams[i]; + AVBitStreamFilterContext *bsfc = ost->bitstream_filters; + while (bsfc) { + AVBitStreamFilterContext *next = bsfc->next; + av_bitstream_filter_close(bsfc); + bsfc = next; + } + ost->bitstream_filters = NULL; + av_frame_free(&ost->filtered_frame); + + av_parser_close(ost->parser); + + av_freep(&ost->forced_keyframes); + av_freep(&ost->avfilter); + av_freep(&ost->logfile_prefix); + + av_freep(&output_streams[i]); + } + for (i = 0; i < nb_input_files; i++) { + avformat_close_input(&input_files[i]->ctx); + av_freep(&input_files[i]); + } + for (i = 0; i < nb_input_streams; i++) { + InputStream *ist = input_streams[i]; + + av_frame_free(&ist->decoded_frame); + av_frame_free(&ist->filter_frame); + av_dict_free(&ist->decoder_opts); + av_freep(&ist->filters); + av_freep(&ist->hwaccel_device); + + av_freep(&input_streams[i]); + } + + if (vstats_file) + fclose(vstats_file); + av_free(vstats_filename); + + av_freep(&input_streams); + av_freep(&input_files); + av_freep(&output_streams); + av_freep(&output_files); + + uninit_opts(); + + avformat_network_deinit(); + + if (received_sigterm) { + av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n", + (int) received_sigterm); + exit (255); + } +} + +void assert_avoptions(AVDictionary *m) +{ + AVDictionaryEntry *t; + if ((t = av_dict_get(m, "", NULL, AV_DICT_IGNORE_SUFFIX))) { + av_log(NULL, AV_LOG_FATAL, "Option %s not found.\n", t->key); + exit_program(1); + } +} + +static void abort_codec_experimental(AVCodec *c, int encoder) +{ + const char *codec_string = encoder ? "encoder" : "decoder"; + AVCodec *codec; + av_log(NULL, AV_LOG_FATAL, "%s '%s' is experimental and might produce bad " + "results.\nAdd '-strict experimental' if you want to use it.\n", + codec_string, c->name); + codec = encoder ? avcodec_find_encoder(c->id) : avcodec_find_decoder(c->id); + if (!(codec->capabilities & CODEC_CAP_EXPERIMENTAL)) + av_log(NULL, AV_LOG_FATAL, "Or use the non experimental %s '%s'.\n", + codec_string, codec->name); + exit_program(1); +} + +/* + * Update the requested input sample format based on the output sample format. + * This is currently only used to request float output from decoders which + * support multiple sample formats, one of which is AV_SAMPLE_FMT_FLT. + * Ideally this will be removed in the future when decoders do not do format + * conversion and only output in their native format. + */ +static void update_sample_fmt(AVCodecContext *dec, AVCodec *dec_codec, + AVCodecContext *enc) +{ + /* if sample formats match or a decoder sample format has already been + requested, just return */ + if (enc->sample_fmt == dec->sample_fmt || + dec->request_sample_fmt > AV_SAMPLE_FMT_NONE) + return; + + /* if decoder supports more than one output format */ + if (dec_codec && dec_codec->sample_fmts && + dec_codec->sample_fmts[0] != AV_SAMPLE_FMT_NONE && + dec_codec->sample_fmts[1] != AV_SAMPLE_FMT_NONE) { + const enum AVSampleFormat *p; + int min_dec = INT_MAX, min_inc = INT_MAX; + enum AVSampleFormat dec_fmt = AV_SAMPLE_FMT_NONE; + enum AVSampleFormat inc_fmt = AV_SAMPLE_FMT_NONE; + + /* find a matching sample format in the encoder */ + for (p = dec_codec->sample_fmts; *p != AV_SAMPLE_FMT_NONE; p++) { + if (*p == enc->sample_fmt) { + dec->request_sample_fmt = *p; + return; + } else { + enum AVSampleFormat dfmt = av_get_packed_sample_fmt(*p); + enum AVSampleFormat efmt = av_get_packed_sample_fmt(enc->sample_fmt); + int fmt_diff = 32 * abs(dfmt - efmt); + if (av_sample_fmt_is_planar(*p) != + av_sample_fmt_is_planar(enc->sample_fmt)) + fmt_diff++; + if (dfmt == efmt) { + min_inc = fmt_diff; + inc_fmt = *p; + } else if (dfmt > efmt) { + if (fmt_diff < min_inc) { + min_inc = fmt_diff; + inc_fmt = *p; + } + } else { + if (fmt_diff < min_dec) { + min_dec = fmt_diff; + dec_fmt = *p; + } + } + } + } + + /* if none match, provide the one that matches quality closest */ + dec->request_sample_fmt = min_inc != INT_MAX ? inc_fmt : dec_fmt; + } +} + +static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost) +{ + AVBitStreamFilterContext *bsfc = ost->bitstream_filters; + AVCodecContext *avctx = ost->st->codec; + int ret; + + /* + * Audio encoders may split the packets -- #frames in != #packets out. + * But there is no reordering, so we can limit the number of output packets + * by simply dropping them here. + * Counting encoded video frames needs to be done separately because of + * reordering, see do_video_out() + */ + if (!(avctx->codec_type == AVMEDIA_TYPE_VIDEO && avctx->codec)) { + if (ost->frame_number >= ost->max_frames) { + av_free_packet(pkt); + return; + } + ost->frame_number++; + } + + while (bsfc) { + AVPacket new_pkt = *pkt; + int a = av_bitstream_filter_filter(bsfc, avctx, NULL, + &new_pkt.data, &new_pkt.size, + pkt->data, pkt->size, + pkt->flags & AV_PKT_FLAG_KEY); + if (a > 0) { + av_free_packet(pkt); + new_pkt.buf = av_buffer_create(new_pkt.data, new_pkt.size, + av_buffer_default_free, NULL, 0); + if (!new_pkt.buf) + exit_program(1); + } else if (a < 0) { + av_log(NULL, AV_LOG_ERROR, "%s failed for stream %d, codec %s", + bsfc->filter->name, pkt->stream_index, + avctx->codec ? avctx->codec->name : "copy"); + print_error("", a); + if (exit_on_error) + exit_program(1); + } + *pkt = new_pkt; + + bsfc = bsfc->next; + } + + if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS) && + ost->last_mux_dts != AV_NOPTS_VALUE && + pkt->dts < ost->last_mux_dts + !(s->oformat->flags & AVFMT_TS_NONSTRICT)) { + av_log(NULL, AV_LOG_WARNING, "Non-monotonous DTS in output stream " + "%d:%d; previous: %"PRId64", current: %"PRId64"; ", + ost->file_index, ost->st->index, ost->last_mux_dts, pkt->dts); + if (exit_on_error) { + av_log(NULL, AV_LOG_FATAL, "aborting.\n"); + exit_program(1); + } + av_log(NULL, AV_LOG_WARNING, "changing to %"PRId64". This may result " + "in incorrect timestamps in the output file.\n", + ost->last_mux_dts + 1); + pkt->dts = ost->last_mux_dts + 1; + if (pkt->pts != AV_NOPTS_VALUE) + pkt->pts = FFMAX(pkt->pts, pkt->dts); + } + ost->last_mux_dts = pkt->dts; + + ost->data_size += pkt->size; + ost->packets_written++; + + pkt->stream_index = ost->index; + ret = av_interleaved_write_frame(s, pkt); + if (ret < 0) { + print_error("av_interleaved_write_frame()", ret); + exit_program(1); + } +} + +static int check_recording_time(OutputStream *ost) +{ + OutputFile *of = output_files[ost->file_index]; + + if (of->recording_time != INT64_MAX && + av_compare_ts(ost->sync_opts - ost->first_pts, ost->st->codec->time_base, of->recording_time, + AV_TIME_BASE_Q) >= 0) { + ost->finished = 1; + return 0; + } + return 1; +} + +static void do_audio_out(AVFormatContext *s, OutputStream *ost, + AVFrame *frame) +{ + AVCodecContext *enc = ost->st->codec; + AVPacket pkt; + int got_packet = 0; + + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + if (frame->pts == AV_NOPTS_VALUE || audio_sync_method < 0) + frame->pts = ost->sync_opts; + ost->sync_opts = frame->pts + frame->nb_samples; + + ost->samples_encoded += frame->nb_samples; + ost->frames_encoded++; + + if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) { + av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n"); + exit_program(1); + } + + if (got_packet) { + if (pkt.pts != AV_NOPTS_VALUE) + pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base); + if (pkt.dts != AV_NOPTS_VALUE) + pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base); + if (pkt.duration > 0) + pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base); + + write_frame(s, &pkt, ost); + } +} + +static void do_subtitle_out(AVFormatContext *s, + OutputStream *ost, + InputStream *ist, + AVSubtitle *sub, + int64_t pts) +{ + static uint8_t *subtitle_out = NULL; + int subtitle_out_max_size = 1024 * 1024; + int subtitle_out_size, nb, i; + AVCodecContext *enc; + AVPacket pkt; + + if (pts == AV_NOPTS_VALUE) { + av_log(NULL, AV_LOG_ERROR, "Subtitle packets must have a pts\n"); + if (exit_on_error) + exit_program(1); + return; + } + + enc = ost->st->codec; + + if (!subtitle_out) { + subtitle_out = av_malloc(subtitle_out_max_size); + } + + /* Note: DVB subtitle need one packet to draw them and one other + packet to clear them */ + /* XXX: signal it in the codec context ? */ + if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE) + nb = 2; + else + nb = 1; + + for (i = 0; i < nb; i++) { + ost->sync_opts = av_rescale_q(pts, ist->st->time_base, enc->time_base); + if (!check_recording_time(ost)) + return; + + sub->pts = av_rescale_q(pts, ist->st->time_base, AV_TIME_BASE_Q); + // start_display_time is required to be 0 + sub->pts += av_rescale_q(sub->start_display_time, (AVRational){ 1, 1000 }, AV_TIME_BASE_Q); + sub->end_display_time -= sub->start_display_time; + sub->start_display_time = 0; + + ost->frames_encoded++; + + subtitle_out_size = avcodec_encode_subtitle(enc, subtitle_out, + subtitle_out_max_size, sub); + if (subtitle_out_size < 0) { + av_log(NULL, AV_LOG_FATAL, "Subtitle encoding failed\n"); + exit_program(1); + } + + av_init_packet(&pkt); + pkt.data = subtitle_out; + pkt.size = subtitle_out_size; + pkt.pts = av_rescale_q(sub->pts, AV_TIME_BASE_Q, ost->st->time_base); + if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE) { + /* XXX: the pts correction is handled here. Maybe handling + it in the codec would be better */ + if (i == 0) + pkt.pts += 90 * sub->start_display_time; + else + pkt.pts += 90 * sub->end_display_time; + } + write_frame(s, &pkt, ost); + } +} + +static void do_video_out(AVFormatContext *s, + OutputStream *ost, + AVFrame *in_picture, + int *frame_size) +{ + int ret, format_video_sync; + AVPacket pkt; + AVCodecContext *enc = ost->st->codec; + + *frame_size = 0; + + format_video_sync = video_sync_method; + if (format_video_sync == VSYNC_AUTO) + format_video_sync = (s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : + (s->oformat->flags & AVFMT_VARIABLE_FPS) ? VSYNC_VFR : VSYNC_CFR; + if (format_video_sync != VSYNC_PASSTHROUGH && + ost->frame_number && + in_picture->pts != AV_NOPTS_VALUE && + in_picture->pts < ost->sync_opts) { + nb_frames_drop++; + av_log(NULL, AV_LOG_WARNING, + "*** dropping frame %d from stream %d at ts %"PRId64"\n", + ost->frame_number, ost->st->index, in_picture->pts); + return; + } + + if (in_picture->pts == AV_NOPTS_VALUE) + in_picture->pts = ost->sync_opts; + ost->sync_opts = in_picture->pts; + + + if (!ost->frame_number) + ost->first_pts = in_picture->pts; + + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + if (ost->frame_number >= ost->max_frames) + return; + + if (s->oformat->flags & AVFMT_RAWPICTURE && + enc->codec->id == AV_CODEC_ID_RAWVIDEO) { + /* raw pictures are written as AVPicture structure to + avoid any copies. We support temporarily the older + method. */ + enc->coded_frame->interlaced_frame = in_picture->interlaced_frame; + enc->coded_frame->top_field_first = in_picture->top_field_first; + pkt.data = (uint8_t *)in_picture; + pkt.size = sizeof(AVPicture); + pkt.pts = av_rescale_q(in_picture->pts, enc->time_base, ost->st->time_base); + pkt.flags |= AV_PKT_FLAG_KEY; + + write_frame(s, &pkt, ost); + } else { + int got_packet; + + if (ost->st->codec->flags & (CODEC_FLAG_INTERLACED_DCT|CODEC_FLAG_INTERLACED_ME) && + ost->top_field_first >= 0) + in_picture->top_field_first = !!ost->top_field_first; + + in_picture->quality = ost->st->codec->global_quality; + if (!enc->me_threshold) + in_picture->pict_type = 0; + if (ost->forced_kf_index < ost->forced_kf_count && + in_picture->pts >= ost->forced_kf_pts[ost->forced_kf_index]) { + in_picture->pict_type = AV_PICTURE_TYPE_I; + ost->forced_kf_index++; + } + + ost->frames_encoded++; + + ret = avcodec_encode_video2(enc, &pkt, in_picture, &got_packet); + if (ret < 0) { + av_log(NULL, AV_LOG_FATAL, "Video encoding failed\n"); + exit_program(1); + } + + if (got_packet) { + if (pkt.pts != AV_NOPTS_VALUE) + pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base); + if (pkt.dts != AV_NOPTS_VALUE) + pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base); + + write_frame(s, &pkt, ost); + *frame_size = pkt.size; + + /* if two pass, output log */ + if (ost->logfile && enc->stats_out) { + fprintf(ost->logfile, "%s", enc->stats_out); + } + } + } + ost->sync_opts++; + /* + * For video, number of frames in == number of packets out. + * But there may be reordering, so we can't throw away frames on encoder + * flush, we need to limit them here, before they go into encoder. + */ + ost->frame_number++; +} + +static double psnr(double d) +{ + return -10.0 * log(d) / log(10.0); +} + +static void do_video_stats(OutputStream *ost, int frame_size) +{ + AVCodecContext *enc; + int frame_number; + double ti1, bitrate, avg_bitrate; + + /* this is executed just the first time do_video_stats is called */ + if (!vstats_file) { + vstats_file = fopen(vstats_filename, "w"); + if (!vstats_file) { + perror("fopen"); + exit_program(1); + } + } + + enc = ost->st->codec; + if (enc->codec_type == AVMEDIA_TYPE_VIDEO) { + frame_number = ost->frame_number; + fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number, enc->coded_frame->quality / (float)FF_QP2LAMBDA); + if (enc->flags&CODEC_FLAG_PSNR) + fprintf(vstats_file, "PSNR= %6.2f ", psnr(enc->coded_frame->error[0] / (enc->width * enc->height * 255.0 * 255.0))); + + fprintf(vstats_file,"f_size= %6d ", frame_size); + /* compute pts value */ + ti1 = ost->sync_opts * av_q2d(enc->time_base); + if (ti1 < 0.01) + ti1 = 0.01; + + bitrate = (frame_size * 8) / av_q2d(enc->time_base) / 1000.0; + avg_bitrate = (double)(ost->data_size * 8) / ti1 / 1000.0; + fprintf(vstats_file, "s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s ", + (double)ost->data_size / 1024, ti1, bitrate, avg_bitrate); + fprintf(vstats_file, "type= %c\n", av_get_picture_type_char(enc->coded_frame->pict_type)); + } +} + +/* + * Read one frame for lavfi output for ost and encode it. + */ +static int poll_filter(OutputStream *ost) +{ + OutputFile *of = output_files[ost->file_index]; + AVFrame *filtered_frame = NULL; + int frame_size, ret; + + if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) { + return AVERROR(ENOMEM); + } + filtered_frame = ost->filtered_frame; + + if (ost->enc->type == AVMEDIA_TYPE_AUDIO && + !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) + ret = av_buffersink_get_samples(ost->filter->filter, filtered_frame, + ost->st->codec->frame_size); + else + ret = av_buffersink_get_frame(ost->filter->filter, filtered_frame); + + if (ret < 0) + return ret; + + if (filtered_frame->pts != AV_NOPTS_VALUE) { + int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time; + filtered_frame->pts = av_rescale_q(filtered_frame->pts, + ost->filter->filter->inputs[0]->time_base, + ost->st->codec->time_base) - + av_rescale_q(start_time, + AV_TIME_BASE_Q, + ost->st->codec->time_base); + } + + switch (ost->filter->filter->inputs[0]->type) { + case AVMEDIA_TYPE_VIDEO: + if (!ost->frame_aspect_ratio) + ost->st->codec->sample_aspect_ratio = filtered_frame->sample_aspect_ratio; + + do_video_out(of->ctx, ost, filtered_frame, &frame_size); + if (vstats_filename && frame_size) + do_video_stats(ost, frame_size); + break; + case AVMEDIA_TYPE_AUDIO: + do_audio_out(of->ctx, ost, filtered_frame); + break; + default: + // TODO support subtitle filters + av_assert0(0); + } + + av_frame_unref(filtered_frame); + + return 0; +} + +/* + * Read as many frames from possible from lavfi and encode them. + * + * Always read from the active stream with the lowest timestamp. If no frames + * are available for it then return EAGAIN and wait for more input. This way we + * can use lavfi sources that generate unlimited amount of frames without memory + * usage exploding. + */ +static int poll_filters(void) +{ + int i, j, ret = 0; + + while (ret >= 0 && !received_sigterm) { + OutputStream *ost = NULL; + int64_t min_pts = INT64_MAX; + + /* choose output stream with the lowest timestamp */ + for (i = 0; i < nb_output_streams; i++) { + int64_t pts = output_streams[i]->sync_opts; + + if (!output_streams[i]->filter || output_streams[i]->finished) + continue; + + pts = av_rescale_q(pts, output_streams[i]->st->codec->time_base, + AV_TIME_BASE_Q); + if (pts < min_pts) { + min_pts = pts; + ost = output_streams[i]; + } + } + + if (!ost) + break; + + ret = poll_filter(ost); + + if (ret == AVERROR_EOF) { + OutputFile *of = output_files[ost->file_index]; + + ost->finished = 1; + + if (of->shortest) { + for (j = 0; j < of->ctx->nb_streams; j++) + output_streams[of->ost_index + j]->finished = 1; + } + + ret = 0; + } else if (ret == AVERROR(EAGAIN)) + return 0; + } + + return ret; +} + +static void print_final_stats(int64_t total_size) +{ + uint64_t video_size = 0, audio_size = 0, extra_size = 0, other_size = 0; + uint64_t data_size = 0; + float percent = -1.0; + int i, j; + + for (i = 0; i < nb_output_streams; i++) { + OutputStream *ost = output_streams[i]; + switch (ost->st->codec->codec_type) { + case AVMEDIA_TYPE_VIDEO: video_size += ost->data_size; break; + case AVMEDIA_TYPE_AUDIO: audio_size += ost->data_size; break; + default: other_size += ost->data_size; break; + } + extra_size += ost->st->codec->extradata_size; + data_size += ost->data_size; + } + + if (data_size && total_size >= data_size) + percent = 100.0 * (total_size - data_size) / data_size; + + av_log(NULL, AV_LOG_INFO, "\n"); + av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB other streams:%1.0fkB global headers:%1.0fkB muxing overhead: ", + video_size / 1024.0, + audio_size / 1024.0, + other_size / 1024.0, + extra_size / 1024.0); + if (percent >= 0.0) + av_log(NULL, AV_LOG_INFO, "%f%%", percent); + else + av_log(NULL, AV_LOG_INFO, "unknown"); + av_log(NULL, AV_LOG_INFO, "\n"); + + /* print verbose per-stream stats */ + for (i = 0; i < nb_input_files; i++) { + InputFile *f = input_files[i]; + uint64_t total_packets = 0, total_size = 0; + + av_log(NULL, AV_LOG_VERBOSE, "Input file #%d (%s):\n", + i, f->ctx->filename); + + for (j = 0; j < f->nb_streams; j++) { + InputStream *ist = input_streams[f->ist_index + j]; + enum AVMediaType type = ist->st->codec->codec_type; + + total_size += ist->data_size; + total_packets += ist->nb_packets; + + av_log(NULL, AV_LOG_VERBOSE, " Input stream #%d:%d (%s): ", + i, j, media_type_string(type)); + av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" packets read (%"PRIu64" bytes); ", + ist->nb_packets, ist->data_size); + + if (ist->decoding_needed) { + av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" frames decoded", + ist->frames_decoded); + if (type == AVMEDIA_TYPE_AUDIO) + av_log(NULL, AV_LOG_VERBOSE, " (%"PRIu64" samples)", ist->samples_decoded); + av_log(NULL, AV_LOG_VERBOSE, "; "); + } + + av_log(NULL, AV_LOG_VERBOSE, "\n"); + } + + av_log(NULL, AV_LOG_VERBOSE, " Total: %"PRIu64" packets (%"PRIu64" bytes) demuxed\n", + total_packets, total_size); + } + + for (i = 0; i < nb_output_files; i++) { + OutputFile *of = output_files[i]; + uint64_t total_packets = 0, total_size = 0; + + av_log(NULL, AV_LOG_VERBOSE, "Output file #%d (%s):\n", + i, of->ctx->filename); + + for (j = 0; j < of->ctx->nb_streams; j++) { + OutputStream *ost = output_streams[of->ost_index + j]; + enum AVMediaType type = ost->st->codec->codec_type; + + total_size += ost->data_size; + total_packets += ost->packets_written; + + av_log(NULL, AV_LOG_VERBOSE, " Output stream #%d:%d (%s): ", + i, j, media_type_string(type)); + if (ost->encoding_needed) { + av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" frames encoded", + ost->frames_encoded); + if (type == AVMEDIA_TYPE_AUDIO) + av_log(NULL, AV_LOG_VERBOSE, " (%"PRIu64" samples)", ost->samples_encoded); + av_log(NULL, AV_LOG_VERBOSE, "; "); + } + + av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" packets muxed (%"PRIu64" bytes); ", + ost->packets_written, ost->data_size); + + av_log(NULL, AV_LOG_VERBOSE, "\n"); + } + + av_log(NULL, AV_LOG_VERBOSE, " Total: %"PRIu64" packets (%"PRIu64" bytes) muxed\n", + total_packets, total_size); + } +} + +static void print_report(int is_last_report, int64_t timer_start) +{ + char buf[1024]; + OutputStream *ost; + AVFormatContext *oc; + int64_t total_size; + AVCodecContext *enc; + int frame_number, vid, i; + double bitrate, ti1, pts; + static int64_t last_time = -1; + static int qp_histogram[52]; + + if (!print_stats && !is_last_report) + return; + + if (!is_last_report) { + int64_t cur_time; + /* display the report every 0.5 seconds */ + cur_time = av_gettime(); + if (last_time == -1) { + last_time = cur_time; + return; + } + if ((cur_time - last_time) < 500000) + return; + last_time = cur_time; + } + + + oc = output_files[0]->ctx; + + total_size = avio_size(oc->pb); + if (total_size <= 0) // FIXME improve avio_size() so it works with non seekable output too + total_size = avio_tell(oc->pb); + if (total_size < 0) { + char errbuf[128]; + av_strerror(total_size, errbuf, sizeof(errbuf)); + av_log(NULL, AV_LOG_VERBOSE, "Bitrate not available, " + "avio_tell() failed: %s\n", errbuf); + total_size = 0; + } + + buf[0] = '\0'; + ti1 = 1e10; + vid = 0; + for (i = 0; i < nb_output_streams; i++) { + float q = -1; + ost = output_streams[i]; + enc = ost->st->codec; + if (!ost->stream_copy && enc->coded_frame) + q = enc->coded_frame->quality / (float)FF_QP2LAMBDA; + if (vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) { + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "q=%2.1f ", q); + } + if (!vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) { + float t = (av_gettime() - timer_start) / 1000000.0; + + frame_number = ost->frame_number; + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "frame=%5d fps=%3d q=%3.1f ", + frame_number, (t > 1) ? (int)(frame_number / t + 0.5) : 0, q); + if (is_last_report) + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "L"); + if (qp_hist) { + int j; + int qp = lrintf(q); + if (qp >= 0 && qp < FF_ARRAY_ELEMS(qp_histogram)) + qp_histogram[qp]++; + for (j = 0; j < 32; j++) + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%X", (int)lrintf(log2(qp_histogram[j] + 1))); + } + if (enc->flags&CODEC_FLAG_PSNR) { + int j; + double error, error_sum = 0; + double scale, scale_sum = 0; + char type[3] = { 'Y','U','V' }; + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "PSNR="); + for (j = 0; j < 3; j++) { + if (is_last_report) { + error = enc->error[j]; + scale = enc->width * enc->height * 255.0 * 255.0 * frame_number; + } else { + error = enc->coded_frame->error[j]; + scale = enc->width * enc->height * 255.0 * 255.0; + } + if (j) + scale /= 4; + error_sum += error; + scale_sum += scale; + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%c:%2.2f ", type[j], psnr(error / scale)); + } + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "*:%2.2f ", psnr(error_sum / scale_sum)); + } + vid = 1; + } + /* compute min output value */ + pts = (double)ost->last_mux_dts * av_q2d(ost->st->time_base); + if ((pts < ti1) && (pts > 0)) + ti1 = pts; + } + if (ti1 < 0.01) + ti1 = 0.01; + + bitrate = (double)(total_size * 8) / ti1 / 1000.0; + + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), + "size=%8.0fkB time=%0.2f bitrate=%6.1fkbits/s", + (double)total_size / 1024, ti1, bitrate); + + if (nb_frames_drop) + snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), " drop=%d", + nb_frames_drop); + + av_log(NULL, AV_LOG_INFO, "%s \r", buf); + + fflush(stderr); + + if (is_last_report) + print_final_stats(total_size); + +} + +static void flush_encoders(void) +{ + int i, ret; + + for (i = 0; i < nb_output_streams; i++) { + OutputStream *ost = output_streams[i]; + AVCodecContext *enc = ost->st->codec; + AVFormatContext *os = output_files[ost->file_index]->ctx; + int stop_encoding = 0; + + if (!ost->encoding_needed) + continue; + + if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <= 1) + continue; + if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE) && enc->codec->id == AV_CODEC_ID_RAWVIDEO) + continue; + + for (;;) { + int (*encode)(AVCodecContext*, AVPacket*, const AVFrame*, int*) = NULL; + const char *desc; + + switch (ost->st->codec->codec_type) { + case AVMEDIA_TYPE_AUDIO: + encode = avcodec_encode_audio2; + desc = "Audio"; + break; + case AVMEDIA_TYPE_VIDEO: + encode = avcodec_encode_video2; + desc = "Video"; + break; + default: + stop_encoding = 1; + } + + if (encode) { + AVPacket pkt; + int got_packet; + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + ret = encode(enc, &pkt, NULL, &got_packet); + if (ret < 0) { + av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc); + exit_program(1); + } + if (ost->logfile && enc->stats_out) { + fprintf(ost->logfile, "%s", enc->stats_out); + } + if (!got_packet) { + stop_encoding = 1; + break; + } + if (pkt.pts != AV_NOPTS_VALUE) + pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base); + if (pkt.dts != AV_NOPTS_VALUE) + pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base); + if (pkt.duration > 0) + pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base); + write_frame(os, &pkt, ost); + } + + if (stop_encoding) + break; + } + } +} + +/* + * Check whether a packet from ist should be written into ost at this time + */ +static int check_output_constraints(InputStream *ist, OutputStream *ost) +{ + OutputFile *of = output_files[ost->file_index]; + int ist_index = input_files[ist->file_index]->ist_index + ist->st->index; + + if (ost->source_index != ist_index) + return 0; + + if (of->start_time != AV_NOPTS_VALUE && ist->last_dts < of->start_time) + return 0; + + return 1; +} + +static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *pkt) +{ + OutputFile *of = output_files[ost->file_index]; + InputFile *f = input_files [ist->file_index]; + int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time; + int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->st->time_base); + AVPacket opkt; + + av_init_packet(&opkt); + + if ((!ost->frame_number && !(pkt->flags & AV_PKT_FLAG_KEY)) && + !ost->copy_initial_nonkeyframes) + return; + + if (of->recording_time != INT64_MAX && + ist->last_dts >= of->recording_time + start_time) { + ost->finished = 1; + return; + } + + if (f->recording_time != INT64_MAX) { + start_time = f->ctx->start_time; + if (f->start_time != AV_NOPTS_VALUE) + start_time += f->start_time; + if (ist->last_dts >= f->recording_time + start_time) { + ost->finished = 1; + return; + } + } + + /* force the input stream PTS */ + if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) + ost->sync_opts++; + + if (pkt->pts != AV_NOPTS_VALUE) + opkt.pts = av_rescale_q(pkt->pts, ist->st->time_base, ost->st->time_base) - ost_tb_start_time; + else + opkt.pts = AV_NOPTS_VALUE; + + if (pkt->dts == AV_NOPTS_VALUE) + opkt.dts = av_rescale_q(ist->last_dts, AV_TIME_BASE_Q, ost->st->time_base); + else + opkt.dts = av_rescale_q(pkt->dts, ist->st->time_base, ost->st->time_base); + opkt.dts -= ost_tb_start_time; + + opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->st->time_base); + opkt.flags = pkt->flags; + + // FIXME remove the following 2 lines they shall be replaced by the bitstream filters + if ( ost->st->codec->codec_id != AV_CODEC_ID_H264 + && ost->st->codec->codec_id != AV_CODEC_ID_MPEG1VIDEO + && ost->st->codec->codec_id != AV_CODEC_ID_MPEG2VIDEO + && ost->st->codec->codec_id != AV_CODEC_ID_VC1 + ) { + if (av_parser_change(ost->parser, ost->st->codec, + &opkt.data, &opkt.size, + pkt->data, pkt->size, + pkt->flags & AV_PKT_FLAG_KEY)) { + opkt.buf = av_buffer_create(opkt.data, opkt.size, av_buffer_default_free, NULL, 0); + if (!opkt.buf) + exit_program(1); + } + } else { + opkt.data = pkt->data; + opkt.size = pkt->size; + } + + write_frame(of->ctx, &opkt, ost); +} + +int guess_input_channel_layout(InputStream *ist) +{ + AVCodecContext *dec = ist->st->codec; + + if (!dec->channel_layout) { + char layout_name[256]; + + dec->channel_layout = av_get_default_channel_layout(dec->channels); + if (!dec->channel_layout) + return 0; + av_get_channel_layout_string(layout_name, sizeof(layout_name), + dec->channels, dec->channel_layout); + av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Input Stream " + "#%d.%d : %s\n", ist->file_index, ist->st->index, layout_name); + } + return 1; +} + +static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output) +{ + AVFrame *decoded_frame, *f; + AVCodecContext *avctx = ist->st->codec; + int i, ret, err = 0, resample_changed; + + if (!ist->decoded_frame && !(ist->decoded_frame = av_frame_alloc())) + return AVERROR(ENOMEM); + if (!ist->filter_frame && !(ist->filter_frame = av_frame_alloc())) + return AVERROR(ENOMEM); + decoded_frame = ist->decoded_frame; + + ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt); + if (!*got_output || ret < 0) { + if (!pkt->size) { + for (i = 0; i < ist->nb_filters; i++) + av_buffersrc_add_frame(ist->filters[i]->filter, NULL); + } + return ret; + } + + ist->samples_decoded += decoded_frame->nb_samples; + ist->frames_decoded++; + + /* if the decoder provides a pts, use it instead of the last packet pts. + the decoder could be delaying output by a packet or more. */ + if (decoded_frame->pts != AV_NOPTS_VALUE) + ist->next_dts = decoded_frame->pts; + else if (pkt->pts != AV_NOPTS_VALUE) + decoded_frame->pts = pkt->pts; + pkt->pts = AV_NOPTS_VALUE; + + resample_changed = ist->resample_sample_fmt != decoded_frame->format || + ist->resample_channels != avctx->channels || + ist->resample_channel_layout != decoded_frame->channel_layout || + ist->resample_sample_rate != decoded_frame->sample_rate; + if (resample_changed) { + char layout1[64], layout2[64]; + + if (!guess_input_channel_layout(ist)) { + av_log(NULL, AV_LOG_FATAL, "Unable to find default channel " + "layout for Input Stream #%d.%d\n", ist->file_index, + ist->st->index); + exit_program(1); + } + decoded_frame->channel_layout = avctx->channel_layout; + + av_get_channel_layout_string(layout1, sizeof(layout1), ist->resample_channels, + ist->resample_channel_layout); + av_get_channel_layout_string(layout2, sizeof(layout2), avctx->channels, + decoded_frame->channel_layout); + + av_log(NULL, AV_LOG_INFO, + "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d chl:%s to rate:%d fmt:%s ch:%d chl:%s\n", + ist->file_index, ist->st->index, + ist->resample_sample_rate, av_get_sample_fmt_name(ist->resample_sample_fmt), + ist->resample_channels, layout1, + decoded_frame->sample_rate, av_get_sample_fmt_name(decoded_frame->format), + avctx->channels, layout2); + + ist->resample_sample_fmt = decoded_frame->format; + ist->resample_sample_rate = decoded_frame->sample_rate; + ist->resample_channel_layout = decoded_frame->channel_layout; + ist->resample_channels = avctx->channels; + + for (i = 0; i < nb_filtergraphs; i++) + if (ist_in_filtergraph(filtergraphs[i], ist) && + configure_filtergraph(filtergraphs[i]) < 0) { + av_log(NULL, AV_LOG_FATAL, "Error reinitializing filters!\n"); + exit_program(1); + } + } + + if (decoded_frame->pts != AV_NOPTS_VALUE) + decoded_frame->pts = av_rescale_q(decoded_frame->pts, + ist->st->time_base, + (AVRational){1, ist->st->codec->sample_rate}); + for (i = 0; i < ist->nb_filters; i++) { + if (i < ist->nb_filters - 1) { + f = ist->filter_frame; + err = av_frame_ref(f, decoded_frame); + if (err < 0) + break; + } else + f = decoded_frame; + + err = av_buffersrc_add_frame(ist->filters[i]->filter, f); + if (err < 0) + break; + } + + av_frame_unref(ist->filter_frame); + av_frame_unref(decoded_frame); + return err < 0 ? err : ret; +} + +static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output) +{ + AVFrame *decoded_frame, *f; + int i, ret = 0, err = 0, resample_changed; + + if (!ist->decoded_frame && !(ist->decoded_frame = av_frame_alloc())) + return AVERROR(ENOMEM); + if (!ist->filter_frame && !(ist->filter_frame = av_frame_alloc())) + return AVERROR(ENOMEM); + decoded_frame = ist->decoded_frame; + + ret = avcodec_decode_video2(ist->st->codec, + decoded_frame, got_output, pkt); + if (!*got_output || ret < 0) { + if (!pkt->size) { + for (i = 0; i < ist->nb_filters; i++) + av_buffersrc_add_frame(ist->filters[i]->filter, NULL); + } + return ret; + } + + ist->frames_decoded++; + + if (ist->hwaccel_retrieve_data && decoded_frame->format == ist->hwaccel_pix_fmt) { + err = ist->hwaccel_retrieve_data(ist->st->codec, decoded_frame); + if (err < 0) + goto fail; + } + ist->hwaccel_retrieved_pix_fmt = decoded_frame->format; + + decoded_frame->pts = guess_correct_pts(&ist->pts_ctx, decoded_frame->pkt_pts, + decoded_frame->pkt_dts); + pkt->size = 0; + + if (ist->st->sample_aspect_ratio.num) + decoded_frame->sample_aspect_ratio = ist->st->sample_aspect_ratio; + + resample_changed = ist->resample_width != decoded_frame->width || + ist->resample_height != decoded_frame->height || + ist->resample_pix_fmt != decoded_frame->format; + if (resample_changed) { + av_log(NULL, AV_LOG_INFO, + "Input stream #%d:%d frame changed from size:%dx%d fmt:%s to size:%dx%d fmt:%s\n", + ist->file_index, ist->st->index, + ist->resample_width, ist->resample_height, av_get_pix_fmt_name(ist->resample_pix_fmt), + decoded_frame->width, decoded_frame->height, av_get_pix_fmt_name(decoded_frame->format)); + + ret = poll_filters(); + if (ret < 0 && (ret != AVERROR_EOF && ret != AVERROR(EAGAIN))) + av_log(NULL, AV_LOG_ERROR, "Error while filtering.\n"); + + ist->resample_width = decoded_frame->width; + ist->resample_height = decoded_frame->height; + ist->resample_pix_fmt = decoded_frame->format; + + for (i = 0; i < nb_filtergraphs; i++) + if (ist_in_filtergraph(filtergraphs[i], ist) && + configure_filtergraph(filtergraphs[i]) < 0) { + av_log(NULL, AV_LOG_FATAL, "Error reinitializing filters!\n"); + exit_program(1); + } + } + + for (i = 0; i < ist->nb_filters; i++) { + if (i < ist->nb_filters - 1) { + f = ist->filter_frame; + err = av_frame_ref(f, decoded_frame); + if (err < 0) + break; + } else + f = decoded_frame; + + err = av_buffersrc_add_frame(ist->filters[i]->filter, f); + if (err < 0) + break; + } + +fail: + av_frame_unref(ist->filter_frame); + av_frame_unref(decoded_frame); + return err < 0 ? err : ret; +} + +static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output) +{ + AVSubtitle subtitle; + int i, ret = avcodec_decode_subtitle2(ist->st->codec, + &subtitle, got_output, pkt); + if (ret < 0) + return ret; + if (!*got_output) + return ret; + + ist->frames_decoded++; + + for (i = 0; i < nb_output_streams; i++) { + OutputStream *ost = output_streams[i]; + + if (!check_output_constraints(ist, ost) || !ost->encoding_needed) + continue; + + do_subtitle_out(output_files[ost->file_index]->ctx, ost, ist, &subtitle, pkt->pts); + } + + avsubtitle_free(&subtitle); + return ret; +} + +/* pkt = NULL means EOF (needed to flush decoder buffers) */ +static int output_packet(InputStream *ist, const AVPacket *pkt) +{ + int i; + int got_output; + AVPacket avpkt; + + if (ist->next_dts == AV_NOPTS_VALUE) + ist->next_dts = ist->last_dts; + + if (pkt == NULL) { + /* EOF handling */ + av_init_packet(&avpkt); + avpkt.data = NULL; + avpkt.size = 0; + goto handle_eof; + } else { + avpkt = *pkt; + } + + if (pkt->dts != AV_NOPTS_VALUE) + ist->next_dts = ist->last_dts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q); + + // while we have more to decode or while the decoder did output something on EOF + while (ist->decoding_needed && (avpkt.size > 0 || (!pkt && got_output))) { + int ret = 0; + handle_eof: + + ist->last_dts = ist->next_dts; + + if (avpkt.size && avpkt.size != pkt->size && + !(ist->dec->capabilities & CODEC_CAP_SUBFRAMES)) { + av_log(NULL, ist->showed_multi_packet_warning ? AV_LOG_VERBOSE : AV_LOG_WARNING, + "Multiple frames in a packet from stream %d\n", pkt->stream_index); + ist->showed_multi_packet_warning = 1; + } + + switch (ist->st->codec->codec_type) { + case AVMEDIA_TYPE_AUDIO: + ret = decode_audio (ist, &avpkt, &got_output); + break; + case AVMEDIA_TYPE_VIDEO: + ret = decode_video (ist, &avpkt, &got_output); + if (avpkt.duration) + ist->next_dts += av_rescale_q(avpkt.duration, ist->st->time_base, AV_TIME_BASE_Q); + else if (ist->st->avg_frame_rate.num) + ist->next_dts += av_rescale_q(1, av_inv_q(ist->st->avg_frame_rate), + AV_TIME_BASE_Q); + else if (ist->st->codec->time_base.num != 0) { + int ticks = ist->st->parser ? ist->st->parser->repeat_pict + 1 : + ist->st->codec->ticks_per_frame; + ist->next_dts += av_rescale_q(ticks, ist->st->codec->time_base, AV_TIME_BASE_Q); + } + break; + case AVMEDIA_TYPE_SUBTITLE: + ret = transcode_subtitles(ist, &avpkt, &got_output); + break; + default: + return -1; + } + + if (ret < 0) + return ret; + // touch data and size only if not EOF + if (pkt) { + avpkt.data += ret; + avpkt.size -= ret; + } + if (!got_output) { + continue; + } + } + + /* handle stream copy */ + if (!ist->decoding_needed) { + ist->last_dts = ist->next_dts; + switch (ist->st->codec->codec_type) { + case AVMEDIA_TYPE_AUDIO: + ist->next_dts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) / + ist->st->codec->sample_rate; + break; + case AVMEDIA_TYPE_VIDEO: + if (ist->st->codec->time_base.num != 0) { + int ticks = ist->st->parser ? ist->st->parser->repeat_pict + 1 : ist->st->codec->ticks_per_frame; + ist->next_dts += ((int64_t)AV_TIME_BASE * + ist->st->codec->time_base.num * ticks) / + ist->st->codec->time_base.den; + } + break; + } + } + for (i = 0; pkt && i < nb_output_streams; i++) { + OutputStream *ost = output_streams[i]; + + if (!check_output_constraints(ist, ost) || ost->encoding_needed) + continue; + + do_streamcopy(ist, ost, pkt); + } + + return 0; +} + +static void print_sdp(void) +{ + char sdp[16384]; + int i; + AVFormatContext **avc = av_malloc(sizeof(*avc) * nb_output_files); + + if (!avc) + exit_program(1); + for (i = 0; i < nb_output_files; i++) + avc[i] = output_files[i]->ctx; + + av_sdp_create(avc, nb_output_files, sdp, sizeof(sdp)); + printf("SDP:\n%s\n", sdp); + fflush(stdout); + av_freep(&avc); +} + +static const HWAccel *get_hwaccel(enum AVPixelFormat pix_fmt) +{ + int i; + for (i = 0; hwaccels[i].name; i++) + if (hwaccels[i].pix_fmt == pix_fmt) + return &hwaccels[i]; + return NULL; +} + +static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat *pix_fmts) +{ + InputStream *ist = s->opaque; + const enum AVPixelFormat *p; + int ret; + + for (p = pix_fmts; *p != -1; p++) { + const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(*p); + const HWAccel *hwaccel; + + if (!(desc->flags & AV_PIX_FMT_FLAG_HWACCEL)) + break; + + hwaccel = get_hwaccel(*p); + if (!hwaccel || + (ist->active_hwaccel_id && ist->active_hwaccel_id != hwaccel->id) || + (ist->hwaccel_id != HWACCEL_AUTO && ist->hwaccel_id != hwaccel->id)) + continue; + + ret = hwaccel->init(s); + if (ret < 0) { + if (ist->hwaccel_id == hwaccel->id) { + av_log(NULL, AV_LOG_FATAL, + "%s hwaccel requested for input stream #%d:%d, " + "but cannot be initialized.\n", hwaccel->name, + ist->file_index, ist->st->index); + exit_program(1); + } + continue; + } + ist->active_hwaccel_id = hwaccel->id; + ist->hwaccel_pix_fmt = *p; + break; + } + + return *p; +} + +static int get_buffer(AVCodecContext *s, AVFrame *frame, int flags) +{ + InputStream *ist = s->opaque; + + if (ist->hwaccel_get_buffer && frame->format == ist->hwaccel_pix_fmt) + return ist->hwaccel_get_buffer(s, frame, flags); + + return avcodec_default_get_buffer2(s, frame, flags); +} + +static int init_input_stream(int ist_index, char *error, int error_len) +{ + int i, ret; + InputStream *ist = input_streams[ist_index]; + if (ist->decoding_needed) { + AVCodec *codec = ist->dec; + if (!codec) { + snprintf(error, error_len, "Decoder (codec id %d) not found for input stream #%d:%d", + ist->st->codec->codec_id, ist->file_index, ist->st->index); + return AVERROR(EINVAL); + } + + /* update requested sample format for the decoder based on the + corresponding encoder sample format */ + for (i = 0; i < nb_output_streams; i++) { + OutputStream *ost = output_streams[i]; + if (ost->source_index == ist_index) { + update_sample_fmt(ist->st->codec, codec, ost->st->codec); + break; + } + } + + ist->st->codec->opaque = ist; + ist->st->codec->get_format = get_format; + ist->st->codec->get_buffer2 = get_buffer; + ist->st->codec->thread_safe_callbacks = 1; + + av_opt_set_int(ist->st->codec, "refcounted_frames", 1, 0); + + if (!av_dict_get(ist->decoder_opts, "threads", NULL, 0)) + av_dict_set(&ist->decoder_opts, "threads", "auto", 0); + if ((ret = avcodec_open2(ist->st->codec, codec, &ist->decoder_opts)) < 0) { + char errbuf[128]; + if (ret == AVERROR_EXPERIMENTAL) + abort_codec_experimental(codec, 0); + + av_strerror(ret, errbuf, sizeof(errbuf)); + + snprintf(error, error_len, + "Error while opening decoder for input stream " + "#%d:%d : %s", + ist->file_index, ist->st->index, errbuf); + return ret; + } + assert_avoptions(ist->decoder_opts); + } + + ist->last_dts = ist->st->avg_frame_rate.num ? - ist->st->codec->has_b_frames * AV_TIME_BASE / av_q2d(ist->st->avg_frame_rate) : 0; + ist->next_dts = AV_NOPTS_VALUE; + init_pts_correction(&ist->pts_ctx); + + return 0; +} + +static InputStream *get_input_stream(OutputStream *ost) +{ + if (ost->source_index >= 0) + return input_streams[ost->source_index]; + + if (ost->filter) { + FilterGraph *fg = ost->filter->graph; + int i; + + for (i = 0; i < fg->nb_inputs; i++) + if (fg->inputs[i]->ist->st->codec->codec_type == ost->st->codec->codec_type) + return fg->inputs[i]->ist; + } + + return NULL; +} + +static void parse_forced_key_frames(char *kf, OutputStream *ost, + AVCodecContext *avctx) +{ + char *p; + int n = 1, i; + int64_t t; + + for (p = kf; *p; p++) + if (*p == ',') + n++; + ost->forced_kf_count = n; + ost->forced_kf_pts = av_malloc(sizeof(*ost->forced_kf_pts) * n); + if (!ost->forced_kf_pts) { + av_log(NULL, AV_LOG_FATAL, "Could not allocate forced key frames array.\n"); + exit_program(1); + } + + p = kf; + for (i = 0; i < n; i++) { + char *next = strchr(p, ','); + + if (next) + *next++ = 0; + + t = parse_time_or_die("force_key_frames", p, 1); + ost->forced_kf_pts[i] = av_rescale_q(t, AV_TIME_BASE_Q, avctx->time_base); + + p = next; + } +} + +static void set_encoder_id(OutputFile *of, OutputStream *ost) +{ + AVDictionaryEntry *e; + + uint8_t *encoder_string; + int encoder_string_len; + int format_flags = 0; + + e = av_dict_get(of->opts, "fflags", NULL, 0); + if (e) { + const AVOption *o = av_opt_find(of->ctx, "fflags", NULL, 0, 0); + if (!o) + return; + av_opt_eval_flags(of->ctx, o, e->value, &format_flags); + } + + encoder_string_len = sizeof(LIBAVCODEC_IDENT) + strlen(ost->enc->name) + 2; + encoder_string = av_mallocz(encoder_string_len); + if (!encoder_string) + exit_program(1); + + if (!(format_flags & AVFMT_FLAG_BITEXACT)) + av_strlcpy(encoder_string, LIBAVCODEC_IDENT " ", encoder_string_len); + av_strlcat(encoder_string, ost->enc->name, encoder_string_len); + av_dict_set(&ost->st->metadata, "encoder", encoder_string, + AV_DICT_DONT_STRDUP_VAL | AV_DICT_DONT_OVERWRITE); +} + +static int transcode_init(void) +{ + int ret = 0, i, j, k; + AVFormatContext *oc; + OutputStream *ost; + InputStream *ist; + char error[1024]; + int want_sdp = 1; + + /* init framerate emulation */ + for (i = 0; i < nb_input_files; i++) { + InputFile *ifile = input_files[i]; + if (ifile->rate_emu) + for (j = 0; j < ifile->nb_streams; j++) + input_streams[j + ifile->ist_index]->start = av_gettime(); + } + + /* output stream init */ + for (i = 0; i < nb_output_files; i++) { + oc = output_files[i]->ctx; + if (!oc->nb_streams && !(oc->oformat->flags & AVFMT_NOSTREAMS)) { + av_dump_format(oc, i, oc->filename, 1); + av_log(NULL, AV_LOG_ERROR, "Output file #%d does not contain any stream\n", i); + return AVERROR(EINVAL); + } + } + + /* init complex filtergraphs */ + for (i = 0; i < nb_filtergraphs; i++) + if ((ret = avfilter_graph_config(filtergraphs[i]->graph, NULL)) < 0) + return ret; + + /* for each output stream, we compute the right encoding parameters */ + for (i = 0; i < nb_output_streams; i++) { + AVCodecContext *enc_ctx; + AVCodecContext *dec_ctx = NULL; + ost = output_streams[i]; + oc = output_files[ost->file_index]->ctx; + ist = get_input_stream(ost); + + if (ost->attachment_filename) + continue; + + enc_ctx = ost->st->codec; + + if (ist) { + dec_ctx = ist->st->codec; + + ost->st->disposition = ist->st->disposition; + enc_ctx->bits_per_raw_sample = dec_ctx->bits_per_raw_sample; + enc_ctx->chroma_sample_location = dec_ctx->chroma_sample_location; + } + + if (ost->stream_copy) { + AVRational sar; + uint64_t extra_size; + + av_assert0(ist && !ost->filter); + + extra_size = (uint64_t)dec_ctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE; + + if (extra_size > INT_MAX) { + return AVERROR(EINVAL); + } + + /* if stream_copy is selected, no need to decode or encode */ + enc_ctx->codec_id = dec_ctx->codec_id; + enc_ctx->codec_type = dec_ctx->codec_type; + + if (!enc_ctx->codec_tag) { + if (!oc->oformat->codec_tag || + av_codec_get_id (oc->oformat->codec_tag, dec_ctx->codec_tag) == enc_ctx->codec_id || + av_codec_get_tag(oc->oformat->codec_tag, dec_ctx->codec_id) <= 0) + enc_ctx->codec_tag = dec_ctx->codec_tag; + } + + enc_ctx->bit_rate = dec_ctx->bit_rate; + enc_ctx->rc_max_rate = dec_ctx->rc_max_rate; + enc_ctx->rc_buffer_size = dec_ctx->rc_buffer_size; + enc_ctx->field_order = dec_ctx->field_order; + enc_ctx->extradata = av_mallocz(extra_size); + if (!enc_ctx->extradata) { + return AVERROR(ENOMEM); + } + memcpy(enc_ctx->extradata, dec_ctx->extradata, dec_ctx->extradata_size); + enc_ctx->extradata_size = dec_ctx->extradata_size; + if (!copy_tb) { + enc_ctx->time_base = dec_ctx->time_base; + enc_ctx->time_base.num *= dec_ctx->ticks_per_frame; + av_reduce(&enc_ctx->time_base.num, &enc_ctx->time_base.den, + enc_ctx->time_base.num, enc_ctx->time_base.den, INT_MAX); + } else + enc_ctx->time_base = ist->st->time_base; + + ost->parser = av_parser_init(enc_ctx->codec_id); + + switch (enc_ctx->codec_type) { + case AVMEDIA_TYPE_AUDIO: + if (audio_volume != 256) { + av_log(NULL, AV_LOG_FATAL, "-acodec copy and -vol are incompatible (frames are not decoded)\n"); + exit_program(1); + } + enc_ctx->channel_layout = dec_ctx->channel_layout; + enc_ctx->sample_rate = dec_ctx->sample_rate; + enc_ctx->channels = dec_ctx->channels; + enc_ctx->frame_size = dec_ctx->frame_size; + enc_ctx->audio_service_type = dec_ctx->audio_service_type; + enc_ctx->block_align = dec_ctx->block_align; + break; + case AVMEDIA_TYPE_VIDEO: + enc_ctx->pix_fmt = dec_ctx->pix_fmt; + enc_ctx->width = dec_ctx->width; + enc_ctx->height = dec_ctx->height; + enc_ctx->has_b_frames = dec_ctx->has_b_frames; + if (ost->frame_aspect_ratio) + sar = av_d2q(ost->frame_aspect_ratio * enc_ctx->height / enc_ctx->width, 255); + else if (ist->st->sample_aspect_ratio.num) + sar = ist->st->sample_aspect_ratio; + else + sar = dec_ctx->sample_aspect_ratio; + ost->st->sample_aspect_ratio = enc_ctx->sample_aspect_ratio = sar; + break; + case AVMEDIA_TYPE_SUBTITLE: + enc_ctx->width = dec_ctx->width; + enc_ctx->height = dec_ctx->height; + break; + case AVMEDIA_TYPE_DATA: + case AVMEDIA_TYPE_ATTACHMENT: + break; + default: + abort(); + } + } else { + if (!ost->enc) { + /* should only happen when a default codec is not present. */ + snprintf(error, sizeof(error), "Automatic encoder selection " + "failed for output stream #%d:%d. Default encoder for " + "format %s is probably disabled. Please choose an " + "encoder manually.\n", ost->file_index, ost->index, + oc->oformat->name); + ret = AVERROR(EINVAL); + goto dump_format; + } + + if (ist) + ist->decoding_needed = 1; + ost->encoding_needed = 1; + + set_encoder_id(output_files[ost->file_index], ost); + + /* + * We want CFR output if and only if one of those is true: + * 1) user specified output framerate with -r + * 2) user specified -vsync cfr + * 3) output format is CFR and the user didn't force vsync to + * something else than CFR + * + * in such a case, set ost->frame_rate + */ + if (enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO && + !ost->frame_rate.num && ist && + (video_sync_method == VSYNC_CFR || + (video_sync_method == VSYNC_AUTO && + !(oc->oformat->flags & (AVFMT_NOTIMESTAMPS | AVFMT_VARIABLE_FPS))))) { + if (ist->framerate.num) + ost->frame_rate = ist->framerate; + else if (ist->st->avg_frame_rate.num) + ost->frame_rate = ist->st->avg_frame_rate; + else { + av_log(NULL, AV_LOG_WARNING, "Constant framerate requested " + "for the output stream #%d:%d, but no information " + "about the input framerate is available. Falling " + "back to a default value of 25fps. Use the -r option " + "if you want a different framerate.\n", + ost->file_index, ost->index); + ost->frame_rate = (AVRational){ 25, 1 }; + } + + if (ost->enc && ost->enc->supported_framerates && !ost->force_fps) { + int idx = av_find_nearest_q_idx(ost->frame_rate, ost->enc->supported_framerates); + ost->frame_rate = ost->enc->supported_framerates[idx]; + } + } + + if (!ost->filter && + (enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO || + enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO)) { + FilterGraph *fg; + fg = init_simple_filtergraph(ist, ost); + if (configure_filtergraph(fg)) { + av_log(NULL, AV_LOG_FATAL, "Error opening filters!\n"); + exit_program(1); + } + } + + switch (enc_ctx->codec_type) { + case AVMEDIA_TYPE_AUDIO: + enc_ctx->sample_fmt = ost->filter->filter->inputs[0]->format; + enc_ctx->sample_rate = ost->filter->filter->inputs[0]->sample_rate; + enc_ctx->channel_layout = ost->filter->filter->inputs[0]->channel_layout; + enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout); + enc_ctx->time_base = (AVRational){ 1, enc_ctx->sample_rate }; + break; + case AVMEDIA_TYPE_VIDEO: + enc_ctx->time_base = ost->filter->filter->inputs[0]->time_base; + + enc_ctx->width = ost->filter->filter->inputs[0]->w; + enc_ctx->height = ost->filter->filter->inputs[0]->h; + enc_ctx->sample_aspect_ratio = ost->st->sample_aspect_ratio = + ost->frame_aspect_ratio ? // overridden by the -aspect cli option + av_d2q(ost->frame_aspect_ratio * enc_ctx->height/enc_ctx->width, 255) : + ost->filter->filter->inputs[0]->sample_aspect_ratio; + enc_ctx->pix_fmt = ost->filter->filter->inputs[0]->format; + + ost->st->avg_frame_rate = ost->frame_rate; + + if (dec_ctx && + (enc_ctx->width != dec_ctx->width || + enc_ctx->height != dec_ctx->height || + enc_ctx->pix_fmt != dec_ctx->pix_fmt)) { + enc_ctx->bits_per_raw_sample = 0; + } + + if (ost->forced_keyframes) + parse_forced_key_frames(ost->forced_keyframes, ost, + ost->st->codec); + break; + case AVMEDIA_TYPE_SUBTITLE: + enc_ctx->time_base = (AVRational){1, 1000}; + break; + default: + abort(); + break; + } + /* two pass mode */ + if ((enc_ctx->flags & (CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2))) { + char logfilename[1024]; + FILE *f; + + snprintf(logfilename, sizeof(logfilename), "%s-%d.log", + ost->logfile_prefix ? ost->logfile_prefix : + DEFAULT_PASS_LOGFILENAME_PREFIX, + i); + if (!strcmp(ost->enc->name, "libx264")) { + av_dict_set(&ost->encoder_opts, "stats", logfilename, AV_DICT_DONT_OVERWRITE); + } else { + if (enc_ctx->flags & CODEC_FLAG_PASS1) { + f = fopen(logfilename, "wb"); + if (!f) { + av_log(NULL, AV_LOG_FATAL, "Cannot write log file '%s' for pass-1 encoding: %s\n", + logfilename, strerror(errno)); + exit_program(1); + } + ost->logfile = f; + } else { + char *logbuffer; + size_t logbuffer_size; + if (cmdutils_read_file(logfilename, &logbuffer, &logbuffer_size) < 0) { + av_log(NULL, AV_LOG_FATAL, "Error reading log file '%s' for pass-2 encoding\n", + logfilename); + exit_program(1); + } + enc_ctx->stats_in = logbuffer; + } + } + } + } + } + + /* open each encoder */ + for (i = 0; i < nb_output_streams; i++) { + ost = output_streams[i]; + if (ost->encoding_needed) { + AVCodec *codec = ost->enc; + AVCodecContext *dec = NULL; + + if ((ist = get_input_stream(ost))) + dec = ist->st->codec; + if (dec && dec->subtitle_header) { + ost->st->codec->subtitle_header = av_malloc(dec->subtitle_header_size); + if (!ost->st->codec->subtitle_header) { + ret = AVERROR(ENOMEM); + goto dump_format; + } + memcpy(ost->st->codec->subtitle_header, dec->subtitle_header, dec->subtitle_header_size); + ost->st->codec->subtitle_header_size = dec->subtitle_header_size; + } + if (!av_dict_get(ost->encoder_opts, "threads", NULL, 0)) + av_dict_set(&ost->encoder_opts, "threads", "auto", 0); + if ((ret = avcodec_open2(ost->st->codec, codec, &ost->encoder_opts)) < 0) { + if (ret == AVERROR_EXPERIMENTAL) + abort_codec_experimental(codec, 1); + snprintf(error, sizeof(error), "Error while opening encoder for output stream #%d:%d - maybe incorrect parameters such as bit_rate, rate, width or height", + ost->file_index, ost->index); + goto dump_format; + } + assert_avoptions(ost->encoder_opts); + if (ost->st->codec->bit_rate && ost->st->codec->bit_rate < 1000) + av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low." + "It takes bits/s as argument, not kbits/s\n"); + } else { + av_opt_set_dict(ost->st->codec, &ost->encoder_opts); + } + } + + /* init input streams */ + for (i = 0; i < nb_input_streams; i++) + if ((ret = init_input_stream(i, error, sizeof(error))) < 0) + goto dump_format; + + /* discard unused programs */ + for (i = 0; i < nb_input_files; i++) { + InputFile *ifile = input_files[i]; + for (j = 0; j < ifile->ctx->nb_programs; j++) { + AVProgram *p = ifile->ctx->programs[j]; + int discard = AVDISCARD_ALL; + + for (k = 0; k < p->nb_stream_indexes; k++) + if (!input_streams[ifile->ist_index + p->stream_index[k]]->discard) { + discard = AVDISCARD_DEFAULT; + break; + } + p->discard = discard; + } + } + + /* open files and write file headers */ + for (i = 0; i < nb_output_files; i++) { + oc = output_files[i]->ctx; + oc->interrupt_callback = int_cb; + if ((ret = avformat_write_header(oc, &output_files[i]->opts)) < 0) { + char errbuf[128]; + av_strerror(ret, errbuf, sizeof(errbuf)); + snprintf(error, sizeof(error), + "Could not write header for output file #%d " + "(incorrect codec parameters ?): %s", + i, errbuf); + ret = AVERROR(EINVAL); + goto dump_format; + } + assert_avoptions(output_files[i]->opts); + if (strcmp(oc->oformat->name, "rtp")) { + want_sdp = 0; + } + } + + dump_format: + /* dump the file output parameters - cannot be done before in case + of stream copy */ + for (i = 0; i < nb_output_files; i++) { + av_dump_format(output_files[i]->ctx, i, output_files[i]->ctx->filename, 1); + } + + /* dump the stream mapping */ + av_log(NULL, AV_LOG_INFO, "Stream mapping:\n"); + for (i = 0; i < nb_input_streams; i++) { + ist = input_streams[i]; + + for (j = 0; j < ist->nb_filters; j++) { + if (ist->filters[j]->graph->graph_desc) { + av_log(NULL, AV_LOG_INFO, " Stream #%d:%d (%s) -> %s", + ist->file_index, ist->st->index, ist->dec ? ist->dec->name : "?", + ist->filters[j]->name); + if (nb_filtergraphs > 1) + av_log(NULL, AV_LOG_INFO, " (graph %d)", ist->filters[j]->graph->index); + av_log(NULL, AV_LOG_INFO, "\n"); + } + } + } + + for (i = 0; i < nb_output_streams; i++) { + ost = output_streams[i]; + + if (ost->attachment_filename) { + /* an attached file */ + av_log(NULL, AV_LOG_INFO, " File %s -> Stream #%d:%d\n", + ost->attachment_filename, ost->file_index, ost->index); + continue; + } + + if (ost->filter && ost->filter->graph->graph_desc) { + /* output from a complex graph */ + av_log(NULL, AV_LOG_INFO, " %s", ost->filter->name); + if (nb_filtergraphs > 1) + av_log(NULL, AV_LOG_INFO, " (graph %d)", ost->filter->graph->index); + + av_log(NULL, AV_LOG_INFO, " -> Stream #%d:%d (%s)\n", ost->file_index, + ost->index, ost->enc ? ost->enc->name : "?"); + continue; + } + + av_log(NULL, AV_LOG_INFO, " Stream #%d:%d -> #%d:%d", + input_streams[ost->source_index]->file_index, + input_streams[ost->source_index]->st->index, + ost->file_index, + ost->index); + if (ost->sync_ist != input_streams[ost->source_index]) + av_log(NULL, AV_LOG_INFO, " [sync #%d:%d]", + ost->sync_ist->file_index, + ost->sync_ist->st->index); + if (ost->stream_copy) + av_log(NULL, AV_LOG_INFO, " (copy)"); + else + av_log(NULL, AV_LOG_INFO, " (%s -> %s)", input_streams[ost->source_index]->dec ? + input_streams[ost->source_index]->dec->name : "?", + ost->enc ? ost->enc->name : "?"); + av_log(NULL, AV_LOG_INFO, "\n"); + } + + if (ret) { + av_log(NULL, AV_LOG_ERROR, "%s\n", error); + return ret; + } + + if (want_sdp) { + print_sdp(); + } + + return 0; +} + +/* Return 1 if there remain streams where more output is wanted, 0 otherwise. */ +static int need_output(void) +{ + int i; + + for (i = 0; i < nb_output_streams; i++) { + OutputStream *ost = output_streams[i]; + OutputFile *of = output_files[ost->file_index]; + AVFormatContext *os = output_files[ost->file_index]->ctx; + + if (ost->finished || + (os->pb && avio_tell(os->pb) >= of->limit_filesize)) + continue; + if (ost->frame_number >= ost->max_frames) { + int j; + for (j = 0; j < of->ctx->nb_streams; j++) + output_streams[of->ost_index + j]->finished = 1; + continue; + } + + return 1; + } + + return 0; +} + +static InputFile *select_input_file(void) +{ + InputFile *ifile = NULL; + int64_t ipts_min = INT64_MAX; + int i; + + for (i = 0; i < nb_input_streams; i++) { + InputStream *ist = input_streams[i]; + int64_t ipts = ist->last_dts; + + if (ist->discard || input_files[ist->file_index]->eagain) + continue; + if (!input_files[ist->file_index]->eof_reached) { + if (ipts < ipts_min) { + ipts_min = ipts; + ifile = input_files[ist->file_index]; + } + } + } + + return ifile; +} + +#if HAVE_PTHREADS +static void *input_thread(void *arg) +{ + InputFile *f = arg; + int ret = 0; + + while (!transcoding_finished && ret >= 0) { + AVPacket pkt; + ret = av_read_frame(f->ctx, &pkt); + + if (ret == AVERROR(EAGAIN)) { + av_usleep(10000); + ret = 0; + continue; + } else if (ret < 0) + break; + + pthread_mutex_lock(&f->fifo_lock); + while (!av_fifo_space(f->fifo)) + pthread_cond_wait(&f->fifo_cond, &f->fifo_lock); + + av_dup_packet(&pkt); + av_fifo_generic_write(f->fifo, &pkt, sizeof(pkt), NULL); + + pthread_mutex_unlock(&f->fifo_lock); + } + + f->finished = 1; + return NULL; +} + +static void free_input_threads(void) +{ + int i; + + if (nb_input_files == 1) + return; + + transcoding_finished = 1; + + for (i = 0; i < nb_input_files; i++) { + InputFile *f = input_files[i]; + AVPacket pkt; + + if (!f->fifo || f->joined) + continue; + + pthread_mutex_lock(&f->fifo_lock); + while (av_fifo_size(f->fifo)) { + av_fifo_generic_read(f->fifo, &pkt, sizeof(pkt), NULL); + av_free_packet(&pkt); + } + pthread_cond_signal(&f->fifo_cond); + pthread_mutex_unlock(&f->fifo_lock); + + pthread_join(f->thread, NULL); + f->joined = 1; + + while (av_fifo_size(f->fifo)) { + av_fifo_generic_read(f->fifo, &pkt, sizeof(pkt), NULL); + av_free_packet(&pkt); + } + av_fifo_free(f->fifo); + } +} + +static int init_input_threads(void) +{ + int i, ret; + + if (nb_input_files == 1) + return 0; + + for (i = 0; i < nb_input_files; i++) { + InputFile *f = input_files[i]; + + if (!(f->fifo = av_fifo_alloc(8*sizeof(AVPacket)))) + return AVERROR(ENOMEM); + + pthread_mutex_init(&f->fifo_lock, NULL); + pthread_cond_init (&f->fifo_cond, NULL); + + if ((ret = pthread_create(&f->thread, NULL, input_thread, f))) + return AVERROR(ret); + } + return 0; +} + +static int get_input_packet_mt(InputFile *f, AVPacket *pkt) +{ + int ret = 0; + + pthread_mutex_lock(&f->fifo_lock); + + if (av_fifo_size(f->fifo)) { + av_fifo_generic_read(f->fifo, pkt, sizeof(*pkt), NULL); + pthread_cond_signal(&f->fifo_cond); + } else { + if (f->finished) + ret = AVERROR_EOF; + else + ret = AVERROR(EAGAIN); + } + + pthread_mutex_unlock(&f->fifo_lock); + + return ret; +} +#endif + +static int get_input_packet(InputFile *f, AVPacket *pkt) +{ + if (f->rate_emu) { + int i; + for (i = 0; i < f->nb_streams; i++) { + InputStream *ist = input_streams[f->ist_index + i]; + int64_t pts = av_rescale(ist->last_dts, 1000000, AV_TIME_BASE); + int64_t now = av_gettime() - ist->start; + if (pts > now) + return AVERROR(EAGAIN); + } + } + +#if HAVE_PTHREADS + if (nb_input_files > 1) + return get_input_packet_mt(f, pkt); +#endif + return av_read_frame(f->ctx, pkt); +} + +static int got_eagain(void) +{ + int i; + for (i = 0; i < nb_input_files; i++) + if (input_files[i]->eagain) + return 1; + return 0; +} + +static void reset_eagain(void) +{ + int i; + for (i = 0; i < nb_input_files; i++) + input_files[i]->eagain = 0; +} + +/* + * Read one packet from an input file and send it for + * - decoding -> lavfi (audio/video) + * - decoding -> encoding -> muxing (subtitles) + * - muxing (streamcopy) + * + * Return + * - 0 -- one packet was read and processed + * - AVERROR(EAGAIN) -- no packets were available for selected file, + * this function should be called again + * - AVERROR_EOF -- this function should not be called again + */ +static int process_input(void) +{ + InputFile *ifile; + AVFormatContext *is; + InputStream *ist; + AVPacket pkt; + int ret, i, j; + + /* select the stream that we must read now */ + ifile = select_input_file(); + /* if none, if is finished */ + if (!ifile) { + if (got_eagain()) { + reset_eagain(); + av_usleep(10000); + return AVERROR(EAGAIN); + } + av_log(NULL, AV_LOG_VERBOSE, "No more inputs to read from.\n"); + return AVERROR_EOF; + } + + is = ifile->ctx; + ret = get_input_packet(ifile, &pkt); + + if (ret == AVERROR(EAGAIN)) { + ifile->eagain = 1; + return ret; + } + if (ret < 0) { + if (ret != AVERROR_EOF) { + print_error(is->filename, ret); + if (exit_on_error) + exit_program(1); + } + ifile->eof_reached = 1; + + for (i = 0; i < ifile->nb_streams; i++) { + ist = input_streams[ifile->ist_index + i]; + if (ist->decoding_needed) + output_packet(ist, NULL); + + /* mark all outputs that don't go through lavfi as finished */ + for (j = 0; j < nb_output_streams; j++) { + OutputStream *ost = output_streams[j]; + + if (ost->source_index == ifile->ist_index + i && + (ost->stream_copy || ost->enc->type == AVMEDIA_TYPE_SUBTITLE)) + ost->finished= 1; + } + } + + return AVERROR(EAGAIN); + } + + reset_eagain(); + + if (do_pkt_dump) { + av_pkt_dump_log2(NULL, AV_LOG_DEBUG, &pkt, do_hex_dump, + is->streams[pkt.stream_index]); + } + /* the following test is needed in case new streams appear + dynamically in stream : we ignore them */ + if (pkt.stream_index >= ifile->nb_streams) + goto discard_packet; + + ist = input_streams[ifile->ist_index + pkt.stream_index]; + + ist->data_size += pkt.size; + ist->nb_packets++; + + if (ist->discard) + goto discard_packet; + + /* add the stream-global side data to the first packet */ + if (ist->nb_packets == 1) + for (i = 0; i < ist->st->nb_side_data; i++) { + AVPacketSideData *src_sd = &ist->st->side_data[i]; + uint8_t *dst_data; + + if (av_packet_get_side_data(&pkt, src_sd->type, NULL)) + continue; + + dst_data = av_packet_new_side_data(&pkt, src_sd->type, src_sd->size); + if (!dst_data) + exit_program(1); + + memcpy(dst_data, src_sd->data, src_sd->size); + } + + if (pkt.dts != AV_NOPTS_VALUE) + pkt.dts += av_rescale_q(ifile->ts_offset, AV_TIME_BASE_Q, ist->st->time_base); + if (pkt.pts != AV_NOPTS_VALUE) + pkt.pts += av_rescale_q(ifile->ts_offset, AV_TIME_BASE_Q, ist->st->time_base); + + if (pkt.pts != AV_NOPTS_VALUE) + pkt.pts *= ist->ts_scale; + if (pkt.dts != AV_NOPTS_VALUE) + pkt.dts *= ist->ts_scale; + + if (pkt.dts != AV_NOPTS_VALUE && ist->next_dts != AV_NOPTS_VALUE && + (is->iformat->flags & AVFMT_TS_DISCONT)) { + int64_t pkt_dts = av_rescale_q(pkt.dts, ist->st->time_base, AV_TIME_BASE_Q); + int64_t delta = pkt_dts - ist->next_dts; + + if ((FFABS(delta) > 1LL * dts_delta_threshold * AV_TIME_BASE || pkt_dts + 1 < ist->last_dts) && !copy_ts) { + ifile->ts_offset -= delta; + av_log(NULL, AV_LOG_DEBUG, + "timestamp discontinuity %"PRId64", new offset= %"PRId64"\n", + delta, ifile->ts_offset); + pkt.dts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base); + if (pkt.pts != AV_NOPTS_VALUE) + pkt.pts -= av_rescale_q(delta, AV_TIME_BASE_Q, ist->st->time_base); + } + } + + ret = output_packet(ist, &pkt); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d\n", + ist->file_index, ist->st->index); + if (exit_on_error) + exit_program(1); + } + +discard_packet: + av_free_packet(&pkt); + + return 0; +} + +/* + * The following code is the main loop of the file converter + */ +static int transcode(void) +{ + int ret, i, need_input = 1; + AVFormatContext *os; + OutputStream *ost; + InputStream *ist; + int64_t timer_start; + + ret = transcode_init(); + if (ret < 0) + goto fail; + + av_log(NULL, AV_LOG_INFO, "Press ctrl-c to stop encoding\n"); + term_init(); + + timer_start = av_gettime(); + +#if HAVE_PTHREADS + if ((ret = init_input_threads()) < 0) + goto fail; +#endif + + while (!received_sigterm) { + /* check if there's any stream where output is still needed */ + if (!need_output()) { + av_log(NULL, AV_LOG_VERBOSE, "No more output streams to write to, finishing.\n"); + break; + } + + /* read and process one input packet if needed */ + if (need_input) { + ret = process_input(); + if (ret == AVERROR_EOF) + need_input = 0; + } + + ret = poll_filters(); + if (ret < 0) { + if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN)) + continue; + + av_log(NULL, AV_LOG_ERROR, "Error while filtering.\n"); + break; + } + + /* dump report by using the output first video and audio streams */ + print_report(0, timer_start); + } +#if HAVE_PTHREADS + free_input_threads(); +#endif + + /* at the end of stream, we must flush the decoder buffers */ + for (i = 0; i < nb_input_streams; i++) { + ist = input_streams[i]; + if (!input_files[ist->file_index]->eof_reached && ist->decoding_needed) { + output_packet(ist, NULL); + } + } + poll_filters(); + flush_encoders(); + + term_exit(); + + /* write the trailer if needed and close file */ + for (i = 0; i < nb_output_files; i++) { + os = output_files[i]->ctx; + av_write_trailer(os); + } + + /* dump report by using the first video and audio streams */ + print_report(1, timer_start); + + /* close each encoder */ + for (i = 0; i < nb_output_streams; i++) { + ost = output_streams[i]; + if (ost->encoding_needed) { + av_freep(&ost->st->codec->stats_in); + avcodec_close(ost->st->codec); + } + } + + /* close each decoder */ + for (i = 0; i < nb_input_streams; i++) { + ist = input_streams[i]; + if (ist->decoding_needed) { + avcodec_close(ist->st->codec); + if (ist->hwaccel_uninit) + ist->hwaccel_uninit(ist->st->codec); + } + } + + /* finished ! */ + ret = 0; + + fail: +#if HAVE_PTHREADS + free_input_threads(); +#endif + + if (output_streams) { + for (i = 0; i < nb_output_streams; i++) { + ost = output_streams[i]; + if (ost) { + if (ost->stream_copy) + av_freep(&ost->st->codec->extradata); + if (ost->logfile) { + fclose(ost->logfile); + ost->logfile = NULL; + } + av_freep(&ost->st->codec->subtitle_header); + av_free(ost->forced_kf_pts); + av_dict_free(&ost->encoder_opts); + av_dict_free(&ost->resample_opts); + } + } + } + return ret; +} + +static int64_t getutime(void) +{ +#if HAVE_GETRUSAGE + struct rusage rusage; + + getrusage(RUSAGE_SELF, &rusage); + return (rusage.ru_utime.tv_sec * 1000000LL) + rusage.ru_utime.tv_usec; +#elif HAVE_GETPROCESSTIMES + HANDLE proc; + FILETIME c, e, k, u; + proc = GetCurrentProcess(); + GetProcessTimes(proc, &c, &e, &k, &u); + return ((int64_t) u.dwHighDateTime << 32 | u.dwLowDateTime) / 10; +#else + return av_gettime(); +#endif +} + +static int64_t getmaxrss(void) +{ +#if HAVE_GETRUSAGE && HAVE_STRUCT_RUSAGE_RU_MAXRSS + struct rusage rusage; + getrusage(RUSAGE_SELF, &rusage); + return (int64_t)rusage.ru_maxrss * 1024; +#elif HAVE_GETPROCESSMEMORYINFO + HANDLE proc; + PROCESS_MEMORY_COUNTERS memcounters; + proc = GetCurrentProcess(); + memcounters.cb = sizeof(memcounters); + GetProcessMemoryInfo(proc, &memcounters, sizeof(memcounters)); + return memcounters.PeakPagefileUsage; +#else + return 0; +#endif +} + +int main(int argc, char **argv) +{ + int ret; + int64_t ti; + + register_exit(avconv_cleanup); + + av_log_set_flags(AV_LOG_SKIP_REPEATED); + parse_loglevel(argc, argv, options); + + avcodec_register_all(); +#if CONFIG_AVDEVICE + avdevice_register_all(); +#endif + avfilter_register_all(); + av_register_all(); + avformat_network_init(); + + show_banner(); + + /* parse options and open all input/output files */ + ret = avconv_parse_options(argc, argv); + if (ret < 0) + exit_program(1); + + if (nb_output_files <= 0 && nb_input_files == 0) { + show_usage(); + av_log(NULL, AV_LOG_WARNING, "Use -h to get full help or, even better, run 'man %s'\n", program_name); + exit_program(1); + } + + /* file converter / grab */ + if (nb_output_files <= 0) { + fprintf(stderr, "At least one output file must be specified\n"); + exit_program(1); + } + + ti = getutime(); + if (transcode() < 0) + exit_program(1); + ti = getutime() - ti; + if (do_benchmark) { + int maxrss = getmaxrss() / 1024; + printf("bench: utime=%0.3fs maxrss=%ikB\n", ti / 1000000.0, maxrss); + } + + exit_program(0); + return 0; +} diff --git a/ffmpeg.h b/avconv.h similarity index 71% rename from ffmpeg.h rename to avconv.h index 3783e2cc50..56ce9f01ff 100644 --- a/ffmpeg.h +++ b/avconv.h @@ -1,29 +1,28 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef FFMPEG_H -#define FFMPEG_H +#ifndef AVCONV_H +#define AVCONV_H #include "config.h" #include #include -#include #if HAVE_PTHREADS #include @@ -40,22 +39,14 @@ #include "libavutil/avutil.h" #include "libavutil/dict.h" -#include "libavutil/eval.h" #include "libavutil/fifo.h" #include "libavutil/pixfmt.h" #include "libavutil/rational.h" -#include "libavutil/threadmessage.h" - -#include "libswresample/swresample.h" #define VSYNC_AUTO -1 #define VSYNC_PASSTHROUGH 0 #define VSYNC_CFR 1 #define VSYNC_VFR 2 -#define VSYNC_VSCFR 0xfe -#define VSYNC_DROP 0xff - -#define MAX_STREAMS 1024 /* arbitrary sanity check value */ enum HWAccelID { HWACCEL_NONE = 0, @@ -82,10 +73,12 @@ typedef struct StreamMap { char *linklabel; /* name of an output link, for mapping lavfi outputs */ } StreamMap; -typedef struct { - int file_idx, stream_idx, channel_idx; // input - int ofile_idx, ostream_idx; // output -} AudioChannelMap; +/* select an input file for an output file */ +typedef struct MetadataMap { + int file; // file index + char type; // type of metadata to copy -- (g)lobal, (s)tream, (c)hapter or (p)rogram + int index; // stream/chapter/program number +} MetadataMap; typedef struct OptionsContext { OptionGroup *g; @@ -124,8 +117,9 @@ typedef struct OptionsContext { /* output options */ StreamMap *stream_maps; int nb_stream_maps; - AudioChannelMap *audio_channel_maps; /* one info entry per -map_channel */ - int nb_audio_channel_maps; /* number of (valid) -map_channel settings */ + /* first item specifies output metadata, second is input */ + MetadataMap (*meta_data_maps)[2]; + int nb_meta_data_maps; int metadata_global_manual; int metadata_streams_manual; int metadata_chapters_manual; @@ -135,7 +129,6 @@ typedef struct OptionsContext { int chapters_input_file; int64_t recording_time; - int64_t stop_time; uint64_t limit_filesize; float mux_preload; float mux_max_delay; @@ -174,8 +167,6 @@ typedef struct OptionsContext { int nb_intra_matrices; SpecifierOpt *inter_matrices; int nb_inter_matrices; - SpecifierOpt *chroma_intra_matrices; - int nb_chroma_intra_matrices; SpecifierOpt *top_field_first; int nb_top_field_first; SpecifierOpt *metadata_map; @@ -184,26 +175,14 @@ typedef struct OptionsContext { int nb_presets; SpecifierOpt *copy_initial_nonkeyframes; int nb_copy_initial_nonkeyframes; - SpecifierOpt *copy_prior_start; - int nb_copy_prior_start; SpecifierOpt *filters; int nb_filters; SpecifierOpt *filter_scripts; int nb_filter_scripts; - SpecifierOpt *reinit_filters; - int nb_reinit_filters; - SpecifierOpt *fix_sub_duration; - int nb_fix_sub_duration; - SpecifierOpt *canvas_sizes; - int nb_canvas_sizes; SpecifierOpt *pass; int nb_pass; SpecifierOpt *passlogfiles; int nb_passlogfiles; - SpecifierOpt *guess_layout_max; - int nb_guess_layout_max; - SpecifierOpt *apad; - int nb_apad; } OptionsContext; typedef struct InputFilter { @@ -228,7 +207,6 @@ typedef struct FilterGraph { const char *graph_desc; AVFilterGraph *graph; - int reconfiguration; InputFilter **inputs; int nb_inputs; @@ -247,23 +225,15 @@ typedef struct InputStream { int64_t start; /* time when read started */ /* predicted dts of the next packet read for this stream or (when there are - * several frames in a packet) of the next frame in current packet (in AV_TIME_BASE units) */ + * several frames in a packet) of the next frame in current packet */ int64_t next_dts; - int64_t dts; ///< dts of the last packet read for this stream (in AV_TIME_BASE units) - - int64_t next_pts; ///< synthetic pts for the next decode frame (in AV_TIME_BASE units) - int64_t pts; ///< current pts of the decoded frame (in AV_TIME_BASE units) - int wrap_correction_done; - - int64_t filter_in_rescale_delta_last; - + /* dts of the last packet read for this stream */ + int64_t last_dts; + PtsCorrectionContext pts_ctx; double ts_scale; - int saw_first_ts; int showed_multi_packet_warning; AVDictionary *decoder_opts; AVRational framerate; /* framerate forced with -r */ - int top_field_first; - int guess_layout_max; int resample_height; int resample_width; @@ -274,29 +244,11 @@ typedef struct InputStream { int resample_channels; uint64_t resample_channel_layout; - int fix_sub_duration; - struct { /* previous decoded subtitle and related variables */ - int got_output; - int ret; - AVSubtitle subtitle; - } prev_sub; - - struct sub2video { - int64_t last_pts; - int64_t end_pts; - AVFrame *frame; - int w, h; - } sub2video; - - int dr1; - /* decoded data from this stream goes into all those filters * currently video and audio only */ InputFilter **filters; int nb_filters; - int reinit_filters; - /* hwaccel options */ enum HWAccelID hwaccel_id; char *hwaccel_device; @@ -324,42 +276,25 @@ typedef struct InputFile { AVFormatContext *ctx; int eof_reached; /* true if eof reached */ int eagain; /* true if last read attempt returned EAGAIN */ - int ist_index; /* index of first stream in input_streams */ - int64_t input_ts_offset; + int ist_index; /* index of first stream in ist_table */ int64_t ts_offset; - int64_t last_ts; int64_t start_time; /* user-specified start time in AV_TIME_BASE or AV_NOPTS_VALUE */ int64_t recording_time; - int nb_streams; /* number of stream that ffmpeg is aware of; may be different + int nb_streams; /* number of stream that avconv is aware of; may be different from ctx.nb_streams if new streams appear during av_read_frame() */ - int nb_streams_warn; /* number of streams that the user was warned of */ int rate_emu; int accurate_seek; #if HAVE_PTHREADS - AVThreadMessageQueue *in_thread_queue; pthread_t thread; /* thread reading from this file */ - int non_blocking; /* reading packets from the thread should not block */ + int finished; /* the thread has exited */ int joined; /* the thread has been joined */ + pthread_mutex_t fifo_lock; /* lock for access to fifo */ + pthread_cond_t fifo_cond; /* the main thread will signal on this cond after reading from fifo */ + AVFifoBuffer *fifo; /* demuxed packets are stored here; freed by the main thread */ #endif } InputFile; -enum forced_keyframes_const { - FKF_N, - FKF_N_FORCED, - FKF_PREV_FORCED_N, - FKF_PREV_FORCED_T, - FKF_T, - FKF_NB -}; - -extern const char *const forced_keyframes_const_names[]; - -typedef enum { - ENCODER_FINISHED = 1, - MUXER_FINISHED = 2, -} OSTFinished ; - typedef struct OutputStream { int file_index; /* file index */ int index; /* stream index in the output file */ @@ -369,6 +304,7 @@ typedef struct OutputStream { int frame_number; /* input pts and corresponding output pts for A/V sync */ + // double sync_ipts; /* dts from the AVPacket of the demuxer in second units */ struct InputStream *sync_ist; /* input stream to sync against */ int64_t sync_opts; /* output frame counter, could be changed to some true timestamp */ // FIXME look at frame_number /* pts of the first frame encoded for this stream, used for limiting @@ -386,41 +322,29 @@ typedef struct OutputStream { int force_fps; int top_field_first; - AVRational frame_aspect_ratio; + float frame_aspect_ratio; /* forced key frames */ int64_t *forced_kf_pts; int forced_kf_count; int forced_kf_index; char *forced_keyframes; - AVExpr *forced_keyframes_pexpr; - double forced_keyframes_expr_const_values[FKF_NB]; - - /* audio only */ - int audio_channels_map[SWR_CH_MAX]; /* list of the channels id to pick from the source stream */ - int audio_channels_mapped; /* number of channels in audio_channels_map */ char *logfile_prefix; FILE *logfile; OutputFilter *filter; char *avfilter; - char *filters; ///< filtergraph associated to the -filter option - char *filters_script; ///< filtergraph script associated to the -filter_script option int64_t sws_flags; AVDictionary *encoder_opts; - AVDictionary *swr_opts; AVDictionary *resample_opts; - char *apad; - OSTFinished finished; /* no more packets should be written for this stream */ - int unavailable; /* true if the steram is unavailable (possibly temporarily) */ + int finished; /* no more packets should be written for this stream */ int stream_copy; const char *attachment_filename; int copy_initial_nonkeyframes; - int copy_prior_start; - int keep_pix_fmt; + enum AVPixelFormat pix_fmts[2]; AVCodecParserContext *parser; @@ -438,9 +362,9 @@ typedef struct OutputFile { AVFormatContext *ctx; AVDictionary *opts; int ost_index; /* index of the first stream in output_streams */ - int64_t recording_time; ///< desired length of the resulting file in microseconds == AV_TIME_BASE units - int64_t start_time; ///< start time in microseconds == AV_TIME_BASE units - uint64_t limit_filesize; /* filesize limit expressed in bytes */ + int64_t recording_time; /* desired length of the resulting file in microseconds */ + int64_t start_time; /* start time in microseconds */ + uint64_t limit_filesize; int shortest; } OutputFile; @@ -462,37 +386,27 @@ extern char *vstats_filename; extern float audio_drift_threshold; extern float dts_delta_threshold; -extern float dts_error_threshold; extern int audio_volume; extern int audio_sync_method; extern int video_sync_method; extern int do_benchmark; -extern int do_benchmark_all; extern int do_deinterlace; extern int do_hex_dump; extern int do_pkt_dump; extern int copy_ts; extern int copy_tb; -extern int debug_ts; extern int exit_on_error; extern int print_stats; extern int qp_hist; -extern int stdin_interaction; -extern int frame_bits_per_raw_sample; -extern AVIOContext *progress_avio; -extern float max_error_rate; extern const AVIOInterruptCB int_cb; extern const OptionDef options[]; + extern const HWAccel hwaccels[]; - -void term_init(void); -void term_exit(void); - -void reset_options(OptionsContext *o, int is_input); +void reset_options(OptionsContext *o); void show_usage(void); void opt_output_file(void *optctx, const char *filename); @@ -501,18 +415,15 @@ void assert_avoptions(AVDictionary *m); int guess_input_channel_layout(InputStream *ist); -enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFormat target); -void choose_sample_fmt(AVStream *st, AVCodec *codec); - int configure_filtergraph(FilterGraph *fg); int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out); int ist_in_filtergraph(FilterGraph *fg, InputStream *ist); FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost); -int ffmpeg_parse_options(int argc, char **argv); +int avconv_parse_options(int argc, char **argv); int vdpau_init(AVCodecContext *s); int dxva2_init(AVCodecContext *s); int vda_init(AVCodecContext *s); -#endif /* FFMPEG_H */ +#endif /* AVCONV_H */ diff --git a/ffmpeg_dxva2.c b/avconv_dxva2.c similarity index 98% rename from ffmpeg_dxva2.c rename to avconv_dxva2.c index 3c91d26f38..4457a21fb0 100644 --- a/ffmpeg_dxva2.c +++ b/avconv_dxva2.c @@ -1,18 +1,18 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -30,7 +30,7 @@ #include #include -#include "ffmpeg.h" +#include "avconv.h" #include "libavcodec/dxva2.h" @@ -121,10 +121,9 @@ static void dxva2_destroy_decoder(AVCodecContext *s) { InputStream *ist = s->opaque; DXVA2Context *ctx = ist->hwaccel_ctx; - int i; if (ctx->surfaces) { - for (i = 0; i < ctx->num_surfaces; i++) { + for (int i = 0; i < ctx->num_surfaces; i++) { if (ctx->surfaces[i]) IDirect3DSurface9_Release(ctx->surfaces[i]); } diff --git a/ffmpeg_filter.c b/avconv_filter.c similarity index 57% rename from ffmpeg_filter.c rename to avconv_filter.c index 6fd170baea..892db00f47 100644 --- a/ffmpeg_filter.c +++ b/avconv_filter.c @@ -1,149 +1,39 @@ /* - * ffmpeg filter configuration + * avconv filter configuration * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include -#include "ffmpeg.h" +#include "avconv.h" #include "libavfilter/avfilter.h" -#include "libavfilter/buffersink.h" #include "libavresample/avresample.h" #include "libavutil/avassert.h" #include "libavutil/avstring.h" -#include "libavutil/bprint.h" #include "libavutil/channel_layout.h" #include "libavutil/opt.h" #include "libavutil/pixdesc.h" #include "libavutil/pixfmt.h" -#include "libavutil/imgutils.h" #include "libavutil/samplefmt.h" -enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodec *codec, enum AVPixelFormat target) -{ - if (codec && codec->pix_fmts) { - const enum AVPixelFormat *p = codec->pix_fmts; - const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target); - int has_alpha = desc ? desc->nb_components % 2 == 0 : 0; - enum AVPixelFormat best= AV_PIX_FMT_NONE; - static const enum AVPixelFormat mjpeg_formats[] = - { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE }; - static const enum AVPixelFormat ljpeg_formats[] = - { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P, - AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE }; - - if (st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) { - if (st->codec->codec_id == AV_CODEC_ID_MJPEG) { - p = mjpeg_formats; - } else if (st->codec->codec_id == AV_CODEC_ID_LJPEG) { - p =ljpeg_formats; - } - } - for (; *p != AV_PIX_FMT_NONE; p++) { - best= avcodec_find_best_pix_fmt_of_2(best, *p, target, has_alpha, NULL); - if (*p == target) - break; - } - if (*p == AV_PIX_FMT_NONE) { - if (target != AV_PIX_FMT_NONE) - av_log(NULL, AV_LOG_WARNING, - "Incompatible pixel format '%s' for codec '%s', auto-selecting format '%s'\n", - av_get_pix_fmt_name(target), - codec->name, - av_get_pix_fmt_name(best)); - return best; - } - } - return target; -} - -void choose_sample_fmt(AVStream *st, AVCodec *codec) -{ - if (codec && codec->sample_fmts) { - const enum AVSampleFormat *p = codec->sample_fmts; - for (; *p != -1; p++) { - if (*p == st->codec->sample_fmt) - break; - } - if (*p == -1) { - if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0])) - av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n"); - if(av_get_sample_fmt_name(st->codec->sample_fmt)) - av_log(NULL, AV_LOG_WARNING, - "Incompatible sample format '%s' for codec '%s', auto-selecting format '%s'\n", - av_get_sample_fmt_name(st->codec->sample_fmt), - codec->name, - av_get_sample_fmt_name(codec->sample_fmts[0])); - st->codec->sample_fmt = codec->sample_fmts[0]; - } - } -} - -static char *choose_pix_fmts(OutputStream *ost) -{ - AVDictionaryEntry *strict_dict = av_dict_get(ost->encoder_opts, "strict", NULL, 0); - if (strict_dict) - // used by choose_pixel_fmt() and below - av_opt_set(ost->st->codec, "strict", strict_dict->value, 0); - - if (ost->keep_pix_fmt) { - if (ost->filter) - avfilter_graph_set_auto_convert(ost->filter->graph->graph, - AVFILTER_AUTO_CONVERT_NONE); - if (ost->st->codec->pix_fmt == AV_PIX_FMT_NONE) - return NULL; - return av_strdup(av_get_pix_fmt_name(ost->st->codec->pix_fmt)); - } - if (ost->st->codec->pix_fmt != AV_PIX_FMT_NONE) { - return av_strdup(av_get_pix_fmt_name(choose_pixel_fmt(ost->st, ost->enc, ost->st->codec->pix_fmt))); - } else if (ost->enc && ost->enc->pix_fmts) { - const enum AVPixelFormat *p; - AVIOContext *s = NULL; - uint8_t *ret; - int len; - - if (avio_open_dyn_buf(&s) < 0) - exit_program(1); - - p = ost->enc->pix_fmts; - if (ost->st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL) { - if (ost->st->codec->codec_id == AV_CODEC_ID_MJPEG) { - p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUV420P, AV_PIX_FMT_YUV422P, AV_PIX_FMT_NONE }; - } else if (ost->st->codec->codec_id == AV_CODEC_ID_LJPEG) { - p = (const enum AVPixelFormat[]) { AV_PIX_FMT_YUVJ420P, AV_PIX_FMT_YUVJ422P, AV_PIX_FMT_YUVJ444P, AV_PIX_FMT_YUV420P, - AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV444P, AV_PIX_FMT_BGRA, AV_PIX_FMT_NONE }; - } - } - - for (; *p != AV_PIX_FMT_NONE; p++) { - const char *name = av_get_pix_fmt_name(*p); - avio_printf(s, "%s|", name); - } - len = avio_close_dyn_buf(s, &ret); - ret[len - 1] = 0; - return ret; - } else - return NULL; -} - /* Define a function for building a string containing a list of * allowed formats. */ #define DEF_CHOOSE_FORMAT(type, var, supported_list, none, get_name) \ @@ -159,7 +49,7 @@ static char *choose_ ## var ## s(OutputStream *ost) \ int len; \ \ if (avio_open_dyn_buf(&s) < 0) \ - exit_program(1); \ + exit(1); \ \ for (p = ost->enc->supported_list; *p != none; p++) { \ get_name(*p); \ @@ -172,8 +62,8 @@ static char *choose_ ## var ## s(OutputStream *ost) \ return NULL; \ } -// DEF_CHOOSE_FORMAT(enum AVPixelFormat, pix_fmt, pix_fmts, AV_PIX_FMT_NONE, -// GET_PIX_FMT_NAME) +DEF_CHOOSE_FORMAT(enum AVPixelFormat, pix_fmt, pix_fmts, AV_PIX_FMT_NONE, + GET_PIX_FMT_NAME) DEF_CHOOSE_FORMAT(enum AVSampleFormat, sample_fmt, sample_fmts, AV_SAMPLE_FMT_NONE, GET_SAMPLE_FMT_NAME) @@ -189,12 +79,12 @@ FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost) FilterGraph *fg = av_mallocz(sizeof(*fg)); if (!fg) - exit_program(1); + exit(1); fg->index = nb_filtergraphs; GROW_ARRAY(fg->outputs, fg->nb_outputs); if (!(fg->outputs[0] = av_mallocz(sizeof(*fg->outputs[0])))) - exit_program(1); + exit(1); fg->outputs[0]->ost = ost; fg->outputs[0]->graph = fg; @@ -202,7 +92,7 @@ FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost) GROW_ARRAY(fg->inputs, fg->nb_inputs); if (!(fg->inputs[0] = av_mallocz(sizeof(*fg->inputs[0])))) - exit_program(1); + exit(1); fg->inputs[0]->ist = ist; fg->inputs[0]->graph = fg; @@ -225,7 +115,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in) if (type != AVMEDIA_TYPE_VIDEO && type != AVMEDIA_TYPE_AUDIO) { av_log(NULL, AV_LOG_FATAL, "Only video and audio filters supported " "currently.\n"); - exit_program(1); + exit(1); } if (in->name) { @@ -235,17 +125,14 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in) int file_idx = strtol(in->name, &p, 0); if (file_idx < 0 || file_idx >= nb_input_files) { - av_log(NULL, AV_LOG_FATAL, "Invalid file index %d in filtergraph description %s.\n", + av_log(NULL, AV_LOG_FATAL, "Invalid file index %d in filtegraph description %s.\n", file_idx, fg->graph_desc); - exit_program(1); + exit(1); } s = input_files[file_idx]->ctx; for (i = 0; i < s->nb_streams; i++) { - enum AVMediaType stream_type = s->streams[i]->codec->codec_type; - if (stream_type != type && - !(stream_type == AVMEDIA_TYPE_SUBTITLE && - type == AVMEDIA_TYPE_VIDEO /* sub2video hack */)) + if (s->streams[i]->codec->codec_type != type) continue; if (check_stream_specifier(s, s->streams[i], *p == ':' ? p + 1 : p) == 1) { st = s->streams[i]; @@ -255,7 +142,7 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in) if (!st) { av_log(NULL, AV_LOG_FATAL, "Stream specifier '%s' in filtergraph description %s " "matches no streams.\n", p, fg->graph_desc); - exit_program(1); + exit(1); } ist = input_streams[input_files[file_idx]->ist_index + st->index]; } else { @@ -269,18 +156,18 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in) av_log(NULL, AV_LOG_FATAL, "Cannot find a matching stream for " "unlabeled input pad %d on filter %s\n", in->pad_idx, in->filter_ctx->name); - exit_program(1); + exit(1); } } av_assert0(ist); ist->discard = 0; - ist->decoding_needed++; + ist->decoding_needed = 1; ist->st->discard = AVDISCARD_NONE; GROW_ARRAY(fg->inputs, fg->nb_inputs); if (!(fg->inputs[fg->nb_inputs - 1] = av_mallocz(sizeof(*fg->inputs[0])))) - exit_program(1); + exit(1); fg->inputs[fg->nb_inputs - 1]->ist = ist; fg->inputs[fg->nb_inputs - 1]->graph = fg; @@ -314,11 +201,11 @@ static int insert_trim(int64_t start_time, int64_t duration, return AVERROR(ENOMEM); if (duration != INT64_MAX) { - ret = av_opt_set_int(ctx, "durationi", duration, + ret = av_opt_set_double(ctx, "duration", (double)duration / 1e6, AV_OPT_SEARCH_CHILDREN); } if (ret >= 0 && start_time != AV_NOPTS_VALUE) { - ret = av_opt_set_int(ctx, "starti", start_time, + ret = av_opt_set_double(ctx, "start", (double)start_time / 1e6, AV_OPT_SEARCH_CHILDREN); } if (ret < 0) { @@ -354,7 +241,6 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, ret = avfilter_graph_create_filter(&ofilter->filter, avfilter_get_by_name("buffersink"), name, NULL, NULL, fg->graph); - if (ret < 0) return ret; @@ -382,21 +268,20 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterContext *filter; snprintf(name, sizeof(name), "pixel format for output stream %d:%d", ost->file_index, ost->index); - ret = avfilter_graph_create_filter(&filter, + if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("format"), "format", pix_fmts, NULL, - fg->graph); - av_freep(&pix_fmts); - if (ret < 0) + fg->graph)) < 0) return ret; if ((ret = avfilter_link(last_filter, pad_idx, filter, 0)) < 0) return ret; last_filter = filter; pad_idx = 0; + av_freep(&pix_fmts); } - if (ost->frame_rate.num && 0) { + if (ost->frame_rate.num) { AVFilterContext *fps; char args[255]; @@ -441,47 +326,13 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, char name[255]; int ret; + snprintf(name, sizeof(name), "output stream %d:%d", ost->file_index, ost->index); ret = avfilter_graph_create_filter(&ofilter->filter, avfilter_get_by_name("abuffersink"), name, NULL, NULL, fg->graph); if (ret < 0) return ret; - if ((ret = av_opt_set_int(ofilter->filter, "all_channel_counts", 1, AV_OPT_SEARCH_CHILDREN)) < 0) - return ret; - -#define AUTO_INSERT_FILTER(opt_name, filter_name, arg) do { \ - AVFilterContext *filt_ctx; \ - \ - av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi " \ - "similarly to -af " filter_name "=%s.\n", arg); \ - \ - ret = avfilter_graph_create_filter(&filt_ctx, \ - avfilter_get_by_name(filter_name), \ - filter_name, arg, NULL, fg->graph); \ - if (ret < 0) \ - return ret; \ - \ - ret = avfilter_link(last_filter, pad_idx, filt_ctx, 0); \ - if (ret < 0) \ - return ret; \ - \ - last_filter = filt_ctx; \ - pad_idx = 0; \ -} while (0) - if (ost->audio_channels_mapped) { - int i; - AVBPrint pan_buf; - av_bprint_init(&pan_buf, 256, 8192); - av_bprintf(&pan_buf, "0x%"PRIx64, - av_get_default_channel_layout(ost->audio_channels_mapped)); - for (i = 0; i < ost->audio_channels_mapped; i++) - if (ost->audio_channels_map[i] != -1) - av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]); - - AUTO_INSERT_FILTER("-map_channel", "pan", pan_buf.str); - av_bprint_finalize(&pan_buf, NULL); - } if (codec->channels && !codec->channel_layout) codec->channel_layout = av_get_default_channel_layout(codec->channels); @@ -492,17 +343,18 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, if (sample_fmts || sample_rates || channel_layouts) { AVFilterContext *format; char args[256]; - args[0] = 0; + int len = 0; if (sample_fmts) - av_strlcatf(args, sizeof(args), "sample_fmts=%s:", + len += snprintf(args + len, sizeof(args) - len, "sample_fmts=%s:", sample_fmts); if (sample_rates) - av_strlcatf(args, sizeof(args), "sample_rates=%s:", + len += snprintf(args + len, sizeof(args) - len, "sample_rates=%s:", sample_rates); if (channel_layouts) - av_strlcatf(args, sizeof(args), "channel_layouts=%s:", + len += snprintf(args + len, sizeof(args) - len, "channel_layouts=%s:", channel_layouts); + args[len - 1] = 0; av_freep(&sample_fmts); av_freep(&sample_rates); @@ -524,27 +376,6 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, pad_idx = 0; } - if (audio_volume != 256 && 0) { - char args[256]; - - snprintf(args, sizeof(args), "%f", audio_volume / 256.); - AUTO_INSERT_FILTER("-vol", "volume", args); - } - - if (ost->apad && of->shortest) { - char args[256]; - int i; - - for (i=0; ictx->nb_streams; i++) - if (of->ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) - break; - - if (ictx->nb_streams) { - snprintf(args, sizeof(args), "%s", ost->apad); - AUTO_INSERT_FILTER("-apad", "apad", args); - } - } - snprintf(name, sizeof(name), "trim for output stream %d:%d", ost->file_index, ost->index); ret = insert_trim(of->start_time, of->recording_time, @@ -566,7 +397,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter, AVIOContext *pb; \ \ if (avio_open_dyn_buf(&pb) < 0) \ - exit_program(1); \ + exit(1); \ \ avio_printf(pb, "%s", ctx->filter->name); \ if (nb_pads > 1) \ @@ -587,42 +418,6 @@ int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOu } } -static int sub2video_prepare(InputStream *ist) -{ - AVFormatContext *avf = input_files[ist->file_index]->ctx; - int i, w, h; - - /* Compute the size of the canvas for the subtitles stream. - If the subtitles codec has set a size, use it. Otherwise use the - maximum dimensions of the video streams in the same file. */ - w = ist->st->codec->width; - h = ist->st->codec->height; - if (!(w && h)) { - for (i = 0; i < avf->nb_streams; i++) { - if (avf->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) { - w = FFMAX(w, avf->streams[i]->codec->width); - h = FFMAX(h, avf->streams[i]->codec->height); - } - } - if (!(w && h)) { - w = FFMAX(w, 720); - h = FFMAX(h, 576); - } - av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h); - } - ist->sub2video.w = ist->st->codec->width = ist->resample_width = w; - ist->sub2video.h = ist->st->codec->height = ist->resample_height = h; - - /* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the - palettes for all rectangles are identical or compatible */ - ist->resample_pix_fmt = ist->st->codec->pix_fmt = AV_PIX_FMT_RGB32; - - ist->sub2video.frame = av_frame_alloc(); - if (!ist->sub2video.frame) - return AVERROR(ENOMEM); - return 0; -} - static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter, AVFilterInOut *in) { @@ -632,46 +427,22 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter, InputFile *f = input_files[ist->file_index]; AVRational tb = ist->framerate.num ? av_inv_q(ist->framerate) : ist->st->time_base; - AVRational fr = ist->framerate; AVRational sar; - AVBPrint args; - char name[255]; + char args[255], name[255]; int ret, pad_idx = 0; - if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - av_log(NULL, AV_LOG_ERROR, "Cannot connect video filter to audio input\n"); - return AVERROR(EINVAL); - } - - if (!fr.num) - fr = av_guess_frame_rate(input_files[ist->file_index]->ctx, ist->st, NULL); - - if (ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) { - ret = sub2video_prepare(ist); - if (ret < 0) - return ret; - } - sar = ist->st->sample_aspect_ratio.num ? ist->st->sample_aspect_ratio : ist->st->codec->sample_aspect_ratio; - if(!sar.den) - sar = (AVRational){0,1}; - av_bprint_init(&args, 0, 1); - av_bprintf(&args, - "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:" - "pixel_aspect=%d/%d:sws_param=flags=%d", ist->resample_width, - ist->resample_height, - ist->hwaccel_retrieve_data ? ist->hwaccel_retrieved_pix_fmt : ist->resample_pix_fmt, - tb.num, tb.den, sar.num, sar.den, - SWS_BILINEAR + ((ist->st->codec->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0)); - if (fr.num && fr.den) - av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den); + snprintf(args, sizeof(args), "%d:%d:%d:%d:%d:%d:%d", ist->st->codec->width, + ist->st->codec->height, + ist->hwaccel_retrieve_data ? ist->hwaccel_retrieved_pix_fmt : ist->st->codec->pix_fmt, + tb.num, tb.den, sar.num, sar.den); snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index, ist->file_index, ist->st->index); if ((ret = avfilter_graph_create_filter(&ifilter->filter, buffer_filt, name, - args.str, NULL, fg->graph)) < 0) + args, NULL, fg->graph)) < 0) return ret; last_filter = ifilter->filter; @@ -692,23 +463,6 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter, last_filter = setpts; } - if (do_deinterlace) { - AVFilterContext *yadif; - - snprintf(name, sizeof(name), "deinterlace input from stream %d:%d", - ist->file_index, ist->st->index); - if ((ret = avfilter_graph_create_filter(&yadif, - avfilter_get_by_name("yadif"), - name, "", NULL, - fg->graph)) < 0) - return ret; - - if ((ret = avfilter_link(last_filter, 0, yadif, 0)) < 0) - return ret; - - last_filter = yadif; - } - snprintf(name, sizeof(name), "trim for input stream %d:%d", ist->file_index, ist->st->index); ret = insert_trim(((f->start_time == AV_NOPTS_VALUE) || !f->accurate_seek) ? @@ -728,87 +482,72 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter, const AVFilter *abuffer_filt = avfilter_get_by_name("abuffer"); InputStream *ist = ifilter->ist; InputFile *f = input_files[ist->file_index]; - AVBPrint args; - char name[255]; + char args[255], name[255]; int ret, pad_idx = 0; - if (ist->st->codec->codec_type != AVMEDIA_TYPE_AUDIO) { - av_log(NULL, AV_LOG_ERROR, "Cannot connect audio filter to non audio input\n"); - return AVERROR(EINVAL); - } - - av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC); - av_bprintf(&args, "time_base=%d/%d:sample_rate=%d:sample_fmt=%s", + snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s" + ":channel_layout=0x%"PRIx64, 1, ist->st->codec->sample_rate, ist->st->codec->sample_rate, - av_get_sample_fmt_name(ist->st->codec->sample_fmt)); - if (ist->st->codec->channel_layout) - av_bprintf(&args, ":channel_layout=0x%"PRIx64, - ist->st->codec->channel_layout); - else - av_bprintf(&args, ":channels=%d", ist->st->codec->channels); + av_get_sample_fmt_name(ist->st->codec->sample_fmt), + ist->st->codec->channel_layout); snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index, ist->file_index, ist->st->index); if ((ret = avfilter_graph_create_filter(&ifilter->filter, abuffer_filt, - name, args.str, NULL, + name, args, NULL, fg->graph)) < 0) return ret; last_filter = ifilter->filter; -#define AUTO_INSERT_FILTER_INPUT(opt_name, filter_name, arg) do { \ - AVFilterContext *filt_ctx; \ - \ - av_log(NULL, AV_LOG_INFO, opt_name " is forwarded to lavfi " \ - "similarly to -af " filter_name "=%s.\n", arg); \ - \ - snprintf(name, sizeof(name), "graph %d %s for input stream %d:%d", \ - fg->index, filter_name, ist->file_index, ist->st->index); \ - ret = avfilter_graph_create_filter(&filt_ctx, \ - avfilter_get_by_name(filter_name), \ - name, arg, NULL, fg->graph); \ - if (ret < 0) \ - return ret; \ - \ - ret = avfilter_link(last_filter, 0, filt_ctx, 0); \ - if (ret < 0) \ - return ret; \ - \ - last_filter = filt_ctx; \ -} while (0) - if (audio_sync_method > 0) { - char args[256] = {0}; + AVFilterContext *async; + int len = 0; - av_strlcatf(args, sizeof(args), "async=%d", audio_sync_method); - if (audio_drift_threshold != 0.1) - av_strlcatf(args, sizeof(args), ":min_hard_comp=%f", audio_drift_threshold); - if (!fg->reconfiguration) - av_strlcatf(args, sizeof(args), ":first_pts=0"); - AUTO_INSERT_FILTER_INPUT("-async", "aresample", args); + av_log(NULL, AV_LOG_WARNING, "-async has been deprecated. Used the " + "asyncts audio filter instead.\n"); + + if (audio_sync_method > 1) + len += snprintf(args + len, sizeof(args) - len, "compensate=1:" + "max_comp=%d:", audio_sync_method); + snprintf(args + len, sizeof(args) - len, "min_delta=%f", + audio_drift_threshold); + + snprintf(name, sizeof(name), "graph %d audio sync for input stream %d:%d", + fg->index, ist->file_index, ist->st->index); + ret = avfilter_graph_create_filter(&async, + avfilter_get_by_name("asyncts"), + name, args, NULL, fg->graph); + if (ret < 0) + return ret; + + ret = avfilter_link(last_filter, 0, async, 0); + if (ret < 0) + return ret; + + last_filter = async; } - -// if (ost->audio_channels_mapped) { -// int i; -// AVBPrint pan_buf; -// av_bprint_init(&pan_buf, 256, 8192); -// av_bprintf(&pan_buf, "0x%"PRIx64, -// av_get_default_channel_layout(ost->audio_channels_mapped)); -// for (i = 0; i < ost->audio_channels_mapped; i++) -// if (ost->audio_channels_map[i] != -1) -// av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]); -// AUTO_INSERT_FILTER_INPUT("-map_channel", "pan", pan_buf.str); -// av_bprint_finalize(&pan_buf, NULL); -// } - if (audio_volume != 256) { - char args[256]; + AVFilterContext *volume; av_log(NULL, AV_LOG_WARNING, "-vol has been deprecated. Use the volume " "audio filter instead.\n"); - snprintf(args, sizeof(args), "%f", audio_volume / 256.); - AUTO_INSERT_FILTER_INPUT("-vol", "volume", args); + snprintf(args, sizeof(args), "volume=%f", audio_volume / 256.0); + + snprintf(name, sizeof(name), "graph %d volume for input stream %d:%d", + fg->index, ist->file_index, ist->st->index); + ret = avfilter_graph_create_filter(&volume, + avfilter_get_by_name("volume"), + name, args, NULL, fg->graph); + if (ret < 0) + return ret; + + ret = avfilter_link(last_filter, 0, volume, 0); + if (ret < 0) + return ret; + + last_filter = volume; } snprintf(name, sizeof(name), "trim for input stream %d:%d", @@ -856,15 +595,6 @@ int configure_filtergraph(FilterGraph *fg) snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags); fg->graph->scale_sws_opts = av_strdup(args); - args[0] = 0; - while ((e = av_dict_get(ost->swr_opts, "", e, - AV_DICT_IGNORE_SUFFIX))) { - av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value); - } - if (strlen(args)) - args[strlen(args)-1] = 0; - av_opt_set(fg->graph, "aresample_swr_opts", args, 0); - args[0] = '\0'; while ((e = av_dict_get(fg->outputs[0]->ost->resample_opts, "", e, AV_DICT_IGNORE_SUFFIX))) { @@ -873,10 +603,6 @@ int configure_filtergraph(FilterGraph *fg) if (strlen(args)) args[strlen(args) - 1] = '\0'; fg->graph->resample_lavr_opts = av_strdup(args); - - e = av_dict_get(ost->encoder_opts, "threads", NULL, 0); - if (e) - av_opt_set(fg->graph, "threads", e->value, 0); } if ((ret = avfilter_graph_parse2(fg->graph, graph_desc, &inputs, &outputs)) < 0) @@ -910,7 +636,7 @@ int configure_filtergraph(FilterGraph *fg) for (cur = outputs; cur;) { GROW_ARRAY(fg->outputs, fg->nb_outputs); if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0])))) - exit_program(1); + exit(1); fg->outputs[fg->nb_outputs - 1]->graph = fg; fg->outputs[fg->nb_outputs - 1]->out_tmp = cur; cur = cur->next; @@ -918,7 +644,6 @@ int configure_filtergraph(FilterGraph *fg) } } - fg->reconfiguration = 1; return 0; } diff --git a/ffmpeg_opt.c b/avconv_opt.c similarity index 69% rename from ffmpeg_opt.c rename to avconv_opt.c index eeacf5828c..12d676054d 100644 --- a/ffmpeg_opt.c +++ b/avconv_opt.c @@ -1,26 +1,26 @@ /* - * ffmpeg option parsing + * avconv option parsing * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include -#include "ffmpeg.h" +#include "avconv.h" #include "cmdutils.h" #include "libavformat/avformat.h" @@ -53,16 +53,6 @@ }\ } -#define MATCH_PER_TYPE_OPT(name, type, outvar, fmtctx, mediatype)\ -{\ - int i;\ - for (i = 0; i < o->nb_ ## name; i++) {\ - char *spec = o->name[i].specifier;\ - if (!strcmp(spec, mediatype))\ - outvar = o->name[i].u.type;\ - }\ -} - const HWAccel hwaccels[] = { #if HAVE_VDPAU_X11 { "vdpau", vdpau_init, HWACCEL_VDPAU, AV_PIX_FMT_VDPAU }, @@ -80,35 +70,25 @@ char *vstats_filename; float audio_drift_threshold = 0.1; float dts_delta_threshold = 10; -float dts_error_threshold = 3600*30; int audio_volume = 256; int audio_sync_method = 0; int video_sync_method = VSYNC_AUTO; -int do_deinterlace = 0; int do_benchmark = 0; -int do_benchmark_all = 0; int do_hex_dump = 0; int do_pkt_dump = 0; int copy_ts = 0; -int copy_tb = -1; -int debug_ts = 0; +int copy_tb = 1; int exit_on_error = 0; -int print_stats = -1; +int print_stats = 1; int qp_hist = 0; -int stdin_interaction = 1; -int frame_bits_per_raw_sample = 0; -float max_error_rate = 2.0/3; - -static int intra_only = 0; static int file_overwrite = 0; -static int no_file_overwrite = 0; +static int file_skip = 0; static int video_discard = 0; static int intra_dc_precision = 8; -static int do_psnr = 0; +static int using_stdin = 0; static int input_sync; -static int override_ffserver = 0; static void uninit_options(OptionsContext *o) { @@ -137,16 +117,14 @@ static void uninit_options(OptionsContext *o) for (i = 0; i < o->nb_stream_maps; i++) av_freep(&o->stream_maps[i].linklabel); av_freep(&o->stream_maps); - av_freep(&o->audio_channel_maps); + av_freep(&o->meta_data_maps); av_freep(&o->streamid_map); - av_freep(&o->attachments); } static void init_options(OptionsContext *o) { memset(o, 0, sizeof(*o)); - o->stop_time = INT64_MAX; o->mux_max_delay = 0.7; o->start_time = AV_NOPTS_VALUE; o->recording_time = INT64_MAX; @@ -173,25 +151,28 @@ static AVDictionary *strip_specifiers(AVDictionary *dict) return ret; } -static int opt_sameq(void *optctx, const char *opt, const char *arg) +static double parse_frame_aspect_ratio(const char *arg) { - av_log(NULL, AV_LOG_ERROR, "Option '%s' was removed. " - "If you are looking for an option to preserve the quality (which is not " - "what -%s was for), use -qscale 0 or an equivalent quality factor option.\n", - opt, opt); - return AVERROR(EINVAL); -} + int x = 0, y = 0; + double ar = 0; + const char *p; + char *end; -static int opt_video_channel(void *optctx, const char *opt, const char *arg) -{ - av_log(NULL, AV_LOG_WARNING, "This option is deprecated, use -channel.\n"); - return opt_default(optctx, "channel", arg); -} + p = strchr(arg, ':'); + if (p) { + x = strtol(arg, &end, 10); + if (end == p) + y = strtol(end + 1, &end, 10); + if (x > 0 && y > 0) + ar = (double)x / (double)y; + } else + ar = strtod(arg, NULL); -static int opt_video_standard(void *optctx, const char *opt, const char *arg) -{ - av_log(NULL, AV_LOG_WARNING, "This option is deprecated, use -standard.\n"); - return opt_default(optctx, "standard", arg); + if (!ar) { + av_log(NULL, AV_LOG_FATAL, "Incorrect aspect ratio specification.\n"); + exit_program(1); + } + return ar; } static int opt_audio_codec(void *optctx, const char *opt, const char *arg) @@ -223,7 +204,7 @@ static int opt_map(void *optctx, const char *opt, const char *arg) OptionsContext *o = optctx; StreamMap *m = NULL; int i, negative = 0, file_idx; - int sync_file_idx = -1, sync_stream_idx = 0; + int sync_file_idx = -1, sync_stream_idx; char *p, *sync; char *map; @@ -321,65 +302,6 @@ static int opt_attach(void *optctx, const char *opt, const char *arg) return 0; } -static int opt_map_channel(void *optctx, const char *opt, const char *arg) -{ - OptionsContext *o = optctx; - int n; - AVStream *st; - AudioChannelMap *m; - - GROW_ARRAY(o->audio_channel_maps, o->nb_audio_channel_maps); - m = &o->audio_channel_maps[o->nb_audio_channel_maps - 1]; - - /* muted channel syntax */ - n = sscanf(arg, "%d:%d.%d", &m->channel_idx, &m->ofile_idx, &m->ostream_idx); - if ((n == 1 || n == 3) && m->channel_idx == -1) { - m->file_idx = m->stream_idx = -1; - if (n == 1) - m->ofile_idx = m->ostream_idx = -1; - return 0; - } - - /* normal syntax */ - n = sscanf(arg, "%d.%d.%d:%d.%d", - &m->file_idx, &m->stream_idx, &m->channel_idx, - &m->ofile_idx, &m->ostream_idx); - - if (n != 3 && n != 5) { - av_log(NULL, AV_LOG_FATAL, "Syntax error, mapchan usage: " - "[file.stream.channel|-1][:syncfile:syncstream]\n"); - exit_program(1); - } - - if (n != 5) // only file.stream.channel specified - m->ofile_idx = m->ostream_idx = -1; - - /* check input */ - if (m->file_idx < 0 || m->file_idx >= nb_input_files) { - av_log(NULL, AV_LOG_FATAL, "mapchan: invalid input file index: %d\n", - m->file_idx); - exit_program(1); - } - if (m->stream_idx < 0 || - m->stream_idx >= input_files[m->file_idx]->nb_streams) { - av_log(NULL, AV_LOG_FATAL, "mapchan: invalid input file stream index #%d.%d\n", - m->file_idx, m->stream_idx); - exit_program(1); - } - st = input_files[m->file_idx]->ctx->streams[m->stream_idx]; - if (st->codec->codec_type != AVMEDIA_TYPE_AUDIO) { - av_log(NULL, AV_LOG_FATAL, "mapchan: stream #%d.%d is not an audio stream.\n", - m->file_idx, m->stream_idx); - exit_program(1); - } - if (m->channel_idx < 0 || m->channel_idx >= st->codec->channels) { - av_log(NULL, AV_LOG_FATAL, "mapchan: invalid audio channel #%d.%d.%d\n", - m->file_idx, m->stream_idx, m->channel_idx); - exit_program(1); - } - return 0; -} - /** * Parse a metadata specifier passed as 'arg' parameter. * @param arg metadata string to parse @@ -417,7 +339,7 @@ static void parse_meta_type(char *arg, char *type, int *index, const char **stre static int copy_metadata(char *outspec, char *inspec, AVFormatContext *oc, AVFormatContext *ic, OptionsContext *o) { AVDictionary **meta_in = NULL; - AVDictionary **meta_out = NULL; + AVDictionary **meta_out; int i, ret = 0; char type_in, type_out; const char *istream_spec = NULL, *ostream_spec = NULL; @@ -426,16 +348,6 @@ static int copy_metadata(char *outspec, char *inspec, AVFormatContext *oc, AVFor parse_meta_type(inspec, &type_in, &idx_in, &istream_spec); parse_meta_type(outspec, &type_out, &idx_out, &ostream_spec); - if (!ic) { - if (type_out == 'g' || !*outspec) - o->metadata_global_manual = 1; - if (type_out == 's' || !*outspec) - o->metadata_streams_manual = 1; - if (type_out == 'c' || !*outspec) - o->metadata_chapters_manual = 1; - return 0; - } - if (type_in == 'g' || type_out == 'g') o->metadata_global_manual = 1; if (type_in == 's' || type_out == 's') @@ -504,20 +416,6 @@ static int copy_metadata(char *outspec, char *inspec, AVFormatContext *oc, AVFor return 0; } -static int opt_recording_timestamp(void *optctx, const char *opt, const char *arg) -{ - OptionsContext *o = optctx; - char buf[128]; - int64_t recording_timestamp = parse_time_or_die(opt, arg, 0) / 1E6; - struct tm time = *gmtime((time_t*)&recording_timestamp); - strftime(buf, sizeof(buf), "creation_time=%FT%T%z", &time); - parse_option(o, "metadata", buf, options); - - av_log(NULL, AV_LOG_WARNING, "%s is deprecated, set the 'creation_time' metadata " - "tag instead.\n", opt); - return 0; -} - static AVCodec *find_codec_or_die(const char *name, enum AVMediaType type, int encoder) { const AVCodecDescriptor *desc; @@ -599,19 +497,8 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic) ist->dec = choose_decoder(o, ic, st); ist->decoder_opts = filter_codec_opts(o->g->codec_opts, ist->st->codec->codec_id, ic, st, ist->dec); - ist->reinit_filters = -1; - MATCH_PER_STREAM_OPT(reinit_filters, i, ist->reinit_filters, ic, st); - - ist->filter_in_rescale_delta_last = AV_NOPTS_VALUE; - switch (dec->codec_type) { case AVMEDIA_TYPE_VIDEO: - if(!ist->dec) - ist->dec = avcodec_find_decoder(dec->codec_id); - if (av_codec_get_lowres(dec)) { - dec->flags |= CODEC_FLAG_EMU_EDGE; - } - ist->resample_height = dec->height; ist->resample_width = dec->width; ist->resample_pix_fmt = dec->pix_fmt; @@ -624,9 +511,6 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic) exit_program(1); } - ist->top_field_first = -1; - MATCH_PER_STREAM_OPT(top_field_first, i, ist->top_field_first, ic, st); - MATCH_PER_STREAM_OPT(hwaccels, str, hwaccel, ic, st); if (hwaccel) { if (!strcmp(hwaccel, "none")) @@ -664,8 +548,6 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic) break; case AVMEDIA_TYPE_AUDIO: - ist->guess_layout_max = INT_MAX; - MATCH_PER_STREAM_OPT(guess_layout_max, i, ist->guess_layout_max, ic, st); guess_input_channel_layout(ist); ist->resample_sample_fmt = dec->sample_fmt; @@ -675,19 +557,7 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic) break; case AVMEDIA_TYPE_DATA: - case AVMEDIA_TYPE_SUBTITLE: { - char *canvas_size = NULL; - if(!ist->dec) - ist->dec = avcodec_find_decoder(dec->codec_id); - MATCH_PER_STREAM_OPT(fix_sub_duration, i, ist->fix_sub_duration, ic, st); - MATCH_PER_STREAM_OPT(canvas_sizes, str, canvas_size, ic, st); - if (canvas_size && - av_parse_video_size(&dec->width, &dec->height, canvas_size) < 0) { - av_log(NULL, AV_LOG_FATAL, "Invalid canvas size: %s.\n", canvas_size); - exit_program(1); - } - break; - } + case AVMEDIA_TYPE_SUBTITLE: case AVMEDIA_TYPE_ATTACHMENT: case AVMEDIA_TYPE_UNKNOWN: break; @@ -699,27 +569,25 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic) static void assert_file_overwrite(const char *filename) { - if (file_overwrite && no_file_overwrite) { + if (file_overwrite && file_skip) { fprintf(stderr, "Error, both -y and -n supplied. Exiting.\n"); exit_program(1); } - if (!file_overwrite) { - const char *proto_name = avio_find_protocol_name(filename); - if (proto_name && !strcmp(proto_name, "file") && avio_check(filename, 0) == 0) { - if (stdin_interaction && !no_file_overwrite) { + if (!file_overwrite && + (strchr(filename, ':') == NULL || filename[1] == ':' || + av_strstart(filename, "file:", NULL))) { + if (avio_check(filename, 0) == 0) { + if (!using_stdin && !file_skip) { fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename); fflush(stderr); - term_exit(); - signal(SIGINT, SIG_DFL); if (!read_yesno()) { - av_log(NULL, AV_LOG_FATAL, "Not overwriting - exiting\n"); + fprintf(stderr, "Not overwriting - exiting\n"); exit_program(1); } - term_init(); } else { - av_log(NULL, AV_LOG_FATAL, "File '%s' already exists. Exiting.\n", filename); + fprintf(stderr,"File '%s' already exists. Exiting.\n", filename); exit_program(1); } } @@ -770,9 +638,6 @@ static int open_input_file(OptionsContext *o, const char *filename) AVDictionary *unused_opts = NULL; AVDictionaryEntry *e = NULL; int orig_nb_streams; // number of streams before avformat_find_stream_info - char * video_codec_name = NULL; - char * audio_codec_name = NULL; - char *subtitle_codec_name = NULL; if (o->format) { if (!(file_iformat = av_find_input_format(o->format))) { @@ -784,8 +649,8 @@ static int open_input_file(OptionsContext *o, const char *filename) if (!strcmp(filename, "-")) filename = "pipe:"; - stdin_interaction &= strncmp(filename, "pipe:", 5) && - strcmp(filename, "/dev/stdin"); + using_stdin |= !strncmp(filename, "pipe:", 5) || + !strcmp(filename, "/dev/stdin"); /* get default parameters from command line */ ic = avformat_alloc_context(); @@ -825,28 +690,10 @@ static int open_input_file(OptionsContext *o, const char *filename) if (o->nb_frame_pix_fmts) av_dict_set(&o->g->format_opts, "pixel_format", o->frame_pix_fmts[o->nb_frame_pix_fmts - 1].u.str, 0); - MATCH_PER_TYPE_OPT(codec_names, str, video_codec_name, ic, "v"); - MATCH_PER_TYPE_OPT(codec_names, str, audio_codec_name, ic, "a"); - MATCH_PER_TYPE_OPT(codec_names, str, subtitle_codec_name, ic, "s"); - - ic->video_codec_id = video_codec_name ? - find_codec_or_die(video_codec_name , AVMEDIA_TYPE_VIDEO , 0)->id : AV_CODEC_ID_NONE; - ic->audio_codec_id = audio_codec_name ? - find_codec_or_die(audio_codec_name , AVMEDIA_TYPE_AUDIO , 0)->id : AV_CODEC_ID_NONE; - ic->subtitle_codec_id= subtitle_codec_name ? - find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0)->id : AV_CODEC_ID_NONE; - - if (video_codec_name) - av_format_set_video_codec (ic, find_codec_or_die(video_codec_name , AVMEDIA_TYPE_VIDEO , 0)); - if (audio_codec_name) - av_format_set_audio_codec (ic, find_codec_or_die(audio_codec_name , AVMEDIA_TYPE_AUDIO , 0)); - if (subtitle_codec_name) - av_format_set_subtitle_codec(ic, find_codec_or_die(subtitle_codec_name, AVMEDIA_TYPE_SUBTITLE, 0)); - ic->flags |= AVFMT_FLAG_NONBLOCK; ic->interrupt_callback = int_cb; - /* open the input file with generic avformat function */ + /* open the input file with generic libav function */ err = avformat_open_input(&ic, filename, file_iformat, &o->g->format_opts); if (err < 0) { print_error(filename, err); @@ -878,7 +725,7 @@ static int open_input_file(OptionsContext *o, const char *filename) /* if seeking requested, we execute it */ if (o->start_time != AV_NOPTS_VALUE) { - ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, timestamp, 0); + ret = av_seek_frame(ic, -1, timestamp, AVSEEK_FLAG_BACKWARD); if (ret < 0) { av_log(NULL, AV_LOG_WARNING, "%s: could not seek to position %0.3f\n", filename, (double)timestamp / AV_TIME_BASE); @@ -901,7 +748,6 @@ static int open_input_file(OptionsContext *o, const char *filename) f->ist_index = nb_input_streams - ic->nb_streams; f->start_time = o->start_time; f->recording_time = o->recording_time; - f->input_ts_offset = o->input_ts_offset; f->ts_offset = o->input_ts_offset - (copy_ts ? 0 : timestamp); f->nb_streams = ic->nb_streams; f->rate_emu = o->rate_emu; @@ -921,13 +767,8 @@ static int open_input_file(OptionsContext *o, const char *filename) const AVClass *class = avcodec_get_class(); const AVOption *option = av_opt_find(&class, e->key, NULL, 0, AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ); - const AVClass *fclass = avformat_get_class(); - const AVOption *foption = av_opt_find(&fclass, e->key, NULL, 0, - AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ); - if (!option || foption) + if (!option) continue; - - if (!(option->flags & AV_OPT_FLAG_DECODING_PARAM)) { av_log(NULL, AV_LOG_ERROR, "Codec AVOption %s (%s) specified for " "input file #%d (%s) is not a decoding option.\n", e->key, @@ -1025,7 +866,7 @@ static void choose_encoder(OptionsContext *o, AVFormatContext *s, OutputStream * } } -static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, enum AVMediaType type, int source_index) +static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, enum AVMediaType type) { OutputStream *ost; AVStream *st = avformat_new_stream(oc, NULL); @@ -1033,7 +874,6 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e char *bsf = NULL, *next, *codec_tag = NULL; AVBitStreamFilterContext *bsfc, *bsfc_prev = NULL; double qscale = -1; - int i; if (!st) { av_log(NULL, AV_LOG_FATAL, "Could not alloc stream.\n"); @@ -1092,16 +932,6 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e ost->max_frames = INT64_MAX; MATCH_PER_STREAM_OPT(max_frames, i64, ost->max_frames, oc, st); - for (i = 0; inb_max_frames; i++) { - char *p = o->max_frames[i].specifier; - if (!*p && type != AVMEDIA_TYPE_VIDEO) { - av_log(NULL, AV_LOG_WARNING, "Applying unspecific -frames to non video streams, maybe you meant -vframes ?\n"); - break; - } - } - - ost->copy_prior_start = -1; - MATCH_PER_STREAM_OPT(copy_prior_start, i, ost->copy_prior_start, oc ,st); MATCH_PER_STREAM_OPT(bitstream_filters, str, bsf, oc, st); while (bsf) { @@ -1139,18 +969,9 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e av_opt_get_int(o->g->sws_opts, "sws_flags", 0, &ost->sws_flags); - av_dict_copy(&ost->swr_opts, o->g->swr_opts, 0); - if (ost->enc && av_get_exact_bits_per_sample(ost->enc->id) == 24) - av_dict_set(&ost->swr_opts, "output_sample_bits", "24", 0); - av_dict_copy(&ost->resample_opts, o->g->resample_opts, 0); - ost->source_index = source_index; - if (source_index >= 0) { - ost->sync_ist = input_streams[source_index]; - input_streams[source_index]->discard = 0; - input_streams[source_index]->st->discard = AVDISCARD_NONE; - } + ost->pix_fmts[0] = ost->pix_fmts[1] = AV_PIX_FMT_NONE; ost->last_mux_dts = AV_NOPTS_VALUE; return ost; @@ -1206,97 +1027,68 @@ static char *get_ost_filters(OptionsContext *o, AVFormatContext *oc, OutputStream *ost) { AVStream *st = ost->st; + char *filter = NULL, *filter_script = NULL; - if (ost->filters_script && ost->filters) { + MATCH_PER_STREAM_OPT(filter_scripts, str, filter_script, oc, st); + MATCH_PER_STREAM_OPT(filters, str, filter, oc, st); + + if (filter_script && filter) { av_log(NULL, AV_LOG_ERROR, "Both -filter and -filter_script set for " "output stream #%d:%d.\n", nb_output_files, st->index); exit_program(1); } - if (ost->filters_script) - return read_file(ost->filters_script); - else if (ost->filters) - return av_strdup(ost->filters); + if (filter_script) + return read_file(filter_script); + else if (filter) + return av_strdup(filter); return av_strdup(st->codec->codec_type == AVMEDIA_TYPE_VIDEO ? "null" : "anull"); } -static void check_streamcopy_filters(OptionsContext *o, AVFormatContext *oc, - const OutputStream *ost, enum AVMediaType type) -{ - if (ost->filters_script || ost->filters) { - av_log(NULL, AV_LOG_ERROR, - "%s '%s' was defined for %s output stream %d:%d but codec copy was selected.\n" - "Filtering and streamcopy cannot be used together.\n", - ost->filters ? "Filtergraph" : "Filtergraph script", - ost->filters ? ost->filters : ost->filters_script, - av_get_media_type_string(type), ost->file_index, ost->index); - exit_program(1); - } -} - -static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, int source_index) +static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc) { AVStream *st; OutputStream *ost; AVCodecContext *video_enc; - char *frame_rate = NULL, *frame_aspect_ratio = NULL; + char *frame_aspect_ratio = NULL; - ost = new_output_stream(o, oc, AVMEDIA_TYPE_VIDEO, source_index); + ost = new_output_stream(o, oc, AVMEDIA_TYPE_VIDEO); st = ost->st; video_enc = st->codec; - MATCH_PER_STREAM_OPT(frame_rates, str, frame_rate, oc, st); - if (frame_rate && av_parse_video_rate(&ost->frame_rate, frame_rate) < 0) { - av_log(NULL, AV_LOG_FATAL, "Invalid framerate value: %s\n", frame_rate); - exit_program(1); - } - MATCH_PER_STREAM_OPT(frame_aspect_ratios, str, frame_aspect_ratio, oc, st); - if (frame_aspect_ratio) { - AVRational q; - if (av_parse_ratio(&q, frame_aspect_ratio, 255, 0, NULL) < 0 || - q.num <= 0 || q.den <= 0) { - av_log(NULL, AV_LOG_FATAL, "Invalid aspect ratio: %s\n", frame_aspect_ratio); - exit_program(1); - } - ost->frame_aspect_ratio = q; - } - - MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st); - MATCH_PER_STREAM_OPT(filters, str, ost->filters, oc, st); + if (frame_aspect_ratio) + ost->frame_aspect_ratio = parse_frame_aspect_ratio(frame_aspect_ratio); if (!ost->stream_copy) { const char *p = NULL; - char *frame_size = NULL; + char *frame_rate = NULL, *frame_size = NULL; char *frame_pix_fmt = NULL; char *intra_matrix = NULL, *inter_matrix = NULL; - char *chroma_intra_matrix = NULL; int do_pass = 0; int i; + MATCH_PER_STREAM_OPT(frame_rates, str, frame_rate, oc, st); + if (frame_rate && av_parse_video_rate(&ost->frame_rate, frame_rate) < 0) { + av_log(NULL, AV_LOG_FATAL, "Invalid framerate value: %s\n", frame_rate); + exit_program(1); + } + MATCH_PER_STREAM_OPT(frame_sizes, str, frame_size, oc, st); if (frame_size && av_parse_video_size(&video_enc->width, &video_enc->height, frame_size) < 0) { av_log(NULL, AV_LOG_FATAL, "Invalid frame size: %s.\n", frame_size); exit_program(1); } - video_enc->bits_per_raw_sample = frame_bits_per_raw_sample; MATCH_PER_STREAM_OPT(frame_pix_fmts, str, frame_pix_fmt, oc, st); - if (frame_pix_fmt && *frame_pix_fmt == '+') { - ost->keep_pix_fmt = 1; - if (!*++frame_pix_fmt) - frame_pix_fmt = NULL; - } if (frame_pix_fmt && (video_enc->pix_fmt = av_get_pix_fmt(frame_pix_fmt)) == AV_PIX_FMT_NONE) { av_log(NULL, AV_LOG_FATAL, "Unknown pixel format requested: %s.\n", frame_pix_fmt); exit_program(1); } st->sample_aspect_ratio = video_enc->sample_aspect_ratio; - if (intra_only) - video_enc->gop_size = 0; MATCH_PER_STREAM_OPT(intra_matrices, str, intra_matrix, oc, st); if (intra_matrix) { if (!(video_enc->intra_matrix = av_mallocz(sizeof(*video_enc->intra_matrix) * 64))) { @@ -1305,16 +1097,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in } parse_matrix_coeffs(video_enc->intra_matrix, intra_matrix); } - MATCH_PER_STREAM_OPT(chroma_intra_matrices, str, chroma_intra_matrix, oc, st); - if (chroma_intra_matrix) { - uint16_t *p = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64); - if (!p) { - av_log(NULL, AV_LOG_FATAL, "Could not allocate memory for intra matrix.\n"); - exit_program(1); - } - av_codec_set_chroma_intra_matrix(video_enc, p); - parse_matrix_coeffs(p, chroma_intra_matrix); - } MATCH_PER_STREAM_OPT(inter_matrices, str, inter_matrix, oc, st); if (inter_matrix) { if (!(video_enc->inter_matrix = av_mallocz(sizeof(*video_enc->inter_matrix) * 64))) { @@ -1332,7 +1114,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in av_log(NULL, AV_LOG_FATAL, "error parsing rc_override\n"); exit_program(1); } - /* FIXME realloc failure */ video_enc->rc_override = av_realloc(video_enc->rc_override, sizeof(RcOverride) * (i + 1)); @@ -1352,19 +1133,13 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in video_enc->rc_override_count = i; video_enc->intra_dc_precision = intra_dc_precision - 8; - if (do_psnr) - video_enc->flags|= CODEC_FLAG_PSNR; - /* two pass mode */ MATCH_PER_STREAM_OPT(pass, i, do_pass, oc, st); if (do_pass) { - if (do_pass & 1) { + if (do_pass == 1) { video_enc->flags |= CODEC_FLAG_PASS1; - av_dict_set(&ost->encoder_opts, "flags", "+pass1", AV_DICT_APPEND); - } - if (do_pass & 2) { + } else { video_enc->flags |= CODEC_FLAG_PASS2; - av_dict_set(&ost->encoder_opts, "flags", "+pass2", AV_DICT_APPEND); } } @@ -1390,28 +1165,21 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in MATCH_PER_STREAM_OPT(copy_initial_nonkeyframes, i, ost->copy_initial_nonkeyframes, oc ,st); } - if (ost->stream_copy) - check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_VIDEO); - return ost; } -static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, int source_index) +static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc) { - int n; AVStream *st; OutputStream *ost; AVCodecContext *audio_enc; - ost = new_output_stream(o, oc, AVMEDIA_TYPE_AUDIO, source_index); + ost = new_output_stream(o, oc, AVMEDIA_TYPE_AUDIO); st = ost->st; audio_enc = st->codec; audio_enc->codec_type = AVMEDIA_TYPE_AUDIO; - MATCH_PER_STREAM_OPT(filter_scripts, str, ost->filters_script, oc, st); - MATCH_PER_STREAM_OPT(filters, str, ost->filters, oc, st); - if (!ost->stream_copy) { char *sample_fmt = NULL; @@ -1426,52 +1194,19 @@ static OutputStream *new_audio_stream(OptionsContext *o, AVFormatContext *oc, in MATCH_PER_STREAM_OPT(audio_sample_rate, i, audio_enc->sample_rate, oc, st); - MATCH_PER_STREAM_OPT(apad, str, ost->apad, oc, st); - ost->apad = av_strdup(ost->apad); - ost->avfilter = get_ost_filters(o, oc, ost); if (!ost->avfilter) exit_program(1); - - /* check for channel mapping for this audio stream */ - for (n = 0; n < o->nb_audio_channel_maps; n++) { - AudioChannelMap *map = &o->audio_channel_maps[n]; - if ((map->ofile_idx == -1 || ost->file_index == map->ofile_idx) && - (map->ostream_idx == -1 || ost->st->index == map->ostream_idx)) { - InputStream *ist; - - if (map->channel_idx == -1) { - ist = NULL; - } else if (ost->source_index < 0) { - av_log(NULL, AV_LOG_FATAL, "Cannot determine input stream for channel mapping %d.%d\n", - ost->file_index, ost->st->index); - continue; - } else { - ist = input_streams[ost->source_index]; - } - - if (!ist || (ist->file_index == map->file_idx && ist->st->index == map->stream_idx)) { - if (ost->audio_channels_mapped < FF_ARRAY_ELEMS(ost->audio_channels_map)) - ost->audio_channels_map[ost->audio_channels_mapped++] = map->channel_idx; - else - av_log(NULL, AV_LOG_FATAL, "Max channel mapping for output %d.%d reached\n", - ost->file_index, ost->st->index); - } - } - } } - if (ost->stream_copy) - check_streamcopy_filters(o, oc, ost, AVMEDIA_TYPE_AUDIO); - return ost; } -static OutputStream *new_data_stream(OptionsContext *o, AVFormatContext *oc, int source_index) +static OutputStream *new_data_stream(OptionsContext *o, AVFormatContext *oc) { OutputStream *ost; - ost = new_output_stream(o, oc, AVMEDIA_TYPE_DATA, source_index); + ost = new_output_stream(o, oc, AVMEDIA_TYPE_DATA); if (!ost->stream_copy) { av_log(NULL, AV_LOG_FATAL, "Data stream encoding not supported yet (only streamcopy)\n"); exit_program(1); @@ -1480,38 +1215,26 @@ static OutputStream *new_data_stream(OptionsContext *o, AVFormatContext *oc, int return ost; } -static OutputStream *new_attachment_stream(OptionsContext *o, AVFormatContext *oc, int source_index) +static OutputStream *new_attachment_stream(OptionsContext *o, AVFormatContext *oc) { - OutputStream *ost = new_output_stream(o, oc, AVMEDIA_TYPE_ATTACHMENT, source_index); + OutputStream *ost = new_output_stream(o, oc, AVMEDIA_TYPE_ATTACHMENT); ost->stream_copy = 1; ost->finished = 1; return ost; } -static OutputStream *new_subtitle_stream(OptionsContext *o, AVFormatContext *oc, int source_index) +static OutputStream *new_subtitle_stream(OptionsContext *o, AVFormatContext *oc) { AVStream *st; OutputStream *ost; AVCodecContext *subtitle_enc; - ost = new_output_stream(o, oc, AVMEDIA_TYPE_SUBTITLE, source_index); + ost = new_output_stream(o, oc, AVMEDIA_TYPE_SUBTITLE); st = ost->st; subtitle_enc = st->codec; subtitle_enc->codec_type = AVMEDIA_TYPE_SUBTITLE; - MATCH_PER_STREAM_OPT(copy_initial_nonkeyframes, i, ost->copy_initial_nonkeyframes, oc, st); - - if (!ost->stream_copy) { - char *frame_size = NULL; - - MATCH_PER_STREAM_OPT(frame_sizes, str, frame_size, oc, st); - if (frame_size && av_parse_video_size(&subtitle_enc->width, &subtitle_enc->height, frame_size) < 0) { - av_log(NULL, AV_LOG_FATAL, "Invalid frame size: %s.\n", frame_size); - exit_program(1); - } - } - return ost; } @@ -1532,7 +1255,7 @@ static int opt_streamid(void *optctx, const char *opt, const char *arg) exit_program(1); } *p++ = '\0'; - idx = parse_number_or_die(opt, idx_str, OPT_INT, 0, MAX_STREAMS-1); + idx = parse_number_or_die(opt, idx_str, OPT_INT, 0, INT_MAX); o->streamid_map = grow_array(o->streamid_map, sizeof(*o->streamid_map), &o->nb_streamid_map, idx+1); o->streamid_map[idx] = parse_number_or_die(opt, p, OPT_INT, 0, INT_MAX); return 0; @@ -1545,7 +1268,7 @@ static int copy_chapters(InputFile *ifile, OutputFile *ofile, int copy_metadata) AVChapter **tmp; int i; - tmp = av_realloc_f(os->chapters, is->nb_chapters + os->nb_chapters, sizeof(*os->chapters)); + tmp = av_realloc(os->chapters, sizeof(*os->chapters) * (is->nb_chapters + os->nb_chapters)); if (!tmp) return AVERROR(ENOMEM); os->chapters = tmp; @@ -1581,46 +1304,6 @@ static int copy_chapters(InputFile *ifile, OutputFile *ofile, int copy_metadata) return 0; } -static int read_ffserver_streams(OptionsContext *o, AVFormatContext *s, const char *filename) -{ - int i, err; - AVFormatContext *ic = avformat_alloc_context(); - - ic->interrupt_callback = int_cb; - err = avformat_open_input(&ic, filename, NULL, NULL); - if (err < 0) - return err; - /* copy stream format */ - for(i=0;inb_streams;i++) { - AVStream *st; - OutputStream *ost; - AVCodec *codec; - AVCodecContext *avctx; - - codec = avcodec_find_encoder(ic->streams[i]->codec->codec_id); - ost = new_output_stream(o, s, codec->type, -1); - st = ost->st; - avctx = st->codec; - ost->enc = codec; - - // FIXME: a more elegant solution is needed - memcpy(st, ic->streams[i], sizeof(AVStream)); - st->cur_dts = 0; - st->info = av_malloc(sizeof(*st->info)); - memcpy(st->info, ic->streams[i]->info, sizeof(*st->info)); - st->codec= avctx; - avcodec_copy_context(st->codec, ic->streams[i]->codec); - - if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && !ost->stream_copy) - choose_sample_fmt(st, codec); - else if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO && !ost->stream_copy) - choose_pixel_fmt(st, codec, st->codec->pix_fmt); - } - - avformat_close_input(&ic); - return err; -} - static void init_output_filter(OutputFilter *ofilter, OptionsContext *o, AVFormatContext *oc) { @@ -1628,8 +1311,8 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o, switch (avfilter_pad_get_type(ofilter->out_tmp->filter_ctx->output_pads, ofilter->out_tmp->pad_idx)) { - case AVMEDIA_TYPE_VIDEO: ost = new_video_stream(o, oc, -1); break; - case AVMEDIA_TYPE_AUDIO: ost = new_audio_stream(o, oc, -1); break; + case AVMEDIA_TYPE_VIDEO: ost = new_video_stream(o, oc); break; + case AVMEDIA_TYPE_AUDIO: ost = new_audio_stream(o, oc); break; default: av_log(NULL, AV_LOG_FATAL, "Only video and audio filters are supported " "currently.\n"); @@ -1648,18 +1331,6 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o, exit_program(1); } - if (ost->avfilter && (ost->filters || ost->filters_script)) { - const char *opt = ost->filters ? "-vf/-af/-filter" : "-filter_script"; - av_log(NULL, AV_LOG_ERROR, - "%s '%s' was specified through the %s option " - "for output stream %d:%d, which is fed from a complex filtergraph.\n" - "%s and -filter_complex cannot be used together for the same stream.\n", - ost->filters ? "Filtergraph" : "Filtergraph script", - ost->filters ? ost->filters : ost->filters_script, - opt, ost->file_index, ost->index, opt); - exit_program(1); - } - if (configure_output_filter(ofilter->graph, ofilter, ofilter->out_tmp) < 0) { av_log(NULL, AV_LOG_FATAL, "Error configuring filter.\n"); exit_program(1); @@ -1694,21 +1365,6 @@ static int open_output_file(OptionsContext *o, const char *filename) exit_program(1); } - if (o->stop_time != INT64_MAX && o->recording_time != INT64_MAX) { - o->stop_time = INT64_MAX; - av_log(NULL, AV_LOG_WARNING, "-t and -to cannot be used together; using -t.\n"); - } - - if (o->stop_time != INT64_MAX && o->recording_time == INT64_MAX) { - int64_t start_time = o->start_time == AV_NOPTS_VALUE ? 0 : o->start_time; - if (o->stop_time <= start_time) { - av_log(NULL, AV_LOG_WARNING, "-to value smaller than -ss; ignoring -to.\n"); - o->stop_time = INT64_MAX; - } else { - o->recording_time = o->stop_time - start_time; - } - } - GROW_ARRAY(output_files, nb_output_files); of = av_mallocz(sizeof(*of)); if (!of) @@ -1725,18 +1381,33 @@ static int open_output_file(OptionsContext *o, const char *filename) if (!strcmp(filename, "-")) filename = "pipe:"; - err = avformat_alloc_output_context2(&oc, NULL, o->format, filename); + oc = avformat_alloc_context(); if (!oc) { - print_error(filename, err); + print_error(filename, AVERROR(ENOMEM)); exit_program(1); } - of->ctx = oc; if (o->recording_time != INT64_MAX) oc->duration = o->recording_time; - file_oformat= oc->oformat; + if (o->format) { + file_oformat = av_guess_format(o->format, NULL, NULL); + if (!file_oformat) { + av_log(NULL, AV_LOG_FATAL, "Requested output format '%s' is not a suitable output format\n", o->format); + exit_program(1); + } + } else { + file_oformat = av_guess_format(NULL, filename, NULL); + if (!file_oformat) { + av_log(NULL, AV_LOG_FATAL, "Unable to find a suitable output format for '%s'\n", + filename); + exit_program(1); + } + } + + oc->oformat = file_oformat; oc->interrupt_callback = int_cb; + av_strlcpy(oc->filename, filename, sizeof(oc->filename)); /* create streams for all unlabeled output pads */ for (i = 0; i < nb_filtergraphs; i++) { @@ -1757,69 +1428,29 @@ static int open_output_file(OptionsContext *o, const char *filename) } } - /* ffserver seeking with date=... needs a date reference */ - if (!strcmp(file_oformat->name, "ffm") && - av_strstart(filename, "http:", NULL)) { - int err = parse_option(o, "metadata", "creation_time=now", options); - if (err < 0) { - print_error(filename, err); - exit_program(1); - } - } - - if (!strcmp(file_oformat->name, "ffm") && !override_ffserver && - av_strstart(filename, "http:", NULL)) { - int j; - /* special case for files sent to ffserver: we get the stream - parameters from ffserver */ - int err = read_ffserver_streams(o, oc, filename); - if (err < 0) { - print_error(filename, err); - exit_program(1); - } - for(j = nb_output_streams - oc->nb_streams; j < nb_output_streams; j++) { - ost = output_streams[j]; - for (i = 0; i < nb_input_streams; i++) { - ist = input_streams[i]; - if(ist->st->codec->codec_type == ost->st->codec->codec_type){ - ost->sync_ist= ist; - ost->source_index= i; - if(ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) ost->avfilter = av_strdup("anull"); - if(ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) ost->avfilter = av_strdup("null"); - ist->discard = 0; - ist->st->discard = AVDISCARD_NONE; - break; - } - } - if(!ost->sync_ist){ - av_log(NULL, AV_LOG_FATAL, "Missing %s stream which is required by this ffm\n", av_get_media_type_string(ost->st->codec->codec_type)); - exit_program(1); - } - } - } else if (!o->nb_stream_maps) { - char *subtitle_codec_name = NULL; + if (!o->nb_stream_maps) { /* pick the "best" stream of each type */ +#define NEW_STREAM(type, index)\ + if (index >= 0) {\ + ost = new_ ## type ## _stream(o, oc);\ + ost->source_index = index;\ + ost->sync_ist = input_streams[index];\ + input_streams[index]->discard = 0;\ + input_streams[index]->st->discard = AVDISCARD_NONE;\ + } /* video: highest resolution */ if (!o->video_disable && oc->oformat->video_codec != AV_CODEC_ID_NONE) { int area = 0, idx = -1; - int qcr = avformat_query_codec(oc->oformat, oc->oformat->video_codec, 0); for (i = 0; i < nb_input_streams; i++) { - int new_area; ist = input_streams[i]; - new_area = ist->st->codec->width * ist->st->codec->height; - if((qcr!=MKTAG('A', 'P', 'I', 'C')) && (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC)) - new_area = 1; if (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && - new_area > area) { - if((qcr==MKTAG('A', 'P', 'I', 'C')) && !(ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC)) - continue; - area = new_area; + ist->st->codec->width * ist->st->codec->height > area) { + area = ist->st->codec->width * ist->st->codec->height; idx = i; } } - if (idx >= 0) - new_video_stream(o, oc, idx); + NEW_STREAM(video, idx); } /* audio: most channels */ @@ -1833,16 +1464,14 @@ static int open_output_file(OptionsContext *o, const char *filename) idx = i; } } - if (idx >= 0) - new_audio_stream(o, oc, idx); + NEW_STREAM(audio, idx); } /* subtitles: pick first */ - MATCH_PER_TYPE_OPT(codec_names, str, subtitle_codec_name, oc, "s"); - if (!o->subtitle_disable && (avcodec_find_encoder(oc->oformat->subtitle_codec) || subtitle_codec_name)) { + if (!o->subtitle_disable && oc->oformat->subtitle_codec != AV_CODEC_ID_NONE) { for (i = 0; i < nb_input_streams; i++) if (input_streams[i]->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) { - new_subtitle_stream(o, oc, i); + NEW_STREAM(subtitle, i); break; } } @@ -1872,34 +1501,29 @@ static int open_output_file(OptionsContext *o, const char *filename) loop_end: if (!ofilter) { av_log(NULL, AV_LOG_FATAL, "Output with label '%s' does not exist " - "in any defined filter graph, or was already used elsewhere.\n", map->linklabel); + "in any defined filter graph.\n", map->linklabel); exit_program(1); } init_output_filter(ofilter, o, oc); } else { - int src_idx = input_files[map->file_index]->ist_index + map->stream_index; - ist = input_streams[input_files[map->file_index]->ist_index + map->stream_index]; - if(o->subtitle_disable && ist->st->codec->codec_type == AVMEDIA_TYPE_SUBTITLE) - continue; - if(o-> audio_disable && ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO) - continue; - if(o-> video_disable && ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) - continue; - if(o-> data_disable && ist->st->codec->codec_type == AVMEDIA_TYPE_DATA) - continue; - switch (ist->st->codec->codec_type) { - case AVMEDIA_TYPE_VIDEO: ost = new_video_stream (o, oc, src_idx); break; - case AVMEDIA_TYPE_AUDIO: ost = new_audio_stream (o, oc, src_idx); break; - case AVMEDIA_TYPE_SUBTITLE: ost = new_subtitle_stream (o, oc, src_idx); break; - case AVMEDIA_TYPE_DATA: ost = new_data_stream (o, oc, src_idx); break; - case AVMEDIA_TYPE_ATTACHMENT: ost = new_attachment_stream(o, oc, src_idx); break; + case AVMEDIA_TYPE_VIDEO: ost = new_video_stream(o, oc); break; + case AVMEDIA_TYPE_AUDIO: ost = new_audio_stream(o, oc); break; + case AVMEDIA_TYPE_SUBTITLE: ost = new_subtitle_stream(o, oc); break; + case AVMEDIA_TYPE_DATA: ost = new_data_stream(o, oc); break; + case AVMEDIA_TYPE_ATTACHMENT: ost = new_attachment_stream(o, oc); break; default: av_log(NULL, AV_LOG_FATAL, "Cannot map stream #%d:%d - unsupported type.\n", map->file_index, map->stream_index); exit_program(1); } + + ost->source_index = input_files[map->file_index]->ist_index + map->stream_index; + ost->sync_ist = input_streams[input_files[map->sync_file_index]->ist_index + + map->sync_stream_index]; + ist->discard = 0; + ist->st->discard = AVDISCARD_NONE; } } } @@ -1928,10 +1552,10 @@ loop_end: } avio_read(pb, attachment, len); - ost = new_attachment_stream(o, oc, -1); + ost = new_attachment_stream(o, oc); ost->stream_copy = 0; + ost->source_index = -1; ost->attachment_filename = o->attachments[i]; - ost->finished = 1; ost->st->codec->extradata = attachment; ost->st->codec->extradata_size = len; @@ -1940,17 +1564,6 @@ loop_end: avio_close(pb); } - for (i = nb_output_streams - oc->nb_streams; i < nb_output_streams; i++) { //for all streams of this output file - AVDictionaryEntry *e; - ost = output_streams[i]; - - if ((ost->stream_copy || ost->attachment_filename) - && (e = av_dict_get(o->g->codec_opts, "flags", NULL, AV_DICT_IGNORE_SUFFIX)) - && (!e->key[5] || check_stream_specifier(oc, ost->st, e->key+6))) - if (av_opt_set(ost->st->codec, "flags", e->value, 0) < 0) - exit_program(1); - } - /* check if all codec options have been used */ unused_opts = strip_specifiers(o->g->codec_opts); for (i = of->ost_index; i < nb_output_streams; i++) { @@ -1975,10 +1588,6 @@ loop_end: exit_program(1); } - // gop_timecode is injected by generic code but not always used - if (!strcmp(e->key, "gop_timecode")) - continue; - av_log(NULL, AV_LOG_WARNING, "Codec AVOption %s (%s) specified for " "output file #%d (%s) has not been used for any stream. The most " "likely reason is either wrong type (e.g. a video option with " @@ -2007,8 +1616,7 @@ loop_end: print_error(filename, err); exit_program(1); } - } else if (strcmp(oc->oformat->name, "image2")==0 && !av_filename_number_test(filename)) - assert_file_overwrite(filename); + } if (o->mux_preload) { uint8_t buf[64]; @@ -2016,6 +1624,7 @@ loop_end: av_dict_set(&of->opts, "preload", buf, 0); } oc->max_delay = (int)(o->mux_max_delay * AV_TIME_BASE); + oc->flags |= AVFMT_FLAG_NONBLOCK; /* copy metadata */ for (i = 0; i < o->nb_metadata_map; i++) { @@ -2052,13 +1661,9 @@ loop_end: !o->metadata_chapters_manual); /* copy global metadata by default */ - if (!o->metadata_global_manual && nb_input_files){ + if (!o->metadata_global_manual && nb_input_files) av_dict_copy(&oc->metadata, input_files[0]->ctx->metadata, AV_DICT_DONT_OVERWRITE); - if(o->recording_time != INT64_MAX) - av_dict_set(&oc->metadata, "duration", NULL, 0); - av_dict_set(&oc->metadata, "creation_time", NULL, 0); - } if (!o->metadata_streams_manual) for (i = of->ost_index; i < nb_output_streams; i++) { InputStream *ist; @@ -2066,8 +1671,6 @@ loop_end: continue; ist = input_streams[output_streams[i]->source_index]; av_dict_copy(&output_streams[i]->st->metadata, ist->st->metadata, AV_DICT_DONT_OVERWRITE); - if (!output_streams[i]->stream_copy) - av_dict_set(&output_streams[i]->st->metadata, "encoder", NULL, 0); } /* process manually set metadata */ @@ -2075,7 +1678,7 @@ loop_end: AVDictionary **m; char type, *val; const char *stream_spec; - int index = 0, j, ret = 0; + int index = 0, j, ret; val = strchr(o->metadata[i].u.str, '='); if (!val) { @@ -2172,19 +1775,19 @@ static int opt_target(void *optctx, const char *opt, const char *arg) parse_option(o, "s", norm == PAL ? "352x288" : "352x240", options); parse_option(o, "r", frame_rates[norm], options); - av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE); + opt_default(NULL, "g", norm == PAL ? "15" : "18"); - av_dict_set(&o->g->codec_opts, "b:v", "1150000", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->codec_opts, "maxrate", "1150000", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->codec_opts, "minrate", "1150000", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->codec_opts, "bufsize", "327680", AV_DICT_DONT_OVERWRITE); // 40*1024*8; + opt_default(NULL, "b", "1150000"); + opt_default(NULL, "maxrate", "1150000"); + opt_default(NULL, "minrate", "1150000"); + opt_default(NULL, "bufsize", "327680"); // 40*1024*8; - av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE); + opt_default(NULL, "b:a", "224000"); parse_option(o, "ar", "44100", options); parse_option(o, "ac", "2", options); - av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->format_opts, "muxrate", "1411200", AV_DICT_DONT_OVERWRITE); // 2352 * 75 * 8; + opt_default(NULL, "packetsize", "2324"); + opt_default(NULL, "muxrate", "1411200"); // 2352 * 75 * 8; /* We have to offset the PTS, so that it is consistent with the SCR. SCR starts at 36000, but the first two packs contain only padding @@ -2200,19 +1803,19 @@ static int opt_target(void *optctx, const char *opt, const char *arg) parse_option(o, "s", norm == PAL ? "480x576" : "480x480", options); parse_option(o, "r", frame_rates[norm], options); - parse_option(o, "pix_fmt", "yuv420p", options); - av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE); + opt_default(NULL, "g", norm == PAL ? "15" : "18"); - av_dict_set(&o->g->codec_opts, "b:v", "2040000", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->codec_opts, "maxrate", "2516000", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1145000; - av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8; - av_dict_set(&o->g->codec_opts, "scan_offset", "1", AV_DICT_DONT_OVERWRITE); + opt_default(NULL, "b", "2040000"); + opt_default(NULL, "maxrate", "2516000"); + opt_default(NULL, "minrate", "0"); // 1145000; + opt_default(NULL, "bufsize", "1835008"); // 224*1024*8; + opt_default(NULL, "flags", "+scan_offset"); - av_dict_set(&o->g->codec_opts, "b:a", "224000", AV_DICT_DONT_OVERWRITE); + + opt_default(NULL, "b:a", "224000"); parse_option(o, "ar", "44100", options); - av_dict_set(&o->g->format_opts, "packetsize", "2324", AV_DICT_DONT_OVERWRITE); + opt_default(NULL, "packetsize", "2324"); } else if (!strcmp(arg, "dvd")) { @@ -2222,18 +1825,17 @@ static int opt_target(void *optctx, const char *opt, const char *arg) parse_option(o, "s", norm == PAL ? "720x576" : "720x480", options); parse_option(o, "r", frame_rates[norm], options); - parse_option(o, "pix_fmt", "yuv420p", options); - av_dict_set(&o->g->codec_opts, "g", norm == PAL ? "15" : "18", AV_DICT_DONT_OVERWRITE); + opt_default(NULL, "g", norm == PAL ? "15" : "18"); - av_dict_set(&o->g->codec_opts, "b:v", "6000000", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->codec_opts, "maxrate", "9000000", AV_DICT_DONT_OVERWRITE); - av_dict_set(&o->g->codec_opts, "minrate", "0", AV_DICT_DONT_OVERWRITE); // 1500000; - av_dict_set(&o->g->codec_opts, "bufsize", "1835008", AV_DICT_DONT_OVERWRITE); // 224*1024*8; + opt_default(NULL, "b", "6000000"); + opt_default(NULL, "maxrate", "9000000"); + opt_default(NULL, "minrate", "0"); // 1500000; + opt_default(NULL, "bufsize", "1835008"); // 224*1024*8; - av_dict_set(&o->g->format_opts, "packetsize", "2048", AV_DICT_DONT_OVERWRITE); // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack. - av_dict_set(&o->g->format_opts, "muxrate", "10080000", AV_DICT_DONT_OVERWRITE); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8 + opt_default(NULL, "packetsize", "2048"); // from www.mpucoder.com: DVD sectors contain 2048 bytes of data, this is also the size of one pack. + opt_default(NULL, "muxrate", "10080000"); // from mplex project: data_rate = 1260000. mux_rate = data_rate * 8 - av_dict_set(&o->g->codec_opts, "b:a", "448000", AV_DICT_DONT_OVERWRITE); + opt_default(NULL, "b:a", "448000"); parse_option(o, "ar", "48000", options); } else if (!strncmp(arg, "dv", 2)) { @@ -2291,120 +1893,22 @@ static int opt_data_frames(void *optctx, const char *opt, const char *arg) return parse_option(o, "frames:d", arg, options); } -static int opt_default_new(OptionsContext *o, const char *opt, const char *arg) -{ - int ret; - AVDictionary *cbak = codec_opts; - AVDictionary *fbak = format_opts; - codec_opts = NULL; - format_opts = NULL; - - ret = opt_default(NULL, opt, arg); - - av_dict_copy(&o->g->codec_opts , codec_opts, 0); - av_dict_copy(&o->g->format_opts, format_opts, 0); - av_dict_free(&codec_opts); - av_dict_free(&format_opts); - codec_opts = cbak; - format_opts = fbak; - - return ret; -} - -static int opt_preset(void *optctx, const char *opt, const char *arg) +static int opt_video_tag(void *optctx, const char *opt, const char *arg) { OptionsContext *o = optctx; - FILE *f=NULL; - char filename[1000], line[1000], tmp_line[1000]; - const char *codec_name = NULL; - - tmp_line[0] = *opt; - tmp_line[1] = 0; - MATCH_PER_TYPE_OPT(codec_names, str, codec_name, NULL, tmp_line); - - if (!(f = get_preset_file(filename, sizeof(filename), arg, *opt == 'f', codec_name))) { - if(!strncmp(arg, "libx264-lossless", strlen("libx264-lossless"))){ - av_log(NULL, AV_LOG_FATAL, "Please use -preset -qp 0\n"); - }else - av_log(NULL, AV_LOG_FATAL, "File for preset '%s' not found\n", arg); - exit_program(1); - } - - while (fgets(line, sizeof(line), f)) { - char *key = tmp_line, *value, *endptr; - - if (strcspn(line, "#\n\r") == 0) - continue; - av_strlcpy(tmp_line, line, sizeof(tmp_line)); - if (!av_strtok(key, "=", &value) || - !av_strtok(value, "\r\n", &endptr)) { - av_log(NULL, AV_LOG_FATAL, "%s: Invalid syntax: '%s'\n", filename, line); - exit_program(1); - } - av_log(NULL, AV_LOG_DEBUG, "ffpreset[%s]: set '%s' = '%s'\n", filename, key, value); - - if (!strcmp(key, "acodec")) opt_audio_codec (o, key, value); - else if (!strcmp(key, "vcodec")) opt_video_codec (o, key, value); - else if (!strcmp(key, "scodec")) opt_subtitle_codec(o, key, value); - else if (!strcmp(key, "dcodec")) opt_data_codec (o, key, value); - else if (opt_default_new(o, key, value) < 0) { - av_log(NULL, AV_LOG_FATAL, "%s: Invalid option or argument: '%s', parsed as '%s' = '%s'\n", - filename, line, key, value); - exit_program(1); - } - } - - fclose(f); - - return 0; + return parse_option(o, "tag:v", arg, options); } -static int opt_old2new(void *optctx, const char *opt, const char *arg) +static int opt_audio_tag(void *optctx, const char *opt, const char *arg) { OptionsContext *o = optctx; - char *s = av_asprintf("%s:%c", opt + 1, *opt); - int ret = parse_option(o, s, arg, options); - av_free(s); - return ret; + return parse_option(o, "tag:a", arg, options); } -static int opt_bitrate(void *optctx, const char *opt, const char *arg) +static int opt_subtitle_tag(void *optctx, const char *opt, const char *arg) { OptionsContext *o = optctx; - if(!strcmp(opt, "b")){ - av_log(NULL, AV_LOG_WARNING, "Please use -b:a or -b:v, -b is ambiguous\n"); - av_dict_set(&o->g->codec_opts, "b:v", arg, 0); - return 0; - } - av_dict_set(&o->g->codec_opts, opt, arg, 0); - return 0; -} - -static int opt_qscale(void *optctx, const char *opt, const char *arg) -{ - OptionsContext *o = optctx; - char *s; - int ret; - if(!strcmp(opt, "qscale")){ - av_log(NULL, AV_LOG_WARNING, "Please use -q:a or -q:v, -qscale is ambiguous\n"); - return parse_option(o, "q:v", arg, options); - } - s = av_asprintf("q%s", opt + 6); - ret = parse_option(o, s, arg, options); - av_free(s); - return ret; -} - -static int opt_profile(void *optctx, const char *opt, const char *arg) -{ - OptionsContext *o = optctx; - if(!strcmp(opt, "profile")){ - av_log(NULL, AV_LOG_WARNING, "Please use -profile:a or -profile:v, -profile is ambiguous\n"); - av_dict_set(&o->g->codec_opts, "profile:v", arg, 0); - return 0; - } - av_dict_set(&o->g->codec_opts, opt, arg, 0); - return 0; + return parse_option(o, "tag:s", arg, options); } static int opt_video_filters(void *optctx, const char *opt, const char *arg) @@ -2424,24 +1928,12 @@ static int opt_vsync(void *optctx, const char *opt, const char *arg) if (!av_strcasecmp(arg, "cfr")) video_sync_method = VSYNC_CFR; else if (!av_strcasecmp(arg, "vfr")) video_sync_method = VSYNC_VFR; else if (!av_strcasecmp(arg, "passthrough")) video_sync_method = VSYNC_PASSTHROUGH; - else if (!av_strcasecmp(arg, "drop")) video_sync_method = VSYNC_DROP; if (video_sync_method == VSYNC_AUTO) video_sync_method = parse_number_or_die("vsync", arg, OPT_INT, VSYNC_AUTO, VSYNC_VFR); return 0; } -static int opt_timecode(void *optctx, const char *opt, const char *arg) -{ - OptionsContext *o = optctx; - char *tcr = av_asprintf("timecode=%s", arg); - int ret = parse_option(o, "metadata:g", tcr, options); - if (ret >= 0) - ret = av_dict_set(&o->g->codec_opts, "gop_timecode", arg, 0); - av_free(tcr); - return 0; -} - static int opt_channel_layout(void *optctx, const char *opt, const char *arg) { OptionsContext *o = optctx; @@ -2457,7 +1949,7 @@ static int opt_channel_layout(void *optctx, const char *opt, const char *arg) return AVERROR(EINVAL); } snprintf(layout_str, sizeof(layout_str), "%"PRIu64, layout); - ret = opt_default_new(o, opt, layout_str); + ret = opt_default(NULL, opt, layout_str); if (ret < 0) return ret; @@ -2570,19 +2062,16 @@ void show_help_default(const char *opt, const char *arg) int flags = AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM; show_help_children(avcodec_get_class(), flags); show_help_children(avformat_get_class(), flags); -#if CONFIG_SWSCALE show_help_children(sws_get_class(), flags); -#endif - show_help_children(swr_get_class(), AV_OPT_FLAG_AUDIO_PARAM); - show_help_children(avfilter_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM); + show_help_children(avfilter_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM); } } void show_usage(void) { - av_log(NULL, AV_LOG_INFO, "Hyper fast Audio and Video encoder\n"); - av_log(NULL, AV_LOG_INFO, "usage: %s [options] [[infile options] -i infile]... {[outfile options] outfile}...\n", program_name); - av_log(NULL, AV_LOG_INFO, "\n"); + printf("Hyper fast Audio and Video encoder\n"); + printf("usage: %s [options] [[infile options] -i infile]... {[outfile options] outfile}...\n", program_name); + printf("\n"); } enum OptGroup { @@ -2628,7 +2117,7 @@ static int open_files(OptionGroupList *l, const char *inout, return 0; } -int ffmpeg_parse_options(int argc, char **argv) +int avconv_parse_options(int argc, char **argv) { OptionParseContext octx; uint8_t error[128]; @@ -2674,23 +2163,6 @@ fail: return ret; } -static int opt_progress(void *optctx, const char *opt, const char *arg) -{ - AVIOContext *avio = NULL; - int ret; - - if (!strcmp(arg, "-")) - arg = "pipe:"; - ret = avio_open2(&avio, arg, AVIO_FLAG_WRITE, &int_cb, NULL); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Failed to open progress URL \"%s\": %s\n", - arg, av_err2str(ret)); - return ret; - } - progress_avio = avio; - return 0; -} - #define OFFSET(x) offsetof(OptionsContext, x) const OptionDef options[] = { /* main options */ @@ -2700,7 +2172,7 @@ const OptionDef options[] = { "force format", "fmt" }, { "y", OPT_BOOL, { &file_overwrite }, "overwrite output files" }, - { "n", OPT_BOOL, { &no_file_overwrite }, + { "n", OPT_BOOL, { &file_skip }, "never overwrite output files" }, { "c", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(codec_names) }, @@ -2715,8 +2187,6 @@ const OptionDef options[] = { OPT_OUTPUT, { .func_arg = opt_map }, "set input stream mapping", "[-]input_file_id[:stream_specifier][,sync_file_id[:stream_specifier]]" }, - { "map_channel", HAS_ARG | OPT_EXPERT | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_map_channel }, - "map an audio channel from one stream to another", "file.stream.channel[:syncfile.syncstream]" }, { "map_metadata", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(metadata_map) }, "set metadata information of outfile from infile", @@ -2728,8 +2198,6 @@ const OptionDef options[] = { OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(recording_time) }, "record or transcode \"duration\" seconds of audio/video", "duration" }, - { "to", HAS_ARG | OPT_TIME | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(stop_time) }, - "record or transcode stop time", "time_stop" }, { "fs", HAS_ARG | OPT_INT64 | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(limit_filesize) }, "set the limit file size in bytes", "limit_size" }, { "ss", HAS_ARG | OPT_TIME | OPT_OFFSET | @@ -2744,8 +2212,6 @@ const OptionDef options[] = { { "itsscale", HAS_ARG | OPT_DOUBLE | OPT_SPEC | OPT_EXPERT | OPT_INPUT, { .off = OFFSET(ts_scale) }, "set the input ts scale", "scale" }, - { "timestamp", HAS_ARG | OPT_PERFILE, { .func_arg = opt_recording_timestamp }, - "set the recording timestamp ('now' to set the current time)", "time" }, { "metadata", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(metadata) }, "add metadata", "string=string" }, { "dframes", HAS_ARG | OPT_PERFILE | OPT_EXPERT | @@ -2753,12 +2219,6 @@ const OptionDef options[] = { "set the number of data frames to record", "number" }, { "benchmark", OPT_BOOL | OPT_EXPERT, { &do_benchmark }, "add timings for benchmarking" }, - { "benchmark_all", OPT_BOOL | OPT_EXPERT, { &do_benchmark_all }, - "add timings for each task" }, - { "progress", HAS_ARG | OPT_EXPERT, { .func_arg = opt_progress }, - "write program-readable progress information", "url" }, - { "stdin", OPT_BOOL | OPT_EXPERT, { &stdin_interaction }, - "enable or disable interaction on standard input" }, { "timelimit", HAS_ARG | OPT_EXPERT, { .func_arg = opt_timelimit }, "set max runtime in seconds", "limit" }, { "dump", OPT_BOOL | OPT_EXPERT, { &do_pkt_dump }, @@ -2779,25 +2239,18 @@ const OptionDef options[] = { "audio drift threshold", "threshold" }, { "copyts", OPT_BOOL | OPT_EXPERT, { ©_ts }, "copy timestamps" }, - { "copytb", HAS_ARG | OPT_INT | OPT_EXPERT, { ©_tb }, - "copy input stream time base when stream copying", "mode" }, + { "copytb", OPT_BOOL | OPT_EXPERT, { ©_tb }, + "copy input stream time base when stream copying" }, { "shortest", OPT_BOOL | OPT_EXPERT | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(shortest) }, "finish encoding within shortest input" }, - { "apad", OPT_STRING | HAS_ARG | OPT_SPEC | - OPT_OUTPUT, { .off = OFFSET(apad) }, - "audio pad", "" }, { "dts_delta_threshold", HAS_ARG | OPT_FLOAT | OPT_EXPERT, { &dts_delta_threshold }, "timestamp discontinuity delta threshold", "threshold" }, - { "dts_error_threshold", HAS_ARG | OPT_FLOAT | OPT_EXPERT, { &dts_error_threshold }, - "timestamp error delta threshold", "threshold" }, { "xerror", OPT_BOOL | OPT_EXPERT, { &exit_on_error }, "exit on error", "error" }, { "copyinkf", OPT_BOOL | OPT_EXPERT | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(copy_initial_nonkeyframes) }, "copy initial non-keyframes" }, - { "copypriorss", OPT_INT | HAS_ARG | OPT_EXPERT | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(copy_prior_start) }, - "copy or discard frames before start time" }, { "frames", OPT_INT64 | HAS_ARG | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(max_frames) }, "set the number of frames to record", "number" }, { "tag", OPT_STRING | HAS_ARG | OPT_SPEC | @@ -2806,21 +2259,15 @@ const OptionDef options[] = { { "q", HAS_ARG | OPT_EXPERT | OPT_DOUBLE | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(qscale) }, "use fixed quality scale (VBR)", "q" }, - { "qscale", HAS_ARG | OPT_EXPERT | OPT_PERFILE | - OPT_OUTPUT, { .func_arg = opt_qscale }, + { "qscale", HAS_ARG | OPT_EXPERT | OPT_DOUBLE | + OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(qscale) }, "use fixed quality scale (VBR)", "q" }, - { "profile", HAS_ARG | OPT_EXPERT | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_profile }, - "set profile", "profile" }, { "filter", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(filters) }, - "set stream filtergraph", "filter_graph" }, + "set stream filterchain", "filter_list" }, { "filter_script", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(filter_scripts) }, "read stream filtergraph description from a file", "filename" }, - { "reinit_filter", HAS_ARG | OPT_INT | OPT_SPEC | OPT_INPUT, { .off = OFFSET(reinit_filters) }, - "reinit filtergraph on input parameter changes", "" }, { "filter_complex", HAS_ARG | OPT_EXPERT, { .func_arg = opt_filter_complex }, "create a complex filtergraph", "graph_description" }, - { "lavfi", HAS_ARG | OPT_EXPERT, { .func_arg = opt_filter_complex }, - "create a complex filtergraph", "graph_description" }, { "filter_complex_script", HAS_ARG | OPT_EXPERT, { .func_arg = opt_filter_complex_script }, "read complex filtergraph description from a file", "filename" }, { "stats", OPT_BOOL, { &print_stats }, @@ -2831,10 +2278,6 @@ const OptionDef options[] = { { "dump_attachment", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_EXPERT | OPT_INPUT, { .off = OFFSET(dump_attachment) }, "extract an attachment into a file", "filename" }, - { "debug_ts", OPT_BOOL | OPT_EXPERT, { &debug_ts }, - "print timestamp debugging info" }, - { "max_error_rate", HAS_ARG | OPT_FLOAT, { &max_error_rate }, - "maximum error rate", "ratio of errors (0.0: no errors, 1.0: 100% errors) above which ffmpeg returns an error instead of success." }, /* video options */ { "vframes", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_video_frames }, @@ -2842,7 +2285,7 @@ const OptionDef options[] = { { "r", OPT_VIDEO | HAS_ARG | OPT_STRING | OPT_SPEC | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(frame_rates) }, "set frame rate (Hz value, fraction or abbreviation)", "rate" }, - { "s", OPT_VIDEO | HAS_ARG | OPT_SUBTITLE | OPT_STRING | OPT_SPEC | + { "s", OPT_VIDEO | HAS_ARG | OPT_STRING | OPT_SPEC | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(frame_sizes) }, "set frame size (WxH or abbreviation)", "size" }, { "aspect", OPT_VIDEO | HAS_ARG | OPT_STRING | OPT_SPEC | @@ -2851,11 +2294,7 @@ const OptionDef options[] = { { "pix_fmt", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_STRING | OPT_SPEC | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(frame_pix_fmts) }, "set pixel format", "format" }, - { "bits_per_raw_sample", OPT_VIDEO | OPT_INT | HAS_ARG, { &frame_bits_per_raw_sample }, - "set the number of bits per raw sample", "number" }, - { "intra", OPT_VIDEO | OPT_BOOL | OPT_EXPERT, { &intra_only }, - "deprecated use -g 1" }, - { "vn", OPT_VIDEO | OPT_BOOL | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT,{ .off = OFFSET(video_disable) }, + { "vn", OPT_VIDEO | OPT_BOOL | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(video_disable) }, "disable video" }, { "vdt", OPT_VIDEO | OPT_INT | HAS_ARG | OPT_EXPERT , { &video_discard }, "discard threshold", "n" }, @@ -2865,43 +2304,30 @@ const OptionDef options[] = { { "vcodec", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_INPUT | OPT_OUTPUT, { .func_arg = opt_video_codec }, "force video codec ('copy' to copy stream)", "codec" }, - { "sameq", OPT_VIDEO | OPT_EXPERT , { .func_arg = opt_sameq }, - "Removed" }, - { "same_quant", OPT_VIDEO | OPT_EXPERT , { .func_arg = opt_sameq }, - "Removed" }, - { "timecode", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_timecode }, - "set initial TimeCode value.", "hh:mm:ss[:;.]ff" }, { "pass", OPT_VIDEO | HAS_ARG | OPT_SPEC | OPT_INT | OPT_OUTPUT, { .off = OFFSET(pass) }, - "select the pass number (1 to 3)", "n" }, + "select the pass number (1 or 2)", "n" }, { "passlogfile", OPT_VIDEO | HAS_ARG | OPT_STRING | OPT_EXPERT | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(passlogfiles) }, "select two pass log file name prefix", "prefix" }, - { "deinterlace", OPT_VIDEO | OPT_BOOL | OPT_EXPERT, { &do_deinterlace }, - "this option is deprecated, use the yadif filter instead" }, - { "psnr", OPT_VIDEO | OPT_BOOL | OPT_EXPERT, { &do_psnr }, - "calculate PSNR of compressed frames" }, { "vstats", OPT_VIDEO | OPT_EXPERT , { &opt_vstats }, "dump video coding statistics to file" }, { "vstats_file", OPT_VIDEO | HAS_ARG | OPT_EXPERT , { opt_vstats_file }, "dump video coding statistics to file", "file" }, { "vf", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_video_filters }, - "set video filters", "filter_graph" }, + "video filters", "filter list" }, { "intra_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(intra_matrices) }, "specify intra matrix coeffs", "matrix" }, { "inter_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_STRING | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(inter_matrices) }, "specify inter matrix coeffs", "matrix" }, - { "chroma_intra_matrix", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_STRING | OPT_SPEC | - OPT_OUTPUT, { .off = OFFSET(chroma_intra_matrices) }, - "specify intra matrix coeffs", "matrix" }, { "top", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_INT| OPT_SPEC | - OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(top_field_first) }, + OPT_OUTPUT, { .off = OFFSET(top_field_first) }, "top=1/bottom=0/auto=-1 field first", "" }, { "dc", OPT_VIDEO | OPT_INT | HAS_ARG | OPT_EXPERT , { &intra_dc_precision }, "intra_dc_precision", "precision" }, { "vtag", OPT_VIDEO | HAS_ARG | OPT_EXPERT | OPT_PERFILE | - OPT_OUTPUT, { .func_arg = opt_old2new }, + OPT_OUTPUT, { .func_arg = opt_video_tag }, "force video tag/fourcc", "fourcc/tag" }, { "qphist", OPT_VIDEO | OPT_BOOL | OPT_EXPERT , { &qp_hist }, "show QP histogram" }, @@ -2914,8 +2340,6 @@ const OptionDef options[] = { { "force_key_frames", OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT | OPT_SPEC | OPT_OUTPUT, { .off = OFFSET(forced_key_frames) }, "force key frames at specified timestamps", "timestamps" }, - { "b", OPT_VIDEO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_bitrate }, - "video bitrate (please use -b:v)", "bitrate" }, { "hwaccel", OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT | OPT_SPEC | OPT_INPUT, { .off = OFFSET(hwaccels) }, "use HW accelerated decoding", "hwaccel name" }, @@ -2934,13 +2358,13 @@ const OptionDef options[] = { { "ac", OPT_AUDIO | HAS_ARG | OPT_INT | OPT_SPEC | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(audio_channels) }, "set number of audio channels", "channels" }, - { "an", OPT_AUDIO | OPT_BOOL | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT,{ .off = OFFSET(audio_disable) }, + { "an", OPT_AUDIO | OPT_BOOL | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(audio_disable) }, "disable audio" }, { "acodec", OPT_AUDIO | HAS_ARG | OPT_PERFILE | OPT_INPUT | OPT_OUTPUT, { .func_arg = opt_audio_codec }, "force audio codec ('copy' to copy stream)", "codec" }, { "atag", OPT_AUDIO | HAS_ARG | OPT_EXPERT | OPT_PERFILE | - OPT_OUTPUT, { .func_arg = opt_old2new }, + OPT_OUTPUT, { .func_arg = opt_audio_tag }, "force audio tag/fourcc", "fourcc/tag" }, { "vol", OPT_AUDIO | HAS_ARG | OPT_INT, { &audio_volume }, "change audio volume (256=normal)" , "volume" }, @@ -2951,27 +2375,17 @@ const OptionDef options[] = { OPT_INPUT | OPT_OUTPUT, { .func_arg = opt_channel_layout }, "set channel layout", "layout" }, { "af", OPT_AUDIO | HAS_ARG | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_audio_filters }, - "set audio filters", "filter_graph" }, - { "guess_layout_max", OPT_AUDIO | HAS_ARG | OPT_INT | OPT_SPEC | OPT_EXPERT | OPT_INPUT, { .off = OFFSET(guess_layout_max) }, - "set the maximum number of channels to try to guess the channel layout" }, + "audio filters", "filter list" }, /* subtitle options */ - { "sn", OPT_SUBTITLE | OPT_BOOL | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(subtitle_disable) }, + { "sn", OPT_SUBTITLE | OPT_BOOL | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(subtitle_disable) }, "disable subtitle" }, { "scodec", OPT_SUBTITLE | HAS_ARG | OPT_PERFILE | OPT_INPUT | OPT_OUTPUT, { .func_arg = opt_subtitle_codec }, "force subtitle codec ('copy' to copy stream)", "codec" }, - { "stag", OPT_SUBTITLE | HAS_ARG | OPT_EXPERT | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_old2new } + { "stag", OPT_SUBTITLE | HAS_ARG | OPT_EXPERT | OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_subtitle_tag } , "force subtitle tag/fourcc", "fourcc/tag" }, - { "fix_sub_duration", OPT_BOOL | OPT_EXPERT | OPT_SUBTITLE | OPT_SPEC | OPT_INPUT, { .off = OFFSET(fix_sub_duration) }, - "fix subtitles duration" }, - { "canvas_size", OPT_SUBTITLE | HAS_ARG | OPT_STRING | OPT_SPEC | OPT_INPUT, { .off = OFFSET(canvas_sizes) }, - "set canvas size (WxH or abbreviation)", "size" }, /* grab options */ - { "vc", HAS_ARG | OPT_EXPERT | OPT_VIDEO, { .func_arg = opt_video_channel }, - "deprecated, use -channel", "channel" }, - { "tvstd", HAS_ARG | OPT_EXPERT | OPT_VIDEO, { .func_arg = opt_video_standard }, - "deprecated, use -standard", "standard" }, { "isync", OPT_BOOL | OPT_EXPERT, { &input_sync }, "this option is deprecated and does nothing", "" }, /* muxer options */ @@ -2979,29 +2393,13 @@ const OptionDef options[] = { "set the maximum demux-decode delay", "seconds" }, { "muxpreload", OPT_FLOAT | HAS_ARG | OPT_EXPERT | OPT_OFFSET | OPT_OUTPUT, { .off = OFFSET(mux_preload) }, "set the initial demux-decode delay", "seconds" }, - { "override_ffserver", OPT_BOOL | OPT_EXPERT | OPT_OUTPUT, { &override_ffserver }, - "override the options from ffserver", "" }, { "bsf", HAS_ARG | OPT_STRING | OPT_SPEC | OPT_EXPERT | OPT_OUTPUT, { .off = OFFSET(bitstream_filters) }, "A comma-separated list of bitstream filters", "bitstream_filters" }, - { "absf", HAS_ARG | OPT_AUDIO | OPT_EXPERT| OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_old2new }, - "deprecated", "audio bitstream_filters" }, - { "vbsf", OPT_VIDEO | HAS_ARG | OPT_EXPERT| OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_old2new }, - "deprecated", "video bitstream_filters" }, - { "apre", HAS_ARG | OPT_AUDIO | OPT_EXPERT| OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_preset }, - "set the audio options to the indicated preset", "preset" }, - { "vpre", OPT_VIDEO | HAS_ARG | OPT_EXPERT| OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_preset }, - "set the video options to the indicated preset", "preset" }, - { "spre", HAS_ARG | OPT_SUBTITLE | OPT_EXPERT| OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_preset }, - "set the subtitle options to the indicated preset", "preset" }, - { "fpre", HAS_ARG | OPT_EXPERT| OPT_PERFILE | OPT_OUTPUT, { .func_arg = opt_preset }, - "set options from indicated preset file", "filename" }, /* data codec support */ { "dcodec", HAS_ARG | OPT_DATA | OPT_PERFILE | OPT_EXPERT | OPT_INPUT | OPT_OUTPUT, { .func_arg = opt_data_codec }, "force data codec ('copy' to copy stream)", "codec" }, - { "dn", OPT_BOOL | OPT_VIDEO | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT, { .off = OFFSET(data_disable) }, - "disable data" }, { NULL, }, }; diff --git a/ffmpeg_vda.c b/avconv_vda.c similarity index 93% rename from ffmpeg_vda.c rename to avconv_vda.c index b9f0975f55..40f87c4f76 100644 --- a/ffmpeg_vda.c +++ b/avconv_vda.c @@ -1,18 +1,18 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -20,7 +20,7 @@ #include "libavcodec/vda.h" #include "libavutil/imgutils.h" -#include "ffmpeg.h" +#include "avconv.h" typedef struct VDAContext { AVFrame *tmp_frame; diff --git a/ffmpeg_vdpau.c b/avconv_vdpau.c similarity index 97% rename from ffmpeg_vdpau.c rename to avconv_vdpau.c index fe0930695e..820678e36a 100644 --- a/ffmpeg_vdpau.c +++ b/avconv_vdpau.c @@ -1,18 +1,18 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -23,7 +23,7 @@ #include -#include "ffmpeg.h" +#include "avconv.h" #include "libavcodec/vdpau.h" diff --git a/avplay.c b/avplay.c new file mode 100644 index 0000000000..b70ee54c7b --- /dev/null +++ b/avplay.c @@ -0,0 +1,3001 @@ +/* + * avplay : Simple Media Player based on the Libav libraries + * Copyright (c) 2003 Fabrice Bellard + * + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "config.h" +#include +#include +#include +#include + +#include "libavutil/avstring.h" +#include "libavutil/colorspace.h" +#include "libavutil/mathematics.h" +#include "libavutil/pixdesc.h" +#include "libavutil/imgutils.h" +#include "libavutil/dict.h" +#include "libavutil/parseutils.h" +#include "libavutil/samplefmt.h" +#include "libavutil/time.h" +#include "libavformat/avformat.h" +#include "libavdevice/avdevice.h" +#include "libswscale/swscale.h" +#include "libavresample/avresample.h" +#include "libavutil/opt.h" +#include "libavcodec/avfft.h" + +#if CONFIG_AVFILTER +# include "libavfilter/avfilter.h" +# include "libavfilter/buffersink.h" +# include "libavfilter/buffersrc.h" +#endif + +#include "cmdutils.h" + +#include +#include + +#ifdef __MINGW32__ +#undef main /* We don't want SDL to override our main() */ +#endif + +#include + +const char program_name[] = "avplay"; +const int program_birth_year = 2003; + +#define MAX_QUEUE_SIZE (15 * 1024 * 1024) +#define MIN_AUDIOQ_SIZE (20 * 16 * 1024) +#define MIN_FRAMES 5 + +/* SDL audio buffer size, in samples. Should be small to have precise + A/V sync as SDL does not have hardware buffer fullness info. */ +#define SDL_AUDIO_BUFFER_SIZE 1024 + +/* no AV sync correction is done if below the AV sync threshold */ +#define AV_SYNC_THRESHOLD 0.01 +/* no AV correction is done if too big error */ +#define AV_NOSYNC_THRESHOLD 10.0 + +#define FRAME_SKIP_FACTOR 0.05 + +/* maximum audio speed change to get correct sync */ +#define SAMPLE_CORRECTION_PERCENT_MAX 10 + +/* we use about AUDIO_DIFF_AVG_NB A-V differences to make the average */ +#define AUDIO_DIFF_AVG_NB 20 + +/* NOTE: the size must be big enough to compensate the hardware audio buffersize size */ +#define SAMPLE_ARRAY_SIZE (2 * 65536) + +static int64_t sws_flags = SWS_BICUBIC; + +typedef struct PacketQueue { + AVPacketList *first_pkt, *last_pkt; + int nb_packets; + int size; + int abort_request; + SDL_mutex *mutex; + SDL_cond *cond; +} PacketQueue; + +#define VIDEO_PICTURE_QUEUE_SIZE 2 +#define SUBPICTURE_QUEUE_SIZE 4 + +typedef struct VideoPicture { + double pts; // presentation timestamp for this picture + double target_clock; // av_gettime() time at which this should be displayed ideally + int64_t pos; // byte position in file + SDL_Overlay *bmp; + int width, height; /* source height & width */ + int allocated; + int reallocate; + enum AVPixelFormat pix_fmt; + + AVRational sar; +} VideoPicture; + +typedef struct SubPicture { + double pts; /* presentation time stamp for this picture */ + AVSubtitle sub; +} SubPicture; + +enum { + AV_SYNC_AUDIO_MASTER, /* default choice */ + AV_SYNC_VIDEO_MASTER, + AV_SYNC_EXTERNAL_CLOCK, /* synchronize to an external clock */ +}; + +typedef struct VideoState { + SDL_Thread *parse_tid; + SDL_Thread *video_tid; + SDL_Thread *refresh_tid; + AVInputFormat *iformat; + int no_background; + int abort_request; + int paused; + int last_paused; + int seek_req; + int seek_flags; + int64_t seek_pos; + int64_t seek_rel; + int read_pause_return; + AVFormatContext *ic; + + int audio_stream; + + int av_sync_type; + double external_clock; /* external clock base */ + int64_t external_clock_time; + + double audio_clock; + double audio_diff_cum; /* used for AV difference average computation */ + double audio_diff_avg_coef; + double audio_diff_threshold; + int audio_diff_avg_count; + AVStream *audio_st; + PacketQueue audioq; + int audio_hw_buf_size; + uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE]; + uint8_t *audio_buf; + uint8_t *audio_buf1; + unsigned int audio_buf_size; /* in bytes */ + int audio_buf_index; /* in bytes */ + AVPacket audio_pkt_temp; + AVPacket audio_pkt; + enum AVSampleFormat sdl_sample_fmt; + uint64_t sdl_channel_layout; + int sdl_channels; + int sdl_sample_rate; + enum AVSampleFormat resample_sample_fmt; + uint64_t resample_channel_layout; + int resample_sample_rate; + AVAudioResampleContext *avr; + AVFrame *frame; + + int show_audio; /* if true, display audio samples */ + int16_t sample_array[SAMPLE_ARRAY_SIZE]; + int sample_array_index; + int last_i_start; + RDFTContext *rdft; + int rdft_bits; + FFTSample *rdft_data; + int xpos; + + SDL_Thread *subtitle_tid; + int subtitle_stream; + int subtitle_stream_changed; + AVStream *subtitle_st; + PacketQueue subtitleq; + SubPicture subpq[SUBPICTURE_QUEUE_SIZE]; + int subpq_size, subpq_rindex, subpq_windex; + SDL_mutex *subpq_mutex; + SDL_cond *subpq_cond; + + double frame_timer; + double frame_last_pts; + double frame_last_delay; + double video_clock; // pts of last decoded frame / predicted pts of next decoded frame + int video_stream; + AVStream *video_st; + PacketQueue videoq; + double video_current_pts; // current displayed pts (different from video_clock if frame fifos are used) + double video_current_pts_drift; // video_current_pts - time (av_gettime) at which we updated video_current_pts - used to have running video pts + int64_t video_current_pos; // current displayed file pos + VideoPicture pictq[VIDEO_PICTURE_QUEUE_SIZE]; + int pictq_size, pictq_rindex, pictq_windex; + SDL_mutex *pictq_mutex; + SDL_cond *pictq_cond; +#if !CONFIG_AVFILTER + struct SwsContext *img_convert_ctx; +#endif + + // QETimer *video_timer; + char filename[1024]; + int width, height, xleft, ytop; + + PtsCorrectionContext pts_ctx; + +#if CONFIG_AVFILTER + AVFilterContext *in_video_filter; // the first filter in the video chain + AVFilterContext *out_video_filter; // the last filter in the video chain +#endif + + float skip_frames; + float skip_frames_index; + int refresh; +} VideoState; + +/* options specified by the user */ +static AVInputFormat *file_iformat; +static const char *input_filename; +static const char *window_title; +static int fs_screen_width; +static int fs_screen_height; +static int screen_width = 0; +static int screen_height = 0; +static int audio_disable; +static int video_disable; +static int wanted_stream[AVMEDIA_TYPE_NB] = { + [AVMEDIA_TYPE_AUDIO] = -1, + [AVMEDIA_TYPE_VIDEO] = -1, + [AVMEDIA_TYPE_SUBTITLE] = -1, +}; +static int seek_by_bytes = -1; +static int display_disable; +static int show_status = 1; +static int av_sync_type = AV_SYNC_AUDIO_MASTER; +static int64_t start_time = AV_NOPTS_VALUE; +static int64_t duration = AV_NOPTS_VALUE; +static int step = 0; +static int workaround_bugs = 1; +static int fast = 0; +static int genpts = 0; +static int idct = FF_IDCT_AUTO; +static enum AVDiscard skip_frame = AVDISCARD_DEFAULT; +static enum AVDiscard skip_idct = AVDISCARD_DEFAULT; +static enum AVDiscard skip_loop_filter = AVDISCARD_DEFAULT; +static int error_concealment = 3; +static int decoder_reorder_pts = -1; +static int autoexit; +static int exit_on_keydown; +static int exit_on_mousedown; +static int loop = 1; +static int framedrop = 1; +static int infinite_buffer = 0; + +static int rdftspeed = 20; +#if CONFIG_AVFILTER +static char *vfilters = NULL; +#endif + +/* current context */ +static int is_full_screen; +static VideoState *cur_stream; +static int64_t audio_callback_time; + +static AVPacket flush_pkt; + +#define FF_ALLOC_EVENT (SDL_USEREVENT) +#define FF_REFRESH_EVENT (SDL_USEREVENT + 1) +#define FF_QUIT_EVENT (SDL_USEREVENT + 2) + +static SDL_Surface *screen; + +static int packet_queue_put(PacketQueue *q, AVPacket *pkt); + +/* packet queue handling */ +static void packet_queue_init(PacketQueue *q) +{ + memset(q, 0, sizeof(PacketQueue)); + q->mutex = SDL_CreateMutex(); + q->cond = SDL_CreateCond(); + packet_queue_put(q, &flush_pkt); +} + +static void packet_queue_flush(PacketQueue *q) +{ + AVPacketList *pkt, *pkt1; + + SDL_LockMutex(q->mutex); + for (pkt = q->first_pkt; pkt != NULL; pkt = pkt1) { + pkt1 = pkt->next; + av_free_packet(&pkt->pkt); + av_freep(&pkt); + } + q->last_pkt = NULL; + q->first_pkt = NULL; + q->nb_packets = 0; + q->size = 0; + SDL_UnlockMutex(q->mutex); +} + +static void packet_queue_end(PacketQueue *q) +{ + packet_queue_flush(q); + SDL_DestroyMutex(q->mutex); + SDL_DestroyCond(q->cond); +} + +static int packet_queue_put(PacketQueue *q, AVPacket *pkt) +{ + AVPacketList *pkt1; + + /* duplicate the packet */ + if (pkt != &flush_pkt && av_dup_packet(pkt) < 0) + return -1; + + pkt1 = av_malloc(sizeof(AVPacketList)); + if (!pkt1) + return -1; + pkt1->pkt = *pkt; + pkt1->next = NULL; + + + SDL_LockMutex(q->mutex); + + if (!q->last_pkt) + + q->first_pkt = pkt1; + else + q->last_pkt->next = pkt1; + q->last_pkt = pkt1; + q->nb_packets++; + q->size += pkt1->pkt.size + sizeof(*pkt1); + /* XXX: should duplicate packet data in DV case */ + SDL_CondSignal(q->cond); + + SDL_UnlockMutex(q->mutex); + return 0; +} + +static void packet_queue_abort(PacketQueue *q) +{ + SDL_LockMutex(q->mutex); + + q->abort_request = 1; + + SDL_CondSignal(q->cond); + + SDL_UnlockMutex(q->mutex); +} + +/* return < 0 if aborted, 0 if no packet and > 0 if packet. */ +static int packet_queue_get(PacketQueue *q, AVPacket *pkt, int block) +{ + AVPacketList *pkt1; + int ret; + + SDL_LockMutex(q->mutex); + + for (;;) { + if (q->abort_request) { + ret = -1; + break; + } + + pkt1 = q->first_pkt; + if (pkt1) { + q->first_pkt = pkt1->next; + if (!q->first_pkt) + q->last_pkt = NULL; + q->nb_packets--; + q->size -= pkt1->pkt.size + sizeof(*pkt1); + *pkt = pkt1->pkt; + av_free(pkt1); + ret = 1; + break; + } else if (!block) { + ret = 0; + break; + } else { + SDL_CondWait(q->cond, q->mutex); + } + } + SDL_UnlockMutex(q->mutex); + return ret; +} + +static inline void fill_rectangle(SDL_Surface *screen, + int x, int y, int w, int h, int color) +{ + SDL_Rect rect; + rect.x = x; + rect.y = y; + rect.w = w; + rect.h = h; + SDL_FillRect(screen, &rect, color); +} + +#define ALPHA_BLEND(a, oldp, newp, s)\ +((((oldp << s) * (255 - (a))) + (newp * (a))) / (255 << s)) + +#define RGBA_IN(r, g, b, a, s)\ +{\ + unsigned int v = ((const uint32_t *)(s))[0];\ + a = (v >> 24) & 0xff;\ + r = (v >> 16) & 0xff;\ + g = (v >> 8) & 0xff;\ + b = v & 0xff;\ +} + +#define YUVA_IN(y, u, v, a, s, pal)\ +{\ + unsigned int val = ((const uint32_t *)(pal))[*(const uint8_t*)(s)];\ + a = (val >> 24) & 0xff;\ + y = (val >> 16) & 0xff;\ + u = (val >> 8) & 0xff;\ + v = val & 0xff;\ +} + +#define YUVA_OUT(d, y, u, v, a)\ +{\ + ((uint32_t *)(d))[0] = (a << 24) | (y << 16) | (u << 8) | v;\ +} + + +#define BPP 1 + +static void blend_subrect(AVPicture *dst, const AVSubtitleRect *rect, int imgw, int imgh) +{ + int wrap, wrap3, width2, skip2; + int y, u, v, a, u1, v1, a1, w, h; + uint8_t *lum, *cb, *cr; + const uint8_t *p; + const uint32_t *pal; + int dstx, dsty, dstw, dsth; + + dstw = av_clip(rect->w, 0, imgw); + dsth = av_clip(rect->h, 0, imgh); + dstx = av_clip(rect->x, 0, imgw - dstw); + dsty = av_clip(rect->y, 0, imgh - dsth); + lum = dst->data[0] + dsty * dst->linesize[0]; + cb = dst->data[1] + (dsty >> 1) * dst->linesize[1]; + cr = dst->data[2] + (dsty >> 1) * dst->linesize[2]; + + width2 = ((dstw + 1) >> 1) + (dstx & ~dstw & 1); + skip2 = dstx >> 1; + wrap = dst->linesize[0]; + wrap3 = rect->pict.linesize[0]; + p = rect->pict.data[0]; + pal = (const uint32_t *)rect->pict.data[1]; /* Now in YCrCb! */ + + if (dsty & 1) { + lum += dstx; + cb += skip2; + cr += skip2; + + if (dstx & 1) { + YUVA_IN(y, u, v, a, p, pal); + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0); + cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0); + cb++; + cr++; + lum++; + p += BPP; + } + for (w = dstw - (dstx & 1); w >= 2; w -= 2) { + YUVA_IN(y, u, v, a, p, pal); + u1 = u; + v1 = v; + a1 = a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + + YUVA_IN(y, u, v, a, p + BPP, pal); + u1 += u; + v1 += v; + a1 += a; + lum[1] = ALPHA_BLEND(a, lum[1], y, 0); + cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1); + cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1); + cb++; + cr++; + p += 2 * BPP; + lum += 2; + } + if (w) { + YUVA_IN(y, u, v, a, p, pal); + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0); + cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0); + p++; + lum++; + } + p += wrap3 - dstw * BPP; + lum += wrap - dstw - dstx; + cb += dst->linesize[1] - width2 - skip2; + cr += dst->linesize[2] - width2 - skip2; + } + for (h = dsth - (dsty & 1); h >= 2; h -= 2) { + lum += dstx; + cb += skip2; + cr += skip2; + + if (dstx & 1) { + YUVA_IN(y, u, v, a, p, pal); + u1 = u; + v1 = v; + a1 = a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + p += wrap3; + lum += wrap; + YUVA_IN(y, u, v, a, p, pal); + u1 += u; + v1 += v; + a1 += a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1); + cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1); + cb++; + cr++; + p += -wrap3 + BPP; + lum += -wrap + 1; + } + for (w = dstw - (dstx & 1); w >= 2; w -= 2) { + YUVA_IN(y, u, v, a, p, pal); + u1 = u; + v1 = v; + a1 = a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + + YUVA_IN(y, u, v, a, p + BPP, pal); + u1 += u; + v1 += v; + a1 += a; + lum[1] = ALPHA_BLEND(a, lum[1], y, 0); + p += wrap3; + lum += wrap; + + YUVA_IN(y, u, v, a, p, pal); + u1 += u; + v1 += v; + a1 += a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + + YUVA_IN(y, u, v, a, p + BPP, pal); + u1 += u; + v1 += v; + a1 += a; + lum[1] = ALPHA_BLEND(a, lum[1], y, 0); + + cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 2); + cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 2); + + cb++; + cr++; + p += -wrap3 + 2 * BPP; + lum += -wrap + 2; + } + if (w) { + YUVA_IN(y, u, v, a, p, pal); + u1 = u; + v1 = v; + a1 = a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + p += wrap3; + lum += wrap; + YUVA_IN(y, u, v, a, p, pal); + u1 += u; + v1 += v; + a1 += a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1); + cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1); + cb++; + cr++; + p += -wrap3 + BPP; + lum += -wrap + 1; + } + p += wrap3 + (wrap3 - dstw * BPP); + lum += wrap + (wrap - dstw - dstx); + cb += dst->linesize[1] - width2 - skip2; + cr += dst->linesize[2] - width2 - skip2; + } + /* handle odd height */ + if (h) { + lum += dstx; + cb += skip2; + cr += skip2; + + if (dstx & 1) { + YUVA_IN(y, u, v, a, p, pal); + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0); + cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0); + cb++; + cr++; + lum++; + p += BPP; + } + for (w = dstw - (dstx & 1); w >= 2; w -= 2) { + YUVA_IN(y, u, v, a, p, pal); + u1 = u; + v1 = v; + a1 = a; + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + + YUVA_IN(y, u, v, a, p + BPP, pal); + u1 += u; + v1 += v; + a1 += a; + lum[1] = ALPHA_BLEND(a, lum[1], y, 0); + cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u, 1); + cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v, 1); + cb++; + cr++; + p += 2 * BPP; + lum += 2; + } + if (w) { + YUVA_IN(y, u, v, a, p, pal); + lum[0] = ALPHA_BLEND(a, lum[0], y, 0); + cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0); + cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0); + } + } +} + +static void free_subpicture(SubPicture *sp) +{ + avsubtitle_free(&sp->sub); +} + +static void video_image_display(VideoState *is) +{ + VideoPicture *vp; + SubPicture *sp; + AVPicture pict; + float aspect_ratio; + int width, height, x, y; + SDL_Rect rect; + int i; + + vp = &is->pictq[is->pictq_rindex]; + if (vp->bmp) { +#if CONFIG_AVFILTER + if (!vp->sar.num) + aspect_ratio = 0; + else + aspect_ratio = av_q2d(vp->sar); +#else + + /* XXX: use variable in the frame */ + if (is->video_st->sample_aspect_ratio.num) + aspect_ratio = av_q2d(is->video_st->sample_aspect_ratio); + else if (is->video_st->codec->sample_aspect_ratio.num) + aspect_ratio = av_q2d(is->video_st->codec->sample_aspect_ratio); + else + aspect_ratio = 0; +#endif + if (aspect_ratio <= 0.0) + aspect_ratio = 1.0; + aspect_ratio *= (float)vp->width / (float)vp->height; + + if (is->subtitle_st) + { + if (is->subpq_size > 0) + { + sp = &is->subpq[is->subpq_rindex]; + + if (vp->pts >= sp->pts + ((float) sp->sub.start_display_time / 1000)) + { + SDL_LockYUVOverlay (vp->bmp); + + pict.data[0] = vp->bmp->pixels[0]; + pict.data[1] = vp->bmp->pixels[2]; + pict.data[2] = vp->bmp->pixels[1]; + + pict.linesize[0] = vp->bmp->pitches[0]; + pict.linesize[1] = vp->bmp->pitches[2]; + pict.linesize[2] = vp->bmp->pitches[1]; + + for (i = 0; i < sp->sub.num_rects; i++) + blend_subrect(&pict, sp->sub.rects[i], + vp->bmp->w, vp->bmp->h); + + SDL_UnlockYUVOverlay (vp->bmp); + } + } + } + + + /* XXX: we suppose the screen has a 1.0 pixel ratio */ + height = is->height; + width = ((int)rint(height * aspect_ratio)) & ~1; + if (width > is->width) { + width = is->width; + height = ((int)rint(width / aspect_ratio)) & ~1; + } + x = (is->width - width) / 2; + y = (is->height - height) / 2; + is->no_background = 0; + rect.x = is->xleft + x; + rect.y = is->ytop + y; + rect.w = width; + rect.h = height; + SDL_DisplayYUVOverlay(vp->bmp, &rect); + } +} + +/* get the current audio output buffer size, in samples. With SDL, we + cannot have a precise information */ +static int audio_write_get_buf_size(VideoState *is) +{ + return is->audio_buf_size - is->audio_buf_index; +} + +static inline int compute_mod(int a, int b) +{ + a = a % b; + if (a >= 0) + return a; + else + return a + b; +} + +static void video_audio_display(VideoState *s) +{ + int i, i_start, x, y1, y, ys, delay, n, nb_display_channels; + int ch, channels, h, h2, bgcolor, fgcolor; + int16_t time_diff; + int rdft_bits, nb_freq; + + for (rdft_bits = 1; (1 << rdft_bits) < 2 * s->height; rdft_bits++) + ; + nb_freq = 1 << (rdft_bits - 1); + + /* compute display index : center on currently output samples */ + channels = s->sdl_channels; + nb_display_channels = channels; + if (!s->paused) { + int data_used = s->show_audio == 1 ? s->width : (2 * nb_freq); + n = 2 * channels; + delay = audio_write_get_buf_size(s); + delay /= n; + + /* to be more precise, we take into account the time spent since + the last buffer computation */ + if (audio_callback_time) { + time_diff = av_gettime() - audio_callback_time; + delay -= (time_diff * s->sdl_sample_rate) / 1000000; + } + + delay += 2 * data_used; + if (delay < data_used) + delay = data_used; + + i_start= x = compute_mod(s->sample_array_index - delay * channels, SAMPLE_ARRAY_SIZE); + if (s->show_audio == 1) { + h = INT_MIN; + for (i = 0; i < 1000; i += channels) { + int idx = (SAMPLE_ARRAY_SIZE + x - i) % SAMPLE_ARRAY_SIZE; + int a = s->sample_array[idx]; + int b = s->sample_array[(idx + 4 * channels) % SAMPLE_ARRAY_SIZE]; + int c = s->sample_array[(idx + 5 * channels) % SAMPLE_ARRAY_SIZE]; + int d = s->sample_array[(idx + 9 * channels) % SAMPLE_ARRAY_SIZE]; + int score = a - d; + if (h < score && (b ^ c) < 0) { + h = score; + i_start = idx; + } + } + } + + s->last_i_start = i_start; + } else { + i_start = s->last_i_start; + } + + bgcolor = SDL_MapRGB(screen->format, 0x00, 0x00, 0x00); + if (s->show_audio == 1) { + fill_rectangle(screen, + s->xleft, s->ytop, s->width, s->height, + bgcolor); + + fgcolor = SDL_MapRGB(screen->format, 0xff, 0xff, 0xff); + + /* total height for one channel */ + h = s->height / nb_display_channels; + /* graph height / 2 */ + h2 = (h * 9) / 20; + for (ch = 0; ch < nb_display_channels; ch++) { + i = i_start + ch; + y1 = s->ytop + ch * h + (h / 2); /* position of center line */ + for (x = 0; x < s->width; x++) { + y = (s->sample_array[i] * h2) >> 15; + if (y < 0) { + y = -y; + ys = y1 - y; + } else { + ys = y1; + } + fill_rectangle(screen, + s->xleft + x, ys, 1, y, + fgcolor); + i += channels; + if (i >= SAMPLE_ARRAY_SIZE) + i -= SAMPLE_ARRAY_SIZE; + } + } + + fgcolor = SDL_MapRGB(screen->format, 0x00, 0x00, 0xff); + + for (ch = 1; ch < nb_display_channels; ch++) { + y = s->ytop + ch * h; + fill_rectangle(screen, + s->xleft, y, s->width, 1, + fgcolor); + } + SDL_UpdateRect(screen, s->xleft, s->ytop, s->width, s->height); + } else { + nb_display_channels= FFMIN(nb_display_channels, 2); + if (rdft_bits != s->rdft_bits) { + av_rdft_end(s->rdft); + av_free(s->rdft_data); + s->rdft = av_rdft_init(rdft_bits, DFT_R2C); + s->rdft_bits = rdft_bits; + s->rdft_data = av_malloc(4 * nb_freq * sizeof(*s->rdft_data)); + } + { + FFTSample *data[2]; + for (ch = 0; ch < nb_display_channels; ch++) { + data[ch] = s->rdft_data + 2 * nb_freq * ch; + i = i_start + ch; + for (x = 0; x < 2 * nb_freq; x++) { + double w = (x-nb_freq) * (1.0 / nb_freq); + data[ch][x] = s->sample_array[i] * (1.0 - w * w); + i += channels; + if (i >= SAMPLE_ARRAY_SIZE) + i -= SAMPLE_ARRAY_SIZE; + } + av_rdft_calc(s->rdft, data[ch]); + } + /* Least efficient way to do this, we should of course + * directly access it but it is more than fast enough. */ + for (y = 0; y < s->height; y++) { + double w = 1 / sqrt(nb_freq); + int a = sqrt(w * sqrt(data[0][2 * y + 0] * data[0][2 * y + 0] + data[0][2 * y + 1] * data[0][2 * y + 1])); + int b = (nb_display_channels == 2 ) ? sqrt(w * sqrt(data[1][2 * y + 0] * data[1][2 * y + 0] + + data[1][2 * y + 1] * data[1][2 * y + 1])) : a; + a = FFMIN(a, 255); + b = FFMIN(b, 255); + fgcolor = SDL_MapRGB(screen->format, a, b, (a + b) / 2); + + fill_rectangle(screen, + s->xpos, s->height-y, 1, 1, + fgcolor); + } + } + SDL_UpdateRect(screen, s->xpos, s->ytop, 1, s->height); + s->xpos++; + if (s->xpos >= s->width) + s->xpos= s->xleft; + } +} + +static int video_open(VideoState *is) +{ + int flags = SDL_HWSURFACE | SDL_ASYNCBLIT | SDL_HWACCEL; + int w,h; + + if (is_full_screen) flags |= SDL_FULLSCREEN; + else flags |= SDL_RESIZABLE; + + if (is_full_screen && fs_screen_width) { + w = fs_screen_width; + h = fs_screen_height; + } else if (!is_full_screen && screen_width) { + w = screen_width; + h = screen_height; +#if CONFIG_AVFILTER + } else if (is->out_video_filter && is->out_video_filter->inputs[0]) { + w = is->out_video_filter->inputs[0]->w; + h = is->out_video_filter->inputs[0]->h; +#else + } else if (is->video_st && is->video_st->codec->width) { + w = is->video_st->codec->width; + h = is->video_st->codec->height; +#endif + } else { + w = 640; + h = 480; + } + if (screen && is->width == screen->w && screen->w == w + && is->height== screen->h && screen->h == h) + return 0; + +#if defined(__APPLE__) && !SDL_VERSION_ATLEAST(1, 2, 14) + /* setting bits_per_pixel = 0 or 32 causes blank video on OS X and older SDL */ + screen = SDL_SetVideoMode(w, h, 24, flags); +#else + screen = SDL_SetVideoMode(w, h, 0, flags); +#endif + if (!screen) { + fprintf(stderr, "SDL: could not set video mode - exiting\n"); + return -1; + } + if (!window_title) + window_title = input_filename; + SDL_WM_SetCaption(window_title, window_title); + + is->width = screen->w; + is->height = screen->h; + + return 0; +} + +/* display the current picture, if any */ +static void video_display(VideoState *is) +{ + if (!screen) + video_open(cur_stream); + if (is->audio_st && is->show_audio) + video_audio_display(is); + else if (is->video_st) + video_image_display(is); +} + +static int refresh_thread(void *opaque) +{ + VideoState *is= opaque; + while (!is->abort_request) { + SDL_Event event; + event.type = FF_REFRESH_EVENT; + event.user.data1 = opaque; + if (!is->refresh) { + is->refresh = 1; + SDL_PushEvent(&event); + } + av_usleep(is->audio_st && is->show_audio ? rdftspeed * 1000 : 5000); // FIXME ideally we should wait the correct time but SDLs event passing is so slow it would be silly + } + return 0; +} + +/* get the current audio clock value */ +static double get_audio_clock(VideoState *is) +{ + double pts; + int hw_buf_size, bytes_per_sec; + pts = is->audio_clock; + hw_buf_size = audio_write_get_buf_size(is); + bytes_per_sec = 0; + if (is->audio_st) { + bytes_per_sec = is->sdl_sample_rate * is->sdl_channels * + av_get_bytes_per_sample(is->sdl_sample_fmt); + } + if (bytes_per_sec) + pts -= (double)hw_buf_size / bytes_per_sec; + return pts; +} + +/* get the current video clock value */ +static double get_video_clock(VideoState *is) +{ + if (is->paused) { + return is->video_current_pts; + } else { + return is->video_current_pts_drift + av_gettime() / 1000000.0; + } +} + +/* get the current external clock value */ +static double get_external_clock(VideoState *is) +{ + int64_t ti; + ti = av_gettime(); + return is->external_clock + ((ti - is->external_clock_time) * 1e-6); +} + +/* get the current master clock value */ +static double get_master_clock(VideoState *is) +{ + double val; + + if (is->av_sync_type == AV_SYNC_VIDEO_MASTER) { + if (is->video_st) + val = get_video_clock(is); + else + val = get_audio_clock(is); + } else if (is->av_sync_type == AV_SYNC_AUDIO_MASTER) { + if (is->audio_st) + val = get_audio_clock(is); + else + val = get_video_clock(is); + } else { + val = get_external_clock(is); + } + return val; +} + +/* seek in the stream */ +static void stream_seek(VideoState *is, int64_t pos, int64_t rel, int seek_by_bytes) +{ + if (!is->seek_req) { + is->seek_pos = pos; + is->seek_rel = rel; + is->seek_flags &= ~AVSEEK_FLAG_BYTE; + if (seek_by_bytes) + is->seek_flags |= AVSEEK_FLAG_BYTE; + is->seek_req = 1; + } +} + +/* pause or resume the video */ +static void stream_pause(VideoState *is) +{ + if (is->paused) { + is->frame_timer += av_gettime() / 1000000.0 + is->video_current_pts_drift - is->video_current_pts; + if (is->read_pause_return != AVERROR(ENOSYS)) { + is->video_current_pts = is->video_current_pts_drift + av_gettime() / 1000000.0; + } + is->video_current_pts_drift = is->video_current_pts - av_gettime() / 1000000.0; + } + is->paused = !is->paused; +} + +static double compute_target_time(double frame_current_pts, VideoState *is) +{ + double delay, sync_threshold, diff; + + /* compute nominal delay */ + delay = frame_current_pts - is->frame_last_pts; + if (delay <= 0 || delay >= 10.0) { + /* if incorrect delay, use previous one */ + delay = is->frame_last_delay; + } else { + is->frame_last_delay = delay; + } + is->frame_last_pts = frame_current_pts; + + /* update delay to follow master synchronisation source */ + if (((is->av_sync_type == AV_SYNC_AUDIO_MASTER && is->audio_st) || + is->av_sync_type == AV_SYNC_EXTERNAL_CLOCK)) { + /* if video is slave, we try to correct big delays by + duplicating or deleting a frame */ + diff = get_video_clock(is) - get_master_clock(is); + + /* skip or repeat frame. We take into account the + delay to compute the threshold. I still don't know + if it is the best guess */ + sync_threshold = FFMAX(AV_SYNC_THRESHOLD, delay); + if (fabs(diff) < AV_NOSYNC_THRESHOLD) { + if (diff <= -sync_threshold) + delay = 0; + else if (diff >= sync_threshold) + delay = 2 * delay; + } + } + is->frame_timer += delay; + + av_dlog(NULL, "video: delay=%0.3f pts=%0.3f A-V=%f\n", + delay, frame_current_pts, -diff); + + return is->frame_timer; +} + +/* called to display each frame */ +static void video_refresh_timer(void *opaque) +{ + VideoState *is = opaque; + VideoPicture *vp; + + SubPicture *sp, *sp2; + + if (is->video_st) { +retry: + if (is->pictq_size == 0) { + // nothing to do, no picture to display in the que + } else { + double time = av_gettime() / 1000000.0; + double next_target; + /* dequeue the picture */ + vp = &is->pictq[is->pictq_rindex]; + + if (time < vp->target_clock) + return; + /* update current video pts */ + is->video_current_pts = vp->pts; + is->video_current_pts_drift = is->video_current_pts - time; + is->video_current_pos = vp->pos; + if (is->pictq_size > 1) { + VideoPicture *nextvp = &is->pictq[(is->pictq_rindex + 1) % VIDEO_PICTURE_QUEUE_SIZE]; + assert(nextvp->target_clock >= vp->target_clock); + next_target= nextvp->target_clock; + } else { + next_target = vp->target_clock + is->video_clock - vp->pts; // FIXME pass durations cleanly + } + if (framedrop && time > next_target) { + is->skip_frames *= 1.0 + FRAME_SKIP_FACTOR; + if (is->pictq_size > 1 || time > next_target + 0.5) { + /* update queue size and signal for next picture */ + if (++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE) + is->pictq_rindex = 0; + + SDL_LockMutex(is->pictq_mutex); + is->pictq_size--; + SDL_CondSignal(is->pictq_cond); + SDL_UnlockMutex(is->pictq_mutex); + goto retry; + } + } + + if (is->subtitle_st) { + if (is->subtitle_stream_changed) { + SDL_LockMutex(is->subpq_mutex); + + while (is->subpq_size) { + free_subpicture(&is->subpq[is->subpq_rindex]); + + /* update queue size and signal for next picture */ + if (++is->subpq_rindex == SUBPICTURE_QUEUE_SIZE) + is->subpq_rindex = 0; + + is->subpq_size--; + } + is->subtitle_stream_changed = 0; + + SDL_CondSignal(is->subpq_cond); + SDL_UnlockMutex(is->subpq_mutex); + } else { + if (is->subpq_size > 0) { + sp = &is->subpq[is->subpq_rindex]; + + if (is->subpq_size > 1) + sp2 = &is->subpq[(is->subpq_rindex + 1) % SUBPICTURE_QUEUE_SIZE]; + else + sp2 = NULL; + + if ((is->video_current_pts > (sp->pts + ((float) sp->sub.end_display_time / 1000))) + || (sp2 && is->video_current_pts > (sp2->pts + ((float) sp2->sub.start_display_time / 1000)))) + { + free_subpicture(sp); + + /* update queue size and signal for next picture */ + if (++is->subpq_rindex == SUBPICTURE_QUEUE_SIZE) + is->subpq_rindex = 0; + + SDL_LockMutex(is->subpq_mutex); + is->subpq_size--; + SDL_CondSignal(is->subpq_cond); + SDL_UnlockMutex(is->subpq_mutex); + } + } + } + } + + /* display picture */ + if (!display_disable) + video_display(is); + + /* update queue size and signal for next picture */ + if (++is->pictq_rindex == VIDEO_PICTURE_QUEUE_SIZE) + is->pictq_rindex = 0; + + SDL_LockMutex(is->pictq_mutex); + is->pictq_size--; + SDL_CondSignal(is->pictq_cond); + SDL_UnlockMutex(is->pictq_mutex); + } + } else if (is->audio_st) { + /* draw the next audio frame */ + + /* if only audio stream, then display the audio bars (better + than nothing, just to test the implementation */ + + /* display picture */ + if (!display_disable) + video_display(is); + } + if (show_status) { + static int64_t last_time; + int64_t cur_time; + int aqsize, vqsize, sqsize; + double av_diff; + + cur_time = av_gettime(); + if (!last_time || (cur_time - last_time) >= 30000) { + aqsize = 0; + vqsize = 0; + sqsize = 0; + if (is->audio_st) + aqsize = is->audioq.size; + if (is->video_st) + vqsize = is->videoq.size; + if (is->subtitle_st) + sqsize = is->subtitleq.size; + av_diff = 0; + if (is->audio_st && is->video_st) + av_diff = get_audio_clock(is) - get_video_clock(is); + printf("%7.2f A-V:%7.3f s:%3.1f aq=%5dKB vq=%5dKB sq=%5dB f=%"PRId64"/%"PRId64" \r", + get_master_clock(is), av_diff, FFMAX(is->skip_frames - 1, 0), aqsize / 1024, + vqsize / 1024, sqsize, is->pts_ctx.num_faulty_dts, is->pts_ctx.num_faulty_pts); + fflush(stdout); + last_time = cur_time; + } + } +} + +static void stream_close(VideoState *is) +{ + VideoPicture *vp; + int i; + /* XXX: use a special url_shutdown call to abort parse cleanly */ + is->abort_request = 1; + SDL_WaitThread(is->parse_tid, NULL); + SDL_WaitThread(is->refresh_tid, NULL); + + /* free all pictures */ + for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++) { + vp = &is->pictq[i]; + if (vp->bmp) { + SDL_FreeYUVOverlay(vp->bmp); + vp->bmp = NULL; + } + } + SDL_DestroyMutex(is->pictq_mutex); + SDL_DestroyCond(is->pictq_cond); + SDL_DestroyMutex(is->subpq_mutex); + SDL_DestroyCond(is->subpq_cond); +#if !CONFIG_AVFILTER + if (is->img_convert_ctx) + sws_freeContext(is->img_convert_ctx); +#endif + av_free(is); +} + +static void do_exit(void) +{ + if (cur_stream) { + stream_close(cur_stream); + cur_stream = NULL; + } + uninit_opts(); + avformat_network_deinit(); + if (show_status) + printf("\n"); + SDL_Quit(); + av_log(NULL, AV_LOG_QUIET, ""); + exit(0); +} + +/* allocate a picture (needs to do that in main thread to avoid + potential locking problems */ +static void alloc_picture(void *opaque) +{ + VideoState *is = opaque; + VideoPicture *vp; + + vp = &is->pictq[is->pictq_windex]; + + if (vp->bmp) + SDL_FreeYUVOverlay(vp->bmp); + +#if CONFIG_AVFILTER + vp->width = is->out_video_filter->inputs[0]->w; + vp->height = is->out_video_filter->inputs[0]->h; + vp->pix_fmt = is->out_video_filter->inputs[0]->format; +#else + vp->width = is->video_st->codec->width; + vp->height = is->video_st->codec->height; + vp->pix_fmt = is->video_st->codec->pix_fmt; +#endif + + vp->bmp = SDL_CreateYUVOverlay(vp->width, vp->height, + SDL_YV12_OVERLAY, + screen); + if (!vp->bmp || vp->bmp->pitches[0] < vp->width) { + /* SDL allocates a buffer smaller than requested if the video + * overlay hardware is unable to support the requested size. */ + fprintf(stderr, "Error: the video system does not support an image\n" + "size of %dx%d pixels. Try using -vf \"scale=w:h\"\n" + "to reduce the image size.\n", vp->width, vp->height ); + do_exit(); + } + + SDL_LockMutex(is->pictq_mutex); + vp->allocated = 1; + SDL_CondSignal(is->pictq_cond); + SDL_UnlockMutex(is->pictq_mutex); +} + +/* The 'pts' parameter is the dts of the packet / pts of the frame and + * guessed if not known. */ +static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, int64_t pos) +{ + VideoPicture *vp; +#if CONFIG_AVFILTER + AVPicture pict_src; +#else + int dst_pix_fmt = AV_PIX_FMT_YUV420P; +#endif + /* wait until we have space to put a new picture */ + SDL_LockMutex(is->pictq_mutex); + + if (is->pictq_size >= VIDEO_PICTURE_QUEUE_SIZE && !is->refresh) + is->skip_frames = FFMAX(1.0 - FRAME_SKIP_FACTOR, is->skip_frames * (1.0 - FRAME_SKIP_FACTOR)); + + while (is->pictq_size >= VIDEO_PICTURE_QUEUE_SIZE && + !is->videoq.abort_request) { + SDL_CondWait(is->pictq_cond, is->pictq_mutex); + } + SDL_UnlockMutex(is->pictq_mutex); + + if (is->videoq.abort_request) + return -1; + + vp = &is->pictq[is->pictq_windex]; + + /* alloc or resize hardware picture buffer */ + if (!vp->bmp || vp->reallocate || +#if CONFIG_AVFILTER + vp->width != is->out_video_filter->inputs[0]->w || + vp->height != is->out_video_filter->inputs[0]->h) { +#else + vp->width != is->video_st->codec->width || + vp->height != is->video_st->codec->height) { +#endif + SDL_Event event; + + vp->allocated = 0; + vp->reallocate = 0; + + /* the allocation must be done in the main thread to avoid + locking problems */ + event.type = FF_ALLOC_EVENT; + event.user.data1 = is; + SDL_PushEvent(&event); + + /* wait until the picture is allocated */ + SDL_LockMutex(is->pictq_mutex); + while (!vp->allocated && !is->videoq.abort_request) { + SDL_CondWait(is->pictq_cond, is->pictq_mutex); + } + SDL_UnlockMutex(is->pictq_mutex); + + if (is->videoq.abort_request) + return -1; + } + + /* if the frame is not skipped, then display it */ + if (vp->bmp) { + AVPicture pict = { { 0 } }; + + /* get a pointer on the bitmap */ + SDL_LockYUVOverlay (vp->bmp); + + pict.data[0] = vp->bmp->pixels[0]; + pict.data[1] = vp->bmp->pixels[2]; + pict.data[2] = vp->bmp->pixels[1]; + + pict.linesize[0] = vp->bmp->pitches[0]; + pict.linesize[1] = vp->bmp->pitches[2]; + pict.linesize[2] = vp->bmp->pitches[1]; + +#if CONFIG_AVFILTER + pict_src.data[0] = src_frame->data[0]; + pict_src.data[1] = src_frame->data[1]; + pict_src.data[2] = src_frame->data[2]; + + pict_src.linesize[0] = src_frame->linesize[0]; + pict_src.linesize[1] = src_frame->linesize[1]; + pict_src.linesize[2] = src_frame->linesize[2]; + + // FIXME use direct rendering + av_picture_copy(&pict, &pict_src, + vp->pix_fmt, vp->width, vp->height); +#else + av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags); + is->img_convert_ctx = sws_getCachedContext(is->img_convert_ctx, + vp->width, vp->height, vp->pix_fmt, vp->width, vp->height, + dst_pix_fmt, sws_flags, NULL, NULL, NULL); + if (is->img_convert_ctx == NULL) { + fprintf(stderr, "Cannot initialize the conversion context\n"); + exit(1); + } + sws_scale(is->img_convert_ctx, src_frame->data, src_frame->linesize, + 0, vp->height, pict.data, pict.linesize); +#endif + /* update the bitmap content */ + SDL_UnlockYUVOverlay(vp->bmp); + + vp->pts = pts; + vp->pos = pos; + + /* now we can update the picture count */ + if (++is->pictq_windex == VIDEO_PICTURE_QUEUE_SIZE) + is->pictq_windex = 0; + SDL_LockMutex(is->pictq_mutex); + vp->target_clock = compute_target_time(vp->pts, is); + + is->pictq_size++; + SDL_UnlockMutex(is->pictq_mutex); + } + return 0; +} + +/* Compute the exact PTS for the picture if it is omitted in the stream. + * The 'pts1' parameter is the dts of the packet / pts of the frame. */ +static int output_picture2(VideoState *is, AVFrame *src_frame, double pts1, int64_t pos) +{ + double frame_delay, pts; + int ret; + + pts = pts1; + + if (pts != 0) { + /* update video clock with pts, if present */ + is->video_clock = pts; + } else { + pts = is->video_clock; + } + /* update video clock for next frame */ + frame_delay = av_q2d(is->video_st->codec->time_base); + /* for MPEG2, the frame can be repeated, so we update the + clock accordingly */ + frame_delay += src_frame->repeat_pict * (frame_delay * 0.5); + is->video_clock += frame_delay; + + ret = queue_picture(is, src_frame, pts, pos); + av_frame_unref(src_frame); + return ret; +} + +static int get_video_frame(VideoState *is, AVFrame *frame, int64_t *pts, AVPacket *pkt) +{ + int got_picture, i; + + if (packet_queue_get(&is->videoq, pkt, 1) < 0) + return -1; + + if (pkt->data == flush_pkt.data) { + avcodec_flush_buffers(is->video_st->codec); + + SDL_LockMutex(is->pictq_mutex); + // Make sure there are no long delay timers (ideally we should just flush the que but thats harder) + for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++) { + is->pictq[i].target_clock= 0; + } + while (is->pictq_size && !is->videoq.abort_request) { + SDL_CondWait(is->pictq_cond, is->pictq_mutex); + } + is->video_current_pos = -1; + SDL_UnlockMutex(is->pictq_mutex); + + init_pts_correction(&is->pts_ctx); + is->frame_last_pts = AV_NOPTS_VALUE; + is->frame_last_delay = 0; + is->frame_timer = (double)av_gettime() / 1000000.0; + is->skip_frames = 1; + is->skip_frames_index = 0; + return 0; + } + + avcodec_decode_video2(is->video_st->codec, frame, &got_picture, pkt); + + if (got_picture) { + if (decoder_reorder_pts == -1) { + *pts = guess_correct_pts(&is->pts_ctx, frame->pkt_pts, frame->pkt_dts); + } else if (decoder_reorder_pts) { + *pts = frame->pkt_pts; + } else { + *pts = frame->pkt_dts; + } + + if (*pts == AV_NOPTS_VALUE) { + *pts = 0; + } + if (is->video_st->sample_aspect_ratio.num) { + frame->sample_aspect_ratio = is->video_st->sample_aspect_ratio; + } + + is->skip_frames_index += 1; + if (is->skip_frames_index >= is->skip_frames) { + is->skip_frames_index -= FFMAX(is->skip_frames, 1.0); + return 1; + } + av_frame_unref(frame); + } + return 0; +} + +#if CONFIG_AVFILTER +static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const char *vfilters) +{ + char sws_flags_str[128]; + char buffersrc_args[256]; + int ret; + AVFilterContext *filt_src = NULL, *filt_out = NULL, *filt_format; + AVCodecContext *codec = is->video_st->codec; + + snprintf(sws_flags_str, sizeof(sws_flags_str), "flags=%"PRId64, sws_flags); + graph->scale_sws_opts = av_strdup(sws_flags_str); + + snprintf(buffersrc_args, sizeof(buffersrc_args), "%d:%d:%d:%d:%d:%d:%d", + codec->width, codec->height, codec->pix_fmt, + is->video_st->time_base.num, is->video_st->time_base.den, + codec->sample_aspect_ratio.num, codec->sample_aspect_ratio.den); + + + if ((ret = avfilter_graph_create_filter(&filt_src, + avfilter_get_by_name("buffer"), + "src", buffersrc_args, NULL, + graph)) < 0) + return ret; + if ((ret = avfilter_graph_create_filter(&filt_out, + avfilter_get_by_name("buffersink"), + "out", NULL, NULL, graph)) < 0) + return ret; + + if ((ret = avfilter_graph_create_filter(&filt_format, + avfilter_get_by_name("format"), + "format", "yuv420p", NULL, graph)) < 0) + return ret; + if ((ret = avfilter_link(filt_format, 0, filt_out, 0)) < 0) + return ret; + + + if (vfilters) { + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + + outputs->name = av_strdup("in"); + outputs->filter_ctx = filt_src; + outputs->pad_idx = 0; + outputs->next = NULL; + + inputs->name = av_strdup("out"); + inputs->filter_ctx = filt_format; + inputs->pad_idx = 0; + inputs->next = NULL; + + if ((ret = avfilter_graph_parse(graph, vfilters, inputs, outputs, NULL)) < 0) + return ret; + } else { + if ((ret = avfilter_link(filt_src, 0, filt_format, 0)) < 0) + return ret; + } + + if ((ret = avfilter_graph_config(graph, NULL)) < 0) + return ret; + + is->in_video_filter = filt_src; + is->out_video_filter = filt_out; + + return ret; +} + +#endif /* CONFIG_AVFILTER */ + +static int video_thread(void *arg) +{ + AVPacket pkt = { 0 }; + VideoState *is = arg; + AVFrame *frame = av_frame_alloc(); + int64_t pts_int; + double pts; + int ret; + +#if CONFIG_AVFILTER + AVFilterGraph *graph = avfilter_graph_alloc(); + AVFilterContext *filt_out = NULL, *filt_in = NULL; + int last_w = is->video_st->codec->width; + int last_h = is->video_st->codec->height; + + if ((ret = configure_video_filters(graph, is, vfilters)) < 0) + goto the_end; + filt_in = is->in_video_filter; + filt_out = is->out_video_filter; +#endif + + for (;;) { +#if CONFIG_AVFILTER + AVRational tb; +#endif + while (is->paused && !is->videoq.abort_request) + SDL_Delay(10); + + av_free_packet(&pkt); + + ret = get_video_frame(is, frame, &pts_int, &pkt); + if (ret < 0) + goto the_end; + + if (!ret) + continue; + +#if CONFIG_AVFILTER + if ( last_w != is->video_st->codec->width + || last_h != is->video_st->codec->height) { + av_dlog(NULL, "Changing size %dx%d -> %dx%d\n", last_w, last_h, + is->video_st->codec->width, is->video_st->codec->height); + avfilter_graph_free(&graph); + graph = avfilter_graph_alloc(); + if ((ret = configure_video_filters(graph, is, vfilters)) < 0) + goto the_end; + filt_in = is->in_video_filter; + filt_out = is->out_video_filter; + last_w = is->video_st->codec->width; + last_h = is->video_st->codec->height; + } + + frame->pts = pts_int; + ret = av_buffersrc_add_frame(filt_in, frame); + if (ret < 0) + goto the_end; + + while (ret >= 0) { + ret = av_buffersink_get_frame(filt_out, frame); + if (ret < 0) { + ret = 0; + break; + } + + pts_int = frame->pts; + tb = filt_out->inputs[0]->time_base; + if (av_cmp_q(tb, is->video_st->time_base)) { + av_unused int64_t pts1 = pts_int; + pts_int = av_rescale_q(pts_int, tb, is->video_st->time_base); + av_dlog(NULL, "video_thread(): " + "tb:%d/%d pts:%"PRId64" -> tb:%d/%d pts:%"PRId64"\n", + tb.num, tb.den, pts1, + is->video_st->time_base.num, is->video_st->time_base.den, pts_int); + } + pts = pts_int * av_q2d(is->video_st->time_base); + ret = output_picture2(is, frame, pts, 0); + } +#else + pts = pts_int * av_q2d(is->video_st->time_base); + ret = output_picture2(is, frame, pts, pkt.pos); +#endif + + if (ret < 0) + goto the_end; + + + if (step) + if (cur_stream) + stream_pause(cur_stream); + } + the_end: +#if CONFIG_AVFILTER + av_freep(&vfilters); + avfilter_graph_free(&graph); +#endif + av_free_packet(&pkt); + av_frame_free(&frame); + return 0; +} + +static int subtitle_thread(void *arg) +{ + VideoState *is = arg; + SubPicture *sp; + AVPacket pkt1, *pkt = &pkt1; + int got_subtitle; + double pts; + int i, j; + int r, g, b, y, u, v, a; + + for (;;) { + while (is->paused && !is->subtitleq.abort_request) { + SDL_Delay(10); + } + if (packet_queue_get(&is->subtitleq, pkt, 1) < 0) + break; + + if (pkt->data == flush_pkt.data) { + avcodec_flush_buffers(is->subtitle_st->codec); + continue; + } + SDL_LockMutex(is->subpq_mutex); + while (is->subpq_size >= SUBPICTURE_QUEUE_SIZE && + !is->subtitleq.abort_request) { + SDL_CondWait(is->subpq_cond, is->subpq_mutex); + } + SDL_UnlockMutex(is->subpq_mutex); + + if (is->subtitleq.abort_request) + return 0; + + sp = &is->subpq[is->subpq_windex]; + + /* NOTE: ipts is the PTS of the _first_ picture beginning in + this packet, if any */ + pts = 0; + if (pkt->pts != AV_NOPTS_VALUE) + pts = av_q2d(is->subtitle_st->time_base) * pkt->pts; + + avcodec_decode_subtitle2(is->subtitle_st->codec, &sp->sub, + &got_subtitle, pkt); + + if (got_subtitle && sp->sub.format == 0) { + sp->pts = pts; + + for (i = 0; i < sp->sub.num_rects; i++) + { + for (j = 0; j < sp->sub.rects[i]->nb_colors; j++) + { + RGBA_IN(r, g, b, a, (uint32_t*)sp->sub.rects[i]->pict.data[1] + j); + y = RGB_TO_Y_CCIR(r, g, b); + u = RGB_TO_U_CCIR(r, g, b, 0); + v = RGB_TO_V_CCIR(r, g, b, 0); + YUVA_OUT((uint32_t*)sp->sub.rects[i]->pict.data[1] + j, y, u, v, a); + } + } + + /* now we can update the picture count */ + if (++is->subpq_windex == SUBPICTURE_QUEUE_SIZE) + is->subpq_windex = 0; + SDL_LockMutex(is->subpq_mutex); + is->subpq_size++; + SDL_UnlockMutex(is->subpq_mutex); + } + av_free_packet(pkt); + } + return 0; +} + +/* copy samples for viewing in editor window */ +static void update_sample_display(VideoState *is, short *samples, int samples_size) +{ + int size, len; + + size = samples_size / sizeof(short); + while (size > 0) { + len = SAMPLE_ARRAY_SIZE - is->sample_array_index; + if (len > size) + len = size; + memcpy(is->sample_array + is->sample_array_index, samples, len * sizeof(short)); + samples += len; + is->sample_array_index += len; + if (is->sample_array_index >= SAMPLE_ARRAY_SIZE) + is->sample_array_index = 0; + size -= len; + } +} + +/* return the new audio buffer size (samples can be added or deleted + to get better sync if video or external master clock) */ +static int synchronize_audio(VideoState *is, short *samples, + int samples_size1, double pts) +{ + int n, samples_size; + double ref_clock; + + n = is->sdl_channels * av_get_bytes_per_sample(is->sdl_sample_fmt); + samples_size = samples_size1; + + /* if not master, then we try to remove or add samples to correct the clock */ + if (((is->av_sync_type == AV_SYNC_VIDEO_MASTER && is->video_st) || + is->av_sync_type == AV_SYNC_EXTERNAL_CLOCK)) { + double diff, avg_diff; + int wanted_size, min_size, max_size, nb_samples; + + ref_clock = get_master_clock(is); + diff = get_audio_clock(is) - ref_clock; + + if (diff < AV_NOSYNC_THRESHOLD) { + is->audio_diff_cum = diff + is->audio_diff_avg_coef * is->audio_diff_cum; + if (is->audio_diff_avg_count < AUDIO_DIFF_AVG_NB) { + /* not enough measures to have a correct estimate */ + is->audio_diff_avg_count++; + } else { + /* estimate the A-V difference */ + avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef); + + if (fabs(avg_diff) >= is->audio_diff_threshold) { + wanted_size = samples_size + ((int)(diff * is->sdl_sample_rate) * n); + nb_samples = samples_size / n; + + min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n; + max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n; + if (wanted_size < min_size) + wanted_size = min_size; + else if (wanted_size > max_size) + wanted_size = max_size; + + /* add or remove samples to correction the synchro */ + if (wanted_size < samples_size) { + /* remove samples */ + samples_size = wanted_size; + } else if (wanted_size > samples_size) { + uint8_t *samples_end, *q; + int nb; + + /* add samples */ + nb = (samples_size - wanted_size); + samples_end = (uint8_t *)samples + samples_size - n; + q = samples_end + n; + while (nb > 0) { + memcpy(q, samples_end, n); + q += n; + nb -= n; + } + samples_size = wanted_size; + } + } + av_dlog(NULL, "diff=%f adiff=%f sample_diff=%d apts=%0.3f vpts=%0.3f %f\n", + diff, avg_diff, samples_size - samples_size1, + is->audio_clock, is->video_clock, is->audio_diff_threshold); + } + } else { + /* too big difference : may be initial PTS errors, so + reset A-V filter */ + is->audio_diff_avg_count = 0; + is->audio_diff_cum = 0; + } + } + + return samples_size; +} + +/* decode one audio frame and returns its uncompressed size */ +static int audio_decode_frame(VideoState *is, double *pts_ptr) +{ + AVPacket *pkt_temp = &is->audio_pkt_temp; + AVPacket *pkt = &is->audio_pkt; + AVCodecContext *dec = is->audio_st->codec; + int n, len1, data_size, got_frame; + double pts; + int new_packet = 0; + int flush_complete = 0; + + for (;;) { + /* NOTE: the audio packet can contain several frames */ + while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) { + int resample_changed, audio_resample; + + if (!is->frame) { + if (!(is->frame = av_frame_alloc())) + return AVERROR(ENOMEM); + } + + if (flush_complete) + break; + new_packet = 0; + len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp); + if (len1 < 0) { + /* if error, we skip the frame */ + pkt_temp->size = 0; + break; + } + + pkt_temp->data += len1; + pkt_temp->size -= len1; + + if (!got_frame) { + /* stop sending empty packets if the decoder is finished */ + if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY) + flush_complete = 1; + continue; + } + data_size = av_samples_get_buffer_size(NULL, dec->channels, + is->frame->nb_samples, + is->frame->format, 1); + + audio_resample = is->frame->format != is->sdl_sample_fmt || + is->frame->channel_layout != is->sdl_channel_layout || + is->frame->sample_rate != is->sdl_sample_rate; + + resample_changed = is->frame->format != is->resample_sample_fmt || + is->frame->channel_layout != is->resample_channel_layout || + is->frame->sample_rate != is->resample_sample_rate; + + if ((!is->avr && audio_resample) || resample_changed) { + int ret; + if (is->avr) + avresample_close(is->avr); + else if (audio_resample) { + is->avr = avresample_alloc_context(); + if (!is->avr) { + fprintf(stderr, "error allocating AVAudioResampleContext\n"); + break; + } + } + if (audio_resample) { + av_opt_set_int(is->avr, "in_channel_layout", is->frame->channel_layout, 0); + av_opt_set_int(is->avr, "in_sample_fmt", is->frame->format, 0); + av_opt_set_int(is->avr, "in_sample_rate", is->frame->sample_rate, 0); + av_opt_set_int(is->avr, "out_channel_layout", is->sdl_channel_layout, 0); + av_opt_set_int(is->avr, "out_sample_fmt", is->sdl_sample_fmt, 0); + av_opt_set_int(is->avr, "out_sample_rate", is->sdl_sample_rate, 0); + + if ((ret = avresample_open(is->avr)) < 0) { + fprintf(stderr, "error initializing libavresample\n"); + break; + } + } + is->resample_sample_fmt = is->frame->format; + is->resample_channel_layout = is->frame->channel_layout; + is->resample_sample_rate = is->frame->sample_rate; + } + + if (audio_resample) { + void *tmp_out; + int out_samples, out_size, out_linesize; + int osize = av_get_bytes_per_sample(is->sdl_sample_fmt); + int nb_samples = is->frame->nb_samples; + + out_size = av_samples_get_buffer_size(&out_linesize, + is->sdl_channels, + nb_samples, + is->sdl_sample_fmt, 0); + tmp_out = av_realloc(is->audio_buf1, out_size); + if (!tmp_out) + return AVERROR(ENOMEM); + is->audio_buf1 = tmp_out; + + out_samples = avresample_convert(is->avr, + &is->audio_buf1, + out_linesize, nb_samples, + is->frame->data, + is->frame->linesize[0], + is->frame->nb_samples); + if (out_samples < 0) { + fprintf(stderr, "avresample_convert() failed\n"); + break; + } + is->audio_buf = is->audio_buf1; + data_size = out_samples * osize * is->sdl_channels; + } else { + is->audio_buf = is->frame->data[0]; + } + + /* if no pts, then compute it */ + pts = is->audio_clock; + *pts_ptr = pts; + n = is->sdl_channels * av_get_bytes_per_sample(is->sdl_sample_fmt); + is->audio_clock += (double)data_size / + (double)(n * is->sdl_sample_rate); +#ifdef DEBUG + { + static double last_clock; + printf("audio: delay=%0.3f clock=%0.3f pts=%0.3f\n", + is->audio_clock - last_clock, + is->audio_clock, pts); + last_clock = is->audio_clock; + } +#endif + return data_size; + } + + /* free the current packet */ + if (pkt->data) + av_free_packet(pkt); + memset(pkt_temp, 0, sizeof(*pkt_temp)); + + if (is->paused || is->audioq.abort_request) { + return -1; + } + + /* read next packet */ + if ((new_packet = packet_queue_get(&is->audioq, pkt, 1)) < 0) + return -1; + + if (pkt->data == flush_pkt.data) { + avcodec_flush_buffers(dec); + flush_complete = 0; + } + + *pkt_temp = *pkt; + + /* if update the audio clock with the pts */ + if (pkt->pts != AV_NOPTS_VALUE) { + is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts; + } + } +} + +/* prepare a new audio buffer */ +static void sdl_audio_callback(void *opaque, Uint8 *stream, int len) +{ + VideoState *is = opaque; + int audio_size, len1; + double pts; + + audio_callback_time = av_gettime(); + + while (len > 0) { + if (is->audio_buf_index >= is->audio_buf_size) { + audio_size = audio_decode_frame(is, &pts); + if (audio_size < 0) { + /* if error, just output silence */ + is->audio_buf = is->silence_buf; + is->audio_buf_size = sizeof(is->silence_buf); + } else { + if (is->show_audio) + update_sample_display(is, (int16_t *)is->audio_buf, audio_size); + audio_size = synchronize_audio(is, (int16_t *)is->audio_buf, audio_size, + pts); + is->audio_buf_size = audio_size; + } + is->audio_buf_index = 0; + } + len1 = is->audio_buf_size - is->audio_buf_index; + if (len1 > len) + len1 = len; + memcpy(stream, (uint8_t *)is->audio_buf + is->audio_buf_index, len1); + len -= len1; + stream += len1; + is->audio_buf_index += len1; + } +} + +/* open a given stream. Return 0 if OK */ +static int stream_component_open(VideoState *is, int stream_index) +{ + AVFormatContext *ic = is->ic; + AVCodecContext *avctx; + AVCodec *codec; + SDL_AudioSpec wanted_spec, spec; + AVDictionary *opts; + AVDictionaryEntry *t = NULL; + + if (stream_index < 0 || stream_index >= ic->nb_streams) + return -1; + avctx = ic->streams[stream_index]->codec; + + opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index], NULL); + + codec = avcodec_find_decoder(avctx->codec_id); + avctx->workaround_bugs = workaround_bugs; + avctx->idct_algo = idct; + avctx->skip_frame = skip_frame; + avctx->skip_idct = skip_idct; + avctx->skip_loop_filter = skip_loop_filter; + avctx->error_concealment = error_concealment; + + if (fast) avctx->flags2 |= CODEC_FLAG2_FAST; + + if (!av_dict_get(opts, "threads", NULL, 0)) + av_dict_set(&opts, "threads", "auto", 0); + if (avctx->codec_type == AVMEDIA_TYPE_VIDEO) + av_dict_set(&opts, "refcounted_frames", "1", 0); + if (!codec || + avcodec_open2(avctx, codec, &opts) < 0) + return -1; + if ((t = av_dict_get(opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) { + av_log(NULL, AV_LOG_ERROR, "Option %s not found.\n", t->key); + return AVERROR_OPTION_NOT_FOUND; + } + + /* prepare audio output */ + if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) { + is->sdl_sample_rate = avctx->sample_rate; + + if (!avctx->channel_layout) + avctx->channel_layout = av_get_default_channel_layout(avctx->channels); + if (!avctx->channel_layout) { + fprintf(stderr, "unable to guess channel layout\n"); + return -1; + } + if (avctx->channels == 1) + is->sdl_channel_layout = AV_CH_LAYOUT_MONO; + else + is->sdl_channel_layout = AV_CH_LAYOUT_STEREO; + is->sdl_channels = av_get_channel_layout_nb_channels(is->sdl_channel_layout); + + wanted_spec.format = AUDIO_S16SYS; + wanted_spec.freq = is->sdl_sample_rate; + wanted_spec.channels = is->sdl_channels; + wanted_spec.silence = 0; + wanted_spec.samples = SDL_AUDIO_BUFFER_SIZE; + wanted_spec.callback = sdl_audio_callback; + wanted_spec.userdata = is; + if (SDL_OpenAudio(&wanted_spec, &spec) < 0) { + fprintf(stderr, "SDL_OpenAudio: %s\n", SDL_GetError()); + return -1; + } + is->audio_hw_buf_size = spec.size; + is->sdl_sample_fmt = AV_SAMPLE_FMT_S16; + is->resample_sample_fmt = is->sdl_sample_fmt; + is->resample_channel_layout = avctx->channel_layout; + is->resample_sample_rate = avctx->sample_rate; + } + + ic->streams[stream_index]->discard = AVDISCARD_DEFAULT; + switch (avctx->codec_type) { + case AVMEDIA_TYPE_AUDIO: + is->audio_stream = stream_index; + is->audio_st = ic->streams[stream_index]; + is->audio_buf_size = 0; + is->audio_buf_index = 0; + + /* init averaging filter */ + is->audio_diff_avg_coef = exp(log(0.01) / AUDIO_DIFF_AVG_NB); + is->audio_diff_avg_count = 0; + /* since we do not have a precise anough audio fifo fullness, + we correct audio sync only if larger than this threshold */ + is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / avctx->sample_rate; + + memset(&is->audio_pkt, 0, sizeof(is->audio_pkt)); + packet_queue_init(&is->audioq); + SDL_PauseAudio(0); + break; + case AVMEDIA_TYPE_VIDEO: + is->video_stream = stream_index; + is->video_st = ic->streams[stream_index]; + + packet_queue_init(&is->videoq); + is->video_tid = SDL_CreateThread(video_thread, is); + break; + case AVMEDIA_TYPE_SUBTITLE: + is->subtitle_stream = stream_index; + is->subtitle_st = ic->streams[stream_index]; + packet_queue_init(&is->subtitleq); + + is->subtitle_tid = SDL_CreateThread(subtitle_thread, is); + break; + default: + break; + } + return 0; +} + +static void stream_component_close(VideoState *is, int stream_index) +{ + AVFormatContext *ic = is->ic; + AVCodecContext *avctx; + + if (stream_index < 0 || stream_index >= ic->nb_streams) + return; + avctx = ic->streams[stream_index]->codec; + + switch (avctx->codec_type) { + case AVMEDIA_TYPE_AUDIO: + packet_queue_abort(&is->audioq); + + SDL_CloseAudio(); + + packet_queue_end(&is->audioq); + av_free_packet(&is->audio_pkt); + if (is->avr) + avresample_free(&is->avr); + av_freep(&is->audio_buf1); + is->audio_buf = NULL; + av_frame_free(&is->frame); + + if (is->rdft) { + av_rdft_end(is->rdft); + av_freep(&is->rdft_data); + is->rdft = NULL; + is->rdft_bits = 0; + } + break; + case AVMEDIA_TYPE_VIDEO: + packet_queue_abort(&is->videoq); + + /* note: we also signal this mutex to make sure we deblock the + video thread in all cases */ + SDL_LockMutex(is->pictq_mutex); + SDL_CondSignal(is->pictq_cond); + SDL_UnlockMutex(is->pictq_mutex); + + SDL_WaitThread(is->video_tid, NULL); + + packet_queue_end(&is->videoq); + break; + case AVMEDIA_TYPE_SUBTITLE: + packet_queue_abort(&is->subtitleq); + + /* note: we also signal this mutex to make sure we deblock the + video thread in all cases */ + SDL_LockMutex(is->subpq_mutex); + is->subtitle_stream_changed = 1; + + SDL_CondSignal(is->subpq_cond); + SDL_UnlockMutex(is->subpq_mutex); + + SDL_WaitThread(is->subtitle_tid, NULL); + + packet_queue_end(&is->subtitleq); + break; + default: + break; + } + + ic->streams[stream_index]->discard = AVDISCARD_ALL; + avcodec_close(avctx); + switch (avctx->codec_type) { + case AVMEDIA_TYPE_AUDIO: + is->audio_st = NULL; + is->audio_stream = -1; + break; + case AVMEDIA_TYPE_VIDEO: + is->video_st = NULL; + is->video_stream = -1; + break; + case AVMEDIA_TYPE_SUBTITLE: + is->subtitle_st = NULL; + is->subtitle_stream = -1; + break; + default: + break; + } +} + +/* since we have only one decoding thread, we can use a global + variable instead of a thread local variable */ +static VideoState *global_video_state; + +static int decode_interrupt_cb(void *ctx) +{ + return global_video_state && global_video_state->abort_request; +} + +/* this thread gets the stream from the disk or the network */ +static int decode_thread(void *arg) +{ + VideoState *is = arg; + AVFormatContext *ic = NULL; + int err, i, ret; + int st_index[AVMEDIA_TYPE_NB]; + AVPacket pkt1, *pkt = &pkt1; + int eof = 0; + int pkt_in_play_range = 0; + AVDictionaryEntry *t; + AVDictionary **opts; + int orig_nb_streams; + + memset(st_index, -1, sizeof(st_index)); + is->video_stream = -1; + is->audio_stream = -1; + is->subtitle_stream = -1; + + global_video_state = is; + + ic = avformat_alloc_context(); + ic->interrupt_callback.callback = decode_interrupt_cb; + err = avformat_open_input(&ic, is->filename, is->iformat, &format_opts); + if (err < 0) { + print_error(is->filename, err); + ret = -1; + goto fail; + } + if ((t = av_dict_get(format_opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) { + av_log(NULL, AV_LOG_ERROR, "Option %s not found.\n", t->key); + ret = AVERROR_OPTION_NOT_FOUND; + goto fail; + } + is->ic = ic; + + if (genpts) + ic->flags |= AVFMT_FLAG_GENPTS; + + opts = setup_find_stream_info_opts(ic, codec_opts); + orig_nb_streams = ic->nb_streams; + + err = avformat_find_stream_info(ic, opts); + if (err < 0) { + fprintf(stderr, "%s: could not find codec parameters\n", is->filename); + ret = -1; + goto fail; + } + for (i = 0; i < orig_nb_streams; i++) + av_dict_free(&opts[i]); + av_freep(&opts); + + if (ic->pb) + ic->pb->eof_reached = 0; // FIXME hack, avplay maybe should not use url_feof() to test for the end + + if (seek_by_bytes < 0) + seek_by_bytes = !!(ic->iformat->flags & AVFMT_TS_DISCONT); + + /* if seeking requested, we execute it */ + if (start_time != AV_NOPTS_VALUE) { + int64_t timestamp; + + timestamp = start_time; + /* add the stream start time */ + if (ic->start_time != AV_NOPTS_VALUE) + timestamp += ic->start_time; + ret = avformat_seek_file(ic, -1, INT64_MIN, timestamp, INT64_MAX, 0); + if (ret < 0) { + fprintf(stderr, "%s: could not seek to position %0.3f\n", + is->filename, (double)timestamp / AV_TIME_BASE); + } + } + + for (i = 0; i < ic->nb_streams; i++) + ic->streams[i]->discard = AVDISCARD_ALL; + if (!video_disable) + st_index[AVMEDIA_TYPE_VIDEO] = + av_find_best_stream(ic, AVMEDIA_TYPE_VIDEO, + wanted_stream[AVMEDIA_TYPE_VIDEO], -1, NULL, 0); + if (!audio_disable) + st_index[AVMEDIA_TYPE_AUDIO] = + av_find_best_stream(ic, AVMEDIA_TYPE_AUDIO, + wanted_stream[AVMEDIA_TYPE_AUDIO], + st_index[AVMEDIA_TYPE_VIDEO], + NULL, 0); + if (!video_disable) + st_index[AVMEDIA_TYPE_SUBTITLE] = + av_find_best_stream(ic, AVMEDIA_TYPE_SUBTITLE, + wanted_stream[AVMEDIA_TYPE_SUBTITLE], + (st_index[AVMEDIA_TYPE_AUDIO] >= 0 ? + st_index[AVMEDIA_TYPE_AUDIO] : + st_index[AVMEDIA_TYPE_VIDEO]), + NULL, 0); + if (show_status) { + av_dump_format(ic, 0, is->filename, 0); + } + + /* open the streams */ + if (st_index[AVMEDIA_TYPE_AUDIO] >= 0) { + stream_component_open(is, st_index[AVMEDIA_TYPE_AUDIO]); + } + + ret = -1; + if (st_index[AVMEDIA_TYPE_VIDEO] >= 0) { + ret = stream_component_open(is, st_index[AVMEDIA_TYPE_VIDEO]); + } + is->refresh_tid = SDL_CreateThread(refresh_thread, is); + if (ret < 0) { + if (!display_disable) + is->show_audio = 2; + } + + if (st_index[AVMEDIA_TYPE_SUBTITLE] >= 0) { + stream_component_open(is, st_index[AVMEDIA_TYPE_SUBTITLE]); + } + + if (is->video_stream < 0 && is->audio_stream < 0) { + fprintf(stderr, "%s: could not open codecs\n", is->filename); + ret = -1; + goto fail; + } + + for (;;) { + if (is->abort_request) + break; + if (is->paused != is->last_paused) { + is->last_paused = is->paused; + if (is->paused) + is->read_pause_return = av_read_pause(ic); + else + av_read_play(ic); + } +#if CONFIG_RTSP_DEMUXER + if (is->paused && !strcmp(ic->iformat->name, "rtsp")) { + /* wait 10 ms to avoid trying to get another packet */ + /* XXX: horrible */ + SDL_Delay(10); + continue; + } +#endif + if (is->seek_req) { + int64_t seek_target = is->seek_pos; + int64_t seek_min = is->seek_rel > 0 ? seek_target - is->seek_rel + 2: INT64_MIN; + int64_t seek_max = is->seek_rel < 0 ? seek_target - is->seek_rel - 2: INT64_MAX; +// FIXME the +-2 is due to rounding being not done in the correct direction in generation +// of the seek_pos/seek_rel variables + + ret = avformat_seek_file(is->ic, -1, seek_min, seek_target, seek_max, is->seek_flags); + if (ret < 0) { + fprintf(stderr, "%s: error while seeking\n", is->ic->filename); + } else { + if (is->audio_stream >= 0) { + packet_queue_flush(&is->audioq); + packet_queue_put(&is->audioq, &flush_pkt); + } + if (is->subtitle_stream >= 0) { + packet_queue_flush(&is->subtitleq); + packet_queue_put(&is->subtitleq, &flush_pkt); + } + if (is->video_stream >= 0) { + packet_queue_flush(&is->videoq); + packet_queue_put(&is->videoq, &flush_pkt); + } + } + is->seek_req = 0; + eof = 0; + } + + /* if the queue are full, no need to read more */ + if (!infinite_buffer && + (is->audioq.size + is->videoq.size + is->subtitleq.size > MAX_QUEUE_SIZE + || ( (is->audioq .size > MIN_AUDIOQ_SIZE || is->audio_stream < 0) + && (is->videoq .nb_packets > MIN_FRAMES || is->video_stream < 0) + && (is->subtitleq.nb_packets > MIN_FRAMES || is->subtitle_stream < 0)))) { + /* wait 10 ms */ + SDL_Delay(10); + continue; + } + if (eof) { + if (is->video_stream >= 0) { + av_init_packet(pkt); + pkt->data = NULL; + pkt->size = 0; + pkt->stream_index = is->video_stream; + packet_queue_put(&is->videoq, pkt); + } + if (is->audio_stream >= 0 && + is->audio_st->codec->codec->capabilities & CODEC_CAP_DELAY) { + av_init_packet(pkt); + pkt->data = NULL; + pkt->size = 0; + pkt->stream_index = is->audio_stream; + packet_queue_put(&is->audioq, pkt); + } + SDL_Delay(10); + if (is->audioq.size + is->videoq.size + is->subtitleq.size == 0) { + if (loop != 1 && (!loop || --loop)) { + stream_seek(cur_stream, start_time != AV_NOPTS_VALUE ? start_time : 0, 0, 0); + } else if (autoexit) { + ret = AVERROR_EOF; + goto fail; + } + } + continue; + } + ret = av_read_frame(ic, pkt); + if (ret < 0) { + if (ret == AVERROR_EOF || (ic->pb && ic->pb->eof_reached)) + eof = 1; + if (ic->pb && ic->pb->error) + break; + SDL_Delay(100); /* wait for user event */ + continue; + } + /* check if packet is in play range specified by user, then queue, otherwise discard */ + pkt_in_play_range = duration == AV_NOPTS_VALUE || + (pkt->pts - ic->streams[pkt->stream_index]->start_time) * + av_q2d(ic->streams[pkt->stream_index]->time_base) - + (double)(start_time != AV_NOPTS_VALUE ? start_time : 0) / 1000000 + <= ((double)duration / 1000000); + if (pkt->stream_index == is->audio_stream && pkt_in_play_range) { + packet_queue_put(&is->audioq, pkt); + } else if (pkt->stream_index == is->video_stream && pkt_in_play_range) { + packet_queue_put(&is->videoq, pkt); + } else if (pkt->stream_index == is->subtitle_stream && pkt_in_play_range) { + packet_queue_put(&is->subtitleq, pkt); + } else { + av_free_packet(pkt); + } + } + /* wait until the end */ + while (!is->abort_request) { + SDL_Delay(100); + } + + ret = 0; + fail: + /* disable interrupting */ + global_video_state = NULL; + + /* close each stream */ + if (is->audio_stream >= 0) + stream_component_close(is, is->audio_stream); + if (is->video_stream >= 0) + stream_component_close(is, is->video_stream); + if (is->subtitle_stream >= 0) + stream_component_close(is, is->subtitle_stream); + if (is->ic) { + avformat_close_input(&is->ic); + } + + if (ret != 0) { + SDL_Event event; + + event.type = FF_QUIT_EVENT; + event.user.data1 = is; + SDL_PushEvent(&event); + } + return 0; +} + +static VideoState *stream_open(const char *filename, AVInputFormat *iformat) +{ + VideoState *is; + + is = av_mallocz(sizeof(VideoState)); + if (!is) + return NULL; + av_strlcpy(is->filename, filename, sizeof(is->filename)); + is->iformat = iformat; + is->ytop = 0; + is->xleft = 0; + + /* start video display */ + is->pictq_mutex = SDL_CreateMutex(); + is->pictq_cond = SDL_CreateCond(); + + is->subpq_mutex = SDL_CreateMutex(); + is->subpq_cond = SDL_CreateCond(); + + is->av_sync_type = av_sync_type; + is->parse_tid = SDL_CreateThread(decode_thread, is); + if (!is->parse_tid) { + av_free(is); + return NULL; + } + return is; +} + +static void stream_cycle_channel(VideoState *is, int codec_type) +{ + AVFormatContext *ic = is->ic; + int start_index, stream_index; + AVStream *st; + + if (codec_type == AVMEDIA_TYPE_VIDEO) + start_index = is->video_stream; + else if (codec_type == AVMEDIA_TYPE_AUDIO) + start_index = is->audio_stream; + else + start_index = is->subtitle_stream; + if (start_index < (codec_type == AVMEDIA_TYPE_SUBTITLE ? -1 : 0)) + return; + stream_index = start_index; + for (;;) { + if (++stream_index >= is->ic->nb_streams) + { + if (codec_type == AVMEDIA_TYPE_SUBTITLE) + { + stream_index = -1; + goto the_end; + } else + stream_index = 0; + } + if (stream_index == start_index) + return; + st = ic->streams[stream_index]; + if (st->codec->codec_type == codec_type) { + /* check that parameters are OK */ + switch (codec_type) { + case AVMEDIA_TYPE_AUDIO: + if (st->codec->sample_rate != 0 && + st->codec->channels != 0) + goto the_end; + break; + case AVMEDIA_TYPE_VIDEO: + case AVMEDIA_TYPE_SUBTITLE: + goto the_end; + default: + break; + } + } + } + the_end: + stream_component_close(is, start_index); + stream_component_open(is, stream_index); +} + + +static void toggle_full_screen(void) +{ +#if defined(__APPLE__) && SDL_VERSION_ATLEAST(1, 2, 14) + /* OS X needs to empty the picture_queue */ + int i; + for (i = 0; i < VIDEO_PICTURE_QUEUE_SIZE; i++) + cur_stream->pictq[i].reallocate = 1; +#endif + is_full_screen = !is_full_screen; + video_open(cur_stream); +} + +static void toggle_pause(void) +{ + if (cur_stream) + stream_pause(cur_stream); + step = 0; +} + +static void step_to_next_frame(void) +{ + if (cur_stream) { + /* if the stream is paused unpause it, then step */ + if (cur_stream->paused) + stream_pause(cur_stream); + } + step = 1; +} + +static void toggle_audio_display(void) +{ + if (cur_stream) { + int bgcolor = SDL_MapRGB(screen->format, 0x00, 0x00, 0x00); + cur_stream->show_audio = (cur_stream->show_audio + 1) % 3; + fill_rectangle(screen, + cur_stream->xleft, cur_stream->ytop, cur_stream->width, cur_stream->height, + bgcolor); + SDL_UpdateRect(screen, cur_stream->xleft, cur_stream->ytop, cur_stream->width, cur_stream->height); + } +} + +static void seek_chapter(VideoState *is, int incr) +{ + int64_t pos = get_master_clock(is) * AV_TIME_BASE; + int i; + + if (!is->ic->nb_chapters) + return; + + /* find the current chapter */ + for (i = 0; i < is->ic->nb_chapters; i++) { + AVChapter *ch = is->ic->chapters[i]; + if (av_compare_ts(pos, AV_TIME_BASE_Q, ch->start, ch->time_base) < 0) { + i--; + break; + } + } + + i += incr; + i = FFMAX(i, 0); + if (i >= is->ic->nb_chapters) + return; + + av_log(NULL, AV_LOG_VERBOSE, "Seeking to chapter %d.\n", i); + stream_seek(is, av_rescale_q(is->ic->chapters[i]->start, is->ic->chapters[i]->time_base, + AV_TIME_BASE_Q), 0, 0); +} + +/* handle an event sent by the GUI */ +static void event_loop(void) +{ + SDL_Event event; + double incr, pos, frac; + + for (;;) { + double x; + SDL_WaitEvent(&event); + switch (event.type) { + case SDL_KEYDOWN: + if (exit_on_keydown) { + do_exit(); + break; + } + switch (event.key.keysym.sym) { + case SDLK_ESCAPE: + case SDLK_q: + do_exit(); + break; + case SDLK_f: + toggle_full_screen(); + break; + case SDLK_p: + case SDLK_SPACE: + toggle_pause(); + break; + case SDLK_s: // S: Step to next frame + step_to_next_frame(); + break; + case SDLK_a: + if (cur_stream) + stream_cycle_channel(cur_stream, AVMEDIA_TYPE_AUDIO); + break; + case SDLK_v: + if (cur_stream) + stream_cycle_channel(cur_stream, AVMEDIA_TYPE_VIDEO); + break; + case SDLK_t: + if (cur_stream) + stream_cycle_channel(cur_stream, AVMEDIA_TYPE_SUBTITLE); + break; + case SDLK_w: + toggle_audio_display(); + break; + case SDLK_PAGEUP: + seek_chapter(cur_stream, 1); + break; + case SDLK_PAGEDOWN: + seek_chapter(cur_stream, -1); + break; + case SDLK_LEFT: + incr = -10.0; + goto do_seek; + case SDLK_RIGHT: + incr = 10.0; + goto do_seek; + case SDLK_UP: + incr = 60.0; + goto do_seek; + case SDLK_DOWN: + incr = -60.0; + do_seek: + if (cur_stream) { + if (seek_by_bytes) { + if (cur_stream->video_stream >= 0 && cur_stream->video_current_pos >= 0) { + pos = cur_stream->video_current_pos; + } else if (cur_stream->audio_stream >= 0 && cur_stream->audio_pkt.pos >= 0) { + pos = cur_stream->audio_pkt.pos; + } else + pos = avio_tell(cur_stream->ic->pb); + if (cur_stream->ic->bit_rate) + incr *= cur_stream->ic->bit_rate / 8.0; + else + incr *= 180000.0; + pos += incr; + stream_seek(cur_stream, pos, incr, 1); + } else { + pos = get_master_clock(cur_stream); + pos += incr; + stream_seek(cur_stream, (int64_t)(pos * AV_TIME_BASE), (int64_t)(incr * AV_TIME_BASE), 0); + } + } + break; + default: + break; + } + break; + case SDL_MOUSEBUTTONDOWN: + if (exit_on_mousedown) { + do_exit(); + break; + } + case SDL_MOUSEMOTION: + if (event.type == SDL_MOUSEBUTTONDOWN) { + x = event.button.x; + } else { + if (event.motion.state != SDL_PRESSED) + break; + x = event.motion.x; + } + if (cur_stream) { + if (seek_by_bytes || cur_stream->ic->duration <= 0) { + uint64_t size = avio_size(cur_stream->ic->pb); + stream_seek(cur_stream, size*x/cur_stream->width, 0, 1); + } else { + int64_t ts; + int ns, hh, mm, ss; + int tns, thh, tmm, tss; + tns = cur_stream->ic->duration / 1000000LL; + thh = tns / 3600; + tmm = (tns % 3600) / 60; + tss = (tns % 60); + frac = x / cur_stream->width; + ns = frac * tns; + hh = ns / 3600; + mm = (ns % 3600) / 60; + ss = (ns % 60); + fprintf(stderr, "Seek to %2.0f%% (%2d:%02d:%02d) of total duration (%2d:%02d:%02d) \n", frac*100, + hh, mm, ss, thh, tmm, tss); + ts = frac * cur_stream->ic->duration; + if (cur_stream->ic->start_time != AV_NOPTS_VALUE) + ts += cur_stream->ic->start_time; + stream_seek(cur_stream, ts, 0, 0); + } + } + break; + case SDL_VIDEORESIZE: + if (cur_stream) { + screen = SDL_SetVideoMode(event.resize.w, event.resize.h, 0, + SDL_HWSURFACE|SDL_RESIZABLE|SDL_ASYNCBLIT|SDL_HWACCEL); + screen_width = cur_stream->width = event.resize.w; + screen_height = cur_stream->height = event.resize.h; + } + break; + case SDL_QUIT: + case FF_QUIT_EVENT: + do_exit(); + break; + case FF_ALLOC_EVENT: + video_open(event.user.data1); + alloc_picture(event.user.data1); + break; + case FF_REFRESH_EVENT: + video_refresh_timer(event.user.data1); + cur_stream->refresh = 0; + break; + default: + break; + } + } +} + +static int opt_frame_size(void *optctx, const char *opt, const char *arg) +{ + av_log(NULL, AV_LOG_ERROR, + "Option '%s' has been removed, use private format options instead\n", opt); + return AVERROR(EINVAL); +} + +static int opt_width(void *optctx, const char *opt, const char *arg) +{ + screen_width = parse_number_or_die(opt, arg, OPT_INT64, 1, INT_MAX); + return 0; +} + +static int opt_height(void *optctx, const char *opt, const char *arg) +{ + screen_height = parse_number_or_die(opt, arg, OPT_INT64, 1, INT_MAX); + return 0; +} + +static int opt_format(void *optctx, const char *opt, const char *arg) +{ + file_iformat = av_find_input_format(arg); + if (!file_iformat) { + fprintf(stderr, "Unknown input format: %s\n", arg); + return AVERROR(EINVAL); + } + return 0; +} + +static int opt_frame_pix_fmt(void *optctx, const char *opt, const char *arg) +{ + av_log(NULL, AV_LOG_ERROR, + "Option '%s' has been removed, use private format options instead\n", opt); + return AVERROR(EINVAL); +} + +static int opt_sync(void *optctx, const char *opt, const char *arg) +{ + if (!strcmp(arg, "audio")) + av_sync_type = AV_SYNC_AUDIO_MASTER; + else if (!strcmp(arg, "video")) + av_sync_type = AV_SYNC_VIDEO_MASTER; + else if (!strcmp(arg, "ext")) + av_sync_type = AV_SYNC_EXTERNAL_CLOCK; + else { + fprintf(stderr, "Unknown value for %s: %s\n", opt, arg); + exit(1); + } + return 0; +} + +static int opt_seek(void *optctx, const char *opt, const char *arg) +{ + start_time = parse_time_or_die(opt, arg, 1); + return 0; +} + +static int opt_duration(void *optctx, const char *opt, const char *arg) +{ + duration = parse_time_or_die(opt, arg, 1); + return 0; +} + +static const OptionDef options[] = { +#include "cmdutils_common_opts.h" + { "x", HAS_ARG, { .func_arg = opt_width }, "force displayed width", "width" }, + { "y", HAS_ARG, { .func_arg = opt_height }, "force displayed height", "height" }, + { "s", HAS_ARG | OPT_VIDEO, { .func_arg = opt_frame_size }, "set frame size (WxH or abbreviation)", "size" }, + { "fs", OPT_BOOL, { &is_full_screen }, "force full screen" }, + { "an", OPT_BOOL, { &audio_disable }, "disable audio" }, + { "vn", OPT_BOOL, { &video_disable }, "disable video" }, + { "ast", OPT_INT | HAS_ARG | OPT_EXPERT, { &wanted_stream[AVMEDIA_TYPE_AUDIO] }, "select desired audio stream", "stream_number" }, + { "vst", OPT_INT | HAS_ARG | OPT_EXPERT, { &wanted_stream[AVMEDIA_TYPE_VIDEO] }, "select desired video stream", "stream_number" }, + { "sst", OPT_INT | HAS_ARG | OPT_EXPERT, { &wanted_stream[AVMEDIA_TYPE_SUBTITLE] }, "select desired subtitle stream", "stream_number" }, + { "ss", HAS_ARG, { .func_arg = opt_seek }, "seek to a given position in seconds", "pos" }, + { "t", HAS_ARG, { .func_arg = opt_duration }, "play \"duration\" seconds of audio/video", "duration" }, + { "bytes", OPT_INT | HAS_ARG, { &seek_by_bytes }, "seek by bytes 0=off 1=on -1=auto", "val" }, + { "nodisp", OPT_BOOL, { &display_disable }, "disable graphical display" }, + { "f", HAS_ARG, { .func_arg = opt_format }, "force format", "fmt" }, + { "pix_fmt", HAS_ARG | OPT_EXPERT | OPT_VIDEO, { .func_arg = opt_frame_pix_fmt }, "set pixel format", "format" }, + { "stats", OPT_BOOL | OPT_EXPERT, { &show_status }, "show status", "" }, + { "bug", OPT_INT | HAS_ARG | OPT_EXPERT, { &workaround_bugs }, "workaround bugs", "" }, + { "fast", OPT_BOOL | OPT_EXPERT, { &fast }, "non spec compliant optimizations", "" }, + { "genpts", OPT_BOOL | OPT_EXPERT, { &genpts }, "generate pts", "" }, + { "drp", OPT_INT | HAS_ARG | OPT_EXPERT, { &decoder_reorder_pts }, "let decoder reorder pts 0=off 1=on -1=auto", ""}, + { "skiploop", OPT_INT | HAS_ARG | OPT_EXPERT, { &skip_loop_filter }, "", "" }, + { "skipframe", OPT_INT | HAS_ARG | OPT_EXPERT, { &skip_frame }, "", "" }, + { "skipidct", OPT_INT | HAS_ARG | OPT_EXPERT, { &skip_idct }, "", "" }, + { "idct", OPT_INT | HAS_ARG | OPT_EXPERT, { &idct }, "set idct algo", "algo" }, + { "ec", OPT_INT | HAS_ARG | OPT_EXPERT, { &error_concealment }, "set error concealment options", "bit_mask" }, + { "sync", HAS_ARG | OPT_EXPERT, { .func_arg = opt_sync }, "set audio-video sync. type (type=audio/video/ext)", "type" }, + { "autoexit", OPT_BOOL | OPT_EXPERT, { &autoexit }, "exit at the end", "" }, + { "exitonkeydown", OPT_BOOL | OPT_EXPERT, { &exit_on_keydown }, "exit on key down", "" }, + { "exitonmousedown", OPT_BOOL | OPT_EXPERT, { &exit_on_mousedown }, "exit on mouse down", "" }, + { "loop", OPT_INT | HAS_ARG | OPT_EXPERT, { &loop }, "set number of times the playback shall be looped", "loop count" }, + { "framedrop", OPT_BOOL | OPT_EXPERT, { &framedrop }, "drop frames when cpu is too slow", "" }, + { "infbuf", OPT_BOOL | OPT_EXPERT, { &infinite_buffer }, "don't limit the input buffer size (useful with realtime streams)", "" }, + { "window_title", OPT_STRING | HAS_ARG, { &window_title }, "set window title", "window title" }, +#if CONFIG_AVFILTER + { "vf", OPT_STRING | HAS_ARG, { &vfilters }, "video filters", "filter list" }, +#endif + { "rdftspeed", OPT_INT | HAS_ARG| OPT_AUDIO | OPT_EXPERT, { &rdftspeed }, "rdft speed", "msecs" }, + { "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, { opt_default }, "generic catch all option", "" }, + { "i", 0, { NULL }, "avconv compatibility dummy option", ""}, + { NULL, }, +}; + +static void show_usage(void) +{ + printf("Simple media player\n"); + printf("usage: %s [options] input_file\n", program_name); + printf("\n"); +} + +void show_help_default(const char *opt, const char *arg) +{ + av_log_set_callback(log_callback_help); + show_usage(); + show_help_options(options, "Main options:", 0, OPT_EXPERT, 0); + show_help_options(options, "Advanced options:", OPT_EXPERT, 0, 0); + printf("\n"); + show_help_children(avcodec_get_class(), AV_OPT_FLAG_DECODING_PARAM); + show_help_children(avformat_get_class(), AV_OPT_FLAG_DECODING_PARAM); +#if !CONFIG_AVFILTER + show_help_children(sws_get_class(), AV_OPT_FLAG_ENCODING_PARAM); +#endif + printf("\nWhile playing:\n" + "q, ESC quit\n" + "f toggle full screen\n" + "p, SPC pause\n" + "a cycle audio channel\n" + "v cycle video channel\n" + "t cycle subtitle channel\n" + "w show audio waves\n" + "s activate frame-step mode\n" + "left/right seek backward/forward 10 seconds\n" + "down/up seek backward/forward 1 minute\n" + "mouse click seek to percentage in file corresponding to fraction of width\n" + ); +} + +static void opt_input_file(void *optctx, const char *filename) +{ + if (input_filename) { + fprintf(stderr, "Argument '%s' provided as input filename, but '%s' was already specified.\n", + filename, input_filename); + exit(1); + } + if (!strcmp(filename, "-")) + filename = "pipe:"; + input_filename = filename; +} + +/* Called from the main */ +int main(int argc, char **argv) +{ + int flags; + + av_log_set_flags(AV_LOG_SKIP_REPEATED); + parse_loglevel(argc, argv, options); + + /* register all codecs, demux and protocols */ + avcodec_register_all(); +#if CONFIG_AVDEVICE + avdevice_register_all(); +#endif +#if CONFIG_AVFILTER + avfilter_register_all(); +#endif + av_register_all(); + avformat_network_init(); + + init_opts(); + + show_banner(); + + parse_options(NULL, argc, argv, options, opt_input_file); + + if (!input_filename) { + show_usage(); + fprintf(stderr, "An input file must be specified\n"); + fprintf(stderr, "Use -h to get full help or, even better, run 'man %s'\n", program_name); + exit(1); + } + + if (display_disable) { + video_disable = 1; + } + flags = SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER; +#if !defined(__MINGW32__) && !defined(__APPLE__) + flags |= SDL_INIT_EVENTTHREAD; /* Not supported on Windows or Mac OS X */ +#endif + if (SDL_Init (flags)) { + fprintf(stderr, "Could not initialize SDL - %s\n", SDL_GetError()); + exit(1); + } + + if (!display_disable) { + const SDL_VideoInfo *vi = SDL_GetVideoInfo(); + fs_screen_width = vi->current_w; + fs_screen_height = vi->current_h; + } + + SDL_EventState(SDL_ACTIVEEVENT, SDL_IGNORE); + SDL_EventState(SDL_SYSWMEVENT, SDL_IGNORE); + SDL_EventState(SDL_USEREVENT, SDL_IGNORE); + + av_init_packet(&flush_pkt); + flush_pkt.data = (uint8_t *)&flush_pkt; + + cur_stream = stream_open(input_filename, file_iformat); + + event_loop(); + + /* never returns */ + + return 0; +} diff --git a/avprobe.c b/avprobe.c new file mode 100644 index 0000000000..c7b3d39cb2 --- /dev/null +++ b/avprobe.c @@ -0,0 +1,969 @@ +/* + * avprobe : Simple Media Prober based on the Libav libraries + * Copyright (c) 2007-2010 Stefano Sabatini + * + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "config.h" + +#include "libavformat/avformat.h" +#include "libavcodec/avcodec.h" +#include "libavutil/avstring.h" +#include "libavutil/opt.h" +#include "libavutil/pixdesc.h" +#include "libavutil/dict.h" +#include "libavutil/libm.h" +#include "libavdevice/avdevice.h" +#include "cmdutils.h" + +const char program_name[] = "avprobe"; +const int program_birth_year = 2007; + +static int do_show_format = 0; +static AVDictionary *fmt_entries_to_show = NULL; +static int nb_fmt_entries_to_show; +static int do_show_packets = 0; +static int do_show_streams = 0; + +static int show_value_unit = 0; +static int use_value_prefix = 0; +static int use_byte_value_binary_prefix = 0; +static int use_value_sexagesimal_format = 0; + +/* globals */ +static const OptionDef *options; + +/* AVprobe context */ +static const char *input_filename; +static AVInputFormat *iformat = NULL; + +static const char *const binary_unit_prefixes [] = { "", "Ki", "Mi", "Gi", "Ti", "Pi" }; +static const char *const decimal_unit_prefixes[] = { "", "K" , "M" , "G" , "T" , "P" }; + +static const char unit_second_str[] = "s" ; +static const char unit_hertz_str[] = "Hz" ; +static const char unit_byte_str[] = "byte" ; +static const char unit_bit_per_second_str[] = "bit/s"; + +static void avprobe_cleanup(int ret) +{ + av_dict_free(&fmt_entries_to_show); +} + +/* + * The output is structured in array and objects that might contain items + * Array could require the objects within to not be named. + * Object could require the items within to be named. + * + * For flat representation the name of each section is saved on prefix so it + * can be rendered in order to represent nested structures (e.g. array of + * objects for the packets list). + * + * Within an array each element can need an unique identifier or an index. + * + * Nesting level is accounted separately. + */ + +typedef enum { + ARRAY, + OBJECT +} PrintElementType; + +typedef struct { + const char *name; + PrintElementType type; + int64_t index; + int64_t nb_elems; +} PrintElement; + +typedef struct { + PrintElement *prefix; + int level; + void (*print_header)(void); + void (*print_footer)(void); + + void (*print_array_header) (const char *name); + void (*print_array_footer) (const char *name); + void (*print_object_header)(const char *name); + void (*print_object_footer)(const char *name); + + void (*print_integer) (const char *key, int64_t value); + void (*print_string) (const char *key, const char *value); +} PrintContext; + +static AVIOContext *probe_out = NULL; +static PrintContext octx; +#define AVP_INDENT() avio_printf(probe_out, "%*c", octx.level * 2, ' ') + +/* + * Default format, INI + * + * - all key and values are utf8 + * - '.' is the subgroup separator + * - newlines and the following characters are escaped + * - '\' is the escape character + * - '#' is the comment + * - '=' is the key/value separators + * - ':' is not used but usually parsed as key/value separator + */ + +static void ini_print_header(void) +{ + avio_printf(probe_out, "# avprobe output\n\n"); +} +static void ini_print_footer(void) +{ + avio_w8(probe_out, '\n'); +} + +static void ini_escape_print(const char *s) +{ + int i = 0; + char c = 0; + + while (c = s[i++]) { + switch (c) { + case '\r': avio_printf(probe_out, "%s", "\\r"); break; + case '\n': avio_printf(probe_out, "%s", "\\n"); break; + case '\f': avio_printf(probe_out, "%s", "\\f"); break; + case '\b': avio_printf(probe_out, "%s", "\\b"); break; + case '\t': avio_printf(probe_out, "%s", "\\t"); break; + case '\\': + case '#' : + case '=' : + case ':' : avio_w8(probe_out, '\\'); + default: + if ((unsigned char)c < 32) + avio_printf(probe_out, "\\x00%02x", c & 0xff); + else + avio_w8(probe_out, c); + break; + } + } +} + +static void ini_print_array_header(const char *name) +{ + if (octx.prefix[octx.level -1].nb_elems) + avio_printf(probe_out, "\n"); +} + +static void ini_print_object_header(const char *name) +{ + int i; + PrintElement *el = octx.prefix + octx.level -1; + + if (el->nb_elems) + avio_printf(probe_out, "\n"); + + avio_printf(probe_out, "["); + + for (i = 1; i < octx.level; i++) { + el = octx.prefix + i; + avio_printf(probe_out, "%s.", el->name); + if (el->index >= 0) + avio_printf(probe_out, "%"PRId64".", el->index); + } + + avio_printf(probe_out, "%s", name); + if (el && el->type == ARRAY) + avio_printf(probe_out, ".%"PRId64"", el->nb_elems); + avio_printf(probe_out, "]\n"); +} + +static void ini_print_integer(const char *key, int64_t value) +{ + ini_escape_print(key); + avio_printf(probe_out, "=%"PRId64"\n", value); +} + + +static void ini_print_string(const char *key, const char *value) +{ + ini_escape_print(key); + avio_printf(probe_out, "="); + ini_escape_print(value); + avio_w8(probe_out, '\n'); +} + +/* + * Alternate format, JSON + */ + +static void json_print_header(void) +{ + avio_printf(probe_out, "{"); +} +static void json_print_footer(void) +{ + avio_printf(probe_out, "}\n"); +} + +static void json_print_array_header(const char *name) +{ + if (octx.prefix[octx.level -1].nb_elems) + avio_printf(probe_out, ",\n"); + AVP_INDENT(); + avio_printf(probe_out, "\"%s\" : ", name); + avio_printf(probe_out, "[\n"); +} + +static void json_print_array_footer(const char *name) +{ + avio_printf(probe_out, "\n"); + AVP_INDENT(); + avio_printf(probe_out, "]"); +} + +static void json_print_object_header(const char *name) +{ + if (octx.prefix[octx.level -1].nb_elems) + avio_printf(probe_out, ",\n"); + AVP_INDENT(); + if (octx.prefix[octx.level -1].type == OBJECT) + avio_printf(probe_out, "\"%s\" : ", name); + avio_printf(probe_out, "{\n"); +} + +static void json_print_object_footer(const char *name) +{ + avio_printf(probe_out, "\n"); + AVP_INDENT(); + avio_printf(probe_out, "}"); +} + +static void json_print_integer(const char *key, int64_t value) +{ + if (octx.prefix[octx.level -1].nb_elems) + avio_printf(probe_out, ",\n"); + AVP_INDENT(); + avio_printf(probe_out, "\"%s\" : %"PRId64"", key, value); +} + +static void json_escape_print(const char *s) +{ + int i = 0; + char c = 0; + + while (c = s[i++]) { + switch (c) { + case '\r': avio_printf(probe_out, "%s", "\\r"); break; + case '\n': avio_printf(probe_out, "%s", "\\n"); break; + case '\f': avio_printf(probe_out, "%s", "\\f"); break; + case '\b': avio_printf(probe_out, "%s", "\\b"); break; + case '\t': avio_printf(probe_out, "%s", "\\t"); break; + case '\\': + case '"' : avio_w8(probe_out, '\\'); + default: + if ((unsigned char)c < 32) + avio_printf(probe_out, "\\u00%02x", c & 0xff); + else + avio_w8(probe_out, c); + break; + } + } +} + +static void json_print_string(const char *key, const char *value) +{ + if (octx.prefix[octx.level -1].nb_elems) + avio_printf(probe_out, ",\n"); + AVP_INDENT(); + avio_w8(probe_out, '\"'); + json_escape_print(key); + avio_printf(probe_out, "\" : \""); + json_escape_print(value); + avio_w8(probe_out, '\"'); +} + +/* + * old-style pseudo-INI + */ +static void old_print_object_header(const char *name) +{ + char *str, *p; + + if (!strcmp(name, "tags")) + return; + + str = p = av_strdup(name); + while (*p) { + *p = av_toupper(*p); + p++; + } + + avio_printf(probe_out, "[%s]\n", str); + av_freep(&str); +} + +static void old_print_object_footer(const char *name) +{ + char *str, *p; + + if (!strcmp(name, "tags")) + return; + + str = p = av_strdup(name); + while (*p) { + *p = av_toupper(*p); + p++; + } + + avio_printf(probe_out, "[/%s]\n", str); + av_freep(&str); +} + +static void old_print_string(const char *key, const char *value) +{ + if (!strcmp(octx.prefix[octx.level - 1].name, "tags")) + avio_printf(probe_out, "TAG:"); + ini_print_string(key, value); +} + +/* + * Simple Formatter for single entries. + */ + +static void show_format_entry_integer(const char *key, int64_t value) +{ + if (key && av_dict_get(fmt_entries_to_show, key, NULL, 0)) { + if (nb_fmt_entries_to_show > 1) + avio_printf(probe_out, "%s=", key); + avio_printf(probe_out, "%"PRId64"\n", value); + } +} + +static void show_format_entry_string(const char *key, const char *value) +{ + if (key && av_dict_get(fmt_entries_to_show, key, NULL, 0)) { + if (nb_fmt_entries_to_show > 1) + avio_printf(probe_out, "%s=", key); + avio_printf(probe_out, "%s\n", value); + } +} + +static void probe_group_enter(const char *name, int type) +{ + int64_t count = -1; + + octx.prefix = + av_realloc(octx.prefix, sizeof(PrintElement) * (octx.level + 1)); + + if (!octx.prefix || !name) { + fprintf(stderr, "Out of memory\n"); + exit_program(1); + } + + if (octx.level) { + PrintElement *parent = octx.prefix + octx.level -1; + if (parent->type == ARRAY) + count = parent->nb_elems; + parent->nb_elems++; + } + + octx.prefix[octx.level++] = (PrintElement){name, type, count, 0}; +} + +static void probe_group_leave(void) +{ + --octx.level; +} + +static void probe_header(void) +{ + if (octx.print_header) + octx.print_header(); + probe_group_enter("root", OBJECT); +} + +static void probe_footer(void) +{ + if (octx.print_footer) + octx.print_footer(); + probe_group_leave(); +} + + +static void probe_array_header(const char *name) +{ + if (octx.print_array_header) + octx.print_array_header(name); + + probe_group_enter(name, ARRAY); +} + +static void probe_array_footer(const char *name) +{ + probe_group_leave(); + if (octx.print_array_footer) + octx.print_array_footer(name); +} + +static void probe_object_header(const char *name) +{ + if (octx.print_object_header) + octx.print_object_header(name); + + probe_group_enter(name, OBJECT); +} + +static void probe_object_footer(const char *name) +{ + probe_group_leave(); + if (octx.print_object_footer) + octx.print_object_footer(name); +} + +static void probe_int(const char *key, int64_t value) +{ + octx.print_integer(key, value); + octx.prefix[octx.level -1].nb_elems++; +} + +static void probe_str(const char *key, const char *value) +{ + octx.print_string(key, value); + octx.prefix[octx.level -1].nb_elems++; +} + +static void probe_dict(AVDictionary *dict, const char *name) +{ + AVDictionaryEntry *entry = NULL; + if (!dict) + return; + probe_object_header(name); + while ((entry = av_dict_get(dict, "", entry, AV_DICT_IGNORE_SUFFIX))) { + probe_str(entry->key, entry->value); + } + probe_object_footer(name); +} + +static char *value_string(char *buf, int buf_size, double val, const char *unit) +{ + if (unit == unit_second_str && use_value_sexagesimal_format) { + double secs; + int hours, mins; + secs = val; + mins = (int)secs / 60; + secs = secs - mins * 60; + hours = mins / 60; + mins %= 60; + snprintf(buf, buf_size, "%d:%02d:%09.6f", hours, mins, secs); + } else if (use_value_prefix) { + const char *prefix_string; + int index; + + if (unit == unit_byte_str && use_byte_value_binary_prefix) { + index = (int) log2(val) / 10; + index = av_clip(index, 0, FF_ARRAY_ELEMS(binary_unit_prefixes) - 1); + val /= pow(2, index * 10); + prefix_string = binary_unit_prefixes[index]; + } else { + index = (int) (log10(val)) / 3; + index = av_clip(index, 0, FF_ARRAY_ELEMS(decimal_unit_prefixes) - 1); + val /= pow(10, index * 3); + prefix_string = decimal_unit_prefixes[index]; + } + snprintf(buf, buf_size, "%.*f%s%s", + index ? 3 : 0, val, + prefix_string, + show_value_unit ? unit : ""); + } else { + snprintf(buf, buf_size, "%f%s", val, show_value_unit ? unit : ""); + } + + return buf; +} + +static char *time_value_string(char *buf, int buf_size, int64_t val, + const AVRational *time_base) +{ + if (val == AV_NOPTS_VALUE) { + snprintf(buf, buf_size, "N/A"); + } else { + value_string(buf, buf_size, val * av_q2d(*time_base), unit_second_str); + } + + return buf; +} + +static char *ts_value_string(char *buf, int buf_size, int64_t ts) +{ + if (ts == AV_NOPTS_VALUE) { + snprintf(buf, buf_size, "N/A"); + } else { + snprintf(buf, buf_size, "%"PRId64, ts); + } + + return buf; +} + +static char *rational_string(char *buf, int buf_size, const char *sep, + const AVRational *rat) +{ + snprintf(buf, buf_size, "%d%s%d", rat->num, sep, rat->den); + return buf; +} + +static char *tag_string(char *buf, int buf_size, int tag) +{ + snprintf(buf, buf_size, "0x%04x", tag); + return buf; +} + +static void show_packet(AVFormatContext *fmt_ctx, AVPacket *pkt) +{ + char val_str[128]; + AVStream *st = fmt_ctx->streams[pkt->stream_index]; + + probe_object_header("packet"); + probe_str("codec_type", media_type_string(st->codec->codec_type)); + probe_int("stream_index", pkt->stream_index); + probe_str("pts", ts_value_string(val_str, sizeof(val_str), pkt->pts)); + probe_str("pts_time", time_value_string(val_str, sizeof(val_str), + pkt->pts, &st->time_base)); + probe_str("dts", ts_value_string(val_str, sizeof(val_str), pkt->dts)); + probe_str("dts_time", time_value_string(val_str, sizeof(val_str), + pkt->dts, &st->time_base)); + probe_str("duration", ts_value_string(val_str, sizeof(val_str), + pkt->duration)); + probe_str("duration_time", time_value_string(val_str, sizeof(val_str), + pkt->duration, + &st->time_base)); + probe_str("size", value_string(val_str, sizeof(val_str), + pkt->size, unit_byte_str)); + probe_int("pos", pkt->pos); + probe_str("flags", pkt->flags & AV_PKT_FLAG_KEY ? "K" : "_"); + probe_object_footer("packet"); +} + +static void show_packets(AVFormatContext *fmt_ctx) +{ + AVPacket pkt; + + av_init_packet(&pkt); + probe_array_header("packets"); + while (!av_read_frame(fmt_ctx, &pkt)) + show_packet(fmt_ctx, &pkt); + probe_array_footer("packets"); +} + +static void show_stream(AVFormatContext *fmt_ctx, int stream_idx) +{ + AVStream *stream = fmt_ctx->streams[stream_idx]; + AVCodecContext *dec_ctx; + const AVCodec *dec; + const char *profile; + char val_str[128]; + AVRational display_aspect_ratio, *sar = NULL; + const AVPixFmtDescriptor *desc; + + probe_object_header("stream"); + + probe_int("index", stream->index); + + if ((dec_ctx = stream->codec)) { + if ((dec = dec_ctx->codec)) { + probe_str("codec_name", dec->name); + probe_str("codec_long_name", dec->long_name); + } else { + probe_str("codec_name", "unknown"); + } + + probe_str("codec_type", media_type_string(dec_ctx->codec_type)); + probe_str("codec_time_base", + rational_string(val_str, sizeof(val_str), + "/", &dec_ctx->time_base)); + + /* print AVI/FourCC tag */ + av_get_codec_tag_string(val_str, sizeof(val_str), dec_ctx->codec_tag); + probe_str("codec_tag_string", val_str); + probe_str("codec_tag", tag_string(val_str, sizeof(val_str), + dec_ctx->codec_tag)); + + /* print profile, if there is one */ + if (dec && (profile = av_get_profile_name(dec, dec_ctx->profile))) + probe_str("profile", profile); + + switch (dec_ctx->codec_type) { + case AVMEDIA_TYPE_VIDEO: + probe_int("width", dec_ctx->width); + probe_int("height", dec_ctx->height); + probe_int("has_b_frames", dec_ctx->has_b_frames); + if (dec_ctx->sample_aspect_ratio.num) + sar = &dec_ctx->sample_aspect_ratio; + else if (stream->sample_aspect_ratio.num) + sar = &stream->sample_aspect_ratio; + + if (sar) { + probe_str("sample_aspect_ratio", + rational_string(val_str, sizeof(val_str), ":", sar)); + av_reduce(&display_aspect_ratio.num, &display_aspect_ratio.den, + dec_ctx->width * sar->num, dec_ctx->height * sar->den, + 1024*1024); + probe_str("display_aspect_ratio", + rational_string(val_str, sizeof(val_str), ":", + &display_aspect_ratio)); + } + desc = av_pix_fmt_desc_get(dec_ctx->pix_fmt); + probe_str("pix_fmt", desc ? desc->name : "unknown"); + probe_int("level", dec_ctx->level); + break; + + case AVMEDIA_TYPE_AUDIO: + probe_str("sample_rate", + value_string(val_str, sizeof(val_str), + dec_ctx->sample_rate, + unit_hertz_str)); + probe_int("channels", dec_ctx->channels); + probe_int("bits_per_sample", + av_get_bits_per_sample(dec_ctx->codec_id)); + break; + } + } else { + probe_str("codec_type", "unknown"); + } + + if (fmt_ctx->iformat->flags & AVFMT_SHOW_IDS) + probe_int("id", stream->id); + probe_str("avg_frame_rate", + rational_string(val_str, sizeof(val_str), "/", + &stream->avg_frame_rate)); + if (dec_ctx->bit_rate) + probe_str("bit_rate", + value_string(val_str, sizeof(val_str), + dec_ctx->bit_rate, unit_bit_per_second_str)); + probe_str("time_base", + rational_string(val_str, sizeof(val_str), "/", + &stream->time_base)); + probe_str("start_time", + time_value_string(val_str, sizeof(val_str), + stream->start_time, &stream->time_base)); + probe_str("duration", + time_value_string(val_str, sizeof(val_str), + stream->duration, &stream->time_base)); + if (stream->nb_frames) + probe_int("nb_frames", stream->nb_frames); + + probe_dict(stream->metadata, "tags"); + + probe_object_footer("stream"); +} + +static void show_format(AVFormatContext *fmt_ctx) +{ + char val_str[128]; + int64_t size = fmt_ctx->pb ? avio_size(fmt_ctx->pb) : -1; + + probe_object_header("format"); + probe_str("filename", fmt_ctx->filename); + probe_int("nb_streams", fmt_ctx->nb_streams); + probe_str("format_name", fmt_ctx->iformat->name); + probe_str("format_long_name", fmt_ctx->iformat->long_name); + probe_str("start_time", + time_value_string(val_str, sizeof(val_str), + fmt_ctx->start_time, &AV_TIME_BASE_Q)); + probe_str("duration", + time_value_string(val_str, sizeof(val_str), + fmt_ctx->duration, &AV_TIME_BASE_Q)); + probe_str("size", + size >= 0 ? value_string(val_str, sizeof(val_str), + size, unit_byte_str) + : "unknown"); + probe_str("bit_rate", + value_string(val_str, sizeof(val_str), + fmt_ctx->bit_rate, unit_bit_per_second_str)); + + probe_dict(fmt_ctx->metadata, "tags"); + + probe_object_footer("format"); +} + +static int open_input_file(AVFormatContext **fmt_ctx_ptr, const char *filename) +{ + int err, i; + AVFormatContext *fmt_ctx = NULL; + AVDictionaryEntry *t; + + if ((err = avformat_open_input(&fmt_ctx, filename, + iformat, &format_opts)) < 0) { + print_error(filename, err); + return err; + } + if ((t = av_dict_get(format_opts, "", NULL, AV_DICT_IGNORE_SUFFIX))) { + av_log(NULL, AV_LOG_ERROR, "Option %s not found.\n", t->key); + return AVERROR_OPTION_NOT_FOUND; + } + + + /* fill the streams in the format context */ + if ((err = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { + print_error(filename, err); + return err; + } + + av_dump_format(fmt_ctx, 0, filename, 0); + + /* bind a decoder to each input stream */ + for (i = 0; i < fmt_ctx->nb_streams; i++) { + AVStream *stream = fmt_ctx->streams[i]; + AVCodec *codec; + + if (stream->codec->codec_id == AV_CODEC_ID_PROBE) { + fprintf(stderr, "Failed to probe codec for input stream %d\n", + stream->index); + } else if (!(codec = avcodec_find_decoder(stream->codec->codec_id))) { + fprintf(stderr, + "Unsupported codec with id %d for input stream %d\n", + stream->codec->codec_id, stream->index); + } else if (avcodec_open2(stream->codec, codec, NULL) < 0) { + fprintf(stderr, "Error while opening codec for input stream %d\n", + stream->index); + } + } + + *fmt_ctx_ptr = fmt_ctx; + return 0; +} + +static void close_input_file(AVFormatContext **ctx_ptr) +{ + int i; + AVFormatContext *fmt_ctx = *ctx_ptr; + + /* close decoder for each stream */ + for (i = 0; i < fmt_ctx->nb_streams; i++) { + AVStream *stream = fmt_ctx->streams[i]; + + avcodec_close(stream->codec); + } + avformat_close_input(ctx_ptr); +} + +static int probe_file(const char *filename) +{ + AVFormatContext *fmt_ctx; + int ret, i; + + if ((ret = open_input_file(&fmt_ctx, filename))) + return ret; + + if (do_show_format) + show_format(fmt_ctx); + + if (do_show_streams) { + probe_array_header("streams"); + for (i = 0; i < fmt_ctx->nb_streams; i++) + show_stream(fmt_ctx, i); + probe_array_footer("streams"); + } + + if (do_show_packets) + show_packets(fmt_ctx); + + close_input_file(&fmt_ctx); + return 0; +} + +static void show_usage(void) +{ + printf("Simple multimedia streams analyzer\n"); + printf("usage: %s [OPTIONS] [INPUT_FILE]\n", program_name); + printf("\n"); +} + +static int opt_format(void *optctx, const char *opt, const char *arg) +{ + iformat = av_find_input_format(arg); + if (!iformat) { + fprintf(stderr, "Unknown input format: %s\n", arg); + return AVERROR(EINVAL); + } + return 0; +} + +static int opt_output_format(void *optctx, const char *opt, const char *arg) +{ + + if (!strcmp(arg, "json")) { + octx.print_header = json_print_header; + octx.print_footer = json_print_footer; + octx.print_array_header = json_print_array_header; + octx.print_array_footer = json_print_array_footer; + octx.print_object_header = json_print_object_header; + octx.print_object_footer = json_print_object_footer; + + octx.print_integer = json_print_integer; + octx.print_string = json_print_string; + } else if (!strcmp(arg, "ini")) { + octx.print_header = ini_print_header; + octx.print_footer = ini_print_footer; + octx.print_array_header = ini_print_array_header; + octx.print_object_header = ini_print_object_header; + + octx.print_integer = ini_print_integer; + octx.print_string = ini_print_string; + } else if (!strcmp(arg, "old")) { + octx.print_header = NULL; + octx.print_object_header = old_print_object_header; + octx.print_object_footer = old_print_object_footer; + + octx.print_string = old_print_string; + } else { + av_log(NULL, AV_LOG_ERROR, "Unsupported formatter %s\n", arg); + return AVERROR(EINVAL); + } + return 0; +} + +static int opt_show_format_entry(void *optctx, const char *opt, const char *arg) +{ + do_show_format = 1; + nb_fmt_entries_to_show++; + octx.print_header = NULL; + octx.print_footer = NULL; + octx.print_array_header = NULL; + octx.print_array_footer = NULL; + octx.print_object_header = NULL; + octx.print_object_footer = NULL; + + octx.print_integer = show_format_entry_integer; + octx.print_string = show_format_entry_string; + av_dict_set(&fmt_entries_to_show, arg, "", 0); + return 0; +} + +static void opt_input_file(void *optctx, const char *arg) +{ + if (input_filename) { + fprintf(stderr, + "Argument '%s' provided as input filename, but '%s' was already specified.\n", + arg, input_filename); + exit_program(1); + } + if (!strcmp(arg, "-")) + arg = "pipe:"; + input_filename = arg; +} + +void show_help_default(const char *opt, const char *arg) +{ + av_log_set_callback(log_callback_help); + show_usage(); + show_help_options(options, "Main options:", 0, 0, 0); + printf("\n"); + show_help_children(avformat_get_class(), AV_OPT_FLAG_DECODING_PARAM); +} + +static int opt_pretty(void *optctx, const char *opt, const char *arg) +{ + show_value_unit = 1; + use_value_prefix = 1; + use_byte_value_binary_prefix = 1; + use_value_sexagesimal_format = 1; + return 0; +} + +static const OptionDef real_options[] = { +#include "cmdutils_common_opts.h" + { "f", HAS_ARG, {.func_arg = opt_format}, "force format", "format" }, + { "of", HAS_ARG, {.func_arg = opt_output_format}, "output the document either as ini or json", "output_format" }, + { "unit", OPT_BOOL, {&show_value_unit}, + "show unit of the displayed values" }, + { "prefix", OPT_BOOL, {&use_value_prefix}, + "use SI prefixes for the displayed values" }, + { "byte_binary_prefix", OPT_BOOL, {&use_byte_value_binary_prefix}, + "use binary prefixes for byte units" }, + { "sexagesimal", OPT_BOOL, {&use_value_sexagesimal_format}, + "use sexagesimal format HOURS:MM:SS.MICROSECONDS for time units" }, + { "pretty", 0, {.func_arg = opt_pretty}, + "prettify the format of displayed values, make it more human readable" }, + { "show_format", OPT_BOOL, {&do_show_format} , "show format/container info" }, + { "show_format_entry", HAS_ARG, {.func_arg = opt_show_format_entry}, + "show a particular entry from the format/container info", "entry" }, + { "show_packets", OPT_BOOL, {&do_show_packets}, "show packets info" }, + { "show_streams", OPT_BOOL, {&do_show_streams}, "show streams info" }, + { "default", HAS_ARG | OPT_AUDIO | OPT_VIDEO | OPT_EXPERT, {.func_arg = opt_default}, + "generic catch all option", "" }, + { NULL, }, +}; + +static int probe_buf_write(void *opaque, uint8_t *buf, int buf_size) +{ + printf("%.*s", buf_size, buf); + return 0; +} + +#define AVP_BUFFSIZE 4096 + +int main(int argc, char **argv) +{ + int ret; + uint8_t *buffer = av_malloc(AVP_BUFFSIZE); + + if (!buffer) + exit(1); + + register_exit(avprobe_cleanup); + + options = real_options; + parse_loglevel(argc, argv, options); + av_register_all(); + avformat_network_init(); + init_opts(); +#if CONFIG_AVDEVICE + avdevice_register_all(); +#endif + + show_banner(); + + octx.print_header = ini_print_header; + octx.print_footer = ini_print_footer; + + octx.print_array_header = ini_print_array_header; + octx.print_object_header = ini_print_object_header; + + octx.print_integer = ini_print_integer; + octx.print_string = ini_print_string; + + parse_options(NULL, argc, argv, options, opt_input_file); + + if (!input_filename) { + show_usage(); + fprintf(stderr, "You have to specify one input file.\n"); + fprintf(stderr, + "Use -h to get full help or, even better, run 'man %s'.\n", + program_name); + exit_program(1); + } + + probe_out = avio_alloc_context(buffer, AVP_BUFFSIZE, 1, NULL, NULL, + probe_buf_write, NULL); + if (!probe_out) + exit_program(1); + + probe_header(); + ret = probe_file(input_filename); + probe_footer(); + avio_flush(probe_out); + avio_close(probe_out); + + avformat_network_deinit(); + + return ret; +} diff --git a/ffserver.c b/avserver.c similarity index 88% rename from ffserver.c rename to avserver.c index a6ab458de6..56bcda9d4d 100644 --- a/ffserver.c +++ b/avserver.c @@ -1,28 +1,24 @@ /* + * Multiple format streaming server * Copyright (c) 2000, 2001, 2002 Fabrice Bellard * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -/** - * @file - * multiple format streaming server based on the FFmpeg libraries - */ - #include "config.h" #if !HAVE_CLOSESOCKET #define closesocket close @@ -31,25 +27,22 @@ #include #include #include "libavformat/avformat.h" -// FIXME those are internal headers, ffserver _really_ shouldn't use them +// FIXME those are internal headers, avserver _really_ shouldn't use them #include "libavformat/ffm.h" #include "libavformat/network.h" #include "libavformat/os_support.h" #include "libavformat/rtpdec.h" #include "libavformat/rtpproto.h" #include "libavformat/rtsp.h" -#include "libavformat/rtspcodes.h" #include "libavformat/avio_internal.h" #include "libavformat/internal.h" #include "libavformat/url.h" -#include "libavutil/avassert.h" #include "libavutil/avstring.h" #include "libavutil/lfg.h" #include "libavutil/dict.h" #include "libavutil/intreadwrite.h" #include "libavutil/mathematics.h" -#include "libavutil/pixdesc.h" #include "libavutil/random_seed.h" #include "libavutil/parseutils.h" #include "libavutil/opt.h" @@ -69,7 +62,7 @@ #include "cmdutils.h" -const char program_name[] = "ffserver"; +const char program_name[] = "avserver"; const int program_birth_year = 2000; static const OptionDef options[]; @@ -210,28 +203,31 @@ typedef struct IPAddressACL { struct in_addr last; } IPAddressACL; -/* description of each stream of the ffserver.conf file */ +/* description of each stream of the avserver.conf file */ typedef struct FFStream { enum StreamType stream_type; char filename[1024]; /* stream filename */ struct FFStream *feed; /* feed we are using (can be null if coming from file) */ AVDictionary *in_opts; /* input parameters */ - AVDictionary *metadata; /* metadata to set on the stream */ AVInputFormat *ifmt; /* if non NULL, force input format */ AVOutputFormat *fmt; IPAddressACL *acl; char dynamic_acl[1024]; int nb_streams; - int prebuffer; /* Number of milliseconds early to start */ + int prebuffer; /* Number of millseconds early to start */ int64_t max_time; /* Number of milliseconds to run */ int send_on_key; AVStream *streams[MAX_STREAMS]; int feed_streams[MAX_STREAMS]; /* index of streams in the feed */ char feed_filename[1024]; /* file name of the feed storage, or input file name for a stream */ - pid_t pid; /* Of ffmpeg process */ - time_t pid_start; /* Of ffmpeg process */ + char author[512]; + char title[512]; + char copyright[512]; + char comment[512]; + pid_t pid; /* of avconv process */ + time_t pid_start; /* of avconv process */ char **child_argv; struct FFStream *next; unsigned bandwidth; /* bandwidth, in kbits/s */ @@ -288,7 +284,8 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url); static void rtsp_cmd_options(HTTPContext *c, const char *url); static void rtsp_cmd_setup(HTTPContext *c, const char *url, RTSPMessageHeader *h); static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h); -static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeader *h, int pause_only); +static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h); +static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader *h); /* SDP handling */ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer, @@ -306,7 +303,7 @@ static const char *my_program_name; static const char *config_filename; -static int ffserver_debug; +static int avserver_debug; static int no_launch; static int need_to_start_children; @@ -324,20 +321,11 @@ static AVLFG random_state; static FILE *logfile = NULL; -static void htmlstrip(char *s) { - while (s && *s) { - s += strspn(s, "0123456789abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ,. "); - if (*s) - *s++ = '?'; - } -} - static int64_t ffm_read_write_index(int fd) { uint8_t buf[8]; - if (lseek(fd, 8, SEEK_SET) < 0) - return AVERROR(EIO); + lseek(fd, 8, SEEK_SET); if (read(fd, buf, 8) != 8) return AVERROR(EIO); return AV_RB64(buf); @@ -350,8 +338,7 @@ static int ffm_write_write_index(int fd, int64_t pos) for(i=0;i<8;i++) buf[i] = (pos >> (56 - i * 8)) & 0xff; - if (lseek(fd, 8, SEEK_SET) < 0) - return AVERROR(EIO); + lseek(fd, 8, SEEK_SET); if (write(fd, buf, 8) != 8) return AVERROR(EIO); return 8; @@ -365,7 +352,7 @@ static void ffm_set_write_index(AVFormatContext *s, int64_t pos, ffm->file_size = file_size; } -/* FIXME: make ffserver work with IPv6 */ +/* FIXME: make avserver work with IPv6 */ /* resolve host with also IP address parsing */ static int resolve_host(struct in_addr *sin_addr, const char *hostname) { @@ -400,14 +387,14 @@ static int resolve_host(struct in_addr *sin_addr, const char *hostname) return 0; } -static char *ctime1(char *buf2, int buf_size) +static char *ctime1(char *buf2) { time_t ti; char *p; ti = time(NULL); p = ctime(&ti); - av_strlcpy(buf2, p, buf_size); + strcpy(buf2, p); p = buf2 + strlen(p) - 1; if (*p == '\n') *p = '\0'; @@ -420,7 +407,7 @@ static void http_vlog(const char *fmt, va_list vargs) if (logfile) { if (print_prefix) { char buf[32]; - ctime1(buf, sizeof(buf)); + ctime1(buf); fprintf(logfile, "%s ", buf); } print_prefix = strstr(fmt, "\n") != NULL; @@ -506,15 +493,14 @@ static void start_children(FFStream *feed) char *slash; int i; - /* replace "ffserver" with "ffmpeg" in the path of current - * program. Ignore user provided path */ av_strlcpy(pathname, my_program_name, sizeof(pathname)); + slash = strrchr(pathname, '/'); if (!slash) slash = pathname; else slash++; - strcpy(slash, "ffmpeg"); + strcpy(slash, "avconv"); http_log("Launch command line: "); http_log("%s ", pathname); @@ -525,7 +511,7 @@ static void start_children(FFStream *feed) for (i = 3; i < 256; i++) close(i); - if (!ffserver_debug) { + if (!avserver_debug) { if (!freopen("/dev/null", "r", stdin)) http_log("failed to redirect STDIN to /dev/null\n;"); if (!freopen("/dev/null", "w", stdout)) @@ -556,8 +542,7 @@ static int socket_open_listen(struct sockaddr_in *my_addr) } tmp = 1; - if (setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp))) - av_log(NULL, AV_LOG_WARNING, "setsockopt SO_REUSEADDR failed\n"); + setsockopt(server_fd, SOL_SOCKET, SO_REUSEADDR, &tmp, sizeof(tmp)); my_addr->sin_family = AF_INET; if (bind (server_fd, (struct sockaddr *) my_addr, sizeof (*my_addr)) < 0) { @@ -573,9 +558,7 @@ static int socket_open_listen(struct sockaddr_in *my_addr) closesocket(server_fd); return -1; } - - if (ff_socket_nonblock(server_fd, 1) < 0) - av_log(NULL, AV_LOG_WARNING, "ff_socket_nonblock failed\n"); + ff_socket_nonblock(server_fd, 1); return server_fd; } @@ -586,17 +569,15 @@ static void start_multicast(void) FFStream *stream; char session_id[32]; HTTPContext *rtp_c; - struct sockaddr_in dest_addr = {0}; + struct sockaddr_in dest_addr; int default_port, stream_index; default_port = 6000; for(stream = first_stream; stream != NULL; stream = stream->next) { if (stream->is_multicast) { - unsigned random0 = av_lfg_get(&random_state); - unsigned random1 = av_lfg_get(&random_state); /* open the RTP connection */ snprintf(session_id, sizeof(session_id), "%08x%08x", - random0, random1); + av_lfg_get(&random_state), av_lfg_get(&random_state)); /* choose a port if none given */ if (stream->multicast_port == 0) { @@ -631,16 +612,17 @@ static void start_multicast(void) } } + /* change state to send data */ rtp_c->state = HTTPSTATE_SEND_DATA; } } } -/* main loop of the HTTP server */ +/* main loop of the http server */ static int http_server(void) { int server_fd = 0, rtsp_server_fd = 0; - int ret, delay; + int ret, delay, delay1; struct pollfd *poll_table, *poll_entry; HTTPContext *c, *c_next; @@ -651,28 +633,22 @@ static int http_server(void) if (my_http_addr.sin_port) { server_fd = socket_open_listen(&my_http_addr); - if (server_fd < 0) { - av_free(poll_table); + if (server_fd < 0) return -1; - } } if (my_rtsp_addr.sin_port) { rtsp_server_fd = socket_open_listen(&my_rtsp_addr); - if (rtsp_server_fd < 0) { - av_free(poll_table); - closesocket(server_fd); + if (rtsp_server_fd < 0) return -1; - } } if (!rtsp_server_fd && !server_fd) { http_log("HTTP and RTSP disabled.\n"); - av_free(poll_table); return -1; } - http_log("FFserver started.\n"); + http_log("AVserver started.\n"); start_children(first_feed); @@ -710,19 +686,18 @@ static int http_server(void) case HTTPSTATE_SEND_DATA: case HTTPSTATE_SEND_DATA_TRAILER: if (!c->is_packetized) { - /* for TCP, we output as much as we can - * (may need to put a limit) */ + /* for TCP, we output as much as we can (may need to put a limit) */ c->poll_entry = poll_entry; poll_entry->fd = fd; poll_entry->events = POLLOUT; poll_entry++; } else { - /* when ffserver is doing the timing, we work by - looking at which packet needs to be sent every + /* when avserver is doing the timing, we work by + looking at which packet need to be sent every 10 ms */ - /* one tick wait XXX: 10 ms assumed */ - if (delay > 10) - delay = 10; + delay1 = 10; /* one tick wait XXX: 10 ms assumed */ + if (delay1 < delay) + delay = delay1; } break; case HTTPSTATE_WAIT_REQUEST: @@ -747,10 +722,8 @@ static int http_server(void) do { ret = poll(poll_table, poll_entry - poll_table, delay); if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) && - ff_neterrno() != AVERROR(EINTR)) { - av_free(poll_table); + ff_neterrno() != AVERROR(EINTR)) return -1; - } } while (ret < 0); cur_time = av_gettime() / 1000; @@ -764,8 +737,8 @@ static int http_server(void) for(c = first_http_ctx; c != NULL; c = c_next) { c_next = c->next; if (handle_connection(c) < 0) { - log_connection(c); /* close and free the connection */ + log_connection(c); close_connection(c); } } @@ -802,19 +775,17 @@ static void start_wait_request(HTTPContext *c, int is_rtsp) static void http_send_too_busy_reply(int fd) { - char buffer[400]; + char buffer[300]; int len = snprintf(buffer, sizeof(buffer), "HTTP/1.0 503 Server too busy\r\n" "Content-type: text/html\r\n" "\r\n" "Too busy\r\n" "

The server is too busy to serve your request at this time.

\r\n" - "

The number of current connections is %u, and this exceeds the limit of %u.

\r\n" + "

The number of current connections is %d, and this exceeds the limit of %d.

\r\n" "\r\n", nb_connections, nb_max_connections); - av_assert0(len < sizeof(buffer)); - if (send(fd, buffer, len, 0) < len) - av_log(NULL, AV_LOG_WARNING, "Could not send too-busy reply, send() failed\n"); + send(fd, buffer, len, 0); } @@ -832,8 +803,7 @@ static void new_connection(int server_fd, int is_rtsp) http_log("error during accept %s\n", strerror(errno)); return; } - if (ff_socket_nonblock(fd, 1) < 0) - av_log(NULL, AV_LOG_WARNING, "ff_socket_nonblock failed\n"); + ff_socket_nonblock(fd, 1); if (nb_connections >= nb_max_connections) { http_send_too_busy_reply(fd); @@ -939,8 +909,6 @@ static void close_connection(HTTPContext *c) for(i=0; inb_streams; i++) av_free(ctx->streams[i]); - av_freep(&ctx->streams); - av_freep(&ctx->priv_data); if (c->stream && !c->post && c->stream->stream_type == STREAM_TYPE_LIVE) current_bandwidth -= c->stream->bandwidth; @@ -1016,7 +984,9 @@ static int handle_connection(HTTPContext *c) if (len < 0) { if (ff_neterrno() != AVERROR(EAGAIN) && ff_neterrno() != AVERROR(EINTR)) { - goto close_connection; + /* error : close connection */ + av_freep(&c->pb_buffer); + return -1; } } else { c->buffer_ptr += len; @@ -1028,8 +998,7 @@ static int handle_connection(HTTPContext *c) /* if error, exit */ if (c->http_error) return -1; - /* all the buffer was sent : synchronize to the incoming - * stream */ + /* all the buffer was sent : synchronize to the incoming stream */ c->state = HTTPSTATE_SEND_DATA_HEADER; c->buffer_ptr = c->buffer_end = c->buffer; } @@ -1040,7 +1009,7 @@ static int handle_connection(HTTPContext *c) case HTTPSTATE_SEND_DATA_HEADER: case HTTPSTATE_SEND_DATA_TRAILER: /* for packetized output, we consider we can always write (the - input streams set the speed). It may be better to verify + input streams sets the speed). It may be better to verify that we do not rely too much on the kernel queues */ if (!c->is_packetized) { if (c->poll_entry->revents & (POLLERR | POLLHUP)) @@ -1074,8 +1043,10 @@ static int handle_connection(HTTPContext *c) break; case RTSPSTATE_SEND_REPLY: - if (c->poll_entry->revents & (POLLERR | POLLHUP)) - goto close_connection; + if (c->poll_entry->revents & (POLLERR | POLLHUP)) { + av_freep(&c->pb_buffer); + return -1; + } /* no need to write if no events */ if (!(c->poll_entry->revents & POLLOUT)) return 0; @@ -1083,7 +1054,9 @@ static int handle_connection(HTTPContext *c) if (len < 0) { if (ff_neterrno() != AVERROR(EAGAIN) && ff_neterrno() != AVERROR(EINTR)) { - goto close_connection; + /* error : close connection */ + av_freep(&c->pb_buffer); + return -1; } } else { c->buffer_ptr += len; @@ -1128,10 +1101,6 @@ static int handle_connection(HTTPContext *c) return -1; } return 0; - -close_connection: - av_freep(&c->pb_buffer); - return -1; } static int extract_rates(char *rates, int ratelen, const char *request) @@ -1338,7 +1307,7 @@ static void parse_acl_row(FFStream *stream, FFStream* feed, IPAddressACL *ext_ac get_arg(arg, sizeof(arg), &p); if (resolve_host(&acl.first, arg) != 0) { - fprintf(stderr, "%s:%d: ACL refers to invalid host or IP address '%s'\n", + fprintf(stderr, "%s:%d: ACL refers to invalid host or ip address '%s'\n", filename, line_num, arg); errors++; } else @@ -1348,7 +1317,7 @@ static void parse_acl_row(FFStream *stream, FFStream* feed, IPAddressACL *ext_ac if (arg[0]) { if (resolve_host(&acl.last, arg) != 0) { - fprintf(stderr, "%s:%d: ACL refers to invalid host or IP address '%s'\n", + fprintf(stderr, "%s:%d: ACL refers to invalid host or ip address '%s'\n", filename, line_num, arg); errors++; } @@ -1378,8 +1347,7 @@ static void parse_acl_row(FFStream *stream, FFStream* feed, IPAddressACL *ext_ac naclp = &(*naclp)->next; *naclp = nacl; - } else - av_free(nacl); + } } } @@ -1471,7 +1439,7 @@ static int validate_acl(FFStream *stream, HTTPContext *c) } /* compute the real filename of a file by matching it without its - extensions to all the stream's filenames */ + extensions to all the stream filenames */ static void compute_real_filename(char *filename, int max_size) { char file1[1024]; @@ -1505,7 +1473,7 @@ enum RedirType { REDIR_SDP, }; -/* parse HTTP request and prepare header */ +/* parse http request and prepare header */ static int http_parse_request(HTTPContext *c) { const char *p; @@ -1542,7 +1510,7 @@ static int http_parse_request(HTTPContext *c) av_strlcpy(c->protocol, protocol, sizeof(c->protocol)); - if (ffserver_debug) + if (avserver_debug) http_log("%s - - New connection: %s %s\n", inet_ntoa(c->from_addr.sin_addr), cmd, url); /* find the filename and the optional info string in the request */ @@ -1611,7 +1579,7 @@ static int http_parse_request(HTTPContext *c) if (stream->stream_type == STREAM_TYPE_REDIRECT) { c->http_error = 301; q = c->buffer; - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 301 Moved\r\n" "Location: %s\r\n" "Content-type: text/html\r\n" @@ -1619,7 +1587,6 @@ static int http_parse_request(HTTPContext *c) "Moved\r\n" "You should be redirected.\r\n" "\r\n", stream->feed_filename, stream->feed_filename); - q += strlen(q); /* prepare output buffer */ c->buffer_ptr = c->buffer; c->buffer_end = q; @@ -1650,7 +1617,7 @@ static int http_parse_request(HTTPContext *c) if (c->post == 0 && max_bandwidth < current_bandwidth) { c->http_error = 503; q = c->buffer; - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 503 Server too busy\r\n" "Content-type: text/html\r\n" "\r\n" @@ -1659,7 +1626,6 @@ static int http_parse_request(HTTPContext *c) "

The bandwidth being served (including your stream) is %"PRIu64"kbit/sec, " "and this exceeds the limit of %"PRIu64"kbit/sec.

\r\n" "\r\n", current_bandwidth, max_bandwidth); - q += strlen(q); /* prepare output buffer */ c->buffer_ptr = c->buffer; c->buffer_end = q; @@ -1702,33 +1668,30 @@ static int http_parse_request(HTTPContext *c) q = c->buffer; switch(redir_type) { case REDIR_ASX: - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 200 ASX Follows\r\n" "Content-type: video/x-ms-asf\r\n" "\r\n" "\r\n" - //"\r\n" + //"\r\n" "\r\n" "\r\n", hostbuf, filename, info); - q += strlen(q); break; case REDIR_RAM: - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 200 RAM Follows\r\n" "Content-type: audio/x-pn-realaudio\r\n" "\r\n" - "# Autogenerated by ffserver\r\n" + "# Autogenerated by avserver\r\n" "http://%s/%s%s\r\n", hostbuf, filename, info); - q += strlen(q); break; case REDIR_ASF: - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 200 ASF Redirect follows\r\n" "Content-type: video/x-ms-asf\r\n" "\r\n" "[Reference]\r\n" "Ref1=http://%s/%s%s\r\n", hostbuf, filename, info); - q += strlen(q); break; case REDIR_RTSP: { @@ -1738,13 +1701,12 @@ static int http_parse_request(HTTPContext *c) p = strrchr(hostname, ':'); if (p) *p = '\0'; - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 200 RTSP Redirect follows\r\n" - /* XXX: incorrect MIME type ? */ + /* XXX: incorrect mime type ? */ "Content-type: application/x-rtsp\r\n" "\r\n" "rtsp://%s:%d/%s\r\n", hostname, ntohs(my_rtsp_addr.sin_port), filename); - q += strlen(q); } break; case REDIR_SDP: @@ -1754,17 +1716,13 @@ static int http_parse_request(HTTPContext *c) socklen_t len; struct sockaddr_in my_addr; - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 200 OK\r\n" "Content-type: application/sdp\r\n" "\r\n"); - q += strlen(q); len = sizeof(my_addr); - - /* XXX: Should probably fail? */ - if (getsockname(c->fd, (struct sockaddr *)&my_addr, &len)) - http_log("getsockname() failed\n"); + getsockname(c->fd, (struct sockaddr *)&my_addr, &len); /* XXX: should use a dynamic buffer */ sdp_data_size = prepare_sdp_description(stream, @@ -1804,7 +1762,7 @@ static int http_parse_request(HTTPContext *c) /* if post, it means a feed is being sent */ if (!stream->is_feed) { /* However it might be a status report from WMP! Let us log the - * data as it might come handy one day. */ + * data as it might come in handy one day. */ const char *logline = 0; int client_id = 0; @@ -1879,13 +1837,13 @@ static int http_parse_request(HTTPContext *c) goto send_error; } - /* prepare HTTP header */ - c->buffer[0] = 0; - av_strlcatf(c->buffer, c->buffer_size, "HTTP/1.0 200 OK\r\n"); + /* prepare http header */ + q = c->buffer; + q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "HTTP/1.0 200 OK\r\n"); mime_type = c->stream->fmt->mime_type; if (!mime_type) mime_type = "application/x-octet-stream"; - av_strlcatf(c->buffer, c->buffer_size, "Pragma: no-cache\r\n"); + q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Pragma: no-cache\r\n"); /* for asf, we need extra headers */ if (!strcmp(c->stream->fmt->name,"asf_stream")) { @@ -1893,11 +1851,10 @@ static int http_parse_request(HTTPContext *c) c->wmp_client_id = av_lfg_get(&random_state); - av_strlcatf(c->buffer, c->buffer_size, "Server: Cougar 4.1.0.3923\r\nCache-Control: no-cache\r\nPragma: client-id=%d\r\nPragma: features=\"broadcast\"\r\n", c->wmp_client_id); + q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Server: Cougar 4.1.0.3923\r\nCache-Control: no-cache\r\nPragma: client-id=%d\r\nPragma: features=\"broadcast\"\r\n", c->wmp_client_id); } - av_strlcatf(c->buffer, c->buffer_size, "Content-Type: %s\r\n", mime_type); - av_strlcatf(c->buffer, c->buffer_size, "\r\n"); - q = c->buffer + strlen(c->buffer); + q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "Content-Type: %s\r\n", mime_type); + q += snprintf(q, q - (char *) c->buffer + c->buffer_size, "\r\n"); /* prepare output buffer */ c->http_error = 0; @@ -1908,8 +1865,7 @@ static int http_parse_request(HTTPContext *c) send_error: c->http_error = 404; q = c->buffer; - htmlstrip(msg); - snprintf(q, c->buffer_size, + q += snprintf(q, c->buffer_size, "HTTP/1.0 404 Not Found\r\n" "Content-type: text/html\r\n" "\r\n" @@ -1917,7 +1873,6 @@ static int http_parse_request(HTTPContext *c) "404 Not Found\n" "%s\n" "\n", msg); - q += strlen(q); /* prepare output buffer */ c->buffer_ptr = c->buffer; c->buffer_end = q; @@ -1958,7 +1913,7 @@ static void compute_status(HTTPContext *c) } avio_printf(pb, "HTTP/1.0 200 OK\r\n"); - avio_printf(pb, "Content-type: text/html\r\n"); + avio_printf(pb, "Content-type: %s\r\n", "text/html"); avio_printf(pb, "Pragma: no-cache\r\n"); avio_printf(pb, "\r\n"); @@ -2067,7 +2022,7 @@ static void compute_status(HTTPContext *c) if (stream->pid) { avio_printf(pb, "Running as pid %d.\n", stream->pid); -#if defined(linux) +#if defined(linux) && !defined(CONFIG_NOCUTILS) { FILE *pid_stat; char ps_cmd[64]; @@ -2082,7 +2037,7 @@ static void compute_status(HTTPContext *c) char cpuperc[10]; char cpuused[64]; - if (fscanf(pid_stat, "%9s %63s", cpuperc, + if (fscanf(pid_stat, "%10s %64s", cpuperc, cpuused) == 2) { avio_printf(pb, "Currently using %s%% of the cpu. Total time used %s.\n", cpuperc, cpuused); @@ -2188,19 +2143,16 @@ static int open_input_stream(HTTPContext *c, const char *info) char buf[128]; char input_filename[1024]; AVFormatContext *s = NULL; - int buf_size, i, ret; + int i, ret; int64_t stream_pos; /* find file name */ if (c->stream->feed) { strcpy(input_filename, c->stream->feed->feed_filename); - buf_size = FFM_PACKET_SIZE; /* compute position (absolute time) */ if (av_find_info_tag(buf, sizeof(buf), "date", info)) { - if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0) { - http_log("Invalid date specification '%s' for stream\n", buf); + if ((ret = av_parse_time(&stream_pos, buf, 0)) < 0) return ret; - } } else if (av_find_info_tag(buf, sizeof(buf), "buffer", info)) { int prebuffer = strtol(buf, 0, 10); stream_pos = av_gettime() - prebuffer * (int64_t)1000000; @@ -2208,41 +2160,31 @@ static int open_input_stream(HTTPContext *c, const char *info) stream_pos = av_gettime() - c->stream->prebuffer * (int64_t)1000; } else { strcpy(input_filename, c->stream->feed_filename); - buf_size = 0; /* compute position (relative time) */ if (av_find_info_tag(buf, sizeof(buf), "date", info)) { - if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0) { - http_log("Invalid date specification '%s' for stream\n", buf); + if ((ret = av_parse_time(&stream_pos, buf, 1)) < 0) return ret; - } } else stream_pos = 0; } - if (!input_filename[0]) { - http_log("No filename was specified for stream\n"); - return AVERROR(EINVAL); - } + if (input_filename[0] == '\0') + return -1; /* open stream */ if ((ret = avformat_open_input(&s, input_filename, c->stream->ifmt, &c->stream->in_opts)) < 0) { - http_log("Could not open input '%s': %s\n", input_filename, av_err2str(ret)); - return ret; + http_log("could not open %s: %d\n", input_filename, ret); + return -1; } - - /* set buffer size */ - if (buf_size > 0) ffio_set_buf_size(s->pb, buf_size); - s->flags |= AVFMT_FLAG_GENPTS; c->fmt_in = s; - if (strcmp(s->iformat->name, "ffm") && - (ret = avformat_find_stream_info(c->fmt_in, NULL)) < 0) { - http_log("Could not find stream info for input '%s'\n", input_filename); + if (strcmp(s->iformat->name, "ffm") && avformat_find_stream_info(c->fmt_in, NULL) < 0) { + http_log("Could not find stream info '%s'\n", input_filename); avformat_close_input(&s); - return ret; + return -1; } - /* choose stream as clock source (we favor the video stream if - * present) for packet sending */ + /* choose stream as clock source (we favorize video stream if + present) for packet sending */ c->pts_stream_index = 0; for(i=0;istream->nb_streams;i++) { if (c->pts_stream_index == 0 && @@ -2291,10 +2233,12 @@ static int http_prepare_data(HTTPContext *c) av_freep(&c->pb_buffer); switch(c->state) { case HTTPSTATE_SEND_DATA_HEADER: - ctx = avformat_alloc_context(); - c->fmt_ctx = *ctx; - av_freep(&ctx); - av_dict_copy(&(c->fmt_ctx.metadata), c->stream->metadata, 0); + memset(&c->fmt_ctx, 0, sizeof(c->fmt_ctx)); + av_dict_set(&c->fmt_ctx.metadata, "author" , c->stream->author , 0); + av_dict_set(&c->fmt_ctx.metadata, "comment" , c->stream->comment , 0); + av_dict_set(&c->fmt_ctx.metadata, "copyright", c->stream->copyright, 0); + av_dict_set(&c->fmt_ctx.metadata, "title" , c->stream->title , 0); + c->fmt_ctx.streams = av_mallocz(sizeof(AVStream *) * c->stream->nb_streams); for(i=0;istream->nb_streams;i++) { @@ -2309,8 +2253,8 @@ static int http_prepare_data(HTTPContext *c) *(c->fmt_ctx.streams[i]) = *src; c->fmt_ctx.streams[i]->priv_data = 0; - /* XXX: should be done in AVStream, not in codec */ - c->fmt_ctx.streams[i]->codec->frame_number = 0; + c->fmt_ctx.streams[i]->codec->frame_number = 0; /* XXX: should be done in + AVStream, not in codec */ } /* set output format parameters */ c->fmt_ctx.oformat = c->stream->fmt; @@ -2326,16 +2270,15 @@ static int http_prepare_data(HTTPContext *c) c->fmt_ctx.pb->seekable = 0; /* - * HACK to avoid MPEG-PS muxer to spit many underflow errors - * Default value from FFmpeg - * Try to set it using configuration option + * HACK to avoid mpeg ps muxer to spit many underflow errors + * Default value from Libav + * Try to set it use configuration option */ c->fmt_ctx.max_delay = (int)(0.7*AV_TIME_BASE); - if ((ret = avformat_write_header(&c->fmt_ctx, NULL)) < 0) { - http_log("Error writing output header for stream '%s': %s\n", - c->stream->filename, av_err2str(ret)); - return ret; + if (avformat_write_header(&c->fmt_ctx, NULL) < 0) { + http_log("Error writing output header\n"); + return -1; } av_dict_free(&c->fmt_ctx.metadata); @@ -2379,7 +2322,7 @@ static int http_prepare_data(HTTPContext *c) goto redo; } else { no_loop: - /* must send trailer now because EOF or error */ + /* must send trailer now because eof or error */ c->state = HTTPSTATE_SEND_DATA_TRAILER; } } @@ -2421,8 +2364,8 @@ static int http_prepare_data(HTTPContext *c) send_it: ist = c->fmt_in->streams[source_index]; /* specific handling for RTP: we use several - * output streams (one for each RTP connection). - * XXX: need more abstract handling */ + output stream (one for each RTP + connection). XXX: need more abstract handling */ if (c->is_packetized) { /* compute send time and duration */ c->cur_pts = av_rescale_q(pkt.dts, ist->time_base, AV_TIME_BASE_Q); @@ -2466,9 +2409,8 @@ static int http_prepare_data(HTTPContext *c) if (pkt.pts != AV_NOPTS_VALUE) pkt.pts = av_rescale_q(pkt.pts, ist->time_base, ost->time_base); pkt.duration = av_rescale_q(pkt.duration, ist->time_base, ost->time_base); - if ((ret = av_write_frame(ctx, &pkt)) < 0) { - http_log("Error writing frame to output for stream '%s': %s\n", - c->stream->filename, av_err2str(ret)); + if (av_write_frame(ctx, &pkt) < 0) { + http_log("Error writing frame to output\n"); c->state = HTTPSTATE_SEND_DATA_TRAILER; } @@ -2512,7 +2454,7 @@ static int http_prepare_data(HTTPContext *c) /* should convert the format at the same time */ /* send data starting at c->buffer_ptr to the output connection - * (either UDP or TCP) */ + (either UDP or TCP connection) */ static int http_send_data(HTTPContext *c) { int len, ret; @@ -2635,26 +2577,19 @@ static int http_send_data(HTTPContext *c) static int http_start_receive_data(HTTPContext *c) { int fd; - int ret; - if (c->stream->feed_opened) { - http_log("Stream feed '%s' was not opened\n", c->stream->feed_filename); - return AVERROR(EINVAL); - } + if (c->stream->feed_opened) + return -1; /* Don't permit writing to this one */ - if (c->stream->readonly) { - http_log("Cannot write to read-only file '%s'\n", c->stream->feed_filename); - return AVERROR(EINVAL); - } + if (c->stream->readonly) + return -1; /* open feed */ fd = open(c->stream->feed_filename, O_RDWR); if (fd < 0) { - ret = AVERROR(errno); - http_log("Could not open feed file '%s': %s\n", - c->stream->feed_filename, strerror(errno)); - return ret; + http_log("Error opening feeder file: %s\n", strerror(errno)); + return -1; } c->feed_fd = fd; @@ -2663,19 +2598,13 @@ static int http_start_receive_data(HTTPContext *c) ffm_write_write_index(c->feed_fd, FFM_PACKET_SIZE); http_log("Truncating feed file '%s'\n", c->stream->feed_filename); if (ftruncate(c->feed_fd, FFM_PACKET_SIZE) < 0) { - ret = AVERROR(errno); - http_log("Error truncating feed file '%s': %s\n", - c->stream->feed_filename, strerror(errno)); - return ret; + http_log("Error truncating feed file: %s\n", strerror(errno)); + return -1; } } else { - ret = ffm_read_write_index(fd); - if (ret < 0) { - http_log("Error reading write index from feed file '%s': %s\n", - c->stream->feed_filename, strerror(errno)); - return ret; - } else { - c->stream->feed_write_index = ret; + if ((c->stream->feed_write_index = ffm_read_write_index(fd)) < 0) { + http_log("Error reading write index from feed file: %s\n", strerror(errno)); + return -1; } } @@ -2757,11 +2686,8 @@ static int http_receive_data(HTTPContext *c) /* a packet has been received : write it in the store, except if header */ if (c->data_count > FFM_PACKET_SIZE) { - /* XXX: use llseek or url_seek - * XXX: Should probably fail? */ - if (lseek(c->feed_fd, feed->feed_write_index, SEEK_SET) == -1) - http_log("Seek to %"PRId64" failed\n", feed->feed_write_index); - + /* XXX: use llseek or url_seek */ + lseek(c->feed_fd, feed->feed_write_index, SEEK_SET); if (write(c->feed_fd, c->buffer, FFM_PACKET_SIZE) < 0) { http_log("Error writing to feed file: %s\n", strerror(errno)); goto fail; @@ -2857,9 +2783,44 @@ static void rtsp_reply_header(HTTPContext *c, enum RTSPStatusCode error_number) struct tm *tm; char buf2[32]; - str = RTSP_STATUS_CODE2STRING(error_number); - if (!str) + switch(error_number) { + case RTSP_STATUS_OK: + str = "OK"; + break; + case RTSP_STATUS_METHOD: + str = "Method Not Allowed"; + break; + case RTSP_STATUS_BANDWIDTH: + str = "Not Enough Bandwidth"; + break; + case RTSP_STATUS_SESSION: + str = "Session Not Found"; + break; + case RTSP_STATUS_STATE: + str = "Method Not Valid in This State"; + break; + case RTSP_STATUS_AGGREGATE: + str = "Aggregate operation not allowed"; + break; + case RTSP_STATUS_ONLY_AGGREGATE: + str = "Only aggregate operation allowed"; + break; + case RTSP_STATUS_TRANSPORT: + str = "Unsupported transport"; + break; + case RTSP_STATUS_INTERNAL: + str = "Internal Server Error"; + break; + case RTSP_STATUS_SERVICE: + str = "Service Unavailable"; + break; + case RTSP_STATUS_VERSION: + str = "RTSP Version not supported"; + break; + default: str = "Unknown Error"; + break; + } avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", error_number, str); avio_printf(c->pb, "CSeq: %d\r\n", c->seq); @@ -2947,9 +2908,9 @@ static int rtsp_parse_request(HTTPContext *c) else if (!strcmp(cmd, "PLAY")) rtsp_cmd_play(c, url, header); else if (!strcmp(cmd, "PAUSE")) - rtsp_cmd_interrupt(c, url, header, 1); + rtsp_cmd_pause(c, url, header); else if (!strcmp(cmd, "TEARDOWN")) - rtsp_cmd_interrupt(c, url, header, 0); + rtsp_cmd_teardown(c, url, header); else rtsp_reply_error(c, RTSP_STATUS_METHOD); @@ -2971,17 +2932,14 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer, { AVFormatContext *avc; AVStream *avs = NULL; - AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL); - AVDictionaryEntry *entry = av_dict_get(stream->metadata, "title", NULL, 0); int i; avc = avformat_alloc_context(); - if (avc == NULL || !rtp_format) { + if (avc == NULL) { return -1; } - avc->oformat = rtp_format; av_dict_set(&avc->metadata, "title", - entry ? entry->value : "No Title", 0); + stream->title[0] ? stream->title : "No Title", 0); avc->nb_streams = stream->nb_streams; if (stream->is_multicast) { snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d", @@ -3033,7 +2991,7 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url) socklen_t len; struct sockaddr_in my_addr; - /* find which URL is asked */ + /* find which url is asked */ av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url); path = path1; if (*path == '/') @@ -3051,7 +3009,7 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url) return; found: - /* prepare the media description in SDP format */ + /* prepare the media description in sdp format */ /* get the host IP */ len = sizeof(my_addr); @@ -3110,7 +3068,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, struct sockaddr_in dest_addr; RTSPActionServerSetup setup; - /* find which URL is asked */ + /* find which url is asked */ av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url); path = path1; if (*path == '/') @@ -3145,14 +3103,11 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, found: /* generate session id if needed */ - if (h->session_id[0] == '\0') { - unsigned random0 = av_lfg_get(&random_state); - unsigned random1 = av_lfg_get(&random_state); + if (h->session_id[0] == '\0') snprintf(h->session_id, sizeof(h->session_id), "%08x%08x", - random0, random1); - } + av_lfg_get(&random_state), av_lfg_get(&random_state)); - /* find RTP session, and create it if none found */ + /* find rtp session, and create it if none found */ rtp_c = find_rtp_session(h->session_id); if (!rtp_c) { /* always prefer UDP */ @@ -3241,7 +3196,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url, } -/* find an RTP connection by using the session ID. Check consistency +/* find an rtp connection by using the session ID. Check consistency with filename */ static HTTPContext *find_rtp_session_with_url(const char *url, const char *session_id) @@ -3256,7 +3211,7 @@ static HTTPContext *find_rtp_session_with_url(const char *url, if (!rtp_c) return NULL; - /* find which URL is asked */ + /* find which url is asked */ av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url); path = path1; if (*path == '/') @@ -3303,7 +3258,7 @@ static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPMessageHeader *h) avio_printf(c->pb, "\r\n"); } -static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeader *h, int pause_only) +static void rtsp_cmd_pause(HTTPContext *c, const char *url, RTSPMessageHeader *h) { HTTPContext *rtp_c; @@ -3313,14 +3268,29 @@ static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeade return; } - if (pause_only) { - if (rtp_c->state != HTTPSTATE_SEND_DATA && - rtp_c->state != HTTPSTATE_WAIT_FEED) { - rtsp_reply_error(c, RTSP_STATUS_STATE); - return; - } - rtp_c->state = HTTPSTATE_READY; - rtp_c->first_pts = AV_NOPTS_VALUE; + if (rtp_c->state != HTTPSTATE_SEND_DATA && + rtp_c->state != HTTPSTATE_WAIT_FEED) { + rtsp_reply_error(c, RTSP_STATUS_STATE); + return; + } + + rtp_c->state = HTTPSTATE_READY; + rtp_c->first_pts = AV_NOPTS_VALUE; + /* now everything is OK, so we can send the connection parameters */ + rtsp_reply_header(c, RTSP_STATUS_OK); + /* session ID */ + avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id); + avio_printf(c->pb, "\r\n"); +} + +static void rtsp_cmd_teardown(HTTPContext *c, const char *url, RTSPMessageHeader *h) +{ + HTTPContext *rtp_c; + + rtp_c = find_rtp_session_with_url(url, h->session_id); + if (!rtp_c) { + rtsp_reply_error(c, RTSP_STATUS_SESSION); + return; } /* now everything is OK, so we can send the connection parameters */ @@ -3329,10 +3299,11 @@ static void rtsp_cmd_interrupt(HTTPContext *c, const char *url, RTSPMessageHeade avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id); avio_printf(c->pb, "\r\n"); - if (!pause_only) - close_connection(rtp_c); + /* abort the session */ + close_connection(rtp_c); } + /********************************************************************/ /* RTP handling */ @@ -3477,8 +3448,7 @@ static int rtp_new_av_stream(HTTPContext *c, ipaddr, ntohs(dest_addr->sin_port), c->stream->filename, stream_index, c->protocol); - /* normally, no packets should be output here, but the packet size may - * be checked */ + /* normally, no packets should be output here, but the packet size may be checked */ if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) { /* XXX: close stream */ goto fail; @@ -3487,7 +3457,6 @@ static int rtp_new_av_stream(HTTPContext *c, fail: if (h) ffurl_close(h); - av_free(st); av_free(ctx); return -1; } @@ -3499,15 +3468,12 @@ static int rtp_new_av_stream(HTTPContext *c, } /********************************************************************/ -/* ffserver initialization */ +/* avserver initialization */ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int copy) { AVStream *fst; - if(stream->nb_streams >= FF_ARRAY_ELEMS(stream->streams)) - return NULL; - fst = av_mallocz(sizeof(AVStream)); if (!fst) return NULL; @@ -3515,13 +3481,13 @@ static AVStream *add_av_stream1(FFStream *stream, AVCodecContext *codec, int cop fst->codec = avcodec_alloc_context3(NULL); memcpy(fst->codec, codec, sizeof(AVCodecContext)); if (codec->extradata_size) { - fst->codec->extradata = av_mallocz(codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE); + fst->codec->extradata = av_malloc(codec->extradata_size); memcpy(fst->codec->extradata, codec->extradata, codec->extradata_size); } } else { /* live streams must use the actual feed's codec since it may be - * updated later to carry extradata needed by them. + * updated later to carry extradata needed by the streams. */ fst->codec = codec; } @@ -3586,7 +3552,7 @@ static void remove_stream(FFStream *stream) } } -/* specific MPEG4 handling : we extract the raw parameters */ +/* specific mpeg4 handling : we extract the raw parameters */ static void extract_mpeg4_header(AVFormatContext *infile) { int mpeg4_count, i, size; @@ -3624,7 +3590,7 @@ static void extract_mpeg4_header(AVFormatContext *infile) p[2] == 0x01 && p[3] == 0xb6) { size = p - pkt.data; // av_hex_dump_log(infile, AV_LOG_DEBUG, pkt.data, size); - st->codec->extradata = av_mallocz(size + FF_INPUT_BUFFER_PADDING_SIZE); + st->codec->extradata = av_malloc(size); st->codec->extradata_size = size; memcpy(st->codec->extradata, pkt.data, size); break; @@ -3658,14 +3624,9 @@ static void build_file_streams(void) av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0); } - if (!stream->feed_filename[0]) { - http_log("Unspecified feed file for stream '%s'\n", stream->filename); - goto fail; - } - - http_log("Opening feed file '%s' for stream '%s'\n", stream->feed_filename, stream->filename); + http_log("Opening file '%s'\n", stream->feed_filename); if ((ret = avformat_open_input(&infile, stream->feed_filename, stream->ifmt, &stream->in_opts)) < 0) { - http_log("Could not open '%s': %s\n", stream->feed_filename, av_err2str(ret)); + http_log("Could not open '%s': %d\n", stream->feed_filename, ret); /* remove stream (no need to spend more time on it) */ fail: remove_stream(stream); @@ -3720,8 +3681,6 @@ static void build_feed_streams(void) int matches = 0; if (avformat_open_input(&s, feed->feed_filename, NULL, NULL) >= 0) { - /* set buffer size */ - ffio_set_buf_size(s->pb, FFM_PACKET_SIZE); /* Now see if it matches */ if (s->nb_streams == feed->nb_streams) { matches = 1; @@ -3790,7 +3749,7 @@ static void build_feed_streams(void) } } if (avio_check(feed->feed_filename, AVIO_FLAG_WRITE) <= 0) { - AVFormatContext *s = avformat_alloc_context(); + AVFormatContext s1 = {0}, *s = &s1; if (feed->readonly) { http_log("Unable to create feed file '%s' as it is marked readonly\n", @@ -3808,15 +3767,12 @@ static void build_feed_streams(void) s->nb_streams = feed->nb_streams; s->streams = feed->streams; if (avformat_write_header(s, NULL) < 0) { - http_log("Container doesn't support the required parameters\n"); + http_log("Container doesn't supports the required parameters\n"); exit(1); } - /* XXX: need better API */ + /* XXX: need better api */ av_freep(&s->priv_data); avio_close(s->pb); - s->streams = NULL; - s->nb_streams = 0; - avformat_free_context(s); } /* get feed size and write index */ fd = open(feed->feed_filename, O_RDONLY); @@ -3865,9 +3821,6 @@ static void add_codec(FFStream *stream, AVCodecContext *av) { AVStream *st; - if(stream->nb_streams >= FF_ARRAY_ELEMS(stream->streams)) - return; - /* compute default parameters */ switch(av->codec_type) { case AVMEDIA_TYPE_AUDIO: @@ -3911,7 +3864,7 @@ static void add_codec(FFStream *stream, AVCodecContext *av) av->rc_buffer_aggressivity = 1.0; if (!av->rc_eq) - av->rc_eq = av_strdup("tex^qComp"); + av->rc_eq = "tex^qComp"; if (!av->i_quant_factor) av->i_quant_factor = -0.8; if (!av->b_quant_factor) @@ -3939,16 +3892,27 @@ static void add_codec(FFStream *stream, AVCodecContext *av) memcpy(st->codec, av, sizeof(AVCodecContext)); } -static enum AVCodecID opt_codec(const char *name, enum AVMediaType type) +static enum AVCodecID opt_audio_codec(const char *arg) { - AVCodec *codec = avcodec_find_encoder_by_name(name); + AVCodec *p= avcodec_find_encoder_by_name(arg); - if (!codec || codec->type != type) + if (p == NULL || p->type != AVMEDIA_TYPE_AUDIO) return AV_CODEC_ID_NONE; - return codec->id; + + return p->id; } -static int ffserver_opt_default(const char *opt, const char *arg, +static enum AVCodecID opt_video_codec(const char *arg) +{ + AVCodec *p= avcodec_find_encoder_by_name(arg); + + if (p == NULL || p->type != AVMEDIA_TYPE_VIDEO) + return AV_CODEC_ID_NONE; + + return p->id; +} + +static int avserver_opt_default(const char *opt, const char *arg, AVCodecContext *avctx, int type) { int ret = 0; @@ -3958,7 +3922,7 @@ static int ffserver_opt_default(const char *opt, const char *arg, return ret; } -static int ffserver_opt_preset(const char *arg, +static int avserver_opt_preset(const char *arg, AVCodecContext *avctx, int type, enum AVCodecID *audio_id, enum AVCodecID *video_id) { @@ -3984,12 +3948,12 @@ static int ffserver_opt_preset(const char *arg, break; } if(!strcmp(tmp, "acodec")){ - *audio_id = opt_codec(tmp2, AVMEDIA_TYPE_AUDIO); + *audio_id = opt_audio_codec(tmp2); }else if(!strcmp(tmp, "vcodec")){ - *video_id = opt_codec(tmp2, AVMEDIA_TYPE_VIDEO); + *video_id = opt_video_codec(tmp2); }else if(!strcmp(tmp, "scodec")){ /* opt_subtitle_codec(tmp2); */ - }else if(ffserver_opt_default(tmp, tmp2, avctx, type) < 0){ + }else if(avserver_opt_default(tmp, tmp2, avctx, type) < 0){ fprintf(stderr, "%s: Invalid option or argument: '%s', parsed as '%s' = '%s'\n", filename, line, tmp, tmp2); ret = 1; break; @@ -4001,7 +3965,8 @@ static int ffserver_opt_preset(const char *arg, return ret; } -static AVOutputFormat *ffserver_guess_format(const char *short_name, const char *filename, const char *mime_type) +static AVOutputFormat *avserver_guess_format(const char *short_name, const char *filename, + const char *mime_type) { AVOutputFormat *fmt = av_guess_format(short_name, filename, mime_type); @@ -4019,12 +3984,12 @@ static AVOutputFormat *ffserver_guess_format(const char *short_name, const char return fmt; } -static void report_config_error(const char *filename, int line_num, int log_level, int *errors, const char *fmt, ...) +static void report_config_error(const char *filename, int line_num, int *errors, const char *fmt, ...) { va_list vl; va_start(vl, fmt); - av_log(NULL, log_level, "%s:%d: ", filename, line_num); - av_vlog(NULL, log_level, fmt, vl); + fprintf(stderr, "%s:%d: ", filename, line_num); + vfprintf(stderr, fmt, vl); va_end(vl); (*errors)++; @@ -4035,23 +4000,21 @@ static int parse_ffconfig(const char *filename) FILE *f; char line[1024]; char cmd[64]; - char arg[1024], arg2[1024]; + char arg[1024]; const char *p; - int val, errors, warnings, line_num; + int val, errors, line_num; FFStream **last_stream, *stream, *redirect; FFStream **last_feed, *feed, *s; AVCodecContext audio_enc, video_enc; enum AVCodecID audio_id, video_id; - int ret = 0; f = fopen(filename, "r"); if (!f) { - ret = AVERROR(errno); - av_log(NULL, AV_LOG_ERROR, "Could not open the configuration file '%s'\n", filename); - return ret; + perror(filename); + return -1; } - errors = warnings = 0; + errors = 0; line_num = 0; first_stream = NULL; last_stream = &first_stream; @@ -4062,9 +4025,8 @@ static int parse_ffconfig(const char *filename) redirect = NULL; audio_id = AV_CODEC_ID_NONE; video_id = AV_CODEC_ID_NONE; -#define ERROR(...) report_config_error(filename, line_num, AV_LOG_ERROR, &errors, __VA_ARGS__) -#define WARNING(...) report_config_error(filename, line_num, AV_LOG_WARNING, &warnings, __VA_ARGS__) +#define ERROR(...) report_config_error(filename, line_num, &errors, __VA_ARGS__) for(;;) { if (fgets(line, sizeof(line), f) == NULL) break; @@ -4089,8 +4051,6 @@ static int parse_ffconfig(const char *filename) if (resolve_host(&my_http_addr.sin_addr, arg) != 0) { ERROR("%s:%d: Invalid host/IP address: %s\n", arg); } - } else if (!av_strcasecmp(cmd, "NoDaemon")) { - WARNING("NoDaemon option has no effect, you should remove it\n"); } else if (!av_strcasecmp(cmd, "RTSPPort")) { get_arg(arg, sizeof(arg), &p); val = atoi(arg); @@ -4121,13 +4081,13 @@ static int parse_ffconfig(const char *filename) } else if (!av_strcasecmp(cmd, "MaxBandwidth")) { int64_t llval; get_arg(arg, sizeof(arg), &p); - llval = strtoll(arg, NULL, 10); + llval = atoll(arg); if (llval < 10 || llval > 10000000) { ERROR("Invalid MaxBandwidth: %s\n", arg); } else max_bandwidth = llval; } else if (!av_strcasecmp(cmd, "CustomLog")) { - if (!ffserver_debug) + if (!avserver_debug) get_arg(logfilename, sizeof(logfilename), &p); } else if (!av_strcasecmp(cmd, "filename, sizeof(feed->filename), &p); q = strrchr(feed->filename, '>'); if (*q) @@ -4153,7 +4109,7 @@ static int parse_ffconfig(const char *filename) } feed->fmt = av_guess_format("ffm", NULL, NULL); - /* default feed file */ + /* defaut feed file */ snprintf(feed->feed_filename, sizeof(feed->feed_filename), "/tmp/%s.ffm", feed->filename); feed->feed_max_size = 5 * 1024 * 1024; @@ -4172,50 +4128,39 @@ static int parse_ffconfig(const char *filename) int i; feed->child_argv = av_mallocz(64 * sizeof(char *)); - if (!feed->child_argv) { - ret = AVERROR(ENOMEM); - goto end; - } + for (i = 0; i < 62; i++) { get_arg(arg, sizeof(arg), &p); if (!arg[0]) break; feed->child_argv[i] = av_strdup(arg); - if (!feed->child_argv[i]) { - ret = AVERROR(ENOMEM); - goto end; - } } - feed->child_argv[i] = - av_asprintf("http://%s:%d/%s", - (my_http_addr.sin_addr.s_addr == INADDR_ANY) ? "127.0.0.1" : - inet_ntoa(my_http_addr.sin_addr), ntohs(my_http_addr.sin_port), - feed->filename); - if (!feed->child_argv[i]) { - ret = AVERROR(ENOMEM); - goto end; - } + feed->child_argv[i] = av_malloc(30 + strlen(feed->filename)); + + snprintf(feed->child_argv[i], 30+strlen(feed->filename), + "http://%s:%d/%s", + (my_http_addr.sin_addr.s_addr == INADDR_ANY) ? "127.0.0.1" : + inet_ntoa(my_http_addr.sin_addr), + ntohs(my_http_addr.sin_port), feed->filename); } - } else if (!av_strcasecmp(cmd, "File") || !av_strcasecmp(cmd, "ReadOnlyFile")) { + } else if (!av_strcasecmp(cmd, "ReadOnlyFile")) { if (feed) { get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p); - feed->readonly = !av_strcasecmp(cmd, "ReadOnlyFile"); + feed->readonly = 1; } else if (stream) { get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p); } + } else if (!av_strcasecmp(cmd, "File")) { + if (feed) { + get_arg(feed->feed_filename, sizeof(feed->feed_filename), &p); + } else if (stream) + get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p); } else if (!av_strcasecmp(cmd, "Truncate")) { if (feed) { get_arg(arg, sizeof(arg), &p); - /* assume Truncate is true in case no argument is specified */ - if (!arg[0]) { - feed->truncate = 1; - } else { - WARNING("Truncate N syntax in configuration file is deprecated, " - "use Truncate alone with no arguments\n"); - feed->truncate = strtod(arg, NULL); - } + feed->truncate = strtod(arg, NULL); } } else if (!av_strcasecmp(cmd, "FileMaxSize")) { if (feed) { @@ -4255,13 +4200,9 @@ static int parse_ffconfig(const char *filename) } else { FFStream *s; stream = av_mallocz(sizeof(FFStream)); - if (!stream) { - ret = AVERROR(ENOMEM); - goto end; - } get_arg(stream->filename, sizeof(stream->filename), &p); q = strrchr(stream->filename, '>'); - if (q) + if (*q) *q = '\0'; for (s = first_stream; s; s = s->next) { @@ -4270,10 +4211,9 @@ static int parse_ffconfig(const char *filename) } } - stream->fmt = ffserver_guess_format(NULL, stream->filename, NULL); + stream->fmt = avserver_guess_format(NULL, stream->filename, NULL); avcodec_get_context_defaults3(&video_enc, NULL); avcodec_get_context_defaults3(&audio_enc, NULL); - audio_id = AV_CODEC_ID_NONE; video_id = AV_CODEC_ID_NONE; if (stream->fmt) { @@ -4296,7 +4236,7 @@ static int parse_ffconfig(const char *filename) sfeed = sfeed->next_feed; } if (!sfeed) - ERROR("Feed with name '%s' for stream '%s' is not defined\n", arg, stream->filename); + ERROR("feed '%s' not defined\n", arg); else stream->feed = sfeed; } @@ -4308,10 +4248,10 @@ static int parse_ffconfig(const char *filename) stream->fmt = NULL; } else { stream->stream_type = STREAM_TYPE_LIVE; - /* JPEG cannot be used here, so use single frame MJPEG */ + /* jpeg cannot be used here, so use single frame jpeg */ if (!strcmp(arg, "jpeg")) strcpy(arg, "mjpeg"); - stream->fmt = ffserver_guess_format(arg, NULL, NULL); + stream->fmt = avserver_guess_format(arg, NULL, NULL); if (!stream->fmt) { ERROR("Unknown Format: %s\n", arg); } @@ -4335,36 +4275,18 @@ static int parse_ffconfig(const char *filename) } else { ERROR("FaviconURL only permitted for status streams\n"); } - } else if (!av_strcasecmp(cmd, "Author") || - !av_strcasecmp(cmd, "Comment") || - !av_strcasecmp(cmd, "Copyright") || - !av_strcasecmp(cmd, "Title")) { - get_arg(arg, sizeof(arg), &p); - - if (stream) { - char key[32]; - int i, ret; - - for (i = 0; i < strlen(cmd); i++) - key[i] = av_tolower(cmd[i]); - key[i] = 0; - WARNING("'%s' option in configuration file is deprecated, " - "use 'Metadata %s VALUE' instead\n", cmd, key); - if ((ret = av_dict_set(&stream->metadata, key, arg, 0)) < 0) { - ERROR("Could not set metadata '%s' to value '%s': %s\n", - key, arg, av_err2str(ret)); - } - } - } else if (!av_strcasecmp(cmd, "Metadata")) { - get_arg(arg, sizeof(arg), &p); - get_arg(arg2, sizeof(arg2), &p); - if (stream) { - int ret; - if ((ret = av_dict_set(&stream->metadata, arg, arg2, 0)) < 0) { - ERROR("Could not set metadata '%s' to value '%s': %s\n", - arg, arg2, av_err2str(ret)); - } - } + } else if (!av_strcasecmp(cmd, "Author")) { + if (stream) + get_arg(stream->author, sizeof(stream->author), &p); + } else if (!av_strcasecmp(cmd, "Comment")) { + if (stream) + get_arg(stream->comment, sizeof(stream->comment), &p); + } else if (!av_strcasecmp(cmd, "Copyright")) { + if (stream) + get_arg(stream->copyright, sizeof(stream->copyright), &p); + } else if (!av_strcasecmp(cmd, "Title")) { + if (stream) + get_arg(stream->title, sizeof(stream->title), &p); } else if (!av_strcasecmp(cmd, "Preroll")) { get_arg(arg, sizeof(arg), &p); if (stream) @@ -4374,13 +4296,13 @@ static int parse_ffconfig(const char *filename) stream->send_on_key = 1; } else if (!av_strcasecmp(cmd, "AudioCodec")) { get_arg(arg, sizeof(arg), &p); - audio_id = opt_codec(arg, AVMEDIA_TYPE_AUDIO); + audio_id = opt_audio_codec(arg); if (audio_id == AV_CODEC_ID_NONE) { ERROR("Unknown AudioCodec: %s\n", arg); } } else if (!av_strcasecmp(cmd, "VideoCodec")) { get_arg(arg, sizeof(arg), &p); - video_id = opt_codec(arg, AVMEDIA_TYPE_VIDEO); + video_id = opt_video_codec(arg); if (video_id == AV_CODEC_ID_NONE) { ERROR("Unknown VideoCodec: %s\n", arg); } @@ -4400,6 +4322,11 @@ static int parse_ffconfig(const char *filename) get_arg(arg, sizeof(arg), &p); if (stream) audio_enc.sample_rate = atoi(arg); + } else if (!av_strcasecmp(cmd, "AudioQuality")) { + get_arg(arg, sizeof(arg), &p); + if (stream) { +// audio_enc.quality = atof(arg) * 1000; + } } else if (!av_strcasecmp(cmd, "VideoBitRateRange")) { if (stream) { int minrate, maxrate; @@ -4441,14 +4368,10 @@ static int parse_ffconfig(const char *filename) } else if (!av_strcasecmp(cmd, "VideoSize")) { get_arg(arg, sizeof(arg), &p); if (stream) { - ret = av_parse_video_size(&video_enc.width, &video_enc.height, arg); - if (ret < 0) { - ERROR("Invalid video size '%s'\n", arg); - } else { - if ((video_enc.width % 16) != 0 || - (video_enc.height % 16) != 0) { - ERROR("Image size must be a multiple of 16\n"); - } + av_parse_video_size(&video_enc.width, &video_enc.height, arg); + if ((video_enc.width % 16) != 0 || + (video_enc.height % 16) != 0) { + ERROR("Image size must be a multiple of 16\n"); } } } else if (!av_strcasecmp(cmd, "VideoFrameRate")) { @@ -4462,14 +4385,6 @@ static int parse_ffconfig(const char *filename) video_enc.time_base.den = frame_rate.num; } } - } else if (!av_strcasecmp(cmd, "PixelFormat")) { - get_arg(arg, sizeof(arg), &p); - if (stream) { - video_enc.pix_fmt = av_get_pix_fmt(arg); - if (video_enc.pix_fmt == AV_PIX_FMT_NONE) { - ERROR("Unknown pixel format: %s\n", arg); - } - } } else if (!av_strcasecmp(cmd, "VideoGopSize")) { get_arg(arg, sizeof(arg), &p); if (stream) @@ -4487,6 +4402,7 @@ static int parse_ffconfig(const char *filename) } } else if (!av_strcasecmp(cmd, "AVOptionVideo") || !av_strcasecmp(cmd, "AVOptionAudio")) { + char arg2[1024]; AVCodecContext *avctx; int type; get_arg(arg, sizeof(arg), &p); @@ -4498,8 +4414,8 @@ static int parse_ffconfig(const char *filename) avctx = &audio_enc; type = AV_OPT_FLAG_AUDIO_PARAM; } - if (ffserver_opt_default(arg, arg2, avctx, type|AV_OPT_FLAG_ENCODING_PARAM)) { - ERROR("Error setting %s option to %s %s\n", cmd, arg, arg2); + if (avserver_opt_default(arg, arg2, avctx, type|AV_OPT_FLAG_ENCODING_PARAM)) { + ERROR("AVOption error: %s %s\n", arg, arg2); } } else if (!av_strcasecmp(cmd, "AVPresetVideo") || !av_strcasecmp(cmd, "AVPresetAudio")) { @@ -4515,7 +4431,7 @@ static int parse_ffconfig(const char *filename) audio_enc.codec_id = audio_id; type = AV_OPT_FLAG_AUDIO_PARAM; } - if (ffserver_opt_preset(arg, avctx, type|AV_OPT_FLAG_ENCODING_PARAM, &audio_id, &video_id)) { + if (avserver_opt_preset(arg, avctx, type|AV_OPT_FLAG_ENCODING_PARAM, &audio_id, &video_id)) { ERROR("AVPreset error: %s\n", arg); } } else if (!av_strcasecmp(cmd, "VideoTag")) { @@ -4630,10 +4546,6 @@ static int parse_ffconfig(const char *filename) ERROR("Already in a tag\n"); } else { redirect = av_mallocz(sizeof(FFStream)); - if (!redirect) { - ret = AVERROR(ENOMEM); - goto end; - } *last_stream = redirect; last_stream = &redirect->next; @@ -4663,12 +4575,9 @@ static int parse_ffconfig(const char *filename) } #undef ERROR -end: fclose(f); - if (ret < 0) - return ret; if (errors) - return AVERROR(EINVAL); + return -1; else return 0; } @@ -4700,13 +4609,13 @@ static void handle_child_exit(int sig) static void opt_debug(void) { - ffserver_debug = 1; + avserver_debug = 1; logfilename[0] = '-'; } void show_help_default(const char *opt, const char *arg) { - printf("usage: ffserver [options]\n" + printf("usage: avserver [options]\n" "Hyper fast multi format Audio/Video streaming server\n"); printf("\n"); show_help_options(options, "Main options:", 0, 0, 0); @@ -4716,22 +4625,21 @@ static const OptionDef options[] = { #include "cmdutils_common_opts.h" { "n", OPT_BOOL, {(void *)&no_launch }, "enable no-launch mode" }, { "d", 0, {(void*)opt_debug}, "enable debug mode" }, - { "f", HAS_ARG | OPT_STRING, {(void*)&config_filename }, "use configfile instead of /etc/ffserver.conf", "configfile" }, + { "f", HAS_ARG | OPT_STRING, {(void*)&config_filename }, "use configfile instead of /etc/avserver.conf", "configfile" }, { NULL }, }; int main(int argc, char **argv) { struct sigaction sigact = { { 0 } }; - int ret = 0; - config_filename = av_strdup("/etc/ffserver.conf"); + config_filename = av_strdup("/etc/avserver.conf"); parse_loglevel(argc, argv, options); av_register_all(); avformat_network_init(); - show_banner(argc, argv, options); + show_banner(); my_program_name = argv[0]; @@ -4745,12 +4653,10 @@ int main(int argc, char **argv) sigact.sa_flags = SA_NOCLDSTOP | SA_RESTART; sigaction(SIGCHLD, &sigact, 0); - if ((ret = parse_ffconfig(config_filename)) < 0) { - fprintf(stderr, "Error reading configuration file '%s': %s\n", - config_filename, av_err2str(ret)); + if (parse_ffconfig(config_filename) < 0) { + fprintf(stderr, "Incorrect config file - exiting.\n"); exit(1); } - av_freep(&config_filename); /* open log file if needed */ if (logfilename[0] != '\0') { diff --git a/cmdutils.c b/cmdutils.c index 1147bc338e..69a11bdd4f 100644 --- a/cmdutils.c +++ b/cmdutils.c @@ -2,20 +2,20 @@ * Various utilities for command line tools * Copyright (c) 2000-2003 Fabrice Bellard * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -30,17 +30,13 @@ references to libraries that are not being built. */ #include "config.h" -#include "compat/va_copy.h" #include "libavformat/avformat.h" #include "libavfilter/avfilter.h" #include "libavdevice/avdevice.h" #include "libavresample/avresample.h" #include "libswscale/swscale.h" -#include "libswresample/swresample.h" -#include "libpostproc/postprocess.h" #include "libavutil/avassert.h" #include "libavutil/avstring.h" -#include "libavutil/bprint.h" #include "libavutil/mathematics.h" #include "libavutil/imgutils.h" #include "libavutil/parseutils.h" @@ -49,8 +45,8 @@ #include "libavutil/dict.h" #include "libavutil/opt.h" #include "libavutil/cpu.h" -#include "libavutil/ffversion.h" #include "cmdutils.h" +#include "version.h" #if CONFIG_NETWORK #include "libavformat/network.h" #endif @@ -59,21 +55,17 @@ #include #endif -static int init_report(const char *env); - struct SwsContext *sws_opts; -AVDictionary *swr_opts; AVDictionary *format_opts, *codec_opts, *resample_opts; -static FILE *report_file; -int hide_banner = 0; +static const int this_year = 2014; void init_opts(void) { - - if(CONFIG_SWSCALE) - sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC, +#if CONFIG_SWSCALE + sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC, NULL, NULL, NULL); +#endif } void uninit_opts(void) @@ -82,8 +74,6 @@ void uninit_opts(void) sws_freeContext(sws_opts); sws_opts = NULL; #endif - - av_dict_free(&swr_opts); av_dict_free(&format_opts); av_dict_free(&codec_opts); av_dict_free(&resample_opts); @@ -94,20 +84,6 @@ void log_callback_help(void *ptr, int level, const char *fmt, va_list vl) vfprintf(stdout, fmt, vl); } -static void log_callback_report(void *ptr, int level, const char *fmt, va_list vl) -{ - va_list vl2; - char line[1024]; - static int print_prefix = 1; - - va_copy(vl2, vl); - av_log_default_callback(ptr, level, fmt, vl); - av_log_format_line(ptr, level, fmt, vl2, line, sizeof(line), &print_prefix); - va_end(vl2); - fputs(line, report_file); - fflush(report_file); -} - static void (*program_exit)(int ret); void register_exit(void (*cb)(int ret)) @@ -188,10 +164,8 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags, void show_help_children(const AVClass *class, int flags) { const AVClass *child = NULL; - if (class->option) { - av_opt_show2(&class, NULL, flags, 0); - printf("\n"); - } + av_opt_show2(&class, NULL, flags, 0); + printf("\n"); while (child = av_opt_child_class_next(class, child)) show_help_children(child, flags); @@ -313,8 +287,7 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt, int ret = po->u.func_arg(optctx, opt, arg); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, - "Failed to set value '%s' for option '%s': %s\n", - arg, opt, av_err2str(ret)); + "Failed to set value '%s' for option '%s'\n", arg, opt); return ret; } } @@ -441,142 +414,65 @@ int locate_option(int argc, char **argv, const OptionDef *options, (po->name && !strcmp(optname, po->name))) return i; - if (po->flags & HAS_ARG) + if (!po || po->flags & HAS_ARG) i++; } return 0; } -static void dump_argument(const char *a) -{ - const unsigned char *p; - - for (p = a; *p; p++) - if (!((*p >= '+' && *p <= ':') || (*p >= '@' && *p <= 'Z') || - *p == '_' || (*p >= 'a' && *p <= 'z'))) - break; - if (!*p) { - fputs(a, report_file); - return; - } - fputc('"', report_file); - for (p = a; *p; p++) { - if (*p == '\\' || *p == '"' || *p == '$' || *p == '`') - fprintf(report_file, "\\%c", *p); - else if (*p < ' ' || *p > '~') - fprintf(report_file, "\\x%02x", *p); - else - fputc(*p, report_file); - } - fputc('"', report_file); -} - void parse_loglevel(int argc, char **argv, const OptionDef *options) { int idx = locate_option(argc, argv, options, "loglevel"); - const char *env; if (!idx) idx = locate_option(argc, argv, options, "v"); if (idx && argv[idx + 1]) opt_loglevel(NULL, "loglevel", argv[idx + 1]); - idx = locate_option(argc, argv, options, "report"); - if ((env = getenv("FFREPORT")) || idx) { - init_report(env); - if (report_file) { - int i; - fprintf(report_file, "Command line:\n"); - for (i = 0; i < argc; i++) { - dump_argument(argv[i]); - fputc(i < argc - 1 ? ' ' : '\n', report_file); - } - fflush(report_file); - } - } - idx = locate_option(argc, argv, options, "hide_banner"); - if (idx) - hide_banner = 1; -} - -static const AVOption *opt_find(void *obj, const char *name, const char *unit, - int opt_flags, int search_flags) -{ - const AVOption *o = av_opt_find(obj, name, unit, opt_flags, search_flags); - if(o && !o->flags) - return NULL; - return o; } #define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0 int opt_default(void *optctx, const char *opt, const char *arg) { const AVOption *o; - int consumed = 0; char opt_stripped[128]; const char *p; const AVClass *cc = avcodec_get_class(), *fc = avformat_get_class(); #if CONFIG_AVRESAMPLE const AVClass *rc = avresample_get_class(); #endif - const AVClass *sc, *swr_class; - - if (!strcmp(opt, "debug") || !strcmp(opt, "fdebug")) - av_log_set_level(AV_LOG_DEBUG); +#if CONFIG_SWSCALE + const AVClass *sc = sws_get_class(); +#endif if (!(p = strchr(opt, ':'))) p = opt + strlen(opt); av_strlcpy(opt_stripped, opt, FFMIN(sizeof(opt_stripped), p - opt + 1)); - if ((o = opt_find(&cc, opt_stripped, NULL, 0, + if ((o = av_opt_find(&cc, opt_stripped, NULL, 0, AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) || ((opt[0] == 'v' || opt[0] == 'a' || opt[0] == 's') && - (o = opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) { + (o = av_opt_find(&cc, opt + 1, NULL, 0, AV_OPT_SEARCH_FAKE_OBJ)))) av_dict_set(&codec_opts, opt, arg, FLAGS); - consumed = 1; - } - if ((o = opt_find(&fc, opt, NULL, 0, - AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) { + else if ((o = av_opt_find(&fc, opt, NULL, 0, + AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) av_dict_set(&format_opts, opt, arg, FLAGS); - if (consumed) - av_log(NULL, AV_LOG_VERBOSE, "Routing option %s to both codec and muxer layer\n", opt); - consumed = 1; - } +#if CONFIG_AVRESAMPLE + else if ((o = av_opt_find(&rc, opt, NULL, 0, + AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) + av_dict_set(&resample_opts, opt, arg, FLAGS); +#endif #if CONFIG_SWSCALE - sc = sws_get_class(); - if (!consumed && opt_find(&sc, opt, NULL, 0, - AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) { + else if ((o = av_opt_find(&sc, opt, NULL, 0, + AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) { // XXX we only support sws_flags, not arbitrary sws options int ret = av_opt_set(sws_opts, opt, arg, 0); if (ret < 0) { av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt); return ret; } - consumed = 1; - } -#endif -#if CONFIG_SWRESAMPLE - swr_class = swr_get_class(); - if (!consumed && (o=opt_find(&swr_class, opt, NULL, 0, - AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) { - struct SwrContext *swr = swr_alloc(); - int ret = av_opt_set(swr, opt, arg, 0); - swr_free(&swr); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt); - return ret; - } - av_dict_set(&swr_opts, opt, arg, FLAGS); - consumed = 1; - } -#endif -#if CONFIG_AVRESAMPLE - if ((o=opt_find(&rc, opt, NULL, 0, - AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) { - av_dict_set(&resample_opts, opt, arg, FLAGS); - consumed = 1; } #endif - if (consumed) + if (o) return 0; return AVERROR_OPTION_NOT_FOUND; } @@ -621,7 +517,6 @@ static void finish_group(OptionParseContext *octx, int group_idx, #if CONFIG_SWSCALE g->sws_opts = sws_opts; #endif - g->swr_opts = swr_opts; g->codec_opts = codec_opts; g->format_opts = format_opts; g->resample_opts = resample_opts; @@ -632,7 +527,6 @@ static void finish_group(OptionParseContext *octx, int group_idx, #if CONFIG_SWSCALE sws_opts = NULL; #endif - swr_opts = NULL; init_opts(); memset(&octx->cur_group, 0, sizeof(octx->cur_group)); @@ -662,7 +556,7 @@ static void init_parse_context(OptionParseContext *octx, memset(octx, 0, sizeof(*octx)); octx->nb_groups = nb_groups; - octx->groups = av_mallocz_array(octx->nb_groups, sizeof(*octx->groups)); + octx->groups = av_mallocz(sizeof(*octx->groups) * octx->nb_groups); if (!octx->groups) exit_program(1); @@ -690,7 +584,6 @@ void uninit_parse_context(OptionParseContext *octx) #if CONFIG_SWSCALE sws_freeContext(l->groups[j].sws_opts); #endif - av_dict_free(&l->groups[j].swr_opts); } av_freep(&l->groups); } @@ -707,7 +600,6 @@ int split_commandline(OptionParseContext *octx, int argc, char *argv[], const OptionGroupDef *groups, int nb_groups) { int optindex = 1; - int dashdash = -2; /* perform system-dependent conversions for arguments list */ prepare_app_arguments(&argc, &argv); @@ -722,12 +614,8 @@ int split_commandline(OptionParseContext *octx, int argc, char *argv[], av_log(NULL, AV_LOG_DEBUG, "Reading option '%s' ...", opt); - if (opt[0] == '-' && opt[1] == '-' && !opt[2]) { - dashdash = optindex; - continue; - } /* unnamed group separators, e.g. output filename */ - if (opt[0] != '-' || !opt[1] || dashdash+1 == optindex) { + if (opt[0] != '-' || !opt[1]) { finish_group(octx, 0, opt); av_log(NULL, AV_LOG_DEBUG, " matched as %s.\n", groups[0].name); continue; @@ -810,13 +698,12 @@ do { \ int opt_cpuflags(void *optctx, const char *opt, const char *arg) { - int ret; - unsigned flags = av_get_cpu_flags(); + int flags = av_parse_cpu_flags(arg); - if ((ret = av_parse_cpu_caps(&flags, arg)) < 0) - return ret; + if (flags < 0) + return flags; - av_force_cpu_flags(flags); + av_set_cpu_flags_mask(flags); return 0; } @@ -834,22 +721,8 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg) }; char *tail; int level; - int flags; int i; - flags = av_log_get_flags(); - tail = strstr(arg, "repeat"); - if (tail) - flags &= ~AV_LOG_SKIP_REPEATED; - else - flags |= AV_LOG_SKIP_REPEATED; - - av_log_set_flags(flags); - if (tail == arg) - arg += 6 + (arg[6]=='+'); - if(tail && !*arg) - return 0; - for (i = 0; i < FF_ARRAY_ELEMS(log_levels); i++) { if (!strcmp(log_levels[i].name, arg)) { av_log_set_level(log_levels[i].level); @@ -869,116 +742,6 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg) return 0; } -static void expand_filename_template(AVBPrint *bp, const char *template, - struct tm *tm) -{ - int c; - - while ((c = *(template++))) { - if (c == '%') { - if (!(c = *(template++))) - break; - switch (c) { - case 'p': - av_bprintf(bp, "%s", program_name); - break; - case 't': - av_bprintf(bp, "%04d%02d%02d-%02d%02d%02d", - tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday, - tm->tm_hour, tm->tm_min, tm->tm_sec); - break; - case '%': - av_bprint_chars(bp, c, 1); - break; - } - } else { - av_bprint_chars(bp, c, 1); - } - } -} - -static int init_report(const char *env) -{ - char *filename_template = NULL; - char *key, *val; - int ret, count = 0; - time_t now; - struct tm *tm; - AVBPrint filename; - - if (report_file) /* already opened */ - return 0; - time(&now); - tm = localtime(&now); - - while (env && *env) { - if ((ret = av_opt_get_key_value(&env, "=", ":", 0, &key, &val)) < 0) { - if (count) - av_log(NULL, AV_LOG_ERROR, - "Failed to parse FFREPORT environment variable: %s\n", - av_err2str(ret)); - break; - } - if (*env) - env++; - count++; - if (!strcmp(key, "file")) { - av_free(filename_template); - filename_template = val; - val = NULL; - } else { - av_log(NULL, AV_LOG_ERROR, "Unknown key '%s' in FFREPORT\n", key); - } - av_free(val); - av_free(key); - } - - av_bprint_init(&filename, 0, 1); - expand_filename_template(&filename, - av_x_if_null(filename_template, "%p-%t.log"), tm); - av_free(filename_template); - if (!av_bprint_is_complete(&filename)) { - av_log(NULL, AV_LOG_ERROR, "Out of memory building report file name\n"); - return AVERROR(ENOMEM); - } - - report_file = fopen(filename.str, "w"); - if (!report_file) { - av_log(NULL, AV_LOG_ERROR, "Failed to open report \"%s\": %s\n", - filename.str, strerror(errno)); - return AVERROR(errno); - } - av_log_set_callback(log_callback_report); - av_log(NULL, AV_LOG_INFO, - "%s started on %04d-%02d-%02d at %02d:%02d:%02d\n" - "Report written to \"%s\"\n", - program_name, - tm->tm_year + 1900, tm->tm_mon + 1, tm->tm_mday, - tm->tm_hour, tm->tm_min, tm->tm_sec, - filename.str); - av_bprint_finalize(&filename, NULL); - return 0; -} - -int opt_report(const char *opt) -{ - return init_report(NULL); -} - -int opt_max_alloc(void *optctx, const char *opt, const char *arg) -{ - char *tail; - size_t max; - - max = strtol(arg, &tail, 10); - if (*tail) { - av_log(NULL, AV_LOG_FATAL, "Invalid max_alloc \"%s\".\n", arg); - exit_program(1); - } - av_max_alloc(max); - return 0; -} - int opt_timelimit(void *optctx, const char *opt, const char *arg) { #if HAVE_SETRLIMIT @@ -1007,7 +770,6 @@ static int warned_cfg = 0; #define INDENT 1 #define SHOW_VERSION 2 #define SHOW_CONFIG 4 -#define SHOW_COPYRIGHT 8 #define PRINT_LIB_INFO(libname, LIBNAME, flags, level) \ if (CONFIG_##LIBNAME) { \ @@ -1015,7 +777,7 @@ static int warned_cfg = 0; if (flags & SHOW_VERSION) { \ unsigned int version = libname##_version(); \ av_log(NULL, level, \ - "%slib%-11s %2d.%3d.%3d / %2d.%3d.%3d\n", \ + "%slib%-10s %2d.%3d.%2d / %2d.%3d.%2d\n", \ indent, #libname, \ LIB##LIBNAME##_VERSION_MAJOR, \ LIB##LIBNAME##_VERSION_MINOR, \ @@ -1024,7 +786,7 @@ static int warned_cfg = 0; } \ if (flags & SHOW_CONFIG) { \ const char *cfg = libname##_configuration(); \ - if (strcmp(FFMPEG_CONFIGURATION, cfg)) { \ + if (strcmp(LIBAV_CONFIGURATION, cfg)) { \ if (!warned_cfg) { \ av_log(NULL, level, \ "%sWARNING: library configuration mismatch\n", \ @@ -1046,88 +808,37 @@ static void print_all_libs_info(int flags, int level) PRINT_LIB_INFO(avfilter, AVFILTER, flags, level); PRINT_LIB_INFO(avresample, AVRESAMPLE, flags, level); PRINT_LIB_INFO(swscale, SWSCALE, flags, level); - PRINT_LIB_INFO(swresample,SWRESAMPLE, flags, level); - PRINT_LIB_INFO(postproc, POSTPROC, flags, level); } -static void print_program_info(int flags, int level) +void show_banner(void) { - const char *indent = flags & INDENT? " " : ""; - - av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name); - if (flags & SHOW_COPYRIGHT) - av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers", - program_birth_year, CONFIG_THIS_YEAR); - av_log(NULL, level, "\n"); - av_log(NULL, level, "%sbuilt on %s %s with %s\n", - indent, __DATE__, __TIME__, CC_IDENT); - - av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent); -} - -static void print_buildconf(int flags, int level) -{ - const char *indent = flags & INDENT ? " " : ""; - char str[] = { FFMPEG_CONFIGURATION }; - char *conflist, *remove_tilde, *splitconf; - - // Change all the ' --' strings to '~--' so that - // they can be identified as tokens. - while ((conflist = strstr(str, " --")) != NULL) { - strncpy(conflist, "~--", 3); - } - - // Compensate for the weirdness this would cause - // when passing 'pkg-config --static'. - while ((remove_tilde = strstr(str, "pkg-config~")) != NULL) { - strncpy(remove_tilde, "pkg-config ", 11); - } - - splitconf = strtok(str, "~"); - av_log(NULL, level, "\n%sconfiguration:\n", indent); - while (splitconf != NULL) { - av_log(NULL, level, "%s%s%s\n", indent, indent, splitconf); - splitconf = strtok(NULL, "~"); - } -} - -void show_banner(int argc, char **argv, const OptionDef *options) -{ - int idx = locate_option(argc, argv, options, "version"); - if (hide_banner || idx) - return; - - print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO); - print_all_libs_info(INDENT|SHOW_CONFIG, AV_LOG_INFO); - print_all_libs_info(INDENT|SHOW_VERSION, AV_LOG_INFO); + av_log(NULL, AV_LOG_INFO, + "%s version " LIBAV_VERSION ", Copyright (c) %d-%d the Libav developers\n", + program_name, program_birth_year, this_year); + av_log(NULL, AV_LOG_INFO, " built on %s %s with %s\n", + __DATE__, __TIME__, CC_IDENT); + av_log(NULL, AV_LOG_VERBOSE, " configuration: " LIBAV_CONFIGURATION "\n"); + print_all_libs_info(INDENT|SHOW_CONFIG, AV_LOG_VERBOSE); + print_all_libs_info(INDENT|SHOW_VERSION, AV_LOG_VERBOSE); } int show_version(void *optctx, const char *opt, const char *arg) { av_log_set_callback(log_callback_help); - print_program_info (0 , AV_LOG_INFO); + printf("%s " LIBAV_VERSION "\n", program_name); print_all_libs_info(SHOW_VERSION, AV_LOG_INFO); return 0; } -int show_buildconf(void *optctx, const char *opt, const char *arg) -{ - av_log_set_callback(log_callback_help); - print_buildconf (INDENT|0, AV_LOG_INFO); - - return 0; -} - int show_license(void *optctx, const char *opt, const char *arg) { -#if CONFIG_NONFREE printf( +#if CONFIG_NONFREE "This version of %s has nonfree parts compiled in.\n" "Therefore it is not legally redistributable.\n", - program_name ); + program_name #elif CONFIG_GPLV3 - printf( "%s is free software; you can redistribute it and/or modify\n" "it under the terms of the GNU General Public License as published by\n" "the Free Software Foundation; either version 3 of the License, or\n" @@ -1140,9 +851,8 @@ int show_license(void *optctx, const char *opt, const char *arg) "\n" "You should have received a copy of the GNU General Public License\n" "along with %s. If not, see .\n", - program_name, program_name, program_name ); + program_name, program_name, program_name #elif CONFIG_GPL - printf( "%s is free software; you can redistribute it and/or modify\n" "it under the terms of the GNU General Public License as published by\n" "the Free Software Foundation; either version 2 of the License, or\n" @@ -1156,9 +866,8 @@ int show_license(void *optctx, const char *opt, const char *arg) "You should have received a copy of the GNU General Public License\n" "along with %s; if not, write to the Free Software\n" "Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA\n", - program_name, program_name, program_name ); + program_name, program_name, program_name #elif CONFIG_LGPLV3 - printf( "%s is free software; you can redistribute it and/or modify\n" "it under the terms of the GNU Lesser General Public License as published by\n" "the Free Software Foundation; either version 3 of the License, or\n" @@ -1171,9 +880,8 @@ int show_license(void *optctx, const char *opt, const char *arg) "\n" "You should have received a copy of the GNU Lesser General Public License\n" "along with %s. If not, see .\n", - program_name, program_name, program_name ); + program_name, program_name, program_name #else - printf( "%s is free software; you can redistribute it and/or\n" "modify it under the terms of the GNU Lesser General Public\n" "License as published by the Free Software Foundation; either\n" @@ -1187,8 +895,9 @@ int show_license(void *optctx, const char *opt, const char *arg) "You should have received a copy of the GNU Lesser General Public\n" "License along with %s; if not, write to the Free Software\n" "Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA\n", - program_name, program_name, program_name ); + program_name, program_name, program_name #endif + ); return 0; } @@ -1243,7 +952,7 @@ int show_formats(void *optctx, const char *opt, const char *arg) #define PRINT_CODEC_SUPPORTED(codec, field, type, list_name, term, get_name) \ if (codec->field) { \ - const type *p = codec->field; \ + const type *p = c->field; \ \ printf(" Supported " list_name ":"); \ while (*p != term) { \ @@ -1261,8 +970,7 @@ static void print_codec(const AVCodec *c) printf("%s %s [%s]:\n", encoder ? "Encoder" : "Decoder", c->name, c->long_name ? c->long_name : ""); - if (c->type == AVMEDIA_TYPE_VIDEO || - c->type == AVMEDIA_TYPE_AUDIO) { + if (c->type == AVMEDIA_TYPE_VIDEO) { printf(" Threading capabilities: "); switch (c->capabilities & (CODEC_CAP_FRAME_THREADS | CODEC_CAP_SLICE_THREADS)) { @@ -1306,9 +1014,7 @@ static char get_media_type_char(enum AVMediaType type) switch (type) { case AVMEDIA_TYPE_VIDEO: return 'V'; case AVMEDIA_TYPE_AUDIO: return 'A'; - case AVMEDIA_TYPE_DATA: return 'D'; case AVMEDIA_TYPE_SUBTITLE: return 'S'; - case AVMEDIA_TYPE_ATTACHMENT:return 'T'; default: return '?'; } } @@ -1324,36 +1030,6 @@ static const AVCodec *next_codec_for_id(enum AVCodecID id, const AVCodec *prev, return NULL; } -static int compare_codec_desc(const void *a, const void *b) -{ - const AVCodecDescriptor * const *da = a; - const AVCodecDescriptor * const *db = b; - - return (*da)->type != (*db)->type ? (*da)->type - (*db)->type : - strcmp((*da)->name, (*db)->name); -} - -static unsigned get_codecs_sorted(const AVCodecDescriptor ***rcodecs) -{ - const AVCodecDescriptor *desc = NULL; - const AVCodecDescriptor **codecs; - unsigned nb_codecs = 0, i = 0; - - while ((desc = avcodec_descriptor_next(desc))) - nb_codecs++; - if (!(codecs = av_calloc(nb_codecs, sizeof(*codecs)))) { - av_log(NULL, AV_LOG_ERROR, "Out of memory\n"); - exit_program(1); - } - desc = NULL; - while ((desc = avcodec_descriptor_next(desc))) - codecs[i++] = desc; - av_assert0(i == nb_codecs); - qsort(codecs, nb_codecs, sizeof(*codecs), compare_codec_desc); - *rcodecs = codecs; - return nb_codecs; -} - static void print_codecs_for_id(enum AVCodecID id, int encoder) { const AVCodec *codec = NULL; @@ -1368,8 +1044,7 @@ static void print_codecs_for_id(enum AVCodecID id, int encoder) int show_codecs(void *optctx, const char *opt, const char *arg) { - const AVCodecDescriptor **codecs; - unsigned i, nb_codecs = get_codecs_sorted(&codecs); + const AVCodecDescriptor *desc = NULL; printf("Codecs:\n" " D..... = Decoding supported\n" @@ -1381,14 +1056,9 @@ int show_codecs(void *optctx, const char *opt, const char *arg) " ....L. = Lossy compression\n" " .....S = Lossless compression\n" " -------\n"); - for (i = 0; i < nb_codecs; i++) { - const AVCodecDescriptor *desc = codecs[i]; + while ((desc = avcodec_descriptor_next(desc))) { const AVCodec *codec = NULL; - if (strstr(desc->name, "_deprecated")) - continue; - - printf(" "); printf(avcodec_find_decoder(desc->id) ? "D" : "."); printf(avcodec_find_encoder(desc->id) ? "E" : "."); @@ -1417,37 +1087,30 @@ int show_codecs(void *optctx, const char *opt, const char *arg) printf("\n"); } - av_free(codecs); return 0; } static void print_codecs(int encoder) { - const AVCodecDescriptor **codecs; - unsigned i, nb_codecs = get_codecs_sorted(&codecs); + const AVCodecDescriptor *desc = NULL; printf("%s:\n" - " V..... = Video\n" - " A..... = Audio\n" - " S..... = Subtitle\n" - " .F.... = Frame-level multithreading\n" - " ..S... = Slice-level multithreading\n" - " ...X.. = Codec is experimental\n" - " ....B. = Supports draw_horiz_band\n" - " .....D = Supports direct rendering method 1\n" - " ------\n", + " V... = Video\n" + " A... = Audio\n" + " S... = Subtitle\n" + " .F.. = Frame-level multithreading\n" + " ..S. = Slice-level multithreading\n" + " ...X = Codec is experimental\n" + " ---\n", encoder ? "Encoders" : "Decoders"); - for (i = 0; i < nb_codecs; i++) { - const AVCodecDescriptor *desc = codecs[i]; + while ((desc = avcodec_descriptor_next(desc))) { const AVCodec *codec = NULL; while ((codec = next_codec_for_id(desc->id, codec, encoder))) { - printf(" %c", get_media_type_char(desc->type)); + printf("%c", get_media_type_char(desc->type)); printf((codec->capabilities & CODEC_CAP_FRAME_THREADS) ? "F" : "."); printf((codec->capabilities & CODEC_CAP_SLICE_THREADS) ? "S" : "."); printf((codec->capabilities & CODEC_CAP_EXPERIMENTAL) ? "X" : "."); - printf((codec->capabilities & CODEC_CAP_DRAW_HORIZ_BAND)?"B" : "."); - printf((codec->capabilities & CODEC_CAP_DR1) ? "D" : "."); printf(" %-20s %s", codec->name, codec->long_name ? codec->long_name : ""); if (strcmp(codec->name, desc->name)) @@ -1456,7 +1119,6 @@ static void print_codecs(int encoder) printf("\n"); } } - av_free(codecs); } int show_decoders(void *optctx, const char *opt, const char *arg) @@ -1500,61 +1162,15 @@ int show_protocols(void *optctx, const char *opt, const char *arg) int show_filters(void *optctx, const char *opt, const char *arg) { const AVFilter av_unused(*filter) = NULL; - char descr[64], *descr_cur; - int i, j; - const AVFilterPad *pad; - printf("Filters:\n" - " T.. = Timeline support\n" - " .S. = Slice threading\n" - " ..C = Commmand support\n" - " A = Audio input/output\n" - " V = Video input/output\n" - " N = Dynamic number and/or type of input/output\n" - " | = Source or sink filter\n"); + printf("Filters:\n"); #if CONFIG_AVFILTER - while ((filter = avfilter_next(filter))) { - descr_cur = descr; - for (i = 0; i < 2; i++) { - if (i) { - *(descr_cur++) = '-'; - *(descr_cur++) = '>'; - } - pad = i ? filter->outputs : filter->inputs; - for (j = 0; pad && pad[j].name; j++) { - if (descr_cur >= descr + sizeof(descr) - 4) - break; - *(descr_cur++) = get_media_type_char(pad[j].type); - } - if (!j) - *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) || - ( i && (filter->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS))) ? 'N' : '|'; - } - *descr_cur = 0; - printf(" %c%c%c %-16s %-10s %s\n", - filter->flags & AVFILTER_FLAG_SUPPORT_TIMELINE ? 'T' : '.', - filter->flags & AVFILTER_FLAG_SLICE_THREADS ? 'S' : '.', - filter->process_command ? 'C' : '.', - filter->name, descr, filter->description); - } + while ((filter = avfilter_next(filter))) + printf("%-16s %s\n", filter->name, filter->description); #endif return 0; } -int show_colors(void *optctx, const char *opt, const char *arg) -{ - const char *name; - const uint8_t *rgb; - int i; - - printf("%-32s #RRGGBB\n", "name"); - - for (i = 0; name = av_get_known_color_name(i, &rgb); i++) - printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]); - - return 0; -} - int show_pix_fmts(void *optctx, const char *opt, const char *arg) { const AVPixFmtDescriptor *pix_desc = NULL; @@ -1588,35 +1204,6 @@ int show_pix_fmts(void *optctx, const char *opt, const char *arg) return 0; } -int show_layouts(void *optctx, const char *opt, const char *arg) -{ - int i = 0; - uint64_t layout, j; - const char *name, *descr; - - printf("Individual channels:\n" - "NAME DESCRIPTION\n"); - for (i = 0; i < 63; i++) { - name = av_get_channel_name((uint64_t)1 << i); - if (!name) - continue; - descr = av_get_channel_description((uint64_t)1 << i); - printf("%-12s%s\n", name, descr); - } - printf("\nStandard channel layouts:\n" - "NAME DECOMPOSITION\n"); - for (i = 0; !av_get_standard_channel_layout(i, &layout, &name); i++) { - if (name) { - printf("%-12s", name); - for (j = 1; j; j <<= 1) - if ((layout & j)) - printf("%s%s", (layout & (j - 1)) ? "+" : "", av_get_channel_name(j)); - printf("\n"); - } - } - return 0; -} - int show_sample_fmts(void *optctx, const char *opt, const char *arg) { int i; @@ -1650,13 +1237,13 @@ static void show_help_codec(const char *name, int encoder) } if (!printed) { - av_log(NULL, AV_LOG_ERROR, "Codec '%s' is known to FFmpeg, " - "but no %s for it are available. FFmpeg might need to be " + av_log(NULL, AV_LOG_ERROR, "Codec '%s' is known to Libav, " + "but no %s for it are available. Libav might need to be " "recompiled with additional external libraries.\n", name, encoder ? "encoders" : "decoders"); } } else { - av_log(NULL, AV_LOG_ERROR, "Codec '%s' is not recognized by FFmpeg.\n", + av_log(NULL, AV_LOG_ERROR, "Codec '%s' is not recognized by Libav.\n", name); } } @@ -1715,7 +1302,6 @@ static void show_help_muxer(const char *name) #if CONFIG_AVFILTER static void show_help_filter(const char *name) { -#if CONFIG_AVFILTER const AVFilter *f = avfilter_get_by_name(name); int i, count; @@ -1727,9 +1313,7 @@ static void show_help_filter(const char *name) return; } - printf("Filter %s\n", f->name); - if (f->description) - printf(" %s\n", f->description); + printf("Filter %s [%s]:\n", f->name, f->description); if (f->flags & AVFILTER_FLAG_SLICE_THREADS) printf(" slice threading supported\n"); @@ -1737,34 +1321,24 @@ static void show_help_filter(const char *name) printf(" Inputs:\n"); count = avfilter_pad_count(f->inputs); for (i = 0; i < count; i++) { - printf(" #%d: %s (%s)\n", i, avfilter_pad_get_name(f->inputs, i), + printf(" %d %s (%s)\n", i, avfilter_pad_get_name(f->inputs, i), media_type_string(avfilter_pad_get_type(f->inputs, i))); } if (f->flags & AVFILTER_FLAG_DYNAMIC_INPUTS) printf(" dynamic (depending on the options)\n"); - else if (!count) - printf(" none (source filter)\n"); printf(" Outputs:\n"); count = avfilter_pad_count(f->outputs); for (i = 0; i < count; i++) { - printf(" #%d: %s (%s)\n", i, avfilter_pad_get_name(f->outputs, i), + printf(" %d %s (%s)\n", i, avfilter_pad_get_name(f->outputs, i), media_type_string(avfilter_pad_get_type(f->outputs, i))); } if (f->flags & AVFILTER_FLAG_DYNAMIC_OUTPUTS) printf(" dynamic (depending on the options)\n"); - else if (!count) - printf(" none (sink filter)\n"); if (f->priv_class) - show_help_children(f->priv_class, AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_FILTERING_PARAM | + show_help_children(f->priv_class, AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM); - if (f->flags & AVFILTER_FLAG_SUPPORT_TIMELINE) - printf("This filter has support for timeline through the 'enable' option.\n"); -#else - av_log(NULL, AV_LOG_ERROR, "Build without libavfilter; " - "can not to satisfy request\n"); -#endif } #endif @@ -1814,7 +1388,7 @@ int read_yesno(void) int cmdutils_read_file(const char *filename, char **bufptr, size_t *size) { int ret; - FILE *f = av_fopen_utf8(filename, "rb"); + FILE *f = fopen(filename, "rb"); if (!f) { av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename, @@ -1824,11 +1398,6 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size) fseek(f, 0, SEEK_END); *size = ftell(f); fseek(f, 0, SEEK_SET); - if (*size == (size_t)-1) { - av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", strerror(errno)); - fclose(f); - return AVERROR(errno); - } *bufptr = av_malloc(*size + 1); if (!*bufptr) { av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n"); @@ -1853,47 +1422,58 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size) return ret; } +void init_pts_correction(PtsCorrectionContext *ctx) +{ + ctx->num_faulty_pts = ctx->num_faulty_dts = 0; + ctx->last_pts = ctx->last_dts = INT64_MIN; +} + +int64_t guess_correct_pts(PtsCorrectionContext *ctx, int64_t reordered_pts, + int64_t dts) +{ + int64_t pts = AV_NOPTS_VALUE; + + if (dts != AV_NOPTS_VALUE) { + ctx->num_faulty_dts += dts <= ctx->last_dts; + ctx->last_dts = dts; + } + if (reordered_pts != AV_NOPTS_VALUE) { + ctx->num_faulty_pts += reordered_pts <= ctx->last_pts; + ctx->last_pts = reordered_pts; + } + if ((ctx->num_faulty_pts<=ctx->num_faulty_dts || dts == AV_NOPTS_VALUE) + && reordered_pts != AV_NOPTS_VALUE) + pts = reordered_pts; + else + pts = dts; + + return pts; +} + FILE *get_preset_file(char *filename, size_t filename_size, const char *preset_name, int is_path, const char *codec_name) { FILE *f = NULL; int i; - const char *base[3] = { getenv("FFMPEG_DATADIR"), + const char *base[3] = { getenv("AVCONV_DATADIR"), getenv("HOME"), - FFMPEG_DATADIR, }; + AVCONV_DATADIR, }; if (is_path) { av_strlcpy(filename, preset_name, filename_size); f = fopen(filename, "r"); } else { -#ifdef _WIN32 - char datadir[MAX_PATH], *ls; - base[2] = NULL; - - if (GetModuleFileNameA(GetModuleHandleA(NULL), datadir, sizeof(datadir) - 1)) - { - for (ls = datadir; ls < datadir + strlen(datadir); ls++) - if (*ls == '\\') *ls = '/'; - - if (ls = strrchr(datadir, '/')) - { - *ls = 0; - strncat(datadir, "/ffpresets", sizeof(datadir) - 1 - strlen(datadir)); - base[2] = datadir; - } - } -#endif for (i = 0; i < 3 && !f; i++) { if (!base[i]) continue; - snprintf(filename, filename_size, "%s%s/%s.ffpreset", base[i], - i != 1 ? "" : "/.ffmpeg", preset_name); + snprintf(filename, filename_size, "%s%s/%s.avpreset", base[i], + i != 1 ? "" : "/.avconv", preset_name); f = fopen(filename, "r"); if (!f && codec_name) { snprintf(filename, filename_size, - "%s%s/%s-%s.ffpreset", - base[i], i != 1 ? "" : "/.ffmpeg", codec_name, + "%s%s/%s-%s.avpreset", + base[i], i != 1 ? "" : "/.avconv", codec_name, preset_name); f = fopen(filename, "r"); } @@ -1905,10 +1485,62 @@ FILE *get_preset_file(char *filename, size_t filename_size, int check_stream_specifier(AVFormatContext *s, AVStream *st, const char *spec) { - int ret = avformat_match_stream_specifier(s, st, spec); - if (ret < 0) - av_log(s, AV_LOG_ERROR, "Invalid stream specifier: %s.\n", spec); - return ret; + if (*spec <= '9' && *spec >= '0') /* opt:index */ + return strtol(spec, NULL, 0) == st->index; + else if (*spec == 'v' || *spec == 'a' || *spec == 's' || *spec == 'd' || + *spec == 't') { /* opt:[vasdt] */ + enum AVMediaType type; + + switch (*spec++) { + case 'v': type = AVMEDIA_TYPE_VIDEO; break; + case 'a': type = AVMEDIA_TYPE_AUDIO; break; + case 's': type = AVMEDIA_TYPE_SUBTITLE; break; + case 'd': type = AVMEDIA_TYPE_DATA; break; + case 't': type = AVMEDIA_TYPE_ATTACHMENT; break; + default: av_assert0(0); + } + if (type != st->codec->codec_type) + return 0; + if (*spec++ == ':') { /* possibly followed by :index */ + int i, index = strtol(spec, NULL, 0); + for (i = 0; i < s->nb_streams; i++) + if (s->streams[i]->codec->codec_type == type && index-- == 0) + return i == st->index; + return 0; + } + return 1; + } else if (*spec == 'p' && *(spec + 1) == ':') { + int prog_id, i, j; + char *endptr; + spec += 2; + prog_id = strtol(spec, &endptr, 0); + for (i = 0; i < s->nb_programs; i++) { + if (s->programs[i]->id != prog_id) + continue; + + if (*endptr++ == ':') { + int stream_idx = strtol(endptr, NULL, 0); + return stream_idx >= 0 && + stream_idx < s->programs[i]->nb_stream_indexes && + st->index == s->programs[i]->stream_index[stream_idx]; + } + + for (j = 0; j < s->programs[i]->nb_stream_indexes; j++) + if (st->index == s->programs[i]->stream_index[j]) + return 1; + } + return 0; + } else if (*spec == 'i' && *(spec + 1) == ':') { + int stream_id; + char *endptr; + spec += 2; + stream_id = strtol(spec, &endptr, 0); + return stream_id == st->id; + } else if (!*spec) /* empty specifier, matches everything */ + return 1; + + av_log(s, AV_LOG_ERROR, "Invalid stream specifier: %s.\n", spec); + return AVERROR(EINVAL); } AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id, @@ -1952,8 +1584,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id, } if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) || - !codec || - (codec->priv_class && + (codec && codec->priv_class && av_opt_find(&codec->priv_class, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ))) av_dict_set(&ret, t->key, t->value, 0); @@ -1976,7 +1607,7 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s, if (!s->nb_streams) return NULL; - opts = av_mallocz_array(s->nb_streams, sizeof(*opts)); + opts = av_mallocz(s->nb_streams * sizeof(*opts)); if (!opts) { av_log(NULL, AV_LOG_ERROR, "Could not alloc memory for stream options.\n"); @@ -2006,3 +1637,15 @@ void *grow_array(void *array, int elem_size, int *size, int new_size) } return array; } + +const char *media_type_string(enum AVMediaType media_type) +{ + switch (media_type) { + case AVMEDIA_TYPE_VIDEO: return "video"; + case AVMEDIA_TYPE_AUDIO: return "audio"; + case AVMEDIA_TYPE_DATA: return "data"; + case AVMEDIA_TYPE_SUBTITLE: return "subtitle"; + case AVMEDIA_TYPE_ATTACHMENT: return "attachment"; + default: return "unknown"; + } +} diff --git a/cmdutils.h b/cmdutils.h index c4a16aac73..d90ad4549d 100644 --- a/cmdutils.h +++ b/cmdutils.h @@ -2,38 +2,33 @@ * Various utilities for command line tools * copyright (c) 2003 Fabrice Bellard * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef FFMPEG_CMDUTILS_H -#define FFMPEG_CMDUTILS_H +#ifndef LIBAV_CMDUTILS_H +#define LIBAV_CMDUTILS_H #include -#include "config.h" #include "libavcodec/avcodec.h" #include "libavfilter/avfilter.h" #include "libavformat/avformat.h" #include "libswscale/swscale.h" -#ifdef _WIN32 -#undef main /* We don't want SDL to override our main() */ -#endif - /** * program name, defined by the program for show_version(). */ @@ -47,9 +42,7 @@ extern const int program_birth_year; extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB]; extern AVFormatContext *avformat_opts; extern struct SwsContext *sws_opts; -extern AVDictionary *swr_opts; extern AVDictionary *format_opts, *codec_opts, *resample_opts; -extern int hide_banner; /** * Register a program-specific cleanup routine. @@ -74,12 +67,12 @@ void uninit_opts(void); /** * Trivial log callback. - * Only suitable for opt_help and similar since it lacks prefix handling. + * Only suitable for show_help and similar since it lacks prefix handling. */ void log_callback_help(void* ptr, int level, const char* fmt, va_list vl); /** - * Override the cpuflags. + * Override the cpuflags mask. */ int opt_cpuflags(void *optctx, const char *opt, const char *arg); @@ -94,18 +87,6 @@ int opt_default(void *optctx, const char *opt, const char *arg); */ int opt_loglevel(void *optctx, const char *opt, const char *arg); -int opt_report(const char *opt); - -int opt_max_alloc(void *optctx, const char *opt, const char *arg); - -int opt_codec_debug(void *optctx, const char *opt, const char *arg); - -#if CONFIG_OPENCL -int opt_opencl(void *optctx, const char *opt, const char *arg); - -int opt_opencl_bench(void *optctx, const char *opt, const char *arg); -#endif - /** * Limit the execution time. */ @@ -139,7 +120,7 @@ double parse_number_or_die(const char *context, const char *numstr, int type, * not zero timestr is interpreted as a duration, otherwise as a * date * - * @see av_parse_time() + * @see parse_date() */ int64_t parse_time_or_die(const char *context, const char *timestr, int is_duration); @@ -170,7 +151,7 @@ typedef struct OptionDef { #define OPT_INT64 0x0400 #define OPT_EXIT 0x0800 #define OPT_DATA 0x1000 -#define OPT_PERFILE 0x2000 /* the option is per-file (currently ffmpeg-only). +#define OPT_PERFILE 0x2000 /* the option is per-file (currently avconv-only). implied by OPT_OFFSET or OPT_SPEC */ #define OPT_OFFSET 0x4000 /* option is specified as an offset in a passed optctx */ #define OPT_SPEC 0x8000 /* option is to be stored in an array of SpecifierOpt. @@ -208,13 +189,13 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags, void show_help_children(const AVClass *class, int flags); /** - * Per-fftool specific help handler. Implemented in each - * fftool, called by show_help(). + * Per-avtool specific help handler. Implemented in each + * avtool, called by show_help(). */ void show_help_default(const char *opt, const char *arg); /** - * Generic -h handler common to all fftools. + * Generic -h handler common to all avtools. */ int show_help(void *optctx, const char *opt, const char *arg); @@ -257,7 +238,7 @@ typedef struct OptionGroupDef { const char *name; /** * Option to be used as group separator. Can be NULL for groups which - * are terminated by a non-option argument (e.g. ffmpeg output files) + * are terminated by a non-option argument (e.g. avconv output files) */ const char *sep; /** @@ -278,7 +259,6 @@ typedef struct OptionGroup { AVDictionary *format_opts; AVDictionary *resample_opts; struct SwsContext *sws_opts; - AVDictionary *swr_opts; } OptionGroup; /** @@ -405,41 +385,30 @@ void print_error(const char *filename, int err); * current version of the repository and of the libav* libraries used by * the program. */ -void show_banner(int argc, char **argv, const OptionDef *options); +void show_banner(void); /** * Print the version of the program to stdout. The version message * depends on the current versions of the repository and of the libav* * libraries. - * This option processing function does not utilize the arguments. */ int show_version(void *optctx, const char *opt, const char *arg); -/** - * Print the build configuration of the program to stdout. The contents - * depend on the definition of FFMPEG_CONFIGURATION. - * This option processing function does not utilize the arguments. - */ -int show_buildconf(void *optctx, const char *opt, const char *arg); - /** * Print the license of the program to stdout. The license depends on * the license of the libraries compiled into the program. - * This option processing function does not utilize the arguments. */ int show_license(void *optctx, const char *opt, const char *arg); /** * Print a listing containing all the formats supported by the * program. - * This option processing function does not utilize the arguments. */ int show_formats(void *optctx, const char *opt, const char *arg); /** * Print a listing containing all the codecs supported by the * program. - * This option processing function does not utilize the arguments. */ int show_codecs(void *optctx, const char *opt, const char *arg); @@ -458,50 +427,33 @@ int show_encoders(void *optctx, const char *opt, const char *arg); /** * Print a listing containing all the filters supported by the * program. - * This option processing function does not utilize the arguments. */ int show_filters(void *optctx, const char *opt, const char *arg); /** * Print a listing containing all the bit stream filters supported by the * program. - * This option processing function does not utilize the arguments. */ int show_bsfs(void *optctx, const char *opt, const char *arg); /** * Print a listing containing all the protocols supported by the * program. - * This option processing function does not utilize the arguments. */ int show_protocols(void *optctx, const char *opt, const char *arg); /** * Print a listing containing all the pixel formats supported by the * program. - * This option processing function does not utilize the arguments. */ int show_pix_fmts(void *optctx, const char *opt, const char *arg); -/** - * Print a listing containing all the standard channel layouts supported by - * the program. - * This option processing function does not utilize the arguments. - */ -int show_layouts(void *optctx, const char *opt, const char *arg); - /** * Print a listing containing all the sample formats supported by the * program. */ int show_sample_fmts(void *optctx, const char *opt, const char *arg); -/** - * Print a listing containing all the color names and values recognized - * by the program. - */ -int show_colors(void *optctx, const char *opt, const char *arg); - /** * Return a positive value if a line read from standard input * starts with [yY], otherwise return 0. @@ -515,19 +467,43 @@ int read_yesno(void); * @param filename file to read from * @param bufptr location where pointer to buffer is returned * @param size location where size of buffer is returned - * @return >= 0 in case of success, a negative value corresponding to an + * @return 0 in case of success, a negative value corresponding to an * AVERROR error code in case of failure. */ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size); +typedef struct PtsCorrectionContext { + int64_t num_faulty_pts; /// Number of incorrect PTS values so far + int64_t num_faulty_dts; /// Number of incorrect DTS values so far + int64_t last_pts; /// PTS of the last frame + int64_t last_dts; /// DTS of the last frame +} PtsCorrectionContext; + +/** + * Reset the state of the PtsCorrectionContext. + */ +void init_pts_correction(PtsCorrectionContext *ctx); + +/** + * Attempt to guess proper monotonic timestamps for decoded video frames + * which might have incorrect times. Input timestamps may wrap around, in + * which case the output will as well. + * + * @param ctx the PtsCorrectionContext carrying stream pts information + * @param pts the pts field of the decoded AVPacket, as passed through + * AVCodecContext.reordered_opaque + * @param dts the dts field of the decoded AVPacket + * @return one of the input values, may be AV_NOPTS_VALUE + */ +int64_t guess_correct_pts(PtsCorrectionContext *ctx, int64_t pts, int64_t dts); + /** * Get a file corresponding to a preset file. * * If is_path is non-zero, look for the file in the path preset_name. - * Otherwise search for a file named arg.ffpreset in the directories - * $FFMPEG_DATADIR (if set), $HOME/.ffmpeg, and in the datadir defined - * at configuration time or in a "ffpresets" folder along the executable - * on win32, in that order. If no such file is found and + * Otherwise search for a file named arg.avpreset in the directories + * $AVCONV_DATADIR (if set), $HOME/.avconv, and in the datadir defined + * at configuration time, in that order. If no such file is found and * codec_name is defined, then search for a file named * codec_name-preset_name.avpreset in the above-mentioned directories. * @@ -553,7 +529,10 @@ FILE *get_preset_file(char *filename, size_t filename_size, */ void *grow_array(void *array, int elem_size, int *size, int new_size); -#define media_type_string av_get_media_type_string +/** + * Get a string describing a media type. + */ +const char *media_type_string(enum AVMediaType media_type); #define GROW_ARRAY(array, nb_elems)\ array = grow_array(array, sizeof(*array), &nb_elems, nb_elems + 1) @@ -576,4 +555,4 @@ void *grow_array(void *array, int elem_size, int *size, int new_size); char name[128];\ av_get_channel_layout_string(name, sizeof(name), 0, ch_layout); -#endif /* CMDUTILS_H */ +#endif /* LIBAV_CMDUTILS_H */ diff --git a/cmdutils_common_opts.h b/cmdutils_common_opts.h index 190dba674f..8693f59c56 100644 --- a/cmdutils_common_opts.h +++ b/cmdutils_common_opts.h @@ -4,7 +4,6 @@ { "help" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" }, { "-help" , OPT_EXIT, {.func_arg = show_help}, "show help", "topic" }, { "version" , OPT_EXIT, {.func_arg = show_version}, "show version" }, - { "buildconf" , OPT_EXIT, {.func_arg = show_buildconf}, "show build configuration" }, { "formats" , OPT_EXIT, {.func_arg = show_formats }, "show available formats" }, { "codecs" , OPT_EXIT, {.func_arg = show_codecs }, "show available codecs" }, { "decoders" , OPT_EXIT, {.func_arg = show_decoders }, "show available decoders" }, @@ -13,16 +12,7 @@ { "protocols" , OPT_EXIT, {.func_arg = show_protocols}, "show available protocols" }, { "filters" , OPT_EXIT, {.func_arg = show_filters }, "show available filters" }, { "pix_fmts" , OPT_EXIT, {.func_arg = show_pix_fmts }, "show available pixel formats" }, - { "layouts" , OPT_EXIT, {.func_arg = show_layouts }, "show standard channel layouts" }, { "sample_fmts", OPT_EXIT, {.func_arg = show_sample_fmts }, "show available audio sample formats" }, - { "colors" , OPT_EXIT, {.func_arg = show_colors }, "show available color names" }, - { "loglevel" , HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" }, - { "v", HAS_ARG, {.func_arg = opt_loglevel}, "set logging level", "loglevel" }, - { "report" , 0, {(void*)opt_report}, "generate a report" }, - { "max_alloc" , HAS_ARG, {.func_arg = opt_max_alloc}, "set maximum size of a single allocated block", "bytes" }, - { "cpuflags" , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" }, - { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner}, "do not show program banner", "hide_banner" }, -#if CONFIG_OPENCL - { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" }, - { "opencl_options", HAS_ARG, {.func_arg = opt_opencl}, "set OpenCL environment options" }, -#endif + { "loglevel" , HAS_ARG, {.func_arg = opt_loglevel}, "set libav* logging level", "loglevel" }, + { "v", HAS_ARG, {.func_arg = opt_loglevel}, "set libav* logging level", "loglevel" }, + { "cpuflags", HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "set CPU flags mask", "mask" }, diff --git a/cmdutils_opencl.c b/cmdutils_opencl.c deleted file mode 100644 index 0fa8cc0ada..0000000000 --- a/cmdutils_opencl.c +++ /dev/null @@ -1,274 +0,0 @@ -/* - * Copyright (C) 2013 Lenny Wang - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include "libavutil/opt.h" -#include "libavutil/time.h" -#include "libavutil/log.h" -#include "libavutil/opencl.h" -#include "cmdutils.h" - -typedef struct { - int platform_idx; - int device_idx; - char device_name[64]; - int64_t runtime; -} OpenCLDeviceBenchmark; - -const char *ocl_bench_source = AV_OPENCL_KERNEL( -inline unsigned char clip_uint8(int a) -{ - if (a & (~0xFF)) - return (-a)>>31; - else - return a; -} - -kernel void unsharp_bench( - global unsigned char *src, - global unsigned char *dst, - global int *mask, - int width, - int height) -{ - int i, j, local_idx, lc_idx, sum = 0; - int2 thread_idx, block_idx, global_idx, lm_idx; - thread_idx.x = get_local_id(0); - thread_idx.y = get_local_id(1); - block_idx.x = get_group_id(0); - block_idx.y = get_group_id(1); - global_idx.x = get_global_id(0); - global_idx.y = get_global_id(1); - local uchar data[32][32]; - local int lc[128]; - - for (i = 0; i <= 1; i++) { - lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y; - lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y; - lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y; - for (j = 0; j <= 1; j++) { - lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x; - lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x; - lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x; - data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x]; - } - } - local_idx = thread_idx.y*16 + thread_idx.x; - if (local_idx < 128) - lc[local_idx] = mask[local_idx]; - barrier(CLK_LOCAL_MEM_FENCE); - - \n#pragma unroll\n - for (i = -4; i <= 4; i++) { - lm_idx.y = 8 + i + thread_idx.y; - \n#pragma unroll\n - for (j = -4; j <= 4; j++) { - lm_idx.x = 8 + j + thread_idx.x; - lc_idx = (i + 4)*8 + j + 4; - sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx]; - } - } - int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8]; - int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16); - if (global_idx.x < width && global_idx.y < height) - dst[global_idx.x + global_idx.y*width] = clip_uint8(res); -} -); - -#define OCLCHECK(method, ... ) \ -do { \ - status = method(__VA_ARGS__); \ - if (status != CL_SUCCESS) { \ - av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n", \ - av_opencl_errstr(status)); \ - ret = AVERROR_EXTERNAL; \ - goto end; \ - } \ -} while (0) - -#define CREATEBUF(out, flags, size) \ -do { \ - out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \ - if (status != CL_SUCCESS) { \ - av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n"); \ - ret = AVERROR_EXTERNAL; \ - goto end; \ - } \ -} while (0) - -static void fill_rand_int(int *data, int n) -{ - int i; - srand(av_gettime()); - for (i = 0; i < n; i++) - data[i] = rand(); -} - -#define OPENCL_NB_ITER 5 -static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env) -{ - int i, arg = 0, width = 1920, height = 1088; - int64_t start, ret = 0; - cl_int status; - size_t kernel_len; - char *inbuf; - int *mask; - int buf_size = width * height * sizeof(char); - int mask_size = sizeof(uint32_t) * 128; - - cl_mem cl_mask, cl_inbuf, cl_outbuf; - cl_kernel kernel = NULL; - cl_program program = NULL; - size_t local_work_size_2d[2] = {16, 16}; - size_t global_work_size_2d[2] = {(size_t)width, (size_t)height}; - - if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) { - av_log(NULL, AV_LOG_ERROR, "Out of memory\n"); - ret = AVERROR(ENOMEM); - goto end; - } - fill_rand_int((int*)inbuf, buf_size/4); - fill_rand_int(mask, mask_size/4); - - CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size); - CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size); - CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size); - - kernel_len = strlen(ocl_bench_source); - program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source, - &kernel_len, &status); - if (status != CL_SUCCESS || !program) { - av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n"); - ret = AVERROR_EXTERNAL; - goto end; - } - status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL); - if (status != CL_SUCCESS) { - av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n"); - ret = AVERROR_EXTERNAL; - goto end; - } - kernel = clCreateKernel(program, "unsharp_bench", &status); - if (status != CL_SUCCESS) { - av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n"); - ret = AVERROR_EXTERNAL; - goto end; - } - - OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0, - buf_size, inbuf, 0, NULL, NULL); - OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0, - mask_size, mask, 0, NULL, NULL); - OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf); - OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf); - OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask); - OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width); - OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height); - - start = av_gettime_relative(); - for (i = 0; i < OPENCL_NB_ITER; i++) - OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL, - global_work_size_2d, local_work_size_2d, 0, NULL, NULL); - clFinish(ext_opencl_env->command_queue); - ret = (av_gettime_relative() - start)/OPENCL_NB_ITER; -end: - if (kernel) - clReleaseKernel(kernel); - if (program) - clReleaseProgram(program); - if (cl_inbuf) - clReleaseMemObject(cl_inbuf); - if (cl_outbuf) - clReleaseMemObject(cl_outbuf); - if (cl_mask) - clReleaseMemObject(cl_mask); - av_free(inbuf); - av_free(mask); - return ret; -} - -static int compare_ocl_device_desc(const void *a, const void *b) -{ - return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime; -} - -int opt_opencl_bench(void *optctx, const char *opt, const char *arg) -{ - int i, j, nb_devices = 0, count = 0; - int64_t score = 0; - AVOpenCLDeviceList *device_list; - AVOpenCLDeviceNode *device_node = NULL; - OpenCLDeviceBenchmark *devices = NULL; - cl_platform_id platform; - - av_opencl_get_device_list(&device_list); - for (i = 0; i < device_list->platform_num; i++) - nb_devices += device_list->platform_node[i]->device_num; - if (!nb_devices) { - av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n"); - return AVERROR(EINVAL); - } - if (!(devices = av_malloc(sizeof(OpenCLDeviceBenchmark) * nb_devices))) { - av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n"); - return AVERROR(ENOMEM); - } - - for (i = 0; i < device_list->platform_num; i++) { - for (j = 0; j < device_list->platform_node[i]->device_num; j++) { - device_node = device_list->platform_node[i]->device_node[j]; - platform = device_list->platform_node[i]->platform_id; - score = av_opencl_benchmark(device_node, platform, run_opencl_bench); - if (score > 0) { - devices[count].platform_idx = i; - devices[count].device_idx = j; - devices[count].runtime = score; - strcpy(devices[count].device_name, device_node->device_name); - count++; - } - } - } - qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc); - fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n"); - for (i = 0; i < count; i++) - fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n", - devices[i].platform_idx, devices[i].device_idx, - devices[i].device_name, devices[i].runtime); - - av_opencl_free_device_list(&device_list); - av_free(devices); - return 0; -} - -int opt_opencl(void *optctx, const char *opt, const char *arg) -{ - char *key, *value; - const char *opts = arg; - int ret = 0; - while (*opts) { - ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value); - if (ret < 0) - return ret; - ret = av_opencl_set_option(key, value); - if (ret < 0) - return ret; - if (*opts) - opts++; - } - return ret; -} diff --git a/common.mak b/common.mak index a493792657..9244fd366d 100644 --- a/common.mak +++ b/common.mak @@ -5,94 +5,9 @@ # first so "all" becomes default target all: all-yes -ifndef SUBDIR - -ifndef V -Q = @ -ECHO = printf "$(1)\t%s\n" $(2) -BRIEF = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES -SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM - -MSG = $@ -M = @$(call ECHO,$(TAG),$@); -$(foreach VAR,$(BRIEF), \ - $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR)))) -$(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR)))) -$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL)) -endif - -ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample - -# NASM requires -I path terminated with / -IFLAGS := -I. -I$(SRC_PATH)/ -CPPFLAGS := $(IFLAGS) $(CPPFLAGS) -CFLAGS += $(ECFLAGS) -CCFLAGS = $(CPPFLAGS) $(CFLAGS) -ASFLAGS := $(CPPFLAGS) $(ASFLAGS) -CXXFLAGS += $(CPPFLAGS) $(CFLAGS) -YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm - -HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS) -LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS) - -define COMPILE - $(call $(1)DEP,$(1)) - $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $< -endef - -COMPILE_C = $(call COMPILE,CC) -COMPILE_CXX = $(call COMPILE,CXX) -COMPILE_S = $(call COMPILE,AS) -COMPILE_HOSTC = $(call COMPILE,HOSTCC) - -%.o: %.c - $(COMPILE_C) - -%.o: %.cpp - $(COMPILE_CXX) - -%.o: %.m - $(COMPILE_C) - -%.s: %.c - $(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $< - -%.o: %.S - $(COMPILE_S) - -%_host.o: %.c - $(COMPILE_HOSTC) - -%.o: %.rc - $(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $< - -%.i: %.c - $(CC) $(CCFLAGS) $(CC_E) $< - -%.h.c: - $(Q)echo '#include "$*.h"' >$@ - -%.ver: %.v - $(Q)sed 's/$$MAJOR/$($(basename $(@F))_VERSION_MAJOR)/' $^ > $@ - -%.c %.h: TAG = GEN - -# Dummy rule to stop make trying to rebuild removed or renamed headers -%.h: - @: - -# Disable suffix rules. Most of the builtin rules are suffix rules, -# so this saves some time on slow systems. -.SUFFIXES: - -# Do not delete intermediate files from chains of implicit rules -$(OBJS): -endif - include $(SRC_PATH)/arch.mak OBJS += $(OBJS-yes) -SLIBOBJS += $(SLIBOBJS-yes) FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS) TESTPROGS += $(TESTPROGS-yes) @@ -100,7 +15,6 @@ LDLIBS = $(FFLIBS:%=%$(BUILDSUF)) FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS) OBJS := $(sort $(OBJS:%=$(SUBDIR)%)) -SLIBOBJS := $(sort $(SLIBOBJS:%=$(SUBDIR)%)) TESTOBJS := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o) TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF)) HOSTOBJS := $(HOSTPROGS:%=$(SUBDIR)%.o) @@ -132,13 +46,12 @@ $(HOSTPROGS): %$(HOSTEXESUF): %.o $(OBJS): | $(sort $(dir $(OBJS))) $(HOBJS): | $(sort $(dir $(HOBJS))) $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS))) -$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS))) $(TESTOBJS): | $(sort $(dir $(TESTOBJS))) $(TOOLOBJS): | tools -OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS)) +OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS)) -CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda +CLEANSUFFIXES = *.d *.o *~ *.h.c *.map *.ver *.gcno *.gcda DISTCLEANSUFFIXES = *.pc LIBSUFFIXES = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a @@ -151,4 +64,4 @@ endef $(eval $(RULES)) --include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) +-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d)) diff --git a/compat/aix/math.h b/compat/aix/math.h index 65a89c4565..380f87804d 100644 --- a/compat/aix/math.h +++ b/compat/aix/math.h @@ -2,25 +2,25 @@ * Work around the class() function in AIX math.h clashing with * identifiers named "class". * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef FFMPEG_COMPAT_AIX_MATH_H -#define FFMPEG_COMPAT_AIX_MATH_H +#ifndef LIBAV_COMPAT_AIX_MATH_H +#define LIBAV_COMPAT_AIX_MATH_H #define class class_in_math_h_causes_problems @@ -28,4 +28,4 @@ #undef class -#endif /* FFMPEG_COMPAT_AIX_MATH_H */ +#endif /* LIBAV_COMPAT_AIX_MATH_H */ diff --git a/compat/avisynth/avisynth_c.h b/compat/avisynth/avisynth_c.h deleted file mode 100644 index 448493b839..0000000000 --- a/compat/avisynth/avisynth_c.h +++ /dev/null @@ -1,880 +0,0 @@ -// Avisynth C Interface Version 0.20 -// Copyright 2003 Kevin Atkinson - -// This program is free software; you can redistribute it and/or modify -// it under the terms of the GNU General Public License as published by -// the Free Software Foundation; either version 2 of the License, or -// (at your option) any later version. -// -// This program is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -// GNU General Public License for more details. -// -// You should have received a copy of the GNU General Public License -// along with this program; if not, write to the Free Software -// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, -// MA 02110-1301 USA, or visit -// http://www.gnu.org/copyleft/gpl.html . -// -// As a special exception, I give you permission to link to the -// Avisynth C interface with independent modules that communicate with -// the Avisynth C interface solely through the interfaces defined in -// avisynth_c.h, regardless of the license terms of these independent -// modules, and to copy and distribute the resulting combined work -// under terms of your choice, provided that every copy of the -// combined work is accompanied by a complete copy of the source code -// of the Avisynth C interface and Avisynth itself (with the version -// used to produce the combined work), being distributed under the -// terms of the GNU General Public License plus this exception. An -// independent module is a module which is not derived from or based -// on Avisynth C Interface, such as 3rd-party filters, import and -// export plugins, or graphical user interfaces. - -// NOTE: this is a partial update of the Avisynth C interface to recognize -// new color spaces added in Avisynth 2.60. By no means is this document -// completely Avisynth 2.60 compliant. - -#ifndef __AVISYNTH_C__ -#define __AVISYNTH_C__ - -#ifdef __cplusplus -# define EXTERN_C extern "C" -#else -# define EXTERN_C -#endif - -#define AVSC_USE_STDCALL 1 - -#ifndef AVSC_USE_STDCALL -# define AVSC_CC __cdecl -#else -# define AVSC_CC __stdcall -#endif - -#define AVSC_INLINE static __inline - -#ifdef AVISYNTH_C_EXPORTS -# define AVSC_EXPORT EXTERN_C -# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name -#else -# define AVSC_EXPORT EXTERN_C __declspec(dllexport) -# ifndef AVSC_NO_DECLSPEC -# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name -# else -# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func) -# endif -#endif - -typedef unsigned char BYTE; -#ifdef __GNUC__ -typedef long long int INT64; -#else -typedef __int64 INT64; -#endif - - -///////////////////////////////////////////////////////////////////// -// -// Constants -// - -#ifndef __AVISYNTH_H__ -enum { AVISYNTH_INTERFACE_VERSION = 4 }; -#endif - -enum {AVS_SAMPLE_INT8 = 1<<0, - AVS_SAMPLE_INT16 = 1<<1, - AVS_SAMPLE_INT24 = 1<<2, - AVS_SAMPLE_INT32 = 1<<3, - AVS_SAMPLE_FLOAT = 1<<4}; - -enum {AVS_PLANAR_Y=1<<0, - AVS_PLANAR_U=1<<1, - AVS_PLANAR_V=1<<2, - AVS_PLANAR_ALIGNED=1<<3, - AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED, - AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED, - AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED, - AVS_PLANAR_A=1<<4, - AVS_PLANAR_R=1<<5, - AVS_PLANAR_G=1<<6, - AVS_PLANAR_B=1<<7, - AVS_PLANAR_A_ALIGNED=AVS_PLANAR_A|AVS_PLANAR_ALIGNED, - AVS_PLANAR_R_ALIGNED=AVS_PLANAR_R|AVS_PLANAR_ALIGNED, - AVS_PLANAR_G_ALIGNED=AVS_PLANAR_G|AVS_PLANAR_ALIGNED, - AVS_PLANAR_B_ALIGNED=AVS_PLANAR_B|AVS_PLANAR_ALIGNED}; - - // Colorspace properties. -enum {AVS_CS_BGR = 1<<28, - AVS_CS_YUV = 1<<29, - AVS_CS_INTERLEAVED = 1<<30, - AVS_CS_PLANAR = 1<<31, - - AVS_CS_SHIFT_SUB_WIDTH = 0, - AVS_CS_SHIFT_SUB_HEIGHT = 1 << 3, - AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4, - - AVS_CS_SUB_WIDTH_MASK = 7 << AVS_CS_SHIFT_SUB_WIDTH, - AVS_CS_SUB_WIDTH_1 = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24 - AVS_CS_SUB_WIDTH_2 = 0 << AVS_CS_SHIFT_SUB_WIDTH, // YV12, I420, YV16 - AVS_CS_SUB_WIDTH_4 = 1 << AVS_CS_SHIFT_SUB_WIDTH, // YUV9, YV411 - - AVS_CS_VPLANEFIRST = 1 << 3, // YV12, YV16, YV24, YV411, YUV9 - AVS_CS_UPLANEFIRST = 1 << 4, // I420 - - AVS_CS_SUB_HEIGHT_MASK = 7 << AVS_CS_SHIFT_SUB_HEIGHT, - AVS_CS_SUB_HEIGHT_1 = 3 << AVS_CS_SHIFT_SUB_HEIGHT, // YV16, YV24, YV411 - AVS_CS_SUB_HEIGHT_2 = 0 << AVS_CS_SHIFT_SUB_HEIGHT, // YV12, I420 - AVS_CS_SUB_HEIGHT_4 = 1 << AVS_CS_SHIFT_SUB_HEIGHT, // YUV9 - - AVS_CS_SAMPLE_BITS_MASK = 7 << AVS_CS_SHIFT_SAMPLE_BITS, - AVS_CS_SAMPLE_BITS_8 = 0 << AVS_CS_SHIFT_SAMPLE_BITS, - AVS_CS_SAMPLE_BITS_16 = 1 << AVS_CS_SHIFT_SAMPLE_BITS, - AVS_CS_SAMPLE_BITS_32 = 2 << AVS_CS_SHIFT_SAMPLE_BITS, - - AVS_CS_PLANAR_MASK = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_BGR | AVS_CS_SAMPLE_BITS_MASK | AVS_CS_SUB_HEIGHT_MASK | AVS_CS_SUB_WIDTH_MASK, - AVS_CS_PLANAR_FILTER = ~( AVS_CS_VPLANEFIRST | AVS_CS_UPLANEFIRST )}; - - // Specific colorformats -enum { - AVS_CS_UNKNOWN = 0, - AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED, - AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED, - AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED, - // AVS_CS_YV12 = 1<<3 Reserved - // AVS_CS_I420 = 1<<4 Reserved - AVS_CS_RAW32 = 1<<5 | AVS_CS_INTERLEAVED, - - AVS_CS_YV24 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_1, // YVU 4:4:4 planar - AVS_CS_YV16 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:2 planar - AVS_CS_YV12 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YVU 4:2:0 planar - AVS_CS_I420 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_UPLANEFIRST | AVS_CS_SUB_HEIGHT_2 | AVS_CS_SUB_WIDTH_2, // YUV 4:2:0 planar - AVS_CS_IYUV = AVS_CS_I420, - AVS_CS_YV411 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_1 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:1 planar - AVS_CS_YUV9 = AVS_CS_PLANAR | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 | AVS_CS_VPLANEFIRST | AVS_CS_SUB_HEIGHT_4 | AVS_CS_SUB_WIDTH_4, // YVU 4:1:0 planar - AVS_CS_Y8 = AVS_CS_PLANAR | AVS_CS_INTERLEAVED | AVS_CS_YUV | AVS_CS_SAMPLE_BITS_8 // Y 4:0:0 planar -}; - -enum { - AVS_IT_BFF = 1<<0, - AVS_IT_TFF = 1<<1, - AVS_IT_FIELDBASED = 1<<2}; - -enum { - AVS_FILTER_TYPE=1, - AVS_FILTER_INPUT_COLORSPACE=2, - AVS_FILTER_OUTPUT_TYPE=9, - AVS_FILTER_NAME=4, - AVS_FILTER_AUTHOR=5, - AVS_FILTER_VERSION=6, - AVS_FILTER_ARGS=7, - AVS_FILTER_ARGS_INFO=8, - AVS_FILTER_ARGS_DESCRIPTION=10, - AVS_FILTER_DESCRIPTION=11}; - -enum { //SUBTYPES - AVS_FILTER_TYPE_AUDIO=1, - AVS_FILTER_TYPE_VIDEO=2, - AVS_FILTER_OUTPUT_TYPE_SAME=3, - AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4}; - -enum { - AVS_CACHE_NOTHING=0, - AVS_CACHE_RANGE=1, - AVS_CACHE_ALL=2, - AVS_CACHE_AUDIO=3, - AVS_CACHE_AUDIO_NONE=4, - AVS_CACHE_AUDIO_AUTO=5 - }; - -#define AVS_FRAME_ALIGN 16 - -typedef struct AVS_Clip AVS_Clip; -typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment; - -///////////////////////////////////////////////////////////////////// -// -// AVS_VideoInfo -// - -// AVS_VideoInfo is layed out identicly to VideoInfo -typedef struct AVS_VideoInfo { - int width, height; // width=0 means no video - unsigned fps_numerator, fps_denominator; - int num_frames; - - int pixel_type; - - int audio_samples_per_second; // 0 means no audio - int sample_type; - INT64 num_audio_samples; - int nchannels; - - // Imagetype properties - - int image_type; -} AVS_VideoInfo; - -// useful functions of the above -AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p) - { return (p->width!=0); } - -AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p) - { return (p->audio_samples_per_second!=0); } - -AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p) - { return !!(p->pixel_type&AVS_CS_BGR); } - -AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p) - { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties - -AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; } - -AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p) - { return !!(p->pixel_type&AVS_CS_YUV ); } - -AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; } - -AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24 & AVS_CS_PLANAR_FILTER); } - -AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16 & AVS_CS_PLANAR_FILTER); } - -AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12 & AVS_CS_PLANAR_FILTER); } - -AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); } - -AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8 & AVS_CS_PLANAR_FILTER); } - -AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property) - { return ((p->pixel_type & property)==property ); } - -AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p) - { return !!(p->pixel_type & AVS_CS_PLANAR); } - -AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space) - { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); } - -AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p) - { return !!(p->image_type & AVS_IT_FIELDBASED); } - -AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p) - { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); } - -AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p) - { return !!(p->image_type & AVS_IT_BFF); } - -AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p) - { return !!(p->image_type & AVS_IT_TFF); } - -AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p) -{ - switch (p->pixel_type) { - case AVS_CS_BGR24: return 24; - case AVS_CS_BGR32: return 32; - case AVS_CS_YUY2: return 16; - case AVS_CS_YV12: - case AVS_CS_I420: return 12; - default: return 0; - } -} -AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels) - { return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes - -AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p) - { return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images - -AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi) - { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); } - -AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p) - { return p->audio_samples_per_second; } - - -AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p) -{ - switch (p->sample_type) { - case AVS_SAMPLE_INT8: return sizeof(signed char); - case AVS_SAMPLE_INT16: return sizeof(signed short); - case AVS_SAMPLE_INT24: return 3; - case AVS_SAMPLE_INT32: return sizeof(signed int); - case AVS_SAMPLE_FLOAT: return sizeof(float); - default: return 0; - } -} -AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p) - { return p->nchannels*avs_bytes_per_channel_sample(p);} - -AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames) - { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); } - -AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples) - { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); } - -AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes) - { return bytes / avs_bytes_per_audio_sample(p); } - -AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples) - { return samples * avs_bytes_per_audio_sample(p); } - -AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p) - { return p->nchannels; } - -AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p) - { return p->sample_type;} - -// useful mutator -AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property) - { p->image_type|=property; } - -AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property) - { p->image_type&=~property; } - -AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased) - { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; } - -AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator) -{ - unsigned x=numerator, y=denominator; - while (y) { // find gcd - unsigned t = x%y; x = y; y = t; - } - p->fps_numerator = numerator/x; - p->fps_denominator = denominator/x; -} - -AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y) -{ - return (x->pixel_type == y->pixel_type) - || (avs_is_yv12(x) && avs_is_yv12(y)); -} - -///////////////////////////////////////////////////////////////////// -// -// AVS_VideoFrame -// - -// VideoFrameBuffer holds information about a memory block which is used -// for video data. For efficiency, instances of this class are not deleted -// when the refcount reaches zero; instead they're stored in a linked list -// to be reused. The instances are deleted when the corresponding AVS -// file is closed. - -// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer -// DO NOT USE THIS STRUCTURE DIRECTLY -typedef struct AVS_VideoFrameBuffer { - BYTE * data; - int data_size; - // sequence_number is incremented every time the buffer is changed, so - // that stale views can tell they're no longer valid. - volatile long sequence_number; - - volatile long refcount; -} AVS_VideoFrameBuffer; - -// VideoFrame holds a "window" into a VideoFrameBuffer. - -// AVS_VideoFrame is layed out identicly to IVideoFrame -// DO NOT USE THIS STRUCTURE DIRECTLY -typedef struct AVS_VideoFrame { - volatile long refcount; - AVS_VideoFrameBuffer * vfb; - int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture. - int row_sizeUV, heightUV; -} AVS_VideoFrame; - -// Access functions for AVS_VideoFrame -AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) { - return p->pitch;} - -AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) { - switch (plane) { - case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;} - return p->pitch;} - -AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) { - return p->row_size; } - -AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) { - int r; - switch (plane) { - case AVS_PLANAR_U: case AVS_PLANAR_V: - if (p->pitchUV) return p->row_sizeUV; - else return 0; - case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED: - if (p->pitchUV) { - r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize - if (r < p->pitchUV) - return r; - return p->row_sizeUV; - } else return 0; - case AVS_PLANAR_Y_ALIGNED: - r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize - if (r <= p->pitch) - return r; - return p->row_size; - } - return p->row_size; -} - -AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) { - return p->height;} - -AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) { - switch (plane) { - case AVS_PLANAR_U: case AVS_PLANAR_V: - if (p->pitchUV) return p->heightUV; - return 0; - } - return p->height;} - -AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) { - return p->vfb->data + p->offset;} - -AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane) -{ - switch (plane) { - case AVS_PLANAR_U: return p->vfb->data + p->offsetU; - case AVS_PLANAR_V: return p->vfb->data + p->offsetV; - default: return p->vfb->data + p->offset;} -} - -AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) { - return (p->refcount == 1 && p->vfb->refcount == 1);} - -AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) -{ - if (avs_is_writable(p)) { - ++p->vfb->sequence_number; - return p->vfb->data + p->offset; - } else - return 0; -} - -AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane) -{ - if (plane==AVS_PLANAR_Y && avs_is_writable(p)) { - ++p->vfb->sequence_number; - return p->vfb->data + p->offset; - } else if (plane==AVS_PLANAR_Y) { - return 0; - } else { - switch (plane) { - case AVS_PLANAR_U: return p->vfb->data + p->offsetU; - case AVS_PLANAR_V: return p->vfb->data + p->offsetV; - default: return p->vfb->data + p->offset; - } - } -} - - -AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *); -// makes a shallow copy of a video frame -AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *); - -#ifndef AVSC_NO_DECLSPEC -AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f) - {avs_release_video_frame(f);} -AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f) - {return avs_copy_video_frame(f);} -#endif - -///////////////////////////////////////////////////////////////////// -// -// AVS_Value -// - -// Treat AVS_Value as a fat pointer. That is use avs_copy_value -// and avs_release_value appropiaty as you would if AVS_Value was -// a pointer. - -// To maintain source code compatibility with future versions of the -// avisynth_c API don't use the AVS_Value directly. Use the helper -// functions below. - -// AVS_Value is layed out identicly to AVSValue -typedef struct AVS_Value AVS_Value; -struct AVS_Value { - short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong - // for some function e'rror - short array_size; - union { - void * clip; // do not use directly, use avs_take_clip - char boolean; - int integer; - float floating_pt; - const char * string; - const AVS_Value * array; - } d; -}; - -// AVS_Value should be initilized with avs_void. -// Should also set to avs_void after the value is released -// with avs_copy_value. Consider it the equalvent of setting -// a pointer to NULL -static const AVS_Value avs_void = {'v'}; - -AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src); -AVSC_API(void, avs_release_value)(AVS_Value); - -AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; } -AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; } -AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; } -AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; } -AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; } -AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; } -AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; } -AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; } - -AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *); -AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *); - -AVSC_INLINE int avs_as_bool(AVS_Value v) - { return v.d.boolean; } -AVSC_INLINE int avs_as_int(AVS_Value v) - { return v.d.integer; } -AVSC_INLINE const char * avs_as_string(AVS_Value v) - { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; } -AVSC_INLINE double avs_as_float(AVS_Value v) - { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; } -AVSC_INLINE const char * avs_as_error(AVS_Value v) - { return avs_is_error(v) ? v.d.string : 0; } -AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v) - { return v.d.array; } -AVSC_INLINE int avs_array_size(AVS_Value v) - { return avs_is_array(v) ? v.array_size : 1; } -AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index) - { return avs_is_array(v) ? v.d.array[index] : v; } - -// only use these functions on an AVS_Value that does not already have -// an active value. Remember, treat AVS_Value as a fat pointer. -AVSC_INLINE AVS_Value avs_new_value_bool(int v0) - { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; } -AVSC_INLINE AVS_Value avs_new_value_int(int v0) - { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; } -AVSC_INLINE AVS_Value avs_new_value_string(const char * v0) - { AVS_Value v; v.type = 's'; v.d.string = v0; return v; } -AVSC_INLINE AVS_Value avs_new_value_float(float v0) - { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;} -AVSC_INLINE AVS_Value avs_new_value_error(const char * v0) - { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; } -#ifndef AVSC_NO_DECLSPEC -AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0) - { AVS_Value v; avs_set_to_clip(&v, v0); return v; } -#endif -AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size) - { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; } - -///////////////////////////////////////////////////////////////////// -// -// AVS_Clip -// - -AVSC_API(void, avs_release_clip)(AVS_Clip *); -AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *); - -AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error - -AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *); - -AVSC_API(int, avs_get_version)(AVS_Clip *); - -AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n); -// The returned video frame must be released with avs_release_video_frame - -AVSC_API(int, avs_get_parity)(AVS_Clip *, int n); -// return field parity if field_based, else parity of first field in frame - -AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf, - INT64 start, INT64 count); -// start and count are in samples - -AVSC_API(int, avs_set_cache_hints)(AVS_Clip *, - int cachehints, int frame_range); - -// This is the callback type used by avs_add_function -typedef AVS_Value (AVSC_CC * AVS_ApplyFunc) - (AVS_ScriptEnvironment *, AVS_Value args, void * user_data); - -typedef struct AVS_FilterInfo AVS_FilterInfo; -struct AVS_FilterInfo -{ - // these members should not be modified outside of the AVS_ApplyFunc callback - AVS_Clip * child; - AVS_VideoInfo vi; - AVS_ScriptEnvironment * env; - AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n); - int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n); - int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf, - INT64 start, INT64 count); - int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints, - int frame_range); - void (AVSC_CC * free_filter)(AVS_FilterInfo *); - - // Should be set when ever there is an error to report. - // It is cleared before any of the above methods are called - const char * error; - // this is to store whatever and may be modified at will - void * user_data; -}; - -// Create a new filter -// fi is set to point to the AVS_FilterInfo so that you can -// modify it once it is initilized. -// store_child should generally be set to true. If it is not -// set than ALL methods (the function pointers) must be defined -// If it is set than you do not need to worry about freeing the child -// clip. -AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e, - AVS_FilterInfo * * fi, - AVS_Value child, int store_child); - -///////////////////////////////////////////////////////////////////// -// -// AVS_ScriptEnvironment -// - -// For GetCPUFlags. These are backwards-compatible with those in VirtualDub. -enum { - /* slowest CPU to support extension */ - AVS_CPU_FORCE = 0x01, // N/A - AVS_CPU_FPU = 0x02, // 386/486DX - AVS_CPU_MMX = 0x04, // P55C, K6, PII - AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon - AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP - AVS_CPU_SSE2 = 0x20, // PIV, Hammer - AVS_CPU_3DNOW = 0x40, // K6-2 - AVS_CPU_3DNOW_EXT = 0x80, // Athlon - AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2, - // which only Hammer will have anyway) - AVS_CPUF_SSE3 = 0x100, // PIV+, K8 Venice - AVS_CPUF_SSSE3 = 0x200, // Core 2 - AVS_CPUF_SSE4 = 0x400, // Penryn, Wolfdale, Yorkfield - AVS_CPUF_SSE4_1 = 0x400, - AVS_CPUF_SSE4_2 = 0x800, // Nehalem -}; - -AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error - -AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *); -AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version); - -AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length); -AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...); - -AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, void* val); - // note: val is really a va_list; I hope everyone typedefs va_list to a pointer - -AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *, - const char * name, const char * params, - AVS_ApplyFunc apply, void * user_data); - -AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name); - -AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name, - AVS_Value args, const char** arg_names); -// The returned value must be be released with avs_release_value - -AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name); -// The returned value must be be released with avs_release_value - -AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val); - -AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val); - -//void avs_push_context(AVS_ScriptEnvironment *, int level=0); -//void avs_pop_context(AVS_ScriptEnvironment *); - -AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *, - const AVS_VideoInfo * vi, int align); -// align should be at least 16 - -#ifndef AVSC_NO_DECLSPEC -AVSC_INLINE -AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env, - const AVS_VideoInfo * vi) - {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);} - -AVSC_INLINE -AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env, - const AVS_VideoInfo * vi) - {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);} -#endif - - -AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf); - -AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, BYTE* dstp, int dst_pitch, const BYTE* srcp, int src_pitch, int row_size, int height); - -typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env); -AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data); - -AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height); -// The returned video frame must be be released - -AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem); - -AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir); - -// avisynth.dll exports this; it's a way to use it as a library, without -// writing an AVS script or without going through AVIFile. -AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version); - -// this symbol is the entry point for the plugin and must -// be defined -AVSC_EXPORT -const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env); - - -AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *); - - -AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV); -// The returned video frame must be be released - -#ifdef AVSC_NO_DECLSPEC -// use LoadLibrary and related functions to dynamically load Avisynth instead of declspec(dllimport) -/* - The following functions needs to have been declared, probably from windows.h - - void* malloc(size_t) - void free(void*); - - HMODULE LoadLibrary(const char*); - void* GetProcAddress(HMODULE, const char*); - FreeLibrary(HMODULE); -*/ - - -typedef struct AVS_Library AVS_Library; - -#define AVSC_DECLARE_FUNC(name) name##_func name - -struct AVS_Library { - HMODULE handle; - - AVSC_DECLARE_FUNC(avs_add_function); - AVSC_DECLARE_FUNC(avs_at_exit); - AVSC_DECLARE_FUNC(avs_bit_blt); - AVSC_DECLARE_FUNC(avs_check_version); - AVSC_DECLARE_FUNC(avs_clip_get_error); - AVSC_DECLARE_FUNC(avs_copy_clip); - AVSC_DECLARE_FUNC(avs_copy_value); - AVSC_DECLARE_FUNC(avs_copy_video_frame); - AVSC_DECLARE_FUNC(avs_create_script_environment); - AVSC_DECLARE_FUNC(avs_delete_script_environment); - AVSC_DECLARE_FUNC(avs_function_exists); - AVSC_DECLARE_FUNC(avs_get_audio); - AVSC_DECLARE_FUNC(avs_get_cpu_flags); - AVSC_DECLARE_FUNC(avs_get_error); - AVSC_DECLARE_FUNC(avs_get_frame); - AVSC_DECLARE_FUNC(avs_get_parity); - AVSC_DECLARE_FUNC(avs_get_var); - AVSC_DECLARE_FUNC(avs_get_version); - AVSC_DECLARE_FUNC(avs_get_video_info); - AVSC_DECLARE_FUNC(avs_invoke); - AVSC_DECLARE_FUNC(avs_make_writable); - AVSC_DECLARE_FUNC(avs_new_c_filter); - AVSC_DECLARE_FUNC(avs_new_video_frame_a); - AVSC_DECLARE_FUNC(avs_release_clip); - AVSC_DECLARE_FUNC(avs_release_value); - AVSC_DECLARE_FUNC(avs_release_video_frame); - AVSC_DECLARE_FUNC(avs_save_string); - AVSC_DECLARE_FUNC(avs_set_cache_hints); - AVSC_DECLARE_FUNC(avs_set_global_var); - AVSC_DECLARE_FUNC(avs_set_memory_max); - AVSC_DECLARE_FUNC(avs_set_to_clip); - AVSC_DECLARE_FUNC(avs_set_var); - AVSC_DECLARE_FUNC(avs_set_working_dir); - AVSC_DECLARE_FUNC(avs_sprintf); - AVSC_DECLARE_FUNC(avs_subframe); - AVSC_DECLARE_FUNC(avs_subframe_planar); - AVSC_DECLARE_FUNC(avs_take_clip); - AVSC_DECLARE_FUNC(avs_vsprintf); -}; - -#undef AVSC_DECLARE_FUNC - - -AVSC_INLINE AVS_Library * avs_load_library() { - AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library)); - if (library == NULL) - return NULL; - library->handle = LoadLibrary("avisynth"); - if (library->handle == NULL) - goto fail; - -#define __AVSC_STRINGIFY(x) #x -#define AVSC_STRINGIFY(x) __AVSC_STRINGIFY(x) -#define AVSC_LOAD_FUNC(name) {\ - library->name = (name##_func) GetProcAddress(library->handle, AVSC_STRINGIFY(name));\ - if (library->name == NULL)\ - goto fail;\ -} - - AVSC_LOAD_FUNC(avs_add_function); - AVSC_LOAD_FUNC(avs_at_exit); - AVSC_LOAD_FUNC(avs_bit_blt); - AVSC_LOAD_FUNC(avs_check_version); - AVSC_LOAD_FUNC(avs_clip_get_error); - AVSC_LOAD_FUNC(avs_copy_clip); - AVSC_LOAD_FUNC(avs_copy_value); - AVSC_LOAD_FUNC(avs_copy_video_frame); - AVSC_LOAD_FUNC(avs_create_script_environment); - AVSC_LOAD_FUNC(avs_delete_script_environment); - AVSC_LOAD_FUNC(avs_function_exists); - AVSC_LOAD_FUNC(avs_get_audio); - AVSC_LOAD_FUNC(avs_get_cpu_flags); - AVSC_LOAD_FUNC(avs_get_error); - AVSC_LOAD_FUNC(avs_get_frame); - AVSC_LOAD_FUNC(avs_get_parity); - AVSC_LOAD_FUNC(avs_get_var); - AVSC_LOAD_FUNC(avs_get_version); - AVSC_LOAD_FUNC(avs_get_video_info); - AVSC_LOAD_FUNC(avs_invoke); - AVSC_LOAD_FUNC(avs_make_writable); - AVSC_LOAD_FUNC(avs_new_c_filter); - AVSC_LOAD_FUNC(avs_new_video_frame_a); - AVSC_LOAD_FUNC(avs_release_clip); - AVSC_LOAD_FUNC(avs_release_value); - AVSC_LOAD_FUNC(avs_release_video_frame); - AVSC_LOAD_FUNC(avs_save_string); - AVSC_LOAD_FUNC(avs_set_cache_hints); - AVSC_LOAD_FUNC(avs_set_global_var); - AVSC_LOAD_FUNC(avs_set_memory_max); - AVSC_LOAD_FUNC(avs_set_to_clip); - AVSC_LOAD_FUNC(avs_set_var); - AVSC_LOAD_FUNC(avs_set_working_dir); - AVSC_LOAD_FUNC(avs_sprintf); - AVSC_LOAD_FUNC(avs_subframe); - AVSC_LOAD_FUNC(avs_subframe_planar); - AVSC_LOAD_FUNC(avs_take_clip); - AVSC_LOAD_FUNC(avs_vsprintf); - -#undef __AVSC_STRINGIFY -#undef AVSC_STRINGIFY -#undef AVSC_LOAD_FUNC - - return library; - -fail: - free(library); - return NULL; -} - -AVSC_INLINE void avs_free_library(AVS_Library *library) { - if (library == NULL) - return; - FreeLibrary(library->handle); - free(library); -} -#endif - -#endif diff --git a/compat/avisynth/avisynth_c_25.h b/compat/avisynth/avisynth_c_25.h deleted file mode 100644 index 9288761331..0000000000 --- a/compat/avisynth/avisynth_c_25.h +++ /dev/null @@ -1,68 +0,0 @@ -// Copyright (c) 2011 FFmpegSource Project -// -// Permission is hereby granted, free of charge, to any person obtaining a copy -// of this software and associated documentation files (the "Software"), to deal -// in the Software without restriction, including without limitation the rights -// to use, copy, modify, merge, publish, distribute, sublicense, and/or sell -// copies of the Software, and to permit persons to whom the Software is -// furnished to do so, subject to the following conditions: -// -// The above copyright notice and this permission notice shall be included in -// all copies or substantial portions of the Software. -// -// THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR -// IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, -// FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE -// AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER -// LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, -// OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN -// THE SOFTWARE. - -/* these are defines/functions that are used and were changed in the switch to 2.6 - * and are needed to maintain full compatility with 2.5 */ - -enum { - AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar - AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar -}; - -AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) { - switch (plane) - { - case AVS_PLANAR_U: case AVS_PLANAR_V: - if (p->pitchUV) - return p->height>>1; - return 0; - } - return p->height;} - -AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) { - int r; - switch (plane) - { - case AVS_PLANAR_U: case AVS_PLANAR_V: - if (p->pitchUV) - return p->row_size>>1; - else - return 0; - case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED: - if (p->pitchUV) - { - r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize - if (r < p->pitchUV) - return r; - return p->row_size>>1; - } - else - return 0; - case AVS_PLANAR_Y_ALIGNED: - r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize - if (r <= p->pitch) - return r; - return p->row_size; - } - return p->row_size; -} - -AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p) - { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); } diff --git a/compat/avisynth/avxsynth_c.h b/compat/avisynth/avxsynth_c.h deleted file mode 100644 index b20f46081e..0000000000 --- a/compat/avisynth/avxsynth_c.h +++ /dev/null @@ -1,728 +0,0 @@ -// Avisynth C Interface Version 0.20 -// Copyright 2003 Kevin Atkinson - -// This program is free software; you can redistribute it and/or modify -// it under the terms of the GNU General Public License as published by -// the Free Software Foundation; either version 2 of the License, or -// (at your option) any later version. -// -// This program is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -// GNU General Public License for more details. -// -// You should have received a copy of the GNU General Public License -// along with this program; if not, write to the Free Software -// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, -// MA 02110-1301 USA, or visit -// http://www.gnu.org/copyleft/gpl.html . -// -// As a special exception, I give you permission to link to the -// Avisynth C interface with independent modules that communicate with -// the Avisynth C interface solely through the interfaces defined in -// avisynth_c.h, regardless of the license terms of these independent -// modules, and to copy and distribute the resulting combined work -// under terms of your choice, provided that every copy of the -// combined work is accompanied by a complete copy of the source code -// of the Avisynth C interface and Avisynth itself (with the version -// used to produce the combined work), being distributed under the -// terms of the GNU General Public License plus this exception. An -// independent module is a module which is not derived from or based -// on Avisynth C Interface, such as 3rd-party filters, import and -// export plugins, or graphical user interfaces. - -#ifndef __AVXSYNTH_C__ -#define __AVXSYNTH_C__ - -#include "windowsPorts/windows2linux.h" -#include - -#ifdef __cplusplus -# define EXTERN_C extern "C" -#else -# define EXTERN_C -#endif - -#define AVSC_USE_STDCALL 1 - -#ifndef AVSC_USE_STDCALL -# define AVSC_CC __cdecl -#else -# define AVSC_CC __stdcall -#endif - -#define AVSC_INLINE static __inline - -#ifdef AVISYNTH_C_EXPORTS -# define AVSC_EXPORT EXTERN_C -# define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name -#else -# define AVSC_EXPORT EXTERN_C __declspec(dllexport) -# ifndef AVSC_NO_DECLSPEC -# define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name -# else -# define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func) -# endif -#endif - -#ifdef __GNUC__ -typedef long long int INT64; -#else -typedef __int64 INT64; -#endif - - -///////////////////////////////////////////////////////////////////// -// -// Constants -// - -#ifndef __AVXSYNTH_H__ -enum { AVISYNTH_INTERFACE_VERSION = 3 }; -#endif - -enum {AVS_SAMPLE_INT8 = 1<<0, - AVS_SAMPLE_INT16 = 1<<1, - AVS_SAMPLE_INT24 = 1<<2, - AVS_SAMPLE_INT32 = 1<<3, - AVS_SAMPLE_FLOAT = 1<<4}; - -enum {AVS_PLANAR_Y=1<<0, - AVS_PLANAR_U=1<<1, - AVS_PLANAR_V=1<<2, - AVS_PLANAR_ALIGNED=1<<3, - AVS_PLANAR_Y_ALIGNED=AVS_PLANAR_Y|AVS_PLANAR_ALIGNED, - AVS_PLANAR_U_ALIGNED=AVS_PLANAR_U|AVS_PLANAR_ALIGNED, - AVS_PLANAR_V_ALIGNED=AVS_PLANAR_V|AVS_PLANAR_ALIGNED}; - - // Colorspace properties. -enum {AVS_CS_BGR = 1<<28, - AVS_CS_YUV = 1<<29, - AVS_CS_INTERLEAVED = 1<<30, - AVS_CS_PLANAR = 1<<31}; - - // Specific colorformats -enum { - AVS_CS_UNKNOWN = 0, - AVS_CS_BGR24 = 1<<0 | AVS_CS_BGR | AVS_CS_INTERLEAVED, - AVS_CS_BGR32 = 1<<1 | AVS_CS_BGR | AVS_CS_INTERLEAVED, - AVS_CS_YUY2 = 1<<2 | AVS_CS_YUV | AVS_CS_INTERLEAVED, - AVS_CS_YV12 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR, // y-v-u, planar - AVS_CS_I420 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR, // y-u-v, planar - AVS_CS_IYUV = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR // same as above -}; - -enum { - AVS_IT_BFF = 1<<0, - AVS_IT_TFF = 1<<1, - AVS_IT_FIELDBASED = 1<<2}; - -enum { - AVS_FILTER_TYPE=1, - AVS_FILTER_INPUT_COLORSPACE=2, - AVS_FILTER_OUTPUT_TYPE=9, - AVS_FILTER_NAME=4, - AVS_FILTER_AUTHOR=5, - AVS_FILTER_VERSION=6, - AVS_FILTER_ARGS=7, - AVS_FILTER_ARGS_INFO=8, - AVS_FILTER_ARGS_DESCRIPTION=10, - AVS_FILTER_DESCRIPTION=11}; - -enum { //SUBTYPES - AVS_FILTER_TYPE_AUDIO=1, - AVS_FILTER_TYPE_VIDEO=2, - AVS_FILTER_OUTPUT_TYPE_SAME=3, - AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4}; - -enum { - AVS_CACHE_NOTHING=0, - AVS_CACHE_RANGE=1, - AVS_CACHE_ALL=2, - AVS_CACHE_AUDIO=3, - AVS_CACHE_AUDIO_NONE=4, - AVS_CACHE_AUDIO_AUTO=5 -}; - -#define AVS_FRAME_ALIGN 16 - -typedef struct AVS_Clip AVS_Clip; -typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment; - -///////////////////////////////////////////////////////////////////// -// -// AVS_VideoInfo -// - -// AVS_VideoInfo is layed out identicly to VideoInfo -typedef struct AVS_VideoInfo { - int width, height; // width=0 means no video - unsigned fps_numerator, fps_denominator; - int num_frames; - - int pixel_type; - - int audio_samples_per_second; // 0 means no audio - int sample_type; - INT64 num_audio_samples; - int nchannels; - - // Imagetype properties - - int image_type; -} AVS_VideoInfo; - -// useful functions of the above -AVSC_INLINE int avs_has_video(const AVS_VideoInfo * p) - { return (p->width!=0); } - -AVSC_INLINE int avs_has_audio(const AVS_VideoInfo * p) - { return (p->audio_samples_per_second!=0); } - -AVSC_INLINE int avs_is_rgb(const AVS_VideoInfo * p) - { return !!(p->pixel_type&AVS_CS_BGR); } - -AVSC_INLINE int avs_is_rgb24(const AVS_VideoInfo * p) - { return (p->pixel_type&AVS_CS_BGR24)==AVS_CS_BGR24; } // Clear out additional properties - -AVSC_INLINE int avs_is_rgb32(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_BGR32) == AVS_CS_BGR32 ; } - -AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p) - { return !!(p->pixel_type&AVS_CS_YUV ); } - -AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p) - { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; } - -AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p) - { return ((p->pixel_type & AVS_CS_YV12) == AVS_CS_YV12)||((p->pixel_type & AVS_CS_I420) == AVS_CS_I420); } - -AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space) - { return ((p->pixel_type & c_space) == c_space); } - -AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property) - { return ((p->pixel_type & property)==property ); } - -AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p) - { return !!(p->pixel_type & AVS_CS_PLANAR); } - -AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p) - { return !!(p->image_type & AVS_IT_FIELDBASED); } - -AVSC_INLINE int avs_is_parity_known(const AVS_VideoInfo * p) - { return ((p->image_type & AVS_IT_FIELDBASED)&&(p->image_type & (AVS_IT_BFF | AVS_IT_TFF))); } - -AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p) - { return !!(p->image_type & AVS_IT_BFF); } - -AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p) - { return !!(p->image_type & AVS_IT_TFF); } - -AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p) -{ - switch (p->pixel_type) { - case AVS_CS_BGR24: return 24; - case AVS_CS_BGR32: return 32; - case AVS_CS_YUY2: return 16; - case AVS_CS_YV12: - case AVS_CS_I420: return 12; - default: return 0; - } -} -AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels) - { return pixels * (avs_bits_per_pixel(p)>>3); } // Will work on planar images, but will return only luma planes - -AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p) - { return avs_bytes_from_pixels(p,p->width); } // Also only returns first plane on planar images - -AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi) - { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p; } return vi->height * ((avs_row_size(vi)+3) & ~3); } - -AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p) - { return p->audio_samples_per_second; } - - -AVSC_INLINE int avs_bytes_per_channel_sample(const AVS_VideoInfo * p) -{ - switch (p->sample_type) { - case AVS_SAMPLE_INT8: return sizeof(signed char); - case AVS_SAMPLE_INT16: return sizeof(signed short); - case AVS_SAMPLE_INT24: return 3; - case AVS_SAMPLE_INT32: return sizeof(signed int); - case AVS_SAMPLE_FLOAT: return sizeof(float); - default: return 0; - } -} -AVSC_INLINE int avs_bytes_per_audio_sample(const AVS_VideoInfo * p) - { return p->nchannels*avs_bytes_per_channel_sample(p);} - -AVSC_INLINE INT64 avs_audio_samples_from_frames(const AVS_VideoInfo * p, INT64 frames) - { return ((INT64)(frames) * p->audio_samples_per_second * p->fps_denominator / p->fps_numerator); } - -AVSC_INLINE int avs_frames_from_audio_samples(const AVS_VideoInfo * p, INT64 samples) - { return (int)(samples * (INT64)p->fps_numerator / (INT64)p->fps_denominator / (INT64)p->audio_samples_per_second); } - -AVSC_INLINE INT64 avs_audio_samples_from_bytes(const AVS_VideoInfo * p, INT64 bytes) - { return bytes / avs_bytes_per_audio_sample(p); } - -AVSC_INLINE INT64 avs_bytes_from_audio_samples(const AVS_VideoInfo * p, INT64 samples) - { return samples * avs_bytes_per_audio_sample(p); } - -AVSC_INLINE int avs_audio_channels(const AVS_VideoInfo * p) - { return p->nchannels; } - -AVSC_INLINE int avs_sample_type(const AVS_VideoInfo * p) - { return p->sample_type;} - -// useful mutator -AVSC_INLINE void avs_set_property(AVS_VideoInfo * p, int property) - { p->image_type|=property; } - -AVSC_INLINE void avs_clear_property(AVS_VideoInfo * p, int property) - { p->image_type&=~property; } - -AVSC_INLINE void avs_set_field_based(AVS_VideoInfo * p, int isfieldbased) - { if (isfieldbased) p->image_type|=AVS_IT_FIELDBASED; else p->image_type&=~AVS_IT_FIELDBASED; } - -AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned denominator) -{ - unsigned x=numerator, y=denominator; - while (y) { // find gcd - unsigned t = x%y; x = y; y = t; - } - p->fps_numerator = numerator/x; - p->fps_denominator = denominator/x; -} - -AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y) -{ - return (x->pixel_type == y->pixel_type) - || (avs_is_yv12(x) && avs_is_yv12(y)); -} - -///////////////////////////////////////////////////////////////////// -// -// AVS_VideoFrame -// - -// VideoFrameBuffer holds information about a memory block which is used -// for video data. For efficiency, instances of this class are not deleted -// when the refcount reaches zero; instead they're stored in a linked list -// to be reused. The instances are deleted when the corresponding AVS -// file is closed. - -// AVS_VideoFrameBuffer is layed out identicly to VideoFrameBuffer -// DO NOT USE THIS STRUCTURE DIRECTLY -typedef struct AVS_VideoFrameBuffer { - unsigned char * data; - int data_size; - // sequence_number is incremented every time the buffer is changed, so - // that stale views can tell they're no longer valid. - long sequence_number; - - long refcount; -} AVS_VideoFrameBuffer; - -// VideoFrame holds a "window" into a VideoFrameBuffer. - -// AVS_VideoFrame is layed out identicly to IVideoFrame -// DO NOT USE THIS STRUCTURE DIRECTLY -typedef struct AVS_VideoFrame { - int refcount; - AVS_VideoFrameBuffer * vfb; - int offset, pitch, row_size, height, offsetU, offsetV, pitchUV; // U&V offsets are from top of picture. -} AVS_VideoFrame; - -// Access functions for AVS_VideoFrame -AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) { - return p->pitch;} - -AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) { - switch (plane) { - case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;} - return p->pitch;} - -AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) { - return p->row_size; } - -AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) { - int r; - switch (plane) { - case AVS_PLANAR_U: case AVS_PLANAR_V: - if (p->pitchUV) return p->row_size>>1; - else return 0; - case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED: - if (p->pitchUV) { - r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize - if (r < p->pitchUV) - return r; - return p->row_size>>1; - } else return 0; - case AVS_PLANAR_Y_ALIGNED: - r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize - if (r <= p->pitch) - return r; - return p->row_size; - } - return p->row_size; -} - -AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) { - return p->height;} - -AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) { - switch (plane) { - case AVS_PLANAR_U: case AVS_PLANAR_V: - if (p->pitchUV) return p->height>>1; - return 0; - } - return p->height;} - -AVSC_INLINE const unsigned char* avs_get_read_ptr(const AVS_VideoFrame * p) { - return p->vfb->data + p->offset;} - -AVSC_INLINE const unsigned char* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane) -{ - switch (plane) { - case AVS_PLANAR_U: return p->vfb->data + p->offsetU; - case AVS_PLANAR_V: return p->vfb->data + p->offsetV; - default: return p->vfb->data + p->offset;} -} - -AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) { - return (p->refcount == 1 && p->vfb->refcount == 1);} - -AVSC_INLINE unsigned char* avs_get_write_ptr(const AVS_VideoFrame * p) -{ - if (avs_is_writable(p)) { - ++p->vfb->sequence_number; - return p->vfb->data + p->offset; - } else - return 0; -} - -AVSC_INLINE unsigned char* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane) -{ - if (plane==AVS_PLANAR_Y && avs_is_writable(p)) { - ++p->vfb->sequence_number; - return p->vfb->data + p->offset; - } else if (plane==AVS_PLANAR_Y) { - return 0; - } else { - switch (plane) { - case AVS_PLANAR_U: return p->vfb->data + p->offsetU; - case AVS_PLANAR_V: return p->vfb->data + p->offsetV; - default: return p->vfb->data + p->offset; - } - } -} - -#if defined __cplusplus -extern "C" -{ -#endif // __cplusplus -AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *); -// makes a shallow copy of a video frame -AVSC_API(AVS_VideoFrame *, avs_copy_video_frame)(AVS_VideoFrame *); -#if defined __cplusplus -} -#endif // __cplusplus - -#ifndef AVSC_NO_DECLSPEC -AVSC_INLINE void avs_release_frame(AVS_VideoFrame * f) - {avs_release_video_frame(f);} -AVSC_INLINE AVS_VideoFrame * avs_copy_frame(AVS_VideoFrame * f) - {return avs_copy_video_frame(f);} -#endif - -///////////////////////////////////////////////////////////////////// -// -// AVS_Value -// - -// Treat AVS_Value as a fat pointer. That is use avs_copy_value -// and avs_release_value appropiaty as you would if AVS_Value was -// a pointer. - -// To maintain source code compatibility with future versions of the -// avisynth_c API don't use the AVS_Value directly. Use the helper -// functions below. - -// AVS_Value is layed out identicly to AVSValue -typedef struct AVS_Value AVS_Value; -struct AVS_Value { - short type; // 'a'rray, 'c'lip, 'b'ool, 'i'nt, 'f'loat, 's'tring, 'v'oid, or 'l'ong - // for some function e'rror - short array_size; - union { - void * clip; // do not use directly, use avs_take_clip - char boolean; - int integer; - INT64 integer64; // match addition of __int64 to avxplugin.h - float floating_pt; - const char * string; - const AVS_Value * array; - } d; -}; - -// AVS_Value should be initilized with avs_void. -// Should also set to avs_void after the value is released -// with avs_copy_value. Consider it the equalvent of setting -// a pointer to NULL -static const AVS_Value avs_void = {'v'}; - -AVSC_API(void, avs_copy_value)(AVS_Value * dest, AVS_Value src); -AVSC_API(void, avs_release_value)(AVS_Value); - -AVSC_INLINE int avs_defined(AVS_Value v) { return v.type != 'v'; } -AVSC_INLINE int avs_is_clip(AVS_Value v) { return v.type == 'c'; } -AVSC_INLINE int avs_is_bool(AVS_Value v) { return v.type == 'b'; } -AVSC_INLINE int avs_is_int(AVS_Value v) { return v.type == 'i'; } -AVSC_INLINE int avs_is_float(AVS_Value v) { return v.type == 'f' || v.type == 'i'; } -AVSC_INLINE int avs_is_string(AVS_Value v) { return v.type == 's'; } -AVSC_INLINE int avs_is_array(AVS_Value v) { return v.type == 'a'; } -AVSC_INLINE int avs_is_error(AVS_Value v) { return v.type == 'e'; } - -#if defined __cplusplus -extern "C" -{ -#endif // __cplusplus -AVSC_API(AVS_Clip *, avs_take_clip)(AVS_Value, AVS_ScriptEnvironment *); -AVSC_API(void, avs_set_to_clip)(AVS_Value *, AVS_Clip *); -#if defined __cplusplus -} -#endif // __cplusplus - -AVSC_INLINE int avs_as_bool(AVS_Value v) - { return v.d.boolean; } -AVSC_INLINE int avs_as_int(AVS_Value v) - { return v.d.integer; } -AVSC_INLINE const char * avs_as_string(AVS_Value v) - { return avs_is_error(v) || avs_is_string(v) ? v.d.string : 0; } -AVSC_INLINE double avs_as_float(AVS_Value v) - { return avs_is_int(v) ? v.d.integer : v.d.floating_pt; } -AVSC_INLINE const char * avs_as_error(AVS_Value v) - { return avs_is_error(v) ? v.d.string : 0; } -AVSC_INLINE const AVS_Value * avs_as_array(AVS_Value v) - { return v.d.array; } -AVSC_INLINE int avs_array_size(AVS_Value v) - { return avs_is_array(v) ? v.array_size : 1; } -AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index) - { return avs_is_array(v) ? v.d.array[index] : v; } - -// only use these functions on am AVS_Value that does not already have -// an active value. Remember, treat AVS_Value as a fat pointer. -AVSC_INLINE AVS_Value avs_new_value_bool(int v0) - { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; } -AVSC_INLINE AVS_Value avs_new_value_int(int v0) - { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; } -AVSC_INLINE AVS_Value avs_new_value_string(const char * v0) - { AVS_Value v; v.type = 's'; v.d.string = v0; return v; } -AVSC_INLINE AVS_Value avs_new_value_float(float v0) - { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;} -AVSC_INLINE AVS_Value avs_new_value_error(const char * v0) - { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; } -#ifndef AVSC_NO_DECLSPEC -AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0) - { AVS_Value v; avs_set_to_clip(&v, v0); return v; } -#endif -AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size) - { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; } - -///////////////////////////////////////////////////////////////////// -// -// AVS_Clip -// -#if defined __cplusplus -extern "C" -{ -#endif // __cplusplus -AVSC_API(void, avs_release_clip)(AVS_Clip *); -AVSC_API(AVS_Clip *, avs_copy_clip)(AVS_Clip *); - -AVSC_API(const char *, avs_clip_get_error)(AVS_Clip *); // return 0 if no error - -AVSC_API(const AVS_VideoInfo *, avs_get_video_info)(AVS_Clip *); - -AVSC_API(int, avs_get_version)(AVS_Clip *); - -AVSC_API(AVS_VideoFrame *, avs_get_frame)(AVS_Clip *, int n); -// The returned video frame must be released with avs_release_video_frame - -AVSC_API(int, avs_get_parity)(AVS_Clip *, int n); -// return field parity if field_based, else parity of first field in frame - -AVSC_API(int, avs_get_audio)(AVS_Clip *, void * buf, - INT64 start, INT64 count); -// start and count are in samples - -AVSC_API(int, avs_set_cache_hints)(AVS_Clip *, - int cachehints, size_t frame_range); -#if defined __cplusplus -} -#endif // __cplusplus - -// This is the callback type used by avs_add_function -typedef AVS_Value (AVSC_CC * AVS_ApplyFunc) - (AVS_ScriptEnvironment *, AVS_Value args, void * user_data); - -typedef struct AVS_FilterInfo AVS_FilterInfo; -struct AVS_FilterInfo -{ - // these members should not be modified outside of the AVS_ApplyFunc callback - AVS_Clip * child; - AVS_VideoInfo vi; - AVS_ScriptEnvironment * env; - AVS_VideoFrame * (AVSC_CC * get_frame)(AVS_FilterInfo *, int n); - int (AVSC_CC * get_parity)(AVS_FilterInfo *, int n); - int (AVSC_CC * get_audio)(AVS_FilterInfo *, void * buf, - INT64 start, INT64 count); - int (AVSC_CC * set_cache_hints)(AVS_FilterInfo *, int cachehints, - int frame_range); - void (AVSC_CC * free_filter)(AVS_FilterInfo *); - - // Should be set when ever there is an error to report. - // It is cleared before any of the above methods are called - const char * error; - // this is to store whatever and may be modified at will - void * user_data; -}; - -// Create a new filter -// fi is set to point to the AVS_FilterInfo so that you can -// modify it once it is initilized. -// store_child should generally be set to true. If it is not -// set than ALL methods (the function pointers) must be defined -// If it is set than you do not need to worry about freeing the child -// clip. -#if defined __cplusplus -extern "C" -{ -#endif // __cplusplus -AVSC_API(AVS_Clip *, avs_new_c_filter)(AVS_ScriptEnvironment * e, - AVS_FilterInfo * * fi, - AVS_Value child, int store_child); -#if defined __cplusplus -} -#endif // __cplusplus - - -///////////////////////////////////////////////////////////////////// -// -// AVS_ScriptEnvironment -// - -// For GetCPUFlags. These are backwards-compatible with those in VirtualDub. -enum { - /* slowest CPU to support extension */ - AVS_CPU_FORCE = 0x01, // N/A - AVS_CPU_FPU = 0x02, // 386/486DX - AVS_CPU_MMX = 0x04, // P55C, K6, PII - AVS_CPU_INTEGER_SSE = 0x08, // PIII, Athlon - AVS_CPU_SSE = 0x10, // PIII, Athlon XP/MP - AVS_CPU_SSE2 = 0x20, // PIV, Hammer - AVS_CPU_3DNOW = 0x40, // K6-2 - AVS_CPU_3DNOW_EXT = 0x80, // Athlon - AVS_CPU_X86_64 = 0xA0, // Hammer (note: equiv. to 3DNow + SSE2, - // which only Hammer will have anyway) -}; - -#if defined __cplusplus -extern "C" -{ -#endif // __cplusplus -AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error - -AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *); -AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version); - -AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length); -AVSC_API(char *, avs_sprintf)(AVS_ScriptEnvironment *, const char * fmt, ...); - -AVSC_API(char *, avs_vsprintf)(AVS_ScriptEnvironment *, const char * fmt, va_list val); - // note: val is really a va_list; I hope everyone typedefs va_list to a pointer - -AVSC_API(int, avs_add_function)(AVS_ScriptEnvironment *, - const char * name, const char * params, - AVS_ApplyFunc apply, void * user_data); - -AVSC_API(int, avs_function_exists)(AVS_ScriptEnvironment *, const char * name); - -AVSC_API(AVS_Value, avs_invoke)(AVS_ScriptEnvironment *, const char * name, - AVS_Value args, const char** arg_names); -// The returned value must be be released with avs_release_value - -AVSC_API(AVS_Value, avs_get_var)(AVS_ScriptEnvironment *, const char* name); -// The returned value must be be released with avs_release_value - -AVSC_API(int, avs_set_var)(AVS_ScriptEnvironment *, const char* name, AVS_Value val); - -AVSC_API(int, avs_set_global_var)(AVS_ScriptEnvironment *, const char* name, const AVS_Value val); - -//void avs_push_context(AVS_ScriptEnvironment *, int level=0); -//void avs_pop_context(AVS_ScriptEnvironment *); - -AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *, - const AVS_VideoInfo * vi, int align); -// align should be at least 16 -#if defined __cplusplus -} -#endif // __cplusplus - -#ifndef AVSC_NO_DECLSPEC -AVSC_INLINE -AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env, - const AVS_VideoInfo * vi) - {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);} - -AVSC_INLINE -AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env, - const AVS_VideoInfo * vi) - {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);} -#endif - -#if defined __cplusplus -extern "C" -{ -#endif // __cplusplus -AVSC_API(int, avs_make_writable)(AVS_ScriptEnvironment *, AVS_VideoFrame * * pvf); - -AVSC_API(void, avs_bit_blt)(AVS_ScriptEnvironment *, unsigned char* dstp, int dst_pitch, const unsigned char* srcp, int src_pitch, int row_size, int height); - -typedef void (AVSC_CC *AVS_ShutdownFunc)(void* user_data, AVS_ScriptEnvironment * env); -AVSC_API(void, avs_at_exit)(AVS_ScriptEnvironment *, AVS_ShutdownFunc function, void * user_data); - -AVSC_API(AVS_VideoFrame *, avs_subframe)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height); -// The returned video frame must be be released - -AVSC_API(int, avs_set_memory_max)(AVS_ScriptEnvironment *, int mem); - -AVSC_API(int, avs_set_working_dir)(AVS_ScriptEnvironment *, const char * newdir); - -// avisynth.dll exports this; it's a way to use it as a library, without -// writing an AVS script or without going through AVIFile. -AVSC_API(AVS_ScriptEnvironment *, avs_create_script_environment)(int version); -#if defined __cplusplus -} -#endif // __cplusplus - -// this symbol is the entry point for the plugin and must -// be defined -AVSC_EXPORT -const char * AVSC_CC avisynth_c_plugin_init(AVS_ScriptEnvironment* env); - - -#if defined __cplusplus -extern "C" -{ -#endif // __cplusplus -AVSC_API(void, avs_delete_script_environment)(AVS_ScriptEnvironment *); - - -AVSC_API(AVS_VideoFrame *, avs_subframe_planar)(AVS_ScriptEnvironment *, AVS_VideoFrame * src, int rel_offset, int new_pitch, int new_row_size, int new_height, int rel_offsetU, int rel_offsetV, int new_pitchUV); -// The returned video frame must be be released -#if defined __cplusplus -} -#endif // __cplusplus - -#endif //__AVXSYNTH_C__ diff --git a/compat/avisynth/windowsPorts/basicDataTypeConversions.h b/compat/avisynth/windowsPorts/basicDataTypeConversions.h deleted file mode 100644 index ff367d5a2a..0000000000 --- a/compat/avisynth/windowsPorts/basicDataTypeConversions.h +++ /dev/null @@ -1,85 +0,0 @@ -#ifndef __DATA_TYPE_CONVERSIONS_H__ -#define __DATA_TYPE_CONVERSIONS_H__ - -#include -#include - -#ifdef __cplusplus -namespace avxsynth { -#endif // __cplusplus - -typedef int64_t __int64; -typedef int32_t __int32; -#ifdef __cplusplus -typedef bool BOOL; -#else -typedef uint32_t BOOL; -#endif // __cplusplus -typedef void* HMODULE; -typedef void* LPVOID; -typedef void* PVOID; -typedef PVOID HANDLE; -typedef HANDLE HWND; -typedef HANDLE HINSTANCE; -typedef void* HDC; -typedef void* HBITMAP; -typedef void* HICON; -typedef void* HFONT; -typedef void* HGDIOBJ; -typedef void* HBRUSH; -typedef void* HMMIO; -typedef void* HACMSTREAM; -typedef void* HACMDRIVER; -typedef void* HIC; -typedef void* HACMOBJ; -typedef HACMSTREAM* LPHACMSTREAM; -typedef void* HACMDRIVERID; -typedef void* LPHACMDRIVER; -typedef unsigned char BYTE; -typedef BYTE* LPBYTE; -typedef char TCHAR; -typedef TCHAR* LPTSTR; -typedef const TCHAR* LPCTSTR; -typedef char* LPSTR; -typedef LPSTR LPOLESTR; -typedef const char* LPCSTR; -typedef LPCSTR LPCOLESTR; -typedef wchar_t WCHAR; -typedef unsigned short WORD; -typedef unsigned int UINT; -typedef UINT MMRESULT; -typedef uint32_t DWORD; -typedef DWORD COLORREF; -typedef DWORD FOURCC; -typedef DWORD HRESULT; -typedef DWORD* LPDWORD; -typedef DWORD* DWORD_PTR; -typedef int32_t LONG; -typedef int32_t* LONG_PTR; -typedef LONG_PTR LRESULT; -typedef uint32_t ULONG; -typedef uint32_t* ULONG_PTR; -//typedef __int64_t intptr_t; -typedef uint64_t _fsize_t; - - -// -// Structures -// - -typedef struct _GUID { - DWORD Data1; - WORD Data2; - WORD Data3; - BYTE Data4[8]; -} GUID; - -typedef GUID REFIID; -typedef GUID CLSID; -typedef CLSID* LPCLSID; -typedef GUID IID; - -#ifdef __cplusplus -}; // namespace avxsynth -#endif // __cplusplus -#endif // __DATA_TYPE_CONVERSIONS_H__ diff --git a/compat/avisynth/windowsPorts/windows2linux.h b/compat/avisynth/windowsPorts/windows2linux.h deleted file mode 100644 index 5476afe847..0000000000 --- a/compat/avisynth/windowsPorts/windows2linux.h +++ /dev/null @@ -1,77 +0,0 @@ -#ifndef __WINDOWS2LINUX_H__ -#define __WINDOWS2LINUX_H__ - -/* - * LINUX SPECIFIC DEFINITIONS -*/ -// -// Data types conversions -// -#include -#include -#include "basicDataTypeConversions.h" - -#ifdef __cplusplus -namespace avxsynth { -#endif // __cplusplus -// -// purposefully define the following MSFT definitions -// to mean nothing (as they do not mean anything on Linux) -// -#define __stdcall -#define __cdecl -#define noreturn -#define __declspec(x) -#define STDAPI extern "C" HRESULT -#define STDMETHODIMP HRESULT __stdcall -#define STDMETHODIMP_(x) x __stdcall - -#define STDMETHOD(x) virtual HRESULT x -#define STDMETHOD_(a, x) virtual a x - -#ifndef TRUE -#define TRUE true -#endif - -#ifndef FALSE -#define FALSE false -#endif - -#define S_OK (0x00000000) -#define S_FALSE (0x00000001) -#define E_NOINTERFACE (0X80004002) -#define E_POINTER (0x80004003) -#define E_FAIL (0x80004005) -#define E_OUTOFMEMORY (0x8007000E) - -#define INVALID_HANDLE_VALUE ((HANDLE)((LONG_PTR)-1)) -#define FAILED(hr) ((hr) & 0x80000000) -#define SUCCEEDED(hr) (!FAILED(hr)) - - -// -// Functions -// -#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d)) -#define MAKEWORD(a,b) ((a << 8) | (b)) - -#define lstrlen strlen -#define lstrcpy strcpy -#define lstrcmpi strcasecmp -#define _stricmp strcasecmp -#define InterlockedIncrement(x) __sync_fetch_and_add((x), 1) -#define InterlockedDecrement(x) __sync_fetch_and_sub((x), 1) -// Windows uses (new, old) ordering but GCC has (old, new) -#define InterlockedCompareExchange(x,y,z) __sync_val_compare_and_swap(x,z,y) - -#define UInt32x32To64(a, b) ( (uint64_t) ( ((uint64_t)((uint32_t)(a))) * ((uint32_t)(b)) ) ) -#define Int64ShrlMod32(a, b) ( (uint64_t) ( (uint64_t)(a) >> (b) ) ) -#define Int32x32To64(a, b) ((__int64)(((__int64)((long)(a))) * ((long)(b)))) - -#define MulDiv(nNumber, nNumerator, nDenominator) (int32_t) (((int64_t) (nNumber) * (int64_t) (nNumerator) + (int64_t) ((nDenominator)/2)) / (int64_t) (nDenominator)) - -#ifdef __cplusplus -}; // namespace avxsynth -#endif // __cplusplus - -#endif // __WINDOWS2LINUX_H__ diff --git a/compat/float/float.h b/compat/float/float.h index 1f0d3ab4b5..c69f728e32 100644 --- a/compat/float/float.h +++ b/compat/float/float.h @@ -1,20 +1,20 @@ /* * Work around broken floating point limits on some systems. * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ diff --git a/compat/float/limits.h b/compat/float/limits.h index 7ea374a8bc..9150bc8134 100644 --- a/compat/float/limits.h +++ b/compat/float/limits.h @@ -1,20 +1,20 @@ /* * Work around broken floating point limits on some systems. * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ diff --git a/compat/getopt.c b/compat/getopt.c index 9a9c5421de..7c646f6a05 100644 --- a/compat/getopt.c +++ b/compat/getopt.c @@ -1,18 +1,18 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -44,7 +44,7 @@ static int getopt(int argc, char *argv[], char *opts) int c; char *cp; - if (sp == 1) { + if (sp == 1) if (optind >= argc || argv[optind][0] != '-' || argv[optind][1] == '\0') return EOF; @@ -52,7 +52,6 @@ static int getopt(int argc, char *argv[], char *opts) optind++; return EOF; } - } optopt = c = argv[optind][sp]; if (c == ':' || (cp = strchr(opts, c)) == NULL) { fprintf(stderr, ": illegal option -- %c\n", c); diff --git a/compat/msvcrt/snprintf.c b/compat/msvcrt/snprintf.c index c64653fe82..0af7b54353 100644 --- a/compat/msvcrt/snprintf.c +++ b/compat/msvcrt/snprintf.c @@ -2,20 +2,20 @@ * C99-compatible snprintf() and vsnprintf() implementations * Copyright (c) 2012 Ronald S. Bultje * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -24,11 +24,10 @@ #include #include -#include "compat/va_copy.h" #include "libavutil/error.h" -#if defined(__MINGW32__) -#define EOVERFLOW EFBIG +#if !defined(va_copy) && defined(_MSC_VER) +#define va_copy(dst, src) ((dst) = (src)) #endif int avpriv_snprintf(char *s, size_t n, const char *fmt, ...) diff --git a/compat/msvcrt/snprintf.h b/compat/msvcrt/snprintf.h deleted file mode 100644 index f02113c5a2..0000000000 --- a/compat/msvcrt/snprintf.h +++ /dev/null @@ -1,38 +0,0 @@ -/* - * C99-compatible snprintf() and vsnprintf() implementations - * Copyright (c) 2012 Ronald S. Bultje - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#ifndef COMPAT_SNPRINTF_H -#define COMPAT_SNPRINTF_H - -#include -#include - -int avpriv_snprintf(char *s, size_t n, const char *fmt, ...); -int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap); - -#undef snprintf -#undef _snprintf -#undef vsnprintf -#define snprintf avpriv_snprintf -#define _snprintf avpriv_snprintf -#define vsnprintf avpriv_vsnprintf - -#endif /* COMPAT_SNPRINTF_H */ diff --git a/compat/os2threads.h b/compat/os2threads.h deleted file mode 100644 index 441ac43710..0000000000 --- a/compat/os2threads.h +++ /dev/null @@ -1,164 +0,0 @@ -/* - * Copyright (c) 2011 KO Myung-Hun - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/** - * @file - * os2threads to pthreads wrapper - */ - -#ifndef AVCODEC_OS2PTHREADS_H -#define AVCODEC_OS2PTHREADS_H - -#define INCL_DOS -#include - -#undef __STRICT_ANSI__ /* for _beginthread() */ -#include - -#include "libavutil/mem.h" - -typedef TID pthread_t; -typedef void pthread_attr_t; - -typedef HMTX pthread_mutex_t; -typedef void pthread_mutexattr_t; - -typedef struct { - HEV event_sem; - int wait_count; -} pthread_cond_t; - -typedef void pthread_condattr_t; - -struct thread_arg { - void *(*start_routine)(void *); - void *arg; -}; - -static void thread_entry(void *arg) -{ - struct thread_arg *thread_arg = arg; - - thread_arg->start_routine(thread_arg->arg); - - av_free(thread_arg); -} - -static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr_t *attr, void *(*start_routine)(void*), void *arg) -{ - struct thread_arg *thread_arg; - - thread_arg = av_mallocz(sizeof(struct thread_arg)); - - thread_arg->start_routine = start_routine; - thread_arg->arg = arg; - - *thread = _beginthread(thread_entry, NULL, 256 * 1024, thread_arg); - - return 0; -} - -static av_always_inline int pthread_join(pthread_t thread, void **value_ptr) -{ - DosWaitThread((PTID)&thread, DCWW_WAIT); - - return 0; -} - -static av_always_inline int pthread_mutex_init(pthread_mutex_t *mutex, const pthread_mutexattr_t *attr) -{ - DosCreateMutexSem(NULL, (PHMTX)mutex, 0, FALSE); - - return 0; -} - -static av_always_inline int pthread_mutex_destroy(pthread_mutex_t *mutex) -{ - DosCloseMutexSem(*(PHMTX)mutex); - - return 0; -} - -static av_always_inline int pthread_mutex_lock(pthread_mutex_t *mutex) -{ - DosRequestMutexSem(*(PHMTX)mutex, SEM_INDEFINITE_WAIT); - - return 0; -} - -static av_always_inline int pthread_mutex_unlock(pthread_mutex_t *mutex) -{ - DosReleaseMutexSem(*(PHMTX)mutex); - - return 0; -} - -static av_always_inline int pthread_cond_init(pthread_cond_t *cond, const pthread_condattr_t *attr) -{ - DosCreateEventSem(NULL, &cond->event_sem, DCE_POSTONE, FALSE); - - cond->wait_count = 0; - - return 0; -} - -static av_always_inline int pthread_cond_destroy(pthread_cond_t *cond) -{ - DosCloseEventSem(cond->event_sem); - - return 0; -} - -static av_always_inline int pthread_cond_signal(pthread_cond_t *cond) -{ - if (cond->wait_count > 0) { - DosPostEventSem(cond->event_sem); - - cond->wait_count--; - } - - return 0; -} - -static av_always_inline int pthread_cond_broadcast(pthread_cond_t *cond) -{ - while (cond->wait_count > 0) { - DosPostEventSem(cond->event_sem); - - cond->wait_count--; - } - - return 0; -} - -static av_always_inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex) -{ - cond->wait_count++; - - pthread_mutex_unlock(mutex); - - DosWaitEventSem(cond->event_sem, SEM_INDEFINITE_WAIT); - - pthread_mutex_lock(mutex); - - return 0; -} - -#endif /* AVCODEC_OS2PTHREADS_H */ diff --git a/compat/plan9/main.c b/compat/plan9/main.c index d46f96d170..97d70678a6 100644 --- a/compat/plan9/main.c +++ b/compat/plan9/main.c @@ -1,18 +1,18 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ diff --git a/compat/strtod.c b/compat/strtod.c index 3a9452eac2..258909f409 100644 --- a/compat/strtod.c +++ b/compat/strtod.c @@ -2,20 +2,20 @@ * C99-compatible strtod() implementation * Copyright (c) 2012 Ronald S. Bultje * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ diff --git a/compat/tms470/math.h b/compat/tms470/math.h index 6234cc597b..b686d4d8a9 100644 --- a/compat/tms470/math.h +++ b/compat/tms470/math.h @@ -1,23 +1,23 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef FFMPEG_COMPAT_TMS470_MATH_H -#define FFMPEG_COMPAT_TMS470_MATH_H +#ifndef LIBAV_COMPAT_TMS470_MATH_H +#define LIBAV_COMPAT_TMS470_MATH_H #include_next @@ -27,4 +27,4 @@ #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 })) #define NAN (*(const float*)((const unsigned []){ 0x7fc00000 })) -#endif /* FFMPEG_COMPAT_TMS470_MATH_H */ +#endif /* LIBAV_COMPAT_TMS470_MATH_H */ diff --git a/compat/va_copy.h b/compat/va_copy.h deleted file mode 100644 index 3cb5ebee33..0000000000 --- a/compat/va_copy.h +++ /dev/null @@ -1,29 +0,0 @@ -/* - * MSVC Compatible va_copy macro - * Copyright (c) 2012 Derek Buitenhuis - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include - -#if !defined(va_copy) && defined(_MSC_VER) -#define va_copy(dst, src) ((dst) = (src)) -#endif -#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3 -#define va_copy(dst, src) __va_copy(dst, src) -#endif diff --git a/compat/w32pthreads.h b/compat/w32pthreads.h index cfb8f6471f..be275edce7 100644 --- a/compat/w32pthreads.h +++ b/compat/w32pthreads.h @@ -4,20 +4,20 @@ * Authors: Steven Walters * Pegasys Inc. * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -26,8 +26,8 @@ * w32threads to pthreads wrapper */ -#ifndef FFMPEG_COMPAT_W32PTHREADS_H -#define FFMPEG_COMPAT_W32PTHREADS_H +#ifndef LIBAV_COMPAT_W32PTHREADS_H +#define LIBAV_COMPAT_W32PTHREADS_H /* Build up a pthread-like API using underlying Windows API. Have only static * methods so as to not conflict with a potentially linked in pthread-win32 @@ -39,7 +39,6 @@ #include #include -#include "libavutil/common.h" #include "libavutil/internal.h" #include "libavutil/mem.h" @@ -134,29 +133,28 @@ typedef struct win32_cond_t { volatile int is_broadcast; } win32_cond_t; -static int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr) +static void pthread_cond_init(pthread_cond_t *cond, const void *unused_attr) { win32_cond_t *win32_cond = NULL; if (cond_init) { cond_init(cond); - return 0; + return; } /* non native condition variables */ win32_cond = av_mallocz(sizeof(win32_cond_t)); if (!win32_cond) - return ENOMEM; + return; cond->ptr = win32_cond; win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL); if (!win32_cond->semaphore) - return ENOMEM; + return; win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL); if (!win32_cond->waiters_done) - return ENOMEM; + return; pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL); pthread_mutex_init(&win32_cond->mtx_broadcast, NULL); - return 0; } static void pthread_cond_destroy(pthread_cond_t *cond) @@ -280,4 +278,4 @@ static void w32thread_init(void) } -#endif /* FFMPEG_COMPAT_W32PTHREADS_H */ +#endif /* LIBAV_COMPAT_W32PTHREADS_H */ diff --git a/configure b/configure index faaee7f715..ccf437f73a 100755 --- a/configure +++ b/configure @@ -1,6 +1,6 @@ #!/bin/sh # -# FFmpeg configure script +# Libav configure script # # Copyright (c) 2000-2002 Fabrice Bellard # Copyright (c) 2005-2008 Diego Biurrun @@ -44,9 +44,9 @@ if test "$E1" != 0 || test "$E2" = 0; then echo "No compatible shell script interpreter found." echo "This configure script requires a POSIX-compatible shell" echo "such as bash or ksh." - echo "THIS IS NOT A BUG IN FFMPEG, DO NOT REPORT IT AS SUCH." + echo "THIS IS NOT A BUG IN LIBAV, DO NOT REPORT IT AS SUCH." echo "Instead, install a working POSIX-compatible shell." - echo "Disabling this configure test will create a broken FFmpeg." + echo "Disabling this configure test will create a broken Libav." if test "$BASH_VERSION" = '2.04.0(1)-release'; then echo "This bash version ($BASH_VERSION) is broken on your platform." echo "Upgrade to a later version if available." @@ -78,17 +78,14 @@ Help options: Standard options: --logfile=FILE log tests and output to FILE [config.log] --disable-logging do not log configure debug information - --fatal-warnings fail if any configure warning is generated --prefix=PREFIX install in PREFIX [$prefix] --bindir=DIR install binaries in DIR [PREFIX/bin] - --datadir=DIR install data files in DIR [PREFIX/share/ffmpeg] - --docdir=DIR install documentation in DIR [PREFIX/share/doc/ffmpeg] + --datadir=DIR install data files in DIR [PREFIX/share/avconv] + --docdir=DIR install documentation in DIR [PREFIX/share/doc/libav] --libdir=DIR install libs in DIR [PREFIX/lib] --shlibdir=DIR install shared libs in DIR [PREFIX/lib] --incdir=DIR install includes in DIR [PREFIX/include] --mandir=DIR install man page in DIR [PREFIX/share/man] - --enable-rpath use rpath to allow installing libraries in paths - not part of the dynamic linker search path Licensing options: --enable-gpl allow use of GPL code, the resulting libs @@ -101,43 +98,31 @@ Configuration options: --disable-static do not build static libraries [no] --enable-shared build shared libraries [no] --enable-small optimize for size instead of speed - --disable-runtime-cpudetect disable detecting cpu capabilities at runtime (smaller binary) + --enable-runtime-cpudetect detect cpu capabilities at runtime (bigger binary) --enable-gray enable full grayscale support (slower color) --disable-swscale-alpha disable alpha channel support in swscale --disable-all disable building components, libraries and programs - --enable-incompatible-libav-abi enable incompatible Libav fork ABI [no] - --enable-raise-major increase major version numbers in sonames [no] Program options: --disable-programs do not build command line programs - --disable-ffmpeg disable ffmpeg build - --disable-ffplay disable ffplay build - --disable-ffprobe disable ffprobe build - --disable-ffserver disable ffserver build - -Documentation options: - --disable-doc do not build documentation - --disable-htmlpages do not build HTML documentation pages - --disable-manpages do not build man documentation pages - --disable-podpages do not build POD documentation pages - --disable-txtpages do not build text documentation pages + --disable-avconv disable avconv build + --disable-avplay disable avplay build + --disable-avprobe disable avprobe build + --disable-avserver disable avserver build Component options: + --disable-doc do not build documentation --disable-avdevice disable libavdevice build --disable-avcodec disable libavcodec build --disable-avformat disable libavformat build --disable-avutil disable libavutil build - --disable-swresample disable libswresample build --disable-swscale disable libswscale build - --disable-postproc disable libpostproc build - --disable-avfilter disable libavfilter build - --enable-avresample enable libavresample build [no] + --disable-avfilter disable video filter support [no] + --disable-avresample disable libavresample build [no] --disable-pthreads disable pthreads [auto] --disable-w32threads disable Win32 threads [auto] - --disable-os2threads disable OS/2 threads [auto] --disable-network disable network support [no] --disable-dct disable DCT code - --disable-dwt disable DWT code --disable-error-resilience disable error resilience code --disable-lsp disable LSP code --disable-lzo disable LZO decoder code @@ -146,10 +131,10 @@ Component options: --disable-fft disable FFT code Hardware accelerators: - --disable-dxva2 disable DXVA2 code [autodetect] - --disable-vaapi disable VAAPI code [autodetect] - --disable-vda disable VDA code [autodetect] - --disable-vdpau disable VDPAU code [autodetect] + --enable-dxva2 enable DXVA2 code + --enable-vaapi enable VAAPI code + --enable-vda enable VDA code + --enable-vdpau enable VDPAU code Individual component options: --disable-everything disable all components listed below @@ -190,56 +175,33 @@ Individual component options: External library support: --enable-avisynth enable reading of AviSynth script files [no] - --disable-bzlib disable bzlib [autodetect] - --enable-fontconfig enable fontconfig + --enable-bzlib enable bzlib [autodetect] --enable-frei0r enable frei0r video filtering --enable-gnutls enable gnutls [no] - --disable-iconv disable iconv [autodetect] - --enable-ladspa enable LADSPA audio filtering - --enable-libaacplus enable AAC+ encoding via libaacplus [no] - --enable-libass enable libass subtitles rendering [no] - --enable-libbluray enable BluRay reading using libbluray [no] --enable-libbs2b enable bs2b DSP library [no] - --enable-libcaca enable textual display using libcaca - --enable-libcelt enable CELT decoding via libcelt [no] --enable-libcdio enable audio CD grabbing with libcdio --enable-libdc1394 enable IIDC-1394 grabbing using libdc1394 and libraw1394 [no] --enable-libfaac enable AAC encoding via libfaac [no] --enable-libfdk-aac enable AAC de/encoding via libfdk-aac [no] - --enable-libflite enable flite (voice synthesis) support via libflite [no] --enable-libfreetype enable libfreetype [no] - --enable-libgme enable Game Music Emu via libgme [no] --enable-libgsm enable GSM de/encoding via libgsm [no] - --enable-libiec61883 enable iec61883 via libiec61883 [no] --enable-libilbc enable iLBC de/encoding via libilbc [no] - --enable-libmodplug enable ModPlug via libmodplug [no] --enable-libmp3lame enable MP3 encoding via libmp3lame [no] - --enable-libnut enable NUT (de)muxing via libnut, - native (de)muxer exists [no] --enable-libopencore-amrnb enable AMR-NB de/encoding via libopencore-amrnb [no] --enable-libopencore-amrwb enable AMR-WB decoding via libopencore-amrwb [no] --enable-libopencv enable video filtering via libopencv [no] --enable-libopenjpeg enable JPEG 2000 de/encoding via OpenJPEG [no] --enable-libopus enable Opus de/encoding via libopus [no] --enable-libpulse enable Pulseaudio input via libpulse [no] - --enable-libquvi enable quvi input via libquvi [no] --enable-librtmp enable RTMP[E] support via librtmp [no] --enable-libschroedinger enable Dirac de/encoding via libschroedinger [no] - --enable-libshine enable fixed-point MP3 encoding via libshine [no] - --enable-libsoxr enable Include libsoxr resampling [no] --enable-libspeex enable Speex de/encoding via libspeex [no] - --enable-libssh enable SFTP protocol via libssh [no] - --enable-libstagefright-h264 enable H.264 decoding via libstagefright [no] --enable-libtheora enable Theora encoding via libtheora [no] --enable-libtwolame enable MP2 encoding via libtwolame [no] - --enable-libutvideo enable Ut Video encoding and decoding via libutvideo [no] - --enable-libv4l2 enable libv4l2/v4l-utils [no] - --enable-libvidstab enable video stabilization using vid.stab [no] --enable-libvo-aacenc enable AAC encoding via libvo-aacenc [no] --enable-libvo-amrwbenc enable AMR-WB encoding via libvo-amrwbenc [no] - --enable-libvorbis enable Vorbis en/decoding via libvorbis, - native implementation exists [no] + --enable-libvorbis enable Vorbis encoding via libvorbis [no] --enable-libvpx enable VP8 and VP9 de/encoding via libvpx [no] --enable-libwavpack enable wavpack encoding via libwavpack [no] --enable-libwebp enable WebP encoding via libwebp [no] @@ -248,23 +210,15 @@ External library support: --enable-libxavs enable AVS encoding via xavs [no] --enable-libxvid enable Xvid encoding via xvidcore, native MPEG-4/Xvid encoder exists [no] - --enable-libzmq enable message passing via libzmq [no] - --enable-libzvbi enable teletext support via libzvbi [no] - --enable-decklink enable Blackmagick DeckLink output [no] - --enable-openal enable OpenAL 1.1 capture support [no] - --enable-opencl enable OpenCL code - --enable-opengl enable OpenGL rendering [no] --enable-openssl enable openssl [no] --enable-x11grab enable X11 grabbing [no] - --disable-xlib disable xlib [autodetect] - --disable-zlib disable zlib [autodetect] + --enable-zlib enable zlib [autodetect] Toolchain options: --arch=ARCH select architecture [$arch] --cpu=CPU select the minimum required CPU (affects instruction selection, may crash on older CPUs) --cross-prefix=PREFIX use PREFIX for compilation tools [$cross_prefix] - --progs-suffix=SUFFIX program name suffix [] --enable-cross-compile assume a cross-compiler is used --sysroot=PATH root of cross-build tree --sysinclude=PATH location of cross-build system headers @@ -272,21 +226,16 @@ Toolchain options: --target-exec=CMD command to run executables on target --target-path=DIR path to view of build directory on target --target-samples=DIR path to samples directory on target - --tempprefix=PATH force fixed dir/prefix instead of mktemp for checks --toolchain=NAME set tool defaults according to NAME --pkg-config=PKGCONFIG use pkg-config tool PKGCONFIG [$pkg_config_default] - --nm=NM use nm tool NM [$nm_default] + --nm=NM use nm tool --ar=AR use archive tool AR [$ar_default] --as=AS use assembler AS [$as_default] - --windres=WINDRES use windows resource compiler WINDRES [$windres_default] - --yasmexe=EXE use yasm-compatible assembler EXE [$yasmexe_default] --cc=CC use C compiler CC [$cc_default] - --cxx=CXX use C compiler CXX [$cxx_default] --dep-cc=DEPCC use dependency generator DEPCC [$cc_default] - --ld=LD use linker LD [$ld_default] + --ld=LD use linker LD --pkg-config=PKGCONF use pkg-config PKGCONF [$pkg_config_default] --pkg-config-flags=FLAGS pass additional flags to pkgconf [] - --ranlib=RANLIB use ranlib RANLIB [$ranlib_default] --host-cc=HOSTCC use host C compiler HOSTCC --host-cflags=HCFLAGS use HCFLAGS when compiling for host --host-cppflags=HCPPFLAGS use HCPPFLAGS when compiling for host @@ -295,7 +244,6 @@ Toolchain options: --host-libs=HLIBS use libs HLIBS when linking for host --host-os=OS compiler host OS [$target_os] --extra-cflags=ECFLAGS add ECFLAGS to CFLAGS [$CFLAGS] - --extra-cxxflags=ECFLAGS add ECFLAGS to CXXFLAGS [$CXXFLAGS] --extra-ldflags=ELDFLAGS add ELDFLAGS to LDFLAGS [$LDFLAGS] --extra-libs=ELIBS add ELIBS [$ELIBS] --extra-version=STRING version string suffix [] @@ -340,27 +288,14 @@ Optimization options (experts only): --disable-neon disable NEON optimizations --disable-inline-asm disable use of inline assembler --disable-yasm disable use of yasm assembler - --disable-mips32r2 disable MIPS32R2 optimizations - --disable-mipsdspr1 disable MIPS DSP ASE R1 optimizations - --disable-mipsdspr2 disable MIPS DSP ASE R2 optimizations - --disable-mipsfpu disable floating point MIPS optimizations - --disable-fast-unaligned consider unaligned accesses slow -Developer options (useful when working on FFmpeg itself): +Developer options (useful when working on Libav itself): --disable-debug disable debugging symbols --enable-debug=LEVEL set the debug level [$debuglevel] --disable-optimizations disable compiler optimizations --enable-extra-warnings enable more compiler warnings - --disable-stripping disable stripping of executables and shared libraries - --assert-level=level 0(default), 1 or 2, amount of assertion testing, - 2 causes a slowdown at runtime. - --enable-memory-poisoning fill heap uninitialized allocated space with arbitrary data - --valgrind=VALGRIND run "make fate" tests through valgrind to detect memory - leaks and errors, using the specified valgrind binary. - Cannot be combined with --target-exec - --enable-ftrapv Trap arithmetic overflows --samples=PATH location of test samples for FATE, if not set use - \$FATE_SAMPLES at make invocation time. + \$LIBAV_SAMPLES at make invocation time. --enable-neon-clobber-test check NEON registers for clobbering (should be used only for debugging purposes) --enable-xmm-clobber-test check XMM registers for clobbering (Win64-only; @@ -407,7 +342,7 @@ die(){ If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the -ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. +libav-tools@libav.org mailing list or IRC #libav on irc.freenode.net. EOF if disabled logging; then cat <>file; - } else if (file ~ /\\.texi\$/) { - pre = -v ? \"\" : \"@c \"; - yesno = \$2; - c2 = tolower(c); - gsub(/_/, \"-\", c2); - printf(\"%s@set %s %s\\n\", pre, c2, yesno) >>file; } } }" @@ -743,10 +672,6 @@ add_cflags(){ append CFLAGS $($cflags_filter "$@") } -add_cxxflags(){ - append CXXFLAGS $($cflags_filter "$@") -} - add_asflags(){ append ASFLAGS $($asflags_filter "$@") } @@ -756,7 +681,7 @@ add_ldflags(){ } add_stripflags(){ - append ASMSTRIPFLAGS "$@" + append STRIPFLAGS "$@" } add_extralibs(){ @@ -786,11 +711,6 @@ check_cmd(){ "$@" >> $logfile 2>&1 } -check_stat(){ - log check_stat "$@" - stat "$1" >> $logfile 2>&1 -} - cc_o(){ eval printf '%s\\n' $CC_O } @@ -806,20 +726,6 @@ check_cc(){ check_cmd $cc $CPPFLAGS $CFLAGS "$@" $CC_C $(cc_o $TMPO) $TMPC } -check_cxx(){ - log check_cxx "$@" - cat > $TMPCPP - log_file $TMPCPP - check_cmd $cxx $CPPFLAGS $CFLAGS $CXXFLAGS "$@" $CXX_C -o $TMPO $TMPCPP -} - -check_oc(){ - log check_oc "$@" - cat > $TMPM - log_file $TMPM - check_cmd $cc -Werror=missing-prototypes $CPPFLAGS $CFLAGS "$@" $CC_C $(cc_o $TMPO) $TMPM -} - check_cpp(){ log check_cpp "$@" cat > $TMPC @@ -860,7 +766,7 @@ check_yasm(){ echo "$1" > $TMPS log_file $TMPS shift 1 - check_cmd $yasmexe $YASMFLAGS -Werror "$@" -o $TMPO $TMPS + check_cmd $yasmexe $YASMFLAGS "$@" -o $TMPO $TMPS } ld_o(){ @@ -869,11 +775,9 @@ ld_o(){ check_ld(){ log check_ld "$@" - type=$1 - shift 1 - flags=$(filter_out '-l*|*.so' $@) - libs=$(filter '-l*|*.so' $@) - check_$type $($cflags_filter $flags) || return + flags=$(filter_out '-l*' "$@") + libs=$(filter '-l*' "$@") + check_cc $($cflags_filter $flags) || return flags=$($ldflags_filter $flags) libs=$($ldflags_filter $libs) check_cmd $ld $LDFLAGS $flags $(ld_o $TMPE) $TMPO $libs $extralibs @@ -920,17 +824,9 @@ check_cflags(){ test_cflags "$@" && add_cflags "$@" } -check_cxxflags(){ - log check_cxxflags "$@" - set -- $($cflags_filter "$@") - check_cxx "$@" <" - echo "int main(void) { return 0; }" - } | check_oc && check_stat "$TMPO" && enable_safe $headers -} - check_func(){ log check_func "$@" func=$1 shift disable $func - check_ld "cc" "$@" < float foo(float f, float g) { return $func($args); } -int main(void){ return (int) foo; } +int main(void){ return 0; } EOF } @@ -1015,26 +899,7 @@ check_func_headers(){ echo "long check_$func(void) { return (long) $func; }" done echo "int main(void) { return 0; }" - } | check_ld "cc" "$@" && enable $funcs && enable_safe $headers -} - -check_class_headers_cpp(){ - log check_class_headers_cpp "$@" - headers=$1 - classes=$2 - shift 2 - { - for hdr in $headers; do - echo "#include <$hdr>" - done - echo "int main(void) { " - i=1 - for class in $classes; do - echo "$class obj$i;" - i=$(expr $i + 1) - done - echo "return 0; }" - } | check_ld "cxx" "$@" && enable $funcs && enable_safe $headers + } | check_ld "$@" && enable $funcs && enable_safe $headers } check_cpp_condition(){ @@ -1066,22 +931,13 @@ check_lib2(){ check_func_headers "$headers" "$funcs" "$@" && add_extralibs "$@" } -check_lib_cpp(){ - log check_lib_cpp "$@" - headers="$1" - classes="$2" - shift 2 - check_class_headers_cpp "$headers" "$classes" "$@" && add_extralibs "$@" -} - check_pkg_config(){ log check_pkg_config "$@" - pkgandversion="$1" - pkg="${1%% *}" + pkg="$1" headers="$2" funcs="$3" shift 3 - check_cmd $pkg_config --exists --print-errors $pkgandversion || return + check_cmd $pkg_config --exists --print-errors $pkg || return pkg_cflags=$($pkg_config --cflags $pkg_config_flags $pkg) pkg_libs=$($pkg_config --libs $pkg_config_flags $pkg) check_func_headers "$headers" "$funcs" $pkg_cflags $pkg_libs "$@" && @@ -1090,7 +946,7 @@ check_pkg_config(){ } check_exec(){ - check_ld "cc" "$@" && { enabled cross_compile || $TMPE >> $logfile 2>&1; } + check_ld "$@" && { enabled cross_compile || $TMPE >> $logfile 2>&1; } } check_exec_crash(){ @@ -1150,7 +1006,7 @@ check_builtin(){ builtin=$3 shift 3 disable "$name" - check_code ld "$headers" "$builtin" "cc" "$@" && enable "$name" + check_code ld "$headers" "$builtin" "$@" && enable "$name" } check_compile_assert(){ @@ -1179,14 +1035,6 @@ require2(){ check_lib2 "$headers" $func "$@" || die "ERROR: $name not found" } -require_cpp(){ - name="$1" - headers="$2" - classes="$3" - shift 3 - check_lib_cpp "$headers" "$classes" "$@" || die "ERROR: $name not found" -} - require_pkg_config(){ pkg="$1" check_pkg_config "$@" || die "ERROR: $pkg not found" @@ -1194,26 +1042,6 @@ require_pkg_config(){ add_extralibs $(get_safe ${pkg}_libs) } -require_libfreetype(){ - log require_libfreetype "$@" - pkg="freetype2" - check_cmd $pkg_config --exists --print-errors $pkg \ - || die "ERROR: $pkg not found" - pkg_cflags=$($pkg_config --cflags $pkg_config_flags $pkg) - pkg_libs=$($pkg_config --libs $pkg_config_flags $pkg) - { - echo "#include " - echo "#include FT_FREETYPE_H" - echo "long check_func(void) { return (long) FT_Init_FreeType; }" - echo "int main(void) { return 0; }" - } | check_ld "cc" $pkg_cflags $pkg_libs \ - && set_safe ${pkg}_cflags $pkg_cflags \ - && set_safe ${pkg}_libs $pkg_libs \ - || die "ERROR: $pkg not found" - add_cflags $(get_safe ${pkg}_cflags) - add_extralibs $(get_safe ${pkg}_libs) -} - hostcc_e(){ eval printf '%s\\n' $HOSTCC_E } @@ -1294,69 +1122,39 @@ COMPONENT_LIST=" " EXAMPLE_LIST=" - avio_reading_example - decoding_encoding_example - demuxing_decoding_example + avcodec_example filter_audio_example - filtering_audio_example - filtering_video_example metadata_example - muxing_example - remuxing_example - resampling_audio_example - scaling_video_example + output_example transcode_aac_example - transcoding_example " EXTERNAL_LIBRARY_LIST=" avisynth bzlib - crystalhd - decklink frei0r gnutls - iconv - ladspa - libaacplus - libass - libbluray libbs2b - libcaca libcdio - libcelt libdc1394 libfaac libfdk_aac - libflite libfontconfig libfreetype - libgme libgsm - libiec61883 libilbc - libmodplug libmp3lame - libnut libopencore_amrnb libopencore_amrwb libopencv libopenjpeg libopus libpulse - libquvi librtmp libschroedinger - libshine - libsoxr libspeex - libssh - libstagefright_h264 libtheora libtwolame - libutvideo - libv4l2 - libvidstab libvo_aacenc libvo_amrwbenc libvorbis @@ -1367,27 +1165,12 @@ EXTERNAL_LIBRARY_LIST=" libx265 libxavs libxvid - libzmq - libzvbi - openal - opencl - opengl openssl x11grab - xlib zlib " -DOCUMENT_LIST=" - doc - htmlpages - manpages - podpages - txtpages -" - FEATURE_LIST=" - ftrapv gray hardcoded_tables runtime_cpudetect @@ -1404,7 +1187,6 @@ HWACCEL_LIST=" vaapi vda vdpau - xvmc " LIBRARY_LIST=" @@ -1414,8 +1196,6 @@ LIBRARY_LIST=" avformat avresample avutil - postproc - swresample swscale " @@ -1426,17 +1206,16 @@ LICENSE_LIST=" " PROGRAM_LIST=" - ffplay - ffprobe - ffserver - ffmpeg + avconv + avplay + avprobe + avserver " SUBSYSTEM_LIST=" dct - dwt + doc error_resilience - fast_unaligned fft lsp lzo @@ -1447,7 +1226,6 @@ SUBSYSTEM_LIST=" CONFIG_LIST=" $COMPONENT_LIST - $DOCUMENT_LIST $EXAMPLE_LIST $EXTERNAL_LIBRARY_LIST $FEATURE_LIST @@ -1456,21 +1234,17 @@ CONFIG_LIST=" $LIBRARY_LIST $PROGRAM_LIST $SUBSYSTEM_LIST - fontconfig - incompatible_libav_abi memalign_hack - memory_poisoning neon_clobber_test pic pod2man - raise_major + texi2html thumb xmm_clobber_test " THREADS_LIST=" pthreads - os2threads w32threads " @@ -1517,13 +1291,6 @@ ARCH_EXT_LIST_ARM=" vfpv3 " -ARCH_EXT_LIST_MIPS=" - mipsfpu - mips32r2 - mipsdspr1 - mipsdspr2 -" - ARCH_EXT_LIST_X86_SIMD=" amd3dnow amd3dnowext @@ -1559,8 +1326,8 @@ ARCH_EXT_LIST=" $ARCH_EXT_LIST_ARM $ARCH_EXT_LIST_PPC $ARCH_EXT_LIST_X86 - $ARCH_EXT_LIST_MIPS loongson + vis " ARCH_FEATURES=" @@ -1570,7 +1337,6 @@ ARCH_FEATURES=" fast_cmov local_aligned_8 local_aligned_16 - local_aligned_32 " BUILTIN_LIST=" @@ -1579,7 +1345,6 @@ BUILTIN_LIST=" MemoryBarrier mm_empty rdtsc - sarestart sync_val_compare_and_swap " HAVE_LIST_CMDLINE=" @@ -1591,17 +1356,14 @@ HAVE_LIST_CMDLINE=" HAVE_LIST_PUB=" bigendian fast_unaligned - incompatible_libav_abi " HEADERS_LIST=" alsa_asoundlib_h altivec_h arpa_inet_h - asm_types_h cdio_paranoia_h cdio_paranoia_paranoia_h - CL_cl_h dev_bktr_ioctl_bt848_h dev_bktr_ioctl_meteor_h dev_ic_bt8xx_h @@ -1610,15 +1372,12 @@ HEADERS_LIST=" direct_h dlfcn_h dxva_h - ES2_gl_h gsm_h io_h mach_mach_time_h machine_ioctl_bt848_h machine_ioctl_meteor_h malloc_h - openjpeg_1_5_openjpeg_h - OpenGL_gl3_h poll_h sndio_h soundcard_h @@ -1630,7 +1389,6 @@ HEADERS_LIST=" sys_time_h sys_un_h sys_videoio_h - termios_h unistd_h windows_h winsock2_h @@ -1639,7 +1397,6 @@ HEADERS_LIST=" MATH_FUNCS=" atanf atan2f - cbrt cbrtf cosf exp2 @@ -1665,9 +1422,7 @@ MATH_FUNCS=" " SYSTEM_FUNCS=" - access aligned_malloc - clock_gettime closesocket CommandLineToArgvW CoTaskMemFree @@ -1686,14 +1441,10 @@ SYSTEM_FUNCS=" getservbyport GetSystemTimeAsFileTime gettimeofday - glob - glXGetProcAddress inet_aton isatty jack_port_get_latency_range - kbhit localtime_r - lzo1x_999_compress mach_absolute_time MapViewOfFile memalign @@ -1701,20 +1452,18 @@ SYSTEM_FUNCS=" mmap mprotect nanosleep - PeekNamedPipe posix_memalign - pthread_cancel sched_getaffinity SetConsoleTextAttribute setmode setrlimit Sleep strerror_r + strptime sysconf sysctl usleep VirtualAlloc - wglGetProcAddress " TOOLCHAIN_FEATURES=" @@ -1724,13 +1473,9 @@ TOOLCHAIN_FEATURES=" ebp_available ebx_available gnu_as - gnu_windres ibm_asm inline_asm_labels - inline_asm_nonlocal_labels - inline_asm_direct_symbol_refs pragma_deprecated - rsync_contimeout symver_asm_label symver_gnu_asm vfp_args @@ -1746,11 +1491,9 @@ TYPES_LIST=" struct_ipv6_mreq struct_pollfd struct_rusage_ru_maxrss - struct_sctp_event_subscribe struct_sockaddr_in6 struct_sockaddr_sa_len struct_sockaddr_storage - struct_stat_st_mtim_tv_nsec struct_v4l2_frmivalenum_discrete " @@ -1771,16 +1514,11 @@ HAVE_LIST=" $TYPES_LIST atomics_native dos_paths - dxva2api_cobj dxva2_lib libc_msvcrt libdc1394_1 libdc1394_2 - makeinfo - perl - pod2man sdl - texi2html threads vdpau_x11 xlib @@ -1793,8 +1531,6 @@ CONFIG_EXTRA=" audio_frame_queue cabac dsputil - exif - frame_thread_encoder gcrypt golomb gplv3 @@ -1807,7 +1543,6 @@ CONFIG_EXTRA=" huffman intrax8 lgplv3 - llviddsp lpc mpegaudio mpegaudiodsp @@ -1837,8 +1572,6 @@ CMDLINE_SELECT=" logging lto optimizations - rpath - stripping " PATHS_LIST=" @@ -1857,22 +1590,18 @@ CMDLINE_SET=" ar arch as - assert_level build_suffix cc cpu cross_prefix - cxx dep_cc extra_version - gas host_cc host_cflags host_ld host_ldflags host_libs host_os - install ld logfile malloc_prefix @@ -1880,26 +1609,19 @@ CMDLINE_SET=" optflags pkg_config pkg_config_flags - progs_suffix random_seed - ranlib samples - strip sysinclude sysroot target_exec target_os target_path target_samples - tempprefix toolchain - valgrind - yasmexe " CMDLINE_APPEND=" extra_cflags - extra_cxxflags host_cppflags " @@ -1917,11 +1639,6 @@ vfpv3_deps="vfp" map 'eval ${v}_inline_deps=inline_asm' $ARCH_EXT_LIST_ARM -mipsfpu_deps="mips" -mips32r2_deps="mips" -mipsdspr1_deps="mips" -mipsdspr2_deps="mips" - altivec_deps="ppc" ppc4xx_deps="ppc" @@ -1965,6 +1682,7 @@ fast_unaligned_if_any="aarch64 ppc x86" need_memalign="altivec neon sse" # system capabilities +log2_deps="!libc_msvcrt" symver_if_any="symver_asm_label symver_gnu_asm" @@ -1979,12 +1697,11 @@ threads_if_any="$THREADS_LIST" # subsystems dct_select="rdft" error_resilience_select="dsputil" -frame_thread_encoder_deps="encoders threads" mdct_select="fft" rdft_select="fft" mpegaudio_select="mpegaudiodsp" mpegaudiodsp_select="dct" -mpegvideo_select="dsputil h264chroma hpeldsp videodsp" +mpegvideo_select="dsputil hpeldsp videodsp" mpegvideoenc_select="dsputil mpegvideo" # decoders / encoders @@ -1992,7 +1709,6 @@ aac_decoder_select="mdct sinewin" aac_encoder_select="audio_frame_queue mdct sinewin" aac_latm_decoder_select="aac_decoder aac_latm_parser" ac3_decoder_select="mdct ac3dsp ac3_parser dsputil" -ac3_fixed_decoder_select="mdct ac3dsp ac3_parser dsputil" ac3_encoder_select="mdct ac3dsp dsputil" ac3_fixed_encoder_select="mdct ac3dsp dsputil" aic_decoder_select="dsputil golomb" @@ -2000,8 +1716,7 @@ alac_encoder_select="lpc" als_decoder_select="dsputil" amrnb_decoder_select="lsp" amrwb_decoder_select="lsp" -amv_decoder_select="sp5x_decoder exif" -amv_encoder_select="aandcttables" +amv_decoder_select="sp5x_decoder" ape_decoder_select="dsputil" asv1_decoder_select="dsputil" asv1_encoder_select="dsputil" @@ -2010,7 +1725,6 @@ asv2_encoder_select="dsputil" atrac1_decoder_select="mdct sinewin" atrac3_decoder_select="mdct" atrac3p_decoder_select="mdct sinewin" -avrn_decoder_select="exif" bink_decoder_select="dsputil hpeldsp" binkaudio_dct_decoder_select="mdct rdft dct sinewin" binkaudio_rdft_decoder_select="mdct rdft sinewin" @@ -2021,18 +1735,17 @@ cook_decoder_select="dsputil mdct sinewin" cscd_decoder_select="lzo" cscd_decoder_suggest="zlib" dca_decoder_select="mdct" -dirac_decoder_select="dsputil dwt golomb videodsp" dnxhd_decoder_select="dsputil" dnxhd_encoder_select="aandcttables dsputil mpegvideoenc" dvvideo_decoder_select="dsputil" dvvideo_encoder_select="dsputil" -dxa_decoder_select="zlib" +dxa_decoder_deps="zlib" eac3_decoder_select="ac3_decoder" eac3_encoder_select="ac3_encoder" eamad_decoder_select="aandcttables dsputil mpegvideo" -eatgq_decoder_select="aandcttables" +eatgq_decoder_select="aandcttables dsputil" eatqi_decoder_select="aandcttables dsputil error_resilience mpegvideo" -exr_decoder_select="zlib" +exr_decoder_deps="zlib" ffv1_decoder_select="golomb rangecoder" ffv1_encoder_select="rangecoder" ffvhuff_decoder_select="huffyuv_decoder" @@ -2040,16 +1753,15 @@ ffvhuff_encoder_select="huffyuv_encoder" fic_decoder_select="golomb" flac_decoder_select="golomb" flac_encoder_select="dsputil golomb lpc" -flashsv_decoder_select="zlib" -flashsv_encoder_select="zlib" -flashsv2_encoder_select="zlib" -flashsv2_decoder_select="zlib" +flashsv_decoder_deps="zlib" +flashsv_encoder_deps="zlib" +flashsv2_decoder_deps="zlib" flv_decoder_select="h263_decoder" flv_encoder_select="h263_encoder" fourxm_decoder_select="dsputil" fraps_decoder_select="dsputil huffman" -g2m_decoder_select="dsputil zlib" -g729_decoder_select="dsputil" +g2m_decoder_deps="zlib" +g2m_decoder_select="dsputil" h261_decoder_select="error_resilience mpegvideo" h261_encoder_select="aandcttables mpegvideoenc" h263_decoder_select="error_resilience h263_parser h263dsp mpegvideo" @@ -2059,8 +1771,8 @@ h263p_encoder_select="h263_encoder" h264_decoder_select="cabac golomb h264chroma h264dsp h264pred h264qpel videodsp" h264_decoder_suggest="error_resilience" hevc_decoder_select="cabac dsputil golomb videodsp" -huffyuv_decoder_select="dsputil llviddsp" -huffyuv_encoder_select="dsputil huffman llviddsp" +huffyuv_decoder_select="dsputil" +huffyuv_encoder_select="dsputil huffman" iac_decoder_select="imc_decoder" imc_decoder_select="dsputil fft mdct sinewin" indeo3_decoder_select="hpeldsp" @@ -2074,7 +1786,7 @@ loco_decoder_select="golomb" mdec_decoder_select="dsputil error_resilience mpegvideo" metasound_decoder_select="lsp mdct sinewin" mimic_decoder_select="dsputil hpeldsp" -mjpeg_decoder_select="dsputil hpeldsp exif" +mjpeg_decoder_select="dsputil hpeldsp" mjpeg_encoder_select="aandcttables mpegvideoenc" mjpegb_decoder_select="mjpeg_decoder" mlp_decoder_select="mlp_parser" @@ -2094,9 +1806,9 @@ mpc8_decoder_select="mpegaudiodsp" mpeg_xvmc_decoder_deps="X11_extensions_XvMClib_h" mpeg_xvmc_decoder_select="mpeg2video_decoder" mpeg1video_decoder_select="error_resilience mpegvideo" -mpeg1video_encoder_select="aandcttables mpegvideoenc h263dsp" +mpeg1video_encoder_select="aandcttables mpegvideoenc" mpeg2video_decoder_select="error_resilience mpegvideo" -mpeg2video_encoder_select="aandcttables mpegvideoenc h263dsp" +mpeg2video_encoder_select="aandcttables mpegvideoenc" mpeg4_decoder_select="h263_decoder mpeg4video_parser" mpeg4_encoder_select="h263_encoder" msmpeg4v1_decoder_select="h263_decoder" @@ -2110,17 +1822,16 @@ nellymoser_decoder_select="mdct sinewin" nellymoser_encoder_select="audio_frame_queue mdct sinewin" nuv_decoder_select="dsputil lzo" on2avc_decoder_select="mdct" -opus_decoder_deps="swresample" -png_decoder_select="zlib" -png_encoder_select="dsputil zlib" +opus_decoder_deps="avresample" +png_decoder_deps="zlib" +png_encoder_deps="zlib" +png_encoder_select="dsputil" prores_decoder_select="dsputil" prores_encoder_select="dsputil" qcelp_decoder_select="lsp" qdm2_decoder_select="mdct rdft mpegaudiodsp" ra_144_encoder_select="audio_frame_queue lpc" ralf_decoder_select="golomb" -rawvideo_decoder_select="dsputil" -rtjpeg_decoder_select="dsputil" rv10_decoder_select="error_resilience h263_decoder h263dsp" rv10_encoder_select="h263_encoder" rv20_decoder_select="error_resilience h263_decoder h263dsp" @@ -2129,11 +1840,6 @@ rv30_decoder_select="error_resilience golomb h264chroma h264pred h264qpel mpegvi rv40_decoder_select="error_resilience golomb h264chroma h264pred h264qpel mpegvideo videodsp" shorten_decoder_select="golomb" sipr_decoder_select="lsp" -snow_decoder_select="dsputil dwt h264qpel hpeldsp rangecoder" -snow_encoder_select="aandcttables dsputil dwt h264qpel hpeldsp mpegvideoenc rangecoder" -sonic_decoder_select="golomb rangecoder" -sonic_encoder_select="golomb rangecoder" -sonic_ls_encoder_select="golomb rangecoder" sp5x_decoder_select="mjpeg_decoder" svq1_decoder_select="hpeldsp" svq1_encoder_select="aandcttables dsputil hpeldsp mpegvideoenc" @@ -2144,10 +1850,10 @@ theora_decoder_select="vp3_decoder" thp_decoder_select="mjpeg_decoder" tiff_decoder_suggest="zlib" tiff_encoder_suggest="zlib" -truehd_decoder_select="mlp_parser" +truehd_decoder_select="mlp_decoder" truemotion2_decoder_select="dsputil" truespeech_decoder_select="dsputil" -tscc_decoder_select="zlib" +tscc_decoder_deps="zlib" twinvq_decoder_select="mdct lsp sinewin" utvideo_decoder_select="dsputil" utvideo_encoder_select="dsputil huffman" @@ -2163,7 +1869,7 @@ vp6a_decoder_select="vp6_decoder" vp6f_decoder_select="vp6_decoder" vp7_decoder_select="h264pred videodsp" vp8_decoder_select="h264pred videodsp" -vp9_decoder_select="videodsp vp9_parser" +vp9_decoder_select="videodsp" webp_decoder_select="vp8_decoder" wmapro_decoder_select="mdct sinewin" wmav1_decoder_select="mdct sinewin" @@ -2177,105 +1883,73 @@ wmv2_decoder_select="h263_decoder intrax8 videodsp" wmv2_encoder_select="h263_encoder" wmv3_decoder_select="vc1_decoder" wmv3image_decoder_select="wmv3_decoder" -zerocodec_decoder_select="zlib" -zlib_decoder_select="zlib" -zlib_encoder_select="zlib" -zmbv_decoder_select="zlib" -zmbv_encoder_select="zlib" +zerocodec_decoder_deps="zlib" +zlib_decoder_deps="zlib" +zlib_encoder_deps="zlib" +zmbv_decoder_deps="zlib" +zmbv_encoder_deps="zlib" # hardware accelerators -crystalhd_deps="libcrystalhd_libcrystalhd_if_h" dxva2_deps="dxva2api_h" vaapi_deps="va_va_h" vda_deps="VideoDecodeAcceleration_VDADecoder_h pthreads" vda_extralibs="-framework CoreFoundation -framework VideoDecodeAcceleration -framework QuartzCore" vdpau_deps="vdpau_vdpau_h vdpau_vdpau_x11_h" -xvmc_deps="X11_extensions_XvMClib_h" h263_vaapi_hwaccel_deps="vaapi" h263_vaapi_hwaccel_select="h263_decoder" h263_vdpau_hwaccel_deps="vdpau" h263_vdpau_hwaccel_select="h263_decoder" -h264_crystalhd_decoder_select="crystalhd h264_mp4toannexb_bsf h264_parser" h264_dxva2_hwaccel_deps="dxva2" h264_dxva2_hwaccel_select="h264_decoder" h264_vaapi_hwaccel_deps="vaapi" h264_vaapi_hwaccel_select="h264_decoder" -h264_vda_decoder_deps="vda" -h264_vda_decoder_select="h264_decoder" h264_vda_hwaccel_deps="vda" h264_vda_hwaccel_select="h264_decoder" h264_vda_old_hwaccel_deps="vda" h264_vda_old_hwaccel_select="h264_decoder" -h264_vdpau_decoder_deps="vdpau" -h264_vdpau_decoder_select="h264_decoder" h264_vdpau_hwaccel_deps="vdpau" h264_vdpau_hwaccel_select="h264_decoder" -mpeg_vdpau_decoder_deps="vdpau" -mpeg_vdpau_decoder_select="mpeg2video_decoder" -mpeg_xvmc_hwaccel_deps="xvmc" -mpeg_xvmc_hwaccel_select="mpeg2video_decoder" -mpeg1_vdpau_decoder_deps="vdpau" -mpeg1_vdpau_decoder_select="mpeg1video_decoder" mpeg1_vdpau_hwaccel_deps="vdpau" mpeg1_vdpau_hwaccel_select="mpeg1video_decoder" -mpeg1_xvmc_hwaccel_deps="xvmc" -mpeg1_xvmc_hwaccel_select="mpeg1video_decoder" -mpeg2_crystalhd_decoder_select="crystalhd" mpeg2_dxva2_hwaccel_deps="dxva2" mpeg2_dxva2_hwaccel_select="mpeg2video_decoder" mpeg2_vaapi_hwaccel_deps="vaapi" mpeg2_vaapi_hwaccel_select="mpeg2video_decoder" mpeg2_vdpau_hwaccel_deps="vdpau" mpeg2_vdpau_hwaccel_select="mpeg2video_decoder" -mpeg2_xvmc_hwaccel_deps="xvmc" -mpeg2_xvmc_hwaccel_select="mpeg2video_decoder" -mpeg4_crystalhd_decoder_select="crystalhd" mpeg4_vaapi_hwaccel_deps="vaapi" mpeg4_vaapi_hwaccel_select="mpeg4_decoder" -mpeg4_vdpau_decoder_deps="vdpau" -mpeg4_vdpau_decoder_select="mpeg4_decoder" mpeg4_vdpau_hwaccel_deps="vdpau" mpeg4_vdpau_hwaccel_select="mpeg4_decoder" -msmpeg4_crystalhd_decoder_select="crystalhd" -vc1_crystalhd_decoder_select="crystalhd" vc1_dxva2_hwaccel_deps="dxva2" vc1_dxva2_hwaccel_select="vc1_decoder" vc1_vaapi_hwaccel_deps="vaapi" vc1_vaapi_hwaccel_select="vc1_decoder" -vc1_vdpau_decoder_deps="vdpau" -vc1_vdpau_decoder_select="vc1_decoder" vc1_vdpau_hwaccel_deps="vdpau" vc1_vdpau_hwaccel_select="vc1_decoder" -wmv3_crystalhd_decoder_select="crystalhd" wmv3_dxva2_hwaccel_select="vc1_dxva2_hwaccel" wmv3_vaapi_hwaccel_select="vc1_vaapi_hwaccel" -wmv3_vdpau_decoder_select="vc1_vdpau_decoder" wmv3_vdpau_hwaccel_select="vc1_vdpau_hwaccel" # parsers h264_parser_select="h264_decoder" -hevc_parser_select="hevc_decoder" mpeg4video_parser_select="error_resilience h263dsp mpegvideo" mpegvideo_parser_select="error_resilience mpegvideo" vc1_parser_select="mpegvideo" # external libraries -libaacplus_encoder_deps="libaacplus" -libcelt_decoder_deps="libcelt" libfaac_encoder_deps="libfaac" libfaac_encoder_select="audio_frame_queue" libfdk_aac_decoder_deps="libfdk_aac" libfdk_aac_encoder_deps="libfdk_aac" libfdk_aac_encoder_select="audio_frame_queue" -libgme_demuxer_deps="libgme" libgsm_decoder_deps="libgsm" libgsm_encoder_deps="libgsm" libgsm_ms_decoder_deps="libgsm" libgsm_ms_encoder_deps="libgsm" libilbc_decoder_deps="libilbc" libilbc_encoder_deps="libilbc" -libmodplug_demuxer_deps="libmodplug" libmp3lame_encoder_deps="libmp3lame" libmp3lame_encoder_select="audio_frame_queue" libopencore_amrnb_decoder_deps="libopencore_amrnb" @@ -2287,21 +1961,16 @@ libopenjpeg_encoder_deps="libopenjpeg" libopus_decoder_deps="libopus" libopus_encoder_deps="libopus" libopus_encoder_select="audio_frame_queue" -libquvi_demuxer_deps="libquvi" libschroedinger_decoder_deps="libschroedinger" libschroedinger_encoder_deps="libschroedinger" -libshine_encoder_deps="libshine" -libshine_encoder_select="audio_frame_queue" libspeex_decoder_deps="libspeex" libspeex_encoder_deps="libspeex" libspeex_encoder_select="audio_frame_queue" -libstagefright_h264_decoder_deps="libstagefright_h264" libtheora_encoder_deps="libtheora" libtwolame_encoder_deps="libtwolame" libvo_aacenc_encoder_deps="libvo_aacenc" libvo_aacenc_encoder_select="audio_frame_queue" libvo_amrwbenc_encoder_deps="libvo_amrwbenc" -libvorbis_decoder_deps="libvorbis" libvorbis_encoder_deps="libvorbis" libvorbis_encoder_select="audio_frame_queue" libvpx_vp8_decoder_deps="libvpx" @@ -2311,39 +1980,29 @@ libvpx_vp9_encoder_deps="libvpx" libwavpack_encoder_deps="libwavpack" libwebp_encoder_deps="libwebp" libx264_encoder_deps="libx264" -libx264rgb_encoder_deps="libx264" libx265_encoder_deps="libx265" libxavs_encoder_deps="libxavs" libxvid_encoder_deps="libxvid" -libutvideo_decoder_deps="libutvideo" -libutvideo_encoder_deps="libutvideo" -libzvbi_teletext_decoder_deps="libzvbi" # demuxers / muxers ac3_demuxer_select="ac3_parser" asf_demuxer_select="riffdec" asf_muxer_select="riffenc" asf_stream_muxer_select="asf_muxer" -avi_demuxer_select="riffdec exif" +avi_demuxer_select="riffdec" avi_muxer_select="riffenc" avisynth_demuxer_deps="avisynth" avisynth_demuxer_select="riffdec" caf_demuxer_select="riffdec" dirac_demuxer_select="dirac_parser" -dts_demuxer_select="dca_parser" -dtshd_demuxer_select="dca_parser" dxa_demuxer_select="riffdec" eac3_demuxer_select="ac3_parser" f4v_muxer_select="mov_muxer" flac_demuxer_select="flac_parser" hds_muxer_select="flv_muxer" hls_muxer_select="mpegts_muxer" -image2_alias_pix_demuxer_select="image2_demuxer" -image2_brender_pix_demuxer_select="image2_demuxer" ipod_muxer_select="mov_muxer" ismv_muxer_select="mov_muxer" -libnut_demuxer_deps="libnut" -libnut_muxer_deps="libnut" matroska_audio_muxer_select="matroska_muxer" matroska_demuxer_select="riffdec" matroska_demuxer_suggest="bzlib lzo zlib" @@ -2373,69 +2032,40 @@ spdif_muxer_select="aac_parser" tak_demuxer_select="tak_parser" tg2_muxer_select="mov_muxer" tgp_muxer_select="mov_muxer" -vobsub_demuxer_select="mpegps_demuxer" w64_demuxer_select="wav_demuxer" -w64_muxer_select="wav_muxer" wav_demuxer_select="riffdec" wav_muxer_select="riffenc" webm_muxer_select="riffenc" wtv_demuxer_select="riffdec" -wtv_muxer_select="riffenc" xmv_demuxer_select="riffdec" xwma_demuxer_select="riffdec" # indevs / outdevs alsa_indev_deps="alsa_asoundlib_h snd_pcm_htimestamp" alsa_outdev_deps="alsa_asoundlib_h" -avfoundation_indev_extralibs="-framework CoreVideo -framework Foundation -framework AVFoundation -framework CoreMedia" -avfoundation_indev_select="avfoundation" bktr_indev_deps_any="dev_bktr_ioctl_bt848_h machine_ioctl_bt848_h dev_video_bktr_ioctl_bt848_h dev_ic_bt8xx_h" -caca_outdev_deps="libcaca" -decklink_outdev_deps="decklink pthreads" -decklink_outdev_extralibs="-lstdc++" -dshow_indev_deps="IBaseFilter" -dshow_indev_extralibs="-lpsapi -lole32 -lstrmiids -luuid" dv1394_indev_deps="dv1394" dv1394_indev_select="dv_demuxer" fbdev_indev_deps="linux_fb_h" -fbdev_outdev_deps="linux_fb_h" -gdigrab_indev_deps="CreateDIBSection" -gdigrab_indev_extralibs="-lgdi32" -gdigrab_indev_select="bmp_decoder" -iec61883_indev_deps="libiec61883" -jack_indev_deps="jack_jack_h sem_timedwait" -lavfi_indev_deps="avfilter" +jack_indev_deps="jack_jack_h pthreads" libcdio_indev_deps="libcdio" libdc1394_indev_deps="libdc1394" -libv4l2_indev_deps="libv4l2" -openal_indev_deps="openal" -opengl_outdev_deps="opengl" oss_indev_deps_any="soundcard_h sys_soundcard_h" oss_outdev_deps_any="soundcard_h sys_soundcard_h" pulse_indev_deps="libpulse" -pulse_outdev_deps="libpulse" -qtkit_indev_extralibs="-framework QTKit -framework Foundation -framework QuartzCore" -qtkit_indev_select="qtkit" -sdl_outdev_deps="sdl" sndio_indev_deps="sndio_h" sndio_outdev_deps="sndio_h" -v4l_indev_deps="linux_videodev_h" v4l2_indev_deps_any="linux_videodev2_h sys_videoio_h" -v4l2_outdev_deps_any="linux_videodev2_h sys_videoio_h" vfwcap_indev_deps="capCreateCaptureWindow vfwcap_defines" vfwcap_indev_extralibs="-lavicap32" -xv_outdev_deps="X11_extensions_Xvlib_h XvGetPortAttribute" -xv_outdev_extralibs="-lXv -lX11 -lXext" -x11grab_indev_deps="x11grab" +x11grab_indev_deps="x11grab XShmCreateImage" # protocols -bluray_protocol_deps="libbluray" ffrtmpcrypt_protocol_deps="!librtmp_protocol" ffrtmpcrypt_protocol_deps_any="gcrypt nettle openssl" ffrtmpcrypt_protocol_select="tcp_protocol" ffrtmphttp_protocol_deps="!librtmp_protocol" ffrtmphttp_protocol_select="http_protocol" -ftp_protocol_select="tcp_protocol" gopher_protocol_select="network" http_protocol_select="tcp_protocol" httpproxy_protocol_select="tcp_protocol" @@ -2445,7 +2075,6 @@ librtmpe_protocol_deps="librtmp" librtmps_protocol_deps="librtmp" librtmpt_protocol_deps="librtmp" librtmpte_protocol_deps="librtmp" -libssh_protocol_deps="libssh" mmsh_protocol_select="http_protocol" mmst_protocol_select="network" rtmp_protocol_deps="!librtmp_protocol" @@ -2468,85 +2097,28 @@ unix_protocol_deps="sys_un_h" unix_protocol_select="network" # filters -aconvert_filter_deps="swresample" -amovie_filter_deps="avcodec avformat" -aresample_filter_deps="swresample" -ass_filter_deps="libass" -asyncts_filter_deps="avresample" -atempo_filter_deps="avcodec" -atempo_filter_select="rdft" -azmq_filter_deps="libzmq" blackframe_filter_deps="gpl" boxblur_filter_deps="gpl" bs2b_filter_deps="libbs2b" -colormatrix_filter_deps="gpl" cropdetect_filter_deps="gpl" -dctdnoiz_filter_deps="avcodec" -dctdnoiz_filter_select="dct" delogo_filter_deps="gpl" -deshake_filter_deps="avcodec" -deshake_filter_select="dsputil" drawtext_filter_deps="libfreetype" -ebur128_filter_deps="gpl" -flite_filter_deps="libflite" frei0r_filter_deps="frei0r dlopen" frei0r_filter_extralibs='$ldl' frei0r_src_filter_deps="frei0r dlopen" frei0r_src_filter_extralibs='$ldl' -geq_filter_deps="gpl" -histeq_filter_deps="gpl" hqdn3d_filter_deps="gpl" interlace_filter_deps="gpl" -kerndeint_filter_deps="gpl" -ladspa_filter_deps="ladspa dlopen" -ladspa_filter_extralibs='$ldl' -mcdeint_filter_deps="avcodec gpl" -movie_filter_deps="avcodec avformat" -mp_filter_deps="gpl avcodec swscale inline_asm" -mpdecimate_filter_deps="gpl avcodec" -mptestsrc_filter_deps="gpl" -negate_filter_deps="lut_filter" -perspective_filter_deps="gpl" ocv_filter_deps="libopencv" -owdenoise_filter_deps="gpl" -pan_filter_deps="swresample" -phase_filter_deps="gpl" -pp_filter_deps="gpl postproc" -pullup_filter_deps="gpl" -removelogo_filter_deps="avcodec avformat swscale" resample_filter_deps="avresample" -sab_filter_deps="gpl swscale" scale_filter_deps="swscale" -smartblur_filter_deps="gpl swscale" -showspectrum_filter_deps="avcodec" -showspectrum_filter_select="rdft" -spp_filter_deps="gpl avcodec" -spp_filter_select="fft" -stereo3d_filter_deps="gpl" -subtitles_filter_deps="avformat avcodec libass" -super2xsai_filter_deps="gpl" -tinterlace_filter_deps="gpl" -vidstabdetect_filter_deps="libvidstab" -vidstabtransform_filter_deps="libvidstab" -pixfmts_super2xsai_test_deps="super2xsai_filter" -tinterlace_merge_test_deps="tinterlace_filter" -tinterlace_pad_test_deps="tinterlace_filter" -zmq_filter_deps="libzmq" # examples -avio_reading="avformat avcodec avutil" avcodec_example_deps="avcodec avutil" -demuxing_decoding_example_deps="avcodec avformat avutil" filter_audio_example_deps="avfilter avutil" -filtering_audio_example_deps="avfilter avcodec avformat avutil" -filtering_video_example_deps="avfilter avcodec avformat avutil" metadata_example_deps="avformat avutil" -muxing_example_deps="avcodec avformat avutil swscale" -remuxing_example_deps="avcodec avformat avutil" -resampling_audio_example_deps="avutil swresample" -scaling_video_example_deps="avutil swscale" -transcode_aac_example_deps="avcodec avformat swresample" -transcoding_example_deps="avfilter avcodec avformat avutil" +output_example_deps="avcodec avformat avutil swscale" +transcode_aac_example_deps="avcodec avformat avresample" # libraries, in linking order avcodec_deps="avutil" @@ -2554,28 +2126,23 @@ avdevice_deps="avformat avcodec avutil" avfilter_deps="avutil" avformat_deps="avcodec avutil" avresample_deps="avutil" -postproc_deps="avutil gpl" -swresample_deps="avutil" swscale_deps="avutil" # programs -ffmpeg_deps="avcodec avfilter avformat swresample" -ffmpeg_select="aformat_filter anull_filter atrim_filter format_filter - null_filter +avconv_deps="avcodec avfilter avformat avresample swscale" +avconv_select="aformat_filter anull_filter asyncts_filter atrim_filter format_filter + fps_filter null_filter resample_filter scale_filter setpts_filter trim_filter" -ffplay_deps="avcodec avformat swscale swresample sdl" -ffplay_libs='$sdl_libs' -ffplay_select="rdft crop_filter transpose_filter hflip_filter vflip_filter rotate_filter" -ffprobe_deps="avcodec avformat" -ffserver_deps="avformat fork sarestart" -ffserver_select="ffm_muxer rtp_protocol rtsp_demuxer" +avplay_deps="avcodec avformat avresample swscale sdl" +avplay_libs='$sdl_libs' +avplay_select="rdft" +avprobe_deps="avcodec avformat" +avserver_deps="avformat fork !shared" +avserver_select="ffm_muxer rtp_protocol rtsp_demuxer" # documentation -podpages_deps="perl" -manpages_deps="perl pod2man" -htmlpages_deps="perl texi2html" -txtpages_deps="perl makeinfo" -doc_deps_any="manpages htmlpages podpages txtpages" +pod2man_deps="doc" +texi2html_deps="doc" # default parameters @@ -2584,8 +2151,8 @@ logfile="config.log" # installation paths prefix_default="/usr/local" bindir_default='${prefix}/bin' -datadir_default='${prefix}/share/ffmpeg' -docdir_default='${prefix}/share/doc/ffmpeg' +datadir_default='${prefix}/share/avconv' +docdir_default='${prefix}/share/doc/libav' incdir_default='${prefix}/include' libdir_default='${prefix}/lib' mandir_default='${prefix}/share/man' @@ -2594,52 +2161,39 @@ shlibdir_default="$libdir_default" # toolchain ar_default="ar" cc_default="gcc" -cxx_default="g++" host_cc_default="gcc" cp_f="cp -f" -install="install" ln_s="ln -s -f" nm_default="nm -g" objformat="elf" pkg_config_default=pkg-config -ranlib_default="ranlib" -strip_default="strip" -yasmexe_default="yasm" -windres_default="windres" +ranlib="ranlib" +strip="strip" +yasmexe="yasm" nogas=":" +# machine +arch_default=$(uname -m) +cpu="generic" + # OS target_os_default=$(tolower $(uname -s)) host_os=$target_os_default -# machine -if test "$target_os_default" = aix; then - arch_default=$(uname -p) -else - arch_default=$(uname -m) -fi -cpu="generic" - # configurable options -enable $PROGRAM_LIST -enable $DOCUMENT_LIST -enable $EXAMPLE_LIST -enable $(filter_out avresample $LIBRARY_LIST) -enable stripping +enable $EXAMPLE_LIST $LIBRARY_LIST $PROGRAM_LIST enable asm enable debug enable doc enable optimizations -enable runtime_cpudetect enable safe_bitstream_reader enable static enable swscale_alpha -# Enable hwaccels by default. -enable dxva2 vaapi vda vdpau xvmc -enable xlib +# By default, enable only those hwaccels that have no external dependencies. +enable dxva2 vdpau # build settings SHFLAGS='-shared -Wl,-soname,$$(@F)' @@ -2665,8 +2219,6 @@ AS_O='-o $@' CC_C='-c' CC_E='-E -o $@' CC_O='-o $@' -CXX_C='-c' -CXX_O='-o $@' LD_O='-o $@' LD_LIB='-l%' LD_PATH='-L' @@ -2701,7 +2253,7 @@ for v in "$@"; do r=${v#*=} l=${v%"$r"} r=$(sh_quote "$r") - FFMPEG_CONFIGURATION="${FFMPEG_CONFIGURATION# } ${l}${r}" + LIBAV_CONFIGURATION="${LIBAV_CONFIGURATION# } ${l}${r}" done find_things(){ @@ -2821,9 +2373,7 @@ for opt do is_in "${thing}s" $COMPONENT_LIST || die_unknown "$opt" eval list=\$$(toupper $thing)_LIST name=$(echo "${optval}" | sed "s/,/_${thing}|/g")_${thing} - list=$(filter "$name" $list) - [ "$list" = "" ] && warn "Option $opt did not match anything" - $action $list + $action $(filter "$name" $list) ;; --enable-?*|--disable-?*) eval $(echo "$opt" | sed 's/--/action=/;s/-/ option=/;s/-/_/g') @@ -2844,8 +2394,6 @@ for opt do ;; --help|-h) show_help ;; - --fatal-warnings) enable fatal_warnings - ;; *) optname="${opt%%=*}" optname="${optname#--}" @@ -2863,7 +2411,7 @@ done disabled logging && logfile=/dev/null -echo "# $0 $FFMPEG_CONFIGURATION" > $logfile +echo "# $0 $LIBAV_CONFIGURATION" > $logfile set >> $logfile test -n "$cross_prefix" && enable cross_compile @@ -2875,17 +2423,13 @@ fi ar_default="${cross_prefix}${ar_default}" cc_default="${cross_prefix}${cc_default}" -cxx_default="${cross_prefix}${cxx_default}" nm_default="${cross_prefix}${nm_default}" pkg_config_default="${cross_prefix}${pkg_config_default}" -ranlib_default="${cross_prefix}${ranlib_default}" -strip_default="${cross_prefix}${strip_default}" -windres_default="${cross_prefix}${windres_default}" +ranlib="${cross_prefix}${ranlib}" +strip="${cross_prefix}${strip}" sysinclude_default="${sysroot}/usr/include" -test -n "$valgrind" && toolchain="valgrind-memcheck" - case "$toolchain" in clang-asan) cc_default="clang" @@ -2899,7 +2443,7 @@ case "$toolchain" in ;; clang-usan) cc_default="clang" - add_cflags -fsanitize=undefined + add_cflags -fsanitize=undefined -O1 add_ldflags -fsanitize=undefined ;; gcc-asan) @@ -2918,17 +2462,17 @@ case "$toolchain" in add_ldflags -fsanitize=undefined ;; valgrind-massif) - target_exec_default=${valgrind:-"valgrind"} - target_exec_args="--tool=massif --alloc-fn=av_malloc --alloc-fn=av_mallocz --alloc-fn=av_calloc --alloc-fn=av_fast_padded_malloc --alloc-fn=av_fast_malloc --alloc-fn=av_realloc_f --alloc-fn=av_fast_realloc --alloc-fn=av_realloc" + target_exec_default="valgrind" + target_exec_args="--alloc-fn=av_malloc --alloc-fn=av_mallocz" ;; valgrind-memcheck) - target_exec_default=${valgrind:-"valgrind"} - target_exec_args="--error-exitcode=1 --malloc-fill=0x2a --track-origins=yes --leak-check=full --gen-suppressions=all --suppressions=$source_path/tests/fate-valgrind.supp" + target_exec_default="valgrind" + target_exec_args="--track-origins=yes --leak-check=full" ;; msvc) # Check whether the current MSVC version needs the C99 converter. # From MSVC 2013 (compiler major version 18) onwards, it does actually - # support enough of C99 to build ffmpeg. Default to the new + # support enough of C99 to build libav. Default to the new # behaviour if the regexp was unable to match anything, since this # successfully parses the version number of existing supported # versions that require the converter (MSVC 2010 and 2012). @@ -2969,7 +2513,7 @@ case "$toolchain" in ;; esac -set_default arch cc cxx pkg_config ranlib strip sysinclude target_exec target_os yasmexe +set_default arch cc pkg_config sysinclude target_exec target_os enabled cross_compile || host_cc_default=$cc set_default host_cc @@ -2992,11 +2536,7 @@ HOSTEXESUF=$(exesuf $host_os) : ${TMPDIR:=$TMP} : ${TMPDIR:=/tmp} -if [ -n "$tempprefix" ] ; then - mktemp(){ - echo $tempprefix.${HOSTNAME}.${UID} - } -elif ! check_cmd mktemp -u XXXXXX; then +if ! check_cmd mktemp -u XXXXXX; then # simple replacement for missing mktemp # NOT SAFE FOR GENERAL USE mktemp(){ @@ -3016,10 +2556,8 @@ trap 'rm -f -- $TMPFILES' EXIT tmpfile TMPASM .asm tmpfile TMPC .c -tmpfile TMPCPP .cpp tmpfile TMPE $EXESUF tmpfile TMPH .h -tmpfile TMPM .m tmpfile TMPO .o tmpfile TMPS .S tmpfile TMPSH .sh @@ -3105,7 +2643,6 @@ msvc_flags(){ -Wall) echo -W4 -wd4244 -wd4127 -wd4018 -wd4389 \ -wd4146 -wd4057 -wd4204 -wd4706 -wd4305 \ -wd4152 -wd4324 -we4013 -wd4100 -wd4214 \ - -wd4554 \ -wd4273 ;; esac done @@ -3123,15 +2660,6 @@ icl_flags(){ done } -icc_flags(){ - for flag; do - case $flag in - -flto) echo -ipo ;; - *) echo $flag ;; - esac - done -} - pgi_flags(){ for flag; do case $flag in @@ -3160,16 +2688,9 @@ suncc_flags(){ prescott|nocona) echo -xarch=sse3 -xchip=pentium4 ;; *-sse3) echo -xarch=sse3 ;; core2) echo -xarch=ssse3 -xchip=core2 ;; - bonnell) echo -xarch=ssse3 ;; - corei7|nehalem) echo -xtarget=nehalem ;; - westmere) echo -xtarget=westmere ;; - silvermont) echo -xarch=sse4_2 ;; - corei7-avx|sandybridge) echo -xtarget=sandybridge ;; - core-avx*|ivybridge|haswell|broadwell) - echo -xarch=avx ;; - amdfam10|barcelona) echo -xtarget=barcelona ;; - btver1) echo -xarch=amdsse4a ;; - btver2|bdver*) echo -xarch=avx ;; + corei7) echo -xarch=sse4_2 -xchip=nehalem ;; + corei7-avx) echo -xarch=avx -xchip=sandybridge ;; + amdfam10|barcelona|bdver*) echo -xarch=sse4_1 ;; athlon-4|athlon-[mx]p) echo -xarch=ssea ;; k8|opteron|athlon64|athlon-fx) echo -xarch=sse2a ;; @@ -3225,9 +2746,7 @@ probe_cc(){ unset _depflags _DEPCMD _DEPFLAGS _flags_filter=echo - if $_cc --version 2>&1 | grep -q '^GNU assembler'; then - true # no-op to avoid reading stdin in following checks - elif $_cc -v 2>&1 | grep -q '^gcc.*LLVM'; then + if $_cc -v 2>&1 | grep -q '^gcc.*LLVM'; then _type=llvm_gcc gcc_extra_ver=$(expr "$($_cc --version | head -n1)" : '.*\((.*)\)') _ident="llvm-gcc $($_cc -dumpversion) $gcc_extra_ver" @@ -3253,7 +2772,6 @@ probe_cc(){ _cflags_speed='-O3' _cflags_size='-Os' _cflags_noopt='-O1' - _flags_filter=icc_flags elif $_cc -v 2>&1 | grep -q xlc; then _type=xlc _ident=$($_cc -qversion 2>/dev/null | head -n1) @@ -3327,7 +2845,7 @@ probe_cc(){ opt_common='-alias=ansi -Mdse -Mlre -Mpre' _cflags_speed="-O3 -Mautoinline -Munroll=c:4 $opt_common" _cflags_size="-O2 -Munroll=c:1 $opt_common" - _cflags_noopt="-O" + _cflags_noopt="-O1" _flags_filter=pgi_flags elif $_cc 2>&1 | grep -q Microsoft; then _type=msvc @@ -3351,7 +2869,6 @@ probe_cc(){ if [ $pfx = hostcc ]; then append _cflags -Dsnprintf=_snprintf fi - disable stripping elif $_cc 2>&1 | grep -q Intel; then _type=icl _ident=$($cc 2>&1 | head -n1) @@ -3372,9 +2889,9 @@ probe_cc(){ _ld_path='-libpath:' # -Qdiag-error to make icl error when seeing certain unknown arguments _flags='-nologo -Qdiag-error:4044,10157' - # -Qvec- -Qsimd- to prevent miscompilation, -GS, fp:precise for consistency + # -Qvec- -Qsimd- to prevent miscompilation, -GS for consistency # with MSVC which enables it by default. - _cflags='-D_USE_MATH_DEFINES -FIstdlib.h -Dstrtoll=_strtoi64 -Qms0 -Qvec- -Qsimd- -GS -fp:precise' + _cflags='-D_USE_MATH_DEFINES -FIstdlib.h -Dstrtoll=_strtoi64 -Qms0 -Qvec- -Qsimd- -GS' if [ $pfx = hostcc ]; then append _cflags -Dsnprintf=_snprintf fi @@ -3426,7 +2943,7 @@ test -n "$cc_type" && enable $cc_type || : ${dep_cc_default:=$cc} : ${ld_default:=$cc} : ${host_ld_default:=$host_cc} -set_default ar as dep_cc ld host_ld windres +set_default ar as dep_cc ld host_ld probe_cc as "$as" asflags_filter=$_flags_filter @@ -3468,7 +2985,6 @@ else fi add_cflags $extra_cflags -add_cxxflags $extra_cxxflags add_asflags $extra_cflags if test -n "$sysroot"; then @@ -3476,9 +2992,6 @@ if test -n "$sysroot"; then gcc|llvm_gcc|clang) add_cppflags --sysroot="$sysroot" add_ldflags --sysroot="$sysroot" -# On Darwin --sysroot may be ignored, -isysroot always affects headers and linking - add_cppflags -isysroot "$sysroot" - add_ldflags -isysroot "$sysroot" ;; tms470) add_cppflags -I"$sysinclude" @@ -3514,7 +3027,7 @@ case "$arch" in aarch64|arm64) arch="aarch64" ;; - arm*|iPad*|iPhone*) + arm*) arch="arm" ;; mips*|IP*) @@ -3538,7 +3051,7 @@ case "$arch" in tilegx|tile-gx) arch="tilegx" ;; - i[3-6]86|i86pc|BePC|x86pc|x86_64|x86_32|amd64) + i[3-6]86|i86pc|BePC|x86pc|x86_64|amd64) arch="x86" ;; esac @@ -3616,10 +3129,7 @@ elif enabled arm; then case "$subarch" in armv5t*) enable fast_clz ;; - armv[6-8]*) - enable fast_clz - disabled fast_unaligned || enable fast_unaligned - ;; + armv[6-8]*) enable fast_clz fast_unaligned ;; esac elif enabled avr32; then @@ -3651,28 +3161,6 @@ elif enabled mips; then cpuflags="-march=$cpu" - case $cpu in - 24kc) - disable mipsfpu - disable mipsdspr1 - disable mipsdspr2 - ;; - 24kf*) - disable mipsdspr1 - disable mipsdspr2 - ;; - 24kec|34kc|1004kc) - disable mipsfpu - disable mipsdspr2 - ;; - 24kef*|34kf*|1004kf*) - disable mipsdspr2 - ;; - 74kc) - disable mipsfpu - ;; - esac - elif enabled ppc; then disable ldbrx @@ -3750,8 +3238,7 @@ elif enabled x86; then disable i686 ;; # targets that do support nopl and conditional mov (cmov) - i686|pentiumpro|pentium[23]|pentium-m|athlon|athlon-tbird|athlon-4|athlon-[mx]p|athlon64*|k8*|opteron*|athlon-fx\ - |core*|atom|bonnell|nehalem|westmere|silvermont|sandybridge|ivybridge|haswell|broadwell|amdfam10|barcelona|b[dt]ver*) + i686|pentiumpro|pentium[23]|pentium-m|athlon|athlon-tbird|athlon-4|athlon-[mx]p|athlon64*|k8*|opteron*|athlon-fx|core2|corei7*|amdfam10|barcelona|atom|bdver*) cpuflags="-march=$cpu" enable i686 enable fast_cmov @@ -3785,7 +3272,6 @@ if test "$?" != 0; then fi add_cppflags -D_ISOC99_SOURCE -add_cxxflags -D__STDC_CONSTANT_MACROS check_cflags -std=c99 check_cc -D_FILE_OFFSET_BITS=64 < @@ -3837,11 +3323,6 @@ case "$arch" in spic=$shared fi ;; - ppc) - check_cc < $$(@:$(SLIBSUF)=.def); lib.exe /machine:$(LIBTARGET) /def:$$(@:$(SLIBSUF)=.def) /out:$(SUBDIR)$(SLIBNAME:$(SLIBSUF)=.lib)' - if enabled x86_64; then - LIBTARGET=x64 - fi - elif check_cmd $dlltool --version; then - SLIB_EXTRA_CMD=-'sed -e "s/ @[^ ]*//" $$(@:$(SLIBSUF)=.orig.def) > $$(@:$(SLIBSUF)=.def); $(DLLTOOL) -m $(LIBTARGET) -d $$(@:$(SLIBSUF)=.def) -l $(SUBDIR)$(SLIBNAME:$(SLIBSUF)=.lib) -D $(SLIBNAME_WITH_MAJOR)' - fi + SLIB_EXTRA_CMD=-'sed -e "s/ @[^ ]*//" $$(@:$(SLIBSUF)=.orig.def) > $$(@:$(SLIBSUF)=.def); $(DLLTOOL) -m $(LIBTARGET) -d $$(@:$(SLIBSUF)=.def) -l $(SUBDIR)$(SLIBNAME:$(SLIBSUF)=.lib) -D $(SLIBNAME_WITH_MAJOR)' SLIB_INSTALL_NAME='$(SLIBNAME_WITH_MAJOR)' SLIB_INSTALL_LINKS= SLIB_INSTALL_EXTRA_SHLIB='$(SLIBNAME:$(SLIBSUF)=.lib)' SLIB_INSTALL_EXTRA_LIB='lib$(SLIBNAME:$(SLIBSUF)=.dll.a) $(SLIBNAME_WITH_MAJOR:$(SLIBSUF)=.def)' SHFLAGS='-shared -Wl,--output-def,$$(@:$(SLIBSUF)=.orig.def) -Wl,--out-implib,$(SUBDIR)lib$(SLIBNAME:$(SLIBSUF)=.dll.a) -Wl,--enable-runtime-pseudo-reloc -Wl,--enable-auto-image-base' objformat="win32" + dlltool="${cross_prefix}dlltool" ranlib=: enable dos_paths ;; @@ -3986,7 +3456,6 @@ case $target_os in SHFLAGS='-shared -Wl,--out-implib,$(SUBDIR)lib$(FULLNAME).dll.a' objformat="win32" enable dos_paths - enabled shared && ! enabled small && check_cmd $windres --version && enable gnu_windres ;; *-dos|freedos|opendos) network_extralibs="-lsocket" @@ -4002,7 +3471,6 @@ case $target_os in ranlib="echo ignoring ranlib" ;; os/2*) - strip="lxlite -CS" ln_s="cp -f" objformat="aout" add_cppflags -D_GNU_SOURCE @@ -4023,7 +3491,6 @@ case $target_os in emximp -o $(SUBDIR)$(LIBPREF)$(NAME)_dll.lib $(SUBDIR)$(NAME).def;' SLIB_INSTALL_EXTRA_LIB='$(LIBPREF)$(NAME)_dll.a $(LIBPREF)$(NAME)_dll.lib' enable dos_paths - enable_weak os2threads ;; gnu/kfreebsd) add_cppflags -D_BSD_SOURCE @@ -4060,7 +3527,7 @@ case $target_os in add_compat strtod.o strtod=avpriv_strtod network_extralibs='-lbsd' exeobjs=compat/plan9/main.o - disable ffserver + disable avserver cp_f='cp' ;; none) @@ -4089,10 +3556,6 @@ probe_libc(){ # MinGW64 is backwards compatible with MinGW32, so check for it first. elif check_${pfx}cpp_condition _mingw.h "defined __MINGW64_VERSION_MAJOR"; then eval ${pfx}libc_type=mingw64 - if check_${pfx}cpp_condition _mingw.h "__MINGW64_VERSION_MAJOR < 3"; then - add_compat msvcrt/snprintf.o - add_cflags "-include $source_path/compat/msvcrt/snprintf.h" - fi add_${pfx}cppflags -U__STRICT_ANSI__ -D__USE_MINGW_ANSI_STDIO=1 eval test \$${pfx_no_}cc_type = "gcc" && add_${pfx}cppflags -D__printf__=__gnu_printf__ @@ -4163,7 +3626,7 @@ esc(){ echo "$*" | sed 's/%/%25/g;s/:/%3a/g' } -echo "config:$arch:$subarch:$cpu:$target_os:$(esc $cc_ident):$(esc $FFMPEG_CONFIGURATION)" >config.fate +echo "config:$arch:$subarch:$cpu:$target_os:$(esc $cc_ident):$(esc $LIBAV_CONFIGURATION)" >config.fate check_cpp_condition stdlib.h "defined(__PIC__) || defined(__pic__) || defined(PIC)" && enable_weak pic @@ -4185,25 +3648,16 @@ die_license_disabled() { enabled $1 || { enabled $2 && die "$2 is $1 and --enable-$1 is not specified."; } } -die_license_disabled_gpl() { - enabled $1 || { enabled $2 && die "$2 is incompatible with the gpl and --enable-$1 is not specified."; } -} - -die_license_disabled gpl frei0r die_license_disabled gpl libcdio -die_license_disabled gpl libutvideo -die_license_disabled gpl libvidstab die_license_disabled gpl libx264 die_license_disabled gpl libx265 die_license_disabled gpl libxavs die_license_disabled gpl libxvid -die_license_disabled gpl libzvbi die_license_disabled gpl x11grab -die_license_disabled nonfree libaacplus die_license_disabled nonfree libfaac -enabled gpl && die_license_disabled_gpl nonfree libfdk_aac -enabled gpl && die_license_disabled_gpl nonfree openssl +die_license_disabled nonfree libfdk_aac +die_license_disabled nonfree openssl die_license_disabled version3 libopencore_amrnb die_license_disabled version3 libopencore_amrwb @@ -4266,8 +3720,6 @@ od -t x1 $TMPO | grep -q '42 *49 *47 *45' && enable bigendian check_inline_asm inline_asm_labels '"1:\n"' -check_inline_asm inline_asm_nonlocal_labels '"Label:\n"' - if enabled aarch64; then enabled armv8 && check_insn armv8 'prfm pldl1strm, [x0]' # internal assembler in clang 3.3 does not support this instruction @@ -4282,10 +3734,7 @@ elif enabled alpha; then elif enabled arm; then - check_cpp_condition stddef.h "defined __thumb__" && check_cc <= 2.0.0" aacplus.h aacplusEncOpen -laacplus -enabled libass && require_pkg_config libass ass/ass.h ass_library_init -enabled libbluray && require_pkg_config libbluray libbluray/bluray.h bd_open enabled libbs2b && require_pkg_config libbs2b bs2b.h bs2b_open -enabled libcelt && require libcelt celt/celt.h celt_decode -lcelt0 && - { check_lib celt/celt.h celt_decoder_create_custom -lcelt0 || - die "ERROR: libcelt must be installed and version must be >= 0.11.0."; } -enabled libcaca && require_pkg_config caca caca.h caca_create_canvas enabled libfaac && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac enabled libfdk_aac && require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac -flite_libs="-lflite_cmu_time_awb -lflite_cmu_us_awb -lflite_cmu_us_kal -lflite_cmu_us_kal16 -lflite_cmu_us_rms -lflite_cmu_us_slt -lflite_usenglish -lflite_cmulex -lflite" -enabled libflite && require2 libflite "flite/flite.h" flite_init $flite_libs -enabled fontconfig && enable libfontconfig enabled libfontconfig && require_pkg_config fontconfig "fontconfig/fontconfig.h" FcInit -enabled libfreetype && require_libfreetype -enabled libgme && require libgme gme/gme.h gme_new_emu -lgme -lstdc++ +enabled libfreetype && require_pkg_config freetype2 "ft2build.h FT_FREETYPE_H" FT_Init_FreeType enabled libgsm && { for gsm_hdr in "gsm.h" "gsm/gsm.h"; do check_lib "${gsm_hdr}" gsm_create -lgsm && break; done || die "ERROR: libgsm not found"; } enabled libilbc && require libilbc ilbc.h WebRtcIlbcfix_InitDecode -lilbc -enabled libmodplug && require libmodplug libmodplug/modplug.h ModPlug_Load -lmodplug enabled libmp3lame && require "libmp3lame >= 3.98.3" lame/lame.h lame_set_VBR_quality -lmp3lame -enabled libnut && require libnut libnut.h nut_demuxer_init -lnut enabled libopencore_amrnb && require libopencore_amrnb opencore-amrnb/interf_dec.h Decoder_Interface_init -lopencore-amrnb enabled libopencore_amrwb && require libopencore_amrwb opencore-amrwb/dec_if.h D_IF_init -lopencore-amrwb -enabled libopencv && require_pkg_config opencv opencv/cxcore.h cvCreateImageHeader -enabled libopenjpeg && { check_lib openjpeg-1.5/openjpeg.h opj_version -lopenjpeg -DOPJ_STATIC || - check_lib openjpeg.h opj_version -lopenjpeg -DOPJ_STATIC || - die "ERROR: libopenjpeg not found"; } +enabled libopencv && require_pkg_config opencv opencv/cv.h cvCreateImageHeader +enabled libopenjpeg && { { check_header openjpeg.h && check_lib2 openjpeg.h opj_version -lopenjpeg -DOPJ_STATIC; } || + { require_pkg_config libopenjpeg1 openjpeg.h opj_version -DOPJ_STATIC; } } enabled libopus && require_pkg_config opus opus_multistream.h opus_multistream_decoder_create enabled libpulse && require_pkg_config libpulse-simple pulse/simple.h pa_simple_new -enabled libquvi && require_pkg_config libquvi quvi/quvi.h quvi_init enabled librtmp && require_pkg_config librtmp librtmp/rtmp.h RTMP_Socket enabled libschroedinger && require_pkg_config schroedinger-1.0 schroedinger/schro.h schro_init -enabled libshine && require_pkg_config shine shine/layer3.h shine_encode_buffer -enabled libsoxr && require libsoxr soxr.h soxr_create -lsoxr -enabled libssh && require_pkg_config libssh libssh/sftp.h sftp_init enabled libspeex && require libspeex speex/speex.h speex_decoder_init -lspeex -enabled libstagefright_h264 && require_cpp libstagefright_h264 "binder/ProcessState.h media/stagefright/MetaData.h - media/stagefright/MediaBufferGroup.h media/stagefright/MediaDebug.h media/stagefright/MediaDefs.h - media/stagefright/OMXClient.h media/stagefright/OMXCodec.h" android::OMXClient -lstagefright -lmedia -lutils -lbinder -lgnustl_static enabled libtheora && require libtheora theora/theoraenc.h th_info_init -ltheoraenc -ltheoradec -logg -enabled libtwolame && require libtwolame twolame.h twolame_init -ltwolame && - { check_lib twolame.h twolame_encode_buffer_float32_interleaved -ltwolame || - die "ERROR: libtwolame must be installed and version must be >= 0.3.10"; } -enabled libutvideo && require_cpp utvideo "stdint.h stdlib.h utvideo/utvideo.h utvideo/Codec.h" 'CCodec*' -lutvideo -lstdc++ -enabled libv4l2 && require_pkg_config libv4l2 libv4l2.h v4l2_ioctl -enabled libvidstab && require_pkg_config "vidstab >= 0.98" vid.stab/libvidstab.h vsMotionDetectInit +enabled libtwolame && require libtwolame twolame.h twolame_init -ltwolame enabled libvo_aacenc && require libvo_aacenc vo-aacenc/voAAC.h voGetAACEncAPI -lvo-aacenc enabled libvo_amrwbenc && require libvo_amrwbenc vo-amrwbenc/enc_if.h E_IF_init -lvo-amrwbenc enabled libvorbis && require libvorbis vorbis/vorbisenc.h vorbis_info_init -lvorbisenc -lvorbis -logg enabled libvpx && { enabled libvpx_vp8_decoder && { check_lib2 "vpx/vpx_decoder.h vpx/vp8dx.h" vpx_codec_dec_init_ver -lvpx || die "ERROR: libvpx decoder version must be >=0.9.1"; } - enabled libvpx_vp8_encoder && { check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_enc_init_ver VP8E_SET_MAX_INTRA_BITRATE_PCT" -lvpx || - die "ERROR: libvpx encoder version must be >=0.9.7"; } + enabled libvpx_vp8_encoder && { check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_enc_init_ver VPX_CQ" -lvpx || + die "ERROR: libvpx encoder version must be >=0.9.6"; } enabled libvpx_vp9_decoder && { check_lib2 "vpx/vpx_decoder.h vpx/vp8dx.h" "vpx_codec_vp9_dx" -lvpx || disable libvpx_vp9_decoder; } - enabled libvpx_vp9_encoder && { check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_vp9_cx VP9E_SET_SVC" -lvpx || disable libvpx_vp9_encoder; } } + enabled libvpx_vp9_encoder && { check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_vp9_cx" -lvpx || disable libvpx_vp9_encoder; } } enabled libwavpack && require libwavpack wavpack/wavpack.h WavpackOpenFileOutput -lwavpack enabled libwebp && require_pkg_config libwebp webp/encode.h WebPGetEncoderVersion enabled libx264 && require libx264 x264.h x264_encoder_encode -lx264 && { check_cpp_condition x264.h "X264_BUILD >= 118" || - die "ERROR: libx264 must be installed and version must be >= 0.118."; } + die "ERROR: libx264 version must be >= 0.118."; } enabled libx265 && require_pkg_config x265 x265.h x265_encoder_encode && { check_cpp_condition x265.h "X265_BUILD >= 13" || die "ERROR: libx265 version must be >= 13."; } enabled libxavs && require libxavs xavs.h xavs_encoder_encode -lxavs enabled libxvid && require libxvid xvid.h xvid_global -lxvidcore -enabled libzmq && require_pkg_config libzmq zmq.h zmq_ctx_new -enabled libzvbi && require libzvbi libzvbi.h vbi_decoder_new -lzvbi -enabled openal && { { for al_libs in "${OPENAL_LIBS}" "-lopenal" "-lOpenAL32"; do - check_lib 'AL/al.h' alGetError "${al_libs}" && break; done } || - die "ERROR: openal not found"; } && - { check_cpp_condition "AL/al.h" "defined(AL_VERSION_1_1)" || - die "ERROR: openal must be installed and version must be 1.1 or compatible"; } -enabled opencl && { check_lib2 OpenCL/cl.h clEnqueueNDRangeKernel -Wl,-framework,OpenCL || - check_lib2 CL/cl.h clEnqueueNDRangeKernel -lOpenCL || - die "ERROR: opencl not found"; } && - { check_cpp_condition "OpenCL/cl.h" "defined(CL_VERSION_1_2)" || - check_cpp_condition "CL/cl.h" "defined(CL_VERSION_1_2)" || - die "ERROR: opencl must be installed and version must be 1.2 or compatible"; } -enabled opengl && { check_lib GL/glx.h glXGetProcAddress "-lGL" || - check_lib2 windows.h wglGetProcAddress "-lopengl32 -lgdi32" || - check_lib2 OpenGL/gl3.h glGetError "-Wl,-framework,OpenGL" || - check_lib2 ES2/gl.h glGetError "-isysroot=${sysroot} -Wl,-framework,OpenGLES" || - die "ERROR: opengl not found." - } enabled openssl && { check_lib openssl/ssl.h SSL_library_init -lssl -lcrypto || check_lib openssl/ssl.h SSL_library_init -lssl32 -leay32 || check_lib openssl/ssl.h SSL_library_init -lssl -lcrypto -lws2_32 -lgdi32 || die "ERROR: openssl not found"; } -enabled qtkit_indev && { check_header_oc QTKit/QTKit.h || disable qtkit_indev; } if enabled gnutls; then { check_lib nettle/bignum.h nettle_mpz_get_str_256 -lnettle -lhogweed -lgmp && enable nettle; } || @@ -4742,33 +4103,18 @@ if enabled libdc1394; then die "ERROR: No version of libdc1394 found " fi -SDL_CONFIG="${cross_prefix}sdl-config" if check_pkg_config sdl SDL_events.h SDL_PollEvent; then check_cpp_condition SDL.h "(SDL_MAJOR_VERSION<<16 | SDL_MINOR_VERSION<<8 | SDL_PATCHLEVEL) >= 0x010201" $sdl_cflags && check_cpp_condition SDL.h "(SDL_MAJOR_VERSION<<16 | SDL_MINOR_VERSION<<8 | SDL_PATCHLEVEL) < 0x010300" $sdl_cflags && enable sdl -else - if "${SDL_CONFIG}" --version > /dev/null 2>&1; then - sdl_cflags=$("${SDL_CONFIG}" --cflags) - sdl_libs=$("${SDL_CONFIG}" --libs) - check_func_headers SDL_version.h SDL_Linked_Version $sdl_cflags $sdl_libs && - check_cpp_condition SDL.h "(SDL_MAJOR_VERSION<<16 | SDL_MINOR_VERSION<<8 | SDL_PATCHLEVEL) >= 0x010201" $sdl_cflags && - check_cpp_condition SDL.h "(SDL_MAJOR_VERSION<<16 | SDL_MINOR_VERSION<<8 | SDL_PATCHLEVEL) < 0x010300" $sdl_cflags && - enable sdl - fi fi -enabled sdl && add_cflags $sdl_cflags && add_extralibs $sdl_libs -texi2html --help 2> /dev/null | grep -q 'init-file' && enable texi2html || disable texi2html -makeinfo --version > /dev/null 2>&1 && enable makeinfo || disable makeinfo -perl -v > /dev/null 2>&1 && enable perl || disable perl pod2man --help > /dev/null 2>&1 && enable pod2man || disable pod2man -rsync --help 2> /dev/null | grep -q 'contimeout' && enable rsync_contimeout || disable rsync_contimeout +texi2html -version > /dev/null 2>&1 && enable texi2html || disable texi2html check_header linux/fb.h -check_header linux/videodev.h check_header linux/videodev2.h -check_code cc linux/videodev2.h "struct v4l2_frmsizeenum vfse; vfse.discrete.width = 0;" && enable_safe struct_v4l2_frmivalenum_discrete +check_struct linux/videodev2.h "struct v4l2_frmivalenum" discrete check_header sys/videoio.h @@ -4777,8 +4123,6 @@ check_func_headers "windows.h vfw.h" capCreateCaptureWindow "$vfwcap_indev_extra # w32api 3.12 had it defined wrong check_cpp_condition vfw.h "WM_CAP_DRIVER_CONNECT > WM_USER" && enable vfwcap_defines -check_type "dshow.h" IBaseFilter - # check for ioctl_meteor.h, ioctl_bt848.h and alternatives { check_header dev/bktr/ioctl_meteor.h && check_header dev/bktr/ioctl_bt848.h; } || @@ -4789,66 +4133,38 @@ check_type "dshow.h" IBaseFilter check_header dev/ic/bt8xx.h check_header sndio.h -if check_struct sys/soundcard.h audio_buf_info bytes; then - enable_safe sys/soundcard.h -else - check_cc -D__BSD_VISIBLE -D__XSI_VISIBLE < - audio_buf_info abc; -EOF -fi +check_header sys/soundcard.h check_header soundcard.h enabled_any alsa_indev alsa_outdev && check_lib2 alsa/asoundlib.h snd_pcm_htimestamp -lasound -enabled jack_indev && check_lib2 jack/jack.h jack_client_open -ljack && check_func sem_timedwait && +enabled jack_indev && check_lib2 jack/jack.h jack_client_open -ljack && check_func jack_port_get_latency_range -ljack enabled_any sndio_indev sndio_outdev && check_lib2 sndio.h sio_open -lsndio if enabled libcdio; then check_lib2 "cdio/cdda.h cdio/paranoia.h" cdio_cddap_open -lcdio_paranoia -lcdio_cdda -lcdio || - check_lib2 "cdio/paranoia/cdda.h cdio/paranoia/paranoia.h" cdio_cddap_open -lcdio_paranoia -lcdio_cdda -lcdio || - die "ERROR: libcdio-paranoia not found" + check_lib2 "cdio/paranoia/cdda.h cdio/paranoia/paranoia.h" cdio_cddap_open -lcdio_paranoia -lcdio_cdda -lcdio fi -enabled xlib && - check_lib X11/Xlib.h XOpenDisplay -lX11 || disable xlib +check_lib X11/Xlib.h XOpenDisplay -lX11 && enable xlib enabled x11grab && require Xext X11/extensions/XShm.h XShmCreateImage -lXext && require Xfixes X11/extensions/Xfixes.h XFixesGetCursorImage -lXfixes && { enabled xlib || die "ERROR: Xlib not found"; } -check_func_headers "windows.h" CreateDIBSection "$gdigrab_indev_extralibs" - -enabled dxva2api_h && - check_cc < -#include -#include -int main(void) { IDirectXVideoDecoder *o = NULL; IDirectXVideoDecoder_Release(o); return 0; } -EOF - -enabled vaapi && - check_lib va/va.h vaInitialize -lva || - disable vaapi - enabled vdpau && check_cpp_condition vdpau/vdpau.h "defined VDP_DECODER_PROFILE_MPEG4_PART2_ASP" || disable vdpau enabled vdpau && enabled xlib && check_func_headers "vdpau/vdpau.h vdpau/vdpau_x11.h" vdp_device_create_x11 -lvdpau && - prepend ffmpeg_libs $($ldflags_filter "-lvdpau") && + prepend avconv_libs $($ldflags_filter "-lvdpau") && enable vdpau_x11 -# Funny iconv installations are not unusual, so check it after all flags have been set -disabled iconv || check_func_headers iconv.h iconv || check_lib2 iconv.h iconv -liconv || disable iconv - enabled debug && add_cflags -g"$debuglevel" && add_asflags -g"$debuglevel" # add some useful compiler flags if supported @@ -4857,13 +4173,12 @@ check_cflags -Wall check_cflags -Wdisabled-optimization check_cflags -Wpointer-arith check_cflags -Wredundant-decls +check_cflags -Wcast-qual check_cflags -Wwrite-strings check_cflags -Wtype-limits check_cflags -Wundef check_cflags -Wmissing-prototypes -check_cflags -Wno-pointer-to-int-cast check_cflags -Wstrict-prototypes -check_cflags -Wempty-body enabled extra_warnings && check_cflags -Winline check_disable_warning(){ @@ -4878,13 +4193,12 @@ check_disable_warning -Wno-pointer-sign # add some linker flags check_ldflags -Wl,--warn-common -check_ldflags -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample -enabled rpath && add_ldflags -Wl,-rpath,$libdir +check_ldflags -Wl,-rpath-link=libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic # add some strip flags # -wN '..@*' is more selective than -x, but not available everywhere. -check_stripflags -wN \'..@*\' || check_stripflags -x +check_stripflags -wN \'..@*\' || check_stripflags -x || strip='true' enabled neon_clobber_test && check_ldflags -Wl,--wrap,avcodec_open2 \ @@ -4894,7 +4208,6 @@ enabled neon_clobber_test && -Wl,--wrap,avcodec_encode_audio2 \ -Wl,--wrap,avcodec_encode_video2 \ -Wl,--wrap,avcodec_encode_subtitle \ - -Wl,--wrap,swr_convert \ -Wl,--wrap,avresample_convert || disable neon_clobber_test @@ -4904,10 +4217,8 @@ enabled xmm_clobber_test && -Wl,--wrap,avcodec_decode_video2 \ -Wl,--wrap,avcodec_decode_subtitle2 \ -Wl,--wrap,avcodec_encode_audio2 \ - -Wl,--wrap,avcodec_encode_video \ -Wl,--wrap,avcodec_encode_video2 \ -Wl,--wrap,avcodec_encode_subtitle \ - -Wl,--wrap,swr_convert \ -Wl,--wrap,avresample_convert \ -Wl,--wrap,sws_scale || disable xmm_clobber_test @@ -4945,39 +4256,28 @@ if enabled lto; then test "$cc_type" != "$ld_type" && die "LTO requires same compiler and linker" check_cflags -flto check_ldflags -flto $cpuflags - disable inline_asm_direct_symbol_refs fi check_optflags $optflags check_optflags -fno-math-errno check_optflags -fno-signed-zeros -enabled ftrapv && check_cflags -ftrapv - -check_cc -mno-red-zone <= 1400" && add_cflags -Qansi-alias - # Some inline asm is not compilable in debug - if enabled debug; then - disable ebp_available - disable ebx_available - fi + # icl will pass the inline asm tests but inline asm is currently + # not supported (build will fail) + disable inline_asm fi - # msvcrt10 x64 incorrectly enables log2, only msvcrt12 onwards actually has log2. - check_cpp_condition crtversion.h "_VC_CRT_MAJOR_VERSION >= 12" || disable log2 fi case $as_type in @@ -5064,8 +4361,6 @@ case $target_os in ;; esac -enable frame_thread_encoder - enabled asm || { arch=c; disable $ARCH_LIST $ARCH_EXT_LIST; } check_deps $CONFIG_LIST \ @@ -5073,48 +4368,13 @@ check_deps $CONFIG_LIST \ $HAVE_LIST \ $ALL_COMPONENTS \ -enabled threads && ! enabled pthreads && ! enabled atomics_native && die "non pthread threading without atomics not supported, try adding --enable-pthreads or --cpu=i486 or higher if you are on x86" - - -if test $target_os = "haiku"; then - disable memalign - disable posix_memalign -fi - -enabled_all dxva2 dxva2api_cobj CoTaskMemFree && - prepend ffmpeg_libs $($ldflags_filter "-lole32") && +enabled_all dxva2 CoTaskMemFree && + prepend avconv_libs $($ldflags_filter "-lole32") && enable dxva2_lib ! enabled_any memalign posix_memalign aligned_malloc && enabled_any $need_memalign && enable memalign_hack -# add_dep lib dep -# -> enable ${lib}_deps_${dep} -# -> add $dep to ${lib}_deps only once -add_dep() { - lib=$1 - dep=$2 - enabled "${lib}_deps_${dep}" && return 0 - enable "${lib}_deps_${dep}" - prepend "${lib}_deps" $dep -} - -# merge deps lib components -# merge all ${component}_deps into ${lib}_deps and ${lib}_deps_* -merge_deps() { - lib=$1 - shift - for comp in $*; do - enabled $comp || continue - eval "dep=\"\$${comp}_deps\"" - for d in $dep; do - add_dep $lib $d - done - done -} - -merge_deps libavfilter $FILTER_LIST - for thread in $THREADS_LIST; do if enabled $thread; then test -n "$thread_type" && @@ -5124,31 +4384,11 @@ for thread in $THREADS_LIST; do done # conditional library dependencies, in linking order -enabled aconvert_filter && prepend avfilter_deps "swresample" -enabled amovie_filter && prepend avfilter_deps "avformat avcodec" -enabled aresample_filter && prepend avfilter_deps "swresample" -enabled asyncts_filter && prepend avfilter_deps "avresample" -enabled atempo_filter && prepend avfilter_deps "avcodec" -enabled decimate_filter && prepend avfilter_deps "avcodec" -enabled deshake_filter && prepend avfilter_deps "avcodec" -enabled ebur128_filter && enabled swresample && prepend avfilter_deps "swresample" -enabled elbg_filter && prepend avfilter_deps "avcodec" -enabled mcdeint_filter && prepend avfilter_deps "avcodec" enabled movie_filter && prepend avfilter_deps "avformat avcodec" -enabled mp_filter && prepend avfilter_deps "avcodec" -enabled pan_filter && prepend avfilter_deps "swresample" -enabled pp_filter && prepend avfilter_deps "postproc" -enabled removelogo_filter && prepend avfilter_deps "avformat avcodec swscale" enabled resample_filter && prepend avfilter_deps "avresample" -enabled sab_filter && prepend avfilter_deps "swscale" enabled scale_filter && prepend avfilter_deps "swscale" -enabled showspectrum_filter && prepend avfilter_deps "avcodec" -enabled smartblur_filter && prepend avfilter_deps "swscale" -enabled subtitles_filter && prepend avfilter_deps "avformat avcodec" -enabled lavfi_indev && prepend avdevice_deps "avfilter" - -enabled opus_decoder && prepend avcodec_deps "swresample" +enabled opus_decoder && prepend avcodec_deps "avresample" expand_deps(){ lib_deps=${1}_deps @@ -5157,9 +4397,6 @@ expand_deps(){ unique $lib_deps } -#we have to remove gpl from the deps here as some code assumes all lib deps are libs -postproc_deps="$(filter_out 'gpl' $postproc_deps)" - map 'expand_deps $v' $LIBRARY_LIST echo "install prefix $prefix" @@ -5174,9 +4411,6 @@ echo "ARCH $arch ($cpu)" if test "$build_suffix" != ""; then echo "build suffix $build_suffix" fi -if test "$progs_suffix" != ""; then - echo "progs suffix $progs_suffix" -fi if test "$extra_version" != ""; then echo "version string suffix $extra_version" fi @@ -5209,13 +4443,6 @@ if enabled arm; then echo "ARMv6T2 enabled ${armv6t2-no}" echo "VFP enabled ${vfp-no}" echo "NEON enabled ${neon-no}" - echo "THUMB enabled ${thumb-no}" -fi -if enabled mips; then - echo "MIPS FPU enabled ${mipsfpu-no}" - echo "MIPS32R2 enabled ${mips32r2-no}" - echo "MIPS DSP R1 enabled ${mipsdspr1-no}" - echo "MIPS DSP R2 enabled ${mipsdspr2-no}" fi if enabled ppc; then echo "AltiVec enabled ${altivec-no}" @@ -5223,23 +4450,15 @@ if enabled ppc; then echo "dcbzl available ${dcbzl-no}" fi echo "debug symbols ${debug-no}" -echo "strip symbols ${stripping-no}" echo "optimize for size ${small-no}" echo "optimizations ${optimizations-no}" echo "static ${static-no}" echo "shared ${shared-no}" -echo "postprocessing support ${postproc-no}" echo "new filter support ${avfilter-no}" echo "network support ${network-no}" echo "threading support ${thread_type-no}" echo "safe bitstream reader ${safe_bitstream_reader-no}" echo "SDL support ${sdl-no}" -echo "opencl enabled ${opencl-no}" -echo "libzvbi enabled ${libzvbi-no}" -echo "texi2html enabled ${texi2html-no}" -echo "perl enabled ${perl-no}" -echo "pod2man enabled ${pod2man-no}" -echo "makeinfo enabled ${makeinfo-no}" test -n "$random_seed" && echo "random seed ${random_seed}" echo @@ -5268,19 +4487,15 @@ fi echo "License: $license" -echo "Creating config.mak, config.h, and doc/config.texi..." +echo "Creating config.mak and config.h..." test -e Makefile || echo "include $source_path/Makefile" > Makefile -enabled stripping || strip="echo skipping strip" - -config_files="$TMPH config.mak doc/config.texi" +config_files="$TMPH config.mak" cat > config.mak <> config.mak eval echo "${lcname}_VERSION_MAJOR=\$${name}_VERSION_MAJOR" >> config.mak @@ -5414,25 +4615,18 @@ map 'print_program_libs $v' $PROGRAM_LIST cat > $TMPH <>$TMPH - test -n "$malloc_prefix" && echo "#define MALLOC_PREFIX $malloc_prefix" >>$TMPH @@ -5443,18 +4637,13 @@ fi enabled getenv || echo "#define getenv(x) NULL" >> $TMPH - -mkdir -p doc -echo "@c auto-generated by configure" > doc/config.texi - print_config ARCH_ "$config_files" $ARCH_LIST print_config HAVE_ "$config_files" $HAVE_LIST print_config CONFIG_ "$config_files" $CONFIG_LIST \ $CONFIG_EXTRA \ $ALL_COMPONENTS \ -echo "#endif /* FFMPEG_CONFIG_H */" >> $TMPH -echo "endif # FFMPEG_CONFIG_MAK" >> config.mak +echo "#endif /* LIBAV_CONFIG_H */" >> $TMPH # Do not overwrite an unchanged config.h to avoid superfluous rebuilds. cp_if_changed $TMPH config.h @@ -5474,15 +4663,12 @@ echo "#endif /* AVUTIL_AVCONFIG_H */" >> $TMPH cp_if_changed $TMPH libavutil/avconfig.h -if test -n "$WARNINGS"; then - printf "\n$WARNINGS" - enabled fatal_warnings && exit 1 -fi +test -n "$WARNINGS" && printf "\n$WARNINGS" # build pkg-config files lib_version(){ - eval printf "\"lib${1}${build_suffix} >= \$LIB$(toupper ${1})_VERSION, \"" + eval printf "\"lib${1} >= \$LIB$(toupper ${1})_VERSION, \"" } pkgconfig_generate(){ @@ -5495,7 +4681,7 @@ pkgconfig_generate(){ requires=${requires%, } enabled ${name#lib} || return 0 mkdir -p $name - cat < $name/$name${build_suffix}.pc + cat < $name/$name.pc prefix=$prefix exec_prefix=\${prefix} libdir=$libdir @@ -5507,36 +4693,30 @@ Version: $version Requires: $(enabled shared || echo $requires) Requires.private: $(enabled shared && echo $requires) Conflicts: -Libs: -L\${libdir} $(enabled rpath && echo "-Wl,-rpath,\${libdir}") -l${shortname} $(enabled shared || echo $libs) +Libs: -L\${libdir} -l${shortname} $(enabled shared || echo $libs) Libs.private: $(enabled shared && echo $libs) Cflags: -I\${includedir} EOF - -mkdir -p doc/examples/pc-uninstalled -includedir=${source_path} -[ "$includedir" = . ] && includedir="\${pcfiledir}/../../.." - cat < doc/examples/pc-uninstalled/$name.pc + cat < $name/$name-uninstalled.pc prefix= exec_prefix= -libdir=\${pcfiledir}/../../../$name -includedir=${includedir} +libdir=\${pcfiledir} +includedir=${source_path} Name: $name Description: $comment Version: $version Requires: $requires Conflicts: -Libs: -L\${libdir} -Wl,-rpath,\${libdir} -l${shortname} $(enabled shared || echo $libs) +Libs: \${libdir}/${LIBPREF}${shortname}${LIBSUF} $libs Cflags: -I\${includedir} EOF } -pkgconfig_generate libavutil "FFmpeg utility library" "$LIBAVUTIL_VERSION" "$LIBM" -pkgconfig_generate libavcodec "FFmpeg codec library" "$LIBAVCODEC_VERSION" "$extralibs" -pkgconfig_generate libavformat "FFmpeg container format library" "$LIBAVFORMAT_VERSION" "$extralibs" -pkgconfig_generate libavdevice "FFmpeg device handling library" "$LIBAVDEVICE_VERSION" "$extralibs" -pkgconfig_generate libavfilter "FFmpeg audio/video filtering library" "$LIBAVFILTER_VERSION" "$extralibs" -pkgconfig_generate libpostproc "FFmpeg postprocessing library" "$LIBPOSTPROC_VERSION" "" -pkgconfig_generate libavresample "Libav audio resampling library" "$LIBAVRESAMPLE_VERSION" "$extralibs" -pkgconfig_generate libswscale "FFmpeg image rescaling library" "$LIBSWSCALE_VERSION" "$LIBM" -pkgconfig_generate libswresample "FFmpeg audio resampling library" "$LIBSWRESAMPLE_VERSION" "$LIBM" +pkgconfig_generate libavutil "Libav utility library" "$LIBAVUTIL_VERSION" "$LIBM" +pkgconfig_generate libavcodec "Libav codec library" "$LIBAVCODEC_VERSION" "$extralibs" +pkgconfig_generate libavformat "Libav container format library" "$LIBAVFORMAT_VERSION" "$extralibs" +pkgconfig_generate libavdevice "Libav device handling library" "$LIBAVDEVICE_VERSION" "$extralibs" +pkgconfig_generate libavfilter "Libav video filtering library" "$LIBAVFILTER_VERSION" "$extralibs" +pkgconfig_generate libavresample "Libav audio resampling library" "$LIBAVRESAMPLE_VERSION" "$extralibs" +pkgconfig_generate libswscale "Libav image rescaling library" "$LIBSWSCALE_VERSION" "$LIBM" diff --git a/doc/APIchanges b/doc/APIchanges index 395083c091..9bc3f1acdc 100644 --- a/doc/APIchanges +++ b/doc/APIchanges @@ -1,5 +1,5 @@ Never assume the API of libav* to be stable unless at least 1 month has passed -since the last major version increase or the API was added. +since the last major version increase. The last version increases were: libavcodec: 2013-03-xx @@ -7,71 +7,49 @@ libavdevice: 2013-03-xx libavfilter: 2013-12-xx libavformat: 2013-03-xx libavresample: 2012-10-05 -libpostproc: 2011-04-18 -libswresample: 2011-09-19 libswscale: 2011-06-20 -libavutil: 2012-10-22 +libavutil: 2013-12-xx API changes, most recent first: -2014-05-26 - xxxxxxx - lavu 52.87.100 - threadmessage.h - Add thread message queue API. - -2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h - Add format_probesize to AVFormatContext. - 2014-05-19 - xxxxxxx - lavf 55.18.0 - avformat.h Add av_stream_get_side_data() to access stream-level side data in the same way as av_packet_get_side_data(). -2014-05-xx - xxxxxxx - lavu 52.86.100 - fifo.h - Add av_fifo_alloc_array() function. - 2014-05-xx - xxxxxxx - lavu 53.15.0 - frame.h, display.h Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level spatial rendering on video frames for proper display. -2014-05-xx - xxxxxxx - lavc 55.64.100 / 55.53.0 - avcodec.h +2014-05-xx - xxxxxxx - lavc 55.53.0 - avcodec.h Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level spatial rendering on video frames for proper display. 2014-05-xx - xxxxxxx - lavf 55.17.1 - avformat.h Deprecate AVStream.pts and the AVFrac struct, which was its only use case. - See use av_stream_get_end_pts() + Those fields were poorly defined and not meant to be public, so there is + no replacement for them. 2014-05-18 - fd05602 - lavc 55.52.0 - avcodec.h Add avcodec_free_context(). From now on it should be used for freeing AVCodecContext. -2014-05-xx - xxxxxxx - lavu 52.84.100 - time.h - Add av_gettime_relative() av_gettime_relative_is_monotonic() - 2014-05-15 - 0c1959b - lavf 55.17.0 - avformat.h Add AVMFT_FLAG_BITEXACT flag. Muxers now use it instead of checking CODEC_FLAG_BITEXACT on the first stream. -2014-05-15 - xxxxxxx - lswr 0.19.100 - swresample.h - Add swr_close() - 2014-05-11 - 66e6c8a - lavu 53.14.0 - pixfmt.h Add AV_PIX_FMT_VDA for new-style VDA acceleration. -2014-05-xx - xxxxxxx - lavu 52.82.0 - fifo.h - Add av_fifo_freep() function. - -2014-05-02 - ba52fb11 - lavu 52.81.0 - opt.h - Add av_opt_set_dict2() function. - -2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h +2014-05-01 - a2941c8 - lavc 55.50.3 - avcodec.h Deprecate CODEC_FLAG_MV0. It is replaced by the flag "mv0" in the "mpv_flags" private option of the mpegvideo encoders. -2014-05-01 - e40ae8c / 6484149 - lavc 55.60.102 / 55.50.2 - avcodec.h +2014-05-01 - 6484149 - lavc 55.50.2 - avcodec.h Deprecate CODEC_FLAG_GMC. It is replaced by the "gmc" private option of the libxvid encoder. -2014-05-01 - 1851643 / b2c3171 - lavc 55.60.101 / 55.50.1 - avcodec.h +2014-05-01 - b2c3171 - lavc 55.50.1 - avcodec.h Deprecate CODEC_FLAG_NORMALIZE_AQP. It is replaced by the flag "naq" in the "mpv_flags" private option of the mpegvideo encoders. @@ -79,528 +57,213 @@ API changes, most recent first: Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing reference-counted frames to encoders. -2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h - Add AVCodecDescriptor.mime_types field. - -2014-04-29 - xxxxxxx - lavu 52.80.0 - hash.h - Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64(). - -2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h +2014-04-28 - ed4b757 - lavc 55.50.0 - dxva2.h Add FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO for old Intel GPUs. -2014-04-22 - 502512e /dac7e8a - lavu 53.13.0 / 52.78.100 - avutil.h +2014-04-22 - 502512e - lavu 53.13.0 - avutil.h Add av_get_time_base_q(). -2014-04-17 - 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h +2014-04-17 - 0983d48 - lavu 53.12.0 - crc.h Add AV_CRC_16_ANSI_LE crc variant. -2014-04-XX - xxxxxxx - lavf xx.xx.1xx - avformat.h - Add av_format_inject_global_side_data() - -2014-04-12 - 4f698be - lavu 52.76.100 - log.h - Add av_log_get_flags() - -2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h - Add avdevice_capabilities_create() function. - Add avdevice_capabilities_free() function. - -2014-04-07 - 0a1cc04 / 8b17243 - lavu 52.75.100 / 53.11.0 - pixfmt.h +2014-04-07 - 8b17243 - lavu 53.11.0 - pixfmt.h Add AV_PIX_FMT_YVYU422 pixel format. -2014-04-04 - c1d0536 / 8542f9c - lavu 52.74.100 / 53.10.0 - replaygain.h +2014-04-04 - 8542f9c - lavu 53.10.0 - replaygain.h Full scale for peak values is now 100000 (instead of UINT32_MAX) and values may overflow. -2014-04-03 - c16e006 / 7763118 - lavu 52.73.100 / 53.9.0 - log.h +2014-04-03 - 7763118 - lavu 53.09.0 - log.h Add AV_LOG(c) macro to have 256 color debug messages. -2014-04-03 - eaed4da9 - lavu 52.72.100 - opt.h - Add AV_OPT_MULTI_COMPONENT_RANGE define to allow return - multi-component option ranges. - -2014-03-29 - cd50a44b - lavu 52.70.100 - mem.h - Add av_dynarray_add_nofree() function. - -2014-02-24 - 3e1f241 / d161ae0 - lavu 52.69.100 / 53.8.0 - frame.h +2014-03-24 - d161ae0 - lavu 53.08.0 - frame.h Add av_frame_remove_side_data() for removing a single side data instance from a frame. -2014-03-24 - 83e8978 / 5a7e35d - lavu 52.68.100 / 53.7.0 - frame.h, replaygain.h +2014-03-24 - 5a7e35d - lavu 53.07.0 - frame.h, replaygain.h Add AV_FRAME_DATA_REPLAYGAIN for exporting replaygain tags. Add a new header replaygain.h with the AVReplayGain struct. -2014-03-24 - 83e8978 / 5a7e35d - lavc 55.54.100 / 55.36.0 - avcodec.h +2014-03-24 - 5a7e35d - lavc 55.36.0 - avcodec.h Add AV_PKT_DATA_REPLAYGAIN for exporting replaygain tags. -2014-03-24 - 595ba3b / 25b3258 - lavf 55.35.100 / 55.13.0 - avformat.h +2014-03-24 - 25b3258 - lavf 55.13.0 - avformat.h Add AVStream.side_data and AVStream.nb_side_data for exporting stream-global side data (e.g. replaygain tags, video rotation) -2014-03-24 - bd34e26 / 0e2c3ee - lavc 55.53.100 / 55.35.0 - avcodec.h +2014-03-24 - 0e2c3ee - lavc 55.35.0 - avcodec.h Give the name AVPacketSideData to the previously anonymous struct used for AVPacket.side_data. -2014-03-18 - 37c07d4 - lsws 2.5.102 - Make gray16 full-scale. +2014-03-16 - 1481d24 - lavu 53.06.0 - pixfmt.h + Add RGBA64 pixel format and variants. -2014-03-16 - 6b1ca17 / 1481d24 - lavu 52.67.100 / 53.6.0 - pixfmt.h - Add RGBA64_LIBAV pixel format and variants for compatibility - -2014-03-11 - 3f3229c - lavf 55.34.101 - avformat.h - Set AVFormatContext.start_time_realtime when demuxing. - -2014-03-03 - 06fed440 - lavd 55.11.100 - avdevice.h - Add av_input_audio_device_next(). - Add av_input_video_device_next(). - Add av_output_audio_device_next(). - Add av_output_video_device_next(). - -2014-02-24 - fff5262 / 1155fd0 - lavu 52.66.100 / 53.5.0 - frame.h +2014-02-24 - 1155fd0 - lavu 53.05.0 - frame.h Add av_frame_copy() for copying the frame data. -2014-02-24 - a66be60 - lswr 0.18.100 - swresample.h - Add swr_is_initialized() for checking whether a resample context is initialized. - -2014-02-22 - 5367c0b / 7e86c27 - lavr 1.2.0 - avresample.h +2014-02-22 - 7e86c27 - lavr 1.2.0 - avresample.h Add avresample_is_open() for checking whether a resample context is open. -2014-02-19 - 6a24d77 / c3ecd96 - lavu 52.65.100 / 53.4.0 - opt.h +2014-02-19 - c3ecd96 - lavu 53.04.0 - opt.h Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only) for reading. -2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h +2014-02-19 - 6bb8720 - lavu 53.03.01 - opt.h Deprecate unused AV_OPT_FLAG_METADATA. -2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h - Add avdevice_list_devices() and avdevice_free_list_devices() +------------------------------8<------------------------------------- + 10 branch was cut here +----------------------------->8-------------------------------------- -2014-02-16 - db3c970 - lavf 55.33.100 - avio.h - Add avio_find_protocol_name() to find out the name of the protocol that would - be selected for a given URL. - -2014-02-15 - a2bc6c1 / c98f316 - lavu 52.64.100 / 53.3.0 - frame.h +2014-02-15 - c98f316 - lavu 53.3.0 - frame.h Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and downmix_info.h API, which identify downmix-related metadata. -2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h - Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame(). - -2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h +2014-02-04 - d9ae103 - lavf 55.11.0 - avformat.h Add AVFormatContext.max_interleave_delta for controlling amount of buffering when interleaving. -2014-02-02 - 5871ee5 - lavf 55.29.100 - avformat.h - Add output_ts_offset muxing option to AVFormatContext. - -2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h - lavf 55.28.100 - avformat.h - Add avdevice_dev_to_app_control_message() function. - -2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h - lavf 55.27.100 - avformat.h - Add avdevice_app_to_dev_control_message() function. - -2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h - Add AVFormatContext option metadata_header_padding to allow control over the - amount of padding added. - -2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h +2014-01-20 - 93c553c - lavc 55.32.1 - avcodec.h Edges are not required anymore on video buffers allocated by get_buffer2() (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width(). -2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h - Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags(). - -2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h - Add av_make_q() function. - -2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h +2014-01-05 - 5b4797a - lavu 53.2.0 - frame.h Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which identifies AVMatrixEncoding data. -2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h +2014-01-05 - 5c437fb - lavu 53.1.0 - channel_layout.h Add values for various Dolby flags to the AVMatrixEncoding enum. -2014-01-04 - b317f94 - lavu 52.60.100 - mathematics.h - Add av_add_stable() function. +2013-12-20 - 2a41826 - lavc 55.30.0 - avcodec.h + Add HEVC profiles -2013-12-22 - 911676c - lavu 52.59.100 - avstring.h - Add av_strnlen() function. - -2013-12-09 - 64f73ac - lavu 52.57.100 - opencl.h - Add av_opencl_benchmark() function. - -2013-11-30 - 82b2e9c - lavu 52.56.100 - ffversion.h - Moves version.h to libavutil/ffversion.h. - Install ffversion.h and make it public. - -2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h +2013-12-11 - b9fb59d,9431356,d7b3ee9 - lavc 55.28.1 - avcodec.h av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and avcodec_free_frame() respectively. The latter three functions are deprecated. -2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h +2013-12-09 - 7e244c6- - lavu 52.20.0 - frame.h Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and stereo3d.h API, that identify codec-independent stereo3d information. -2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h +2013-11-26 - 1eaac1d- - lavu 52.19.0 - frame.h Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which identifies ATSC A53 Part 4 Closed Captions data. -2013-11-22 - 6859065 - lavu 52.54.100 - avstring.h - Add av_utf8_decode() function. - -2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h - Add HEVC profiles - -2013-11-20 - c28b61c - lavc 55.44.100 - avcodec.h - Add av_packet_{un,}pack_dictionary() - Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value - strings between a stream and the application. - -2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h +2013-11-14 - cce3e0a - lavu 52.18.0 - mem.h Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil. -2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h +2013-11-14 - 8941971 - lavc 55.27.0 - avcodec.h Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate' private option of the mpegvideo encoder family. -2013-11-14 - 31c09b7 / 728c465 - lavc 55.42.100 / 55.26.0 - vdpau.h +2013-11-14 - 728c465 - lavc 55.26.0 - vdpau.h Add av_vdpau_get_profile(). Add av_vdpau_alloc_context(). This function must from now on be used for allocating AVVDPAUContext. -2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h - lavu 52.51.100 - frame.h +2013-11-04 - cd8f772 - lavc 55.25.0 - avcodec.h Add ITU-R BT.2020 and other not yet included values to color primaries, transfer characteristics and colorspaces. -2013-11-04 - 85cabf1 - lavu 52.50.100 - avutil.h - Add av_fopen_utf8() - -2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h +2013-10-31 - 28096e0 - lavu 52.17.0 - frame.h Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT. -2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h - Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2. - -2013-10-27 - d61617a - lavu 52.48.100 - parseutils.h - Add av_get_known_color_name(). - -2013-10-17 - 8696e51 - lavu 52.47.100 - opt.h - Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers - av_opt_get_channel_layout() and av_opt_set_channel_layout(). - -2013-10-06 - ccf96f8 -libswscale 2.5.101 - options.c - Change default scaler to bicubic - -2013-10-03 - e57dba0 - lavc 55.34.100 - avcodec.h - Add av_codec_get_max_lowres() - -2013-10-02 - 5082fcc - lavf 55.19.100 - avformat.h - Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific - decoders - -2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h +2013-09-28 - 0767bfd - lavfi 3.11.0 - avfilter.h Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading implementations. -2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h +2013-09-21 - e208e6d - lavu 52.16.0 - pixfmt.h Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and AV_PIX_FMT_NV20. -2013-09-16 - c74c3fb / 3feb3d6 - lavu 52.44.100 / 52.15.0 - mem.h +2013-09-16 - 3feb3d6 - lavu 52.15.0 - mem.h Add av_reallocp. -2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h - avcodec_close() argument can be NULL. - -2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h - avformat_close_input() argument can be NULL and point on NULL. - -2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h - Add av_format_get_probe_score(). - -2013-08-15 - 1e0e193 - lsws 2.5.100 - - Add a sws_dither AVOption, allowing to set the dither algorithm used - -2013-08-11 - d404fe35 - lavc 55.27.100 - vdpau.h - Add a render2 alternative to the render callback function. - -2013-08-11 - af05edc - lavc 55.26.100 - vdpau.h - Add allocation function for AVVDPAUContext, allowing - to extend it in the future without breaking ABI/API. - -2013-08-10 - 67a580f / 5a9a9d4 - lavc 55.25.100 / 55.16.0 - avcodec.h +2013-08-10 - 5a9a9d4 - lavc 55.16.0 - avcodec.h Extend AVPacket API with av_packet_unref, av_packet_ref, av_packet_move_ref, av_packet_copy_props, av_packet_free_side_data. -2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h +2013-08-05 - f824535 - lavc 55.13.0 - avcodec.h Deprecate the bitstream-related members from struct AVVDPAUContext. - The bitstream buffers no longer need to be explicitly freed. + The bistream buffers no longer need to be explicitly freed. -2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h +2013-08-05 - 549294f - lavc 55.12.0 - avcodec.h Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL and select the AV_PIX_FMT_VDPAU format with get_format() instead. -2013-08-05 - 4ee0984 / a0ad5d0 - lavu 52.41.100 / 52.14.0 - pixfmt.h +2013-08-05 - a0ad5d0 - lavu 52.14.0 - pixfmt.h Deprecate AV_PIX_FMT_VDPAU_*. Use AV_PIX_FMT_VDPAU instead. -2013-08-02 - 82fdfe8 / a8b1927 - lavc 55.20.100 / 55.11.0 - avcodec.h +2013-08-02 - a8b1927 - lavc 55.11.0 - avcodec.h Add output_picture_number to AVCodecParserContext. -2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h - Add avcodec_chroma_pos_to_enum() - Add avcodec_enum_to_chroma_pos() - -2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h - Deprecate avfilter_graph_parse() in favor of the equivalent - avfilter_graph_parse_ptr(). - -2013-06-24 - af5f9c0 / 95d5246 - lavc 55.17.100 / 55.10.0 - avcodec.h +2013-06-24 - 95d5246 - lavc 55.10.0 - avcodec.h Add MPEG-2 AAC profiles -2013-06-25 - af5f9c0 / 95d5246 - lavf 55.10.100 - avformat.h - Add AV_DISPOSITION_* flags to indicate text track kind. - -2013-06-15 - 99b8cd0 - lavu 52.36.100 - Add AVRIPEMD: - av_ripemd_alloc() - av_ripemd_init() - av_ripemd_update() - av_ripemd_final() - -2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h +2013-06-04 - fc962d4 - lavu 52.13.0 - mem.h Add av_realloc_array and av_reallocp_array -2013-05-30 - 682b227 - lavu 52.35.100 - Add AVSHA512: - av_sha512_alloc() - av_sha512_init() - av_sha512_update() - av_sha512_final() - -2013-05-24 - 8d4e969 / 129bb23 - lavfi 3.10.0 / 3.70.100 - avfilter.h +2013-05-24 - 129bb23 - lavfi 3.10.0 - avfilter.h Add support for slice multithreading to lavfi. Filters supporting threading are marked with AVFILTER_FLAG_SLICE_THREADS. New fields AVFilterContext.thread_type, AVFilterGraph.thread_type and AVFilterGraph.nb_threads (accessible directly or through AVOptions) may be used to configure multithreading. -2013-05-24 - fe40a9f / 2a6eaea - lavu 52.12.0 / 52.34.100 - cpu.h +2013-05-24 - 2a6eaea - lavu 52.12.0 - cpu.h Add av_cpu_count() function for getting the number of logical CPUs. -2013-05-24 - 0c25c39 / b493847 - lavc 55.7.0 / 55.12.100 - avcodec.h +2013-05-24 - b493847 - lavc 55.7.0 - avcodec.h Add picture_structure to AVCodecParserContext. -2013-05-17 - 3a751ea - lavu 52.33.100 - opt.h - Add AV_OPT_TYPE_COLOR value to AVOptionType enum. - -2013-05-13 - e398416 - lavu 52.31.100 - mem.h - Add av_dynarray2_add(). - -2013-05-12 - 1776177 - lavfi 3.65.100 - Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags. - -2013-04-19 - 380cfce - lavc 55.4.100 - Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec. - -2013-04-18 - 7c1a002 - lavf 55.3.100 - The matroska demuxer can now output proper verbatim ASS packets. It will - become the default starting lavf 56.0.100. - -2013-04-10 - af0d270 - lavu 25.26.100 - avutil.h,opt.h - Add av_int_list_length() - and av_opt_set_int_list(). - -2013-03-30 - 5c73645 - lavu 52.24.100 - samplefmt.h - Add av_samples_alloc_array_and_samples(). - -2013-03-29 - ef7b6b4 - lavf 55.1.100 - avformat.h - Add av_guess_frame_rate() - -2013-03-20 - 8d928a9 - lavu 52.22.100 - opt.h - Add AV_OPT_TYPE_DURATION value to AVOptionType enum. - -2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h - Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum. - -2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h - Add av_escape() and av_bprint_escape() API. - -2013-02-24 - b59cd08 - lavfi 3.41.100 - buffersink.h - Add sample_rates field to AVABufferSinkParams. - -2013-01-17 - a1a707f - lavf 54.61.100 - Add av_codec_get_tag2(). - -2013-01-01 - 2eb2e17 - lavfi 3.34.100 - Add avfilter_get_audio_buffer_ref_from_arrays_channels. - -2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h - Add AVFilterLink.channels, avfilter_link_get_channels() - and avfilter_ref_get_channels(). - -2012-12-15 - 96d815fc - lavc 54.80.100 - avcodec.h - Add pkt_size field to AVFrame. - -2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h - Add the following convenience functions to opt.h: - av_opt_get_image_size - av_opt_get_pixel_fmt - av_opt_get_sample_fmt - av_opt_set_image_size - av_opt_set_pixel_fmt - av_opt_set_sample_fmt - -2012-11-17 - 4cd74c81 - lavu 52.8.100 - bprint.h - Add av_bprint_strftime(). - -2012-11-15 - 92648107 - lavu 52.7.100 - opt.h - Add av_opt_get_key_value(). - -2012-11-13 - 79456652 - lavfi 3.23.100 - avfilter.h - Add channels field to AVFilterBufferRefAudioProps. - -2012-11-03 - 481fdeee - lavu 52.3.100 - opt.h - Add AV_OPT_TYPE_SAMPLE_FMT value to AVOptionType enum. - -2012-10-21 - 6fb2fd8 - lavc 54.68.100 - avcodec.h - lavfi 3.20.100 - avfilter.h - Add AV_PKT_DATA_STRINGS_METADATA side data type, used to transmit key/value - strings between AVPacket and AVFrame, and add metadata field to - AVCodecContext (which shall not be accessed by users; see AVFrame metadata - instead). - -2012-09-27 - a70b493 - lavd 54.3.100 - version.h - Add LIBAVDEVICE_IDENT symbol. - -2012-09-27 - a70b493 - lavfi 3.18.100 - version.h - Add LIBAVFILTER_IDENT symbol. - -2012-09-27 - a70b493 - libswr 0.16.100 - version.h - Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD - and LIBSWRESAMPLE_IDENT symbols. - -2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h - Add av_small_strptime() time parsing function. - - Can be used as a stripped-down replacement for strptime(), on - systems which do not support it. - -2012-08-25 - 2626cc4 - lavf 54.28.100 - Matroska demuxer now identifies SRT subtitles as AV_CODEC_ID_SUBRIP instead - of AV_CODEC_ID_TEXT. - -2012-08-13 - 5c0d8bc - lavfi 3.8.100 - avfilter.h - Add avfilter_get_class() function, and priv_class field to AVFilter - struct. - -2012-08-12 - a25346e - lavu 51.69.100 - opt.h - Add AV_OPT_FLAG_FILTERING_PARAM symbol in opt.h. - -2012-07-31 - 23fc4dd - lavc 54.46.100 - Add channels field to AVFrame. - -2012-07-30 - f893904 - lavu 51.66.100 - Add av_get_channel_description() - and av_get_standard_channel_layout() functions. - -2012-07-21 - 016a472 - lavc 54.43.100 - Add decode_error_flags field to AVFrame. - -2012-07-20 - b062936 - lavf 54.18.100 - Add avformat_match_stream_specifier() function. - -2012-07-14 - f49ec1b - lavc 54.38.100 - avcodec.h - Add metadata to AVFrame, and the accessor functions - av_frame_get_metadata() and av_frame_set_metadata(). - -2012-07-10 - 0e003d8 - lavc 54.33.100 - Add av_fast_padded_mallocz(). - -2012-07-10 - 21d5609 - lavfi 3.2.0 - avfilter.h - Add init_opaque() callback to AVFilter struct. - -2012-06-26 - e6674e4 - lavu 51.63.100 - imgutils.h - Add functions to libavutil/imgutils.h: - av_image_get_buffer_size() - av_image_fill_arrays() - av_image_copy_to_buffer() - -2012-06-24 - c41899a - lavu 51.62.100 - version.h - version moved from avutil.h to version.h - -2012-04-11 - 359abb1 - lavu 51.58.100 - error.h - Add av_make_error_string() and av_err2str() utilities to - libavutil/error.h. - -2012-06-05 - 62b39d4 - lavc 54.24.100 - Add pkt_duration field to AVFrame. - -2012-05-24 - f2ee065 - lavu 51.54.100 - Move AVPALETTE_SIZE and AVPALETTE_COUNT macros from - libavcodec/avcodec.h to libavutil/pixfmt.h. - -2012-05-14 - 94a9ac1 - lavf 54.5.100 - Add av_guess_sample_aspect_ratio() function. - -2012-04-20 - 65fa7bc - lavfi 2.70.100 - Add avfilter_unref_bufferp() to avfilter.h. - -2012-04-13 - 162e400 - lavfi 2.68.100 - Install libavfilter/asrc_abuffer.h public header. - -2012-03-26 - a67d9cf - lavfi 2.66.100 - Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions. - -2013-05-15 - ff46809 / e6c4ac7 - lavu 52.32.100 / 52.11.0 - pixdesc.h +2013-05-15 - e6c4ac7 - lavu 52.11.0 - pixdesc.h Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*. -2013-04-03 - 6fc58a8 / 507b1e4 - lavc 55.7.100 / 55.4.0 - avcodec.h +2013-04-03 - 507b1e4 - lavc 55.4.0 - avcodec.h Add field_order to AVCodecParserContext. -2013-04-19 - f4b05cd / 5e83d9a - lavc 55.5.100 / 55.2.0 - avcodec.h +2013-04-19 - 5e83d9a - lavc 55.2.0 - avcodec.h Add CODEC_FLAG_UNALIGNED to allow decoders to produce unaligned output. -2013-04-11 - lavfi 3.53.100 / 3.8.0 - 231fd44 / 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate +2013-04-11 - lavfi 3.8.0 + 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate avfilterhraph.h, user applications should include just avfilter.h - 86070b8 / bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and + bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and avfilter_graph_add_filter(). - 4fde705 / 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the + 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the filter. - 710b0aa / 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter(). - 46de9ba / 1ba95a9 - Add avfilter_init_dict(). - 16fc24b / 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags. - f4db6bf / 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs. - 835cc0f / fa2a34c - Add avfilter_next(), deprecate av_filter_next(). + 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter(). + 1ba95a9 - Add avfilter_init_dict(). + 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags. + 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs. + fa2a34c - Add avfilter_next(), deprecate av_filter_next(). Deprecate avfilter_uninit(). -2013-04-09 - lavfi 3.51.100 / 3.7.0 - avfilter.h - 0594ef0 / b439c99 - Add AVFilter.priv_class for exporting filter options through the +2013-04-09 - lavfi 3.7.0 - avfilter.h + b439c99 - Add AVFilter.priv_class for exporting filter options through the AVOptions API in the similar way private options work in lavc and lavf. - 44d4488 / 8114c10 - Add avfilter_get_class(). + 8114c10 - Add avfilter_get_class(). Switch all filters to use AVOptions. -2013-03-19 - 17ebef2 / 2c328a9 - lavu 52.20.100 / 52.9.0 - pixdesc.h +2013-03-19 - 2c328a9 - lavu 52.9.0 - pixdesc.h Add av_pix_fmt_count_planes() function for counting planes in a pixel format. -2013-03-16 - ecade98 / 42c7c61 - lavfi 3.47.100 / 3.6.0 +2013-03-16 - 42c7c61 - lavfi 3.6.0 Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count. -2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.100 / 55.0.0, lavf 55.0.100 / 55.0.0, -lavd 54.4.100 / 54.0.0, lavfi 3.5.0 - 36099df / 8e401db, 532f31a / 1cec062 - add a new API for reference counted buffers and buffer +2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.0, lavf 55.0.0, +lavd 54.0.0, lavfi 3.5.0 + 8e401db, 1cec062 - add a new API for reference counted buffers and buffer pools (new header libavutil/buffer.h). - 2653e12 / 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data. + 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data. Add av_packet_from_data() function for constructing packets from av_malloc()ed data. - c4e8821 / 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add + 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add AVFrame.buf/extended_buf to allow reference counting for the AVFrame data. Add new API for working with reference-counted AVFrames. - 80e9e63 / 759001c - add the refcounted_frames field to AVCodecContext to make audio and + 759001c - add the refcounted_frames field to AVCodecContext to make audio and video decoders return reference-counted frames. Add get_buffer2() callback to AVCodecContext which allocates reference-counted frames. Add avcodec_default_get_buffer2() as the default get_buffer2() @@ -618,63 +281,68 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 * qscale_table, qstride, qscale_type, mbskip_table, motion_val, mb_type, dct_coeff, ref_index -- mpegvideo-specific tables, which are not exported anymore. - a05a44e / 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add + 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add av_buffersrc_add_frame(), deprecate av_buffersrc_buffer(). Add av_buffersink_get_frame() and av_buffersink_get_samples(), deprecate av_buffersink_read() and av_buffersink_read_samples(). Deprecate AVFilterBufferRef and all functions for working with it. -2013-03-17 - 6c17ff8 / 12c5c1d - lavu 52.19.100 / 52.8.0 - avstring.h +2013-03-17 - 12c5c1d - lavu 52.8.0 - avstring.h Add av_isdigit, av_isgraph, av_isspace, av_isxdigit. -2013-02-23 - 71cf094 / 9f12235 - lavfi 3.40.100 / 3.4.0 - avfiltergraph.h +2013-02-23 - 9f12235 - lavfi 3.4.0 - avfiltergraph.h Add resample_lavr_opts to AVFilterGraph for setting libavresample options for auto-inserted resample filters. -2013-01-25 - e7e14bc / 38c1466 - lavu 52.17.100 / 52.7.0 - dict.h +2013-01-25 - 38c1466 - lavu 52.7.0 - dict.h Add av_dict_parse_string() to set multiple key/value pairs at once from a string. -2013-01-25 - 25be630 / b85a5e8 - lavu 52.16.100 / 52.6.0 - avstring.h +2013-01-25 - b85a5e8 - lavu 52.6.0 - avstring.h Add av_strnstr() -2013-01-15 - e7e0186 / 8ee288d - lavu 52.15.100 / 52.5.0 - hmac.h +2013-01-15 - 8ee288d - lavu 52.5.0 - hmac.h Add AVHMAC. -2013-01-13 - 8ee7b38 / 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h +2013-01-13 - 44e065d - lavc 54.36.0 - vdpau.h Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding. -2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h +2013-01-12 - 169fb94 - lavu 52.4.0 - pixdesc.h Add AV_PIX_FMT_VDPAU flag. -2013-01-07 - 249fca3 / 074a00d - lavr 1.1.0 +2013-01-07 - 074a00d - lavr 1.1.0 Add avresample_set_channel_mapping() for input channel reordering, duplication, and silencing. -2012-12-29 - 2ce43b3 / d8fd06c - lavu 52.13.100 / 52.3.0 - avstring.h +------------------------------8<------------------------------------- + 9 branch was cut here +----------------------------->8-------------------------------------- + +2012-12-29 - d8fd06c - lavu 52.3.0 - avstring.h Add av_basename() and av_dirname(). -2012-11-11 - 03b0787 / 5980f5d - lavu 52.6.100 / 52.2.0 - audioconvert.h +2012-11-11 - 5980f5d - lavu 52.2.0 - audioconvert.h Rename audioconvert.h to channel_layout.h. audioconvert.h is now deprecated. -2012-11-05 - 7d26be6 / dfde8a3 - lavu 52.5.100 / 52.1.0 - intmath.h +2012-11-05 - dfde8a3 - lavu 52.1.0 - intmath.h Add av_ctz() for trailing zero bit count -2012-10-21 - e3a91c5 / a893655 - lavu 51.77.100 / 51.45.0 - error.h +2012-10-21 - a893655 - lavu 51.45.0 - error.h Add AVERROR_EXPERIMENTAL -2012-10-12 - a33ed6b / d2fcb35 - lavu 51.76.100 / 51.44.0 - pixdesc.h +2012-10-12 - d2fcb35 - lavu 51.44.0 - pixdesc.h Add functions for accessing pixel format descriptors. Accessing the av_pix_fmt_descriptors array directly is now deprecated. -2012-10-11 - f391e40 / 9a92aea - lavu 51.75.100 / 51.43.0 - aes.h, md5.h, sha.h, tree.h +2012-10-11 - 9a92aea - lavu 51.43.0 - aes.h, md5.h, sha.h, tree.h Add functions for allocating the opaque contexts for the algorithms, + deprecate the context size variables. -2012-10-10 - de31814 / b522000 - lavf 54.32.100 / 54.18.0 - avio.h +2012-10-10 - b522000 - lavf 54.18.0 - avio.h Add avio_closep to complement avio_close. -2012-10-08 - ae77266 / 78071a1 - lavu 51.74.100 / 51.42.0 - pixfmt.h +2012-10-08 - 78071a1 - lavu 51.42.0 - pixfmt.h Rename PixelFormat to AVPixelFormat and all PIX_FMT_* to AV_PIX_FMT_*. To provide backwards compatibility, PixelFormat is now #defined as AVPixelFormat. @@ -682,23 +350,23 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 'PixelFormat' identifier. Such code should either #undef PixelFormat or stop using the PixelFormat name. -2012-10-05 - 55c49af / e7ba5b1 - lavr 1.0.0 - avresample.h +2012-10-05 - e7ba5b1 - lavr 1.0.0 - avresample.h Data planes parameters to avresample_convert() and avresample_read() are now uint8_t** instead of void**. Libavresample is now stable. -2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h +2012-09-24 - a42aada - lavc 54.28.0 - avcodec.h Add avcodec_free_frame(). This function must now be used for freeing an AVFrame. -2012-09-12 - e3e09f2 / 8919fee - lavu 51.73.100 / 51.41.0 - audioconvert.h +2012-09-12 - 8919fee - lavu 51.41.0 - audioconvert.h Added AV_CH_LOW_FREQUENCY_2 channel mask value. -2012-09-04 - b21b5b0 / 686a329 - lavu 51.71.100 / 51.40.0 - opt.h +2012-09-04 - 686a329 - lavu 51.40.0 - opt.h Reordered the fields in default_val in AVOption, changed which default_val field is used for which AVOptionType. -2012-08-30 - 98298eb / a231832 - lavc 54.54.101 / 54.26.1 - avcodec.h +2012-08-30 - a231832 - lavc 54.26.1 - avcodec.h Add codec descriptor properties AV_CODEC_PROP_LOSSY and AV_CODEC_PROP_LOSSLESS. @@ -706,90 +374,101 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 Add codec descriptors for accessing codec properties without having to refer to a specific decoder or encoder. - f5f3684 / c223d79 - Add an AVCodecDescriptor struct and functions + c223d79 - Add an AVCodecDescriptor struct and functions avcodec_descriptor_get() and avcodec_descriptor_next(). - f5f3684 / 51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY. - 6c180b3 / 91e59fe - Add avcodec_descriptor_get_by_name(). + 51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY. + 91e59fe - Add avcodec_descriptor_get_by_name(). -2012-08-08 - f5f3684 / 987170c - lavu 51.68.100 / 51.38.0 - dict.h + +2012-08-08 - 1d9c2dc - lavu 51.39 - avutil.h + Don't implicitly include libavutil/common.h in avutil.h + +2012-08-08 - 987170c - lavu 51.38 - dict.h Add av_dict_count(). -2012-08-07 - 7a72695 / 104e10f - lavc 54.51.100 / 54.25.0 - avcodec.h +2012-08-07 - 104e10f - lavc 54.25 - avcodec.h Rename CodecID to AVCodecID and all CODEC_ID_* to AV_CODEC_ID_*. To provide backwards compatibility, CodecID is now #defined as AVCodecID. Note that this can break user code that includes avcodec.h and uses the 'CodecID' identifier. Such code should either #undef CodecID or stop using the CodecID name. -2012-08-03 - e776ee8 / 239fdf1 - lavu 51.66.101 / 51.37.1 - cpu.h +2012-08-03 - 239fdf1 - lavu 51.37.1 - cpu.h lsws 2.1.1 - swscale.h Rename AV_CPU_FLAG_MMX2 ---> AV_CPU_FLAG_MMXEXT. Rename SWS_CPU_CAPS_MMX2 ---> SWS_CPU_CAPS_MMXEXT. -2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h +2012-07-29 - 681ed00 - lavf 54.13.0 - avformat.h Add AVFMT_FLAG_NOBUFFER for low latency use cases. -2012-07-10 - fbe0245 / f3e5e6f - lavu 51.65.100 / 51.37.0 +2012-07-20 - b70d89a - lavfi 3.0.0 - avfilter.h + Add avfilter_unref_bufferp(). + +2012-07-10 - 5fade8a - lavu 51.37.0 Add av_malloc_array() and av_mallocz_array() -2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0 +2012-06-22 - d3d3a32 - lavu 51.34.0 Add av_usleep() -2012-06-20 - 4da42eb / ae0a301 - lavu 51.60.100 / 51.33.0 +2012-06-20 - ae0a301 - lavu 51.33.0 Move av_gettime() to libavutil, add libavutil/time.h -2012-06-09 - 82edf67 / 3971be0 - lavr 0.0.3 +2012-06-09 - 3971be0 - lavr 0.0.3 Add a parameter to avresample_build_matrix() for Dolby/DPLII downmixing. -2012-06-12 - c7b9eab / 9baeff9 - lavfi 2.79.100 / 2.23.0 - avfilter.h +2012-06-12 - 9baeff9 - lavfi 2.23.0 - avfilter.h Add AVFilterContext.nb_inputs/outputs. Deprecate AVFilterContext.input/output_count. -2012-06-12 - c7b9eab / 84b9fbe - lavfi 2.79.100 / 2.22.0 - avfilter.h +2012-06-12 - 84b9fbe - lavfi 2.22.0 - avfilter.h Add avfilter_pad_get_type() and avfilter_pad_get_name(). Those should now be used instead of accessing AVFilterPad members directly. -2012-06-12 - 3630a07 / b0f0dfc - lavu 51.57.100 / 51.32.0 - audioconvert.h +2012-06-12 - b0f0dfc - lavu 51.32.0 - audioconvert.h Add av_get_channel_layout_channel_index(), av_get_channel_name() and av_channel_layout_extract_channel(). -2012-05-25 - 53ce990 / 154486f - lavu 51.55.100 / 51.31.0 - opt.h +2012-05-25 - 154486f - lavu 51.31.0 - opt.h Add av_opt_set_bin() -2012-05-15 - lavfi 2.74.100 / 2.17.0 +2012-05-26 - e9cef89 - lavf 54.3.0 + Add AVFMT_TS_NONSTRICT format flag to indicate that a muxer supports + non-increasing monotone timestamps. + +2012-05-15 - lavfi 2.17.0 Add support for audio filters - 61930bd / ac71230, 1cbf7fb / a2cd9be - add video/audio buffer sink in a new installed + ac71230/a2cd9be - add video/audio buffer sink in a new installed header buffersink.h - 1cbf7fb / 720c6b7 - add av_buffersrc_write_frame(), deprecate + 720c6b7 - add av_buffersrc_write_frame(), deprecate av_vsrc_buffer_add_frame() - 61930bd / ab16504 - add avfilter_copy_buf_props() - 61930bd / 9453c9e - add extended_data to AVFilterBuffer - 61930bd / 1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays() + ab16504 - add avfilter_copy_buf_props() + 9453c9e - add extended_data to AVFilterBuffer + 1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays() -2012-05-09 - lavu 51.53.100 / 51.30.0 - samplefmt.h - 61930bd / 142e740 - add av_samples_copy() - 61930bd / 6d7f617 - add av_samples_set_silence() +2012-05-09 - lavu 51.30.0 - samplefmt.h + 142e740 - add av_samples_copy() + 6d7f617 - add av_samples_set_silence() -2012-05-09 - 61930bd / a5117a2 - lavc 54.21.101 / 54.13.1 +2012-05-09 - a5117a2 - lavc 54.13.1 For audio formats with fixed frame size, the last frame no longer needs to be padded with silence, libavcodec will handle this internally (effectively all encoders behave as if they had CODEC_CAP_SMALL_LAST_FRAME set). -2012-05-07 - 653d117 / 828bd08 - lavc 54.20.100 / 54.13.0 - avcodec.h +2012-05-07 - 828bd08 - lavc 54.13.0 - avcodec.h Add sample_rate and channel_layout fields to AVFrame. -2012-05-01 - 2330eb1 / 4010d72 - lavr 0.0.1 +2012-05-01 - 4010d72 - lavr 0.0.1 Change AV_MIX_COEFF_TYPE_Q6 to AV_MIX_COEFF_TYPE_Q8. -2012-04-25 - e890b68 / 3527a73 - lavu 51.48.100 / 51.29.0 - cpu.h +2012-04-25 - 3527a73 - lavu 51.29.0 - cpu.h Add av_parse_cpu_flags() -2012-04-24 - 3ead79e / c8af852 - lavr 0.0.0 +2012-04-24 - c8af852 - lavr 0.0.0 Add libavresample audio conversion library -2012-04-20 - 3194ab7 / 0c0d1bc - lavu 51.47.100 / 51.28.0 - audio_fifo.h +2012-04-20 - 0c0d1bc - lavu 51.28.0 - audio_fifo.h Add audio FIFO functions: av_audio_fifo_free() av_audio_fifo_alloc() @@ -801,155 +480,137 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 av_audio_fifo_size() av_audio_fifo_space() -2012-04-14 - lavfi 2.70.100 / 2.16.0 - avfiltergraph.h - 7432bcf / d7bcc71 Add avfilter_graph_parse2(). +2012-04-14 - lavfi 2.16.0 - avfiltergraph.h + d7bcc71 Add avfilter_graph_parse2(). + 91d3cbe Add avfilter_inout_alloc() and avfilter_inout_free(). -2012-04-08 - 6bfb304 / 4d693b0 - lavu 51.46.100 / 51.27.0 - samplefmt.h +2012-04-08 - 4d693b0 - lavu 51.27.0 - samplefmt.h Add av_get_packed_sample_fmt() and av_get_planar_sample_fmt() -2012-03-21 - b75c67d - lavu 51.43.100 - Add bprint.h for bprint API. +2012-04-05 - 5cc51a5 - lavu 51.26.0 - audioconvert.h + Add av_get_default_channel_layout() -2012-02-21 - 9cbf17e - lavc 54.4.100 - Add av_get_pcm_codec() function. - -2012-02-16 - 560b224 - libswr 0.7.100 - Add swr_set_matrix() function. - -2012-02-09 - c28e7af - lavu 51.39.100 - Add a new installed header libavutil/timestamp.h with timestamp - utilities. - -2012-02-06 - 70ffda3 - lavu 51.38.100 - Add av_parse_ratio() function to parseutils.h. - -2012-02-06 - 70ffda3 - lavu 51.38.100 - Add AV_LOG_MAX_OFFSET macro to log.h. - -2012-02-02 - 0eaa123 - lavu 51.37.100 - Add public timecode helpers. - -2012-01-24 - 0c3577b - lavfi 2.60.100 - Add avfilter_graph_dump. - -2012-03-20 - 0ebd836 / 3c90cc2 - lavfo 54.2.0 +2012-03-20 - 3c90cc2 - lavfo 54.2.0 Deprecate av_read_packet(), use av_read_frame() with AVFMT_FLAG_NOPARSE | AVFMT_FLAG_NOFILLIN in AVFormatContext.flags -2012-03-05 - lavc 54.10.100 / 54.8.0 - f095391 / 6699d07 Add av_get_exact_bits_per_sample() - f095391 / 9524cf7 Add av_get_audio_frame_duration() +2012-03-06 - 4d851f8 - lavu 51.25.0 - cpu.h + Add av_set_cpu_flags_mask(). -2012-03-04 - 2af8f2c / 44fe77b - lavc 54.8.100 / 54.7.0 - avcodec.h +2012-03-05 - lavc 54.8.0 + 6699d07 Add av_get_exact_bits_per_sample() + 9524cf7 Add av_get_audio_frame_duration() + +2012-03-04 - 44fe77b - lavc 54.7.0 - avcodec.h Add av_codec_is_encoder/decoder(). -2012-03-01 - 1eb7f39 / 442c132 - lavc 54.5.100 / 54.3.0 - avcodec.h +2012-03-01 - 442c132 - lavc 54.3.0 - avcodec.h Add av_packet_shrink_side_data. -2012-02-29 - 79ae084 / dd2a4bc - lavf 54.2.100 / 54.2.0 - avformat.h +2012-02-29 - dd2a4bc - lavf 54.2.0 - avformat.h Add AVStream.attached_pic and AV_DISPOSITION_ATTACHED_PIC, used for dealing with attached pictures/cover art. -2012-02-25 - 305e4b3 / c9bca80 - lavu 51.41.100 / 51.24.0 - error.h +2012-02-25 - c9bca80 - lavu 51.24.0 - error.h Add AVERROR_UNKNOWN NOTE: this was backported to 0.8 -2012-02-20 - eadd426 / e9cda85 - lavc 54.2.100 / 54.2.0 +2012-02-20 - e9cda85 - lavc 54.2.0 Add duration field to AVCodecParserContext -2012-02-20 - eadd426 / 0b42a93 - lavu 51.40.100 / 51.23.1 - mathematics.h +2012-02-20 - 0b42a93 - lavu 51.23.1 - mathematics.h Add av_rescale_q_rnd() -2012-02-08 - f2b20b7 / 38d5533 - lavu 51.38.101 / 51.22.1 - pixdesc.h +2012-02-08 - 38d5533 - lavu 51.22.1 - pixdesc.h Add PIX_FMT_PSEUDOPAL flag. -2012-02-08 - f2b20b7 / 52f82a1 - lavc 54.2.100 / 54.1.0 +2012-02-08 - 52f82a1 - lavc 54.01.0 Add avcodec_encode_video2() and deprecate avcodec_encode_video(). -2012-02-01 - 4c677df / 316fc74 - lavc 54.1.0 +2012-02-01 - 316fc74 - lavc 54.01.0 Add av_fast_padded_malloc() as alternative for av_realloc() when aligned memory is required. The buffer will always have FF_INPUT_BUFFER_PADDING_SIZE zero-padded bytes at the end. -2012-01-31 - a369a6b / dd6d3b0 - lavf 54.1.0 +2012-01-31 - dd6d3b0 - lavf 54.01.0 Add avformat_get_riff_video_tags() and avformat_get_riff_audio_tags(). NOTE: this was backported to 0.8 -2012-01-31 - a369a6b / af08d9a - lavc 54.1.0 +2012-01-31 - af08d9a - lavc 54.01.0 Add avcodec_is_open() function. NOTE: this was backported to 0.8 -2012-01-30 - 151ecc2 / 8b93312 - lavu 51.36.100 / 51.22.0 - intfloat.h +2012-01-30 - 8b93312 - lavu 51.22.0 - intfloat.h Add a new installed header libavutil/intfloat.h with int/float punning functions. NOTE: this was backported to 0.8 -2012-01-25 - lavf 53.31.100 / 53.22.0 - 3c5fe5b / f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible +2012-01-25 - lavf 53.22.0 + f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible buffered data within a muxer. Added AVFMT_ALLOW_FLUSH for muxers supporting it (av_write_frame makes sure it is called only for muxers with this flag). -2012-01-15 - lavc 53.56.105 / 53.34.0 +------------------------------8<------------------------------------- + 0.8 branch was cut here +----------------------------->8-------------------------------------- + +2012-01-15 - lavc 53.34.0 New audio encoding API: - 67f5650 / b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio + b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio encoders. - 67f5650 / 5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function. - 67f5650 / b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio(). + 5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function. + b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio(). Add AVCodec.encode2(). -2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0 +2012-01-12 - 3167dc9 - lavfi 2.15.0 Add a new installed header -- libavfilter/version.h -- with version macros. -2011-12-08 - a502939 - lavfi 2.52.0 - Add av_buffersink_poll_frame() to buffersink.h. - -2011-12-08 - 26c6fec - lavu 51.31.0 - Add av_log_format_line. - -2011-12-03 - 976b095 - lavu 51.30.0 - Add AVERROR_BUG. - -2011-11-24 - 573ffbb - lavu 51.28.1 - Add av_get_alt_sample_fmt() to samplefmt.h. - -2011-11-03 - 96949da - lavu 51.23.0 - Add av_strcasecmp() and av_strncasecmp() to avstring.h. - -2011-10-20 - b35e9e1 - lavu 51.22.0 - Add av_strtok() to avstring.h. - -2012-01-03 - ad1c8dd / b73ec05 - lavu 51.34.100 / 51.21.0 +2011-01-03 - b73ec05 - lavu 51.21.0 Add av_popcount64 -2011-12-18 - 7c29313 / 8400b12 - lavc 53.46.1 / 53.28.1 +2011-12-25 - lavfi 2.14.0 + e1d9dbf Add a new installed header - buffersrc.h + It contains a new function av_buffersrc_buffer() that allows passing + frames to the 'buffer' filter, but unlike av_vsrc_buffer_add_frame() + it allows for direct rendering. + 1c9e340 Add avfilter_copy_frame_props() for copying properties from + AVFrame to AVFilterBufferRef. + +2011-12-25 - lavc 53.31.0 + Add the following new fields to AVFrame: + b58dbb5 sample_aspect_ratio + 3a2ddf7 width, height + 8a4a5f6 format + +2011-12-18 - 8400b12 - lavc 53.28.1 Deprecate AVFrame.age. The field is unused. -2011-12-12 - 8bc7fe4 / 5266045 - lavf 53.25.0 / 53.17.0 +2011-12-12 - 5266045 - lavf 53.17.0 Add avformat_close_input(). Deprecate av_close_input_file() and av_close_input_stream(). -2011-12-02 - e4de716 / 0eea212 - lavc 53.40.0 / 53.25.0 +2011-12-02 - 0eea212 - lavc 53.25.0 Add nb_samples and extended_data fields to AVFrame. Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE. Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4(). avcodec_decode_audio4() writes output samples to an AVFrame, which allows audio decoders to use get_buffer(). -2011-12-04 - e4de716 / 560f773 - lavc 53.40.0 / 53.24.0 +2011-12-04 - 560f773 - lavc 53.24.0 Change AVFrame.data[4]/base[4]/linesize[4]/error[4] to [8] at next major bump. Change AVPicture.data[4]/linesize[4] to [8] at next major bump. Change AVCodecContext.error[4] to [8] at next major bump. Add AV_NUM_DATA_POINTERS to simplify the bump transition. -2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0 +2011-11-23 - bbb46f3 - lavu 51.18.0 Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and av_samples_alloc(), to samplefmt.h. -2011-11-23 - 8e576d5 / 8889cc4 - lavu 51.27.0 / 51.17.0 +2011-11-23 - 8889cc4 - lavu 51.17.0 Add planar sample formats and av_sample_fmt_is_planar() to samplefmt.h. -2011-11-19 - dbb38bc / f3a29b7 - lavc 53.36.0 / 53.21.0 +2011-11-19 - f3a29b7 - lavc 53.21.0 Move some AVCodecContext fields to a new private struct, AVCodecInternal, which is accessed from a new field, AVCodecContext.internal. - fields moved: @@ -957,364 +618,220 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 AVCodecContext.internal_buffer_count --> AVCodecInternal.buffer_count AVCodecContext.is_copy --> AVCodecInternal.is_copy -2011-11-16 - 8709ba9 / 6270671 - lavu 51.26.0 / 51.16.0 +2011-11-16 - 6270671 - lavu 51.16.0 Add av_timegm() -2011-11-13 - lavf 53.21.0 / 53.15.0 +2011-11-13 - lavf 53.15.0 New interrupt callback API, allowing per-AVFormatContext/AVIOContext interrupt callbacks. - 5f268ca / 6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to + 6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to AVFormatContext. - 5f268ca / 1dee0ac Add avio_open2() with additional parameters. Those are + 1dee0ac Add avio_open2() with additional parameters. Those are an interrupt callback and an options AVDictionary. This will allow passing AVOptions to protocols after lavf 54.0. -2011-11-06 - 13b7781 / ba04ecf - lavu 51.24.0 / 51.14.0 +2011-11-06 - ba04ecf - lavu 51.14.0 Add av_strcasecmp() and av_strncasecmp() to avstring.h. -2011-11-06 - 13b7781 / 07b172f - lavu 51.24.0 / 51.13.0 +2011-11-06 - 07b172f - lavu 51.13.0 Add av_toupper()/av_tolower() -2011-11-05 - d8cab5c / b6d08f4 - lavf 53.19.0 / 53.13.0 +2011-11-05 - b6d08f4 - lavf 53.13.0 Add avformat_network_init()/avformat_network_deinit() -2011-10-27 - 6faf0a2 / 512557b - lavc 53.24.0 / 53.15.0 +2011-10-27 - 512557b - lavc 53.15.0 Remove avcodec_parse_frame. Deprecate AVCodecContext.parse_only and CODEC_CAP_PARSE_ONLY. -2011-10-19 - d049257 / 569129a - lavf 53.17.0 / 53.10.0 +2011-10-19 - 569129a - lavf 53.10.0 Add avformat_new_stream(). Deprecate av_new_stream(). -2011-10-13 - 91eb1b1 / b631fba - lavf 53.16.0 / 53.9.0 +2011-10-13 - b631fba - lavf 53.9.0 Add AVFMT_NO_BYTE_SEEK AVInputFormat flag. -2011-10-12 - lavu 51.21.0 / 51.12.0 +2011-10-12 - lavu 51.12.0 AVOptions API rewrite. - - f884ef0 / 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_* + - 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_* - new setting/getting functions with slightly different semantics: - f884ef0 / dac66da av_set_string3 -> av_opt_set + dac66da av_set_string3 -> av_opt_set av_set_double -> av_opt_set_double av_set_q -> av_opt_set_q av_set_int -> av_opt_set_int - f884ef0 / 41d9d51 av_get_string -> av_opt_get + 41d9d51 av_get_string -> av_opt_get av_get_double -> av_opt_get_double av_get_q -> av_opt_get_q av_get_int -> av_opt_get_int - - f884ef0 / 8c5dcaa trivial rename av_next_option -> av_opt_next - - f884ef0 / 641c7af new functions - av_opt_child_next, av_opt_child_class_next + - 8c5dcaa trivial rename av_next_option -> av_opt_next + - 641c7af new functions - av_opt_child_next, av_opt_child_class_next and av_opt_find2() -2011-09-22 - a70e787 - lavu 51.17.0 - Add av_x_if_null(). - -2011-09-18 - 645cebb - lavc 53.16.0 - Add showall flag2 - -2011-09-16 - ea8de10 - lavfi 2.42.0 - Add avfilter_all_channel_layouts. - -2011-09-16 - 9899037 - lavfi 2.41.0 - Rename avfilter_all_* function names to avfilter_make_all_*. - - In particular, apply the renames: - avfilter_all_formats -> avfilter_make_all_formats - avfilter_all_channel_layouts -> avfilter_make_all_channel_layouts - avfilter_all_packing_formats -> avfilter_make_all_packing_formats - -2011-09-12 - 4381bdd - lavfi 2.40.0 - Change AVFilterBufferRefAudioProps.sample_rate type from uint32_t to int. - -2011-09-12 - 2c03174 - lavfi 2.40.0 - Simplify signature for avfilter_get_audio_buffer(), make it - consistent with avfilter_get_video_buffer(). - -2011-09-06 - 4f7dfe1 - lavfi 2.39.0 - Rename libavfilter/vsink_buffer.h to libavfilter/buffersink.h. - -2011-09-06 - c4415f6 - lavfi 2.38.0 - Unify video and audio sink API. - - In particular, add av_buffersink_get_buffer_ref(), deprecate - av_vsink_buffer_get_video_buffer_ref() and change the value for the - opaque field passed to the abuffersink init function. - -2011-09-04 - 61e2e29 - lavu 51.16.0 - Add av_asprintf(). - -2011-08-22 - dacd827 - lavf 53.10.0 - Add av_find_program_from_stream(). - -2011-08-20 - 69e2c1a - lavu 51.13.0 - Add av_get_media_type_string(). - -2011-09-03 - 1889c67 / fb4ca26 - lavc 53.13.0 - lavf 53.11.0 +2011-09-03 - fb4ca26 - lavc 53.10.0 + lavf 53.6.0 lsws 2.1.0 Add {avcodec,avformat,sws}_get_class(). -2011-08-03 - 1889c67 / c11fb82 - lavu 51.15.0 +2011-09-03 - c11fb82 - lavu 51.10.0 Add AV_OPT_SEARCH_FAKE_OBJ flag for av_opt_find() function. -2011-08-14 - 323b930 - lavu 51.12.0 - Add av_fifo_peek2(), deprecate av_fifo_peek(). - -2011-08-26 - lavu 51.14.0 / 51.9.0 - - 976a8b2 / add41de..976a8b2 / abc78a5 Do not include intfloat_readwrite.h, +2011-08-26 - lavu 51.9.0 + - f2011ed Add av_fifo_peek2(), deprecate av_fifo_peek(). + - add41de..abc78a5 Do not include intfloat_readwrite.h, mathematics.h, rational.h, pixfmt.h, or log.h from avutil.h. -2011-08-16 - 27fbe31 / 48f9e45 - lavf 53.11.0 / 53.8.0 +2011-08-16 - 48f9e45 - lavf 53.4.0 Add avformat_query_codec(). -2011-08-16 - 27fbe31 / bca06e7 - lavc 53.11.0 +2011-08-16 - bca06e7 - lavc 53.8.0 Add avcodec_get_type(). -2011-08-06 - 0cb233c / 2f63440 - lavf 53.7.0 +2011-08-06 - 2f63440 - lavf 53.4.0 Add error_recognition to AVFormatContext. -2011-08-02 - 1d186e9 / 9d39cbf - lavc 53.9.1 +2011-08-02 - 9d39cbf - lavc 53.7.1 Add AV_PKT_FLAG_CORRUPT AVPacket flag. -2011-07-16 - b57df29 - lavfi 2.27.0 - Add audio packing negotiation fields and helper functions. - - In particular, add AVFilterPacking enum, planar, in_packings and - out_packings fields to AVFilterLink, and the functions: - avfilter_set_common_packing_formats() - avfilter_all_packing_formats() - -2011-07-10 - 3602ad7 / a67c061 - lavf 53.6.0 +2011-07-10 - a67c061 - lavf 53.3.0 Add avformat_find_stream_info(), deprecate av_find_stream_info(). NOTE: this was backported to 0.7 -2011-07-10 - 3602ad7 / 0b950fe - lavc 53.8.0 +2011-07-10 - 0b950fe - lavc 53.6.0 Add avcodec_open2(), deprecate avcodec_open(). NOTE: this was backported to 0.7 Add avcodec_alloc_context3. Deprecate avcodec_alloc_context() and avcodec_alloc_context2(). -2011-07-01 - b442ca6 - lavf 53.5.0 - avformat.h - Add function av_get_output_timestamp(). - -2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h - Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum. - -2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h - Add layout negotiation fields and helper functions. - - In particular, add in_chlayouts and out_chlayouts to AVFilterLink, - and the functions: - avfilter_set_common_sample_formats() - avfilter_set_common_channel_layouts() - avfilter_all_channel_layouts() - -2011-06-19 - 527ca39 - lavfi 2.22.0 - AVFilterFormats - Change type of AVFilterFormats.formats from int * to int64_t *, - and update formats handling API accordingly. - - avfilter_make_format_list() still takes a int32_t array and converts - it to int64_t. A new function, avfilter_make_format64_list(), that - takes int64_t arrays has been added. - -2011-06-19 - 44f669e - lavfi 2.21.0 - vsink_buffer.h - Add video sink buffer and vsink_buffer.h public header. - -2011-06-12 - 9fdf772 - lavfi 2.18.0 - avcodec.h - Add avfilter_get_video_buffer_ref_from_frame() function in - libavfilter/avcodec.h. - -2011-06-12 - c535494 - lavfi 2.17.0 - avfiltergraph.h - Add avfilter_inout_alloc() and avfilter_inout_free() functions. - -2011-06-12 - 6119b23 - lavfi 2.16.0 - avfilter_graph_parse() - Change avfilter_graph_parse() signature. - -2011-06-23 - 686959e / 67e9ae1 - lavu 51.10.0 / 51.8.0 - attributes.h +2011-06-23 - 67e9ae1 - lavu 51.8.0 - attributes.h Add av_printf_format(). -2011-06-16 - 2905e3f / 05e84c9, 2905e3f / 25de595 - lavf 53.4.0 / 53.2.0 - avformat.h +------------------------------8<------------------------------------- + 0.7 branch was cut here +----------------------------->8-------------------------------------- + +2011-06-16 - 05e84c9, 25de595 - lavf 53.2.0 - avformat.h Add avformat_open_input and avformat_write_header(). Deprecate av_open_input_stream, av_open_input_file, AVFormatParameters and av_write_header. -2011-06-16 - 2905e3f / 7e83e1c, 2905e3f / dc59ec5 - lavu 51.9.0 / 51.7.0 - opt.h +2011-06-16 - 7e83e1c, dc59ec5 - lavu 51.7.0 - opt.h Add av_opt_set_dict() and av_opt_find(). Deprecate av_find_opt(). Add AV_DICT_APPEND flag. -2011-06-10 - 45fb647 / cb7c11c - lavu 51.6.0 - opt.h +2011-06-10 - cb7c11c - lavu 51.6.0 - opt.h Add av_opt_flag_is_set(). -2011-06-10 - c381960 - lavfi 2.15.0 - avfilter_get_audio_buffer_ref_from_arrays - Add avfilter_get_audio_buffer_ref_from_arrays() to avfilter.h. - -2011-06-09 - f9ecb84 / d9f80ea - lavu 51.8.0 - AVMetadata +2011-06-08 - d9f80ea - lavu 51.5.0 - AVMetadata Move AVMetadata from lavf to lavu and rename it to AVDictionary -- new installed header dict.h. All av_metadata_* functions renamed to av_dict_*. -2011-06-07 - d552f61 / a6703fa - lavu 51.8.0 - av_get_bytes_per_sample() +2011-06-07 - a6703fa - lavu 51.4.0 - av_get_bytes_per_sample() Add av_get_bytes_per_sample() in libavutil/samplefmt.h. Deprecate av_get_bits_per_sample_fmt(). -2011-06-05 - f956924 / b39b062 - lavu 51.8.0 - opt.h +2011-06-05 - b39b062 - lavu 51.3.0 - opt.h Add av_opt_free convenience function. -2011-06-06 - 95a0242 - lavfi 2.14.0 - AVFilterBufferRefAudioProps - Remove AVFilterBufferRefAudioProps.size, and use nb_samples in - avfilter_get_audio_buffer() and avfilter_default_get_audio_buffer() in - place of size. - -2011-06-06 - 0bc2cca - lavu 51.6.0 - av_samples_alloc() - Switch nb_channels and nb_samples parameters order in - av_samples_alloc(). - -2011-06-06 - e1c7414 - lavu 51.5.0 - av_samples_* - Change the data layout created by av_samples_fill_arrays() and - av_samples_alloc(). - -2011-06-06 - 27bcf55 - lavfi 2.13.0 - vsrc_buffer.h - Make av_vsrc_buffer_add_video_buffer_ref() accepts an additional - flags parameter in input. - -2011-06-03 - e977ca2 - lavfi 2.12.0 - avfilter_link_free() - Add avfilter_link_free() function. - -2011-06-02 - 5ad38d9 - lavu 51.4.0 - av_force_cpu_flags() - Add av_cpu_flags() in libavutil/cpu.h. - -2011-05-28 - e71f260 - lavu 51.3.0 - pixdesc.h +2011-05-28 - 0420bd7 - lavu 51.2.0 - pixdesc.h Add av_get_pix_fmt_name() in libavutil/pixdesc.h, and deprecate avcodec_get_pix_fmt_name() in libavcodec/avcodec.h in its favor. -2011-05-25 - 39e4206 / 30315a8 - lavf 53.3.0 - avformat.h +2011-05-25 - 30315a8 - lavf 53.1.0 - avformat.h Add fps_probe_size to AVFormatContext. -2011-05-22 - 5ecdfd0 - lavf 53.2.0 - avformat.h - Introduce avformat_alloc_output_context2() and deprecate - avformat_alloc_output_context(). - -2011-05-22 - 83db719 - lavfi 2.10.0 - vsrc_buffer.h - Make libavfilter/vsrc_buffer.h public. - -2011-05-19 - c000a9f - lavfi 2.8.0 - avcodec.h - Add av_vsrc_buffer_add_frame() to libavfilter/avcodec.h. - -2011-05-14 - 9fdf772 - lavfi 2.6.0 - avcodec.h - Add avfilter_get_video_buffer_ref_from_frame() to libavfilter/avcodec.h. - -2011-05-18 - 75a37b5 / 64150ff - lavc 53.7.0 - AVCodecContext.request_sample_fmt +2011-05-18 - 64150ff - lavc 53.4.0 - AVCodecContext.request_sample_fmt Add request_sample_fmt field to AVCodecContext. -2011-05-10 - 59eb12f / 188dea1 - lavc 53.6.0 - avcodec.h +2011-05-10 - 188dea1 - lavc 53.3.0 - avcodec.h Deprecate AVLPCType and the following fields in AVCodecContext: lpc_coeff_precision, prediction_order_method, min_partition_order, max_partition_order, lpc_type, lpc_passes. Corresponding FLAC encoder options should be used instead. -2011-05-07 - 9fdf772 - lavfi 2.5.0 - avcodec.h - Add libavfilter/avcodec.h header and avfilter_copy_frame_props() - function. - -2011-05-07 - 18ded93 - lavc 53.5.0 - AVFrame - Add format field to AVFrame. - -2011-05-07 - 22333a6 - lavc 53.4.0 - AVFrame - Add width and height fields to AVFrame. - -2011-05-01 - 35fe66a - lavfi 2.4.0 - avfilter.h - Rename AVFilterBufferRefVideoProps.pixel_aspect to - sample_aspect_ratio. - -2011-05-01 - 77e9dee - lavc 53.3.0 - AVFrame - Add a sample_aspect_ratio field to AVFrame. - -2011-05-01 - 1ba5727 - lavc 53.2.0 - AVFrame - Add a pkt_pos field to AVFrame. - -2011-04-29 - 35ceaa7 - lavu 51.2.0 - mem.h - Add av_dynarray_add function for adding - an element to a dynamic array. - -2011-04-26 - d7e5aeb / bebe72f - lavu 51.1.0 - avutil.h +2011-04-26 - bebe72f - lavu 51.1.0 - avutil.h Add AVPictureType enum and av_get_picture_type_char(), deprecate FF_*_TYPE defines and av_get_pict_type_char() defined in libavcodec/avcodec.h. -2011-04-26 - d7e5aeb / 10d3940 - lavfi 2.3.0 - avfilter.h +2011-04-26 - 10d3940 - lavfi 2.3.0 - avfilter.h Add pict_type and key_frame fields to AVFilterBufferRefVideo. -2011-04-26 - d7e5aeb / 7a11c82 - lavfi 2.2.0 - vsrc_buffer +2011-04-26 - 7a11c82 - lavfi 2.2.0 - vsrc_buffer Add sample_aspect_ratio fields to vsrc_buffer arguments -2011-04-21 - 8772156 / 94f7451 - lavc 53.1.0 - avcodec.h +2011-04-21 - 94f7451 - lavc 53.1.0 - avcodec.h Add CODEC_CAP_SLICE_THREADS for codecs supporting sliced threading. 2011-04-15 - lavc 52.120.0 - avcodec.h AVPacket structure got additional members for passing side information: - c407984 / 4de339e introduce side information for AVPacket - c407984 / 2d8591c make containers pass palette change in AVPacket + 4de339e introduce side information for AVPacket + 2d8591c make containers pass palette change in AVPacket 2011-04-12 - lavf 52.107.0 - avio.h Avio cleanup, part II - deprecate the entire URLContext API: - c55780d / 175389c add avio_check as a replacement for url_exist - 9891004 / ff1ec0c add avio_pause and avio_seek_time as replacements + 175389c add avio_check as a replacement for url_exist + ff1ec0c add avio_pause and avio_seek_time as replacements for _av_url_read_fseek/fpause - d4d0932 / cdc6a87 deprecate av_protocol_next(), avio_enum_protocols + cdc6a87 deprecate av_protocol_next(), avio_enum_protocols should be used instead. - c88caa5 / 80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb. - c88caa5 / f87b1b3 rename open flags: URL_* -> AVIO_* - d4d0932 / f8270bb add avio_enum_protocols. - d4d0932 / 5593f03 deprecate URLProtocol. - d4d0932 / c486dad deprecate URLContext. - d4d0932 / 026e175 deprecate the typedef for URLInterruptCB - c88caa5 / 8e76a19 deprecate av_register_protocol2. - 11d7841 / b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME - 11d7841 / 1305d93 deprecate av_url_read_seek - 11d7841 / fa104e1 deprecate av_url_read_pause - 434f248 / 727c7aa deprecate url_get_filename(). - 434f248 / 5958df3 deprecate url_max_packet_size(). - 434f248 / 1869ea0 deprecate url_get_file_handle(). - 434f248 / 32a97d4 deprecate url_filesize(). - 434f248 / e52a914 deprecate url_close(). - 434f248 / 58a48c6 deprecate url_seek(). - 434f248 / 925e908 deprecate url_write(). - 434f248 / dce3756 deprecate url_read_complete(). - 434f248 / bc371ac deprecate url_read(). - 434f248 / 0589da0 deprecate url_open(). - 434f248 / 62eaaea deprecate url_connect. - 434f248 / 5652bb9 deprecate url_alloc. - 434f248 / 333e894 deprecate url_open_protocol - 434f248 / e230705 deprecate url_poll and URLPollEntry + 80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb. + f87b1b3 rename open flags: URL_* -> AVIO_* + f8270bb add avio_enum_protocols. + 5593f03 deprecate URLProtocol. + c486dad deprecate URLContext. + 026e175 deprecate the typedef for URLInterruptCB + 8e76a19 deprecate av_register_protocol2. + b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME + 1305d93 deprecate av_url_read_seek + fa104e1 deprecate av_url_read_pause + 727c7aa deprecate url_get_filename(). + 5958df3 deprecate url_max_packet_size(). + 1869ea0 deprecate url_get_file_handle(). + 32a97d4 deprecate url_filesize(). + e52a914 deprecate url_close(). + 58a48c6 deprecate url_seek(). + 925e908 deprecate url_write(). + dce3756 deprecate url_read_complete(). + bc371ac deprecate url_read(). + 0589da0 deprecate url_open(). + 62eaaea deprecate url_connect. + 5652bb9 deprecate url_alloc. + 333e894 deprecate url_open_protocol + e230705 deprecate url_poll and URLPollEntry + +2011-04-10 - lavu 50.40.0 - pixfmt.h + Add PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats 2011-04-08 - lavf 52.106.0 - avformat.h Minor avformat.h cleanup: - d4d0932 / a9bf9d8 deprecate av_guess_image2_codec - d4d0932 / c3675df rename avf_sdp_create->av_sdp_create + a9bf9d8 deprecate av_guess_image2_codec + c3675df rename avf_sdp_create->av_sdp_create 2011-04-03 - lavf 52.105.0 - avio.h Large-scale renaming/deprecating of AVIOContext-related functions: - 2cae980 / 724f6a0 deprecate url_fdopen - 2cae980 / 403ee83 deprecate url_open_dyn_packet_buf - 2cae980 / 6dc7d80 rename url_close_dyn_buf -> avio_close_dyn_buf - 2cae980 / b92c545 rename url_open_dyn_buf -> avio_open_dyn_buf - 2cae980 / 8978fed introduce an AVIOContext.seekable field as a replacement for + 724f6a0 deprecate url_fdopen + 403ee83 deprecate url_open_dyn_packet_buf + 6dc7d80 rename url_close_dyn_buf -> avio_close_dyn_buf + b92c545 rename url_open_dyn_buf -> avio_open_dyn_buf + 8978fed introduce an AVIOContext.seekable field as a replacement for AVIOContext.is_streamed and url_is_streamed() - 1caa412 / b64030f deprecate get_checksum() - 1caa412 / 4c4427a deprecate init_checksum() - 2fd41c9 / 4ec153b deprecate udp_set_remote_url/get_local_port - 4fa0e24 / 933e90a deprecate av_url_read_fseek/fpause - 4fa0e24 / 8d9769a deprecate url_fileno - 0fecf26 / b7f2fdd rename put_flush_packet -> avio_flush - 0fecf26 / 35f1023 deprecate url_close_buf - 0fecf26 / 83fddae deprecate url_open_buf - 0fecf26 / d9d86e0 rename url_fprintf -> avio_printf - 0fecf26 / 59f65d9 deprecate url_setbufsize - 6947b0c / 3e68b3b deprecate url_ferror + b64030f deprecate get_checksum() + 4c4427a deprecate init_checksum() + 4ec153b deprecate udp_set_remote_url/get_local_port + 933e90a deprecate av_url_read_fseek/fpause + 8d9769a deprecate url_fileno + b7f2fdd rename put_flush_packet -> avio_flush + 35f1023 deprecate url_close_buf + 83fddae deprecate url_open_buf + d9d86e0 rename url_fprintf -> avio_printf + 59f65d9 deprecate url_setbufsize + 3e68b3b deprecate url_ferror + 66e5b1d deprecate url_feof e8bb2e2 deprecate url_fget_max_packet_size 76aa876 rename url_fsize -> avio_size e519753 deprecate url_fgetc @@ -1335,12 +852,9 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 b3db9ce deprecate get_partial_buffer 8d9ac96 rename av_alloc_put_byte -> avio_alloc_context -2011-03-25 - 27ef7b1 / 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type +2011-03-25 - 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type Add audio_service_type field to AVCodecContext. -2011-03-17 - e309fdc - lavu 50.40.0 - pixfmt.h - Add PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats - 2011-03-02 - 863c471 - lavf 52.103.0 - av_pkt_dump2, av_pkt_dump_log2 Add new functions av_pkt_dump2, av_pkt_dump_log2 that uses the source stream timebase for outputting timestamps. Deprecate @@ -1373,11 +887,11 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 2011-02-10 - 12c14cd - lavf 52.99.0 - AVStream.disposition Add AV_DISPOSITION_HEARING_IMPAIRED and AV_DISPOSITION_VISUAL_IMPAIRED. -2011-02-09 - c0b102c - lavc 52.112.0 - avcodec_thread_init() +2011-02-09 - 5592734 - lavc 52.112.0 - avcodec_thread_init() Deprecate avcodec_thread_init()/avcodec_thread_free() use; instead set thread_count before calling avcodec_open. -2011-02-09 - 37b00b4 - lavc 52.111.0 - threading API +2011-02-09 - 778b08a - lavc 52.111.0 - threading API Add CODEC_CAP_FRAME_THREADS with new restrictions on get_buffer()/ release_buffer()/draw_horiz_band() callbacks for appropriate codecs. Add thread_type and active_thread_type fields to AVCodecContext. @@ -1407,12 +921,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0 2011-02-02 - dfd2a00 - lavu 50.37.0 - log.h Make av_dlog public. -2011-01-31 - 7b3ea55 - lavfi 1.76.0 - vsrc_buffer - Add sample_aspect_ratio fields to vsrc_buffer arguments - -2011-01-31 - 910b5b8 - lavfi 1.75.0 - AVFilterLink sample_aspect_ratio - Add sample_aspect_ratio field to AVFilterLink. - 2011-01-15 - a242ac3 - lavfi 1.74.0 - AVFilterBufferRefAudioProps Rename AVFilterBufferRefAudioProps.samples_nb to nb_samples. diff --git a/doc/Doxyfile b/doc/Doxyfile index 96327ff6e5..58f7dfbece 100644 --- a/doc/Doxyfile +++ b/doc/Doxyfile @@ -25,7 +25,7 @@ DOXYFILE_ENCODING = UTF-8 # The PROJECT_NAME tag is a single word (or a sequence of words surrounded # by quotes) that should identify the project. -PROJECT_NAME = FFmpeg +PROJECT_NAME = Libav # The PROJECT_NUMBER tag can be used to enter a project or revision number. # This could be handy for archiving the generated documentation or @@ -33,7 +33,7 @@ PROJECT_NAME = FFmpeg PROJECT_NUMBER = -# With the PROJECT_LOGO tag one can specify a logo or icon that is included +# With the PROJECT_LOGO tag one can specify an logo or icon that is included # in the documentation. The maximum height of the logo should not exceed 55 # pixels and the maximum width should not exceed 200 pixels. Doxygen will # copy the logo to the output directory. @@ -639,14 +639,14 @@ EXCLUDE_SYMBOLS = # directories that contain example code fragments that are included (see # the \include command). -EXAMPLE_PATH = doc/examples/ +EXAMPLE_PATH = # If the value of the EXAMPLE_PATH tag contains directories, you can use the # EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp # and *.h) to filter out the source-files in the directories. If left # blank all files are included. -EXAMPLE_PATTERNS = *.c +EXAMPLE_PATTERNS = *-example.c # If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be # searched for input files to be used with the \include or \dontinclude @@ -709,7 +709,7 @@ INLINE_SOURCES = NO # doxygen to hide any special comment blocks from generated source code # fragments. Normal C and C++ comments will always remain visible. -STRIP_CODE_COMMENTS = NO +STRIP_CODE_COMMENTS = YES # If the REFERENCED_BY_RELATION tag is set to YES # then for each documented function all documented @@ -818,7 +818,7 @@ HTML_STYLESHEET = # 180 is cyan, 240 is blue, 300 purple, and 360 is red again. # The allowed range is 0 to 359. -#HTML_COLORSTYLE_HUE = 120 +HTML_COLORSTYLE_HUE = 120 # The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of # the colors in the HTML output. For a value of 0 the output will use diff --git a/doc/Makefile b/doc/Makefile index 10f03cc8cb..1c205a5236 100644 --- a/doc/Makefile +++ b/doc/Makefile @@ -1,76 +1,39 @@ -LIBRARIES-$(CONFIG_AVUTIL) += libavutil -LIBRARIES-$(CONFIG_SWSCALE) += libswscale -LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample -LIBRARIES-$(CONFIG_AVCODEC) += libavcodec -LIBRARIES-$(CONFIG_AVFORMAT) += libavformat -LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice -LIBRARIES-$(CONFIG_AVFILTER) += libavfilter - -COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils -COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler -COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler -COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters -COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols -COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices -COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters - -MANPAGES1 = $(AVPROGS-yes:%=doc/%.1) $(AVPROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1) -MANPAGES3 = $(LIBRARIES-yes:%=doc/%.3) -MANPAGES = $(MANPAGES1) $(MANPAGES3) -PODPAGES = $(AVPROGS-yes:%=doc/%.pod) $(AVPROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod) -HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \ +ALLMANPAGES = $(AVBASENAMES:%=%.1) +MANPAGES = $(AVPROGS-yes:%=doc/%.1) +PODPAGES = $(AVPROGS-yes:%=doc/%.pod) +HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) \ doc/developer.html \ doc/faq.html \ doc/fate.html \ doc/general.html \ doc/git-howto.html \ + doc/libavfilter.html \ doc/nut.html \ doc/platform.html \ -TXTPAGES = doc/fate.txt \ - - -DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES) -DOCS-$(CONFIG_PODPAGES) += $(PODPAGES) -DOCS-$(CONFIG_MANPAGES) += $(MANPAGES) -DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES) +DOCS-$(CONFIG_POD2MAN) += $(MANPAGES) $(PODPAGES) +DOCS-$(CONFIG_TEXI2HTML) += $(HTMLPAGES) DOCS = $(DOCS-yes) -DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading -DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec -DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding -DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio -DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio -DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video -DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata -DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing -DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing -DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio -DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video -DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac -DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding -ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes) +DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec +DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio +DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata +DOC_EXAMPLES-$(CONFIG_OUTPUT_EXAMPLE) += output +DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac +ALL_DOC_EXAMPLES = avcodec filter_audio metadata output transcode_aac -DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF)) -ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF)) -ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF)) -PROGS += $(DOC_EXAMPLES) +DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(EXESUF)) +ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES:%=doc/examples/%$(EXESUF)) +PROGS += $(DOC_EXAMPLES) -all-$(CONFIG_DOC): doc - -doc: documentation +all: $(DOCS) apidoc: doc/doxy/html documentation: $(DOCS) examples: $(DOC_EXAMPLES) -TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d) - -doc/%.txt: TAG = TXT -doc/%.txt: doc/%.texi - $(Q)$(TEXIDEP) - $(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null +TEXIDEP = awk '/^@include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d) GENTEXI = format codec GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi) @@ -82,87 +45,53 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF) doc/%.html: TAG = HTML doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI) $(Q)$(TEXIDEP) - $(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $< - -doc/%-all.html: TAG = HTML -doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI) - $(Q)$(TEXIDEP) - $(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $< + $(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $< doc/%.pod: TAG = POD doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI) $(Q)$(TEXIDEP) - $(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@ + $(M)$(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@ -doc/%-all.pod: TAG = POD -doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI) - $(Q)$(TEXIDEP) - $(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@ - -doc/%.1 doc/%.3: TAG = MAN +doc/%.1: TAG = MAN doc/%.1: doc/%.pod $(GENTEXI) $(M)pod2man --section=1 --center=" " --release=" " $< > $@ -doc/%.3: doc/%.pod $(GENTEXI) - $(M)pod2man --section=3 --center=" " --release=" " $< > $@ $(DOCS) doc/doxy/html: | doc/ $(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples OBJDIRS += doc/examples DOXY_INPUT = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)) +DOXY_TEMPLATES = doxy_stylesheet.css footer.html header.html +DOXY_TEMPLATES := $(addprefix $(SRC_PATH)/doc/doxy/, $(DOXY_TEMPLATES)) -doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(DOXY_INPUT) +doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(DOXY_INPUT) $(DOXY_TEMPLATES) $(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXY_INPUT) -install-doc: install-html install-man +install-progs-$(CONFIG_POD2MAN): install-man +install-progs-$(CONFIG_TEXI2HTML): install-doc -install-html: - -install-man: - -ifdef CONFIG_HTMLPAGES -install-progs-$(CONFIG_DOC): install-html - -install-html: $(HTMLPAGES) +install-doc: $(HTMLPAGES) $(Q)mkdir -p "$(DOCDIR)" $(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)" -endif - -ifdef CONFIG_MANPAGES -install-progs-$(CONFIG_DOC): install-man install-man: $(MANPAGES) $(Q)mkdir -p "$(MANDIR)/man1" - $(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1" - $(Q)mkdir -p "$(MANDIR)/man3" - $(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3" -endif + $(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1" -uninstall: uninstall-doc +uninstall: uninstall-doc uninstall-man -uninstall-doc: uninstall-html uninstall-man - -uninstall-html: +uninstall-doc: $(RM) -r "$(DOCDIR)" uninstall-man: - $(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1)) - $(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3)) + $(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES)) -clean:: docclean - -distclean:: docclean - $(RM) doc/config.texi - -examplesclean: - $(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G) - $(RM) $(CLEANSUFFIXES:%=doc/examples/%) - -docclean: examplesclean - $(RM) $(CLEANSUFFIXES:%=doc/%) - $(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi +clean:: + $(RM) $(ALL_DOC_EXAMPLES) + $(RM) $(CLEANSUFFIXES:%=doc/%) $(CLEANSUFFIXES:%=doc/examples/%) + $(RM) doc/*.html doc/*.pod doc/*.1 doc/avoptions_*.texi $(RM) -r doc/doxy/html -include $(wildcard $(DOCS:%=%.d)) -.PHONY: apidoc doc documentation +.PHONY: apidoc documentation diff --git a/doc/RELEASE_NOTES b/doc/RELEASE_NOTES index b3338b9fee..478bc1d20e 100644 --- a/doc/RELEASE_NOTES +++ b/doc/RELEASE_NOTES @@ -1,16 +1,84 @@ Release Notes ============= -* 2.2 "Muybridge" March, 2014 - +* 10 "Eks" General notes ------------- -See the Changelog file for a list of significant changes. Note, there -are many more new features and bugfixes than whats listed there. -Bugreports against FFmpeg git master or the most recent FFmpeg release are -accepted. If you are experiencing issues with any formally released version of -FFmpeg, please try git master to check if the issue still exists. If it does, -make your report against the development code following the usual bug reporting +One of the main features of this release is the addition of reference-counted +data buffers to Libav and their use in various structures. Specifically, the +data buffers used by AVPacket and AVFrame can now be reference counted, which +should allow to significantly simplify many use cases. In addition, +reference-counted AVFrames can now be used in libavfilter, avoiding the need +for a separate libavfilter-specific frame structure. Frames can now be passed +straight from the decoders into filters or from filters to encoders. + +These additions made it necessary to bump the major versions of libavcodec, +libavformat, libavdevice, libavfilter, and libavutil, which was accompanied by +dropping some old deprecated APIs. These libraries are thus not ABI- or API- +compatible with the previous release. All the other libraries (libavresample +and libswscale) should be both ABI- and API-compatible. + +Another major point is the inclusion of the HEVC (AKA H.265, the successor of +H.264) decoder in the main codebase. It was started in 2012 as a Libav Google +Summer of Code project by Guillaume Martres and subsequently completed with +the assistance of the OpenHEVC project and several Libav developers. + +As usual, this release also contains support for other new formats, many smaller +new features and countless bug fixes. We can highlight a native VP9 decoder, +with encoding provided through libvpx, native decoders for WebP, JPEG 2000, and +AIC, as well as improved WavPack support with encoding through libwavpack, +support for more AAC flavors (LD - low delay, ELD - enhanced low delay), slice +multithreading in libavfilter, or muxing chapters in ASF. Furthermore a few new +filters have been introduced, namely compand, to change audio dynamics, framepack, +to create stereoscopic videos, asetpts, to set audio pts, and interlace, to convert +progressive video to interlaced. Finally there is more fine-grained detection of +host and target libc, which should allow better portability to various cross +compilation scenarios. + +See the Changelog file for a fuller list of significant changes. + +Please note that our policy on bug reports has not changed. We still only accept +bug reports against HEAD of the Libav trunk repository. If you are experiencing +issues with any formally released version of Libav, please try a current version +of the development code to check if the issue still exists. If it does, make +your report against the development code following the usual bug reporting guidelines. + + +API changes +----------- + +A number of additional APIs have been introduced and some existing functions +have been deprecated and are scheduled for removal in the next release. +Significant API changes include: + +[libavutil] ++ added the reference-counted buffers API (buffers.h) ++ moved the AVFrame struct to libavutil and added a new API for working with + reference-counted AVFrames (frame.h) + +[libavcodec] ++ added an API for working with reference-counted AVPackets (av_packet_*) ++- converted VDPAU to the hwaccel framework; the old way of using VDPAU is no + longer supported +- old audio encoding and decoding APIs removed +- old video encoding API removed +- deprecated enum CodecID removed (enum AVCodecID should be used instead) +- deprecated audio resampling API removed (libavresample should be used + instead) + +[libavfilter] ++- replaced AVFilterBufferRef with AVFrame; AVFilterBufferRef and everything + related to it still exists, but is deprecated ++ converted all filters to use the AVOptions system for configuration, it is + now possible to query the supported options, their values and set them + directly with av_opt_* ++ added a slice multithreading framework ++- merged avfiltergraph.h to avfilter.h, using AVFilterGraph is now explicitly + mandatory (it was implicitly required even before); added new API for + allocating and initializing filters + +Please see the file doc/APIchanges for details along with similar +programmer-centric information. diff --git a/doc/authors.texi b/doc/authors.texi deleted file mode 100644 index 6c8c1d7efa..0000000000 --- a/doc/authors.texi +++ /dev/null @@ -1,11 +0,0 @@ -@chapter Authors - -The FFmpeg developers. - -For details about the authorship, see the Git history of the project -(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command -@command{git log} in the FFmpeg source directory, or browsing the -online repository at @url{http://source.ffmpeg.org}. - -Maintainers for the specific components are listed in the file -@file{MAINTAINERS} in the source code tree. diff --git a/doc/ffmpeg.texi b/doc/avconv.texi similarity index 57% rename from doc/ffmpeg.texi rename to doc/avconv.texi index 0f6c093493..6efec55978 100644 --- a/doc/ffmpeg.texi +++ b/doc/avconv.texi @@ -1,8 +1,8 @@ \input texinfo @c -*- texinfo -*- -@settitle ffmpeg Documentation +@settitle avconv Documentation @titlepage -@center @titlefont{ffmpeg Documentation} +@center @titlefont{avconv Documentation} @end titlepage @top @@ -11,31 +11,37 @@ @chapter Synopsis -ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ... +The generic syntax is: + +@example +@c man begin SYNOPSIS +avconv [global options] [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}... +@c man end +@end example @chapter Description @c man begin DESCRIPTION -@command{ffmpeg} is a very fast video and audio converter that can also grab from +avconv is a very fast video and audio converter that can also grab from a live audio/video source. It can also convert between arbitrary sample rates and resize video on the fly with a high quality polyphase filter. -@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular +avconv reads from an arbitrary number of input "files" (which can be regular files, pipes, network streams, grabbing devices, etc.), specified by the @code{-i} option, and writes to an arbitrary number of output "files", which are specified by a plain output filename. Anything found on the command line which cannot be interpreted as an option is considered to be an output filename. -Each input or output file can, in principle, contain any number of streams of -different types (video/audio/subtitle/attachment/data). The allowed number and/or -types of streams may be limited by the container format. Selecting which -streams from which inputs will go into which output is either done automatically -or with the @code{-map} option (see the Stream selection chapter). +Each input or output file can in principle contain any number of streams of +different types (video/audio/subtitle/attachment/data). Allowed number and/or +types of streams can be limited by the container format. Selecting, which +streams from which inputs go into output, is done either automatically or with +the @code{-map} option (see the Stream selection chapter). To refer to input files in options, you must use their indices (0-based). E.g. -the first input file is @code{0}, the second is @code{1}, etc. Similarly, streams +the first input file is @code{0}, the second is @code{1} etc. Similarly, streams within a file are referred to by their indices. E.g. @code{2:3} refers to the -fourth stream in the third input file. Also see the Stream specifiers chapter. +fourth stream in the third input file. See also the Stream specifiers chapter. As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same @@ -50,22 +56,22 @@ options apply ONLY to the next input or output file and are reset between files. @itemize @item -To set the video bitrate of the output file to 64 kbit/s: +To set the video bitrate of the output file to 64kbit/s: @example -ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi +avconv -i input.avi -b 64k output.avi @end example @item To force the frame rate of the output file to 24 fps: @example -ffmpeg -i input.avi -r 24 output.avi +avconv -i input.avi -r 24 output.avi @end example @item To force the frame rate of the input file (valid for raw formats only) to 1 fps and the frame rate of the output file to 24 fps: @example -ffmpeg -r 1 -i input.m2v -r 24 output.avi +avconv -r 1 -i input.m2v -r 24 output.avi @end example @end itemize @@ -76,7 +82,7 @@ The format option may be needed for raw input files. @chapter Detailed description @c man begin DETAILED DESCRIPTION -The transcoding process in @command{ffmpeg} for each output can be described by +The transcoding process in @command{avconv} for each output can be described by the following diagram: @example @@ -100,22 +106,22 @@ the following diagram: @end example -@command{ffmpeg} calls the libavformat library (containing demuxers) to read +@command{avconv} calls the libavformat library (containing demuxers) to read input files and get packets containing encoded data from them. When there are -multiple input files, @command{ffmpeg} tries to keep them synchronized by +multiple input files, @command{avconv} tries to keep them synchronized by tracking lowest timestamp on any active input stream. Encoded packets are then passed to the decoder (unless streamcopy is selected for the stream, see further for a description). The decoder produces uncompressed frames (raw video/PCM audio/...) which can be processed further by -filtering (see next section). After filtering, the frames are passed to the -encoder, which encodes them and outputs encoded packets. Finally those are +filtering (see next section). After filtering the frames are passed to the +encoder, which encodes them and outputs encoded packets again. Finally those are passed to the muxer, which writes the encoded packets to the output file. @section Filtering -Before encoding, @command{ffmpeg} can process raw audio and video frames using +Before encoding, @command{avconv} can process raw audio and video frames using filters from the libavfilter library. Several chained filters form a filter -graph. @command{ffmpeg} distinguishes between two types of filtergraphs: +graph. @command{avconv} distinguishes between two types of filtergraphs - simple and complex. @subsection Simple filtergraphs @@ -127,11 +133,11 @@ an additional step between decoding and encoding: _________ ______________ | | | | | decoded | | encoded data | -| frames |\ _ | packets | -|_________| \ /||______________| +| frames |\ /| packets | +|_________| \ / |______________| \ __________ / - simple _\|| | / encoder - filtergraph | filtered |/ + simple \ | | / encoder + filtergraph \| filtered |/ | frames | |__________| @@ -156,7 +162,7 @@ only sets timestamps and otherwise passes the frames unchanged. @subsection Complex filtergraphs Complex filtergraphs are those which cannot be described as simply a linear -processing chain applied to one stream. This is the case, for example, when the graph has +processing chain applied to one stream. This is the case e.g. when the graph has more than one input and/or output, or when output stream type is different from input. They can be represented with the following diagram: @@ -181,21 +187,19 @@ input. They can be represented with the following diagram: @end example Complex filtergraphs are configured with the @option{-filter_complex} option. -Note that this option is global, since a complex filtergraph, by its nature, +Note that this option is global, since a complex filtergraph by its nature cannot be unambiguously associated with a single stream or file. -The @option{-lavfi} option is equivalent to @option{-filter_complex}. - A trivial example of a complex filtergraph is the @code{overlay} filter, which has two video inputs and one video output, containing one video overlaid on top of the other. Its audio counterpart is the @code{amix} filter. @section Stream copy Stream copy is a mode selected by supplying the @code{copy} parameter to the -@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding +@option{-codec} option. It makes @command{avconv} omit the decoding and encoding step for the specified stream, so it does only demuxing and muxing. It is useful for changing the container format or modifying container-level metadata. The -diagram above will, in this case, simplify to this: +diagram above will in this case simplify to this: @example _______ ______________ ________ @@ -207,7 +211,7 @@ diagram above will, in this case, simplify to this: @end example Since there is no decoding or encoding, it is very fast and there is no quality -loss. However, it might not work in some cases because of many factors. Applying +loss. However it might not work in some cases because of many factors. Applying filters is obviously also impossible, since filters work on uncompressed data. @c man end DETAILED DESCRIPTION @@ -215,14 +219,12 @@ filters is obviously also impossible, since filters work on uncompressed data. @chapter Stream selection @c man begin STREAM SELECTION -By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle) -present in the input files and adds them to each output file. It picks the -"best" of each based upon the following criteria: for video, it is the stream -with the highest resolution, for audio, it is the stream with the most channels, for -subtitles, it is the first subtitle stream. In the case where several streams of -the same type rate equally, the stream with the lowest index is chosen. +By default avconv tries to pick the "best" stream of each type present in input +files and add them to each output file. For video, this means the highest +resolution, for audio the highest channel count. For subtitle it's simply the +first subtitle stream. -You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For +You can disable some of those defaults by using @code{-vn/-an/-sn} options. For full manual control, use the @code{-map} option, which disables the defaults just described. @@ -231,15 +233,15 @@ described. @chapter Options @c man begin OPTIONS -@include fftools-common-opts.texi +@include avtools-common-opts.texi @section Main options @table @option @item -f @var{fmt} (@emph{input/output}) -Force input or output file format. The format is normally auto detected for input -files and guessed from the file extension for output files, so this option is not +Force input or output file format. The format is normally autodetected for input +files and guessed from file extension for output files, so this option is not needed in most cases. @item -i @var{filename} (@emph{input}) @@ -249,53 +251,39 @@ input file name Overwrite output files without asking. @item -n (@emph{global}) -Do not overwrite output files, and exit immediately if a specified -output file already exists. +Immediately exit when output files already exist. @item -c[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream}) @itemx -codec[:@var{stream_specifier}] @var{codec} (@emph{input/output,per-stream}) Select an encoder (when used before an output file) or a decoder (when used before an input file) for one or more streams. @var{codec} is the name of a decoder/encoder or a special value @code{copy} (output only) to indicate that -the stream is not to be re-encoded. +the stream is not to be reencoded. For example @example -ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT +avconv -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT @end example encodes all video streams with libx264 and copies all audio streams. For each stream, the last matching @code{c} option is applied, so @example -ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT +avconv -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT @end example will copy all the streams except the second video, which will be encoded with libx264, and the 138th audio, which will be encoded with libvorbis. -@item -t @var{duration} (@emph{input/output}) -When used as an input option (before @code{-i}), limit the @var{duration} of -data read from the input file. - -When used as an output option (before an output filename), stop writing the -output after its duration reaches @var{duration}. - +@item -t @var{duration} (@emph{output}) +Stop writing the output after its duration reaches @var{duration}. @var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form. --to and -t are mutually exclusive and -t has priority. - -@item -to @var{position} (@emph{output}) -Stop writing the output at @var{position}. -@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form. - --to and -t are mutually exclusive and -t has priority. - @item -fs @var{limit_size} (@emph{output}) -Set the file size limit, expressed in bytes. +Set the file size limit. @item -ss @var{position} (@emph{input/output}) When used as an input option (before @code{-i}), seeks in this input file to @var{position}. Note the in most formats it is not possible to seek exactly, so -@command{ffmpeg} will seek to the closest seek point before @var{position}. +@command{avconv} will seek to the closest seek point before @var{position}. When transcoding and @option{-accurate_seek} is enabled (the default), this extra segment between the seek point and @var{position} will be decoded and discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it @@ -307,20 +295,11 @@ input until the timestamps reach @var{position}. @var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form. @item -itsoffset @var{offset} (@emph{input}) -Set the input time offset. - -@var{offset} must be a time duration specification, -see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. - -The offset is added to the timestamps of the input files. Specifying -a positive offset means that the corresponding streams are delayed by -the time duration specified in @var{offset}. - -@item -timestamp @var{date} (@emph{output}) -Set the recording timestamp in the container. - -@var{date} must be a time duration specification, -see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}. +Set the input time offset in seconds. +@code{[-]hh:mm:ss[.xxx]} syntax is also supported. +The offset is added to the timestamps of the input files. +Specifying a positive offset means that the corresponding +streams are delayed by @var{offset} seconds. @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata}) Set a metadata key/value pair. @@ -334,12 +313,12 @@ also possible to delete metadata by using an empty value. For example, for setting the title in the output file: @example -ffmpeg -i in.avi -metadata title="my title" out.flv +avconv -i in.avi -metadata title="my title" out.flv @end example To set the language of the first audio stream: @example -ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT +avconv -i INPUT -metadata:s:a:0 language=eng OUTPUT @end example @item -target @var{type} (@emph{output}) @@ -349,14 +328,14 @@ Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv}, (bitrate, codecs, buffer sizes) are then set automatically. You can just type: @example -ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg +avconv -i myfile.avi -target vcd /tmp/vcd.mpg @end example Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in: @example -ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg +avconv -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg @end example @item -dframes @var{number} (@emph{output}) @@ -367,28 +346,16 @@ Stop writing to the stream after @var{framecount} frames. @item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream}) @itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream}) -Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is +Use fixed quality scale (VBR). The meaning of @var{q} is codec-dependent. -If @var{qscale} is used without a @var{stream_specifier} then it applies only -to the video stream, this is to maintain compatibility with previous behavior -and as specifying the same codec specific value to 2 different codecs that is -audio and video generally is not what is intended when no stream_specifier is -used. -@anchor{filter_option} -@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream}) -Create the filtergraph specified by @var{filtergraph} and use it to -filter the stream. +@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream}) +@var{filter_graph} is a description of the filter graph to apply to +the stream. Use @code{-filters} to show all the available filters +(including also sources and sinks). -@var{filtergraph} is a description of the filtergraph to apply to -the stream, and must have a single input and a single output of the -same type of the stream. In the filtergraph, the input is associated -to the label @code{in}, and the output to the label @code{out}. See -the ffmpeg-filters manual for more information about the filtergraph -syntax. - -See the @ref{filter_complex_option,,-filter_complex option} if you -want to create filtergraphs with multiple inputs and/or outputs. +See also the @option{-filter_complex} option if you want to create filter graphs +with multiple inputs and/or outputs. @item -filter_script[:@var{stream_specifier}] @var{filename} (@emph{output,per-stream}) This option is similar to @option{-filter}, the only difference is that its @@ -399,34 +366,7 @@ read. Specify the preset for matching stream(s). @item -stats (@emph{global}) -Print encoding progress/statistics. It is on by default, to explicitly -disable it you need to specify @code{-nostats}. - -@item -progress @var{url} (@emph{global}) -Send program-friendly progress information to @var{url}. - -Progress information is written approximately every second and at the end of -the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key} -consists of only alphanumeric characters. The last key of a sequence of -progress information is always "progress". - -@item -stdin -Enable interaction on standard input. On by default unless standard input is -used as an input. To explicitly disable interaction you need to specify -@code{-nostdin}. - -Disabling interaction on standard input is useful, for example, if -ffmpeg is in the background process group. Roughly the same result can -be achieved with @code{ffmpeg ... < /dev/null} but it requires a -shell. - -@item -debug_ts (@emph{global}) -Print timestamp information. It is off by default. This option is -mostly useful for testing and debugging purposes, and the output -format may change from one version to another, so it should not be -employed by portable scripts. - -See also the option @code{-fdebug ts}. +Print encoding progress/statistics. On by default. @item -attach @var{filename} (@emph{output}) Add an attachment to the output file. This is supported by a few formats @@ -439,7 +379,7 @@ with @code{-map} or automatic mappings). Note that for Matroska you also have to set the mimetype metadata tag: @example -ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv +avconv -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv @end example (assuming that the attachment stream will be third in the output file). @@ -450,11 +390,11 @@ will be used. E.g. to extract the first attachment to a file named 'out.ttf': @example -ffmpeg -dump_attachment:t:0 out.ttf -i INPUT +avconv -dump_attachment:t:0 out.ttf INPUT @end example To extract all attachments to files determined by the @code{filename} tag: @example -ffmpeg -dump_attachment:t "" -i INPUT +avconv -dump_attachment:t "" INPUT @end example Technical note -- attachments are implemented as codec extradata, so this @@ -475,7 +415,8 @@ As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate @var{fps}. As an output option, duplicate or drop input frames to achieve constant output -frame rate @var{fps}. +frame rate @var{fps} (note that this actually causes the @code{fps} filter to be +inserted to the end of the corresponding filtergraph). @item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream}) Set frame size. @@ -488,7 +429,68 @@ As an output option, this inserts the @code{scale} video filter to the @emph{end} of the corresponding filtergraph. Please use the @code{scale} filter directly to insert it at the beginning or some other place. -The format is @samp{wxh} (default - same as source). +The format is @samp{wxh} (default - same as source). The following +abbreviations are recognized: +@table @samp +@item sqcif +128x96 +@item qcif +176x144 +@item cif +352x288 +@item 4cif +704x576 +@item 16cif +1408x1152 +@item qqvga +160x120 +@item qvga +320x240 +@item vga +640x480 +@item svga +800x600 +@item xga +1024x768 +@item uxga +1600x1200 +@item qxga +2048x1536 +@item sxga +1280x1024 +@item qsxga +2560x2048 +@item hsxga +5120x4096 +@item wvga +852x480 +@item wxga +1366x768 +@item wsxga +1600x1024 +@item wuxga +1920x1200 +@item woxga +2560x1600 +@item wqsxga +3200x2048 +@item wquxga +3840x2400 +@item whsxga +6400x4096 +@item whuxga +7680x4800 +@item cga +320x200 +@item ega +640x350 +@item hd480 +852x480 +@item hd720 +1280x720 +@item hd1080 +1920x1080 +@end table @item -aspect[:@var{stream_specifier}] @var{aspect} (@emph{output,per-stream}) Set the video display aspect ratio specified by @var{aspect}. @@ -498,10 +500,6 @@ form @var{num}:@var{den}, where @var{num} and @var{den} are the numerator and denominator of the aspect ratio. For example "4:3", "16:9", "1.3333", and "1.7777" are valid argument values. -If used together with @option{-vcodec copy}, it will affect the aspect ratio -stored at container level, but not the aspect ratio stored in encoded -frames, if it exists. - @item -vn (@emph{output}) Disable video recording. @@ -517,21 +515,22 @@ at the exact requested bitrate. On pass 1, you may just deactivate audio and set output to null, examples for Windows and Unix: @example -ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL -ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null +avconv -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL +avconv -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null @end example @item -passlogfile[:@var{stream_specifier}] @var{prefix} (@emph{output,per-stream}) Set two-pass log file name prefix to @var{prefix}, the default file name -prefix is ``ffmpeg2pass''. The complete file name will be +prefix is ``av2pass''. The complete file name will be @file{PREFIX-N.log}, where N is a number specific to the output -stream +stream. -@item -vf @var{filtergraph} (@emph{output}) -Create the filtergraph specified by @var{filtergraph} and use it to -filter the stream. +@item -vf @var{filter_graph} (@emph{output}) +@var{filter_graph} is a description of the filter graph to apply to +the input video. +Use the option "-filters" to show all the available filters (including +also sources and sinks). This is an alias for @code{-filter:v}. -This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}. @end table @section Advanced Video Options @@ -540,33 +539,14 @@ This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter optio @item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream}) Set pixel format. Use @code{-pix_fmts} to show all the supported pixel formats. -If the selected pixel format can not be selected, ffmpeg will print a -warning and select the best pixel format supported by the encoder. -If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error -if the requested pixel format can not be selected, and automatic conversions -inside filtergraphs are disabled. -If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format -as the input (or graph output) and automatic conversions are disabled. - @item -sws_flags @var{flags} (@emph{input/output}) Set SwScaler flags. @item -vdt @var{n} Discard threshold. @item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream}) -Rate control override for specific intervals, formatted as "int,int,int" -list separated with slashes. Two first values are the beginning and -end frame numbers, last one is quantizer to use if positive, or quality -factor if negative. +rate control override for specific intervals -@item -ilme -Force interlacing support in encoder (MPEG-2 and MPEG-4 only). -Use this option if your input file is interlaced and you want -to keep the interlaced format for minimum losses. -The alternative is to deinterlace the input stream with -@option{-deinterlace}, but deinterlacing introduces losses. -@item -psnr -Calculate PSNR of compressed frames. @item -vstats Dump video coding statistics to @file{vstats_HHMMSS.log}. @item -vstats_file @var{file} @@ -578,61 +558,13 @@ Intra_dc_precision. @item -vtag @var{fourcc/tag} (@emph{output}) Force video tag/fourcc. This is an alias for @code{-tag:v}. @item -qphist (@emph{global}) -Show QP histogram -@item -vbsf @var{bitstream_filter} -Deprecated see -bsf - +Show QP histogram. @item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream}) -@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream}) Force key frames at the specified timestamps, more precisely at the first frames after each specified time. - -If the argument is prefixed with @code{expr:}, the string @var{expr} -is interpreted like an expression and is evaluated for each frame. A -key frame is forced in case the evaluation is non-zero. - -If one of the times is "@code{chapters}[@var{delta}]", it is expanded into -the time of the beginning of all chapters in the file, shifted by -@var{delta}, expressed as a time in seconds. This option can be useful to ensure that a seek point is present at a chapter mark or any other designated place in the output file. - -For example, to insert a key frame at 5 minutes, plus key frames 0.1 second -before the beginning of every chapter: -@example --force_key_frames 0:05:00,chapters-0.1 -@end example - -The expression in @var{expr} can contain the following constants: -@table @option -@item n -the number of current processed frame, starting from 0 -@item n_forced -the number of forced frames -@item prev_forced_n -the number of the previous forced frame, it is @code{NAN} when no -keyframe was forced yet -@item prev_forced_t -the time of the previous forced frame, it is @code{NAN} when no -keyframe was forced yet -@item t -the time of the current processed frame -@end table - -For example to force a key frame every 5 seconds, you can specify: -@example --force_key_frames expr:gte(t,n_forced*5) -@end example - -To force a key frame 5 seconds after the time of the last forced one, -starting from second 13: -@example --force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5)) -@end example - -Note that forcing too many keyframes is very harmful for the lookahead -algorithms of certain encoders: using fixed-GOP options or similar -would be more efficient. +The timestamps must be specified in ascending order. @item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream}) When doing stream copy, copy also non-key frames found at the @@ -662,7 +594,7 @@ This option has no effect if the selected hwaccel is not available or not supported by the chosen decoder. Note that most acceleration methods are intended for playback and will not be -faster than software decoding on modern CPUs. Additionally, @command{ffmpeg} +faster than software decoding on modern CPUs. Additionally, @command{avconv} will usually need to copy the decoded frames from the GPU memory into the system memory, resulting in further performance loss. This option is thus mainly useful for testing. @@ -709,12 +641,11 @@ Set the audio codec. This is an alias for @code{-codec:a}. @item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream}) Set the audio sample format. Use @code{-sample_fmts} to get a list of supported sample formats. - -@item -af @var{filtergraph} (@emph{output}) -Create the filtergraph specified by @var{filtergraph} and use it to -filter the stream. - -This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}. +@item -af @var{filter_graph} (@emph{output}) +@var{filter_graph} is a description of the filter graph to apply to +the input audio. +Use the option "-filters" to show all the available filters (including +also sources and sinks). This is an alias for @code{-filter:a}. @end table @section Advanced Audio options: @@ -722,14 +653,6 @@ This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter optio @table @option @item -atag @var{fourcc/tag} (@emph{output}) Force audio tag/fourcc. This is an alias for @code{-tag:a}. -@item -absf @var{bitstream_filter} -Deprecated, see -bsf -@item -guess_layout_max @var{channels} (@emph{input,per-stream}) -If some input channel layout is not known, try to guess only if it -corresponds to at most the specified number of channels. For example, 2 -tells to @command{ffmpeg} to recognize 1 channel as mono and 2 channels as -stereo but not 6 channels as 5.1. The default is to always try to guess. Use -0 to disable all guessing. @end table @section Subtitle options: @@ -739,30 +662,6 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use Set the subtitle codec. This is an alias for @code{-codec:s}. @item -sn (@emph{output}) Disable subtitle recording. -@item -sbsf @var{bitstream_filter} -Deprecated, see -bsf -@end table - -@section Advanced Subtitle options: - -@table @option - -@item -fix_sub_duration -Fix subtitles durations. For each subtitle, wait for the next packet in the -same stream and adjust the duration of the first to avoid overlap. This is -necessary with some subtitles codecs, especially DVB subtitles, because the -duration in the original packet is only a rough estimate and the end is -actually marked by an empty subtitle frame. Failing to use this option when -necessary can result in exaggerated durations or muxing failures due to -non-monotonic timestamps. - -Note that this option will delay the output of all data until the next -subtitle packet is decoded: it may increase memory consumption and latency a -lot. - -@item -canvas_size @var{size} -Set the size of the canvas used to render subtitles. - @end table @section Advanced options @@ -790,7 +689,7 @@ graphs (see the @option{-filter_complex} option) to the output file. For example, to map ALL streams from the first input file to output @example -ffmpeg -i INPUT -map 0 output +avconv -i INPUT -map 0 output @end example For example, if you have two audio streams in the first input file, @@ -798,7 +697,7 @@ these streams are identified by "0:0" and "0:1". You can use @code{-map} to select which streams to place in an output file. For example: @example -ffmpeg -i INPUT -map 0:1 out.wav +avconv -i INPUT -map 0:1 out.wav @end example will map the input stream in @file{INPUT} identified by "0:1" to the (single) output stream in @file{out.wav}. @@ -808,77 +707,21 @@ For example, to select the stream with index 2 from input file index 6 from input @file{b.mov} (specified by the identifier "1:6"), and copy them to the output file @file{out.mov}: @example -ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov +avconv -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov @end example To select all video and the third audio stream from an input file: @example -ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT +avconv -i INPUT -map 0:v -map 0:a:2 OUTPUT @end example To map all the streams except the second audio, use negative mappings @example -ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT +avconv -i INPUT -map 0 -map -0:a:1 OUTPUT @end example Note that using this option disables the default mappings for this output file. -@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}] -Map an audio channel from a given input to an output. If -@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will -be mapped on all the audio streams. - -Using "-1" instead of -@var{input_file_id}.@var{stream_specifier}.@var{channel_id} will map a muted -channel. - -For example, assuming @var{INPUT} is a stereo audio file, you can switch the -two audio channels with the following command: -@example -ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT -@end example - -If you want to mute the first channel and keep the second: -@example -ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT -@end example - -The order of the "-map_channel" option specifies the order of the channels in -the output stream. The output channel layout is guessed from the number of -channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac" -in combination of "-map_channel" makes the channel gain levels to be updated if -input and output channel layouts don't match (for instance two "-map_channel" -options and "-ac 6"). - -You can also extract each channel of an input to specific outputs; the following -command extracts two channels of the @var{INPUT} audio stream (file 0, stream 0) -to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1} outputs: -@example -ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1 -@end example - -The following example splits the channels of a stereo input into two separate -streams, which are put into the same output file: -@example -ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg -@end example - -Note that currently each output stream can only contain channels from a single -input stream; you can't for example use "-map_channel" to pick multiple input -audio channels contained in different streams (from the same or different files) -and merge them into a single output stream. It is therefore not currently -possible, for example, to turn two separate mono streams into a single stereo -stream. However splitting a stereo stream into two single channel mono streams -is possible. - -If you need this feature, a possible workaround is to use the @emph{amerge} -filter. For example, if you need to merge a media (here @file{input.mkv}) with 2 -mono audio streams into one single stereo channel audio stream (and keep the -video stream), you can use the following command: -@example -ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv -@end example - @item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata}) Set metadata information of the next output file from @var{infile}. Note that those are file indices (zero-based), not filenames. @@ -910,12 +753,12 @@ file index can be used to create a dummy mapping that just disables automatic co For example to copy metadata from the first stream of the input file to global metadata of the output file: @example -ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3 +avconv -i in.ogg -map_metadata 0:s:0 out.mp3 @end example To do the reverse, i.e. copy global metadata to all audio streams: @example -ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv +avconv -i in.mkv -map_metadata:s:a 0:g out.mkv @end example Note that simple @code{0} would work as well in this example, since global metadata is assumed by default. @@ -925,63 +768,41 @@ Copy chapters from input file with index @var{input_file_index} to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying. - +@item -debug +Print specific debug info. @item -benchmark (@emph{global}) Show benchmarking information at the end of an encode. Shows CPU time used and maximum memory consumption. Maximum memory consumption is not supported on all systems, it will usually display as 0 if not supported. -@item -benchmark_all (@emph{global}) -Show benchmarking information during the encode. -Shows CPU time used in various steps (audio/video encode/decode). @item -timelimit @var{duration} (@emph{global}) -Exit after ffmpeg has been running for @var{duration} seconds. +Exit after avconv has been running for @var{duration} seconds. @item -dump (@emph{global}) Dump each input packet to stderr. @item -hex (@emph{global}) When dumping packets, also dump the payload. @item -re (@emph{input}) -Read input at native frame rate. Mainly used to simulate a grab device. +Read input at native frame rate. Mainly used to simulate a grab device or live input stream (e.g. when reading from a file). Should not be used with actual grab devices or live input streams (where it can cause packet loss). -By default @command{ffmpeg} attempts to read the input(s) as fast as possible. -This option will slow down the reading of the input(s) to the native frame rate -of the input(s). It is useful for real-time output (e.g. live streaming). -@item -loop_input -Loop over the input stream. Currently it works only for image -streams. This option is used for automatic FFserver testing. -This option is deprecated, use -loop 1. -@item -loop_output @var{number_of_times} -Repeatedly loop output for formats that support looping such as animated GIF -(0 will loop the output infinitely). -This option is deprecated, use -loop. @item -vsync @var{parameter} Video sync method. -For compatibility reasons old values can be specified as numbers. -Newly added values will have to be specified as strings always. @table @option -@item 0, passthrough +@item passthrough Each frame is passed with its timestamp from the demuxer to the muxer. -@item 1, cfr +@item cfr Frames will be duplicated and dropped to achieve exactly the requested -constant frame rate. -@item 2, vfr +constant framerate. +@item vfr Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp. -@item drop -As passthrough but destroys all timestamps, making the muxer generate -fresh timestamps based on frame-rate. -@item -1, auto +@item auto Chooses between 1 and 2 depending on muxer capabilities. This is the default method. @end table -Note that the timestamps may be further modified by the muxer, after this. -For example, in the case that the format option @option{avoid_negative_ts} -is enabled. - With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one. @@ -991,47 +812,11 @@ Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps the parameter is the maximum samples per second by which the audio is changed. -async 1 is a special case where only the start of the audio stream is corrected without any later correction. - -Note that the timestamps may be further modified by the muxer, after this. -For example, in the case that the format option @option{avoid_negative_ts} -is enabled. - -This option has been deprecated. Use the @code{aresample} audio filter instead. - +This option has been deprecated. Use the @code{asyncts} audio filter instead. @item -copyts -Do not process input timestamps, but keep their values without trying -to sanitize them. In particular, do not remove the initial start time -offset value. - -Note that, depending on the @option{vsync} option or on specific muxer -processing (e.g. in case the format option @option{avoid_negative_ts} -is enabled) the output timestamps may mismatch with the input -timestamps even when this option is selected. - -@item -copytb @var{mode} -Specify how to set the encoder timebase when stream copying. @var{mode} is an -integer numeric value, and can assume one of the following values: - -@table @option -@item 1 -Use the demuxer timebase. - -The time base is copied to the output encoder from the corresponding input -demuxer. This is sometimes required to avoid non monotonically increasing -timestamps when copying video streams with variable frame rate. - -@item 0 -Use the decoder timebase. - -The time base is copied to the output encoder from the corresponding input -decoder. - -@item -1 -Try to make the choice automatically, in order to generate a sane output. -@end table - -Default value is -1. - +Copy timestamps from input to output. +@item -copytb +Copy input stream time base from input to output when stream copying. @item -shortest (@emph{output}) Finish encoding when the shortest input stream ends. @item -dts_delta_threshold @@ -1049,37 +834,28 @@ may be reassigned to a different value. For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for an output mpegts file: @example -ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts +avconv -i infile -streamid 0:33 -streamid 1:36 out.ts @end example @item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream}) -Set bitstream filters for matching streams. @var{bitstream_filters} is +Set bitstream filters for matching streams. @var{bistream_filters} is a comma-separated list of bitstream filters. Use the @code{-bsfs} option to get the list of bitstream filters. @example -ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 +avconv -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 @end example @example -ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt +avconv -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt @end example @item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream}) Force a tag/fourcc for matching streams. -@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff} -Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';' -(or '.') for drop. -@example -ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg -@end example - -@anchor{filter_complex_option} @item -filter_complex @var{filtergraph} (@emph{global}) -Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or +Define a complex filter graph, i.e. one with arbitrary number of inputs and/or outputs. For simple graphs -- those with one input and one output of the same type -- see the @option{-filter} options. @var{filtergraph} is a description of -the filtergraph, as described in the ``Filtergraph syntax'' section of the -ffmpeg-filters manual. +the filter graph, as described in @ref{Filtergraph syntax}. Input link labels must refer to input streams using the @code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map} @@ -1095,7 +871,7 @@ normal input files. For example, to overlay an image over video @example -ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map +avconv -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map '[out]' out.mkv @end example Here @code{[0:v]} refers to the first video stream in the first input file, @@ -1106,25 +882,21 @@ of overlay. Assuming there is only one video stream in each input file, we can omit input labels, so the above is equivalent to @example -ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map +avconv -i video.mkv -i image.png -filter_complex 'overlay[out]' -map '[out]' out.mkv @end example Furthermore we can omit the output label and the single output from the filter graph will be added to the output file automatically, so we can simply write @example -ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv +avconv -i video.mkv -i image.png -filter_complex 'overlay' out.mkv @end example To generate 5 seconds of pure red video using lavfi @code{color} source: @example -ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv +avconv -filter_complex 'color=red' -t 5 out.mkv @end example -@item -lavfi @var{filtergraph} (@emph{global}) -Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or -outputs. Equivalent to @option{-filter_complex}. - @item -filter_complex_script @var{filename} (@emph{global}) This option is similar to @option{-filter_complex}, the only difference is that its argument is the name of the file from which a complex filtergraph @@ -1136,66 +908,7 @@ This option enables or disables accurate seeking in input files with the transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful e.g. when copying some streams and transcoding the others. -@item -override_ffserver (@emph{global}) -Overrides the input specifications from @command{ffserver}. Using this -option you can map any input stream to @command{ffserver} and control -many aspects of the encoding from @command{ffmpeg}. Without this -option @command{ffmpeg} will transmit to @command{ffserver} what is -requested by @command{ffserver}. - -The option is intended for cases where features are needed that cannot be -specified to @command{ffserver} but can be to @command{ffmpeg}. - @end table - -As a special exception, you can use a bitmap subtitle stream as input: it -will be converted into a video with the same size as the largest video in -the file, or 720x576 if no video is present. Note that this is an -experimental and temporary solution. It will be removed once libavfilter has -proper support for subtitles. - -For example, to hardcode subtitles on top of a DVB-T recording stored in -MPEG-TS format, delaying the subtitles by 1 second: -@example -ffmpeg -i input.ts -filter_complex \ - '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \ - -sn -map '#0x2dc' output.mkv -@end example -(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video, -audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too) - -@section Preset files -A preset file contains a sequence of @var{option}=@var{value} pairs, -one for each line, specifying a sequence of options which would be -awkward to specify on the command line. Lines starting with the hash -('#') character are ignored and are used to provide comments. Check -the @file{presets} directory in the FFmpeg source tree for examples. - -Preset files are specified with the @code{vpre}, @code{apre}, -@code{spre}, and @code{fpre} options. The @code{fpre} option takes the -filename of the preset instead of a preset name as input and can be -used for any kind of codec. For the @code{vpre}, @code{apre}, and -@code{spre} options, the options specified in a preset file are -applied to the currently selected codec of the same type as the preset -option. - -The argument passed to the @code{vpre}, @code{apre}, and @code{spre} -preset options identifies the preset file to use according to the -following rules: - -First ffmpeg searches for a file named @var{arg}.ffpreset in the -directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in -the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg}) -or in a @file{ffpresets} folder along the executable on win32, -in that order. For example, if the argument is @code{libvpx-1080p}, it will -search for the file @file{libvpx-1080p.ffpreset}. - -If no such file is found, then ffmpeg will search for a file named -@var{codec_name}-@var{arg}.ffpreset in the above-mentioned -directories, where @var{codec_name} is the name of the codec to which -the preset file options will be applied. For example, if you select -the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p}, -then it will search for the file @file{libvpx-1080p.ffpreset}. @c man end OPTIONS @chapter Tips @@ -1203,13 +916,13 @@ then it will search for the file @file{libvpx-1080p.ffpreset}. @itemize @item -For streaming at very low bitrates, use a low frame rate +For streaming at very low bitrate application, use a low frame rate and a small GOP size. This is especially true for RealVideo where the Linux player does not seem to be very fast, so it can miss frames. An example is: @example -ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm +avconv -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm @end example @item @@ -1248,48 +961,43 @@ A preset file contains a sequence of @var{option=value} pairs, one for each line, specifying a sequence of options which can be specified also on the command line. Lines starting with the hash ('#') character are ignored and are used to provide comments. Empty lines are also ignored. Check the -@file{presets} directory in the FFmpeg source tree for examples. +@file{presets} directory in the Libav source tree for examples. Preset files are specified with the @code{pre} option, this option takes a -preset name as input. FFmpeg searches for a file named @var{preset_name}.avpreset in -the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in -the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg}) +preset name as input. Avconv searches for a file named @var{preset_name}.avpreset in +the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.avconv}, and in +the data directory defined at configuration time (usually @file{$PREFIX/share/avconv}) in that order. For example, if the argument is @code{libx264-max}, it will search for the file @file{libx264-max.avpreset}. @section Video and Audio grabbing -If you specify the input format and device then ffmpeg can grab video +If you specify the input format and device then avconv can grab video and audio directly. @example -ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg -@end example - -Or with an ALSA audio source (mono input, card id 1) instead of OSS: -@example -ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg +avconv -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg @end example Note that you must activate the right video source and channel before -launching ffmpeg with any TV viewer such as +launching avconv with any TV viewer such as @uref{http://linux.bytesex.org/xawtv/, xawtv} by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer. @section X11 grabbing -Grab the X11 display with ffmpeg via +Grab the X11 display with avconv via @example -ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg +avconv -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg @end example 0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. @example -ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg +avconv -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg @end example 0.0 is display.screen number of your X11 server, same as the DISPLAY environment @@ -1297,7 +1005,7 @@ variable. 10 is the x-offset and 20 the y-offset for the grabbing. @section Video and Audio file format conversion -Any supported file format and protocol can serve as input to ffmpeg: +Any supported file format and protocol can serve as input to avconv: Examples: @itemize @@ -1305,7 +1013,7 @@ Examples: You can use YUV files as input: @example -ffmpeg -i /tmp/test%d.Y /tmp/out.mpg +avconv -i /tmp/test%d.Y /tmp/out.mpg @end example It will use the files: @@ -1317,13 +1025,13 @@ It will use the files: The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the @option{-s} option -if ffmpeg cannot guess it. +if avconv cannot guess it. @item You can input from a raw YUV420P file: @example -ffmpeg -i /tmp/test.yuv /tmp/out.avi +avconv -i /tmp/test.yuv /tmp/out.avi @end example test.yuv is a file containing raw YUV planar data. Each frame is composed @@ -1334,14 +1042,14 @@ horizontal resolution. You can output to a raw YUV420P file: @example -ffmpeg -i mydivx.avi hugefile.yuv +avconv -i mydivx.avi hugefile.yuv @end example @item You can set several input files and output files: @example -ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg +avconv -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg @end example Converts the audio file a.wav and the raw YUV video file a.yuv @@ -1351,7 +1059,7 @@ to MPEG file a.mpg. You can also do audio and video conversions at the same time: @example -ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2 +avconv -i /tmp/a.wav -ar 22050 /tmp/a.mp2 @end example Converts a.wav to MPEG audio at 22050 Hz sample rate. @@ -1361,7 +1069,7 @@ You can encode to several formats at the same time and define a mapping from input stream to output streams: @example -ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2 +avconv -i /tmp/a.wav -map 0:a -b 64k /tmp/a.mp2 -map 0:a -b 128k /tmp/b.mp2 @end example Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map @@ -1372,7 +1080,7 @@ stream, in the order of the definition of output streams. You can transcode decrypted VOBs: @example -ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi +avconv -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi @end example This is a typical DVD ripping example; the input is a VOB file, the @@ -1384,14 +1092,14 @@ to enable LAME support by passing @code{--enable-libmp3lame} to configure. The mapping is particularly useful for DVD transcoding to get the desired audio language. -NOTE: To see the supported input formats, use @code{ffmpeg -formats}. +NOTE: To see the supported input formats, use @code{avconv -formats}. @item You can extract images from a video, or create a video from many images: For extracting images from a video: @example -ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg +avconv -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg @end example This will extract one video frame per second from the video and will @@ -1404,7 +1112,7 @@ combination with -ss to start extracting from a certain point in time. For creating a video from many images: @example -ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi +avconv -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi @end example The syntax @code{foo-%03d.jpeg} specifies to use a decimal number @@ -1412,21 +1120,11 @@ composed of three digits padded with zeroes to express the sequence number. It is the same syntax supported by the C printf function, but only formats accepting a normal integer are suitable. -When importing an image sequence, -i also supports expanding -shell-like wildcard patterns (globbing) internally, by selecting the -image2-specific @code{-pattern_type glob} option. - -For example, for creating a video from filenames matching the glob pattern -@code{foo-*.jpeg}: -@example -ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi -@end example - @item You can put many streams of the same type in the output: @example -ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut +avconv -i test1.avi -i test2.avi -map 0.3 -map 0.2 -map 0.1 -map 0.0 -c copy test12.nut @end example The resulting output file @file{test12.avi} will contain first four streams from @@ -1435,86 +1133,43 @@ the input file in reverse order. @item To force CBR video output: @example -ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v +avconv -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v @end example @item The four options lmin, lmax, mblmin and mblmax use 'lambda' units, but you may use the QP2LAMBDA constant to easily convert from 'q' units: @example -ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext +avconv -i src.ext -lmax 21*QP2LAMBDA dst.ext @end example @end itemize @c man end EXAMPLES -@include config.texi -@ifset config-all -@ifset config-avutil -@include utils.texi -@end ifset -@ifset config-avcodec -@include codecs.texi -@include bitstream_filters.texi -@end ifset -@ifset config-avformat -@include formats.texi +@include eval.texi +@include decoders.texi +@include encoders.texi +@include demuxers.texi +@include muxers.texi +@include indevs.texi +@include outdevs.texi @include protocols.texi -@end ifset -@ifset config-avdevice -@include devices.texi -@end ifset -@ifset config-swresample -@include resampler.texi -@end ifset -@ifset config-swscale -@include scaler.texi -@end ifset -@ifset config-avfilter +@include bitstream_filters.texi @include filters.texi -@end ifset -@end ifset - -@chapter See Also - -@ifhtml -@ifset config-all -@url{ffmpeg.html,ffmpeg} -@end ifset -@ifset config-not-all -@url{ffmpeg-all.html,ffmpeg-all}, -@end ifset -@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-utils.html,ffmpeg-utils}, -@url{ffmpeg-scaler.html,ffmpeg-scaler}, -@url{ffmpeg-resampler.html,ffmpeg-resampler}, -@url{ffmpeg-codecs.html,ffmpeg-codecs}, -@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, -@url{ffmpeg-formats.html,ffmpeg-formats}, -@url{ffmpeg-devices.html,ffmpeg-devices}, -@url{ffmpeg-protocols.html,ffmpeg-protocols}, -@url{ffmpeg-filters.html,ffmpeg-filters} -@end ifhtml - -@ifnothtml -@ifset config-all -ffmpeg(1), -@end ifset -@ifset config-not-all -ffmpeg-all(1), -@end ifset -ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), -ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), -ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) -@end ifnothtml - -@include authors.texi +@include metadata.texi @ignore -@setfilename ffmpeg -@settitle ffmpeg video converter +@setfilename avconv +@settitle avconv video converter + +@c man begin SEEALSO +avplay(1), avprobe(1) and the Libav HTML documentation +@c man end + +@c man begin AUTHORS +The Libav developers +@c man end @end ignore diff --git a/doc/avplay.texi b/doc/avplay.texi new file mode 100644 index 0000000000..71981f6d5d --- /dev/null +++ b/doc/avplay.texi @@ -0,0 +1,184 @@ +\input texinfo @c -*- texinfo -*- + +@settitle avplay Documentation +@titlepage +@center @titlefont{avplay Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +@example +@c man begin SYNOPSIS +avplay [options] @file{input_file} +@c man end +@end example + +@chapter Description +@c man begin DESCRIPTION + +AVplay is a very simple and portable media player using the Libav +libraries and the SDL library. It is mostly used as a testbed for the +various Libav APIs. +@c man end + +@chapter Options +@c man begin OPTIONS + +@include avtools-common-opts.texi + +@section Main options + +@table @option +@item -x @var{width} +Force displayed width. +@item -y @var{height} +Force displayed height. +@item -s @var{size} +This option has been removed. Use private format options for specifying the +input video size. For example with the rawvideo demuxer you need to specify the +option @var{video_size}. +@item -an +Disable audio. +@item -vn +Disable video. +@item -ss @var{pos} +Seek to a given position in seconds. +@item -t @var{duration} +play seconds of audio/video +@item -bytes +Seek by bytes. +@item -nodisp +Disable graphical display. +@item -f @var{fmt} +Force format. +@item -window_title @var{title} +Set window title (default is the input filename). +@item -loop @var{number} +Loops movie playback times. 0 means forever. +@item -vf @var{filter_graph} +@var{filter_graph} is a description of the filter graph to apply to +the input video. +Use the option "-filters" to show all the available filters (including +also sources and sinks). + +@end table + +@section Advanced options +@table @option +@item -pix_fmt @var{format} +This option has been removed. Use private options for specifying the +input pixel format. For example with the rawvideo demuxer you need to specify +the option @var{pixel_format}. +@item -stats +Show the stream duration, the codec parameters, the current position in +the stream and the audio/video synchronisation drift. +@item -bug +Work around bugs. +@item -fast +Non-spec-compliant optimizations. +@item -genpts +Generate pts. +@item -rtp_tcp +Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful +if you are streaming with the RTSP protocol. +@item -sync @var{type} +Set the master clock to audio (@code{type=audio}), video +(@code{type=video}) or external (@code{type=ext}). Default is audio. The +master clock is used to control audio-video synchronization. Most media +players use audio as master clock, but in some cases (streaming or high +quality broadcast) it is necessary to change that. This option is mainly +used for debugging purposes. +@item -threads @var{count} +Set the thread count. +@item -ast @var{audio_stream_number} +Select the desired audio stream number, counting from 0. The number +refers to the list of all the input audio streams. If it is greater +than the number of audio streams minus one, then the last one is +selected, if it is negative the audio playback is disabled. +@item -vst @var{video_stream_number} +Select the desired video stream number, counting from 0. The number +refers to the list of all the input video streams. If it is greater +than the number of video streams minus one, then the last one is +selected, if it is negative the video playback is disabled. +@item -sst @var{subtitle_stream_number} +Select the desired subtitle stream number, counting from 0. The number +refers to the list of all the input subtitle streams. If it is greater +than the number of subtitle streams minus one, then the last one is +selected, if it is negative the subtitle rendering is disabled. +@item -autoexit +Exit when video is done playing. +@item -exitonkeydown +Exit if any key is pressed. +@item -exitonmousedown +Exit if any mouse button is pressed. +@end table + +@section While playing + +@table @key +@item q, ESC +Quit. + +@item f +Toggle full screen. + +@item p, SPC +Pause. + +@item a +Cycle audio channel. + +@item v +Cycle video channel. + +@item t +Cycle subtitle channel. + +@item w +Show audio waves. + +@item left/right +Seek backward/forward 10 seconds. + +@item down/up +Seek backward/forward 1 minute. + +@item PGDOWN/PGUP +Seek to the previous/next chapter. + +@item mouse click +Seek to percentage in file corresponding to fraction of width. + +@end table + +@c man end + +@include eval.texi +@include decoders.texi +@include demuxers.texi +@include muxers.texi +@include indevs.texi +@include outdevs.texi +@include protocols.texi +@include filters.texi + +@ignore + +@setfilename avplay +@settitle AVplay media player + +@c man begin SEEALSO +avconv(1), avprobe(1) and the Libav HTML documentation +@c man end + +@c man begin AUTHORS +The Libav developers +@c man end + +@end ignore + +@bye diff --git a/doc/avprobe.texi b/doc/avprobe.texi new file mode 100644 index 0000000000..7e6fedf5c4 --- /dev/null +++ b/doc/avprobe.texi @@ -0,0 +1,141 @@ +\input texinfo @c -*- texinfo -*- + +@settitle avprobe Documentation +@titlepage +@center @titlefont{avprobe Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +The generic syntax is: + +@example +@c man begin SYNOPSIS +avprobe [options] [@file{input_file}] +@c man end +@end example + +@chapter Description +@c man begin DESCRIPTION + +avprobe gathers information from multimedia streams and prints it in +human- and machine-readable fashion. + +For example it can be used to check the format of the container used +by a multimedia stream and the format and type of each media stream +contained in it. + +If a filename is specified in input, avprobe will try to open and +probe the file content. If the file cannot be opened or recognized as +a multimedia file, a positive exit code is returned. + +avprobe may be employed both as a standalone application or in +combination with a textual filter, which may perform more +sophisticated processing, e.g. statistical processing or plotting. + +Options are used to list some of the formats supported by avprobe or +for specifying which information to display, and for setting how +avprobe will show it. + +avprobe output is designed to be easily parsable by any INI or JSON +parsers. + +@c man end + +@chapter Options +@c man begin OPTIONS + +@include avtools-common-opts.texi + +@section Main options + +@table @option + +@item -f @var{format} +Force format to use. + +@item -of @var{formatter} +Use a specific formatter to output the document. The following +formatters are available +@table @option +@item ini + +@item json + +@item old +Pseudo-INI format that used to be the only one available in old +avprobe versions. +@end table + +@item -unit +Show the unit of the displayed values. + +@item -prefix +Use SI prefixes for the displayed values. +Unless the "-byte_binary_prefix" option is used all the prefixes +are decimal. + +@item -byte_binary_prefix +Force the use of binary prefixes for byte values. + +@item -sexagesimal +Use sexagesimal format HH:MM:SS.MICROSECONDS for time values. + +@item -pretty +Prettify the format of the displayed values, it corresponds to the +options "-unit -prefix -byte_binary_prefix -sexagesimal". + +@item -show_format +Show information about the container format of the input multimedia +stream. + +All the container format information is printed within a section with +name "FORMAT". + +@item -show_format_entry @var{name} +Like @option{-show_format}, but only prints the specified entry of the +container format information, rather than all. This option may be given more +than once, then all specified entries will be shown. + +@item -show_packets +Show information about each packet contained in the input multimedia +stream. + +The information for each single packet is printed within a dedicated +section with name "PACKET". + +@item -show_streams +Show information about each media stream contained in the input +multimedia stream. + +Each media stream information is printed within a dedicated section +with name "STREAM". + +@end table +@c man end + +@include demuxers.texi +@include muxers.texi +@include protocols.texi +@include indevs.texi + +@ignore + +@setfilename avprobe +@settitle avprobe media prober + +@c man begin SEEALSO +avconv(1), avplay(1) and the Libav HTML documentation +@c man end + +@c man begin AUTHORS +The Libav developers +@c man end + +@end ignore + +@bye diff --git a/doc/ffserver.conf b/doc/avserver.conf similarity index 91% rename from doc/ffserver.conf rename to doc/avserver.conf index 094c093af2..e1cd9fb126 100644 --- a/doc/ffserver.conf +++ b/doc/avserver.conf @@ -12,7 +12,7 @@ BindAddress 0.0.0.0 # MaxClients maximum limit. MaxHTTPConnections 2000 -# Number of simultaneous requests that can be handled. Since FFServer +# Number of simultaneous requests that can be handled. Since AVServer # is very fast, it is more likely that you will want to leave this high # and use MaxBandwidth, below. MaxClients 1000 @@ -25,20 +25,21 @@ MaxBandwidth 1000 # '-' is the standard output. CustomLog - + ################################################################## # Definition of the live feeds. Each live feed contains one video -# and/or audio sequence coming from an ffmpeg encoder or another -# ffserver. This sequence may be encoded simultaneously with several +# and/or audio sequence coming from an avconv encoder or another +# avserver. This sequence may be encoded simultaneously with several # codecs at several resolutions. -# You must use 'ffmpeg' to send a live feed to ffserver. In this +# You must use 'avconv' to send a live feed to avserver. In this # example, you can type: # -# ffmpeg http://localhost:8090/feed1.ffm +# avconv http://localhost:8090/feed1.ffm -# ffserver can also do time shifting. It means that it can stream any +# avserver can also do time shifting. It means that it can stream any # previously recorded live stream. The request should contain: # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify # a path where the feed is stored on disk. You also specify the @@ -51,10 +52,10 @@ FileMaxSize 200K # ReadOnlyFile /saved/specialvideo.ffm # This marks the file as readonly and it will not be deleted or updated. -# Specify launch in order to start ffmpeg automatically. -# First ffmpeg must be defined with an appropriate path if needed, +# Specify launch in order to start avconv automatically. +# First avconv must be defined with an appropriate path if needed, # after that options can follow, but avoid adding the http:// field -#Launch ffmpeg +#Launch avconv # Only allow connections from localhost to the feed. ACL allow 127.0.0.1 @@ -65,7 +66,7 @@ ACL allow 127.0.0.1 ################################################################## # Now you can define each stream which will be generated from the # original audio and video stream. Each format has a filename (here -# 'test1.mpg'). FFServer will send this stream when answering a +# 'test1.mpg'). AVServer will send this stream when answering a # request containing this filename. @@ -235,7 +236,7 @@ StartSendOnKey # #Feed feed1.ffm -#Metadata title "Stream title" +#Title "Stream title" #AudioBitRate 64 #AudioChannels 2 #AudioSampleRate 44100 @@ -280,10 +281,10 @@ StartSendOnKey # #File "/usr/local/httpd/htdocs/test.asf" #NoAudio -#Metadata author "Me" -#Metadata copyright "Super MegaCorp" -#Metadata title "Test stream from disk" -#Metadata comment "Test comment" +#Author "Me" +#Copyright "Super MegaCorp" +#Title "Test stream from disk" +#Comment "Test comment" # @@ -330,7 +331,7 @@ StartSendOnKey # multicast address with MulticastAddress. The port and the TTL can # also be set. # -# An SDP file is automatically generated by ffserver by adding the +# An SDP file is automatically generated by avserver by adding the # 'sdp' extension to the stream name (here # http://localhost:8090/test1-sdp.sdp). You should usually give this # file to your player to play the stream. @@ -367,5 +368,5 @@ ACL allow 192.168.0.0 192.168.255.255 # Redirect index.html to the appropriate site -URL http://www.ffmpeg.org/ +URL http://www.libav.org/ diff --git a/doc/avserver.texi b/doc/avserver.texi new file mode 100644 index 0000000000..88e6221f6a --- /dev/null +++ b/doc/avserver.texi @@ -0,0 +1,276 @@ +\input texinfo @c -*- texinfo -*- + +@settitle avserver Documentation +@titlepage +@center @titlefont{avserver Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsys + +The generic syntax is: + +@example +@c man begin SYNOPSIS +avserver [options] +@c man end +@end example + +@chapter Description +@c man begin DESCRIPTION + +WARNING: avserver is unmaintained, largely broken and in need of a +complete rewrite. It probably won't work for you. Use at your own +risk. + +avserver is a streaming server for both audio and video. It supports +several live feeds, streaming from files and time shifting on live feeds +(you can seek to positions in the past on each live feed, provided you +specify a big enough feed storage in avserver.conf). + +This documentation covers only the streaming aspects of avserver / +avconv. All questions about parameters for avconv, codec questions, +etc. are not covered here. Read @file{avconv.html} for more +information. + +@section How does it work? + +avserver receives prerecorded files or FFM streams from some avconv +instance as input, then streams them over RTP/RTSP/HTTP. + +An avserver instance will listen on some port as specified in the +configuration file. You can launch one or more instances of avconv and +send one or more FFM streams to the port where avserver is expecting +to receive them. Alternately, you can make avserver launch such avconv +instances at startup. + +Input streams are called feeds, and each one is specified by a +section in the configuration file. + +For each feed you can have different output streams in various +formats, each one specified by a section in the configuration +file. + +@section Status stream + +avserver supports an HTTP interface which exposes the current status +of the server. + +Simply point your browser to the address of the special status stream +specified in the configuration file. + +For example if you have: +@example + +Format status + +# Only allow local people to get the status +ACL allow localhost +ACL allow 192.168.0.0 192.168.255.255 + +@end example + +then the server will post a page with the status information when +the special stream @file{status.html} is requested. + +@section What can this do? + +When properly configured and running, you can capture video and audio in real +time from a suitable capture card, and stream it out over the Internet to +either Windows Media Player or RealAudio player (with some restrictions). + +It can also stream from files, though that is currently broken. Very often, a +web server can be used to serve up the files just as well. + +It can stream prerecorded video from .ffm files, though it is somewhat tricky +to make it work correctly. + +@section What do I need? + +I use Linux on a 900 MHz Duron with a cheapo Bt848 based TV capture card. I'm +using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true, +I needed some special drivers for my motherboard-based sound card.] + +I understand that FreeBSD systems work just fine as well. + +@section How do I make it work? + +First, build the kit. It *really* helps to have installed LAME first. Then when +you run the avserver ./configure, make sure that you have the +@code{--enable-libmp3lame} flag turned on. + +LAME is important as it allows for streaming audio to Windows Media Player. +Don't ask why the other audio types do not work. + +As a simple test, just run the following two command lines where INPUTFILE +is some file which you can decode with avconv: + +@example +./avserver -f doc/avserver.conf & +./avconv -i INPUTFILE http://localhost:8090/feed1.ffm +@end example + +At this point you should be able to go to your Windows machine and fire up +Windows Media Player (WMP). Go to Open URL and enter + +@example + http://:8090/test.asf +@end example + +You should (after a short delay) see video and hear audio. + +WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to +transfer the entire file before starting to play. +The same is true of AVI files. + +@section What happens next? + +You should edit the avserver.conf file to suit your needs (in terms of +frame rates etc). Then install avserver and avconv, write a script to start +them up, and off you go. + +@section Troubleshooting + +@subsection I don't hear any audio, but video is fine. + +Maybe you didn't install LAME, or got your ./configure statement wrong. Check +the avconv output to see if a line referring to MP3 is present. If not, then +your configuration was incorrect. If it is, then maybe your wiring is not +set up correctly. Maybe the sound card is not getting data from the right +input source. Maybe you have a really awful audio interface (like I do) +that only captures in stereo and also requires that one channel be flipped. +If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before +starting avconv. + +@subsection The audio and video lose sync after a while. + +Yes, they do. + +@subsection After a long while, the video update rate goes way down in WMP. + +Yes, it does. Who knows why? + +@subsection WMP 6.4 behaves differently to WMP 7. + +Yes, it does. Any thoughts on this would be gratefully received. These +differences extend to embedding WMP into a web page. [There are two +object IDs that you can use: The old one, which does not play well, and +the new one, which does (both tested on the same system). However, +I suspect that the new one is not available unless you have installed WMP 7]. + +@section What else can it do? + +You can replay video from .ffm files that was recorded earlier. +However, there are a number of caveats, including the fact that the +avserver parameters must match the original parameters used to record the +file. If they do not, then avserver deletes the file before recording into it. +(Now that I write this, it seems broken). + +You can fiddle with many of the codec choices and encoding parameters, and +there are a bunch more parameters that you cannot control. Post a message +to the mailing list if there are some 'must have' parameters. Look in +avserver.conf for a list of the currently available controls. + +It will automatically generate the ASX or RAM files that are often used +in browsers. These files are actually redirections to the underlying ASF +or RM file. The reason for this is that the browser often fetches the +entire file before starting up the external viewer. The redirection files +are very small and can be transferred quickly. [The stream itself is +often 'infinite' and thus the browser tries to download it and never +finishes.] + +@section Tips + +* When you connect to a live stream, most players (WMP, RA, etc) want to +buffer a certain number of seconds of material so that they can display the +signal continuously. However, avserver (by default) starts sending data +in realtime. This means that there is a pause of a few seconds while the +buffering is being done by the player. The good news is that this can be +cured by adding a '?buffer=5' to the end of the URL. This means that the +stream should start 5 seconds in the past -- and so the first 5 seconds +of the stream are sent as fast as the network will allow. It will then +slow down to real time. This noticeably improves the startup experience. + +You can also add a 'Preroll 15' statement into the avserver.conf that will +add the 15 second prebuffering on all requests that do not otherwise +specify a time. In addition, avserver will skip frames until a key_frame +is found. This further reduces the startup delay by not transferring data +that will be discarded. + +* You may want to adjust the MaxBandwidth in the avserver.conf to limit +the amount of bandwidth consumed by live streams. + +@section Why does the ?buffer / Preroll stop working after a time? + +It turns out that (on my machine at least) the number of frames successfully +grabbed is marginally less than the number that ought to be grabbed. This +means that the timestamp in the encoded data stream gets behind realtime. +This means that if you say 'Preroll 10', then when the stream gets 10 +or more seconds behind, there is no Preroll left. + +Fixing this requires a change in the internals of how timestamps are +handled. + +@section Does the @code{?date=} stuff work. + +Yes (subject to the limitation outlined above). Also note that whenever you +start avserver, it deletes the ffm file (if any parameters have changed), +thus wiping out what you had recorded before. + +The format of the @code{?date=xxxxxx} is fairly flexible. You should use one +of the following formats (the 'T' is literal): + +@example +* YYYY-MM-DDTHH:MM:SS (localtime) +* YYYY-MM-DDTHH:MM:SSZ (UTC) +@end example + +You can omit the YYYY-MM-DD, and then it refers to the current day. However +note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this +may be in the future and so is unlikely to be useful. + +You use this by adding the ?date= to the end of the URL for the stream. +For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}. +@c man end + +@chapter Options +@c man begin OPTIONS + +@include avtools-common-opts.texi + +@section Main options + +@table @option +@item -f @var{configfile} +Use @file{configfile} instead of @file{/etc/avserver.conf}. +@item -n +Enable no-launch mode. This option disables all the Launch directives +within the various sections. Since avserver will not launch +any avconv instances, you will have to launch them manually. +@item -d +Enable debug mode. This option increases log verbosity, directs log +messages to stdout. +@end table +@c man end + +@ignore + +@setfilename avserver +@settitle avserver video server + +@c man begin SEEALSO + +avconv(1), avplay(1), avprobe(1), the @file{avserver.conf} +example and the Libav HTML documentation +@c man end + +@c man begin AUTHORS +The Libav developers +@c man end + +@end ignore + +@bye diff --git a/doc/avtools-common-opts.texi b/doc/avtools-common-opts.texi new file mode 100644 index 0000000000..3bf321f36d --- /dev/null +++ b/doc/avtools-common-opts.texi @@ -0,0 +1,186 @@ +All the numerical options, if not specified otherwise, accept in input +a string representing a number, which may contain one of the +SI unit prefixes, for example 'K', 'M', 'G'. +If 'i' is appended after the prefix, binary prefixes are used, +which are based on powers of 1024 instead of powers of 1000. +The 'B' postfix multiplies the value by 8, and can be +appended after a unit prefix or used alone. This allows using for +example 'KB', 'MiB', 'G' and 'B' as number postfix. + +Options which do not take arguments are boolean options, and set the +corresponding value to true. They can be set to false by prefixing +with "no" the option name, for example using "-nofoo" in the +command line will set to false the boolean option with name "foo". + +@anchor{Stream specifiers} +@section Stream specifiers +Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers +are used to precisely specify which stream(s) does a given option belong to. + +A stream specifier is a string generally appended to the option name and +separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains +@code{a:1} stream specifer, which matches the second audio stream. Therefore it +would select the ac3 codec for the second audio stream. + +A stream specifier can match several stream, the option is then applied to all +of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio +streams. + +An empty stream specifier matches all streams, for example @code{-codec copy} +or @code{-codec: copy} would copy all the streams without reencoding. + +Possible forms of stream specifiers are: +@table @option +@item @var{stream_index} +Matches the stream with this index. E.g. @code{-threads:1 4} would set the +thread count for the second stream to 4. +@item @var{stream_type}[:@var{stream_index}] +@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle, +'d' for data and 't' for attachments. If @var{stream_index} is given, then +matches stream number @var{stream_index} of this type. Otherwise matches all +streams of this type. +@item p:@var{program_id}[:@var{stream_index}] +If @var{stream_index} is given, then matches stream number @var{stream_index} in +program with id @var{program_id}. Otherwise matches all streams in this program. +@item i:@var{stream_id} +Match the stream by stream id (e.g. PID in MPEG-TS container). +@end table +@section Generic options + +These options are shared amongst the av* tools. + +@table @option + +@item -L +Show license. + +@item -h, -?, -help, --help [@var{arg}] +Show help. An optional parameter may be specified to print help about a specific +item. + +Possible values of @var{arg} are: +@table @option +@item decoder=@var{decoder_name} +Print detailed information about the decoder named @var{decoder_name}. Use the +@option{-decoders} option to get a list of all decoders. + +@item encoder=@var{encoder_name} +Print detailed information about the encoder named @var{encoder_name}. Use the +@option{-encoders} option to get a list of all encoders. + +@item demuxer=@var{demuxer_name} +Print detailed information about the demuxer named @var{demuxer_name}. Use the +@option{-formats} option to get a list of all demuxers and muxers. + +@item muxer=@var{muxer_name} +Print detailed information about the muxer named @var{muxer_name}. Use the +@option{-formats} option to get a list of all muxers and demuxers. + +@item filter=@var{filter_name} +Print detailed information about the filter name @var{filter_name}. Use the +@option{-filters} option to get a list of all filters. + +@end table + +@item -version +Show version. + +@item -formats +Show available formats. + +The fields preceding the format names have the following meanings: +@table @samp +@item D +Decoding available +@item E +Encoding available +@end table + +@item -codecs +Show all codecs known to libavcodec. + +Note that the term 'codec' is used throughout this documentation as a shortcut +for what is more correctly called a media bitstream format. + +@item -decoders +Show available decoders. + +@item -encoders +Show all available encoders. + +@item -bsfs +Show available bitstream filters. + +@item -protocols +Show available protocols. + +@item -filters +Show available libavfilter filters. + +@item -pix_fmts +Show available pixel formats. + +@item -sample_fmts +Show available sample formats. + +@item -loglevel @var{loglevel} | -v @var{loglevel} +Set the logging level used by the library. +@var{loglevel} is a number or a string containing one of the following values: +@table @samp +@item quiet +@item panic +@item fatal +@item error +@item warning +@item info +@item verbose +@item debug +@end table + +By default the program logs to stderr, if coloring is supported by the +terminal, colors are used to mark errors and warnings. Log coloring +can be disabled setting the environment variable +@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting +the environment variable @env{AV_LOG_FORCE_COLOR}. +The use of the environment variable @env{NO_COLOR} is deprecated and +will be dropped in a following Libav version. + +@item -cpuflags mask (@emph{global}) +Set a mask that's applied to autodetected CPU flags. This option is intended +for testing. Do not use it unless you know what you're doing. + +@end table + +@section AVOptions + +These options are provided directly by the libavformat, libavdevice and +libavcodec libraries. To see the list of available AVOptions, use the +@option{-help} option. They are separated into two categories: +@table @option +@item generic +These options can be set for any container, codec or device. Generic options +are listed under AVFormatContext options for containers/devices and under +AVCodecContext options for codecs. +@item private +These options are specific to the given container, device or codec. Private +options are listed under their corresponding containers/devices/codecs. +@end table + +For example to write an ID3v2.3 header instead of a default ID3v2.4 to +an MP3 file, use the @option{id3v2_version} private option of the MP3 +muxer: +@example +avconv -i input.flac -id3v2_version 3 out.mp3 +@end example + +All codec AVOptions are obviously per-stream, so the chapter on stream +specifiers applies to them + +Note @option{-nooption} syntax cannot be used for boolean AVOptions, +use @option{-option 0}/@option{-option 1}. + +Note2 old undocumented way of specifying per-stream AVOptions by prepending +v/a/s to the options name is now obsolete and will be removed soon. + +@include avoptions_codec.texi +@include avoptions_format.texi diff --git a/doc/avutil.txt b/doc/avutil.txt new file mode 100644 index 0000000000..0847683d1d --- /dev/null +++ b/doc/avutil.txt @@ -0,0 +1,36 @@ +AVUtil +====== +libavutil is a small lightweight library of generally useful functions. +It is not a library for code needed by both libavcodec and libavformat. + + +Overview: +========= +adler32.c adler32 checksum +aes.c AES encryption and decryption +fifo.c resizeable first in first out buffer +intfloat_readwrite.c portable reading and writing of floating point values +log.c "printf" with context and level +md5.c MD5 Message-Digest Algorithm +rational.c code to perform exact calculations with rational numbers +tree.c generic AVL tree +crc.c generic CRC checksumming code +integer.c 128bit integer math +lls.c +mathematics.c greatest common divisor, integer sqrt, integer log2, ... +mem.c memory allocation routines with guaranteed alignment + +Headers: +bswap.h big/little/native-endian conversion code +x86_cpu.h a few useful macros for unifying x86-64 and x86-32 code +avutil.h +common.h +intreadwrite.h reading and writing of unaligned big/little/native-endian integers + + +Goals: +====== +* Modular (few interdependencies and the possibility of disabling individual parts during ./configure) +* Small (source and object) +* Efficient (low CPU and memory usage) +* Useful (avoid useless features almost no one needs) diff --git a/doc/bitstream_filters.texi b/doc/bitstream_filters.texi index 9bcb12c734..6e7f8781ee 100644 --- a/doc/bitstream_filters.texi +++ b/doc/bitstream_filters.texi @@ -1,7 +1,7 @@ @chapter Bitstream Filters @c man begin BITSTREAM FILTERS -When you configure your FFmpeg build, all the supported bitstream +When you configure your Libav build, all the supported bitstream filters are enabled by default. You can list all available ones using the configure option @code{--list-bsfs}. @@ -10,70 +10,19 @@ You can disable all the bitstream filters using the configure option the option @code{--enable-bsf=BSF}, or you can disable a particular bitstream filter using the option @code{--disable-bsf=BSF}. -The option @code{-bsfs} of the ff* tools will display the list of +The option @code{-bsfs} of the av* tools will display the list of all the supported bitstream filters included in your build. Below is a description of the currently available bitstream filters. @section aac_adtstoasc -Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration -bitstream filter. - -This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 -ADTS header and removes the ADTS header. - -This is required for example when copying an AAC stream from a raw -ADTS AAC container to a FLV or a MOV/MP4 file. - @section chomp -Remove zero padding at the end of a packet. - -@section dump_extra - -Add extradata to the beginning of the filtered packets. - -The additional argument specifies which packets should be filtered. -It accepts the values: -@table @samp -@item a -add extradata to all key packets, but only if @var{local_header} is -set in the @option{flags2} codec context field - -@item k -add extradata to all key packets - -@item e -add extradata to all packets -@end table - -If not specified it is assumed @samp{k}. - -For example the following @command{ffmpeg} command forces a global -header (thus disabling individual packet headers) in the H.264 packets -generated by the @code{libx264} encoder, but corrects them by adding -the header stored in extradata to the key packets: -@example -ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts -@end example +@section dump_extradata @section h264_mp4toannexb -Convert an H.264 bitstream from length prefixed mode to start code -prefixed mode (as defined in the Annex B of the ITU-T H.264 -specification). - -This is required by some streaming formats, typically the MPEG-2 -transport stream format ("mpegts"). - -For example to remux an MP4 file containing an H.264 stream to mpegts -format with @command{ffmpeg}, you can use the command: - -@example -ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts -@end example - @section imx_dump_header @section mjpeg2jpeg @@ -85,7 +34,7 @@ JPEG image. The individual frames can be extracted without loss, e.g. by @example -ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg +avconv -i ../some_mjpeg.avi -c:v copy frames_%d.jpg @end example Unfortunately, these chunks are incomplete JPEG images, because @@ -108,19 +57,21 @@ stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images. @example -ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg +avconv -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg exiftran -i -9 frame*.jpg -ffmpeg -i frame_%d.jpg -c:v copy rotated.avi +avconv -i frame_%d.jpg -c:v copy rotated.avi @end example @section mjpega_dump_header @section movsub +@section mp3_header_compress + @section mp3_header_decompress @section noise -@section remove_extra +@section remove_extradata @c man end BITSTREAM FILTERS diff --git a/doc/build_system.txt b/doc/build_system.txt index 36c141e9e4..c3dede7cde 100644 --- a/doc/build_system.txt +++ b/doc/build_system.txt @@ -1,4 +1,4 @@ -FFmpeg currently uses a custom build system, this text attempts to document +Libav currently uses a custom build system, this text attempts to document some of its obscure features and options. Makefile variables: @@ -9,19 +9,13 @@ V DESTDIR Destination directory for the install targets, useful to prepare packages - or install FFmpeg in cross-environments. + or install Libav in cross-environments. Makefile targets: all Default target, builds all the libraries and the executables. -fate - Run the fate test suite, note you must have installed it - -fate-list - Will list all fate/regression test targets - install Install headers, libraries and programs. @@ -33,18 +27,3 @@ libavcodec/api-example libswscale/swscale-test Build the swscale self-test (useful also as example). - - -Useful standard make commands: -make -t - Touch all files that otherwise would be build, this is useful to reduce - unneeded rebuilding when changing headers, but note you must force rebuilds - of files that actually need it by hand then. - -make -j - rebuild with multiple jobs at the same time. Faster on multi processor systems - -make -k - continue build in case of errors, this is useful for the regression tests - sometimes but note it will still not run all reg tests. - diff --git a/doc/codecs.texi b/doc/codecs.texi deleted file mode 100644 index ec69a581ba..0000000000 --- a/doc/codecs.texi +++ /dev/null @@ -1,1107 +0,0 @@ -@anchor{codec-options} -@chapter Codec Options -@c man begin CODEC OPTIONS - -libavcodec provides some generic global options, which can be set on -all the encoders and decoders. In addition each codec may support -so-called private options, which are specific for a given codec. - -Sometimes, a global option may only affect a specific kind of codec, -and may be unsensical or ignored by another, so you need to be aware -of the meaning of the specified options. Also some options are -meant only for decoding or encoding. - -Options may be set by specifying -@var{option} @var{value} in the -FFmpeg tools, or by setting the value explicitly in the -@code{AVCodecContext} options or using the @file{libavutil/opt.h} API -for programmatic use. - -The list of supported options follow: - -@table @option -@item b @var{integer} (@emph{encoding,audio,video}) -Set bitrate in bits/s. Default value is 200K. - -@item ab @var{integer} (@emph{encoding,audio}) -Set audio bitrate (in bits/s). Default value is 128K. - -@item bt @var{integer} (@emph{encoding,video}) -Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate -tolerance specifies how far ratecontrol is willing to deviate from the -target average bitrate value. This is not related to min/max -bitrate. Lowering tolerance too much has an adverse effect on quality. - -@item flags @var{flags} (@emph{decoding/encoding,audio,video,subtitles}) -Set generic flags. - -Possible values: -@table @samp -@item mv4 -Use four motion vector by macroblock (mpeg4). -@item qpel -Use 1/4 pel motion compensation. -@item loop -Use loop filter. -@item qscale -Use fixed qscale. -@item gmc -Use gmc. -@item mv0 -Always try a mb with mv=<0,0>. -@item input_preserved - -@item pass1 -Use internal 2pass ratecontrol in first pass mode. -@item pass2 -Use internal 2pass ratecontrol in second pass mode. -@item gray -Only decode/encode grayscale. -@item emu_edge -Do not draw edges. -@item psnr -Set error[?] variables during encoding. -@item truncated - -@item naq -Normalize adaptive quantization. -@item ildct -Use interlaced DCT. -@item low_delay -Force low delay. -@item global_header -Place global headers in extradata instead of every keyframe. -@item bitexact -Use only bitexact stuff (except (I)DCT). -@item aic -Apply H263 advanced intra coding / mpeg4 ac prediction. -@item cbp -Deprecated, use mpegvideo private options instead. -@item qprd -Deprecated, use mpegvideo private options instead. -@item ilme -Apply interlaced motion estimation. -@item cgop -Use closed gop. -@end table - -@item me_method @var{integer} (@emph{encoding,video}) -Set motion estimation method. - -Possible values: -@table @samp -@item zero -zero motion estimation (fastest) -@item full -full motion estimation (slowest) -@item epzs -EPZS motion estimation (default) -@item esa -esa motion estimation (alias for full) -@item tesa -tesa motion estimation -@item dia -dia motion estimation (alias for epzs) -@item log -log motion estimation -@item phods -phods motion estimation -@item x1 -X1 motion estimation -@item hex -hex motion estimation -@item umh -umh motion estimation -@item iter -iter motion estimation -@end table - -@item extradata_size @var{integer} -Set extradata size. - -@item time_base @var{rational number} -Set codec time base. - -It is the fundamental unit of time (in seconds) in terms of which -frame timestamps are represented. For fixed-fps content, timebase -should be @code{1 / frame_rate} and timestamp increments should be -identically 1. - -@item g @var{integer} (@emph{encoding,video}) -Set the group of picture size. Default value is 12. - -@item ar @var{integer} (@emph{decoding/encoding,audio}) -Set audio sampling rate (in Hz). - -@item ac @var{integer} (@emph{decoding/encoding,audio}) -Set number of audio channels. - -@item cutoff @var{integer} (@emph{encoding,audio}) -Set cutoff bandwidth. - -@item frame_size @var{integer} (@emph{encoding,audio}) -Set audio frame size. - -Each submitted frame except the last must contain exactly frame_size -samples per channel. May be 0 when the codec has -CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not -restricted. It is set by some decoders to indicate constant frame -size. - -@item frame_number @var{integer} -Set the frame number. - -@item delay @var{integer} - -@item qcomp @var{float} (@emph{encoding,video}) -Set video quantizer scale compression (VBR). It is used as a constant -in the ratecontrol equation. Recommended range for default rc_eq: -0.0-1.0. - -@item qblur @var{float} (@emph{encoding,video}) -Set video quantizer scale blur (VBR). - -@item qmin @var{integer} (@emph{encoding,video}) -Set min video quantizer scale (VBR). Must be included between -1 and -69, default value is 2. - -@item qmax @var{integer} (@emph{encoding,video}) -Set max video quantizer scale (VBR). Must be included between -1 and -1024, default value is 31. - -@item qdiff @var{integer} (@emph{encoding,video}) -Set max difference between the quantizer scale (VBR). - -@item bf @var{integer} (@emph{encoding,video}) -Set max number of B frames between non-B-frames. - -Must be an integer between -1 and 16. 0 means that B-frames are -disabled. If a value of -1 is used, it will choose an automatic value -depending on the encoder. - -Default value is 0. - -@item b_qfactor @var{float} (@emph{encoding,video}) -Set qp factor between P and B frames. - -@item rc_strategy @var{integer} (@emph{encoding,video}) -Set ratecontrol method. - -@item b_strategy @var{integer} (@emph{encoding,video}) -Set strategy to choose between I/P/B-frames. - -@item ps @var{integer} (@emph{encoding,video}) -Set RTP payload size in bytes. - -@item mv_bits @var{integer} -@item header_bits @var{integer} -@item i_tex_bits @var{integer} -@item p_tex_bits @var{integer} -@item i_count @var{integer} -@item p_count @var{integer} -@item skip_count @var{integer} -@item misc_bits @var{integer} -@item frame_bits @var{integer} -@item codec_tag @var{integer} -@item bug @var{flags} (@emph{decoding,video}) -Workaround not auto detected encoder bugs. - -Possible values: -@table @samp -@item autodetect - -@item old_msmpeg4 -some old lavc generated msmpeg4v3 files (no autodetection) -@item xvid_ilace -Xvid interlacing bug (autodetected if fourcc==XVIX) -@item ump4 -(autodetected if fourcc==UMP4) -@item no_padding -padding bug (autodetected) -@item amv - -@item ac_vlc -illegal vlc bug (autodetected per fourcc) -@item qpel_chroma - -@item std_qpel -old standard qpel (autodetected per fourcc/version) -@item qpel_chroma2 - -@item direct_blocksize -direct-qpel-blocksize bug (autodetected per fourcc/version) -@item edge -edge padding bug (autodetected per fourcc/version) -@item hpel_chroma - -@item dc_clip - -@item ms -Workaround various bugs in microsoft broken decoders. -@item trunc -trancated frames -@end table - -@item lelim @var{integer} (@emph{encoding,video}) -Set single coefficient elimination threshold for luminance (negative -values also consider DC coefficient). - -@item celim @var{integer} (@emph{encoding,video}) -Set single coefficient elimination threshold for chrominance (negative -values also consider dc coefficient) - -@item strict @var{integer} (@emph{decoding/encoding,audio,video}) -Specify how strictly to follow the standards. - -Possible values: -@table @samp -@item very -strictly conform to a older more strict version of the spec or reference software -@item strict -strictly conform to all the things in the spec no matter what consequences -@item normal - -@item unofficial -allow unofficial extensions -@item experimental -allow non standardized experimental things, experimental -(unfinished/work in progress/not well tested) decoders and encoders. -Note: experimental decoders can pose a security risk, do not use this for -decoding untrusted input. -@end table - -@item b_qoffset @var{float} (@emph{encoding,video}) -Set QP offset between P and B frames. - -@item err_detect @var{flags} (@emph{decoding,audio,video}) -Set error detection flags. - -Possible values: -@table @samp -@item crccheck -verify embedded CRCs -@item bitstream -detect bitstream specification deviations -@item buffer -detect improper bitstream length -@item explode -abort decoding on minor error detection -@item ignore_err -ignore decoding errors, and continue decoding. -This is useful if you want to analyze the content of a video and thus want -everything to be decoded no matter what. This option will not result in a video -that is pleasing to watch in case of errors. -@item careful -consider things that violate the spec and have not been seen in the wild as errors -@item compliant -consider all spec non compliancies as errors -@item aggressive -consider things that a sane encoder should not do as an error -@end table - -@item has_b_frames @var{integer} - -@item block_align @var{integer} - -@item mpeg_quant @var{integer} (@emph{encoding,video}) -Use MPEG quantizers instead of H.263. - -@item qsquish @var{float} (@emph{encoding,video}) -How to keep quantizer between qmin and qmax (0 = clip, 1 = use -differentiable function). - -@item rc_qmod_amp @var{float} (@emph{encoding,video}) -Set experimental quantizer modulation. - -@item rc_qmod_freq @var{integer} (@emph{encoding,video}) -Set experimental quantizer modulation. - -@item rc_override_count @var{integer} - -@item rc_eq @var{string} (@emph{encoding,video}) -Set rate control equation. When computing the expression, besides the -standard functions defined in the section 'Expression Evaluation', the -following functions are available: bits2qp(bits), qp2bits(qp). Also -the following constants are available: iTex pTex tex mv fCode iCount -mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex -avgTex. - -@item maxrate @var{integer} (@emph{encoding,audio,video}) -Set max bitrate tolerance (in bits/s). Requires bufsize to be set. - -@item minrate @var{integer} (@emph{encoding,audio,video}) -Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR -encode. It is of little use elsewise. - -@item bufsize @var{integer} (@emph{encoding,audio,video}) -Set ratecontrol buffer size (in bits). - -@item rc_buf_aggressivity @var{float} (@emph{encoding,video}) -Currently useless. - -@item i_qfactor @var{float} (@emph{encoding,video}) -Set QP factor between P and I frames. - -@item i_qoffset @var{float} (@emph{encoding,video}) -Set QP offset between P and I frames. - -@item rc_init_cplx @var{float} (@emph{encoding,video}) -Set initial complexity for 1-pass encoding. - -@item dct @var{integer} (@emph{encoding,video}) -Set DCT algorithm. - -Possible values: -@table @samp -@item auto -autoselect a good one (default) -@item fastint -fast integer -@item int -accurate integer -@item mmx - -@item altivec - -@item faan -floating point AAN DCT -@end table - -@item lumi_mask @var{float} (@emph{encoding,video}) -Compress bright areas stronger than medium ones. - -@item tcplx_mask @var{float} (@emph{encoding,video}) -Set temporal complexity masking. - -@item scplx_mask @var{float} (@emph{encoding,video}) -Set spatial complexity masking. - -@item p_mask @var{float} (@emph{encoding,video}) -Set inter masking. - -@item dark_mask @var{float} (@emph{encoding,video}) -Compress dark areas stronger than medium ones. - -@item idct @var{integer} (@emph{decoding/encoding,video}) -Select IDCT implementation. - -Possible values: -@table @samp -@item auto - -@item int - -@item simple - -@item simplemmx - -@item arm - -@item altivec - -@item sh4 - -@item simplearm - -@item simplearmv5te - -@item simplearmv6 - -@item simpleneon - -@item simplealpha - -@item ipp - -@item xvidmmx - -@item faani -floating point AAN IDCT -@end table - -@item slice_count @var{integer} - -@item ec @var{flags} (@emph{decoding,video}) -Set error concealment strategy. - -Possible values: -@table @samp -@item guess_mvs -iterative motion vector (MV) search (slow) -@item deblock -use strong deblock filter for damaged MBs -@item favor_inter -favor predicting from the previous frame instead of the current -@end table - -@item bits_per_coded_sample @var{integer} - -@item pred @var{integer} (@emph{encoding,video}) -Set prediction method. - -Possible values: -@table @samp -@item left - -@item plane - -@item median - -@end table - -@item aspect @var{rational number} (@emph{encoding,video}) -Set sample aspect ratio. - -@item debug @var{flags} (@emph{decoding/encoding,audio,video,subtitles}) -Print specific debug info. - -Possible values: -@table @samp -@item pict -picture info -@item rc -rate control -@item bitstream - -@item mb_type -macroblock (MB) type -@item qp -per-block quantization parameter (QP) -@item mv -motion vector -@item dct_coeff - -@item skip - -@item startcode - -@item pts - -@item er -error recognition -@item mmco -memory management control operations (H.264) -@item bugs - -@item vis_qp -visualize quantization parameter (QP), lower QP are tinted greener -@item vis_mb_type -visualize block types -@item buffers -picture buffer allocations -@item thread_ops -threading operations -@end table - -@item vismv @var{integer} (@emph{decoding,video}) -Visualize motion vectors (MVs). - -Possible values: -@table @samp -@item pf -forward predicted MVs of P-frames -@item bf -forward predicted MVs of B-frames -@item bb -backward predicted MVs of B-frames -@end table - -@item cmp @var{integer} (@emph{encoding,video}) -Set full pel me compare function. - -Possible values: -@table @samp -@item sad -sum of absolute differences, fast (default) -@item sse -sum of squared errors -@item satd -sum of absolute Hadamard transformed differences -@item dct -sum of absolute DCT transformed differences -@item psnr -sum of squared quantization errors (avoid, low quality) -@item bit -number of bits needed for the block -@item rd -rate distortion optimal, slow -@item zero -0 -@item vsad -sum of absolute vertical differences -@item vsse -sum of squared vertical differences -@item nsse -noise preserving sum of squared differences -@item w53 -5/3 wavelet, only used in snow -@item w97 -9/7 wavelet, only used in snow -@item dctmax - -@item chroma - -@end table - -@item subcmp @var{integer} (@emph{encoding,video}) -Set sub pel me compare function. - -Possible values: -@table @samp -@item sad -sum of absolute differences, fast (default) -@item sse -sum of squared errors -@item satd -sum of absolute Hadamard transformed differences -@item dct -sum of absolute DCT transformed differences -@item psnr -sum of squared quantization errors (avoid, low quality) -@item bit -number of bits needed for the block -@item rd -rate distortion optimal, slow -@item zero -0 -@item vsad -sum of absolute vertical differences -@item vsse -sum of squared vertical differences -@item nsse -noise preserving sum of squared differences -@item w53 -5/3 wavelet, only used in snow -@item w97 -9/7 wavelet, only used in snow -@item dctmax - -@item chroma - -@end table - -@item mbcmp @var{integer} (@emph{encoding,video}) -Set macroblock compare function. - -Possible values: -@table @samp -@item sad -sum of absolute differences, fast (default) -@item sse -sum of squared errors -@item satd -sum of absolute Hadamard transformed differences -@item dct -sum of absolute DCT transformed differences -@item psnr -sum of squared quantization errors (avoid, low quality) -@item bit -number of bits needed for the block -@item rd -rate distortion optimal, slow -@item zero -0 -@item vsad -sum of absolute vertical differences -@item vsse -sum of squared vertical differences -@item nsse -noise preserving sum of squared differences -@item w53 -5/3 wavelet, only used in snow -@item w97 -9/7 wavelet, only used in snow -@item dctmax - -@item chroma - -@end table - -@item ildctcmp @var{integer} (@emph{encoding,video}) -Set interlaced dct compare function. - -Possible values: -@table @samp -@item sad -sum of absolute differences, fast (default) -@item sse -sum of squared errors -@item satd -sum of absolute Hadamard transformed differences -@item dct -sum of absolute DCT transformed differences -@item psnr -sum of squared quantization errors (avoid, low quality) -@item bit -number of bits needed for the block -@item rd -rate distortion optimal, slow -@item zero -0 -@item vsad -sum of absolute vertical differences -@item vsse -sum of squared vertical differences -@item nsse -noise preserving sum of squared differences -@item w53 -5/3 wavelet, only used in snow -@item w97 -9/7 wavelet, only used in snow -@item dctmax - -@item chroma - -@end table - -@item dia_size @var{integer} (@emph{encoding,video}) -Set diamond type & size for motion estimation. - -@item last_pred @var{integer} (@emph{encoding,video}) -Set amount of motion predictors from the previous frame. - -@item preme @var{integer} (@emph{encoding,video}) -Set pre motion estimation. - -@item precmp @var{integer} (@emph{encoding,video}) -Set pre motion estimation compare function. - -Possible values: -@table @samp -@item sad -sum of absolute differences, fast (default) -@item sse -sum of squared errors -@item satd -sum of absolute Hadamard transformed differences -@item dct -sum of absolute DCT transformed differences -@item psnr -sum of squared quantization errors (avoid, low quality) -@item bit -number of bits needed for the block -@item rd -rate distortion optimal, slow -@item zero -0 -@item vsad -sum of absolute vertical differences -@item vsse -sum of squared vertical differences -@item nsse -noise preserving sum of squared differences -@item w53 -5/3 wavelet, only used in snow -@item w97 -9/7 wavelet, only used in snow -@item dctmax - -@item chroma - -@end table - -@item pre_dia_size @var{integer} (@emph{encoding,video}) -Set diamond type & size for motion estimation pre-pass. - -@item subq @var{integer} (@emph{encoding,video}) -Set sub pel motion estimation quality. - -@item dtg_active_format @var{integer} - -@item me_range @var{integer} (@emph{encoding,video}) -Set limit motion vectors range (1023 for DivX player). - -@item ibias @var{integer} (@emph{encoding,video}) -Set intra quant bias. - -@item pbias @var{integer} (@emph{encoding,video}) -Set inter quant bias. - -@item color_table_id @var{integer} - -@item global_quality @var{integer} (@emph{encoding,audio,video}) - -@item coder @var{integer} (@emph{encoding,video}) - -Possible values: -@table @samp -@item vlc -variable length coder / huffman coder -@item ac -arithmetic coder -@item raw -raw (no encoding) -@item rle -run-length coder -@item deflate -deflate-based coder -@end table - -@item context @var{integer} (@emph{encoding,video}) -Set context model. - -@item slice_flags @var{integer} - -@item xvmc_acceleration @var{integer} - -@item mbd @var{integer} (@emph{encoding,video}) -Set macroblock decision algorithm (high quality mode). - -Possible values: -@table @samp -@item simple -use mbcmp (default) -@item bits -use fewest bits -@item rd -use best rate distortion -@end table - -@item stream_codec_tag @var{integer} - -@item sc_threshold @var{integer} (@emph{encoding,video}) -Set scene change threshold. - -@item lmin @var{integer} (@emph{encoding,video}) -Set min lagrange factor (VBR). - -@item lmax @var{integer} (@emph{encoding,video}) -Set max lagrange factor (VBR). - -@item nr @var{integer} (@emph{encoding,video}) -Set noise reduction. - -@item rc_init_occupancy @var{integer} (@emph{encoding,video}) -Set number of bits which should be loaded into the rc buffer before -decoding starts. - -@item flags2 @var{flags} (@emph{decoding/encoding,audio,video}) - -Possible values: -@table @samp -@item fast -Allow non spec compliant speedup tricks. -@item sgop -Deprecated, use mpegvideo private options instead. -@item noout -Skip bitstream encoding. -@item ignorecrop -Ignore cropping information from sps. -@item local_header -Place global headers at every keyframe instead of in extradata. -@item chunks -Frame data might be split into multiple chunks. -@item showall -Show all frames before the first keyframe. -@item skiprd -Deprecated, use mpegvideo private options instead. -@end table - -@item error @var{integer} (@emph{encoding,video}) - -@item qns @var{integer} (@emph{encoding,video}) -Deprecated, use mpegvideo private options instead. - -@item threads @var{integer} (@emph{decoding/encoding,video}) - -Possible values: -@table @samp -@item auto -detect a good number of threads -@end table - -@item me_threshold @var{integer} (@emph{encoding,video}) -Set motion estimation threshold. - -@item mb_threshold @var{integer} (@emph{encoding,video}) -Set macroblock threshold. - -@item dc @var{integer} (@emph{encoding,video}) -Set intra_dc_precision. - -@item nssew @var{integer} (@emph{encoding,video}) -Set nsse weight. - -@item skip_top @var{integer} (@emph{decoding,video}) -Set number of macroblock rows at the top which are skipped. - -@item skip_bottom @var{integer} (@emph{decoding,video}) -Set number of macroblock rows at the bottom which are skipped. - -@item profile @var{integer} (@emph{encoding,audio,video}) - -Possible values: -@table @samp -@item unknown - -@item aac_main - -@item aac_low - -@item aac_ssr - -@item aac_ltp - -@item aac_he - -@item aac_he_v2 - -@item aac_ld - -@item aac_eld - -@item mpeg2_aac_low - -@item mpeg2_aac_he - -@item dts - -@item dts_es - -@item dts_96_24 - -@item dts_hd_hra - -@item dts_hd_ma - -@end table - -@item level @var{integer} (@emph{encoding,audio,video}) - -Possible values: -@table @samp -@item unknown - -@end table - -@item lowres @var{integer} (@emph{decoding,audio,video}) -Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions. - -@item skip_threshold @var{integer} (@emph{encoding,video}) -Set frame skip threshold. - -@item skip_factor @var{integer} (@emph{encoding,video}) -Set frame skip factor. - -@item skip_exp @var{integer} (@emph{encoding,video}) -Set frame skip exponent. -Negative values behave identical to the corresponding positive ones, except -that the score is normalized. -Positive values exist primarly for compatibility reasons and are not so useful. - -@item skipcmp @var{integer} (@emph{encoding,video}) -Set frame skip compare function. - -Possible values: -@table @samp -@item sad -sum of absolute differences, fast (default) -@item sse -sum of squared errors -@item satd -sum of absolute Hadamard transformed differences -@item dct -sum of absolute DCT transformed differences -@item psnr -sum of squared quantization errors (avoid, low quality) -@item bit -number of bits needed for the block -@item rd -rate distortion optimal, slow -@item zero -0 -@item vsad -sum of absolute vertical differences -@item vsse -sum of squared vertical differences -@item nsse -noise preserving sum of squared differences -@item w53 -5/3 wavelet, only used in snow -@item w97 -9/7 wavelet, only used in snow -@item dctmax - -@item chroma - -@end table - -@item border_mask @var{float} (@emph{encoding,video}) -Increase the quantizer for macroblocks close to borders. - -@item mblmin @var{integer} (@emph{encoding,video}) -Set min macroblock lagrange factor (VBR). - -@item mblmax @var{integer} (@emph{encoding,video}) -Set max macroblock lagrange factor (VBR). - -@item mepc @var{integer} (@emph{encoding,video}) -Set motion estimation bitrate penalty compensation (1.0 = 256). - -@item skip_loop_filter @var{integer} (@emph{decoding,video}) -@item skip_idct @var{integer} (@emph{decoding,video}) -@item skip_frame @var{integer} (@emph{decoding,video}) - -Make decoder discard processing depending on the frame type selected -by the option value. - -@option{skip_loop_filter} skips frame loop filtering, @option{skip_idct} -skips frame IDCT/dequantization, @option{skip_frame} skips decoding. - -Possible values: -@table @samp -@item none -Discard no frame. - -@item default -Discard useless frames like 0-sized frames. - -@item noref -Discard all non-reference frames. - -@item bidir -Discard all bidirectional frames. - -@item nokey -Discard all frames excepts keyframes. - -@item all -Discard all frames. -@end table - -Default value is @samp{default}. - -@item bidir_refine @var{integer} (@emph{encoding,video}) -Refine the two motion vectors used in bidirectional macroblocks. - -@item brd_scale @var{integer} (@emph{encoding,video}) -Downscale frames for dynamic B-frame decision. - -@item keyint_min @var{integer} (@emph{encoding,video}) -Set minimum interval between IDR-frames. - -@item refs @var{integer} (@emph{encoding,video}) -Set reference frames to consider for motion compensation. - -@item chromaoffset @var{integer} (@emph{encoding,video}) -Set chroma qp offset from luma. - -@item trellis @var{integer} (@emph{encoding,audio,video}) -Set rate-distortion optimal quantization. - -@item sc_factor @var{integer} (@emph{encoding,video}) -Set value multiplied by qscale for each frame and added to -scene_change_score. - -@item mv0_threshold @var{integer} (@emph{encoding,video}) -@item b_sensitivity @var{integer} (@emph{encoding,video}) -Adjust sensitivity of b_frame_strategy 1. - -@item compression_level @var{integer} (@emph{encoding,audio,video}) -@item min_prediction_order @var{integer} (@emph{encoding,audio}) -@item max_prediction_order @var{integer} (@emph{encoding,audio}) -@item timecode_frame_start @var{integer} (@emph{encoding,video}) -Set GOP timecode frame start number, in non drop frame format. - -@item request_channels @var{integer} (@emph{decoding,audio}) -Set desired number of audio channels. - -@item bits_per_raw_sample @var{integer} -@item channel_layout @var{integer} (@emph{decoding/encoding,audio}) - -Possible values: -@table @samp -@end table -@item request_channel_layout @var{integer} (@emph{decoding,audio}) - -Possible values: -@table @samp -@end table -@item rc_max_vbv_use @var{float} (@emph{encoding,video}) -@item rc_min_vbv_use @var{float} (@emph{encoding,video}) -@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video}) -@item color_primaries @var{integer} (@emph{decoding/encoding,video}) -@item color_trc @var{integer} (@emph{decoding/encoding,video}) -@item colorspace @var{integer} (@emph{decoding/encoding,video}) -@item color_range @var{integer} (@emph{decoding/encoding,video}) -@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video}) - -@item log_level_offset @var{integer} -Set the log level offset. - -@item slices @var{integer} (@emph{encoding,video}) -Number of slices, used in parallelized encoding. - -@item thread_type @var{flags} (@emph{decoding/encoding,video}) -Select multithreading type. - -Possible values: -@table @samp -@item slice - -@item frame - -@end table -@item audio_service_type @var{integer} (@emph{encoding,audio}) -Set audio service type. - -Possible values: -@table @samp -@item ma -Main Audio Service -@item ef -Effects -@item vi -Visually Impaired -@item hi -Hearing Impaired -@item di -Dialogue -@item co -Commentary -@item em -Emergency -@item vo -Voice Over -@item ka -Karaoke -@end table - -@item request_sample_fmt @var{sample_fmt} (@emph{decoding,audio}) -Set sample format audio decoders should prefer. Default value is -@code{none}. - -@item pkt_timebase @var{rational number} - -@item sub_charenc @var{encoding} (@emph{decoding,subtitles}) -Set the input subtitles character encoding. - -@item field_order @var{field_order} (@emph{video}) -Set/override the field order of the video. -Possible values: -@table @samp -@item progressive -Progressive video -@item tt -Interlaced video, top field coded and displayed first -@item bb -Interlaced video, bottom field coded and displayed first -@item tb -Interlaced video, top coded first, bottom displayed first -@item bt -Interlaced video, bottom coded first, top displayed first -@end table - -@item skip_alpha @var{integer} (@emph{decoding,video}) -Set to 1 to disable processing alpha (transparency). This works like the -@samp{gray} flag in the @option{flags} option which skips chroma information -instead of alpha. Default is 0. -@end table - -@c man end CODEC OPTIONS - -@ifclear config-writeonly -@include decoders.texi -@end ifclear -@ifclear config-readonly -@include encoders.texi -@end ifclear diff --git a/doc/decoders.texi b/doc/decoders.texi index 9309b82a58..c3573a6315 100644 --- a/doc/decoders.texi +++ b/doc/decoders.texi @@ -1,10 +1,10 @@ @chapter Decoders @c man begin DECODERS -Decoders are configured elements in FFmpeg which allow the decoding of +Decoders are configured elements in Libav which allow the decoding of multimedia streams. -When you configure your FFmpeg build, all the supported native decoders +When you configure your Libav build, all the supported native decoders are enabled by default. Decoders requiring an external library must be enabled manually via the corresponding @code{--enable-lib} option. You can list all available decoders using the configure option @code{--list-decoders}. @@ -14,41 +14,11 @@ You can disable all the decoders with the configure option with the options @code{--enable-decoder=@var{DECODER}} / @code{--disable-decoder=@var{DECODER}}. -The option @code{-decoders} of the ff* tools will display the list of +The option @code{-decoders} of the av* tools will display the list of enabled decoders. @c man end DECODERS -@chapter Video Decoders -@c man begin VIDEO DECODERS - -A description of some of the currently available video decoders -follows. - -@section rawvideo - -Raw video decoder. - -This decoder decodes rawvideo streams. - -@subsection Options - -@table @option -@item top @var{top_field_first} -Specify the assumed field type of the input video. -@table @option -@item -1 -the video is assumed to be progressive (default) -@item 0 -bottom-field-first is assumed -@item 1 -top-field-first is assumed -@end table - -@end table - -@c man end VIDEO DECODERS - @chapter Audio Decoders @c man begin AUDIO DECODERS @@ -62,7 +32,7 @@ AC-3 audio decoder. This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet). -@subsection AC-3 Decoder Options +@subsubsection AC-3 Decoder Options @table @option @@ -83,151 +53,4 @@ Loud sounds are fully compressed. Soft sounds are enhanced. @end table -@section ffwavesynth - -Internal wave synthetizer. - -This decoder generates wave patterns according to predefined sequences. Its -use is purely internal and the format of the data it accepts is not publicly -documented. - -@section libcelt - -libcelt decoder wrapper. - -libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. -Requires the presence of the libcelt headers and library during configuration. -You need to explicitly configure the build with @code{--enable-libcelt}. - -@section libgsm - -libgsm decoder wrapper. - -libgsm allows libavcodec to decode the GSM full rate audio codec. Requires -the presence of the libgsm headers and library during configuration. You need -to explicitly configure the build with @code{--enable-libgsm}. - -This decoder supports both the ordinary GSM and the Microsoft variant. - -@section libilbc - -libilbc decoder wrapper. - -libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) -audio codec. Requires the presence of the libilbc headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libilbc}. - -@subsection Options - -The following option is supported by the libilbc wrapper. - -@table @option -@item enhance - -Enable the enhancement of the decoded audio when set to 1. The default -value is 0 (disabled). - -@end table - -@section libopencore-amrnb - -libopencore-amrnb decoder wrapper. - -libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate -Narrowband audio codec. Using it requires the presence of the -libopencore-amrnb headers and library during configuration. You need to -explicitly configure the build with @code{--enable-libopencore-amrnb}. - -An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB -without this library. - -@section libopencore-amrwb - -libopencore-amrwb decoder wrapper. - -libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate -Wideband audio codec. Using it requires the presence of the -libopencore-amrwb headers and library during configuration. You need to -explicitly configure the build with @code{--enable-libopencore-amrwb}. - -An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB -without this library. - -@section libopus - -libopus decoder wrapper. - -libopus allows libavcodec to decode the Opus Interactive Audio Codec. -Requires the presence of the libopus headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libopus}. - @c man end AUDIO DECODERS - -@chapter Subtitles Decoders -@c man begin SUBTILES DECODERS - -@section dvdsub - -This codec decodes the bitmap subtitles used in DVDs; the same subtitles can -also be found in VobSub file pairs and in some Matroska files. - -@subsection Options - -@table @option -@item palette -Specify the global palette used by the bitmaps. When stored in VobSub, the -palette is normally specified in the index file; in Matroska, the palette is -stored in the codec extra-data in the same format as in VobSub. In DVDs, the -palette is stored in the IFO file, and therefore not available when reading -from dumped VOB files. - -The format for this option is a string containing 16 24-bits hexadecimal -numbers (without 0x prefix) separated by comas, for example @code{0d00ee, -ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, -7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}. -@end table - -@section libzvbi-teletext - -Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext -subtitles. Requires the presence of the libzvbi headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libzvbi}. - -@subsection Options - -@table @option -@item txt_page -List of teletext page numbers to decode. You may use the special * string to -match all pages. Pages that do not match the specified list are dropped. -Default value is *. -@item txt_chop_top -Discards the top teletext line. Default value is 1. -@item txt_format -Specifies the format of the decoded subtitles. The teletext decoder is capable -of decoding the teletext pages to bitmaps or to simple text, you should use -"bitmap" for teletext pages, because certain graphics and colors cannot be -expressed in simple text. You might use "text" for teletext based subtitles if -your application can handle simple text based subtitles. Default value is -bitmap. -@item txt_left -X offset of generated bitmaps, default is 0. -@item txt_top -Y offset of generated bitmaps, default is 0. -@item txt_chop_spaces -Chops leading and trailing spaces and removes empty lines from the generated -text. This option is useful for teletext based subtitles where empty spaces may -be present at the start or at the end of the lines or empty lines may be -present between the subtitle lines because of double-sized teletext charactes. -Default value is 1. -@item txt_duration -Sets the display duration of the decoded teletext pages or subtitles in -miliseconds. Default value is 30000 which is 30 seconds. -@item txt_transparent -Force transparent background of the generated teletext bitmaps. Default value -is 0 which means an opaque (black) background. -@end table - -@c man end SUBTILES DECODERS diff --git a/doc/default.css b/doc/default.css deleted file mode 100644 index bf50200c28..0000000000 --- a/doc/default.css +++ /dev/null @@ -1,165 +0,0 @@ -a.summary-letter { - text-decoration: none; -} - -a { - color: #2D6198; -} - -a:visited { - color: #884488; -} - -#banner { - background-color: white; - position: relative; - text-align: center; -} - -#banner img { - margin-bottom: 1px; - margin-top: 5px; -} - -#body { - margin-left: 1em; - margin-right: 1em; -} - -body { - background-color: #313131; - margin: 0; - text-align: justify; -} - -.center { - margin-left: auto; - margin-right: auto; - text-align: center; -} - -#container { - background-color: white; - color: #202020; - margin-left: 1em; - margin-right: 1em; -} - -#footer { - text-align: center; -} - -h1 a, h2 a, h3 a, h4 a { - text-decoration: inherit; - color: inherit; -} - -h1, h2, h3, h4 { - padding-left: 0.4em; - border-radius: 4px; - padding-bottom: 0.25em; - padding-top: 0.25em; - border: 1px solid #6A996A; -} - -h1 { - background-color: #7BB37B; - color: #151515; - font-size: 1.2em; - padding-bottom: 0.3em; - padding-top: 0.3em; -} - -h2 { - color: #313131; - font-size: 1.0em; - background-color: #ABE3AB; -} - -h3 { - color: #313131; - font-size: 0.9em; - margin-bottom: -6px; - background-color: #BBF3BB; -} - -h4 { - color: #313131; - font-size: 0.8em; - margin-bottom: -8px; - background-color: #D1FDD1; -} - -img { - border: 0; -} - -#navbar { - background-color: #738073; - border-bottom: 1px solid #5C665C; - border-top: 1px solid #5C665C; - margin-top: 12px; - padding: 0.3em; - position: relative; - text-align: center; -} - -#navbar a, #navbar_secondary a { - color: white; - padding: 0.3em; - text-decoration: none; -} - -#navbar a:hover, #navbar_secondary a:hover { - background-color: #313131; - color: white; - text-decoration: none; -} - -#navbar_secondary { - background-color: #738073; - border-bottom: 1px solid #5C665C; - border-left: 1px solid #5C665C; - border-right: 1px solid #5C665C; - padding: 0.3em; - position: relative; - text-align: center; -} - -p { - margin-left: 1em; - margin-right: 1em; -} - -pre { - margin-left: 3em; - margin-right: 3em; - padding: 0.3em; - border: 1px solid #bbb; - background-color: #f7f7f7; -} - -dl dt { - font-weight: bold; -} - -#proj_desc { - font-size: 1.2em; -} - -#repos { - margin-left: 1em; - margin-right: 1em; - border-collapse: collapse; - border: solid 1px #6A996A; -} - -#repos th { - background-color: #7BB37B; - border: solid 1px #6A996A; -} - -#repos td { - padding: 0.2em; - border: solid 1px #6A996A; -} diff --git a/doc/demuxers.texi b/doc/demuxers.texi index d51b9d0748..2f2f464404 100644 --- a/doc/demuxers.texi +++ b/doc/demuxers.texi @@ -1,143 +1,93 @@ @chapter Demuxers @c man begin DEMUXERS -Demuxers are configured elements in FFmpeg that can read the +Demuxers are configured elements in Libav which allow to read the multimedia streams from a particular type of file. -When you configure your FFmpeg build, all the supported demuxers +When you configure your Libav build, all the supported demuxers are enabled by default. You can list all available ones using the -configure option @code{--list-demuxers}. +configure option "--list-demuxers". You can disable all the demuxers using the configure option -@code{--disable-demuxers}, and selectively enable a single demuxer with -the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it -with the option @code{--disable-demuxer=@var{DEMUXER}}. +"--disable-demuxers", and selectively enable a single demuxer with +the option "--enable-demuxer=@var{DEMUXER}", or disable it +with the option "--disable-demuxer=@var{DEMUXER}". -The option @code{-formats} of the ff* tools will display the list of +The option "-formats" of the av* tools will display the list of enabled demuxers. The description of some of the currently available demuxers follows. +@section image2 + +Image file demuxer. + +This demuxer reads from a list of image files specified by a pattern. + +The pattern may contain the string "%d" or "%0@var{N}d", which +specifies the position of the characters representing a sequential +number in each filename matched by the pattern. If the form +"%d0@var{N}d" is used, the string representing the number in each +filename is 0-padded and @var{N} is the total number of 0-padded +digits representing the number. The literal character '%' can be +specified in the pattern with the string "%%". + +If the pattern contains "%d" or "%0@var{N}d", the first filename of +the file list specified by the pattern must contain a number +inclusively contained between 0 and 4, all the following numbers must +be sequential. This limitation may be hopefully fixed. + +The pattern may contain a suffix which is used to automatically +determine the format of the images contained in the files. + +For example the pattern "img-%03d.bmp" will match a sequence of +filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ..., +@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a +sequence of filenames of the form @file{i%m%g-1.jpg}, +@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc. + +The size, the pixel format, and the format of each image must be the +same for all the files in the sequence. + +The following example shows how to use @command{avconv} for creating a +video from the images in the file sequence @file{img-001.jpeg}, +@file{img-002.jpeg}, ..., assuming an input framerate of 10 frames per +second: +@example +avconv -i 'img-%03d.jpeg' -r 10 out.mkv +@end example + +Note that the pattern must not necessarily contain "%d" or +"%0@var{N}d", for example to convert a single image file +@file{img.jpeg} you can employ the command: +@example +avconv -i img.jpeg img.png +@end example + +@table @option +@item -pixel_format @var{format} +Set the pixel format (for raw image) +@item -video_size @var{size} +Set the frame size (for raw image) +@item -framerate @var{rate} +Set the frame rate +@item -loop @var{bool} +Loop over the images +@item -start_number @var{start} +Specify the first number in the sequence +@end table + @section applehttp Apple HTTP Live Streaming demuxer. This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant index number. By setting -the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), +the discard flags on AVStreams (by pressing 'a' or 'v' in avplay), the caller can decide which variant streams to actually receive. The total bitrate of the variant that the stream belongs to is available in a metadata key named "variant_bitrate". -@section asf - -Advanced Systems Format demuxer. - -This demuxer is used to demux ASF files and MMS network streams. - -@table @option -@item -no_resync_search @var{bool} -Do not try to resynchronize by looking for a certain optional start code. -@end table - -@anchor{concat} -@section concat - -Virtual concatenation script demuxer. - -This demuxer reads a list of files and other directives from a text file and -demuxes them one after the other, as if all their packet had been muxed -together. - -The timestamps in the files are adjusted so that the first file starts at 0 -and each next file starts where the previous one finishes. Note that it is -done globally and may cause gaps if all streams do not have exactly the same -length. - -All files must have the same streams (same codecs, same time base, etc.). - -The duration of each file is used to adjust the timestamps of the next file: -if the duration is incorrect (because it was computed using the bit-rate or -because the file is truncated, for example), it can cause artifacts. The -@code{duration} directive can be used to override the duration stored in -each file. - -@subsection Syntax - -The script is a text file in extended-ASCII, with one directive per line. -Empty lines, leading spaces and lines starting with '#' are ignored. The -following directive is recognized: - -@table @option - -@item @code{file @var{path}} -Path to a file to read; special characters and spaces must be escaped with -backslash or single quotes. - -All subsequent file-related directives apply to that file. - -@item @code{ffconcat version 1.0} -Identify the script type and version. It also sets the @option{safe} option -to 1 if it was to its default -1. - -To make FFmpeg recognize the format automatically, this directive must -appears exactly as is (no extra space or byte-order-mark) on the very first -line of the script. - -@item @code{duration @var{dur}} -Duration of the file. This information can be specified from the file; -specifying it here may be more efficient or help if the information from the -file is not available or accurate. - -If the duration is set for all files, then it is possible to seek in the -whole concatenated video. - -@item @code{stream} -Introduce a stream in the virtual file. -All subsequent stream-related directives apply to the last introduced -stream. -Some streams properties must be set in order to allow identifying the -matching streams in the subfiles. -If no streams are defined in the script, the streams from the first file are -copied. - -@item @code{exact_stream_id @var{id}} -Set the id of the stream. -If this directive is given, the string with the corresponding id in the -subfiles will be used. -This is especially useful for MPEG-PS (VOB) files, where the order of the -streams is not reliable. - -@end table - -@subsection Options - -This demuxer accepts the following option: - -@table @option - -@item safe -If set to 1, reject unsafe file paths. A file path is considered safe if it -does not contain a protocol specification and is relative and all components -only contain characters from the portable character set (letters, digits, -period, underscore and hyphen) and have no period at the beginning of a -component. - -If set to 0, any file name is accepted. - -The default is -1, it is equivalent to 1 if the format was automatically -probed and 0 otherwise. - -@item auto_convert -If set to 1, try to perform automatic conversions on packet data to make the -streams concatenable. - -Currently, the only conversion is adding the h264_mp4toannexb bitstream -filter to H.264 streams in MP4 format. This is necessary in particular if -there are resolution changes. - -@end table - @section flv Adobe Flash Video Format demuxer. @@ -149,253 +99,15 @@ This demuxer is used to demux FLV files and RTMP network streams. Allocate the streams according to the onMetaData array content. @end table -@section libgme +@section asf -The Game Music Emu library is a collection of video game music file emulators. +Advanced Systems Format demuxer. -See @url{http://code.google.com/p/game-music-emu/} for more information. - -Some files have multiple tracks. The demuxer will pick the first track by -default. The @option{track_index} option can be used to select a different -track. Track indexes start at 0. The demuxer exports the number of tracks as -@var{tracks} meta data entry. - -For very large files, the @option{max_size} option may have to be adjusted. - -@section libquvi - -Play media from Internet services using the quvi project. - -The demuxer accepts a @option{format} option to request a specific quality. It -is by default set to @var{best}. - -See @url{http://quvi.sourceforge.net/} for more information. - -FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be -enabled. - -@section image2 - -Image file demuxer. - -This demuxer reads from a list of image files specified by a pattern. -The syntax and meaning of the pattern is specified by the -option @var{pattern_type}. - -The pattern may contain a suffix which is used to automatically -determine the format of the images contained in the files. - -The size, the pixel format, and the format of each image must be the -same for all the files in the sequence. - -This demuxer accepts the following options: -@table @option -@item framerate -Set the frame rate for the video stream. It defaults to 25. -@item loop -If set to 1, loop over the input. Default value is 0. -@item pattern_type -Select the pattern type used to interpret the provided filename. - -@var{pattern_type} accepts one of the following values. -@table @option -@item sequence -Select a sequence pattern type, used to specify a sequence of files -indexed by sequential numbers. - -A sequence pattern may contain the string "%d" or "%0@var{N}d", which -specifies the position of the characters representing a sequential -number in each filename matched by the pattern. If the form -"%d0@var{N}d" is used, the string representing the number in each -filename is 0-padded and @var{N} is the total number of 0-padded -digits representing the number. The literal character '%' can be -specified in the pattern with the string "%%". - -If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of -the file list specified by the pattern must contain a number -inclusively contained between @var{start_number} and -@var{start_number}+@var{start_number_range}-1, and all the following -numbers must be sequential. - -For example the pattern "img-%03d.bmp" will match a sequence of -filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ..., -@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a -sequence of filenames of the form @file{i%m%g-1.jpg}, -@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc. - -Note that the pattern must not necessarily contain "%d" or -"%0@var{N}d", for example to convert a single image file -@file{img.jpeg} you can employ the command: -@example -ffmpeg -i img.jpeg img.png -@end example - -@item glob -Select a glob wildcard pattern type. - -The pattern is interpreted like a @code{glob()} pattern. This is only -selectable if libavformat was compiled with globbing support. - -@item glob_sequence @emph{(deprecated, will be removed)} -Select a mixed glob wildcard/sequence pattern. - -If your version of libavformat was compiled with globbing support, and -the provided pattern contains at least one glob meta character among -@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is -interpreted like a @code{glob()} pattern, otherwise it is interpreted -like a sequence pattern. - -All glob special characters @code{%*?[]@{@}} must be prefixed -with "%". To escape a literal "%" you shall use "%%". - -For example the pattern @code{foo-%*.jpeg} will match all the -filenames prefixed by "foo-" and terminating with ".jpeg", and -@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with -"foo-", followed by a sequence of three characters, and terminating -with ".jpeg". - -This pattern type is deprecated in favor of @var{glob} and -@var{sequence}. -@end table - -Default value is @var{glob_sequence}. -@item pixel_format -Set the pixel format of the images to read. If not specified the pixel -format is guessed from the first image file in the sequence. -@item start_number -Set the index of the file matched by the image file pattern to start -to read from. Default value is 0. -@item start_number_range -Set the index interval range to check when looking for the first image -file in the sequence, starting from @var{start_number}. Default value -is 5. -@item ts_from_file -If set to 1, will set frame timestamp to modification time of image file. Note -that monotonity of timestamps is not provided: images go in the same order as -without this option. Default value is 0. -If set to 2, will set frame timestamp to the modification time of the image file in -nanosecond precision. -@item video_size -Set the video size of the images to read. If not specified the video -size is guessed from the first image file in the sequence. -@end table - -@subsection Examples - -@itemize -@item -Use @command{ffmpeg} for creating a video from the images in the file -sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an -input frame rate of 10 frames per second: -@example -ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv -@end example - -@item -As above, but start by reading from a file with index 100 in the sequence: -@example -ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv -@end example - -@item -Read images matching the "*.png" glob pattern , that is all the files -terminating with the ".png" suffix: -@example -ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv -@end example -@end itemize - -@section mpegts - -MPEG-2 transport stream demuxer. +This demuxer is used to demux ASF files and MMS network streams. @table @option - -@item fix_teletext_pts -Overrides teletext packet PTS and DTS values with the timestamps calculated -from the PCR of the first program which the teletext stream is part of and is -not discarded. Default value is 1, set this option to 0 if you want your -teletext packet PTS and DTS values untouched. +@item -no_resync_search @var{bool} +Do not try to resynchronize by looking for a certain optional start code. @end table -@section rawvideo - -Raw video demuxer. - -This demuxer allows one to read raw video data. Since there is no header -specifying the assumed video parameters, the user must specify them -in order to be able to decode the data correctly. - -This demuxer accepts the following options: -@table @option - -@item framerate -Set input video frame rate. Default value is 25. - -@item pixel_format -Set the input video pixel format. Default value is @code{yuv420p}. - -@item video_size -Set the input video size. This value must be specified explicitly. -@end table - -For example to read a rawvideo file @file{input.raw} with -@command{ffplay}, assuming a pixel format of @code{rgb24}, a video -size of @code{320x240}, and a frame rate of 10 images per second, use -the command: -@example -ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw -@end example - -@section sbg - -SBaGen script demuxer. - -This demuxer reads the script language used by SBaGen -@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG -script looks like that: -@example --SE -a: 300-2.5/3 440+4.5/0 -b: 300-2.5/0 440+4.5/3 -off: - -NOW == a -+0:07:00 == b -+0:14:00 == a -+0:21:00 == b -+0:30:00 off -@end example - -A SBG script can mix absolute and relative timestamps. If the script uses -either only absolute timestamps (including the script start time) or only -relative ones, then its layout is fixed, and the conversion is -straightforward. On the other hand, if the script mixes both kind of -timestamps, then the @var{NOW} reference for relative timestamps will be -taken from the current time of day at the time the script is read, and the -script layout will be frozen according to that reference. That means that if -the script is directly played, the actual times will match the absolute -timestamps up to the sound controller's clock accuracy, but if the user -somehow pauses the playback or seeks, all times will be shifted accordingly. - -@section tedcaptions - -JSON captions used for @url{http://www.ted.com/, TED Talks}. - -TED does not provide links to the captions, but they can be guessed from the -page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree -contains a bookmarklet to expose them. - -This demuxer accepts the following option: -@table @option -@item start_time -Set the start time of the TED talk, in milliseconds. The default is 15000 -(15s). It is used to sync the captions with the downloadable videos, because -they include a 15s intro. -@end table - -Example: convert the captions to a format most players understand: -@example -ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt -@end example - -@c man end DEMUXERS +@c man end INPUT DEVICES diff --git a/doc/developer.texi b/doc/developer.texi index 1e1d3b8c8f..1ffdff10c9 100644 --- a/doc/developer.texi +++ b/doc/developer.texi @@ -11,44 +11,87 @@ @chapter Developers Guide -@section Notes for external developers +@section API -This document is mostly useful for internal FFmpeg developers. -External developers who need to use the API in their application should -refer to the API doxygen documentation in the public headers, and -check the examples in @file{doc/examples} and in the source code to -see how the public API is employed. - -You can use the FFmpeg libraries in your commercial program, but you -are encouraged to @emph{publish any patch you make}. In this case the -best way to proceed is to send your patches to the ffmpeg-devel -mailing list following the guidelines illustrated in the remainder of -this document. - -For more detailed legal information about the use of FFmpeg in -external programs read the @file{LICENSE} file in the source tree and -consult @url{http://ffmpeg.org/legal.html}. - -@section Contributing - -There are 3 ways by which code gets into ffmpeg. @itemize @bullet -@item Submitting Patches to the main developer mailing list - see @ref{Submitting patches} for details. -@item Directly committing changes to the main tree. -@item Committing changes to a git clone, for example on github.com or - gitorious.org. And asking us to merge these changes. +@item libavcodec is the library containing the codecs (both encoding and +decoding). Look at @file{libavcodec/apiexample.c} to see how to use it. + +@item libavformat is the library containing the file format handling (mux and +demux code for several formats). Look at @file{avplay.c} to use it in a +player. See @file{libavformat/output-example.c} to use it to generate +audio or video streams. @end itemize -Whichever way, changes should be reviewed by the maintainer of the code -before they are committed. And they should follow the @ref{Coding Rules}. -The developer making the commit and the author are responsible for their changes -and should try to fix issues their commit causes. +@section Integrating libav in your program + +Shared libraries should be used whenever is possible in order to reduce +the effort distributors have to pour to support programs and to ensure +only the public API is used. + +You can use Libav in your commercial program, but you must abide to the +license, LGPL or GPL depending on the specific features used, please refer +to @uref{http://libav.org/legal.html, our legal page} for a quick checklist and to +the following links for the exact text of each license: +@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.GPLv2, GPL version 2}, +@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.GPLv3, GPL version 3}, +@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.LGPLv2.1, LGPL version 2.1}, +@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.LGPLv3, LGPL version 3}. +Any modification to the source code can be suggested for inclusion. +The best way to proceed is to send your patches to the +@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} +mailing list. @anchor{Coding Rules} @section Coding Rules @subsection Code formatting conventions +The code is written in K&R C style. That means the following: + +@itemize @bullet +@item +The control statements are formatted by putting space between the statement +and parenthesis in the following way: +@example +for (i = 0; i < filter->input_count; i++) @{ +@end example + +@item +The case statement is always located at the same level as the switch itself: +@example +switch (link->init_state) @{ +case AVLINK_INIT: + continue; +case AVLINK_STARTINIT: + av_log(filter, AV_LOG_INFO, "circular filter chain detected"); + return 0; +@end example + +@item +Braces in function declarations are written on the new line: +@example +const char *avfilter_configuration(void) +@{ + return LIBAV_CONFIGURATION; +@} +@end example + +@item +Do not check for NULL values by comparison, @samp{if (p)} and +@samp{if (!p)} are correct; @samp{if (p == NULL)} and @samp{if (p != NULL)} +are not. + +@item +In case of a single-statement if, no curly braces are required: +@example +if (!pic || !picref) + goto fail; +@end example + +@item +Do not put spaces immediately inside parentheses. @samp{if (ret)} is +a valid style; @samp{if ( ret )} is not. +@end itemize There are the following guidelines regarding the indentation in files: @@ -67,7 +110,7 @@ and only if this improves readability. @end itemize The presentation is one inspired by 'indent -i4 -kr -nut'. -The main priority in FFmpeg is simplicity and small code size in order to +The main priority in Libav is simplicity and small code size in order to minimize the bug count. @subsection Comments @@ -112,7 +155,7 @@ int myfunc(int my_parameter) @subsection C language features -FFmpeg is programmed in the ISO C90 language with a few additional +Libav is programmed in the ISO C90 language with a few additional features from ISO C99, namely: @itemize @bullet @@ -154,8 +197,8 @@ GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}). @subsection Naming conventions All names should be composed with underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is an acceptable function name and -@samp{AVFilterGetVideo} is not. The exception from this are type names, like -for example structs and enums; they should always be in the CamelCase +@samp{AVFilterGetVideo} is not. The only exception are structure +names; they should always be CamelCase. There are the following conventions for naming variables and functions: @@ -178,13 +221,8 @@ across multiple libraries, use @code{avpriv_} as prefix, for example, @samp{avpriv_aac_parse_header}. @item -Each library has its own prefix for public symbols, in addition to the -commonly used @code{av_} (@code{avformat_} for libavformat, -@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc). -Check the existing code and choose names accordingly. -Note that some symbols without these prefixes are also exported for -retro-compatibility reasons. These exceptions are declared in the -@code{lib/lib.v} files. +For externally visible symbols, each library has its own prefix. Check +the existing code and choose names accordingly. @end itemize Furthermore, name space reserved for the system should not be invaded. @@ -208,10 +246,10 @@ should also be avoided if they don't make the code easier to understand. @end itemize @subsection Editor configuration -In order to configure Vim to follow FFmpeg formatting conventions, paste +In order to configure Vim to follow Libav formatting conventions, paste the following snippet into your @file{.vimrc}: @example -" indentation rules for FFmpeg: 4 spaces, no tabs +" Indentation rules for Libav: 4 spaces, no tabs. set expandtab set shiftwidth=4 set softtabstop=4 @@ -228,7 +266,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@ 5 lines) chunk of code, -then either do NOT change the indentation of the inner part within (do not -move it to the right)! or do so in a separate commit - -@item -Always fill out the commit log message. Describe in a few lines what you -changed and why. You can refer to mailing list postings if you fix a -particular bug. Comments such as "fixed!" or "Changed it." are unacceptable. -Recommended format: -area changed: Short 1 line description - -details describing what and why and giving references. - -@item -Make sure the author of the commit is set correctly. (see git commit --author) -If you apply a patch, send an -answer to ffmpeg-devel (or wherever you got the patch from) saying that -you applied the patch. +Patches that change behavior of the programs (renaming options etc) or +public API or ABI should be discussed in depth and possible few days should +pass between discussion and commit. +Changes to the build system (Makefiles, configure script) which alter +the expected behavior should be considered in the same regard. @item When applying patches that have been discussed (at length) on the mailing list, reference the thread in the log message. @item -Do NOT commit to code actively maintained by others without permission. -Send a patch to ffmpeg-devel instead. If no one answers within a reasonable -timeframe (12h for build failures and security fixes, 3 days small changes, -1 week for big patches) then commit your patch if you think it is OK. -Also note, the maintainer can simply ask for more time to review! - -@item -Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits -are sent there and reviewed by all the other developers. Bugs and possible -improvements or general questions regarding commits are discussed there. We -expect you to react if problems with your code are uncovered. +Subscribe to the +@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} and +@uref{https://lists.libav.org/mailman/listinfo/libav-commits, libav-commits} +mailing lists. +Bugs and possible improvements or general questions regarding commits +are discussed on libav-devel. We expect you to react if problems with +your code are uncovered. @item Update the documentation if you change behavior or add features. If you are -unsure how best to do this, send a patch to ffmpeg-devel, the documentation -maintainer(s) will review and commit your stuff. +unsure how best to do this, send an [RFC] patch to libav-devel. @item -Try to keep important discussions and requests (also) on the public -developer mailing list, so that all developers can benefit from them. +All discussions and decisions should be reported on the public developer +mailing list, so that there is a reference to them. +Other media (e.g. IRC) should be used for coordination and immediate +collaboration. @item Never write to unallocated memory, never write over the end of arrays, always check values read from some untrusted source before using them -as array index or other risky things. +as array index or other risky things. Always use valgrind to double-check. @item -Remember to check if you need to bump versions for the specific libav* +Remember to check if you need to bump versions for the specific libav parts (libavutil, libavcodec, libavformat) you are changing. You need to change the version integer. Incrementing the first component means no backward compatibility to @@ -374,59 +385,52 @@ Incrementing the second component means backward compatible change (e.g. addition of a function to the public API or extension of an existing data structure). Incrementing the third component means a noteworthy binary compatible -change (e.g. encoder bug fix that matters for the decoder). The third -component always starts at 100 to distinguish FFmpeg from Libav. +change (e.g. encoder bug fix that matters for the decoder). @item -Compiler warnings indicate potential bugs or code with bad style. If a type of -warning always points to correct and clean code, that warning should -be disabled, not the code changed. -Thus the remaining warnings can either be bugs or correct code. +Compiler warnings indicate potential bugs or code with bad style. If it is a bug, the bug has to be fixed. If it is not, the code should be changed to not generate a warning unless that causes a slowdown or obfuscates the code. +If a type of warning leads to too many false positives, that warning +should be disabled, not the code changed. @item -Make sure that no parts of the codebase that you maintain are missing from the -@file{MAINTAINERS} file. If something that you want to maintain is missing add it with -your name after it. -If at some point you no longer want to maintain some code, then please help -finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file. +If you add a new file, give it a proper license header. Do not copy and +paste it from a random place, use an existing file as template. @end enumerate We think our rules are not too hard. If you have comments, contact us. -@anchor{Submitting patches} @section Submitting patches First, read the @ref{Coding Rules} above if you did not yet, in particular the rules regarding patch submission. -When you submit your patch, please use @code{git format-patch} or -@code{git send-email}. We cannot read other diffs :-) - -Also please do not submit a patch which contains several unrelated changes. +As stated already, please do not submit a patch which contains several +unrelated changes. Split it into separate, self-contained pieces. This does not mean splitting file by file. Instead, make the patch as small as possible while still keeping it as a logical unit that contains an individual change, even if it spans multiple files. This makes reviewing your patches much easier for us and greatly increases your chances of getting your patch applied. -Use the patcheck tool of FFmpeg to check your patch. +Use the patcheck tool of Libav to check your patch. The tool is located in the tools directory. -Run the @ref{Regression tests} before submitting a patch in order to verify +Run the @ref{Regression Tests} before submitting a patch in order to verify it does not cause unexpected problems. It also helps quite a bit if you tell us what the patch does (for example 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant -and has no lrint()') +and has no lrint()'). This kind of explanation should be the body of the +commit message. Also please if you send several patches, send each patch as a separate mail, do not attach several unrelated patches to the same mail. Patches should be posted to the -@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel} +@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} mailing list. Use @code{git send-email} when possible since it will properly send patches without requiring extra care. If you cannot, then send patches as base64-encoded attachments, so your patch is not trashed during @@ -435,8 +439,8 @@ transmission. Your patch will be reviewed on the mailing list. You will likely be asked to make some changes and are expected to send in an improved version that incorporates the requests from the review. This process may go through -several iterations. Once your patch is deemed good enough, some developer -will pick it up and commit it to the official FFmpeg tree. +several iterations. Once your patch is deemed good enough, it will be +committed to the official Libav tree. Give us a few days to react. But if some time passes without reaction, send a reminder by email. Your patch should eventually be dealt with. @@ -470,8 +474,8 @@ even if it is only a decoder? @item Did you add a rule to compile the appropriate files in the Makefile? -Remember to do this even if you're just adding a format to a file that is -already being compiled by some other rule, like a raw demuxer. +Remember to do this even if you are just adding a format to a file that +is already being compiled by some other rule, like a raw demuxer. @item Did you add an entry to the table of supported formats or codecs in @@ -498,25 +502,15 @@ Did you make sure it compiles standalone, i.e. with @enumerate @item -Does @code{make fate} pass with the patch applied? +Does @code{make check} pass with the patch applied? @item -Was the patch generated with git format-patch or send-email? +Is the patch against latest Libav git master branch? @item -Did you sign off your patch? (git commit -s) -See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning -of sign off. - -@item -Did you provide a clear git commit log message? - -@item -Is the patch against latest FFmpeg git master branch? - -@item -Are you subscribed to ffmpeg-devel? -(the list is subscribers only due to spam) +Are you subscribed to the +@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} +mailing list? (Only list subscribers are allowed to post.) @item Have you checked that the changes are minimal, so that the same cannot be @@ -559,7 +553,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug? If the patch fixes a bug, did you provide enough information, including a sample, so the bug can be reproduced and the fix can be verified? Note please do not attach samples >100k to mails but rather provide a -URL, you can upload to ftp://upload.ffmpeg.org +URL, you can upload to ftp://upload.libav.org @item Did you provide a verbose summary about what the patch does change? @@ -577,7 +571,7 @@ patch easily? @item If you added a new file, did you insert a license header? It should be -taken from FFmpeg, not randomly copied and pasted from somewhere else. +taken from Libav, not randomly copied and pasted from somewhere else. @item You should maintain alphabetical order in alphabetically ordered lists as @@ -587,25 +581,17 @@ long as doing so does not break API/ABI compatibility. Lines with similar content should be aligned vertically when doing so improves readability. -@item -Consider to add a regression test for your code. - -@item -If you added YASM code please check that things still work with --disable-yasm - @item Make sure you check the return values of function and return appropriate -error codes. Especially memory allocation functions like @code{av_malloc()} +error codes. Especially memory allocation functions like @code{malloc()} are notoriously left unchecked, which is a serious problem. - -@item -Test your code with valgrind and or Address Sanitizer to ensure it's free -of leaks, out of array accesses, etc. @end enumerate @section Patch review process -All patches posted to ffmpeg-devel will be reviewed, unless they contain a +All patches posted to the +@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} +mailing list will be reviewed, unless they contain a clear note that the patch is not for the git master branch. Reviews and comments will be posted as replies to the patch on the mailing list. The patch submitter then has to take care of every comment, @@ -619,43 +605,27 @@ After a patch is approved it will be committed to the repository. We will review all submitted patches, but sometimes we are quite busy so especially for large patches this can take several weeks. -If you feel that the review process is too slow and you are willing to try to -take over maintainership of the area of code you change then just clone -git master and maintain the area of code there. We will merge each area from -where its best maintained. +When resubmitting patches, if their size grew or during the review different +issues arisen please split the patch so each issue has a specific patch. -When resubmitting patches, please do not make any significant changes -not related to the comments received during review. Such patches will -be rejected. Instead, submit significant changes or new features as -separate patches. +@anchor{Regression Tests} +@section Regression Tests -@anchor{Regression tests} -@section Regression tests +Before submitting a patch (or committing to the repository), you should at +least make sure that it does not break anything. -Before submitting a patch (or committing to the repository), you should at least -test that you did not break anything. +If the code changed has already a test present in FATE you should run it, +otherwise it is advised to add it. -Running 'make fate' accomplishes this, please see @url{fate.html} for details. - -[Of course, some patches may change the results of the regression tests. In -this case, the reference results of the regression tests shall be modified -accordingly]. - -@subsection Adding files to the fate-suite dataset - -When there is no muxer or encoder available to generate test media for a -specific test then the media has to be inlcuded in the fate-suite. -First please make sure that the sample file is as small as possible to test the -respective decoder or demuxer sufficiently. Large files increase network -bandwidth and disk space requirements. -Once you have a working fate test and fate sample, provide in the commit -message or introductionary message for the patch series that you post to -the ffmpeg-devel mailing list, a direct link to download the sample media. +Improvements to codec or demuxer might change the FATE results. Make sure +to commit the update reference with the change and to explain in the comment +why the expected result changed. +Please refer to @url{fate.html}. @subsection Visualizing Test Coverage -The FFmpeg build system allows visualizing the test coverage in an easy +The Libav build system allows visualizing the test coverage in an easy manner with the coverage tools @code{gcov}/@code{lcov}. This involves the following steps: @@ -667,7 +637,7 @@ the following steps: @item Run your test case, either manually or via FATE. This can be either the full FATE regression suite, or any arbitrary invocation of any - front-end tool provided by FFmpeg, in any combination. + front-end tool provided by Libav, in any combination. @item Run @code{make lcov} to generate coverage data in HTML format. @@ -696,11 +666,11 @@ your configure line instead. @anchor{Release process} @section Release process -FFmpeg maintains a set of @strong{release branches}, which are the +Libav maintains a set of @strong{release branches}, which are the recommended deliverable for system integrators and distributors (such as -Linux distributions, etc.). At regular times, a @strong{release +Linux distributions, etc.). At irregular times, a @strong{release manager} prepares, tests and publishes tarballs on the -@url{http://ffmpeg.org} website. +@url{http://libav.org} website. There are two kinds of releases: @@ -715,7 +685,7 @@ which are named @code{release/X}, with @code{X} being the release version number. @end enumerate -Note that we promise to our users that shared libraries from any FFmpeg +Note that we promise to our users that shared libraries from any Libav release never break programs that have been @strong{compiled} against previous versions of @strong{the same release series} in any case! @@ -723,7 +693,7 @@ However, from time to time, we do make API changes that require adaptations in applications. Such changes are only allowed in (new) major releases and require further steps such as bumping library version numbers and/or adjustments to the symbol versioning file. Please discuss such changes -on the @strong{ffmpeg-devel} mailing list in time to allow forward planning. +on the @strong{libav-devel} mailing list in time to allow forward planning. @anchor{Criteria for Point Releases} @subsection Criteria for Point Releases @@ -737,7 +707,7 @@ Fixes a security issue, preferably identified by a @strong{CVE number} issued by @url{http://cve.mitre.org/}. @item -Fixes a documented bug in @url{https://trac.ffmpeg.org}. +Fixes a documented bug in @url{http://bugzilla.libav.org}. @item Improves the included documentation. @@ -749,6 +719,11 @@ point releases of the same release branch. The order for checking the rules is (1 OR 2 OR 3) AND 4. +All Libav developers are welcome to nominate commits that they push to +@code{master} by mailing the @strong{libav-stable} mailing list. The +easiest way to do so is to include @code{CC: libav-stable@@libav.org} in +the commit message. + @subsection Release Checklist @@ -760,31 +735,43 @@ Ensure that the @file{RELEASE} file contains the version number for the upcoming release. @item -Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}. +File a release tracking bug in @url{http://bugzilla.libav.org}. Make +sure that the bug has an alias named @code{ReleaseX.Y} for the +@code{X.Y} release. @item Announce the intent to do a release to the mailing list. @item -Make sure all relevant security fixes have been backported. See -@url{https://ffmpeg.org/security.html}. +Reassign unresolved blocking bugs from previous release +tracking bugs to the new bug. + +@item +Review patch nominations that reach the @strong{libav-stable} +mailing list, and push patches that fulfill the stable release +criteria to the release branch. @item Ensure that the FATE regression suite still passes in the release branch on at least @strong{i386} and @strong{amd64} -(cf. @ref{Regression tests}). +(cf. @ref{Regression Tests}). @item -Prepare the release tarballs in @code{bz2} and @code{gz} formats, and -supplementing files that contain @code{gpg} signatures +Prepare the release tarballs in @code{xz} and @code{gz} formats, and +supplementing files that contain @code{md5} and @code{sha1} +checksums. @item -Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and -push an annotated tag in the form @code{nX}, with @code{X} +Publish the tarballs at @url{http://libav.org/releases}. Create and +push an annotated tag in the form @code{vX}, with @code{X} containing the version number. @item -Propose and send a patch to the @strong{ffmpeg-devel} mailing list +Build the tarballs with the Windows binaries, and publish them at +@url{http://win32.libav.org/releases}. + +@item +Propose and send a patch to the @strong{libav-devel} mailing list with a news entry for the website. @item diff --git a/doc/devices.texi b/doc/devices.texi deleted file mode 100644 index 5e74a962d7..0000000000 --- a/doc/devices.texi +++ /dev/null @@ -1,25 +0,0 @@ -@chapter Device Options -@c man begin DEVICE OPTIONS - -The libavdevice library provides the same interface as -libavformat. Namely, an input device is considered like a demuxer, and -an output device like a muxer, and the interface and generic device -options are the same provided by libavformat (see the ffmpeg-formats -manual). - -In addition each input or output device may support so-called private -options, which are specific for that component. - -Options may be set by specifying -@var{option} @var{value} in the -FFmpeg tools, or by setting the value explicitly in the device -@code{AVFormatContext} options or using the @file{libavutil/opt.h} API -for programmatic use. - -@c man end DEVICE OPTIONS - -@ifclear config-writeonly -@include indevs.texi -@end ifclear -@ifclear config-readonly -@include outdevs.texi -@end ifclear diff --git a/doc/doxy-wrapper.sh b/doc/doxy-wrapper.sh index 13be8cb950..d38dd0bcdd 100755 --- a/doc/doxy-wrapper.sh +++ b/doc/doxy-wrapper.sh @@ -9,4 +9,7 @@ doxygen - < li > a { + display: block; +} +.tablist > li > a:hover { + text-decoration: none; + background-color: #eeeeee; +} +.tablist > .pull-right { + float: right; +} +.tablist-header { + display: block; + padding: 3px 15px; + font-size: 11px; + font-weight: bold; + line-height: 20px; + color: #999999; + text-shadow: 0 1px 0 rgba(255, 255, 255, 0.5); + text-transform: uppercase; +} +.tablist li + .tablist-header { + margin-top: 9px; +} +.tablist-list { + padding-left: 15px; + padding-right: 15px; + margin-bottom: 0; +} +.tablist-list > li > a, +.tablist-list .tablist-header { + margin-left: -15px; + margin-right: -15px; + text-shadow: 0 1px 0 rgba(255, 255, 255, 0.5); +} +.tablist-list > li > a { + padding: 3px 15px; +} +.tablist-list > .current > a, +.tablist-list > .current > a:hover { + color: #ffffff; + text-shadow: 0 -1px 0 rgba(0, 0, 0, 0.2); + background-color: #0088cc; +} +.tablist-list [class^="icon-"] { + margin-right: 2px; +} +.tablist-list .divider { + *width: 100%; + height: 1px; + margin: 9px 1px; + *margin: -5px 0 5px; + overflow: hidden; + background-color: #e5e5e5; + border-bottom: 1px solid #ffffff; +} +.tablist-tabs, +.tablist { + *zoom: 1; +} +.tablist-tabs:before, +.tablist:before, +.tablist-tabs:after, +.tablist:after { + display: table; + content: ""; + line-height: 0; +} +.tablist-tabs:after, +.tablist:after { + clear: both; +} +.tablist-tabs > li, +.tablist > li { + float: left; +} +.tablist-tabs > li > a, +.tablist > li > a { + padding-right: 12px; + padding-left: 12px; + margin-right: 2px; + line-height: 14px; +} +.tablist-tabs { + border-bottom: 1px solid #ddd; +} +.tablist-tabs > li { + margin-bottom: -1px; +} +.tablist-tabs > li > a { + padding-top: 8px; + padding-bottom: 8px; + line-height: 20px; + border: 1px solid transparent; + -webkit-border-radius: 4px 4px 0 0; + -moz-border-radius: 4px 4px 0 0; + border-radius: 4px 4px 0 0; +} +.tablist-tabs > li > a:hover { + border-color: #eeeeee #eeeeee #dddddd; +} +.tablist-tabs > .current > a, +.tablist-tabs > .current > a:hover { + color: #555555; + background-color: #ffffff; + border: 1px solid #ddd; + border-bottom-color: transparent; + cursor: default; +} +.tablist > li > a { + padding-top: 8px; + padding-bottom: 8px; + margin-top: 2px; + margin-bottom: 2px; + -webkit-border-radius: 5px; + -moz-border-radius: 5px; + border-radius: 5px; +} +.tablist > .current > a, +.tablist > .current > a:hover { + color: #ffffff; + background-color: #0088cc; +} +.tablist-stacked > li { + float: none; +} +.tablist-stacked > li > a { + margin-right: 0; +} +.tablist-tabs.tablist-stacked { + border-bottom: 0; +} +.tablist-tabs.tablist-stacked > li > a { + border: 1px solid #ddd; + -webkit-border-radius: 0; + -moz-border-radius: 0; + border-radius: 0; +} +.tablist-tabs.tablist-stacked > li:first-child > a { + -webkit-border-top-right-radius: 4px; + -moz-border-radius-topright: 4px; + border-top-right-radius: 4px; + -webkit-border-top-left-radius: 4px; + -moz-border-radius-topleft: 4px; + border-top-left-radius: 4px; +} +.tablist-tabs.tablist-stacked > li:last-child > a { + -webkit-border-bottom-right-radius: 4px; + -moz-border-radius-bottomright: 4px; + border-bottom-right-radius: 4px; + -webkit-border-bottom-left-radius: 4px; + -moz-border-radius-bottomleft: 4px; + border-bottom-left-radius: 4px; +} +.tablist-tabs.tablist-stacked > li > a:hover { + border-color: #ddd; + z-index: 2; +} +.tablist.tablist-stacked > li > a { + margin-bottom: 3px; +} +.tablist.tablist-stacked > li:last-child > a { + margin-bottom: 1px; +} +.tablist-tabs .dropdown-menu { + -webkit-border-radius: 0 0 6px 6px; + -moz-border-radius: 0 0 6px 6px; + border-radius: 0 0 6px 6px; +} +.tablist .dropdown-menu { + -webkit-border-radius: 6px; + -moz-border-radius: 6px; + border-radius: 6px; +} +.tablist .dropdown-toggle .caret { + border-top-color: #0088cc; + border-bottom-color: #0088cc; + margin-top: 6px; +} +.tablist .dropdown-toggle:hover .caret { + border-top-color: #005580; + border-bottom-color: #005580; +} +/* move down carets for tabs */ +.tablist-tabs .dropdown-toggle .caret { + margin-top: 8px; +} +.tablist .current .dropdown-toggle .caret { + border-top-color: #fff; + border-bottom-color: #fff; +} +.tablist-tabs .current .dropdown-toggle .caret { + border-top-color: #555555; + border-bottom-color: #555555; +} +.tablist > .dropdown.current > a:hover { + cursor: pointer; +} +.tablist-tabs .open .dropdown-toggle, +.tablist .open .dropdown-toggle, +.tablist > li.dropdown.open.current > a:hover { + color: #ffffff; + background-color: #999999; + border-color: #999999; +} +.tablist li.dropdown.open .caret, +.tablist li.dropdown.open.current .caret, +.tablist li.dropdown.open a:hover .caret { + border-top-color: #ffffff; + border-bottom-color: #ffffff; + opacity: 1; + filter: alpha(opacity=100); +} +.tabs-stacked .open > a:hover { + border-color: #999999; +} +.tab-content > .tab-pane, +.pill-content > .pill-pane { + display: none; +} +.tab-content > .current, +.pill-content > .current { + display: block; +} +.tabs-below > .tablist-tabs { + border-top: 1px solid #ddd; +} +.tabs-below > .tablist-tabs > li { + margin-top: -1px; + margin-bottom: 0; +} +.tabs-below > .tablist-tabs > li > a { + -webkit-border-radius: 0 0 4px 4px; + -moz-border-radius: 0 0 4px 4px; + border-radius: 0 0 4px 4px; +} +.tabs-below > .tablist-tabs > li > a:hover { + border-bottom-color: transparent; + border-top-color: #ddd; +} +.tabs-below > .tablist-tabs > .current > a, +.tabs-below > .tablist-tabs > .current > a:hover { + border-color: transparent #ddd #ddd #ddd; +} +.tabs-left > .tablist-tabs > li, +.tabs-right > .tablist-tabs > li { + float: none; +} +.tabs-left > .tablist-tabs > li > a, +.tabs-right > .tablist-tabs > li > a { + min-width: 74px; + margin-right: 0; + margin-bottom: 3px; +} +.tabs-left > .tablist-tabs { + float: left; + margin-right: 19px; + border-right: 1px solid #ddd; +} +.tabs-left > .tablist-tabs > li > a { + margin-right: -1px; + -webkit-border-radius: 4px 0 0 4px; + -moz-border-radius: 4px 0 0 4px; + border-radius: 4px 0 0 4px; +} +.tabs-left > .tablist-tabs > li > a:hover { + border-color: #eeeeee #dddddd #eeeeee #eeeeee; +} +.tabs-left > .tablist-tabs .current > a, +.tabs-left > .tablist-tabs .current > a:hover { + border-color: #ddd transparent #ddd #ddd; + *border-right-color: #ffffff; +} +.tabs-right > .tablist-tabs { + float: right; + margin-left: 19px; + border-left: 1px solid #ddd; +} +.tabs-right > .tablist-tabs > li > a { + margin-left: -1px; + -webkit-border-radius: 0 4px 4px 0; + -moz-border-radius: 0 4px 4px 0; + border-radius: 0 4px 4px 0; +} +.tabs-right > .tablist-tabs > li > a:hover { + border-color: #eeeeee #eeeeee #eeeeee #dddddd; +} +.tabs-right > .tablist-tabs .current > a, +.tabs-right > .tablist-tabs .current > a:hover { + border-color: #ddd #ddd #ddd transparent; + *border-left-color: #ffffff; +} +.tablist > .disabled > a { + color: #999999; +} +.tablist > .disabled > a:hover { + text-decoration: none; + background-color: transparent; + cursor: default; +} +.tablistbar { + overflow: visible; + margin-bottom: 20px; + color: #ffffff; + *position: relative; + *z-index: 2; +} +.tablistbar-inner { + min-height: 40px; + padding-left: 20px; + padding-right: 20px; + background-color: #034c03; + background-image: -moz-linear-gradient(top, #024002, #045f04); + background-image: -webkit-gradient(linear, 0 0, 0 100%, from(#024002), to(#045f04)); + background-image: -webkit-linear-gradient(top, #024002, #045f04); + background-image: -o-linear-gradient(top, #024002, #045f04); + background-image: linear-gradient(to bottom, #024002, #045f04); + background-repeat: repeat-x; + filter: progid:DXImageTransform.Microsoft.gradient(startColorstr='#ff024002', endColorstr='#ff045f04', GradientType=0); + border: 1px solid #022402; + -webkit-border-radius: 4px; + -moz-border-radius: 4px; + border-radius: 4px; + -webkit-box-shadow: 0 1px 4px rgba(0, 0, 0, 0.065); + -moz-box-shadow: 0 1px 4px rgba(0, 0, 0, 0.065); + box-shadow: 0 1px 4px rgba(0, 0, 0, 0.065); + *zoom: 1; +} +.tablistbar-inner:before, +.tablistbar-inner:after { + display: table; + content: ""; + line-height: 0; +} +.tablistbar-inner:after { + clear: both; +} +.tablistbar .container { + width: auto; +} +.tablist-collapse.collapse { + height: auto; +} +.tablistbar .brand { + float: left; + display: block; + padding: 10px 20px 10px; + margin-left: -20px; + font-size: 20px; + font-weight: 200; + color: #ffffff; + text-shadow: 0 1px 0 #024002; +} +.tablistbar .brand:hover { + text-decoration: none; +} +.tablistbar-text { + margin-bottom: 0; + line-height: 40px; +} +.tablistbar-link { + color: #ffffff; +} +.tablistbar-link:hover { + color: #333333; +} +.tablistbar .tablist { + position: relative; + left: 0; + display: block; + float: left; + margin: 0 10px 0 0; +} +.tablistbar .tablist.pull-right { + float: right; + margin-right: 0; +} +.tablistbar .tablist > li { + float: left; +} +.tablistbar .tablist > li > a { + float: none; + padding: 10px 15px 10px; + color: #ffffff; + text-decoration: none; + text-shadow: 0 1px 0 #024002; +} +.tablistbar .tablist .dropdown-toggle .caret { + margin-top: 8px; +} +.tablistbar .tablist > li > a:focus, +.tablistbar .tablist > li > a:hover { + background-color: transparent; + color: white; + text-decoration: none; +} +.tablistbar .tablist > .current > a, +.tablistbar .tablist > .current > a:hover, +.tablistbar .tablist > .current > a:focus { + color: #555555; + text-decoration: none; + background-color: #034703; + -webkit-box-shadow: inset 0 3px 8px rgba(0, 0, 0, 0.125); + -moz-box-shadow: inset 0 3px 8px rgba(0, 0, 0, 0.125); + box-shadow: inset 0 3px 8px rgba(0, 0, 0, 0.125); +} +.tablistbar .btn-navbar { + display: none; + float: right; + padding: 7px 10px; + margin-left: 5px; + margin-right: 5px; + color: #ffffff; + text-shadow: 0 -1px 0 rgba(0, 0, 0, 0.25); + background-color: #023402; + background-image: -moz-linear-gradient(top, #012701, #034703); + background-image: -webkit-gradient(linear, 0 0, 0 100%, from(#012701), to(#034703)); + background-image: -webkit-linear-gradient(top, #012701, #034703); + background-image: -o-linear-gradient(top, #012701, #034703); + background-image: linear-gradient(to bottom, #012701, #034703); + background-repeat: repeat-x; + filter: progid:DXImageTransform.Microsoft.gradient(startColorstr='#ff012701', endColorstr='#ff034703', GradientType=0); + border-color: #034703 #034703 #000000; + border-color: rgba(0, 0, 0, 0.1) rgba(0, 0, 0, 0.1) rgba(0, 0, 0, 0.25); + *background-color: #034703; + /* Darken IE7 buttons by default so they stand out more given they won't have borders */ + + filter: progid:DXImageTransform.Microsoft.gradient(enabled = false); + -webkit-box-shadow: inset 0 1px 0 rgba(255, 255, 255, 0.1), 0 1px 0 rgba(255, 255, 255, 0.075); + -moz-box-shadow: inset 0 1px 0 rgba(255, 255, 255, 0.1), 0 1px 0 rgba(255, 255, 255, 0.075); + box-shadow: inset 0 1px 0 rgba(255, 255, 255, 0.1), 0 1px 0 rgba(255, 255, 255, 0.075); +} +.tablistbar .tablist > li > .dropdown-menu:before { + content: ''; + display: inline-block; + border-left: 7px solid transparent; + border-right: 7px solid transparent; + border-bottom: 7px solid #ccc; + border-bottom-color: rgba(0, 0, 0, 0.2); + position: absolute; + top: -7px; + left: 9px; +} +.tablistbar .tablist > li > .dropdown-menu:after { + content: ''; + display: inline-block; + border-left: 6px solid transparent; + border-right: 6px solid transparent; + border-bottom: 6px solid #ffffff; + position: absolute; + top: -6px; + left: 10px; +} +.tablistbar .tablist li.dropdown.open > .dropdown-toggle, +.tablistbar .tablist li.dropdown.current > .dropdown-toggle, +.tablistbar .tablist li.dropdown.open.current > .dropdown-toggle { + background-color: #034703; + color: #555555; +} +.tablistbar .tablist li.dropdown > .dropdown-toggle .caret { + border-top-color: #ffffff; + border-bottom-color: #ffffff; +} +.tablistbar .tablist li.dropdown.open > .dropdown-toggle .caret, +.tablistbar .tablist li.dropdown.current > .dropdown-toggle .caret, +.tablistbar .tablist li.dropdown.open.current > .dropdown-toggle .caret { + border-top-color: #555555; + border-bottom-color: #555555; +} +.tablistbar .pull-right > li > .dropdown-menu, +.tablistbar .tablist > li > .dropdown-menu.pull-right { + left: auto; + right: 0; +} +.tablistbar .pull-right > li > .dropdown-menu:before, +.tablistbar .tablist > li > .dropdown-menu.pull-right:before { + left: auto; + right: 12px; +} +.tablistbar .pull-right > li > .dropdown-menu:after, +.tablistbar .tablist > li > .dropdown-menu.pull-right:after { + left: auto; + right: 13px; +} +.tablistbar .pull-right > li > .dropdown-menu .dropdown-menu, +.tablistbar .tablist > li > .dropdown-menu.pull-right .dropdown-menu { + left: auto; + right: 100%; + margin-left: 0; + margin-right: -1px; + -webkit-border-radius: 6px 0 6px 6px; + -moz-border-radius: 6px 0 6px 6px; + border-radius: 6px 0 6px 6px; +} +.breadcrumb { + padding: 8px 15px; + margin: 0 0 20px; + list-style: none; + background-color: #f5f5f5; + -webkit-border-radius: 4px; + -moz-border-radius: 4px; + border-radius: 4px; +} +.breadcrumb li { + display: inline-block; + *display: inline; + /* IE7 inline-block hack */ + + *zoom: 1; + text-shadow: 0 1px 0 #ffffff; +} +.breadcrumb .divider { + padding: 0 5px; + color: #ccc; +} +.breadcrumb .current { + color: #999999; +} +.pagination-right { + text-align: right; +} +.fade { + opacity: 0; + -webkit-transition: opacity 0.15s linear; + -moz-transition: opacity 0.15s linear; + -o-transition: opacity 0.15s linear; + transition: opacity 0.15s linear; +} +.fade.in { + opacity: 1; +} +.collapse { + position: relative; + height: 0; + overflow: hidden; + -webkit-transition: height 0.35s ease; + -moz-transition: height 0.35s ease; + -o-transition: height 0.35s ease; + transition: height 0.35s ease; +} +.collapse.in { + height: auto; +} +.hidden { + display: none; + visibility: hidden; +} +.visible-phone { + display: none !important; +} +.visible-tablet { + display: none !important; +} +.hidden-desktop { + display: none !important; +} +.visible-desktop { + display: inherit !important; +} +@media (min-width: 768px) and (max-width: 979px) { + .hidden-desktop { + display: inherit !important; + } + .visible-desktop { + display: none !important ; + } + .visible-tablet { + display: inherit !important; + } + .hidden-tablet { + display: none !important; + } +} +@media (max-width: 767px) { + .hidden-desktop { + display: inherit !important; + } + .visible-desktop { + display: none !important; + } + .visible-phone { + display: inherit !important; + } + .hidden-phone { + display: none !important; + } +} +@media (max-width: 767px) { + body { + padding-left: 20px; + padding-right: 20px; + } + .container { + width: auto; + } + .row, + .thumbnails { + margin-left: 0; + } +} +@media (max-width: 480px) { + .tablist-collapse { + -webkit-transform: translate3d(0, 0, 0); + } + .page-header h1 small { + display: block; + line-height: 20px; + } +} +@media (min-width: 768px) and (max-width: 979px) { + .row { + margin-left: -20px; + *zoom: 1; + } + .row:before, + .row:after { + display: table; + content: ""; + line-height: 0; + } + .row:after { + clear: both; + } + [class*="span"] { + float: left; + min-height: 1px; + margin-left: 20px; + } + .container { + width: 724px; + } +} +@media (min-width: 1200px) { + .row { + margin-left: -30px; + *zoom: 1; + } + .row:before, + .row:after { + display: table; + content: ""; + line-height: 0; + } + .row:after { + clear: both; + } + [class*="span"] { + float: left; + min-height: 1px; + margin-left: 30px; + } + .container { + width: 1070px; + } +} +@media (max-width: 979px) { + body { + padding-top: 0; + } +} +@media (min-width: 980px) { + .tablist-collapse.collapse { + height: auto !important; + overflow: visible !important; + } +} +.tablistbar .brand { + padding: 5px; + margin-left: 0; +} +.tablistbar .brand img { + width: 30px; + vertical-align: middle; +} + +h1 small { + font-size: 18px; +} + +h1 small, +h2 small, +h3 small, +h4 small, +h5 small, +h6 small, +.page-header small { + line-height: 0.8; + font-weight: normal; + color: #999999; + display:block; + vertical-align: middle; +} + +.page-header h1, h1:first-child { + font-size: 40px; + padding-bottom: 5px; +} + +.page-header h1 { + border-bottom: 1px solid #999999; + padding-bottom: 9px; +} + +.page-header img { + height: 80px; + padding-bottom: 5px; +} + +.page-header small { + line-height: 1.1; + font-size: 18px; +} + +h2, +h3, +h4, +div.ah, +.title { + border-color: #D6E9C6; + color: #468847; + border-style: solid; + border-width: 0 0 1px; + padding-left: 0.5em; +} + + +.google { + color: white; +} + +.breadcrumb { + font-size: 11px; + padding-top: 2px; + padding-bottom: 2px; +} + +h1 a, +h2 a, +h3 a, +h4 a { + color: inherit; +} + +.tablistbar-inner a { + font-weight: bold; +} + +.list-2panes:before, +.list-2panes:after { + display: table; + content: ""; + line-height: 0; +} + +.list-2panes:after { + clear:both; +} + +.list-2panes li { + width: 470px; + width: 470px; + float: left; + margin-left: 30px; + min-height: 1px; +} +/* The standard CSS for doxygen */ + +/* @group Heading Levels */ + + +dt { + font-weight: bold; +} + +div.multicol { + -moz-column-gap: 1em; + -webkit-column-gap: 1em; + -moz-column-count: 3; + -webkit-column-count: 3; +} + +p.startli, p.startdd, p.starttd { + margin-top: 2px; +} + +p.endli { + margin-bottom: 0px; +} + +p.enddd { + margin-bottom: 4px; +} + +p.endtd { + margin-bottom: 2px; +} + +/* @end */ + +caption { + font-weight: bold; +} + +span.legend { + font-size: 70%; + text-align: center; +} + +h3.version { + font-size: 90%; + text-align: center; +} + +div.qindex, div.tablisttab{ + background-color: #EBF6EB; + border: 1px solid #A3D7A3; + text-align: center; +} + +div.qindex, div.tablistpath { + width: 100%; + line-height: 140%; +} + +div.tablisttab { + margin-right: 15px; +} + +/* @group Link Styling */ + +a { + color: #3D8C3D; + font-weight: normal; + text-decoration: none; +} + +.contents a:visited { + color: #46A246; +} + +a:hover { + text-decoration: underline; +} + +a.qindex { + font-weight: bold; +} + +a.qindexHL { + font-weight: bold; + background-color: #9CD49C; + color: #ffffff; + border: 1px double #86CA86; +} + +.contents a.qindexHL:visited { + color: #ffffff; +} + +a.el { + font-weight: bold; +} + +a.elRef { +} + +a.code { + color: #4665A2; +} + +a.codeRef { + color: #4665A2; +} + +/* @end */ + +dl.el { + margin-left: -1cm; +} + +.fragment { + font-family: monospace, fixed; + font-size: 105%; +} + +pre.fragment { + border: 1px solid #C4E5C4; + background-color: #FBFDFB; + padding: 4px 6px; + margin: 4px 8px 4px 2px; + overflow: auto; + word-wrap: break-word; + font-size: 9pt; + line-height: 125%; +} + +div.groupHeader { + margin-left: 16px; + margin-top: 12px; + font-weight: bold; +} + +div.groupText { + margin-left: 16px; + font-style: italic; +} + +div.contents { + margin-top: 10px; + margin-left: 8px; + margin-right: 8px; +} + +td.indexkey { + white-space: nowrap; + vertical-align: top; +} + + +tr.memlist { + background-color: #EEF7EE; +} + +p.formulaDsp { + text-align: center; +} + +img.formulaDsp { + +} + +img.formulaInl { + vertical-align: middle; +} + +div.center { + text-align: center; + margin-top: 0px; + margin-bottom: 0px; + padding: 0px; +} + +div.center img { + border: 0px; +} + +#footer { + margin: -10px 1em 0; + padding-top: 20px; + text-align: center; + font-size: small; +} + +address.footer { + background-color: #ffffff; + text-align: center; +} + +img.footer { + border: 0px; + vertical-align: middle; +} + +/* @group Code Colorization */ + +span.keyword { + color: #008000 +} + +span.keywordtype { + color: #604020 +} + +span.keywordflow { + color: #e08000 +} + +span.comment { + color: #800000 +} + +span.preprocessor { + color: #806020 +} + +span.stringliteral { + color: #002080 +} + +span.charliteral { + color: #008080 +} + +span.vhdldigit { + color: #ff00ff +} + +span.vhdlchar { + color: #000000 +} + +span.vhdlkeyword { + color: #700070 +} + +span.vhdllogic { + color: #ff0000 +} + +/* @end */ + +/* +.search { + color: #003399; + font-weight: bold; +} + +form.search { + margin-bottom: 0px; + margin-top: 0px; +} + +input.search { + font-size: 75%; + color: #000080; + font-weight: normal; + background-color: #e8eef2; +} +*/ + +td.tiny { + font-size: 75%; +} + +.dirtab { + padding: 4px; + border-collapse: collapse; + border: 1px solid #A3D7A3; +} + +th.dirtab { + background: #EBF6EB; + font-weight: bold; +} + +hr { + height: 0px; + border: none; + border-top: 1px solid #4AAA4A; +} + +hr.footer { + height: 1px; +} + +/* @group Member Descriptions */ + +table.memberdecls { + border-spacing: 0px; + padding: 0px; +} + +.mdescLeft, .mdescRight, +.memItemLeft, .memItemRight, +.memTemplItemLeft, .memTemplItemRight, .memTemplParams { + background-color: #F9FCF9; + border: none; + margin: 4px; + padding: 1px 0 0 8px; +} + +.mdescLeft, .mdescRight { + padding: 0px 8px 4px 8px; + color: #555; +} + +.memItemLeft, .memItemRight, .memTemplParams { + border-top: 1px solid #C4E5C4; +} + +.memItemLeft, .memTemplItemLeft { + white-space: nowrap; +} + +.memItemRight { + width: 100%; +} + +.memTemplParams { + color: #46A246; + white-space: nowrap; +} + +/* @end */ + +/* @group Member Details */ + +/* Styles for detailed member documentation */ + +.memtemplate { + font-size: 80%; + color: #46A246; + font-weight: normal; + margin-left: 9px; +} + +.memnav { + background-color: #EBF6EB; + border: 1px solid #A3D7A3; + text-align: center; + margin: 2px; + margin-right: 15px; + padding: 2px; +} + +.mempage { + width: 100%; +} + +.memitem { + padding: 0; + margin-bottom: 10px; + margin-right: 5px; +} + +.memname { + white-space: nowrap; + font-weight: bold; + margin-left: 6px; +} + +.memproto, dl.reflist dt { + border-top: 1px solid #A8D9A8; + border-left: 1px solid #A8D9A8; + border-right: 1px solid #A8D9A8; + padding: 6px 0px 6px 0px; + color: #255525; + font-weight: bold; + text-shadow: 0px 1px 1px rgba(255, 255, 255, 0.9); + /* opera specific markup */ + box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + border-top-right-radius: 8px; + border-top-left-radius: 8px; + /* firefox specific markup */ + -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; + -moz-border-radius-topright: 8px; + -moz-border-radius-topleft: 8px; + /* webkit specific markup */ + -webkit-box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + -webkit-border-top-right-radius: 8px; + -webkit-border-top-left-radius: 8px; + background-repeat:repeat-x; + background-color: #E2F2E2; + +} + +.memdoc, dl.reflist dd { + border-bottom: 1px solid #A8D9A8; + border-left: 1px solid #A8D9A8; + border-right: 1px solid #A8D9A8; + padding: 2px 5px; + background-color: #FBFDFB; + border-top-width: 0; + /* opera specific markup */ + border-bottom-left-radius: 8px; + border-bottom-right-radius: 8px; + box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + /* firefox specific markup */ + -moz-border-radius-bottomleft: 8px; + -moz-border-radius-bottomright: 8px; + -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; + background-image: -moz-linear-gradient(center top, #FFFFFF 0%, #FFFFFF 60%, #F7FBF7 95%, #EEF7EE); + /* webkit specific markup */ + -webkit-border-bottom-left-radius: 8px; + -webkit-border-bottom-right-radius: 8px; + -webkit-box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); + background-image: -webkit-gradient(linear,center top,center bottom,from(#FFFFFF), color-stop(0.6,#FFFFFF), color-stop(0.60,#FFFFFF), color-stop(0.95,#F7FBF7), to(#EEF7EE)); +} + +dl.reflist dt { + padding: 5px; +} + +dl.reflist dd { + margin: 0px 0px 10px 0px; + padding: 5px; +} + +.paramkey { + text-align: right; +} + +.paramtype { + white-space: nowrap; +} + +.paramname { + color: #602020; + white-space: nowrap; +} +.paramname em { + font-style: normal; +} + +.params, .retval, .exception, .tparams { + border-spacing: 6px 2px; +} + +.params .paramname, .retval .paramname { + font-weight: bold; + vertical-align: top; +} + +.params .paramtype { + font-style: italic; + vertical-align: top; +} + +.params .paramdir { + font-family: "courier new",courier,monospace; + vertical-align: top; +} + + + + +/* @end */ + +/* @group Directory (tree) */ + +/* for the tree view */ + +.ftvtree { + font-family: sans-serif; + margin: 0px; +} + +/* these are for tree view when used as main index */ + +.directory { + font-size: 9pt; + font-weight: bold; + margin: 5px; +} + +.directory h3 { + margin: 0px; + margin-top: 1em; + font-size: 11pt; +} + +/* +The following two styles can be used to replace the root node title +with an image of your choice. Simply uncomment the next two styles, +specify the name of your image and be sure to set 'height' to the +proper pixel height of your image. +*/ + +/* +.directory h3.swap { + height: 61px; + background-repeat: no-repeat; + background-image: url("yourimage.gif"); +} +.directory h3.swap span { + display: none; +} +*/ + +.directory > h3 { + margin-top: 0; +} + +.directory p { + margin: 0px; + white-space: nowrap; +} + +.directory div { + display: none; + margin: 0px; +} + +.directory img { + vertical-align: -30%; +} + +/* these are for tree view when not used as main index */ + +.directory-alt { + font-size: 100%; + font-weight: bold; +} + +.directory-alt h3 { + margin: 0px; + margin-top: 1em; + font-size: 11pt; +} + +.directory-alt > h3 { + margin-top: 0; +} + +.directory-alt p { + margin: 0px; + white-space: nowrap; +} + +.directory-alt div { + display: none; + margin: 0px; +} + +.directory-alt img { + vertical-align: -30%; +} + +/* @end */ + +div.dynheader { + margin-top: 8px; +} + +address { + font-style: normal; + color: #2A612A; +} + +table.doxtable { + border-collapse:collapse; +} + +table.doxtable td, table.doxtable th { + border: 1px solid #2D682D; + padding: 3px 7px 2px; +} + +table.doxtable th { + background-color: #377F37; + color: #FFFFFF; + font-size: 110%; + padding-bottom: 4px; + padding-top: 5px; + text-align:left; +} + +table.fieldtable { + width: 100%; + margin-bottom: 10px; + border: 1px solid #A8D9A8; + border-spacing: 0px; + -moz-border-radius: 4px; + -webkit-border-radius: 4px; + border-radius: 4px; + -moz-box-shadow: rgba(0, 0, 0, 0.15) 2px 2px 2px; + -webkit-box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); + box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); +} + +.fieldtable td, .fieldtable th { + padding: 3px 7px 2px; +} + +.fieldtable td.fieldtype, .fieldtable td.fieldname { + white-space: nowrap; + border-right: 1px solid #A8D9A8; + border-bottom: 1px solid #A8D9A8; + vertical-align: top; +} + +.fieldtable td.fielddoc { + border-bottom: 1px solid #A8D9A8; + width: 100%; +} + +.fieldtable tr:last-child td { + border-bottom: none; +} + +.fieldtable th { + background-repeat:repeat-x; + background-color: #E2F2E2; + font-size: 90%; + color: #255525; + padding-bottom: 4px; + padding-top: 5px; + text-align:left; + -moz-border-radius-topleft: 4px; + -moz-border-radius-topright: 4px; + -webkit-border-top-left-radius: 4px; + -webkit-border-top-right-radius: 4px; + border-top-left-radius: 4px; + border-top-right-radius: 4px; + border-bottom: 1px solid #A8D9A8; +} + + +.tabsearch { + top: 0px; + left: 10px; + height: 36px; + z-index: 101; + overflow: hidden; + font-size: 13px; +} + +.tablistpath ul +{ + font-size: 11px; + background-repeat:repeat-x; + height:30px; + line-height:30px; + color:#8ACC8A; + border:solid 1px #C2E4C2; + overflow:hidden; + margin:0px; + padding:0px; +} + +.tablistpath li +{ + list-style-type:none; + float:left; + padding-left:10px; + padding-right:15px; + background-repeat:no-repeat; + background-position:right; + color:#367C36; +} + +.tablistpath li.tablistelem a +{ + height:32px; + display:block; + text-decoration: none; + outline: none; +} + +.tablistpath li.tablistelem a:hover +{ + color:#68BD68; +} + +.tablistpath li.footer +{ + list-style-type:none; + float:right; + padding-left:10px; + padding-right:15px; + background-image:none; + background-repeat:no-repeat; + background-position:right; + color:#367C36; + font-size: 8pt; +} + + +div.summary +{ + margin-top: 12px; + text-align: center; +} + +div.summary a +{ + white-space: nowrap; +} + +div.ingroups +{ + margin-left: 5px; + font-size: 8pt; + padding-left: 5px; + width: 50%; + text-align: left; +} + +div.ingroups a +{ + white-space: nowrap; +} + +div.headertitle +{ + padding: 5px 5px 5px 7px; +} + +dl +{ + padding: 0 0 0 10px; +} + +dl.note, dl.warning, dl.attention, dl.pre, dl.post, dl.invariant, dl.deprecated, dl.todo, dl.test, dl.bug +{ + border-left:4px solid; + padding: 0 0 0 6px; +} + +dl.note +{ + border-color: #D0C000; +} + +dl.warning, dl.attention +{ + border-color: #FF0000; +} + +dl.pre, dl.post, dl.invariant +{ + border-color: #00D000; +} + +dl.deprecated +{ + border-color: #505050; +} + +dl.todo +{ + border-color: #00C0E0; +} + +dl.test +{ + border-color: #3030E0; +} + +dl.bug +{ + border-color: #C08050; +} + +#projectlogo +{ + text-align: center; + vertical-align: bottom; + border-collapse: separate; +} + +#projectlogo img +{ + border: 0px none; +} + +#projectname +{ + font: 300% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 2px 0px; +} + +#projectbrief +{ + font: 120% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 0px; +} + +#projectnumber +{ + font: 50% Tahoma, Arial,sans-serif; + margin: 0px; + padding: 0px; +} + +#titlearea +{ + padding: 0px; + margin: 0px; + width: 100%; + border-bottom: 1px solid #53B453; +} + +.image +{ + text-align: center; +} + +.dotgraph +{ + text-align: center; +} + +.mscgraph +{ + text-align: center; +} + +.caption +{ + font-weight: bold; +} + +div.zoom +{ + border: 1px solid #90CE90; +} + +dl.citelist { + margin-bottom:50px; +} + +dl.citelist dt { + color:#337533; + float:left; + font-weight:bold; + margin-right:10px; + padding:5px; +} + +dl.citelist dd { + margin:2px 0; + padding:5px 0; +} + +@media print +{ + #top { display: none; } + #side-nav { display: none; } + #nav-path { display: none; } + body { overflow:visible; } + h1, h2, h3, h4, h5, h6 { page-break-after: avoid; } + .summary { display: none; } + .memitem { page-break-inside: avoid; } + #doc-content + { + margin-left:0 !important; + height:auto !important; + width:auto !important; + overflow:inherit; + display:inline; + } + pre.fragment + { + overflow: visible; + text-wrap: unrestricted; + white-space: -moz-pre-wrap; /* Moz */ + white-space: -pre-wrap; /* Opera 4-6 */ + white-space: -o-pre-wrap; /* Opera 7 */ + white-space: pre-wrap; /* CSS3 */ + word-wrap: break-word; /* IE 5.5+ */ + } +} + +#proj_desc { + font-size: 1.2em; +} diff --git a/doc/doxy/footer.html b/doc/doxy/footer.html new file mode 100644 index 0000000000..101e6fe70b --- /dev/null +++ b/doc/doxy/footer.html @@ -0,0 +1,9 @@ + +
+ + Generated on $datetime for $projectname by doxygen $doxygenversion + +
+ + + diff --git a/doc/doxy/header.html b/doc/doxy/header.html new file mode 100644 index 0000000000..312990cdbc --- /dev/null +++ b/doc/doxy/header.html @@ -0,0 +1,16 @@ + + + + + +$projectname: $title +$title + + + + + +
+ + +
diff --git a/doc/encoders.texi b/doc/encoders.texi index d48df7d302..94d834077a 100644 --- a/doc/encoders.texi +++ b/doc/encoders.texi @@ -1,10 +1,10 @@ @chapter Encoders @c man begin ENCODERS -Encoders are configured elements in FFmpeg which allow the encoding of +Encoders are configured elements in Libav which allow the encoding of multimedia streams. -When you configure your FFmpeg build, all the supported native encoders +When you configure your Libav build, all the supported native encoders are enabled by default. Encoders requiring an external library must be enabled manually via the corresponding @code{--enable-lib} option. You can list all available encoders using the configure option @code{--list-encoders}. @@ -14,7 +14,7 @@ You can disable all the encoders with the configure option with the options @code{--enable-encoder=@var{ENCODER}} / @code{--disable-encoder=@var{ENCODER}}. -The option @code{-encoders} of the ff* tools will display the list of +The option @code{-encoders} of the av* tools will display the list of enabled encoders. @c man end ENCODERS @@ -25,88 +25,6 @@ enabled encoders. A description of some of the currently available audio encoders follows. -@anchor{aacenc} -@section aac - -Advanced Audio Coding (AAC) encoder. - -This encoder is an experimental FFmpeg-native AAC encoder. Currently only the -low complexity (AAC-LC) profile is supported. To use this encoder, you must set -@option{strict} option to @samp{experimental} or lower. - -As this encoder is experimental, unexpected behavior may exist from time to -time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned -that it has a worse quality reported by some users. - -@c todo @ref{libaacplus} -See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}. - -@subsection Options - -@table @option -@item b -Set bit rate in bits/s. Setting this automatically activates constant bit rate -(CBR) mode. - -@item q -Set quality for variable bit rate (VBR) mode. This option is valid only using -the @command{ffmpeg} command-line tool. For library interface users, use -@option{global_quality}. - -@item stereo_mode -Set stereo encoding mode. Possible values: - -@table @samp -@item auto -Automatically selected by the encoder. - -@item ms_off -Disable middle/side encoding. This is the default. - -@item ms_force -Force middle/side encoding. -@end table - -@item aac_coder -Set AAC encoder coding method. Possible values: - -@table @samp -@item faac -FAAC-inspired method. - -This method is a simplified reimplementation of the method used in FAAC, which -sets thresholds proportional to the band energies, and then decreases all the -thresholds with quantizer steps to find the appropriate quantization with -distortion below threshold band by band. - -The quality of this method is comparable to the two loop searching method -descibed below, but somewhat a little better and slower. - -@item anmr -Average noise to mask ratio (ANMR) trellis-based solution. - -This has a theoretic best quality out of all the coding methods, but at the -cost of the slowest speed. - -@item twoloop -Two loop searching (TLS) method. - -This method first sets quantizers depending on band thresholds and then tries -to find an optimal combination by adding or subtracting a specific value from -all quantizers and adjusting some individual quantizer a little. - -This method produces similar quality with the FAAC method and is the default. - -@item fast -Constant quantizer method. - -This method sets a constant quantizer for all bands. This is the fastest of all -the methods, yet produces the worst quality. - -@end table - -@end table - @section ac3 and ac3_fixed AC-3 audio encoders. @@ -451,7 +369,7 @@ is highly recommended that it be left as enabled except for testing purposes. @end table -@subsection Floating-Point-Only AC-3 Encoding Options +@subheading Floating-Point-Only AC-3 Encoding Options These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being @@ -494,764 +412,32 @@ Selected by Encoder (default) @end table -@anchor{libfaac} -@section libfaac - -libfaac AAC (Advanced Audio Coding) encoder wrapper. - -Requires the presence of the libfaac headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libfaac --enable-nonfree}. - -This encoder is considered to be of higher quality with respect to the -@ref{aacenc,,the native experimental FFmpeg AAC encoder}. - -For more information see the libfaac project at -@url{http://www.audiocoding.com/faac.html/}. - -@subsection Options - -The following shared FFmpeg codec options are recognized. - -The following options are supported by the libfaac wrapper. The -@command{faac}-equivalent of the options are listed in parentheses. - -@table @option -@item b (@emph{-b}) -Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate -is not explicitly specified, it is automatically set to a suitable -value depending on the selected profile. @command{faac} bitrate is -expressed in kilobits/s. - -Note that libfaac does not support CBR (Constant Bit Rate) but only -ABR (Average Bit Rate). - -If VBR mode is enabled this option is ignored. - -@item ar (@emph{-R}) -Set audio sampling rate (in Hz). - -@item ac (@emph{-c}) -Set the number of audio channels. - -@item cutoff (@emph{-C}) -Set cutoff frequency. If not specified (or explicitly set to 0) it -will use a value automatically computed by the library. Default value -is 0. - -@item profile -Set audio profile. - -The following profiles are recognized: -@table @samp -@item aac_main -Main AAC (Main) - -@item aac_low -Low Complexity AAC (LC) - -@item aac_ssr -Scalable Sample Rate (SSR) - -@item aac_ltp -Long Term Prediction (LTP) -@end table - -If not specified it is set to @samp{aac_low}. - -@item flags +qscale -Set constant quality VBR (Variable Bit Rate) mode. - -@item global_quality -Set quality in VBR mode as an integer number of lambda units. - -Only relevant when VBR mode is enabled with @code{flags +qscale}. The -value is converted to QP units by dividing it by @code{FF_QP2LAMBDA}, -and used to set the quality value used by libfaac. A reasonable range -for the option value in QP units is [10-500], the higher the value the -higher the quality. - -@item q (@emph{-q}) -Enable VBR mode when set to a non-negative value, and set constant -quality value as a double floating point value in QP units. - -The value sets the quality value used by libfaac. A reasonable range -for the option value is [10-500], the higher the value the higher the -quality. - -This option is valid only using the @command{ffmpeg} command-line -tool. For library interface users, use @option{global_quality}. -@end table - -@subsection Examples - -@itemize -@item -Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4) -container: -@example -ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a -@end example - -@item -Use @command{ffmpeg} to convert an audio file to VBR AAC, using the -LTP AAC profile: -@example -ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a -@end example -@end itemize - -@anchor{libfdk-aac-enc} -@section libfdk_aac - -libfdk-aac AAC (Advanced Audio Coding) encoder wrapper. - -The libfdk-aac library is based on the Fraunhofer FDK AAC code from -the Android project. - -Requires the presence of the libfdk-aac headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libfdk-aac}. The library is also incompatible with GPL, -so if you allow the use of GPL, you should configure with -@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}. - -This encoder is considered to be of higher quality with respect to -both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and -@ref{libfaac}. - -VBR encoding, enabled through the @option{vbr} or @option{flags -+qscale} options, is experimental and only works with some -combinations of parameters. - -Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or -higher. - -For more information see the fdk-aac project at -@url{http://sourceforge.net/p/opencore-amr/fdk-aac/}. - -@subsection Options - -The following options are mapped on the shared FFmpeg codec options. - -@table @option -@item b -Set bit rate in bits/s. If the bitrate is not explicitly specified, it -is automatically set to a suitable value depending on the selected -profile. - -In case VBR mode is enabled the option is ignored. - -@item ar -Set audio sampling rate (in Hz). - -@item channels -Set the number of audio channels. - -@item flags +qscale -Enable fixed quality, VBR (Variable Bit Rate) mode. -Note that VBR is implicitly enabled when the @option{vbr} value is -positive. - -@item cutoff -Set cutoff frequency. If not specified (or explicitly set to 0) it -will use a value automatically computed by the library. Default value -is 0. - -@item profile -Set audio profile. - -The following profiles are recognized: -@table @samp -@item aac_low -Low Complexity AAC (LC) - -@item aac_he -High Efficiency AAC (HE-AAC) - -@item aac_he_v2 -High Efficiency AAC version 2 (HE-AACv2) - -@item aac_ld -Low Delay AAC (LD) - -@item aac_eld -Enhanced Low Delay AAC (ELD) -@end table - -If not specified it is set to @samp{aac_low}. -@end table - -The following are private options of the libfdk_aac encoder. - -@table @option -@item afterburner -Enable afterburner feature if set to 1, disabled if set to 0. This -improves the quality but also the required processing power. - -Default value is 1. - -@item eld_sbr -Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled -if set to 0. - -Default value is 0. - -@item signaling -Set SBR/PS signaling style. - -It can assume one of the following values: -@table @samp -@item default -choose signaling implicitly (explicit hierarchical by default, -implicit if global header is disabled) - -@item implicit -implicit backwards compatible signaling - -@item explicit_sbr -explicit SBR, implicit PS signaling - -@item explicit_hierarchical -explicit hierarchical signaling -@end table - -Default value is @samp{default}. - -@item latm -Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0. - -Default value is 0. - -@item header_period -Set StreamMuxConfig and PCE repetition period (in frames) for sending -in-band configuration buffers within LATM/LOAS transport layer. - -Must be a 16-bits non-negative integer. - -Default value is 0. - -@item vbr -Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty -good) and 5 is highest quality. A value of 0 will disable VBR, and CBR -(Constant Bit Rate) is enabled. - -Currently only the @samp{aac_low} profile supports VBR encoding. - -VBR modes 1-5 correspond to roughly the following average bit rates: - -@table @samp -@item 1 -32 kbps/channel -@item 2 -40 kbps/channel -@item 3 -48-56 kbps/channel -@item 4 -64 kbps/channel -@item 5 -about 80-96 kbps/channel -@end table - -Default value is 0. -@end table - -@subsection Examples - -@itemize -@item -Use @command{ffmpeg} to convert an audio file to VBR AAC in an M4A (MP4) -container: -@example -ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a -@end example - -@item -Use @command{ffmpeg} to convert an audio file to CBR 64k kbps AAC, using the -High-Efficiency AAC profile: -@example -ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a -@end example -@end itemize - -@anchor{libmp3lame} -@section libmp3lame - -LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper. - -Requires the presence of the libmp3lame headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libmp3lame}. - -See @ref{libshine} for a fixed-point MP3 encoder, although with a -lower quality. - -@subsection Options - -The following options are supported by the libmp3lame wrapper. The -@command{lame}-equivalent of the options are listed in parentheses. - -@table @option -@item b (@emph{-b}) -Set bitrate expressed in bits/s for CBR or ABR. LAME @code{bitrate} is -expressed in kilobits/s. - -@item q (@emph{-V}) -Set constant quality setting for VBR. This option is valid only -using the @command{ffmpeg} command-line tool. For library interface -users, use @option{global_quality}. - -@item compression_level (@emph{-q}) -Set algorithm quality. Valid arguments are integers in the 0-9 range, -with 0 meaning highest quality but slowest, and 9 meaning fastest -while producing the worst quality. - -@item reservoir -Enable use of bit reservoir when set to 1. Default value is 1. LAME -has this enabled by default, but can be overridden by use -@option{--nores} option. - -@item joint_stereo (@emph{-m j}) -Enable the encoder to use (on a frame by frame basis) either L/R -stereo or mid/side stereo. Default value is 1. - -@item abr (@emph{--abr}) -Enable the encoder to use ABR when set to 1. The @command{lame} -@option{--abr} sets the target bitrate, while this options only -tells FFmpeg to use ABR still relies on @option{b} to set bitrate. - -@end table - -@section libopencore-amrnb - -OpenCORE Adaptive Multi-Rate Narrowband encoder. - -Requires the presence of the libopencore-amrnb headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libopencore-amrnb --enable-version3}. - -This is a mono-only encoder. Officially it only supports 8000Hz sample rate, -but you can override it by setting @option{strict} to @samp{unofficial} or -lower. - -@subsection Options - -@table @option - -@item b -Set bitrate in bits per second. Only the following bitrates are supported, -otherwise libavcodec will round to the nearest valid bitrate. - -@table @option -@item 4750 -@item 5150 -@item 5900 -@item 6700 -@item 7400 -@item 7950 -@item 10200 -@item 12200 -@end table - -@item dtx -Allow discontinuous transmission (generate comfort noise) when set to 1. The -default value is 0 (disabled). - -@end table - -@anchor{libshine} -@section libshine - -Shine Fixed-Point MP3 encoder wrapper. - -Shine is a fixed-point MP3 encoder. It has a far better performance on -platforms without an FPU, e.g. armel CPUs, and some phones and tablets. -However, as it is more targeted on performance than quality, it is not on par -with LAME and other production-grade encoders quality-wise. Also, according to -the project's homepage, this encoder may not be free of bugs as the code was -written a long time ago and the project was dead for at least 5 years. - -This encoder only supports stereo and mono input. This is also CBR-only. - -The original project (last updated in early 2007) is at -@url{http://sourceforge.net/projects/libshine-fxp/}. We only support the -updated fork by the Savonet/Liquidsoap project at @url{https://github.com/savonet/shine}. - -Requires the presence of the libshine headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libshine}. - -See also @ref{libmp3lame}. - -@subsection Options - -The following options are supported by the libshine wrapper. The -@command{shineenc}-equivalent of the options are listed in parentheses. - -@table @option -@item b (@emph{-b}) -Set bitrate expressed in bits/s for CBR. @command{shineenc} @option{-b} option -is expressed in kilobits/s. - -@end table - -@section libtwolame - -TwoLAME MP2 encoder wrapper. - -Requires the presence of the libtwolame headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libtwolame}. - -@subsection Options - -The following options are supported by the libtwolame wrapper. The -@command{twolame}-equivalent options follow the FFmpeg ones and are in -parentheses. - -@table @option -@item b (@emph{-b}) -Set bitrate expressed in bits/s for CBR. @command{twolame} @option{b} -option is expressed in kilobits/s. Default value is 128k. - -@item q (@emph{-V}) -Set quality for experimental VBR support. Maximum value range is -from -50 to 50, useful range is from -10 to 10. The higher the -value, the better the quality. This option is valid only using the -@command{ffmpeg} command-line tool. For library interface users, -use @option{global_quality}. - -@item mode (@emph{--mode}) -Set the mode of the resulting audio. Possible values: - -@table @samp -@item auto -Choose mode automatically based on the input. This is the default. -@item stereo -Stereo -@item joint_stereo -Joint stereo -@item dual_channel -Dual channel -@item mono -Mono -@end table - -@item psymodel (@emph{--psyc-mode}) -Set psychoacoustic model to use in encoding. The argument must be -an integer between -1 and 4, inclusive. The higher the value, the -better the quality. The default value is 3. - -@item energy_levels (@emph{--energy}) -Enable energy levels extensions when set to 1. The default value is -0 (disabled). - -@item error_protection (@emph{--protect}) -Enable CRC error protection when set to 1. The default value is 0 -(disabled). - -@item copyright (@emph{--copyright}) -Set MPEG audio copyright flag when set to 1. The default value is 0 -(disabled). - -@item original (@emph{--original}) -Set MPEG audio original flag when set to 1. The default value is 0 -(disabled). - -@end table - -@anchor{libvo-aacenc} -@section libvo-aacenc - -VisualOn AAC encoder. - -Requires the presence of the libvo-aacenc headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libvo-aacenc --enable-version3}. - -This encoder is considered to be worse than the -@ref{aacenc,,native experimental FFmpeg AAC encoder}, according to -multiple sources. - -@subsection Options - -The VisualOn AAC encoder only support encoding AAC-LC and up to 2 -channels. It is also CBR-only. - -@table @option - -@item b -Set bit rate in bits/s. - -@end table - -@section libvo-amrwbenc - -VisualOn Adaptive Multi-Rate Wideband encoder. - -Requires the presence of the libvo-amrwbenc headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libvo-amrwbenc --enable-version3}. - -This is a mono-only encoder. Officially it only supports 16000Hz sample -rate, but you can override it by setting @option{strict} to -@samp{unofficial} or lower. - -@subsection Options - -@table @option - -@item b -Set bitrate in bits/s. Only the following bitrates are supported, otherwise -libavcodec will round to the nearest valid bitrate. - -@table @samp -@item 6600 -@item 8850 -@item 12650 -@item 14250 -@item 15850 -@item 18250 -@item 19850 -@item 23050 -@item 23850 -@end table - -@item dtx -Allow discontinuous transmission (generate comfort noise) when set to 1. The -default value is 0 (disabled). - -@end table - -@section libopus - -libopus Opus Interactive Audio Codec encoder wrapper. - -Requires the presence of the libopus headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libopus}. - -@subsection Option Mapping - -Most libopus options are modeled after the @command{opusenc} utility from -opus-tools. The following is an option mapping chart describing options -supported by the libopus wrapper, and their @command{opusenc}-equivalent -in parentheses. - -@table @option - -@item b (@emph{bitrate}) -Set the bit rate in bits/s. FFmpeg's @option{b} option is -expressed in bits/s, while @command{opusenc}'s @option{bitrate} in -kilobits/s. - -@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr}) -Set VBR mode. The FFmpeg @option{vbr} option has the following -valid arguments, with the their @command{opusenc} equivalent options -in parentheses: - -@table @samp -@item off (@emph{hard-cbr}) -Use constant bit rate encoding. - -@item on (@emph{vbr}) -Use variable bit rate encoding (the default). - -@item constrained (@emph{cvbr}) -Use constrained variable bit rate encoding. -@end table - -@item compression_level (@emph{comp}) -Set encoding algorithm complexity. Valid options are integers in -the 0-10 range. 0 gives the fastest encodes but lower quality, while 10 -gives the highest quality but slowest encoding. The default is 10. - -@item frame_duration (@emph{framesize}) -Set maximum frame size, or duration of a frame in milliseconds. The -argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller -frame sizes achieve lower latency but less quality at a given bitrate. -Sizes greater than 20ms are only interesting at fairly low bitrates. -The default is 20ms. - -@item packet_loss (@emph{expect-loss}) -Set expected packet loss percentage. The default is 0. - -@item application (N.A.) -Set intended application type. Valid options are listed below: - -@table @samp -@item voip -Favor improved speech intelligibility. -@item audio -Favor faithfulness to the input (the default). -@item lowdelay -Restrict to only the lowest delay modes. -@end table - -@item cutoff (N.A.) -Set cutoff bandwidth in Hz. The argument must be exactly one of the -following: 4000, 6000, 8000, 12000, or 20000, corresponding to -narrowband, mediumband, wideband, super wideband, and fullband -respectively. The default is 0 (cutoff disabled). - -@end table - -@section libvorbis - -libvorbis encoder wrapper. - -Requires the presence of the libvorbisenc headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libvorbis}. - -@subsection Options - -The following options are supported by the libvorbis wrapper. The -@command{oggenc}-equivalent of the options are listed in parentheses. - -To get a more accurate and extensive documentation of the libvorbis -options, consult the libvorbisenc's and @command{oggenc}'s documentations. -See @url{http://xiph.org/vorbis/}, -@url{http://wiki.xiph.org/Vorbis-tools}, and oggenc(1). - -@table @option -@item b (@emph{-b}) -Set bitrate expressed in bits/s for ABR. @command{oggenc} @option{-b} is -expressed in kilobits/s. - -@item q (@emph{-q}) -Set constant quality setting for VBR. The value should be a float -number in the range of -1.0 to 10.0. The higher the value, the better -the quality. The default value is @samp{3.0}. - -This option is valid only using the @command{ffmpeg} command-line tool. -For library interface users, use @option{global_quality}. - -@item cutoff (@emph{--advanced-encode-option lowpass_frequency=N}) -Set cutoff bandwidth in Hz, a value of 0 disables cutoff. @command{oggenc}'s -related option is expressed in kHz. The default value is @samp{0} (cutoff -disabled). - -@item minrate (@emph{-m}) -Set minimum bitrate expressed in bits/s. @command{oggenc} @option{-m} is -expressed in kilobits/s. - -@item maxrate (@emph{-M}) -Set maximum bitrate expressed in bits/s. @command{oggenc} @option{-M} is -expressed in kilobits/s. This only has effect on ABR mode. - -@item iblock (@emph{--advanced-encode-option impulse_noisetune=N}) -Set noise floor bias for impulse blocks. The value is a float number from --15.0 to 0.0. A negative bias instructs the encoder to pay special attention -to the crispness of transients in the encoded audio. The tradeoff for better -transient response is a higher bitrate. - -@end table - -@anchor{libwavpack} @section libwavpack A wrapper providing WavPack encoding through libwavpack. Only lossless mode using 32-bit integer samples is supported currently. - -Requires the presence of the libwavpack headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libwavpack}. - -Note that a libavcodec-native encoder for the WavPack codec exists so users can -encode audios with this codec without using this encoder. See @ref{wavpackenc}. - -@subsection Options - -@command{wavpack} command line utility's corresponding options are listed in -parentheses, if any. +The @option{compression_level} option can be used to control speed vs. +compression tradeoff, with the values mapped to libwavpack as follows: @table @option -@item frame_size (@emph{--blocksize}) -Default is 32768. -@item compression_level -Set speed vs. compression tradeoff. Acceptable arguments are listed below: - -@table @samp -@item 0 (@emph{-f}) -Fast mode. +@item 0 +Fast mode - corresponding to the wavpack @option{-f} option. @item 1 Normal (default) settings. -@item 2 (@emph{-h}) -High quality. +@item 2 +High quality - corresponding to the wavpack @option{-h} option. -@item 3 (@emph{-hh}) -Very high quality. +@item 3 +Very high quality - corresponding to the wavpack @option{-hh} option. -@item 4-8 (@emph{-hh -x}@var{EXTRAPROC}) -Same as @samp{3}, but with extra processing enabled. - -@samp{4} is the same as @option{-x2} and @samp{8} is the same as @option{-x6}. - -@end table -@end table - -@anchor{wavpackenc} -@section wavpack - -WavPack lossless audio encoder. - -This is a libavcodec-native WavPack encoder. There is also an encoder based on -libwavpack, but there is virtually no reason to use that encoder. - -See also @ref{libwavpack}. - -@subsection Options - -The equivalent options for @command{wavpack} command line utility are listed in -parentheses. - -@subsubsection Shared options - -The following shared options are effective for this encoder. Only special notes -about this particular encoder will be documented here. For the general meaning -of the options, see @ref{codec-options,,the Codec Options chapter}. - -@table @option -@item frame_size (@emph{--blocksize}) -For this encoder, the range for this option is between 128 and 131072. Default -is automatically decided based on sample rate and number of channel. - -For the complete formula of calculating default, see -@file{libavcodec/wavpackenc.c}. - -@item compression_level (@emph{-f}, @emph{-h}, @emph{-hh}, and @emph{-x}) -This option's syntax is consistent with @ref{libwavpack}'s. -@end table - -@subsubsection Private options - -@table @option -@item joint_stereo (@emph{-j}) -Set whether to enable joint stereo. Valid values are: - -@table @samp -@item on (@emph{1}) -Force mid/side audio encoding. -@item off (@emph{0}) -Force left/right audio encoding. -@item auto -Let the encoder decide automatically. -@end table - -@item optimize_mono -Set whether to enable optimization for mono. This option is only effective for -non-mono streams. Available values: - -@table @samp -@item on -enabled -@item off -disabled -@end table +@item 4-8 +Same as 3, but with extra processing enabled - corresponding to the wavpack +@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as +@option{-x6}. @end table @@ -1260,194 +446,6 @@ disabled @chapter Video Encoders @c man begin VIDEO ENCODERS -A description of some of the currently available video encoders -follows. - -@section libtheora - -libtheora Theora encoder wrapper. - -Requires the presence of the libtheora headers and library during -configuration. You need to explicitly configure the build with -@code{--enable-libtheora}. - -For more information about the libtheora project see -@url{http://www.theora.org/}. - -@subsection Options - -The following global options are mapped to internal libtheora options -which affect the quality and the bitrate of the encoded stream. - -@table @option -@item b -Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In -case VBR (Variable Bit Rate) mode is enabled this option is ignored. - -@item flags -Used to enable constant quality mode (VBR) encoding through the -@option{qscale} flag, and to enable the @code{pass1} and @code{pass2} -modes. - -@item g -Set the GOP size. - -@item global_quality -Set the global quality as an integer in lambda units. - -Only relevant when VBR mode is enabled with @code{flags +qscale}. The -value is converted to QP units by dividing it by @code{FF_QP2LAMBDA}, -clipped in the [0 - 10] range, and then multiplied by 6.3 to get a -value in the native libtheora range [0-63]. A higher value corresponds -to a higher quality. - -@item q -Enable VBR mode when set to a non-negative value, and set constant -quality value as a double floating point value in QP units. - -The value is clipped in the [0-10] range, and then multiplied by 6.3 -to get a value in the native libtheora range [0-63]. - -This option is valid only using the @command{ffmpeg} command-line -tool. For library interface users, use @option{global_quality}. -@end table - -@subsection Examples - -@itemize -@item -Set maximum constant quality (VBR) encoding with @command{ffmpeg}: -@example -ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg -@end example - -@item -Use @command{ffmpeg} to convert a CBR 1000 kbps Theora video stream: -@example -ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg -@end example -@end itemize - -@section libvpx - -VP8 format supported through libvpx. - -Requires the presence of the libvpx headers and library during configuration. -You need to explicitly configure the build with @code{--enable-libvpx}. - -@subsection Options - -Mapping from FFmpeg to libvpx options with conversion notes in parentheses. - -@table @option - -@item threads -g_threads - -@item profile -g_profile - -@item vb -rc_target_bitrate - -@item g -kf_max_dist - -@item keyint_min -kf_min_dist - -@item qmin -rc_min_quantizer - -@item qmax -rc_max_quantizer - -@item bufsize, vb -rc_buf_sz -@code{(bufsize * 1000 / vb)} - -rc_buf_optimal_sz -@code{(bufsize * 1000 / vb * 5 / 6)} - -@item rc_init_occupancy, vb -rc_buf_initial_sz -@code{(rc_init_occupancy * 1000 / vb)} - -@item rc_buffer_aggressivity -rc_undershoot_pct - -@item skip_threshold -rc_dropframe_thresh - -@item qcomp -rc_2pass_vbr_bias_pct - -@item maxrate, vb -rc_2pass_vbr_maxsection_pct -@code{(maxrate * 100 / vb)} - -@item minrate, vb -rc_2pass_vbr_minsection_pct -@code{(minrate * 100 / vb)} - -@item minrate, maxrate, vb -@code{VPX_CBR} -@code{(minrate == maxrate == vb)} - -@item crf -@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL} - -@item quality -@table @option -@item @var{best} -@code{VPX_DL_BEST_QUALITY} -@item @var{good} -@code{VPX_DL_GOOD_QUALITY} -@item @var{realtime} -@code{VPX_DL_REALTIME} -@end table - -@item speed -@code{VP8E_SET_CPUUSED} - -@item nr -@code{VP8E_SET_NOISE_SENSITIVITY} - -@item mb_threshold -@code{VP8E_SET_STATIC_THRESHOLD} - -@item slices -@code{VP8E_SET_TOKEN_PARTITIONS} - -@item max-intra-rate -@code{VP8E_SET_MAX_INTRA_BITRATE_PCT} - -@item force_key_frames -@code{VPX_EFLAG_FORCE_KF} - -@item Alternate reference frame related -@table @option -@item vp8flags altref -@code{VP8E_SET_ENABLEAUTOALTREF} -@item @var{arnr_max_frames} -@code{VP8E_SET_ARNR_MAXFRAMES} -@item @var{arnr_type} -@code{VP8E_SET_ARNR_TYPE} -@item @var{arnr_strength} -@code{VP8E_SET_ARNR_STRENGTH} -@item @var{rc_lookahead} -g_lag_in_frames -@end table - -@item vp8flags error_resilient -g_error_resilient - -@end table - -For more information about libvpx see: -@url{http://www.webmproject.org/} - - @section libwebp libwebp WebP Image encoder wrapper @@ -1507,489 +505,6 @@ Small-sized colorful images Text-like @end table -@end table - -@section libx264, libx264rgb - -x264 H.264/MPEG-4 AVC encoder wrapper. - -This encoder requires the presence of the libx264 headers and library -during configuration. You need to explicitly configure the build with -@code{--enable-libx264}. - -libx264 supports an impressive number of features, including 8x8 and -4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC -entropy coding, interlacing (MBAFF), lossless mode, psy optimizations -for detail retention (adaptive quantization, psy-RD, psy-trellis). - -Many libx264 encoder options are mapped to FFmpeg global codec -options, while unique encoder options are provided through private -options. Additionally the @option{x264opts} and @option{x264-params} -private options allows one to pass a list of key=value tuples as accepted -by the libx264 @code{x264_param_parse} function. - -The x264 project website is at -@url{http://www.videolan.org/developers/x264.html}. - -The libx264rgb encoder is the same as libx264, except it accepts packed RGB -pixel formats as input instead of YUV. - -@subsection Supported Pixel Formats - -x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at -x264's configure time. FFmpeg only supports one bit depth in one particular -build. In other words, it is not possible to build one FFmpeg with multiple -versions of x264 with different bit depths. - -@subsection Options - -The following options are supported by the libx264 wrapper. The -@command{x264}-equivalent options or values are listed in parentheses -for easy migration. - -To reduce the duplication of documentation, only the private options -and some others requiring special attention are documented here. For -the documentation of the undocumented generic options, see -@ref{codec-options,,the Codec Options chapter}. - -To get a more accurate and extensive documentation of the libx264 -options, invoke the command @command{x264 --full-help} or consult -the libx264 documentation. - -@table @option -@item b (@emph{bitrate}) -Set bitrate in bits/s. Note that FFmpeg's @option{b} option is -expressed in bits/s, while @command{x264}'s @option{bitrate} is in -kilobits/s. - -@item bf (@emph{bframes}) - -@item g (@emph{keyint}) - -@item qmin (@emph{qpmin}) -Minimum quantizer scale. - -@item qmax (@emph{qpmax}) -Maximum quantizer scale. - -@item qdiff (@emph{qpstep}) -Maximum difference between quantizer scales. - -@item qblur (@emph{qblur}) -Quantizer curve blur - -@item qcomp (@emph{qcomp}) -Quantizer curve compression factor - -@item refs (@emph{ref}) -Number of reference frames each P-frame can use. The range is from @var{0-16}. - -@item sc_threshold (@emph{scenecut}) -Sets the threshold for the scene change detection. - -@item trellis (@emph{trellis}) -Performs Trellis quantization to increase efficiency. Enabled by default. - -@item nr (@emph{nr}) - -@item me_range (@emph{merange}) -Maximum range of the motion search in pixels. - -@item me_method (@emph{me}) -Set motion estimation method. Possible values in the decreasing order -of speed: - -@table @samp -@item dia (@emph{dia}) -@item epzs (@emph{dia}) -Diamond search with radius 1 (fastest). @samp{epzs} is an alias for -@samp{dia}. -@item hex (@emph{hex}) -Hexagonal search with radius 2. -@item umh (@emph{umh}) -Uneven multi-hexagon search. -@item esa (@emph{esa}) -Exhaustive search. -@item tesa (@emph{tesa}) -Hadamard exhaustive search (slowest). -@end table - -@item subq (@emph{subme}) -Sub-pixel motion estimation method. - -@item b_strategy (@emph{b-adapt}) -Adaptive B-frame placement decision algorithm. Use only on first-pass. - -@item keyint_min (@emph{min-keyint}) -Minimum GOP size. - -@item coder -Set entropy encoder. Possible values: - -@table @samp -@item ac -Enable CABAC. - -@item vlc -Enable CAVLC and disable CABAC. It generates the same effect as -@command{x264}'s @option{--no-cabac} option. -@end table - -@item cmp -Set full pixel motion estimation comparation algorithm. Possible values: - -@table @samp -@item chroma -Enable chroma in motion estimation. - -@item sad -Ignore chroma in motion estimation. It generates the same effect as -@command{x264}'s @option{--no-chroma-me} option. -@end table - -@item threads (@emph{threads}) -Number of encoding threads. - -@item thread_type -Set multithreading technique. Possible values: - -@table @samp -@item slice -Slice-based multithreading. It generates the same effect as -@command{x264}'s @option{--sliced-threads} option. -@item frame -Frame-based multithreading. -@end table - -@item flags -Set encoding flags. It can be used to disable closed GOP and enable -open GOP by setting it to @code{-cgop}. The result is similar to -the behavior of @command{x264}'s @option{--open-gop} option. - -@item rc_init_occupancy (@emph{vbv-init}) - -@item preset (@emph{preset}) -Set the encoding preset. - -@item tune (@emph{tune}) -Set tuning of the encoding params. - -@item profile (@emph{profile}) -Set profile restrictions. - -@item fastfirstpass -Enable fast settings when encoding first pass, when set to 1. When set -to 0, it has the same effect of @command{x264}'s -@option{--slow-firstpass} option. - -@item crf (@emph{crf}) -Set the quality for constant quality mode. - -@item crf_max (@emph{crf-max}) -In CRF mode, prevents VBV from lowering quality beyond this point. - -@item qp (@emph{qp}) -Set constant quantization rate control method parameter. - -@item aq-mode (@emph{aq-mode}) -Set AQ method. Possible values: - -@table @samp -@item none (@emph{0}) -Disabled. - -@item variance (@emph{1}) -Variance AQ (complexity mask). - -@item autovariance (@emph{2}) -Auto-variance AQ (experimental). -@end table - -@item aq-strength (@emph{aq-strength}) -Set AQ strength, reduce blocking and blurring in flat and textured areas. - -@item psy -Use psychovisual optimizations when set to 1. When set to 0, it has the -same effect as @command{x264}'s @option{--no-psy} option. - -@item psy-rd (@emph{psy-rd}) -Set strength of psychovisual optimization, in -@var{psy-rd}:@var{psy-trellis} format. - -@item rc-lookahead (@emph{rc-lookahead}) -Set number of frames to look ahead for frametype and ratecontrol. - -@item weightb -Enable weighted prediction for B-frames when set to 1. When set to 0, -it has the same effect as @command{x264}'s @option{--no-weightb} option. - -@item weightp (@emph{weightp}) -Set weighted prediction method for P-frames. Possible values: - -@table @samp -@item none (@emph{0}) -Disabled -@item simple (@emph{1}) -Enable only weighted refs -@item smart (@emph{2}) -Enable both weighted refs and duplicates -@end table - -@item ssim (@emph{ssim}) -Enable calculation and printing SSIM stats after the encoding. - -@item intra-refresh (@emph{intra-refresh}) -Enable the use of Periodic Intra Refresh instead of IDR frames when set -to 1. - -@item bluray-compat (@emph{bluray-compat}) -Configure the encoder to be compatible with the bluray standard. -It is a shorthand for setting "bluray-compat=1 force-cfr=1". - -@item b-bias (@emph{b-bias}) -Set the influence on how often B-frames are used. - -@item b-pyramid (@emph{b-pyramid}) -Set method for keeping of some B-frames as references. Possible values: - -@table @samp -@item none (@emph{none}) -Disabled. -@item strict (@emph{strict}) -Strictly hierarchical pyramid. -@item normal (@emph{normal}) -Non-strict (not Blu-ray compatible). -@end table - -@item mixed-refs -Enable the use of one reference per partition, as opposed to one -reference per macroblock when set to 1. When set to 0, it has the -same effect as @command{x264}'s @option{--no-mixed-refs} option. - -@item 8x8dct -Enable adaptive spatial transform (high profile 8x8 transform) -when set to 1. When set to 0, it has the same effect as -@command{x264}'s @option{--no-8x8dct} option. - -@item fast-pskip -Enable early SKIP detection on P-frames when set to 1. When set -to 0, it has the same effect as @command{x264}'s -@option{--no-fast-pskip} option. - -@item aud (@emph{aud}) -Enable use of access unit delimiters when set to 1. - -@item mbtree -Enable use macroblock tree ratecontrol when set to 1. When set -to 0, it has the same effect as @command{x264}'s -@option{--no-mbtree} option. - -@item deblock (@emph{deblock}) -Set loop filter parameters, in @var{alpha}:@var{beta} form. - -@item cplxblur (@emph{cplxblur}) -Set fluctuations reduction in QP (before curve compression). - -@item partitions (@emph{partitions}) -Set partitions to consider as a comma-separated list of. Possible -values in the list: - -@table @samp -@item p8x8 -8x8 P-frame partition. -@item p4x4 -4x4 P-frame partition. -@item b8x8 -4x4 B-frame partition. -@item i8x8 -8x8 I-frame partition. -@item i4x4 -4x4 I-frame partition. -(Enabling @samp{p4x4} requires @samp{p8x8} to be enabled. Enabling -@samp{i8x8} requires adaptive spatial transform (@option{8x8dct} -option) to be enabled.) -@item none (@emph{none}) -Do not consider any partitions. -@item all (@emph{all}) -Consider every partition. -@end table - -@item direct-pred (@emph{direct}) -Set direct MV prediction mode. Possible values: - -@table @samp -@item none (@emph{none}) -Disable MV prediction. -@item spatial (@emph{spatial}) -Enable spatial predicting. -@item temporal (@emph{temporal}) -Enable temporal predicting. -@item auto (@emph{auto}) -Automatically decided. -@end table - -@item slice-max-size (@emph{slice-max-size}) -Set the limit of the size of each slice in bytes. If not specified -but RTP payload size (@option{ps}) is specified, that is used. - -@item stats (@emph{stats}) -Set the file name for multi-pass stats. - -@item nal-hrd (@emph{nal-hrd}) -Set signal HRD information (requires @option{vbv-bufsize} to be set). -Possible values: - -@table @samp -@item none (@emph{none}) -Disable HRD information signaling. -@item vbr (@emph{vbr}) -Variable bit rate. -@item cbr (@emph{cbr}) -Constant bit rate (not allowed in MP4 container). -@end table - -@item x264opts (N.A.) -Set any x264 option, see @command{x264 --fullhelp} for a list. - -Argument is a list of @var{key}=@var{value} couples separated by -":". In @var{filter} and @var{psy-rd} options that use ":" as a separator -themselves, use "," instead. They accept it as well since long ago but this -is kept undocumented for some reason. - -For example to specify libx264 encoding options with @command{ffmpeg}: -@example -ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv -@end example - -@item x264-params (N.A.) -Override the x264 configuration using a :-separated list of key=value -parameters. - -This option is functionally the same as the @option{x264opts}, but is -duplicated for compatibility with the Libav fork. - -For example to specify libx264 encoding options with @command{ffmpeg}: -@example -ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\ -cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\ -no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT -@end example -@end table - -Encoding ffpresets for common usages are provided so they can be used with the -general presets system (e.g. passing the @option{pre} option). - -@section libxvid - -Xvid MPEG-4 Part 2 encoder wrapper. - -This encoder requires the presence of the libxvidcore headers and library -during configuration. You need to explicitly configure the build with -@code{--enable-libxvid --enable-gpl}. - -The native @code{mpeg4} encoder supports the MPEG-4 Part 2 format, so -users can encode to this format without this library. - -@subsection Options - -The following options are supported by the libxvid wrapper. Some of -the following options are listed but are not documented, and -correspond to shared codec options. See @ref{codec-options,,the Codec -Options chapter} for their documentation. The other shared options -which are not listed have no effect for the libxvid encoder. - -@table @option -@item b - -@item g - -@item qmin - -@item qmax - -@item mpeg_quant - -@item threads - -@item bf - -@item b_qfactor - -@item b_qoffset - -@item flags -Set specific encoding flags. Possible values: - -@table @samp - -@item mv4 -Use four motion vector by macroblock. - -@item aic -Enable high quality AC prediction. - -@item gray -Only encode grayscale. - -@item gmc -Enable the use of global motion compensation (GMC). - -@item qpel -Enable quarter-pixel motion compensation. - -@item cgop -Enable closed GOP. - -@item global_header -Place global headers in extradata instead of every keyframe. - -@end table - -@item trellis - -@item me_method -Set motion estimation method. Possible values in decreasing order of -speed and increasing order of quality: - -@table @samp -@item zero -Use no motion estimation (default). - -@item phods -@item x1 -@item log -Enable advanced diamond zonal search for 16x16 blocks and half-pixel -refinement for 16x16 blocks. @samp{x1} and @samp{log} are aliases for -@samp{phods}. - -@item epzs -Enable all of the things described above, plus advanced diamond zonal -search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion -estimation on chroma planes. - -@item full -Enable all of the things described above, plus extended 16x16 and 8x8 -blocks search. -@end table - -@item mbd -Set macroblock decision algorithm. Possible values in the increasing -order of quality: - -@table @samp -@item simple -Use macroblock comparing function algorithm (default). - -@item bits -Enable rate distortion-based half pixel and quarter pixel refinement for -16x16 blocks. - -@item rd -Enable all of the things described above, plus rate distortion-based -half pixel and quarter pixel refinement for 8x8 blocks, and rate -distortion-based search using square pattern. -@end table - @item lumi_aq Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled). @@ -2042,27 +557,205 @@ fastest. @end table -@section png +@section libx264 -PNG image encoder. +x264 H.264/MPEG-4 AVC encoder wrapper -@subsection Private options +x264 supports an impressive number of features, including 8x8 and 4x4 adaptive +spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, +interlacing (MBAFF), lossless mode, psy optimizations for detail retention +(adaptive quantization, psy-RD, psy-trellis). +The Libav wrapper provides a mapping for most of them using global options +that match those of the encoders and provides private options for the unique +encoder options. Additionally an expert override is provided to directly pass +a list of key=value tuples as accepted by x264_param_parse. + +@subsection Option Mapping + +The following options are supported by the x264 wrapper, the x264-equivalent +options follow the Libav ones. + +@multitable { } { } { } +@item b @tab bitrate +@tab Libav @code{b} option is expressed in bits/s, x264 @code{bitrate} in kilobits/s. +@item bf @tab bframes +@tab Maximum number of B-frames. +@item g @tab keyint +@tab Maximum GOP size. +@item qmin @tab qpmin +@tab Minimum quantizer scale. +@item qmax @tab qpmax +@tab Maximum quantizer scale. +@item qdiff @tab qpstep +@tab Maximum difference between quantizer scales. +@item qblur @tab qblur +@tab Quantizer curve blur +@item qcomp @tab qcomp +@tab Quantizer curve compression factor +@item refs @tab ref +@tab Number of reference frames each P-frame can use. The range is from @var{0-16}. +@item sc_threshold @tab scenecut +@tab Sets the threshold for the scene change detection. +@item trellis @tab trellis +@tab Performs Trellis quantization to increase efficiency. Enabled by default. +@item nr @tab nr +@tab Noise reduction. +@item me_range @tab merange +@tab Maximum range of the motion search in pixels. +@item me_method @tab me +@tab Full-pixel motion estimation method. +@item subq @tab subme +@tab Sub-pixel motion estimation method. +@item b_strategy @tab b-adapt +@tab Adaptive B-frame placement decision algorithm. Use only on first-pass. +@item keyint_min @tab min-keyint +@tab Minimum GOP size. +@item coder @tab cabac +@tab Set coder to @code{ac} to use CABAC. +@item cmp @tab chroma-me +@tab Set to @code{chroma} to use chroma motion estimation. +@item threads @tab threads +@tab Number of encoding threads. +@item thread_type @tab sliced_threads +@tab Set to @code{slice} to use sliced threading instead of frame threading. +@item flags -cgop @tab open-gop +@tab Set @code{-cgop} to use recovery points to close GOPs. +@item rc_init_occupancy @tab vbv-init +@tab Initial buffer occupancy. +@end multitable + +@subsection Private Options @table @option -@item dpi @var{integer} -Set physical density of pixels, in dots per inch, unset by default -@item dpm @var{integer} -Set physical density of pixels, in dots per meter, unset by default +@item -preset @var{string} +Set the encoding preset (cf. x264 --fullhelp). +@item -tune @var{string} +Tune the encoding params (cf. x264 --fullhelp). +@item -profile @var{string} +Set profile restrictions (cf. x264 --fullhelp). +@item -fastfirstpass @var{integer} +Use fast settings when encoding first pass. +@item -crf @var{float} +Select the quality for constant quality mode. +@item -crf_max @var{float} +In CRF mode, prevents VBV from lowering quality beyond this point. +@item -qp @var{integer} +Constant quantization parameter rate control method. +@item -aq-mode @var{integer} +AQ method + +Possible values: +@table @samp +@item none + +@item variance +Variance AQ (complexity mask). +@item autovariance +Auto-variance AQ (experimental). @end table +@item -aq-strength @var{float} +AQ strength, reduces blocking and blurring in flat and textured areas. +@item -psy @var{integer} +Use psychovisual optimizations. +@item -psy-rd @var{string} +Strength of psychovisual optimization, in : format. +@item -rc-lookahead @var{integer} +Number of frames to look ahead for frametype and ratecontrol. +@item -weightb @var{integer} +Weighted prediction for B-frames. +@item -weightp @var{integer} +Weighted prediction analysis method. + +Possible values: +@table @samp +@item none + +@item simple + +@item smart + +@end table +@item -ssim @var{integer} +Calculate and print SSIM stats. +@item -intra-refresh @var{integer} +Use Periodic Intra Refresh instead of IDR frames. +@item -bluray-compat @var{integer} +Configure the encoder to be compatible with the bluray standard. +It is a shorthand for setting "bluray-compat=1 force-cfr=1". +@item -b-bias @var{integer} +Influences how often B-frames are used. +@item -b-pyramid @var{integer} +Keep some B-frames as references. + +Possible values: +@table @samp +@item none + +@item strict +Strictly hierarchical pyramid. +@item normal +Non-strict (not Blu-ray compatible). +@end table +@item -mixed-refs @var{integer} +One reference per partition, as opposed to one reference per macroblock. +@item -8x8dct @var{integer} +High profile 8x8 transform. +@item -fast-pskip @var{integer} +@item -aud @var{integer} +Use access unit delimiters. +@item -mbtree @var{integer} +Use macroblock tree ratecontrol. +@item -deblock @var{string} +Loop filter parameters, in form. +@item -cplxblur @var{float} +Reduce fluctuations in QP (before curve compression). +@item -partitions @var{string} +A comma-separated list of partitions to consider, possible values: p8x8, p4x4, b8x8, i8x8, i4x4, none, all. +@item -direct-pred @var{integer} +Direct MV prediction mode + +Possible values: +@table @samp +@item none + +@item spatial + +@item temporal + +@item auto + +@end table +@item -slice-max-size @var{integer} +Limit the size of each slice in bytes. +@item -stats @var{string} +Filename for 2 pass stats. +@item -nal-hrd @var{integer} +Signal HRD information (requires vbv-bufsize; cbr not allowed in .mp4). + +Possible values: +@table @samp +@item none + +@item vbr + +@item cbr + +@end table +@item -x264-params @var{string} +Override the x264 configuration using a :-separated list of key=value parameters. +@example +-x264-params level=30:bframes=0:weightp=0:cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 +@end example +@end table + +Encoding avpresets for common usages are provided so they can be used with the +general presets system (e.g. passing the @code{-pre} option). @section ProRes Apple ProRes encoder. -FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder. -The used encoder can be chosen with the @code{-vcodec} option. - -@subsection Private Options for prores-ks +@subsection Private Options @table @option @item profile @var{integer} diff --git a/doc/errno.txt b/doc/errno.txt deleted file mode 100644 index 31cab26fc2..0000000000 --- a/doc/errno.txt +++ /dev/null @@ -1,174 +0,0 @@ -The following table lists most error codes found in various operating -systems supported by FFmpeg. - - OS -Code Std F LBMWwb Text (YMMV) - -E2BIG POSIX ++++++ Argument list too long -EACCES POSIX ++++++ Permission denied -EADDRINUSE POSIX +++..+ Address in use -EADDRNOTAVAIL POSIX +++..+ Cannot assign requested address -EADV +..... Advertise error -EAFNOSUPPORT POSIX +++..+ Address family not supported -EAGAIN POSIX + ++++++ Resource temporarily unavailable -EALREADY POSIX +++..+ Operation already in progress -EAUTH .++... Authentication error -EBADARCH ..+... Bad CPU type in executable -EBADE +..... Invalid exchange -EBADEXEC ..+... Bad executable -EBADF POSIX ++++++ Bad file descriptor -EBADFD +..... File descriptor in bad state -EBADMACHO ..+... Malformed Macho file -EBADMSG POSIX ++4... Bad message -EBADR +..... Invalid request descriptor -EBADRPC .++... RPC struct is bad -EBADRQC +..... Invalid request code -EBADSLT +..... Invalid slot -EBFONT +..... Bad font file format -EBUSY POSIX - ++++++ Device or resource busy -ECANCELED POSIX +++... Operation canceled -ECHILD POSIX ++++++ No child processes -ECHRNG +..... Channel number out of range -ECOMM +..... Communication error on send -ECONNABORTED POSIX +++..+ Software caused connection abort -ECONNREFUSED POSIX - +++ss+ Connection refused -ECONNRESET POSIX +++..+ Connection reset -EDEADLK POSIX ++++++ Resource deadlock avoided -EDEADLOCK +..++. File locking deadlock error -EDESTADDRREQ POSIX +++... Destination address required -EDEVERR ..+... Device error -EDOM C89 - ++++++ Numerical argument out of domain -EDOOFUS .F.... Programming error -EDOTDOT +..... RFS specific error -EDQUOT POSIX +++... Disc quota exceeded -EEXIST POSIX ++++++ File exists -EFAULT POSIX - ++++++ Bad address -EFBIG POSIX - ++++++ File too large -EFTYPE .++... Inappropriate file type or format -EHOSTDOWN +++... Host is down -EHOSTUNREACH POSIX +++..+ No route to host -EHWPOISON +..... Memory page has hardware error -EIDRM POSIX +++... Identifier removed -EILSEQ C99 ++++++ Illegal byte sequence -EINPROGRESS POSIX - +++ss+ Operation in progress -EINTR POSIX - ++++++ Interrupted system call -EINVAL POSIX + ++++++ Invalid argument -EIO POSIX + ++++++ I/O error -EISCONN POSIX +++..+ Socket is already connected -EISDIR POSIX ++++++ Is a directory -EISNAM +..... Is a named type file -EKEYEXPIRED +..... Key has expired -EKEYREJECTED +..... Key was rejected by service -EKEYREVOKED +..... Key has been revoked -EL2HLT +..... Level 2 halted -EL2NSYNC +..... Level 2 not synchronized -EL3HLT +..... Level 3 halted -EL3RST +..... Level 3 reset -ELIBACC +..... Can not access a needed shared library -ELIBBAD +..... Accessing a corrupted shared library -ELIBEXEC +..... Cannot exec a shared library directly -ELIBMAX +..... Too many shared libraries -ELIBSCN +..... .lib section in a.out corrupted -ELNRNG +..... Link number out of range -ELOOP POSIX +++..+ Too many levels of symbolic links -EMEDIUMTYPE +..... Wrong medium type -EMFILE POSIX ++++++ Too many open files -EMLINK POSIX ++++++ Too many links -EMSGSIZE POSIX +++..+ Message too long -EMULTIHOP POSIX ++4... Multihop attempted -ENAMETOOLONG POSIX - ++++++ Filen ame too long -ENAVAIL +..... No XENIX semaphores available -ENEEDAUTH .++... Need authenticator -ENETDOWN POSIX +++..+ Network is down -ENETRESET SUSv3 +++..+ Network dropped connection on reset -ENETUNREACH POSIX +++..+ Network unreachable -ENFILE POSIX ++++++ Too many open files in system -ENOANO +..... No anode -ENOATTR .++... Attribute not found -ENOBUFS POSIX - +++..+ No buffer space available -ENOCSI +..... No CSI structure available -ENODATA XSR +N4... No message available -ENODEV POSIX - ++++++ No such device -ENOENT POSIX - ++++++ No such file or directory -ENOEXEC POSIX ++++++ Exec format error -ENOFILE ...++. No such file or directory -ENOKEY +..... Required key not available -ENOLCK POSIX ++++++ No locks available -ENOLINK POSIX ++4... Link has been severed -ENOMEDIUM +..... No medium found -ENOMEM POSIX ++++++ Not enough space -ENOMSG POSIX +++..+ No message of desired type -ENONET +..... Machine is not on the network -ENOPKG +..... Package not installed -ENOPROTOOPT POSIX +++..+ Protocol not available -ENOSPC POSIX ++++++ No space left on device -ENOSR XSR +N4... No STREAM resources -ENOSTR XSR +N4... Not a STREAM -ENOSYS POSIX + ++++++ Function not implemented -ENOTBLK +++... Block device required -ENOTCONN POSIX +++..+ Socket is not connected -ENOTDIR POSIX ++++++ Not a directory -ENOTEMPTY POSIX ++++++ Directory not empty -ENOTNAM +..... Not a XENIX named type file -ENOTRECOVERABLE SUSv4 - +..... State not recoverable -ENOTSOCK POSIX +++..+ Socket operation on non-socket -ENOTSUP POSIX +++... Operation not supported -ENOTTY POSIX ++++++ Inappropriate I/O control operation -ENOTUNIQ +..... Name not unique on network -ENXIO POSIX ++++++ No such device or address -EOPNOTSUPP POSIX +++..+ Operation not supported (on socket) -EOVERFLOW POSIX +++..+ Value too large to be stored in data type -EOWNERDEAD SUSv4 +..... Owner died -EPERM POSIX - ++++++ Operation not permitted -EPFNOSUPPORT +++..+ Protocol family not supported -EPIPE POSIX - ++++++ Broken pipe -EPROCLIM .++... Too many processes -EPROCUNAVAIL .++... Bad procedure for program -EPROGMISMATCH .++... Program version wrong -EPROGUNAVAIL .++... RPC prog. not avail -EPROTO POSIX ++4... Protocol error -EPROTONOSUPPORT POSIX - +++ss+ Protocol not supported -EPROTOTYPE POSIX +++..+ Protocol wrong type for socket -EPWROFF ..+... Device power is off -ERANGE C89 - ++++++ Result too large -EREMCHG +..... Remote address changed -EREMOTE +++... Object is remote -EREMOTEIO +..... Remote I/O error -ERESTART +..... Interrupted system call should be restarted -ERFKILL +..... Operation not possible due to RF-kill -EROFS POSIX ++++++ Read-only file system -ERPCMISMATCH .++... RPC version wrong -ESHLIBVERS ..+... Shared library version mismatch -ESHUTDOWN +++..+ Cannot send after socket shutdown -ESOCKTNOSUPPORT +++... Socket type not supported -ESPIPE POSIX ++++++ Illegal seek -ESRCH POSIX ++++++ No such process -ESRMNT +..... Srmount error -ESTALE POSIX +++..+ Stale NFS file handle -ESTRPIPE +..... Streams pipe error -ETIME XSR +N4... Stream ioctl timeout -ETIMEDOUT POSIX - +++ss+ Connection timed out -ETOOMANYREFS +++... Too many references: cannot splice -ETXTBSY POSIX +++... Text file busy -EUCLEAN +..... Structure needs cleaning -EUNATCH +..... Protocol driver not attached -EUSERS +++... Too many users -EWOULDBLOCK POSIX +++..+ Operation would block -EXDEV POSIX ++++++ Cross-device link -EXFULL +..... Exchange full - -Notations: - -F: used in FFmpeg (-: a few times, +: a lot) - -SUSv3: Single Unix Specification, version 3 -SUSv4: Single Unix Specification, version 4 -XSR: XSI STREAMS (obsolete) - -OS: availability on some supported operating systems -L: GNU/Linux -B: BSD (F: FreeBSD, N: NetBSD) -M: MacOS X -W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h) -w: Mingw32 (3.17) and Mingw64 (2.0.1) -b: BeOS diff --git a/doc/eval.texi b/doc/eval.texi new file mode 100644 index 0000000000..e1fd7ee484 --- /dev/null +++ b/doc/eval.texi @@ -0,0 +1,156 @@ +@chapter Expression Evaluation +@c man begin EXPRESSION EVALUATION + +When evaluating an arithmetic expression, Libav uses an internal +formula evaluator, implemented through the @file{libavutil/eval.h} +interface. + +An expression may contain unary, binary operators, constants, and +functions. + +Two expressions @var{expr1} and @var{expr2} can be combined to form +another expression "@var{expr1};@var{expr2}". +@var{expr1} and @var{expr2} are evaluated in turn, and the new +expression evaluates to the value of @var{expr2}. + +The following binary operators are available: @code{+}, @code{-}, +@code{*}, @code{/}, @code{^}. + +The following unary operators are available: @code{+}, @code{-}. + +The following functions are available: +@table @option +@item sinh(x) +@item cosh(x) +@item tanh(x) +@item sin(x) +@item cos(x) +@item tan(x) +@item atan(x) +@item asin(x) +@item acos(x) +@item exp(x) +@item log(x) +@item abs(x) +@item squish(x) +@item gauss(x) +@item isinf(x) +Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise. +@item isnan(x) +Return 1.0 if @var{x} is NAN, 0.0 otherwise. + +@item mod(x, y) +@item max(x, y) +@item min(x, y) +@item eq(x, y) +@item gte(x, y) +@item gt(x, y) +@item lte(x, y) +@item lt(x, y) +@item st(var, expr) +Allow to store the value of the expression @var{expr} in an internal +variable. @var{var} specifies the number of the variable where to +store the value, and it is a value ranging from 0 to 9. The function +returns the value stored in the internal variable. + +@item ld(var) +Allow to load the value of the internal variable with number +@var{var}, which was previously stored with st(@var{var}, @var{expr}). +The function returns the loaded value. + +@item while(cond, expr) +Evaluate expression @var{expr} while the expression @var{cond} is +non-zero, and returns the value of the last @var{expr} evaluation, or +NAN if @var{cond} was always false. + +@item ceil(expr) +Round the value of expression @var{expr} upwards to the nearest +integer. For example, "ceil(1.5)" is "2.0". + +@item floor(expr) +Round the value of expression @var{expr} downwards to the nearest +integer. For example, "floor(-1.5)" is "-2.0". + +@item trunc(expr) +Round the value of expression @var{expr} towards zero to the nearest +integer. For example, "trunc(-1.5)" is "-1.0". + +@item sqrt(expr) +Compute the square root of @var{expr}. This is equivalent to +"(@var{expr})^.5". + +@item not(expr) +Return 1.0 if @var{expr} is zero, 0.0 otherwise. +@end table + +Note that: + +@code{*} works like AND + +@code{+} works like OR + +thus +@example +if A then B else C +@end example +is equivalent to +@example +A*B + not(A)*C +@end example + +In your C code, you can extend the list of unary and binary functions, +and define recognized constants, so that they are available for your +expressions. + +The evaluator also recognizes the International System number +postfixes. If 'i' is appended after the postfix, powers of 2 are used +instead of powers of 10. The 'B' postfix multiplies the value for 8, +and can be appended after another postfix or used alone. This allows +using for example 'KB', 'MiB', 'G' and 'B' as postfix. + +Follows the list of available International System postfixes, with +indication of the corresponding powers of 10 and of 2. +@table @option +@item y +-24 / -80 +@item z +-21 / -70 +@item a +-18 / -60 +@item f +-15 / -50 +@item p +-12 / -40 +@item n +-9 / -30 +@item u +-6 / -20 +@item m +-3 / -10 +@item c +-2 +@item d +-1 +@item h +2 +@item k +3 / 10 +@item K +3 / 10 +@item M +6 / 20 +@item G +9 / 30 +@item T +12 / 40 +@item P +15 / 40 +@item E +18 / 50 +@item Z +21 / 60 +@item Y +24 / 70 +@end table + +@c man end diff --git a/doc/examples/Makefile b/doc/examples/Makefile deleted file mode 100644 index 03c7021e8b..0000000000 --- a/doc/examples/Makefile +++ /dev/null @@ -1,42 +0,0 @@ -# use pkg-config for getting CFLAGS and LDLIBS -FFMPEG_LIBS= libavdevice \ - libavformat \ - libavfilter \ - libavcodec \ - libswresample \ - libswscale \ - libavutil \ - -CFLAGS += -Wall -g -CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS) -LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS) - -EXAMPLES= avio_reading \ - decoding_encoding \ - demuxing_decoding \ - filtering_video \ - filtering_audio \ - metadata \ - muxing \ - remuxing \ - resampling_audio \ - scaling_video \ - transcode_aac \ - transcoding \ - -OBJS=$(addsuffix .o,$(EXAMPLES)) - -# the following examples make explicit use of the math library -avcodec: LDLIBS += -lm -muxing: LDLIBS += -lm -resampling_audio: LDLIBS += -lm - -.phony: all clean-test clean - -all: $(OBJS) $(EXAMPLES) - -clean-test: - $(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg - -clean: clean-test - $(RM) $(EXAMPLES) $(OBJS) diff --git a/doc/examples/README b/doc/examples/README deleted file mode 100644 index c1ce619d35..0000000000 --- a/doc/examples/README +++ /dev/null @@ -1,23 +0,0 @@ -FFmpeg examples README ----------------------- - -Both following use cases rely on pkg-config and make, thus make sure -that you have them installed and working on your system. - - -Method 1: build the installed examples in a generic read/write user directory - -Copy to a read/write user directory and just use "make", it will link -to the libraries on your system, assuming the PKG_CONFIG_PATH is -correctly configured. - -Method 2: build the examples in-tree - -Assuming you are in the source FFmpeg checkout directory, you need to build -FFmpeg (no need to make install in any prefix). Then just run "make examples". -This will build the examples using the FFmpeg build system. You can clean those -examples using "make examplesclean" - -If you want to try the dedicated Makefile examples (to emulate the first -method), go into doc/examples and run a command such as -PKG_CONFIG_PATH=pc-uninstalled make. diff --git a/doc/examples/decoding_encoding.c b/doc/examples/avcodec.c similarity index 56% rename from doc/examples/decoding_encoding.c rename to doc/examples/avcodec.c index 0585e518ad..1478881c25 100644 --- a/doc/examples/decoding_encoding.c +++ b/doc/examples/avcodec.c @@ -1,44 +1,47 @@ /* - * Copyright (c) 2001 Fabrice Bellard + * copyright (c) 2001 Fabrice Bellard * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: + * This file is part of Libav. * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. + * Libav is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * libavcodec API use example. * - * @example decoding_encoding.c - * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...), - * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the + * @example avcodec.c + * Note that this library only handles codecs (mpeg, mpeg4, etc...), + * not file formats (avi, vob, etc...). See library 'libavformat' for the * format handling */ -#include +#include +#include +#include -#include -#include -#include -#include -#include -#include -#include +#ifdef HAVE_AV_CONFIG_H +#undef HAVE_AV_CONFIG_H +#endif + +#include "libavcodec/avcodec.h" +#include "libavutil/channel_layout.h" +#include "libavutil/common.h" +#include "libavutil/imgutils.h" +#include "libavutil/mathematics.h" +#include "libavutil/samplefmt.h" #define INBUF_SIZE 4096 #define AUDIO_INBUF_SIZE 20480 @@ -112,20 +115,16 @@ static void audio_encode_example(const char *filename) uint16_t *samples; float t, tincr; - printf("Encode audio file %s\n", filename); + printf("Audio encoding\n"); /* find the MP2 encoder */ codec = avcodec_find_encoder(AV_CODEC_ID_MP2); if (!codec) { - fprintf(stderr, "Codec not found\n"); + fprintf(stderr, "codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate audio codec context\n"); - exit(1); - } /* put sample parameters */ c->bit_rate = 64000; @@ -133,7 +132,7 @@ static void audio_encode_example(const char *filename) /* check that the encoder supports s16 pcm input */ c->sample_fmt = AV_SAMPLE_FMT_S16; if (!check_sample_fmt(codec, c->sample_fmt)) { - fprintf(stderr, "Encoder does not support sample format %s", + fprintf(stderr, "encoder does not support %s", av_get_sample_fmt_name(c->sample_fmt)); exit(1); } @@ -145,20 +144,20 @@ static void audio_encode_example(const char *filename) /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); + fprintf(stderr, "could not open codec\n"); exit(1); } f = fopen(filename, "wb"); if (!f) { - fprintf(stderr, "Could not open %s\n", filename); + fprintf(stderr, "could not open %s\n", filename); exit(1); } /* frame containing input raw audio */ frame = av_frame_alloc(); if (!frame) { - fprintf(stderr, "Could not allocate audio frame\n"); + fprintf(stderr, "could not allocate audio frame\n"); exit(1); } @@ -170,13 +169,9 @@ static void audio_encode_example(const char *filename) * we calculate the size of the samples buffer in bytes */ buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, c->sample_fmt, 0); - if (buffer_size < 0) { - fprintf(stderr, "Could not get sample buffer size\n"); - exit(1); - } samples = av_malloc(buffer_size); if (!samples) { - fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", + fprintf(stderr, "could not allocate %d bytes for samples buffer\n", buffer_size); exit(1); } @@ -184,14 +179,14 @@ static void audio_encode_example(const char *filename) ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (const uint8_t*)samples, buffer_size, 0); if (ret < 0) { - fprintf(stderr, "Could not setup audio frame\n"); + fprintf(stderr, "could not setup audio frame\n"); exit(1); } /* encode a single tone sound */ t = 0; tincr = 2 * M_PI * 440.0 / c->sample_rate; - for (i = 0; i < 200; i++) { + for(i=0;i<200;i++) { av_init_packet(&pkt); pkt.data = NULL; // packet data will be allocated by the encoder pkt.size = 0; @@ -206,7 +201,7 @@ static void audio_encode_example(const char *filename) /* encode the samples */ ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); if (ret < 0) { - fprintf(stderr, "Error encoding audio frame\n"); + fprintf(stderr, "error encoding audio frame\n"); exit(1); } if (got_output) { @@ -214,20 +209,6 @@ static void audio_encode_example(const char *filename) av_free_packet(&pkt); } } - - /* get the delayed frames */ - for (got_output = 1; got_output; i++) { - ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); - if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); - exit(1); - } - - if (got_output) { - fwrite(pkt.data, 1, pkt.size, f); - av_free_packet(&pkt); - } - } fclose(f); av_freep(&samples); @@ -251,30 +232,26 @@ static void audio_decode_example(const char *outfilename, const char *filename) av_init_packet(&avpkt); - printf("Decode audio file %s to %s\n", filename, outfilename); + printf("Audio decoding\n"); /* find the mpeg audio decoder */ codec = avcodec_find_decoder(AV_CODEC_ID_MP2); if (!codec) { - fprintf(stderr, "Codec not found\n"); + fprintf(stderr, "codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate audio codec context\n"); - exit(1); - } /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); + fprintf(stderr, "could not open codec\n"); exit(1); } f = fopen(filename, "rb"); if (!f) { - fprintf(stderr, "Could not open %s\n", filename); + fprintf(stderr, "could not open %s\n", filename); exit(1); } outfile = fopen(outfilename, "wb"); @@ -292,7 +269,7 @@ static void audio_decode_example(const char *outfilename, const char *filename) if (!decoded_frame) { if (!(decoded_frame = av_frame_alloc())) { - fprintf(stderr, "Could not allocate audio frame\n"); + fprintf(stderr, "out of memory\n"); exit(1); } } @@ -307,17 +284,10 @@ static void audio_decode_example(const char *outfilename, const char *filename) int data_size = av_samples_get_buffer_size(NULL, c->channels, decoded_frame->nb_samples, c->sample_fmt, 1); - if (data_size < 0) { - /* This should not occur, checking just for paranoia */ - fprintf(stderr, "Failed to calculate data size\n"); - exit(1); - } fwrite(decoded_frame->data[0], 1, data_size, outfile); } avpkt.size -= len; avpkt.data += len; - avpkt.dts = - avpkt.pts = AV_NOPTS_VALUE; if (avpkt.size < AUDIO_REFILL_THRESH) { /* Refill the input buffer, to avoid trying to decode * incomplete frames. Instead of this, one could also use @@ -343,30 +313,27 @@ static void audio_decode_example(const char *outfilename, const char *filename) /* * Video encoding example */ -static void video_encode_example(const char *filename, int codec_id) +static void video_encode_example(const char *filename) { AVCodec *codec; AVCodecContext *c= NULL; int i, ret, x, y, got_output; FILE *f; - AVFrame *frame; + AVFrame *picture; AVPacket pkt; uint8_t endcode[] = { 0, 0, 1, 0xb7 }; - printf("Encode video file %s\n", filename); + printf("Video encoding\n"); /* find the mpeg1 video encoder */ - codec = avcodec_find_encoder(codec_id); + codec = avcodec_find_encoder(AV_CODEC_ID_MPEG1VIDEO); if (!codec) { - fprintf(stderr, "Codec not found\n"); + fprintf(stderr, "codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate video codec context\n"); - exit(1); - } + picture = av_frame_alloc(); /* put sample parameters */ c->bit_rate = 400000; @@ -374,52 +341,35 @@ static void video_encode_example(const char *filename, int codec_id) c->width = 352; c->height = 288; /* frames per second */ - c->time_base = (AVRational){1,25}; - /* emit one intra frame every ten frames - * check frame pict_type before passing frame - * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I - * then gop_size is ignored and the output of encoder - * will always be I frame irrespective to gop_size - */ - c->gop_size = 10; - c->max_b_frames = 1; + c->time_base= (AVRational){1,25}; + c->gop_size = 10; /* emit one intra frame every ten frames */ + c->max_b_frames=1; c->pix_fmt = AV_PIX_FMT_YUV420P; - if (codec_id == AV_CODEC_ID_H264) - av_opt_set(c->priv_data, "preset", "slow", 0); - /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); + fprintf(stderr, "could not open codec\n"); exit(1); } f = fopen(filename, "wb"); if (!f) { - fprintf(stderr, "Could not open %s\n", filename); + fprintf(stderr, "could not open %s\n", filename); exit(1); } - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate video frame\n"); - exit(1); - } - frame->format = c->pix_fmt; - frame->width = c->width; - frame->height = c->height; - - /* the image can be allocated by any means and av_image_alloc() is - * just the most convenient way if av_malloc() is to be used */ - ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height, + ret = av_image_alloc(picture->data, picture->linesize, c->width, c->height, c->pix_fmt, 32); if (ret < 0) { - fprintf(stderr, "Could not allocate raw picture buffer\n"); + fprintf(stderr, "could not alloc raw picture buffer\n"); exit(1); } + picture->format = c->pix_fmt; + picture->width = c->width; + picture->height = c->height; /* encode 1 second of video */ - for (i = 0; i < 25; i++) { + for(i=0;i<25;i++) { av_init_packet(&pkt); pkt.data = NULL; // packet data will be allocated by the encoder pkt.size = 0; @@ -427,31 +377,31 @@ static void video_encode_example(const char *filename, int codec_id) fflush(stdout); /* prepare a dummy image */ /* Y */ - for (y = 0; y < c->height; y++) { - for (x = 0; x < c->width; x++) { - frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3; + for(y=0;yheight;y++) { + for(x=0;xwidth;x++) { + picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3; } } /* Cb and Cr */ - for (y = 0; y < c->height/2; y++) { - for (x = 0; x < c->width/2; x++) { - frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2; - frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5; + for(y=0;yheight/2;y++) { + for(x=0;xwidth/2;x++) { + picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2; + picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5; } } - frame->pts = i; + picture->pts = i; /* encode the image */ - ret = avcodec_encode_video2(c, &pkt, frame, &got_output); + ret = avcodec_encode_video2(c, &pkt, picture, &got_output); if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); + fprintf(stderr, "error encoding frame\n"); exit(1); } if (got_output) { - printf("Write frame %3d (size=%5d)\n", i, pkt.size); + printf("encoding frame %3d (size=%5d)\n", i, pkt.size); fwrite(pkt.data, 1, pkt.size, f); av_free_packet(&pkt); } @@ -463,12 +413,12 @@ static void video_encode_example(const char *filename, int codec_id) ret = avcodec_encode_video2(c, &pkt, NULL, &got_output); if (ret < 0) { - fprintf(stderr, "Error encoding frame\n"); + fprintf(stderr, "error encoding frame\n"); exit(1); } if (got_output) { - printf("Write frame %3d (size=%5d)\n", i, pkt.size); + printf("encoding frame %3d (size=%5d)\n", i, pkt.size); fwrite(pkt.data, 1, pkt.size, f); av_free_packet(&pkt); } @@ -480,8 +430,8 @@ static void video_encode_example(const char *filename, int codec_id) avcodec_close(c); av_free(c); - av_freep(&frame->data[0]); - av_frame_free(&frame); + av_freep(&picture->data[0]); + av_frame_free(&picture); printf("\n"); } @@ -495,49 +445,22 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize, FILE *f; int i; - f = fopen(filename,"w"); - fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255); - for (i = 0; i < ysize; i++) - fwrite(buf + i * wrap, 1, xsize, f); + f=fopen(filename,"w"); + fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255); + for(i=0;idata[0], frame->linesize[0], - avctx->width, avctx->height, buf); - (*frame_count)++; - } - if (pkt->data) { - pkt->size -= len; - pkt->data += len; - } - return 0; -} - static void video_decode_example(const char *outfilename, const char *filename) { AVCodec *codec; AVCodecContext *c= NULL; - int frame_count; + int frame, got_picture, len; FILE *f; - AVFrame *frame; + AVFrame *picture; uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE]; + char buf[1024]; AVPacket avpkt; av_init_packet(&avpkt); @@ -545,20 +468,17 @@ static void video_decode_example(const char *outfilename, const char *filename) /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */ memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE); - printf("Decode video file %s to %s\n", filename, outfilename); + printf("Video decoding\n"); /* find the mpeg1 video decoder */ codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO); if (!codec) { - fprintf(stderr, "Codec not found\n"); + fprintf(stderr, "codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); - if (!c) { - fprintf(stderr, "Could not allocate video codec context\n"); - exit(1); - } + picture = av_frame_alloc(); if(codec->capabilities&CODEC_CAP_TRUNCATED) c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */ @@ -569,24 +489,20 @@ static void video_decode_example(const char *outfilename, const char *filename) /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "Could not open codec\n"); + fprintf(stderr, "could not open codec\n"); exit(1); } + /* the codec gives us the frame size, in samples */ + f = fopen(filename, "rb"); if (!f) { - fprintf(stderr, "Could not open %s\n", filename); + fprintf(stderr, "could not open %s\n", filename); exit(1); } - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate video frame\n"); - exit(1); - } - - frame_count = 0; - for (;;) { + frame = 0; + for(;;) { avpkt.size = fread(inbuf, 1, INBUF_SIZE, f); if (avpkt.size == 0) break; @@ -607,9 +523,26 @@ static void video_decode_example(const char *outfilename, const char *filename) /* here, we use a stream based decoder (mpeg1video), so we feed decoder and see if it could decode a frame */ avpkt.data = inbuf; - while (avpkt.size > 0) - if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0) + while (avpkt.size > 0) { + len = avcodec_decode_video2(c, picture, &got_picture, &avpkt); + if (len < 0) { + fprintf(stderr, "Error while decoding frame %d\n", frame); exit(1); + } + if (got_picture) { + printf("saving frame %3d\n", frame); + fflush(stdout); + + /* the picture is allocated by the decoder. no need to + free it */ + snprintf(buf, sizeof(buf), outfilename, frame); + pgm_save(picture->data[0], picture->linesize[0], + c->width, c->height, buf); + frame++; + } + avpkt.size -= len; + avpkt.data += len; + } } /* some codecs, such as MPEG, transmit the I and P frame with a @@ -617,48 +550,46 @@ static void video_decode_example(const char *outfilename, const char *filename) chance to get the last frame of the video */ avpkt.data = NULL; avpkt.size = 0; - decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1); + len = avcodec_decode_video2(c, picture, &got_picture, &avpkt); + if (got_picture) { + printf("saving last frame %3d\n", frame); + fflush(stdout); + + /* the picture is allocated by the decoder. no need to + free it */ + snprintf(buf, sizeof(buf), outfilename, frame); + pgm_save(picture->data[0], picture->linesize[0], + c->width, c->height, buf); + frame++; + } fclose(f); avcodec_close(c); av_free(c); - av_frame_free(&frame); + av_frame_free(&picture); printf("\n"); } int main(int argc, char **argv) { - const char *output_type; + const char *filename; /* register all the codecs */ avcodec_register_all(); - if (argc < 2) { - printf("usage: %s output_type\n" - "API example program to decode/encode a media stream with libavcodec.\n" - "This program generates a synthetic stream and encodes it to a file\n" - "named test.h264, test.mp2 or test.mpg depending on output_type.\n" - "The encoded stream is then decoded and written to a raw data output.\n" - "output_type must be choosen between 'h264', 'mp2', 'mpg'.\n", - argv[0]); - return 1; - } - output_type = argv[1]; + if (argc <= 1) { + audio_encode_example("/tmp/test.mp2"); + audio_decode_example("/tmp/test.sw", "/tmp/test.mp2"); - if (!strcmp(output_type, "h264")) { - video_encode_example("test.h264", AV_CODEC_ID_H264); - } else if (!strcmp(output_type, "mp2")) { - audio_encode_example("test.mp2"); - audio_decode_example("test.sw", "test.mp2"); - } else if (!strcmp(output_type, "mpg")) { - video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO); - video_decode_example("test%02d.pgm", "test.mpg"); + video_encode_example("/tmp/test.mpg"); + filename = "/tmp/test.mpg"; } else { - fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n", - output_type); - return 1; + filename = argv[1]; } + // audio_decode_example("/tmp/test.sw", filename); + video_decode_example("/tmp/test%d.pgm", filename); + return 0; } diff --git a/doc/examples/avio_reading.c b/doc/examples/avio_reading.c deleted file mode 100644 index 02474e907a..0000000000 --- a/doc/examples/avio_reading.c +++ /dev/null @@ -1,134 +0,0 @@ -/* - * Copyright (c) 2014 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * libavformat AVIOContext API example. - * - * Make libavformat demuxer access media content through a custom - * AVIOContext read callback. - * @example avio_reading.c - */ - -#include -#include -#include -#include - -struct buffer_data { - uint8_t *ptr; - size_t size; ///< size left in the buffer -}; - -static int read_packet(void *opaque, uint8_t *buf, int buf_size) -{ - struct buffer_data *bd = (struct buffer_data *)opaque; - buf_size = FFMIN(buf_size, bd->size); - - printf("ptr:%p size:%zu\n", bd->ptr, bd->size); - - /* copy internal buffer data to buf */ - memcpy(buf, bd->ptr, buf_size); - bd->ptr += buf_size; - bd->size -= buf_size; - - return buf_size; -} - -int main(int argc, char *argv[]) -{ - AVFormatContext *fmt_ctx = NULL; - AVIOContext *avio_ctx = NULL; - uint8_t *buffer = NULL, *avio_ctx_buffer = NULL; - size_t buffer_size, avio_ctx_buffer_size = 4096; - char *input_filename = NULL; - int ret = 0; - struct buffer_data bd = { 0 }; - - if (argc != 2) { - fprintf(stderr, "usage: %s input_file\n" - "API example program to show how to read from a custom buffer " - "accessed through AVIOContext.\n", argv[0]); - return 1; - } - input_filename = argv[1]; - - /* register codecs and formats and other lavf/lavc components*/ - av_register_all(); - - /* slurp file content into buffer */ - ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL); - if (ret < 0) - goto end; - - /* fill opaque structure used by the AVIOContext read callback */ - bd.ptr = buffer; - bd.size = buffer_size; - - if (!(fmt_ctx = avformat_alloc_context())) { - ret = AVERROR(ENOMEM); - goto end; - } - - avio_ctx_buffer = av_malloc(avio_ctx_buffer_size); - if (!avio_ctx_buffer) { - ret = AVERROR(ENOMEM); - goto end; - } - avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size, - 0, &bd, &read_packet, NULL, NULL); - if (!avio_ctx) { - ret = AVERROR(ENOMEM); - goto end; - } - fmt_ctx->pb = avio_ctx; - - ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL); - if (ret < 0) { - fprintf(stderr, "Could not open input\n"); - goto end; - } - - ret = avformat_find_stream_info(fmt_ctx, NULL); - if (ret < 0) { - fprintf(stderr, "Could not find stream information\n"); - goto end; - } - - av_dump_format(fmt_ctx, 0, input_filename, 0); - -end: - avformat_close_input(&fmt_ctx); - /* note: the internal buffer could have changed, and be != avio_ctx_buffer */ - if (avio_ctx) { - av_freep(&avio_ctx->buffer); - av_freep(&avio_ctx); - } - av_file_unmap(buffer, buffer_size); - - if (ret < 0) { - fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); - return 1; - } - - return 0; -} diff --git a/doc/examples/demuxing_decoding.c b/doc/examples/demuxing_decoding.c deleted file mode 100644 index 2ce4018c79..0000000000 --- a/doc/examples/demuxing_decoding.c +++ /dev/null @@ -1,386 +0,0 @@ -/* - * Copyright (c) 2012 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * Demuxing and decoding example. - * - * Show how to use the libavformat and libavcodec API to demux and - * decode audio and video data. - * @example demuxing_decoding.c - */ - -#include -#include -#include -#include - -static AVFormatContext *fmt_ctx = NULL; -static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx; -static AVStream *video_stream = NULL, *audio_stream = NULL; -static const char *src_filename = NULL; -static const char *video_dst_filename = NULL; -static const char *audio_dst_filename = NULL; -static FILE *video_dst_file = NULL; -static FILE *audio_dst_file = NULL; - -static uint8_t *video_dst_data[4] = {NULL}; -static int video_dst_linesize[4]; -static int video_dst_bufsize; - -static int video_stream_idx = -1, audio_stream_idx = -1; -static AVFrame *frame = NULL; -static AVPacket pkt; -static int video_frame_count = 0; -static int audio_frame_count = 0; - -/* The different ways of decoding and managing data memory. You are not - * supposed to support all the modes in your application but pick the one most - * appropriate to your needs. Look for the use of api_mode in this example to - * see what are the differences of API usage between them */ -enum { - API_MODE_OLD = 0, /* old method, deprecated */ - API_MODE_NEW_API_REF_COUNT = 1, /* new method, using the frame reference counting */ - API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */ -}; - -static int api_mode = API_MODE_OLD; - -static int decode_packet(int *got_frame, int cached) -{ - int ret = 0; - int decoded = pkt.size; - - *got_frame = 0; - - if (pkt.stream_index == video_stream_idx) { - /* decode video frame */ - ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); - if (ret < 0) { - fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); - return ret; - } - - if (*got_frame) { - printf("video_frame%s n:%d coded_n:%d pts:%s\n", - cached ? "(cached)" : "", - video_frame_count++, frame->coded_picture_number, - av_ts2timestr(frame->pts, &video_dec_ctx->time_base)); - - /* copy decoded frame to destination buffer: - * this is required since rawvideo expects non aligned data */ - av_image_copy(video_dst_data, video_dst_linesize, - (const uint8_t **)(frame->data), frame->linesize, - video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height); - - /* write to rawvideo file */ - fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file); - } - } else if (pkt.stream_index == audio_stream_idx) { - /* decode audio frame */ - ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt); - if (ret < 0) { - fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret)); - return ret; - } - /* Some audio decoders decode only part of the packet, and have to be - * called again with the remainder of the packet data. - * Sample: fate-suite/lossless-audio/luckynight-partial.shn - * Also, some decoders might over-read the packet. */ - decoded = FFMIN(ret, pkt.size); - - if (*got_frame) { - size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format); - printf("audio_frame%s n:%d nb_samples:%d pts:%s\n", - cached ? "(cached)" : "", - audio_frame_count++, frame->nb_samples, - av_ts2timestr(frame->pts, &audio_dec_ctx->time_base)); - - /* Write the raw audio data samples of the first plane. This works - * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However, - * most audio decoders output planar audio, which uses a separate - * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P). - * In other words, this code will write only the first audio channel - * in these cases. - * You should use libswresample or libavfilter to convert the frame - * to packed data. */ - fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file); - } - } - - /* If we use the new API with reference counting, we own the data and need - * to de-reference it when we don't use it anymore */ - if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT) - av_frame_unref(frame); - - return decoded; -} - -static int open_codec_context(int *stream_idx, - AVFormatContext *fmt_ctx, enum AVMediaType type) -{ - int ret; - AVStream *st; - AVCodecContext *dec_ctx = NULL; - AVCodec *dec = NULL; - AVDictionary *opts = NULL; - - ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); - if (ret < 0) { - fprintf(stderr, "Could not find %s stream in input file '%s'\n", - av_get_media_type_string(type), src_filename); - return ret; - } else { - *stream_idx = ret; - st = fmt_ctx->streams[*stream_idx]; - - /* find decoder for the stream */ - dec_ctx = st->codec; - dec = avcodec_find_decoder(dec_ctx->codec_id); - if (!dec) { - fprintf(stderr, "Failed to find %s codec\n", - av_get_media_type_string(type)); - return AVERROR(EINVAL); - } - - /* Init the decoders, with or without reference counting */ - if (api_mode == API_MODE_NEW_API_REF_COUNT) - av_dict_set(&opts, "refcounted_frames", "1", 0); - if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) { - fprintf(stderr, "Failed to open %s codec\n", - av_get_media_type_string(type)); - return ret; - } - } - - return 0; -} - -static int get_format_from_sample_fmt(const char **fmt, - enum AVSampleFormat sample_fmt) -{ - int i; - struct sample_fmt_entry { - enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; - } sample_fmt_entries[] = { - { AV_SAMPLE_FMT_U8, "u8", "u8" }, - { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, - { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, - { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, - { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, - }; - *fmt = NULL; - - for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { - struct sample_fmt_entry *entry = &sample_fmt_entries[i]; - if (sample_fmt == entry->sample_fmt) { - *fmt = AV_NE(entry->fmt_be, entry->fmt_le); - return 0; - } - } - - fprintf(stderr, - "sample format %s is not supported as output format\n", - av_get_sample_fmt_name(sample_fmt)); - return -1; -} - -int main (int argc, char **argv) -{ - int ret = 0, got_frame; - - if (argc != 4 && argc != 5) { - fprintf(stderr, "usage: %s [-refcount=] " - "input_file video_output_file audio_output_file\n" - "API example program to show how to read frames from an input file.\n" - "This program reads frames from a file, decodes them, and writes decoded\n" - "video frames to a rawvideo file named video_output_file, and decoded\n" - "audio frames to a rawaudio file named audio_output_file.\n\n" - "If the -refcount option is specified, the program use the\n" - "reference counting frame system which allows keeping a copy of\n" - "the data for longer than one decode call. If unset, it's using\n" - "the classic old method.\n" - "\n", argv[0]); - exit(1); - } - if (argc == 5) { - const char *mode = argv[1] + strlen("-refcount="); - if (!strcmp(mode, "old")) api_mode = API_MODE_OLD; - else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT; - else if (!strcmp(mode, "new_refcount")) api_mode = API_MODE_NEW_API_REF_COUNT; - else { - fprintf(stderr, "unknow mode '%s'\n", mode); - exit(1); - } - argv++; - } - src_filename = argv[1]; - video_dst_filename = argv[2]; - audio_dst_filename = argv[3]; - - /* register all formats and codecs */ - av_register_all(); - - /* open input file, and allocate format context */ - if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) { - fprintf(stderr, "Could not open source file %s\n", src_filename); - exit(1); - } - - /* retrieve stream information */ - if (avformat_find_stream_info(fmt_ctx, NULL) < 0) { - fprintf(stderr, "Could not find stream information\n"); - exit(1); - } - - if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) { - video_stream = fmt_ctx->streams[video_stream_idx]; - video_dec_ctx = video_stream->codec; - - video_dst_file = fopen(video_dst_filename, "wb"); - if (!video_dst_file) { - fprintf(stderr, "Could not open destination file %s\n", video_dst_filename); - ret = 1; - goto end; - } - - /* allocate image where the decoded image will be put */ - ret = av_image_alloc(video_dst_data, video_dst_linesize, - video_dec_ctx->width, video_dec_ctx->height, - video_dec_ctx->pix_fmt, 1); - if (ret < 0) { - fprintf(stderr, "Could not allocate raw video buffer\n"); - goto end; - } - video_dst_bufsize = ret; - } - - if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) { - audio_stream = fmt_ctx->streams[audio_stream_idx]; - audio_dec_ctx = audio_stream->codec; - audio_dst_file = fopen(audio_dst_filename, "wb"); - if (!audio_dst_file) { - fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename); - ret = 1; - goto end; - } - } - - /* dump input information to stderr */ - av_dump_format(fmt_ctx, 0, src_filename, 0); - - if (!audio_stream && !video_stream) { - fprintf(stderr, "Could not find audio or video stream in the input, aborting\n"); - ret = 1; - goto end; - } - - /* When using the new API, you need to use the libavutil/frame.h API, while - * the classic frame management is available in libavcodec */ - if (api_mode == API_MODE_OLD) - frame = avcodec_alloc_frame(); - else - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate frame\n"); - ret = AVERROR(ENOMEM); - goto end; - } - - /* initialize packet, set data to NULL, let the demuxer fill it */ - av_init_packet(&pkt); - pkt.data = NULL; - pkt.size = 0; - - if (video_stream) - printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename); - if (audio_stream) - printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename); - - /* read frames from the file */ - while (av_read_frame(fmt_ctx, &pkt) >= 0) { - AVPacket orig_pkt = pkt; - do { - ret = decode_packet(&got_frame, 0); - if (ret < 0) - break; - pkt.data += ret; - pkt.size -= ret; - } while (pkt.size > 0); - av_free_packet(&orig_pkt); - } - - /* flush cached frames */ - pkt.data = NULL; - pkt.size = 0; - do { - decode_packet(&got_frame, 1); - } while (got_frame); - - printf("Demuxing succeeded.\n"); - - if (video_stream) { - printf("Play the output video file with the command:\n" - "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", - av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height, - video_dst_filename); - } - - if (audio_stream) { - enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt; - int n_channels = audio_dec_ctx->channels; - const char *fmt; - - if (av_sample_fmt_is_planar(sfmt)) { - const char *packed = av_get_sample_fmt_name(sfmt); - printf("Warning: the sample format the decoder produced is planar " - "(%s). This example will output the first channel only.\n", - packed ? packed : "?"); - sfmt = av_get_packed_sample_fmt(sfmt); - n_channels = 1; - } - - if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0) - goto end; - - printf("Play the output audio file with the command:\n" - "ffplay -f %s -ac %d -ar %d %s\n", - fmt, n_channels, audio_dec_ctx->sample_rate, - audio_dst_filename); - } - -end: - avcodec_close(video_dec_ctx); - avcodec_close(audio_dec_ctx); - avformat_close_input(&fmt_ctx); - if (video_dst_file) - fclose(video_dst_file); - if (audio_dst_file) - fclose(audio_dst_file); - if (api_mode == API_MODE_OLD) - avcodec_free_frame(&frame); - else - av_frame_free(&frame); - av_free(video_dst_data[0]); - - return ret < 0; -} diff --git a/doc/examples/filter_audio.c b/doc/examples/filter_audio.c index 8451f9cba2..71f985ee13 100644 --- a/doc/examples/filter_audio.c +++ b/doc/examples/filter_audio.c @@ -1,20 +1,20 @@ /* * copyright (c) 2013 Andrew Kelley * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ diff --git a/doc/examples/filtering_audio.c b/doc/examples/filtering_audio.c deleted file mode 100644 index 46595fb3b8..0000000000 --- a/doc/examples/filtering_audio.c +++ /dev/null @@ -1,280 +0,0 @@ -/* - * Copyright (c) 2010 Nicolas George - * Copyright (c) 2011 Stefano Sabatini - * Copyright (c) 2012 ClĂ©ment BÅ“sch - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * API example for audio decoding and filtering - * @example filtering_audio.c - */ - -#include - -#include -#include -#include -#include -#include -#include -#include - -static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono"; -static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -"; - -static AVFormatContext *fmt_ctx; -static AVCodecContext *dec_ctx; -AVFilterContext *buffersink_ctx; -AVFilterContext *buffersrc_ctx; -AVFilterGraph *filter_graph; -static int audio_stream_index = -1; - -static int open_input_file(const char *filename) -{ - int ret; - AVCodec *dec; - - if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); - return ret; - } - - if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); - return ret; - } - - /* select the audio stream */ - ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n"); - return ret; - } - audio_stream_index = ret; - dec_ctx = fmt_ctx->streams[audio_stream_index]->codec; - av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0); - - /* init the audio decoder */ - if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n"); - return ret; - } - - return 0; -} - -static int init_filters(const char *filters_descr) -{ - char args[512]; - int ret = 0; - AVFilter *abuffersrc = avfilter_get_by_name("abuffer"); - AVFilter *abuffersink = avfilter_get_by_name("abuffersink"); - AVFilterInOut *outputs = avfilter_inout_alloc(); - AVFilterInOut *inputs = avfilter_inout_alloc(); - static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 }; - static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 }; - static const int out_sample_rates[] = { 8000, -1 }; - const AVFilterLink *outlink; - AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base; - - filter_graph = avfilter_graph_alloc(); - if (!outputs || !inputs || !filter_graph) { - ret = AVERROR(ENOMEM); - goto end; - } - - /* buffer audio source: the decoded frames from the decoder will be inserted here. */ - if (!dec_ctx->channel_layout) - dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels); - snprintf(args, sizeof(args), - "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64, - time_base.num, time_base.den, dec_ctx->sample_rate, - av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout); - ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in", - args, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n"); - goto end; - } - - /* buffer audio sink: to terminate the filter chain. */ - ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out", - NULL, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n"); - goto end; - } - - ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1, - AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n"); - goto end; - } - - ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1, - AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n"); - goto end; - } - - ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1, - AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n"); - goto end; - } - - /* Endpoints for the filter graph. */ - outputs->name = av_strdup("in"); - outputs->filter_ctx = buffersrc_ctx; - outputs->pad_idx = 0; - outputs->next = NULL; - - inputs->name = av_strdup("out"); - inputs->filter_ctx = buffersink_ctx; - inputs->pad_idx = 0; - inputs->next = NULL; - - if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr, - &inputs, &outputs, NULL)) < 0) - goto end; - - if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) - goto end; - - /* Print summary of the sink buffer - * Note: args buffer is reused to store channel layout string */ - outlink = buffersink_ctx->inputs[0]; - av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout); - av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n", - (int)outlink->sample_rate, - (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"), - args); - -end: - avfilter_inout_free(&inputs); - avfilter_inout_free(&outputs); - - return ret; -} - -static void print_frame(const AVFrame *frame) -{ - const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame)); - const uint16_t *p = (uint16_t*)frame->data[0]; - const uint16_t *p_end = p + n; - - while (p < p_end) { - fputc(*p & 0xff, stdout); - fputc(*p>>8 & 0xff, stdout); - p++; - } - fflush(stdout); -} - -int main(int argc, char **argv) -{ - int ret; - AVPacket packet0, packet; - AVFrame *frame = av_frame_alloc(); - AVFrame *filt_frame = av_frame_alloc(); - int got_frame; - - if (!frame || !filt_frame) { - perror("Could not allocate frame"); - exit(1); - } - if (argc != 2) { - fprintf(stderr, "Usage: %s file | %s\n", argv[0], player); - exit(1); - } - - av_register_all(); - avfilter_register_all(); - - if ((ret = open_input_file(argv[1])) < 0) - goto end; - if ((ret = init_filters(filter_descr)) < 0) - goto end; - - /* read all packets */ - packet0.data = NULL; - packet.data = NULL; - while (1) { - if (!packet0.data) { - if ((ret = av_read_frame(fmt_ctx, &packet)) < 0) - break; - packet0 = packet; - } - - if (packet.stream_index == audio_stream_index) { - got_frame = 0; - ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n"); - continue; - } - packet.size -= ret; - packet.data += ret; - - if (got_frame) { - /* push the audio data from decoded frame into the filtergraph */ - if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) { - av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n"); - break; - } - - /* pull filtered audio from the filtergraph */ - while (1) { - ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); - if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) - break; - if (ret < 0) - goto end; - print_frame(filt_frame); - av_frame_unref(filt_frame); - } - } - - if (packet.size <= 0) - av_free_packet(&packet0); - } else { - /* discard non-wanted packets */ - av_free_packet(&packet0); - } - } -end: - avfilter_graph_free(&filter_graph); - avcodec_close(dec_ctx); - avformat_close_input(&fmt_ctx); - av_frame_free(&frame); - av_frame_free(&filt_frame); - - if (ret < 0 && ret != AVERROR_EOF) { - fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); - exit(1); - } - - exit(0); -} diff --git a/doc/examples/filtering_video.c b/doc/examples/filtering_video.c deleted file mode 100644 index 8d595735b3..0000000000 --- a/doc/examples/filtering_video.c +++ /dev/null @@ -1,261 +0,0 @@ -/* - * Copyright (c) 2010 Nicolas George - * Copyright (c) 2011 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * API example for decoding and filtering - * @example filtering_video.c - */ - -#define _XOPEN_SOURCE 600 /* for usleep */ -#include - -#include -#include -#include -#include -#include -#include -#include - -const char *filter_descr = "scale=78:24"; - -static AVFormatContext *fmt_ctx; -static AVCodecContext *dec_ctx; -AVFilterContext *buffersink_ctx; -AVFilterContext *buffersrc_ctx; -AVFilterGraph *filter_graph; -static int video_stream_index = -1; -static int64_t last_pts = AV_NOPTS_VALUE; - -static int open_input_file(const char *filename) -{ - int ret; - AVCodec *dec; - - if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); - return ret; - } - - if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); - return ret; - } - - /* select the video stream */ - ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n"); - return ret; - } - video_stream_index = ret; - dec_ctx = fmt_ctx->streams[video_stream_index]->codec; - av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0); - - /* init the video decoder */ - if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n"); - return ret; - } - - return 0; -} - -static int init_filters(const char *filters_descr) -{ - char args[512]; - int ret = 0; - AVFilter *buffersrc = avfilter_get_by_name("buffer"); - AVFilter *buffersink = avfilter_get_by_name("buffersink"); - AVFilterInOut *outputs = avfilter_inout_alloc(); - AVFilterInOut *inputs = avfilter_inout_alloc(); - enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE }; - - filter_graph = avfilter_graph_alloc(); - if (!outputs || !inputs || !filter_graph) { - ret = AVERROR(ENOMEM); - goto end; - } - - /* buffer video source: the decoded frames from the decoder will be inserted here. */ - snprintf(args, sizeof(args), - "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d", - dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt, - dec_ctx->time_base.num, dec_ctx->time_base.den, - dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den); - - ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", - args, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n"); - goto end; - } - - /* buffer video sink: to terminate the filter chain. */ - ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", - NULL, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n"); - goto end; - } - - ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts, - AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n"); - goto end; - } - - /* Endpoints for the filter graph. */ - outputs->name = av_strdup("in"); - outputs->filter_ctx = buffersrc_ctx; - outputs->pad_idx = 0; - outputs->next = NULL; - - inputs->name = av_strdup("out"); - inputs->filter_ctx = buffersink_ctx; - inputs->pad_idx = 0; - inputs->next = NULL; - - if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr, - &inputs, &outputs, NULL)) < 0) - goto end; - - if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) - goto end; - -end: - avfilter_inout_free(&inputs); - avfilter_inout_free(&outputs); - - return ret; -} - -static void display_frame(const AVFrame *frame, AVRational time_base) -{ - int x, y; - uint8_t *p0, *p; - int64_t delay; - - if (frame->pts != AV_NOPTS_VALUE) { - if (last_pts != AV_NOPTS_VALUE) { - /* sleep roughly the right amount of time; - * usleep is in microseconds, just like AV_TIME_BASE. */ - delay = av_rescale_q(frame->pts - last_pts, - time_base, AV_TIME_BASE_Q); - if (delay > 0 && delay < 1000000) - usleep(delay); - } - last_pts = frame->pts; - } - - /* Trivial ASCII grayscale display. */ - p0 = frame->data[0]; - puts("\033c"); - for (y = 0; y < frame->height; y++) { - p = p0; - for (x = 0; x < frame->width; x++) - putchar(" .-+#"[*(p++) / 52]); - putchar('\n'); - p0 += frame->linesize[0]; - } - fflush(stdout); -} - -int main(int argc, char **argv) -{ - int ret; - AVPacket packet; - AVFrame *frame = av_frame_alloc(); - AVFrame *filt_frame = av_frame_alloc(); - int got_frame; - - if (!frame || !filt_frame) { - perror("Could not allocate frame"); - exit(1); - } - if (argc != 2) { - fprintf(stderr, "Usage: %s file\n", argv[0]); - exit(1); - } - - av_register_all(); - avfilter_register_all(); - - if ((ret = open_input_file(argv[1])) < 0) - goto end; - if ((ret = init_filters(filter_descr)) < 0) - goto end; - - /* read all packets */ - while (1) { - if ((ret = av_read_frame(fmt_ctx, &packet)) < 0) - break; - - if (packet.stream_index == video_stream_index) { - got_frame = 0; - ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Error decoding video\n"); - break; - } - - if (got_frame) { - frame->pts = av_frame_get_best_effort_timestamp(frame); - - /* push the decoded frame into the filtergraph */ - if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) { - av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n"); - break; - } - - /* pull filtered frames from the filtergraph */ - while (1) { - ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); - if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) - break; - if (ret < 0) - goto end; - display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base); - av_frame_unref(filt_frame); - } - av_frame_unref(frame); - } - } - av_free_packet(&packet); - } -end: - avfilter_graph_free(&filter_graph); - avcodec_close(dec_ctx); - avformat_close_input(&fmt_ctx); - av_frame_free(&frame); - av_frame_free(&filt_frame); - - if (ret < 0 && ret != AVERROR_EOF) { - fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); - exit(1); - } - - exit(0); -} diff --git a/doc/examples/metadata.c b/doc/examples/metadata.c index f73c267369..f4c6eee9c3 100644 --- a/doc/examples/metadata.c +++ b/doc/examples/metadata.c @@ -22,8 +22,8 @@ /** * @file - * Shows how the metadata API can be used in application programs. * @example metadata.c + * Shows how the metadata API can be used in application programs. */ #include @@ -51,6 +51,6 @@ int main (int argc, char **argv) while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX))) printf("%s=%s\n", tag->key, tag->value); - avformat_close_input(&fmt_ctx); + avformat_free_context(fmt_ctx); return 0; } diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c deleted file mode 100644 index ad8e027148..0000000000 --- a/doc/examples/muxing.c +++ /dev/null @@ -1,606 +0,0 @@ -/* - * Copyright (c) 2003 Fabrice Bellard - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * libavformat API example. - * - * Output a media file in any supported libavformat format. The default - * codecs are used. - * @example muxing.c - */ - -#include -#include -#include -#include - -#include -#include -#include -#include -#include -#include - -static int audio_is_eof, video_is_eof; - -#define STREAM_DURATION 10.0 -#define STREAM_FRAME_RATE 25 /* 25 images/s */ -#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */ - -static int sws_flags = SWS_BICUBIC; - -static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt) -{ - AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base; - - printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n", - av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base), - av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base), - av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base), - pkt->stream_index); -} - -static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt) -{ - /* rescale output packet timestamp values from codec to stream timebase */ - pkt->pts = av_rescale_q_rnd(pkt->pts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - pkt->dts = av_rescale_q_rnd(pkt->dts, *time_base, st->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - pkt->duration = av_rescale_q(pkt->duration, *time_base, st->time_base); - pkt->stream_index = st->index; - - /* Write the compressed frame to the media file. */ - log_packet(fmt_ctx, pkt); - return av_interleaved_write_frame(fmt_ctx, pkt); -} - -/* Add an output stream. */ -static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec, - enum AVCodecID codec_id) -{ - AVCodecContext *c; - AVStream *st; - - /* find the encoder */ - *codec = avcodec_find_encoder(codec_id); - if (!(*codec)) { - fprintf(stderr, "Could not find encoder for '%s'\n", - avcodec_get_name(codec_id)); - exit(1); - } - - st = avformat_new_stream(oc, *codec); - if (!st) { - fprintf(stderr, "Could not allocate stream\n"); - exit(1); - } - st->id = oc->nb_streams-1; - c = st->codec; - - switch ((*codec)->type) { - case AVMEDIA_TYPE_AUDIO: - c->sample_fmt = (*codec)->sample_fmts ? - (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP; - c->bit_rate = 64000; - c->sample_rate = 44100; - c->channels = 2; - break; - - case AVMEDIA_TYPE_VIDEO: - c->codec_id = codec_id; - - c->bit_rate = 400000; - /* Resolution must be a multiple of two. */ - c->width = 352; - c->height = 288; - /* timebase: This is the fundamental unit of time (in seconds) in terms - * of which frame timestamps are represented. For fixed-fps content, - * timebase should be 1/framerate and timestamp increments should be - * identical to 1. */ - c->time_base.den = STREAM_FRAME_RATE; - c->time_base.num = 1; - c->gop_size = 12; /* emit one intra frame every twelve frames at most */ - c->pix_fmt = STREAM_PIX_FMT; - if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) { - /* just for testing, we also add B frames */ - c->max_b_frames = 2; - } - if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) { - /* Needed to avoid using macroblocks in which some coeffs overflow. - * This does not happen with normal video, it just happens here as - * the motion of the chroma plane does not match the luma plane. */ - c->mb_decision = 2; - } - break; - - default: - break; - } - - /* Some formats want stream headers to be separate. */ - if (oc->oformat->flags & AVFMT_GLOBALHEADER) - c->flags |= CODEC_FLAG_GLOBAL_HEADER; - - return st; -} - -/**************************************************************/ -/* audio output */ - -static float t, tincr, tincr2; - -AVFrame *audio_frame; -static uint8_t **src_samples_data; -static int src_samples_linesize; -static int src_nb_samples; - -static int max_dst_nb_samples; -uint8_t **dst_samples_data; -int dst_samples_linesize; -int dst_samples_size; -int samples_count; - -struct SwrContext *swr_ctx = NULL; - -static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st) -{ - AVCodecContext *c; - int ret; - - c = st->codec; - - /* allocate and init a re-usable frame */ - audio_frame = av_frame_alloc(); - if (!audio_frame) { - fprintf(stderr, "Could not allocate audio frame\n"); - exit(1); - } - - /* open it */ - ret = avcodec_open2(c, codec, NULL); - if (ret < 0) { - fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret)); - exit(1); - } - - /* init signal generator */ - t = 0; - tincr = 2 * M_PI * 110.0 / c->sample_rate; - /* increment frequency by 110 Hz per second */ - tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate; - - src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ? - 10000 : c->frame_size; - - ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels, - src_nb_samples, AV_SAMPLE_FMT_S16, 0); - if (ret < 0) { - fprintf(stderr, "Could not allocate source samples\n"); - exit(1); - } - - /* compute the number of converted samples: buffering is avoided - * ensuring that the output buffer will contain at least all the - * converted input samples */ - max_dst_nb_samples = src_nb_samples; - - /* create resampler context */ - if (c->sample_fmt != AV_SAMPLE_FMT_S16) { - swr_ctx = swr_alloc(); - if (!swr_ctx) { - fprintf(stderr, "Could not allocate resampler context\n"); - exit(1); - } - - /* set options */ - av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0); - av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0); - av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0); - av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0); - - /* initialize the resampling context */ - if ((ret = swr_init(swr_ctx)) < 0) { - fprintf(stderr, "Failed to initialize the resampling context\n"); - exit(1); - } - - ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels, - max_dst_nb_samples, c->sample_fmt, 0); - if (ret < 0) { - fprintf(stderr, "Could not allocate destination samples\n"); - exit(1); - } - } else { - dst_samples_data = src_samples_data; - } - dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples, - c->sample_fmt, 0); -} - -/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and - * 'nb_channels' channels. */ -static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels) -{ - int j, i, v; - int16_t *q; - - q = samples; - for (j = 0; j < frame_size; j++) { - v = (int)(sin(t) * 10000); - for (i = 0; i < nb_channels; i++) - *q++ = v; - t += tincr; - tincr += tincr2; - } -} - -static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush) -{ - AVCodecContext *c; - AVPacket pkt = { 0 }; // data and size must be 0; - int got_packet, ret, dst_nb_samples; - - av_init_packet(&pkt); - c = st->codec; - - if (!flush) { - get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels); - - /* convert samples from native format to destination codec format, using the resampler */ - if (swr_ctx) { - /* compute destination number of samples */ - dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples, - c->sample_rate, c->sample_rate, AV_ROUND_UP); - if (dst_nb_samples > max_dst_nb_samples) { - av_free(dst_samples_data[0]); - ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels, - dst_nb_samples, c->sample_fmt, 0); - if (ret < 0) - exit(1); - max_dst_nb_samples = dst_nb_samples; - dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples, - c->sample_fmt, 0); - } - - /* convert to destination format */ - ret = swr_convert(swr_ctx, - dst_samples_data, dst_nb_samples, - (const uint8_t **)src_samples_data, src_nb_samples); - if (ret < 0) { - fprintf(stderr, "Error while converting\n"); - exit(1); - } - } else { - dst_nb_samples = src_nb_samples; - } - - audio_frame->nb_samples = dst_nb_samples; - audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base); - avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt, - dst_samples_data[0], dst_samples_size, 0); - samples_count += dst_nb_samples; - } - - ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet); - if (ret < 0) { - fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret)); - exit(1); - } - - if (!got_packet) { - if (flush) - audio_is_eof = 1; - return; - } - - ret = write_frame(oc, &c->time_base, st, &pkt); - if (ret < 0) { - fprintf(stderr, "Error while writing audio frame: %s\n", - av_err2str(ret)); - exit(1); - } -} - -static void close_audio(AVFormatContext *oc, AVStream *st) -{ - avcodec_close(st->codec); - if (dst_samples_data != src_samples_data) { - av_free(dst_samples_data[0]); - av_free(dst_samples_data); - } - av_free(src_samples_data[0]); - av_free(src_samples_data); - av_frame_free(&audio_frame); -} - -/**************************************************************/ -/* video output */ - -static AVFrame *frame; -static AVPicture src_picture, dst_picture; -static int frame_count; - -static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st) -{ - int ret; - AVCodecContext *c = st->codec; - - /* open the codec */ - ret = avcodec_open2(c, codec, NULL); - if (ret < 0) { - fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret)); - exit(1); - } - - /* allocate and init a re-usable frame */ - frame = av_frame_alloc(); - if (!frame) { - fprintf(stderr, "Could not allocate video frame\n"); - exit(1); - } - frame->format = c->pix_fmt; - frame->width = c->width; - frame->height = c->height; - - /* Allocate the encoded raw picture. */ - ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height); - if (ret < 0) { - fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret)); - exit(1); - } - - /* If the output format is not YUV420P, then a temporary YUV420P - * picture is needed too. It is then converted to the required - * output format. */ - if (c->pix_fmt != AV_PIX_FMT_YUV420P) { - ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height); - if (ret < 0) { - fprintf(stderr, "Could not allocate temporary picture: %s\n", - av_err2str(ret)); - exit(1); - } - } - - /* copy data and linesize picture pointers to frame */ - *((AVPicture *)frame) = dst_picture; -} - -/* Prepare a dummy image. */ -static void fill_yuv_image(AVPicture *pict, int frame_index, - int width, int height) -{ - int x, y, i; - - i = frame_index; - - /* Y */ - for (y = 0; y < height; y++) - for (x = 0; x < width; x++) - pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3; - - /* Cb and Cr */ - for (y = 0; y < height / 2; y++) { - for (x = 0; x < width / 2; x++) { - pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2; - pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5; - } - } -} - -static void write_video_frame(AVFormatContext *oc, AVStream *st, int flush) -{ - int ret; - static struct SwsContext *sws_ctx; - AVCodecContext *c = st->codec; - - if (!flush) { - if (c->pix_fmt != AV_PIX_FMT_YUV420P) { - /* as we only generate a YUV420P picture, we must convert it - * to the codec pixel format if needed */ - if (!sws_ctx) { - sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P, - c->width, c->height, c->pix_fmt, - sws_flags, NULL, NULL, NULL); - if (!sws_ctx) { - fprintf(stderr, - "Could not initialize the conversion context\n"); - exit(1); - } - } - fill_yuv_image(&src_picture, frame_count, c->width, c->height); - sws_scale(sws_ctx, - (const uint8_t * const *)src_picture.data, src_picture.linesize, - 0, c->height, dst_picture.data, dst_picture.linesize); - } else { - fill_yuv_image(&dst_picture, frame_count, c->width, c->height); - } - } - - if (oc->oformat->flags & AVFMT_RAWPICTURE && !flush) { - /* Raw video case - directly store the picture in the packet */ - AVPacket pkt; - av_init_packet(&pkt); - - pkt.flags |= AV_PKT_FLAG_KEY; - pkt.stream_index = st->index; - pkt.data = dst_picture.data[0]; - pkt.size = sizeof(AVPicture); - - ret = av_interleaved_write_frame(oc, &pkt); - } else { - AVPacket pkt = { 0 }; - int got_packet; - av_init_packet(&pkt); - - /* encode the image */ - frame->pts = frame_count; - ret = avcodec_encode_video2(c, &pkt, flush ? NULL : frame, &got_packet); - if (ret < 0) { - fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret)); - exit(1); - } - /* If size is zero, it means the image was buffered. */ - - if (got_packet) { - ret = write_frame(oc, &c->time_base, st, &pkt); - } else { - if (flush) - video_is_eof = 1; - ret = 0; - } - } - - if (ret < 0) { - fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret)); - exit(1); - } - frame_count++; -} - -static void close_video(AVFormatContext *oc, AVStream *st) -{ - avcodec_close(st->codec); - av_free(src_picture.data[0]); - av_free(dst_picture.data[0]); - av_frame_free(&frame); -} - -/**************************************************************/ -/* media file output */ - -int main(int argc, char **argv) -{ - const char *filename; - AVOutputFormat *fmt; - AVFormatContext *oc; - AVStream *audio_st, *video_st; - AVCodec *audio_codec, *video_codec; - double audio_time, video_time; - int flush, ret; - - /* Initialize libavcodec, and register all codecs and formats. */ - av_register_all(); - - if (argc != 2) { - printf("usage: %s output_file\n" - "API example program to output a media file with libavformat.\n" - "This program generates a synthetic audio and video stream, encodes and\n" - "muxes them into a file named output_file.\n" - "The output format is automatically guessed according to the file extension.\n" - "Raw images can also be output by using '%%d' in the filename.\n" - "\n", argv[0]); - return 1; - } - - filename = argv[1]; - - /* allocate the output media context */ - avformat_alloc_output_context2(&oc, NULL, NULL, filename); - if (!oc) { - printf("Could not deduce output format from file extension: using MPEG.\n"); - avformat_alloc_output_context2(&oc, NULL, "mpeg", filename); - } - if (!oc) - return 1; - - fmt = oc->oformat; - - /* Add the audio and video streams using the default format codecs - * and initialize the codecs. */ - video_st = NULL; - audio_st = NULL; - - if (fmt->video_codec != AV_CODEC_ID_NONE) - video_st = add_stream(oc, &video_codec, fmt->video_codec); - if (fmt->audio_codec != AV_CODEC_ID_NONE) - audio_st = add_stream(oc, &audio_codec, fmt->audio_codec); - - /* Now that all the parameters are set, we can open the audio and - * video codecs and allocate the necessary encode buffers. */ - if (video_st) - open_video(oc, video_codec, video_st); - if (audio_st) - open_audio(oc, audio_codec, audio_st); - - av_dump_format(oc, 0, filename, 1); - - /* open the output file, if needed */ - if (!(fmt->flags & AVFMT_NOFILE)) { - ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE); - if (ret < 0) { - fprintf(stderr, "Could not open '%s': %s\n", filename, - av_err2str(ret)); - return 1; - } - } - - /* Write the stream header, if any. */ - ret = avformat_write_header(oc, NULL); - if (ret < 0) { - fprintf(stderr, "Error occurred when opening output file: %s\n", - av_err2str(ret)); - return 1; - } - - flush = 0; - while ((video_st && !video_is_eof) || (audio_st && !audio_is_eof)) { - /* Compute current audio and video time. */ - audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY; - video_time = (video_st && !video_is_eof) ? video_st->pts.val * av_q2d(video_st->time_base) : INFINITY; - - if (!flush && - (!audio_st || audio_time >= STREAM_DURATION) && - (!video_st || video_time >= STREAM_DURATION)) { - flush = 1; - } - - /* write interleaved audio and video frames */ - if (audio_st && !audio_is_eof && audio_time <= video_time) { - write_audio_frame(oc, audio_st, flush); - } else if (video_st && !video_is_eof && video_time < audio_time) { - write_video_frame(oc, video_st, flush); - } - } - - /* Write the trailer, if any. The trailer must be written before you - * close the CodecContexts open when you wrote the header; otherwise - * av_write_trailer() may try to use memory that was freed on - * av_codec_close(). */ - av_write_trailer(oc); - - /* Close each codec. */ - if (video_st) - close_video(oc, video_st); - if (audio_st) - close_audio(oc, audio_st); - - if (!(fmt->flags & AVFMT_NOFILE)) - /* Close the output file. */ - avio_close(oc->pb); - - /* free the stream */ - avformat_free_context(oc); - - return 0; -} diff --git a/doc/examples/output.c b/doc/examples/output.c new file mode 100644 index 0000000000..dd0e6a224e --- /dev/null +++ b/doc/examples/output.c @@ -0,0 +1,527 @@ +/* + * Copyright (c) 2003 Fabrice Bellard + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavformat API example. + * + * @example output.c + * Output a media file in any supported libavformat format. The default + * codecs are used. + */ + +#include +#include +#include +#include + +#include "libavutil/mathematics.h" +#include "libavformat/avformat.h" +#include "libswscale/swscale.h" + +/* 5 seconds stream duration */ +#define STREAM_DURATION 5.0 +#define STREAM_FRAME_RATE 25 /* 25 images/s */ +#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE)) +#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */ + +static int sws_flags = SWS_BICUBIC; + +/**************************************************************/ +/* audio output */ + +static float t, tincr, tincr2; +static int16_t *samples; +static int audio_input_frame_size; + +/* + * add an audio output stream + */ +static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id) +{ + AVCodecContext *c; + AVStream *st; + AVCodec *codec; + + /* find the audio encoder */ + codec = avcodec_find_encoder(codec_id); + if (!codec) { + fprintf(stderr, "codec not found\n"); + exit(1); + } + + st = avformat_new_stream(oc, codec); + if (!st) { + fprintf(stderr, "Could not alloc stream\n"); + exit(1); + } + + c = st->codec; + + /* put sample parameters */ + c->sample_fmt = AV_SAMPLE_FMT_S16; + c->bit_rate = 64000; + c->sample_rate = 44100; + c->channels = 2; + + // some formats want stream headers to be separate + if (oc->oformat->flags & AVFMT_GLOBALHEADER) + c->flags |= CODEC_FLAG_GLOBAL_HEADER; + + return st; +} + +static void open_audio(AVFormatContext *oc, AVStream *st) +{ + AVCodecContext *c; + + c = st->codec; + + /* open it */ + if (avcodec_open2(c, NULL, NULL) < 0) { + fprintf(stderr, "could not open codec\n"); + exit(1); + } + + /* init signal generator */ + t = 0; + tincr = 2 * M_PI * 110.0 / c->sample_rate; + /* increment frequency by 110 Hz per second */ + tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate; + + if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE) + audio_input_frame_size = 10000; + else + audio_input_frame_size = c->frame_size; + samples = av_malloc(audio_input_frame_size * + av_get_bytes_per_sample(c->sample_fmt) * + c->channels); +} + +/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and + * 'nb_channels' channels. */ +static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels) +{ + int j, i, v; + int16_t *q; + + q = samples; + for (j = 0; j < frame_size; j++) { + v = (int)(sin(t) * 10000); + for (i = 0; i < nb_channels; i++) + *q++ = v; + t += tincr; + tincr += tincr2; + } +} + +static void write_audio_frame(AVFormatContext *oc, AVStream *st) +{ + AVCodecContext *c; + AVPacket pkt = { 0 }; // data and size must be 0; + AVFrame *frame = av_frame_alloc(); + int got_packet; + + av_init_packet(&pkt); + c = st->codec; + + get_audio_frame(samples, audio_input_frame_size, c->channels); + frame->nb_samples = audio_input_frame_size; + avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, + (uint8_t *)samples, + audio_input_frame_size * + av_get_bytes_per_sample(c->sample_fmt) * + c->channels, 1); + + avcodec_encode_audio2(c, &pkt, frame, &got_packet); + if (!got_packet) + return; + + pkt.stream_index = st->index; + + /* Write the compressed frame to the media file. */ + if (av_interleaved_write_frame(oc, &pkt) != 0) { + fprintf(stderr, "Error while writing audio frame\n"); + exit(1); + } + av_frame_free(&frame); +} + +static void close_audio(AVFormatContext *oc, AVStream *st) +{ + avcodec_close(st->codec); + + av_free(samples); +} + +/**************************************************************/ +/* video output */ + +static AVFrame *picture, *tmp_picture; +static int frame_count; + +/* Add a video output stream. */ +static AVStream *add_video_stream(AVFormatContext *oc, enum AVCodecID codec_id) +{ + AVCodecContext *c; + AVStream *st; + AVCodec *codec; + + /* find the video encoder */ + codec = avcodec_find_encoder(codec_id); + if (!codec) { + fprintf(stderr, "codec not found\n"); + exit(1); + } + + st = avformat_new_stream(oc, codec); + if (!st) { + fprintf(stderr, "Could not alloc stream\n"); + exit(1); + } + + c = st->codec; + + /* Put sample parameters. */ + c->bit_rate = 400000; + /* Resolution must be a multiple of two. */ + c->width = 352; + c->height = 288; + /* timebase: This is the fundamental unit of time (in seconds) in terms + * of which frame timestamps are represented. For fixed-fps content, + * timebase should be 1/framerate and timestamp increments should be + * identical to 1. */ + c->time_base.den = STREAM_FRAME_RATE; + c->time_base.num = 1; + c->gop_size = 12; /* emit one intra frame every twelve frames at most */ + c->pix_fmt = STREAM_PIX_FMT; + if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) { + /* just for testing, we also add B frames */ + c->max_b_frames = 2; + } + if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) { + /* Needed to avoid using macroblocks in which some coeffs overflow. + * This does not happen with normal video, it just happens here as + * the motion of the chroma plane does not match the luma plane. */ + c->mb_decision = 2; + } + /* Some formats want stream headers to be separate. */ + if (oc->oformat->flags & AVFMT_GLOBALHEADER) + c->flags |= CODEC_FLAG_GLOBAL_HEADER; + + return st; +} + +static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height) +{ + AVFrame *picture; + uint8_t *picture_buf; + int size; + + picture = av_frame_alloc(); + if (!picture) + return NULL; + size = avpicture_get_size(pix_fmt, width, height); + picture_buf = av_malloc(size); + if (!picture_buf) { + av_free(picture); + return NULL; + } + avpicture_fill((AVPicture *)picture, picture_buf, + pix_fmt, width, height); + return picture; +} + +static void open_video(AVFormatContext *oc, AVStream *st) +{ + AVCodecContext *c; + + c = st->codec; + + /* open the codec */ + if (avcodec_open2(c, NULL, NULL) < 0) { + fprintf(stderr, "could not open codec\n"); + exit(1); + } + + /* Allocate the encoded raw picture. */ + picture = alloc_picture(c->pix_fmt, c->width, c->height); + if (!picture) { + fprintf(stderr, "Could not allocate picture\n"); + exit(1); + } + + /* If the output format is not YUV420P, then a temporary YUV420P + * picture is needed too. It is then converted to the required + * output format. */ + tmp_picture = NULL; + if (c->pix_fmt != AV_PIX_FMT_YUV420P) { + tmp_picture = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height); + if (!tmp_picture) { + fprintf(stderr, "Could not allocate temporary picture\n"); + exit(1); + } + } +} + +/* Prepare a dummy image. */ +static void fill_yuv_image(AVFrame *pict, int frame_index, + int width, int height) +{ + int x, y, i; + + i = frame_index; + + /* Y */ + for (y = 0; y < height; y++) + for (x = 0; x < width; x++) + pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3; + + /* Cb and Cr */ + for (y = 0; y < height / 2; y++) { + for (x = 0; x < width / 2; x++) { + pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2; + pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5; + } + } +} + +static void write_video_frame(AVFormatContext *oc, AVStream *st) +{ + int ret; + AVCodecContext *c; + static struct SwsContext *img_convert_ctx; + + c = st->codec; + + if (frame_count >= STREAM_NB_FRAMES) { + /* No more frames to compress. The codec has a latency of a few + * frames if using B-frames, so we get the last frames by + * passing the same picture again. */ + } else { + if (c->pix_fmt != AV_PIX_FMT_YUV420P) { + /* as we only generate a YUV420P picture, we must convert it + * to the codec pixel format if needed */ + if (img_convert_ctx == NULL) { + img_convert_ctx = sws_getContext(c->width, c->height, + AV_PIX_FMT_YUV420P, + c->width, c->height, + c->pix_fmt, + sws_flags, NULL, NULL, NULL); + if (img_convert_ctx == NULL) { + fprintf(stderr, + "Cannot initialize the conversion context\n"); + exit(1); + } + } + fill_yuv_image(tmp_picture, frame_count, c->width, c->height); + sws_scale(img_convert_ctx, tmp_picture->data, tmp_picture->linesize, + 0, c->height, picture->data, picture->linesize); + } else { + fill_yuv_image(picture, frame_count, c->width, c->height); + } + } + + if (oc->oformat->flags & AVFMT_RAWPICTURE) { + /* Raw video case - the API will change slightly in the near + * future for that. */ + AVPacket pkt; + av_init_packet(&pkt); + + pkt.flags |= AV_PKT_FLAG_KEY; + pkt.stream_index = st->index; + pkt.data = (uint8_t *)picture; + pkt.size = sizeof(AVPicture); + + ret = av_interleaved_write_frame(oc, &pkt); + } else { + AVPacket pkt = { 0 }; + int got_packet; + av_init_packet(&pkt); + + /* encode the image */ + ret = avcodec_encode_video2(c, &pkt, picture, &got_packet); + /* If size is zero, it means the image was buffered. */ + if (!ret && got_packet && pkt.size) { + if (pkt.pts != AV_NOPTS_VALUE) { + pkt.pts = av_rescale_q(pkt.pts, + c->time_base, st->time_base); + } + if (pkt.dts != AV_NOPTS_VALUE) { + pkt.dts = av_rescale_q(pkt.dts, + c->time_base, st->time_base); + } + pkt.stream_index = st->index; + + /* Write the compressed frame to the media file. */ + ret = av_interleaved_write_frame(oc, &pkt); + } else { + ret = 0; + } + } + if (ret != 0) { + fprintf(stderr, "Error while writing video frame\n"); + exit(1); + } + frame_count++; +} + +static void close_video(AVFormatContext *oc, AVStream *st) +{ + avcodec_close(st->codec); + av_free(picture->data[0]); + av_free(picture); + if (tmp_picture) { + av_free(tmp_picture->data[0]); + av_free(tmp_picture); + } +} + +/**************************************************************/ +/* media file output */ + +int main(int argc, char **argv) +{ + const char *filename; + AVOutputFormat *fmt; + AVFormatContext *oc; + AVStream *audio_st, *video_st; + double audio_pts, video_pts; + int i; + + /* Initialize libavcodec, and register all codecs and formats. */ + av_register_all(); + + if (argc != 2) { + printf("usage: %s output_file\n" + "API example program to output a media file with libavformat.\n" + "The output format is automatically guessed according to the file extension.\n" + "Raw images can also be output by using '%%d' in the filename\n" + "\n", argv[0]); + return 1; + } + + filename = argv[1]; + + /* Autodetect the output format from the name. default is MPEG. */ + fmt = av_guess_format(NULL, filename, NULL); + if (!fmt) { + printf("Could not deduce output format from file extension: using MPEG.\n"); + fmt = av_guess_format("mpeg", NULL, NULL); + } + if (!fmt) { + fprintf(stderr, "Could not find suitable output format\n"); + return 1; + } + + /* Allocate the output media context. */ + oc = avformat_alloc_context(); + if (!oc) { + fprintf(stderr, "Memory error\n"); + return 1; + } + oc->oformat = fmt; + snprintf(oc->filename, sizeof(oc->filename), "%s", filename); + + /* Add the audio and video streams using the default format codecs + * and initialize the codecs. */ + video_st = NULL; + audio_st = NULL; + if (fmt->video_codec != AV_CODEC_ID_NONE) { + video_st = add_video_stream(oc, fmt->video_codec); + } + if (fmt->audio_codec != AV_CODEC_ID_NONE) { + audio_st = add_audio_stream(oc, fmt->audio_codec); + } + + /* Now that all the parameters are set, we can open the audio and + * video codecs and allocate the necessary encode buffers. */ + if (video_st) + open_video(oc, video_st); + if (audio_st) + open_audio(oc, audio_st); + + av_dump_format(oc, 0, filename, 1); + + /* open the output file, if needed */ + if (!(fmt->flags & AVFMT_NOFILE)) { + if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) { + fprintf(stderr, "Could not open '%s'\n", filename); + return 1; + } + } + + /* Write the stream header, if any. */ + avformat_write_header(oc, NULL); + + for (;;) { + /* Compute current audio and video time. */ + if (audio_st) + audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den; + else + audio_pts = 0.0; + + if (video_st) + video_pts = (double)video_st->pts.val * video_st->time_base.num / + video_st->time_base.den; + else + video_pts = 0.0; + + if ((!audio_st || audio_pts >= STREAM_DURATION) && + (!video_st || video_pts >= STREAM_DURATION)) + break; + + /* write interleaved audio and video frames */ + if (!video_st || (video_st && audio_st && audio_pts < video_pts)) { + write_audio_frame(oc, audio_st); + } else { + write_video_frame(oc, video_st); + } + } + + /* Write the trailer, if any. The trailer must be written before you + * close the CodecContexts open when you wrote the header; otherwise + * av_write_trailer() may try to use memory that was freed on + * av_codec_close(). */ + av_write_trailer(oc); + + /* Close each codec. */ + if (video_st) + close_video(oc, video_st); + if (audio_st) + close_audio(oc, audio_st); + + /* Free the streams. */ + for (i = 0; i < oc->nb_streams; i++) { + av_freep(&oc->streams[i]->codec); + av_freep(&oc->streams[i]); + } + + if (!(fmt->flags & AVFMT_NOFILE)) + /* Close the output file. */ + avio_close(oc->pb); + + /* free the stream */ + av_free(oc); + + return 0; +} diff --git a/doc/examples/remuxing.c b/doc/examples/remuxing.c deleted file mode 100644 index 39d9dee862..0000000000 --- a/doc/examples/remuxing.c +++ /dev/null @@ -1,165 +0,0 @@ -/* - * Copyright (c) 2013 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * libavformat/libavcodec demuxing and muxing API example. - * - * Remux streams from one container format to another. - * @example remuxing.c - */ - -#include -#include - -static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag) -{ - AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base; - - printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n", - tag, - av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base), - av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base), - av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base), - pkt->stream_index); -} - -int main(int argc, char **argv) -{ - AVOutputFormat *ofmt = NULL; - AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL; - AVPacket pkt; - const char *in_filename, *out_filename; - int ret, i; - - if (argc < 3) { - printf("usage: %s input output\n" - "API example program to remux a media file with libavformat and libavcodec.\n" - "The output format is guessed according to the file extension.\n" - "\n", argv[0]); - return 1; - } - - in_filename = argv[1]; - out_filename = argv[2]; - - av_register_all(); - - if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) { - fprintf(stderr, "Could not open input file '%s'", in_filename); - goto end; - } - - if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) { - fprintf(stderr, "Failed to retrieve input stream information"); - goto end; - } - - av_dump_format(ifmt_ctx, 0, in_filename, 0); - - avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename); - if (!ofmt_ctx) { - fprintf(stderr, "Could not create output context\n"); - ret = AVERROR_UNKNOWN; - goto end; - } - - ofmt = ofmt_ctx->oformat; - - for (i = 0; i < ifmt_ctx->nb_streams; i++) { - AVStream *in_stream = ifmt_ctx->streams[i]; - AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec); - if (!out_stream) { - fprintf(stderr, "Failed allocating output stream\n"); - ret = AVERROR_UNKNOWN; - goto end; - } - - ret = avcodec_copy_context(out_stream->codec, in_stream->codec); - if (ret < 0) { - fprintf(stderr, "Failed to copy context from input to output stream codec context\n"); - goto end; - } - out_stream->codec->codec_tag = 0; - if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER) - out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER; - } - av_dump_format(ofmt_ctx, 0, out_filename, 1); - - if (!(ofmt->flags & AVFMT_NOFILE)) { - ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE); - if (ret < 0) { - fprintf(stderr, "Could not open output file '%s'", out_filename); - goto end; - } - } - - ret = avformat_write_header(ofmt_ctx, NULL); - if (ret < 0) { - fprintf(stderr, "Error occurred when opening output file\n"); - goto end; - } - - while (1) { - AVStream *in_stream, *out_stream; - - ret = av_read_frame(ifmt_ctx, &pkt); - if (ret < 0) - break; - - in_stream = ifmt_ctx->streams[pkt.stream_index]; - out_stream = ofmt_ctx->streams[pkt.stream_index]; - - log_packet(ifmt_ctx, &pkt, "in"); - - /* copy packet */ - pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base); - pkt.pos = -1; - log_packet(ofmt_ctx, &pkt, "out"); - - ret = av_interleaved_write_frame(ofmt_ctx, &pkt); - if (ret < 0) { - fprintf(stderr, "Error muxing packet\n"); - break; - } - av_free_packet(&pkt); - } - - av_write_trailer(ofmt_ctx); -end: - - avformat_close_input(&ifmt_ctx); - - /* close output */ - if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE)) - avio_close(ofmt_ctx->pb); - avformat_free_context(ofmt_ctx); - - if (ret < 0 && ret != AVERROR_EOF) { - fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); - return 1; - } - - return 0; -} diff --git a/doc/examples/resampling_audio.c b/doc/examples/resampling_audio.c deleted file mode 100644 index f743cbe550..0000000000 --- a/doc/examples/resampling_audio.c +++ /dev/null @@ -1,215 +0,0 @@ -/* - * Copyright (c) 2012 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @example resampling_audio.c - * libswresample API use example. - */ - -#include -#include -#include -#include - -static int get_format_from_sample_fmt(const char **fmt, - enum AVSampleFormat sample_fmt) -{ - int i; - struct sample_fmt_entry { - enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; - } sample_fmt_entries[] = { - { AV_SAMPLE_FMT_U8, "u8", "u8" }, - { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, - { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, - { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, - { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, - }; - *fmt = NULL; - - for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { - struct sample_fmt_entry *entry = &sample_fmt_entries[i]; - if (sample_fmt == entry->sample_fmt) { - *fmt = AV_NE(entry->fmt_be, entry->fmt_le); - return 0; - } - } - - fprintf(stderr, - "Sample format %s not supported as output format\n", - av_get_sample_fmt_name(sample_fmt)); - return AVERROR(EINVAL); -} - -/** - * Fill dst buffer with nb_samples, generated starting from t. - */ -static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t) -{ - int i, j; - double tincr = 1.0 / sample_rate, *dstp = dst; - const double c = 2 * M_PI * 440.0; - - /* generate sin tone with 440Hz frequency and duplicated channels */ - for (i = 0; i < nb_samples; i++) { - *dstp = sin(c * *t); - for (j = 1; j < nb_channels; j++) - dstp[j] = dstp[0]; - dstp += nb_channels; - *t += tincr; - } -} - -int main(int argc, char **argv) -{ - int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND; - int src_rate = 48000, dst_rate = 44100; - uint8_t **src_data = NULL, **dst_data = NULL; - int src_nb_channels = 0, dst_nb_channels = 0; - int src_linesize, dst_linesize; - int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples; - enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16; - const char *dst_filename = NULL; - FILE *dst_file; - int dst_bufsize; - const char *fmt; - struct SwrContext *swr_ctx; - double t; - int ret; - - if (argc != 2) { - fprintf(stderr, "Usage: %s output_file\n" - "API example program to show how to resample an audio stream with libswresample.\n" - "This program generates a series of audio frames, resamples them to a specified " - "output format and rate and saves them to an output file named output_file.\n", - argv[0]); - exit(1); - } - dst_filename = argv[1]; - - dst_file = fopen(dst_filename, "wb"); - if (!dst_file) { - fprintf(stderr, "Could not open destination file %s\n", dst_filename); - exit(1); - } - - /* create resampler context */ - swr_ctx = swr_alloc(); - if (!swr_ctx) { - fprintf(stderr, "Could not allocate resampler context\n"); - ret = AVERROR(ENOMEM); - goto end; - } - - /* set options */ - av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0); - av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0); - - av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0); - av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0); - - /* initialize the resampling context */ - if ((ret = swr_init(swr_ctx)) < 0) { - fprintf(stderr, "Failed to initialize the resampling context\n"); - goto end; - } - - /* allocate source and destination samples buffers */ - - src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout); - ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels, - src_nb_samples, src_sample_fmt, 0); - if (ret < 0) { - fprintf(stderr, "Could not allocate source samples\n"); - goto end; - } - - /* compute the number of converted samples: buffering is avoided - * ensuring that the output buffer will contain at least all the - * converted input samples */ - max_dst_nb_samples = dst_nb_samples = - av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); - - /* buffer is going to be directly written to a rawaudio file, no alignment */ - dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout); - ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels, - dst_nb_samples, dst_sample_fmt, 0); - if (ret < 0) { - fprintf(stderr, "Could not allocate destination samples\n"); - goto end; - } - - t = 0; - do { - /* generate synthetic audio */ - fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t); - - /* compute destination number of samples */ - dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) + - src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); - if (dst_nb_samples > max_dst_nb_samples) { - av_free(dst_data[0]); - ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels, - dst_nb_samples, dst_sample_fmt, 1); - if (ret < 0) - break; - max_dst_nb_samples = dst_nb_samples; - } - - /* convert to destination format */ - ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples); - if (ret < 0) { - fprintf(stderr, "Error while converting\n"); - goto end; - } - dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, - ret, dst_sample_fmt, 1); - if (dst_bufsize < 0) { - fprintf(stderr, "Could not get sample buffer size\n"); - goto end; - } - printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret); - fwrite(dst_data[0], 1, dst_bufsize, dst_file); - } while (t < 10); - - if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0) - goto end; - fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n" - "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n", - fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename); - -end: - if (dst_file) - fclose(dst_file); - - if (src_data) - av_freep(&src_data[0]); - av_freep(&src_data); - - if (dst_data) - av_freep(&dst_data[0]); - av_freep(&dst_data); - - swr_free(&swr_ctx); - return ret < 0; -} diff --git a/doc/examples/scaling_video.c b/doc/examples/scaling_video.c deleted file mode 100644 index fcb98b7489..0000000000 --- a/doc/examples/scaling_video.c +++ /dev/null @@ -1,141 +0,0 @@ -/* - * Copyright (c) 2012 Stefano Sabatini - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * libswscale API use example. - * @example scaling_video.c - */ - -#include -#include -#include - -static void fill_yuv_image(uint8_t *data[4], int linesize[4], - int width, int height, int frame_index) -{ - int x, y; - - /* Y */ - for (y = 0; y < height; y++) - for (x = 0; x < width; x++) - data[0][y * linesize[0] + x] = x + y + frame_index * 3; - - /* Cb and Cr */ - for (y = 0; y < height / 2; y++) { - for (x = 0; x < width / 2; x++) { - data[1][y * linesize[1] + x] = 128 + y + frame_index * 2; - data[2][y * linesize[2] + x] = 64 + x + frame_index * 5; - } - } -} - -int main(int argc, char **argv) -{ - uint8_t *src_data[4], *dst_data[4]; - int src_linesize[4], dst_linesize[4]; - int src_w = 320, src_h = 240, dst_w, dst_h; - enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24; - const char *dst_size = NULL; - const char *dst_filename = NULL; - FILE *dst_file; - int dst_bufsize; - struct SwsContext *sws_ctx; - int i, ret; - - if (argc != 3) { - fprintf(stderr, "Usage: %s output_file output_size\n" - "API example program to show how to scale an image with libswscale.\n" - "This program generates a series of pictures, rescales them to the given " - "output_size and saves them to an output file named output_file\n." - "\n", argv[0]); - exit(1); - } - dst_filename = argv[1]; - dst_size = argv[2]; - - if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) { - fprintf(stderr, - "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n", - dst_size); - exit(1); - } - - dst_file = fopen(dst_filename, "wb"); - if (!dst_file) { - fprintf(stderr, "Could not open destination file %s\n", dst_filename); - exit(1); - } - - /* create scaling context */ - sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt, - dst_w, dst_h, dst_pix_fmt, - SWS_BILINEAR, NULL, NULL, NULL); - if (!sws_ctx) { - fprintf(stderr, - "Impossible to create scale context for the conversion " - "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n", - av_get_pix_fmt_name(src_pix_fmt), src_w, src_h, - av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h); - ret = AVERROR(EINVAL); - goto end; - } - - /* allocate source and destination image buffers */ - if ((ret = av_image_alloc(src_data, src_linesize, - src_w, src_h, src_pix_fmt, 16)) < 0) { - fprintf(stderr, "Could not allocate source image\n"); - goto end; - } - - /* buffer is going to be written to rawvideo file, no alignment */ - if ((ret = av_image_alloc(dst_data, dst_linesize, - dst_w, dst_h, dst_pix_fmt, 1)) < 0) { - fprintf(stderr, "Could not allocate destination image\n"); - goto end; - } - dst_bufsize = ret; - - for (i = 0; i < 100; i++) { - /* generate synthetic video */ - fill_yuv_image(src_data, src_linesize, src_w, src_h, i); - - /* convert to destination format */ - sws_scale(sws_ctx, (const uint8_t * const*)src_data, - src_linesize, 0, src_h, dst_data, dst_linesize); - - /* write scaled image to file */ - fwrite(dst_data[0], 1, dst_bufsize, dst_file); - } - - fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n" - "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", - av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename); - -end: - if (dst_file) - fclose(dst_file); - av_freep(&src_data[0]); - av_freep(&dst_data[0]); - sws_freeContext(sws_ctx); - return ret < 0; -} diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c index bf0128f68d..6206afe2e4 100644 --- a/doc/examples/transcode_aac.c +++ b/doc/examples/transcode_aac.c @@ -1,18 +1,18 @@ /* - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -21,7 +21,7 @@ * simple audio converter * * @example transcode_aac.c - * Convert an input audio file to AAC in an MP4 container using FFmpeg. + * Convert an input audio file to AAC in an MP4 container using Libav. * @author Andreas Unterweger (dustsigns@gmail.com) */ @@ -33,12 +33,11 @@ #include "libavcodec/avcodec.h" #include "libavutil/audio_fifo.h" -#include "libavutil/avassert.h" #include "libavutil/avstring.h" #include "libavutil/frame.h" #include "libavutil/opt.h" -#include "libswresample/swresample.h" +#include "libavresample/avresample.h" /** The output bit rate in kbit/s */ #define OUTPUT_BIT_RATE 48000 @@ -227,46 +226,52 @@ static int init_input_frame(AVFrame **frame) /** * Initialize the audio resampler based on the input and output codec settings. * If the input and output sample formats differ, a conversion is required - * libswresample takes care of this, but requires initialization. + * libavresample takes care of this, but requires initialization. */ static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - SwrContext **resample_context) + AVAudioResampleContext **resample_context) { + /** + * Only initialize the resampler if it is necessary, i.e., + * if and only if the sample formats differ. + */ + if (input_codec_context->sample_fmt != output_codec_context->sample_fmt || + input_codec_context->channels != output_codec_context->channels) { int error; + /** Create a resampler context for the conversion. */ + if (!(*resample_context = avresample_alloc_context())) { + fprintf(stderr, "Could not allocate resample context\n"); + return AVERROR(ENOMEM); + } + /** - * Create a resampler context for the conversion. * Set the conversion parameters. * Default channel layouts based on the number of channels * are assumed for simplicity (they are sometimes not detected * properly by the demuxer and/or decoder). */ - *resample_context = swr_alloc_set_opts(NULL, - av_get_default_channel_layout(output_codec_context->channels), - output_codec_context->sample_fmt, - output_codec_context->sample_rate, - av_get_default_channel_layout(input_codec_context->channels), - input_codec_context->sample_fmt, - input_codec_context->sample_rate, - 0, NULL); - if (!*resample_context) { - fprintf(stderr, "Could not allocate resample context\n"); - return AVERROR(ENOMEM); - } - /** - * Perform a sanity check so that the number of converted samples is - * not greater than the number of samples to be converted. - * If the sample rates differ, this case has to be handled differently - */ - av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate); + av_opt_set_int(*resample_context, "in_channel_layout", + av_get_default_channel_layout(input_codec_context->channels), 0); + av_opt_set_int(*resample_context, "out_channel_layout", + av_get_default_channel_layout(output_codec_context->channels), 0); + av_opt_set_int(*resample_context, "in_sample_rate", + input_codec_context->sample_rate, 0); + av_opt_set_int(*resample_context, "out_sample_rate", + output_codec_context->sample_rate, 0); + av_opt_set_int(*resample_context, "in_sample_fmt", + input_codec_context->sample_fmt, 0); + av_opt_set_int(*resample_context, "out_sample_fmt", + output_codec_context->sample_fmt, 0); /** Open the resampler with the specified parameters. */ - if ((error = swr_init(*resample_context)) < 0) { + if ((error = avresample_open(*resample_context)) < 0) { fprintf(stderr, "Could not open resample context\n"); - swr_free(resample_context); + avresample_free(resample_context); return error; } + } return 0; } @@ -385,21 +390,30 @@ static int init_converted_samples(uint8_t ***converted_input_samples, * The conversion happens on a per-frame basis, the size of which is specified * by frame_size. */ -static int convert_samples(const uint8_t **input_data, +static int convert_samples(uint8_t **input_data, uint8_t **converted_data, const int frame_size, - SwrContext *resample_context) + AVAudioResampleContext *resample_context) { int error; /** Convert the samples using the resampler. */ - if ((error = swr_convert(resample_context, - converted_data, frame_size, - input_data , frame_size)) < 0) { + if ((error = avresample_convert(resample_context, converted_data, 0, + frame_size, input_data, 0, frame_size)) < 0) { fprintf(stderr, "Could not convert input samples (error '%s')\n", get_error_text(error)); return error; } + /** + * Perform a sanity check so that the number of converted samples is + * not greater than the number of samples to be converted. + * If the sample rates differ, this case has to be handled differently + */ + if (avresample_available(resample_context)) { + fprintf(stderr, "Converted samples left over\n"); + return AVERROR_EXIT; + } + return 0; } @@ -436,7 +450,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - SwrContext *resampler_context, + AVAudioResampleContext *resampler_context, int *finished) { /** Temporary storage of the input samples of the frame read from the file. */ @@ -473,7 +487,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, * Convert the input samples to the desired output sample format. * This requires a temporary storage provided by converted_input_samples. */ - if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples, + if (convert_samples(input_frame->extended_data, converted_input_samples, input_frame->nb_samples, resampler_context)) goto cleanup; @@ -635,7 +649,7 @@ int main(int argc, char **argv) { AVFormatContext *input_format_context = NULL, *output_format_context = NULL; AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL; - SwrContext *resample_context = NULL; + AVAudioResampleContext *resample_context = NULL; AVAudioFifo *fifo = NULL; int ret = AVERROR_EXIT; @@ -739,7 +753,10 @@ int main(int argc, char **argv) cleanup: if (fifo) av_audio_fifo_free(fifo); - swr_free(&resample_context); + if (resample_context) { + avresample_close(resample_context); + avresample_free(&resample_context); + } if (output_codec_context) avcodec_close(output_codec_context); if (output_format_context) { diff --git a/doc/examples/transcoding.c b/doc/examples/transcoding.c deleted file mode 100644 index 5194de33f4..0000000000 --- a/doc/examples/transcoding.c +++ /dev/null @@ -1,597 +0,0 @@ -/* - * Copyright (c) 2010 Nicolas George - * Copyright (c) 2011 Stefano Sabatini - * Copyright (c) 2014 Andrey Utkin - * - * Permission is hereby granted, free of charge, to any person obtaining a copy - * of this software and associated documentation files (the "Software"), to deal - * in the Software without restriction, including without limitation the rights - * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell - * copies of the Software, and to permit persons to whom the Software is - * furnished to do so, subject to the following conditions: - * - * The above copyright notice and this permission notice shall be included in - * all copies or substantial portions of the Software. - * - * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR - * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, - * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL - * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER - * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, - * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN - * THE SOFTWARE. - */ - -/** - * @file - * API example for demuxing, decoding, filtering, encoding and muxing - * @example doc/examples/transcoding.c - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -static AVFormatContext *ifmt_ctx; -static AVFormatContext *ofmt_ctx; -typedef struct FilteringContext { - AVFilterContext *buffersink_ctx; - AVFilterContext *buffersrc_ctx; - AVFilterGraph *filter_graph; -} FilteringContext; -static FilteringContext *filter_ctx; - -static int open_input_file(const char *filename) -{ - int ret; - unsigned int i; - - ifmt_ctx = NULL; - if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); - return ret; - } - - if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); - return ret; - } - - for (i = 0; i < ifmt_ctx->nb_streams; i++) { - AVStream *stream; - AVCodecContext *codec_ctx; - stream = ifmt_ctx->streams[i]; - codec_ctx = stream->codec; - /* Reencode video & audio and remux subtitles etc. */ - if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO - || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) { - /* Open decoder */ - ret = avcodec_open2(codec_ctx, - avcodec_find_decoder(codec_ctx->codec_id), NULL); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i); - return ret; - } - } - } - - av_dump_format(ifmt_ctx, 0, filename, 0); - return 0; -} - -static int open_output_file(const char *filename) -{ - AVStream *out_stream; - AVStream *in_stream; - AVCodecContext *dec_ctx, *enc_ctx; - AVCodec *encoder; - int ret; - unsigned int i; - - ofmt_ctx = NULL; - avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename); - if (!ofmt_ctx) { - av_log(NULL, AV_LOG_ERROR, "Could not create output context\n"); - return AVERROR_UNKNOWN; - } - - - for (i = 0; i < ifmt_ctx->nb_streams; i++) { - out_stream = avformat_new_stream(ofmt_ctx, NULL); - if (!out_stream) { - av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n"); - return AVERROR_UNKNOWN; - } - - in_stream = ifmt_ctx->streams[i]; - dec_ctx = in_stream->codec; - enc_ctx = out_stream->codec; - - if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO - || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) { - /* in this example, we choose transcoding to same codec */ - encoder = avcodec_find_encoder(dec_ctx->codec_id); - - /* In this example, we transcode to same properties (picture size, - * sample rate etc.). These properties can be changed for output - * streams easily using filters */ - if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) { - enc_ctx->height = dec_ctx->height; - enc_ctx->width = dec_ctx->width; - enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio; - /* take first format from list of supported formats */ - enc_ctx->pix_fmt = encoder->pix_fmts[0]; - /* video time_base can be set to whatever is handy and supported by encoder */ - enc_ctx->time_base = dec_ctx->time_base; - } else { - enc_ctx->sample_rate = dec_ctx->sample_rate; - enc_ctx->channel_layout = dec_ctx->channel_layout; - enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout); - /* take first format from list of supported formats */ - enc_ctx->sample_fmt = encoder->sample_fmts[0]; - enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate}; - } - - /* Third parameter can be used to pass settings to encoder */ - ret = avcodec_open2(enc_ctx, encoder, NULL); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i); - return ret; - } - } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) { - av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i); - return AVERROR_INVALIDDATA; - } else { - /* if this stream must be remuxed */ - ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec, - ifmt_ctx->streams[i]->codec); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n"); - return ret; - } - } - - if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER) - enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER; - - } - av_dump_format(ofmt_ctx, 0, filename, 1); - - if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) { - ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename); - return ret; - } - } - - /* init muxer, write output file header */ - ret = avformat_write_header(ofmt_ctx, NULL); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n"); - return ret; - } - - return 0; -} - -static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx, - AVCodecContext *enc_ctx, const char *filter_spec) -{ - char args[512]; - int ret = 0; - AVFilter *buffersrc = NULL; - AVFilter *buffersink = NULL; - AVFilterContext *buffersrc_ctx = NULL; - AVFilterContext *buffersink_ctx = NULL; - AVFilterInOut *outputs = avfilter_inout_alloc(); - AVFilterInOut *inputs = avfilter_inout_alloc(); - AVFilterGraph *filter_graph = avfilter_graph_alloc(); - - if (!outputs || !inputs || !filter_graph) { - ret = AVERROR(ENOMEM); - goto end; - } - - if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) { - buffersrc = avfilter_get_by_name("buffer"); - buffersink = avfilter_get_by_name("buffersink"); - if (!buffersrc || !buffersink) { - av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n"); - ret = AVERROR_UNKNOWN; - goto end; - } - - snprintf(args, sizeof(args), - "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d", - dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt, - dec_ctx->time_base.num, dec_ctx->time_base.den, - dec_ctx->sample_aspect_ratio.num, - dec_ctx->sample_aspect_ratio.den); - - ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", - args, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n"); - goto end; - } - - ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", - NULL, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n"); - goto end; - } - - ret = av_opt_set_bin(buffersink_ctx, "pix_fmts", - (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt), - AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n"); - goto end; - } - } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) { - buffersrc = avfilter_get_by_name("abuffer"); - buffersink = avfilter_get_by_name("abuffersink"); - if (!buffersrc || !buffersink) { - av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n"); - ret = AVERROR_UNKNOWN; - goto end; - } - - if (!dec_ctx->channel_layout) - dec_ctx->channel_layout = - av_get_default_channel_layout(dec_ctx->channels); - snprintf(args, sizeof(args), - "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64, - dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate, - av_get_sample_fmt_name(dec_ctx->sample_fmt), - dec_ctx->channel_layout); - ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", - args, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n"); - goto end; - } - - ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", - NULL, NULL, filter_graph); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n"); - goto end; - } - - ret = av_opt_set_bin(buffersink_ctx, "sample_fmts", - (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt), - AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n"); - goto end; - } - - ret = av_opt_set_bin(buffersink_ctx, "channel_layouts", - (uint8_t*)&enc_ctx->channel_layout, - sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n"); - goto end; - } - - ret = av_opt_set_bin(buffersink_ctx, "sample_rates", - (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate), - AV_OPT_SEARCH_CHILDREN); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n"); - goto end; - } - } else { - ret = AVERROR_UNKNOWN; - goto end; - } - - /* Endpoints for the filter graph. */ - outputs->name = av_strdup("in"); - outputs->filter_ctx = buffersrc_ctx; - outputs->pad_idx = 0; - outputs->next = NULL; - - inputs->name = av_strdup("out"); - inputs->filter_ctx = buffersink_ctx; - inputs->pad_idx = 0; - inputs->next = NULL; - - if (!outputs->name || !inputs->name) { - ret = AVERROR(ENOMEM); - goto end; - } - - if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec, - &inputs, &outputs, NULL)) < 0) - goto end; - - if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) - goto end; - - /* Fill FilteringContext */ - fctx->buffersrc_ctx = buffersrc_ctx; - fctx->buffersink_ctx = buffersink_ctx; - fctx->filter_graph = filter_graph; - -end: - avfilter_inout_free(&inputs); - avfilter_inout_free(&outputs); - - return ret; -} - -static int init_filters(void) -{ - const char *filter_spec; - unsigned int i; - int ret; - filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx)); - if (!filter_ctx) - return AVERROR(ENOMEM); - - for (i = 0; i < ifmt_ctx->nb_streams; i++) { - filter_ctx[i].buffersrc_ctx = NULL; - filter_ctx[i].buffersink_ctx = NULL; - filter_ctx[i].filter_graph = NULL; - if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO - || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)) - continue; - - - if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) - filter_spec = "null"; /* passthrough (dummy) filter for video */ - else - filter_spec = "anull"; /* passthrough (dummy) filter for audio */ - ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec, - ofmt_ctx->streams[i]->codec, filter_spec); - if (ret) - return ret; - } - return 0; -} - -static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) { - int ret; - int got_frame_local; - AVPacket enc_pkt; - int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) = - (ifmt_ctx->streams[stream_index]->codec->codec_type == - AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2; - - if (!got_frame) - got_frame = &got_frame_local; - - av_log(NULL, AV_LOG_INFO, "Encoding frame\n"); - /* encode filtered frame */ - enc_pkt.data = NULL; - enc_pkt.size = 0; - av_init_packet(&enc_pkt); - ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt, - filt_frame, got_frame); - av_frame_free(&filt_frame); - if (ret < 0) - return ret; - if (!(*got_frame)) - return 0; - - /* prepare packet for muxing */ - enc_pkt.stream_index = stream_index; - enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts, - ofmt_ctx->streams[stream_index]->codec->time_base, - ofmt_ctx->streams[stream_index]->time_base, - AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts, - ofmt_ctx->streams[stream_index]->codec->time_base, - ofmt_ctx->streams[stream_index]->time_base, - AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - enc_pkt.duration = av_rescale_q(enc_pkt.duration, - ofmt_ctx->streams[stream_index]->codec->time_base, - ofmt_ctx->streams[stream_index]->time_base); - - av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n"); - /* mux encoded frame */ - ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt); - return ret; -} - -static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index) -{ - int ret; - AVFrame *filt_frame; - - av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n"); - /* push the decoded frame into the filtergraph */ - ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx, - frame, 0); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n"); - return ret; - } - - /* pull filtered frames from the filtergraph */ - while (1) { - filt_frame = av_frame_alloc(); - if (!filt_frame) { - ret = AVERROR(ENOMEM); - break; - } - av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n"); - ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx, - filt_frame); - if (ret < 0) { - /* if no more frames for output - returns AVERROR(EAGAIN) - * if flushed and no more frames for output - returns AVERROR_EOF - * rewrite retcode to 0 to show it as normal procedure completion - */ - if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) - ret = 0; - av_frame_free(&filt_frame); - break; - } - - filt_frame->pict_type = AV_PICTURE_TYPE_NONE; - ret = encode_write_frame(filt_frame, stream_index, NULL); - if (ret < 0) - break; - } - - return ret; -} - -static int flush_encoder(unsigned int stream_index) -{ - int ret; - int got_frame; - - if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities & - CODEC_CAP_DELAY)) - return 0; - - while (1) { - av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index); - ret = encode_write_frame(NULL, stream_index, &got_frame); - if (ret < 0) - break; - if (!got_frame) - return 0; - } - return ret; -} - -int main(int argc, char **argv) -{ - int ret; - AVPacket packet = { .data = NULL, .size = 0 }; - AVFrame *frame = NULL; - enum AVMediaType type; - unsigned int stream_index; - unsigned int i; - int got_frame; - int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *); - - if (argc != 3) { - av_log(NULL, AV_LOG_ERROR, "Usage: %s \n", argv[0]); - return 1; - } - - av_register_all(); - avfilter_register_all(); - - if ((ret = open_input_file(argv[1])) < 0) - goto end; - if ((ret = open_output_file(argv[2])) < 0) - goto end; - if ((ret = init_filters()) < 0) - goto end; - - /* read all packets */ - while (1) { - if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0) - break; - stream_index = packet.stream_index; - type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type; - av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n", - stream_index); - - if (filter_ctx[stream_index].filter_graph) { - av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n"); - frame = av_frame_alloc(); - if (!frame) { - ret = AVERROR(ENOMEM); - break; - } - packet.dts = av_rescale_q_rnd(packet.dts, - ifmt_ctx->streams[stream_index]->time_base, - ifmt_ctx->streams[stream_index]->codec->time_base, - AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - packet.pts = av_rescale_q_rnd(packet.pts, - ifmt_ctx->streams[stream_index]->time_base, - ifmt_ctx->streams[stream_index]->codec->time_base, - AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 : - avcodec_decode_audio4; - ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame, - &got_frame, &packet); - if (ret < 0) { - av_frame_free(&frame); - av_log(NULL, AV_LOG_ERROR, "Decoding failed\n"); - break; - } - - if (got_frame) { - frame->pts = av_frame_get_best_effort_timestamp(frame); - ret = filter_encode_write_frame(frame, stream_index); - av_frame_free(&frame); - if (ret < 0) - goto end; - } else { - av_frame_free(&frame); - } - } else { - /* remux this frame without reencoding */ - packet.dts = av_rescale_q_rnd(packet.dts, - ifmt_ctx->streams[stream_index]->time_base, - ofmt_ctx->streams[stream_index]->time_base, - AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - packet.pts = av_rescale_q_rnd(packet.pts, - ifmt_ctx->streams[stream_index]->time_base, - ofmt_ctx->streams[stream_index]->time_base, - AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); - - ret = av_interleaved_write_frame(ofmt_ctx, &packet); - if (ret < 0) - goto end; - } - av_free_packet(&packet); - } - - /* flush filters and encoders */ - for (i = 0; i < ifmt_ctx->nb_streams; i++) { - /* flush filter */ - if (!filter_ctx[i].filter_graph) - continue; - ret = filter_encode_write_frame(NULL, i); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n"); - goto end; - } - - /* flush encoder */ - ret = flush_encoder(i); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n"); - goto end; - } - } - - av_write_trailer(ofmt_ctx); -end: - av_free_packet(&packet); - av_frame_free(&frame); - for (i = 0; i < ifmt_ctx->nb_streams; i++) { - avcodec_close(ifmt_ctx->streams[i]->codec); - if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec) - avcodec_close(ofmt_ctx->streams[i]->codec); - if (filter_ctx && filter_ctx[i].filter_graph) - avfilter_graph_free(&filter_ctx[i].filter_graph); - } - av_free(filter_ctx); - avformat_close_input(&ifmt_ctx); - if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) - avio_close(ofmt_ctx->pb); - avformat_free_context(ofmt_ctx); - - if (ret < 0) - av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret)); - - return ret ? 1 : 0; -} diff --git a/doc/faq.texi b/doc/faq.texi index 74c259f2bb..b400124f69 100644 --- a/doc/faq.texi +++ b/doc/faq.texi @@ -1,8 +1,8 @@ \input texinfo @c -*- texinfo -*- -@settitle FFmpeg FAQ +@settitle Libav FAQ @titlepage -@center @titlefont{FFmpeg FAQ} +@center @titlefont{Libav FAQ} @end titlepage @top @@ -11,23 +11,23 @@ @chapter General Questions -@section Why doesn't FFmpeg support feature [xyz]? +@section Why doesn't Libav support feature [xyz]? -Because no one has taken on that task yet. FFmpeg development is +Because no one has taken on that task yet. Libav development is driven by the tasks that are important to the individual developers. If there is a feature that is important to you, the best way to get it implemented is to undertake the task yourself or sponsor a developer. -@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it? +@section Libav does not support codec XXX. Can you include a Windows DLL loader to support it? No. Windows DLLs are not portable, bloated and often slow. -Moreover FFmpeg strives to support all codecs natively. +Moreover Libav strives to support all codecs natively. A DLL loader is not conducive to that goal. -@section I cannot read this file although this format seems to be supported by ffmpeg. +@section I cannot read this file although this format seems to be supported by avconv. -Even if ffmpeg can read the container format, it may not support all its -codecs. Please consult the supported codec list in the ffmpeg +Even if avconv can read the container format, it may not support all its +codecs. Please consult the supported codec list in the avconv documentation. @section Which codecs are supported by Windows? @@ -79,25 +79,8 @@ not a bug they should fix: Then again, some of them do not know the difference between an undecidable problem and an NP-hard problem... -@section I have installed this library with my distro's package manager. Why does @command{configure} not see it? - -Distributions usually split libraries in several packages. The main package -contains the files necessary to run programs using the library. The -development package contains the files necessary to build programs using the -library. Sometimes, docs and/or data are in a separate package too. - -To build FFmpeg, you need to install the development package. It is usually -called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the -build is finished, but be sure to keep the main package. - @chapter Usage -@section ffmpeg does not work; what is wrong? - -Try a @code{make distclean} in the ffmpeg source directory before the build. -If this does not help see -(@url{http://ffmpeg.org/bugreports.html}). - @section How do I encode single pictures into movies? First, rename your pictures to follow a numerical sequence. @@ -105,21 +88,12 @@ For example, img1.jpg, img2.jpg, img3.jpg,... Then you may run: @example -ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg + avconv -f image2 -i img%d.jpg /tmp/a.mpg @end example Notice that @samp{%d} is replaced by the image number. -@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc. - -Use the @option{-start_number} option to declare a starting number for -the sequence. This is useful if your sequence does not start with -@file{img001.jpg} but is still in a numerical order. The following -example will start with @file{img100.jpg}: - -@example -ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg -@end example +@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc... If you have large number of pictures to rename, you can use the following command to ease the burden. The command, using the bourne @@ -128,7 +102,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of @file{img001.jpg}, @file{img002.jpg} and so on. @example -x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done + x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done @end example If you want to sequence them by oldest modified first, substitute @@ -137,23 +111,17 @@ If you want to sequence them by oldest modified first, substitute Then run: @example -ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg + avconv -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg @end example -The same logic is used for any image format that ffmpeg reads. - -You can also use @command{cat} to pipe images to ffmpeg: - -@example -cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg -@end example +The same logic is used for any image format that avconv reads. @section How do I encode movie to single pictures? Use: @example -ffmpeg -i movie.mpg movie%d.jpg + avconv -i movie.mpg movie%d.jpg @end example The @file{movie.mpg} used as input will be converted to @@ -169,7 +137,7 @@ to force the encoding. Applying that to the previous example: @example -ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg + avconv -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg @end example Beware that there is no "jpeg" codec. Use "mjpeg" instead. @@ -188,12 +156,12 @@ Use @file{-} as file name. Try '-f image2 test%d.jpg'. -@section Why can I not change the frame rate? +@section Why can I not change the framerate? -Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates. +Some codecs, like MPEG-1/2, only allow a small number of fixed framerates. Choose a different codec with the -c:v command line option. -@section How do I encode Xvid or DivX video with ffmpeg? +@section How do I encode Xvid or DivX video with avconv? Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4 standard (note that there are many other coding formats that use this @@ -214,24 +182,24 @@ things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'. but beware the '-g 100' might cause problems with some decoders. Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd. -@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong? +@section Interlaced video looks very bad when encoded with avconv, what is wrong? You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced material, and try '-top 0/1' if the result looks really messed-up. @section How can I read DirectShow files? -If you have built FFmpeg with @code{./configure --enable-avisynth} +If you have built Libav with @code{./configure --enable-avisynth} (only possible on MinGW/Cygwin platforms), then you may use any file that DirectShow can read as input. Just create an "input.avs" text file with this single line ... @example -DirectShowSource("C:\path to your file\yourfile.asf") + DirectShowSource("C:\path to your file\yourfile.asf") @end example -... and then feed that text file to ffmpeg: +... and then feed that text file to avconv: @example -ffmpeg -i input.avs + avconv -i input.avs @end example For ANY other help on AviSynth, please visit the @@ -239,67 +207,8 @@ For ANY other help on AviSynth, please visit the @section How can I join video files? -To "join" video files is quite ambiguous. The following list explains the -different kinds of "joining" and points out how those are addressed in -FFmpeg. To join video files may mean: - -@itemize - -@item -To put them one after the other: this is called to @emph{concatenate} them -(in short: concat) and is addressed -@ref{How can I concatenate video files, in this very faq}. - -@item -To put them together in the same file, to let the user choose between the -different versions (example: different audio languages): this is called to -@emph{multiplex} them together (in short: mux), and is done by simply -invoking ffmpeg with several @option{-i} options. - -@item -For audio, to put all channels together in a single stream (example: two -mono streams into one stereo stream): this is sometimes called to -@emph{merge} them, and can be done using the -@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter. - -@item -For audio, to play one on top of the other: this is called to @emph{mix} -them, and can be done by first merging them into a single stream and then -using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix -the channels at will. - -@item -For video, to display both together, side by side or one on top of a part of -the other; it can be done using the -@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter. - -@end itemize - -@anchor{How can I concatenate video files} -@section How can I concatenate video files? - -There are several solutions, depending on the exact circumstances. - -@subsection Concatenating using the concat @emph{filter} - -FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat, -@code{concat}} filter designed specifically for that, with examples in the -documentation. This operation is recommended if you need to re-encode. - -@subsection Concatenating using the concat @emph{demuxer} - -FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat, -@code{concat}} demuxer which you can use when you want to avoid a re-encode and -your format doesn't support file level concatenation. - -@subsection Concatenating using the concat @emph{protocol} (file level) - -FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat, -@code{concat}} protocol designed specifically for that, with examples in the -documentation. - -A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate -video by merely concatenating the files containing them. +A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by +merely concatenating them. Hence you may concatenate your multimedia files by first transcoding them to these privileged formats, then using the humble @code{cat} command (or the @@ -307,38 +216,27 @@ equally humble @code{copy} under Windows), and finally transcoding back to your format of choice. @example -ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg -ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg +avconv -i input1.avi intermediate1.mpg +avconv -i input2.avi intermediate2.mpg cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg -ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi +avconv -i intermediate_all.mpg output.avi @end example -Additionally, you can use the @code{concat} protocol instead of @code{cat} or -@code{copy} which will avoid creation of a potentially huge intermediate file. +Notice that you should set a reasonably high bitrate for your intermediate and +output files, if you want to preserve video quality. -@example -ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg -ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg -ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg -ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi -@end example - -Note that you may need to escape the character "|" which is special for many -shells. - -Another option is usage of named pipes, should your platform support it: +Also notice that you may avoid the huge intermediate files by taking advantage +of named pipes, should your platform support it: @example mkfifo intermediate1.mpg mkfifo intermediate2.mpg -ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null & -ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null & +avconv -i input1.avi -y intermediate1.mpg < /dev/null & +avconv -i input2.avi -y intermediate2.mpg < /dev/null & cat intermediate1.mpg intermediate2.mpg |\ -ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi +avconv -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi @end example -@subsection Concatenating using raw audio and video - Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also allow concatenation, and the transcoding step is almost lossless. When using multiple yuv4mpegpipe(s), the first line needs to be discarded @@ -346,8 +244,7 @@ from all but the first stream. This can be accomplished by piping through @code{tail} as seen below. Note that when piping through @code{tail} you must use command grouping, @code{@{ ;@}}, to background properly. -For example, let's say we want to concatenate two FLV files into an -output.flv file: +For example, let's say we want to join two FLV files into an output.flv file: @example mkfifo temp1.a @@ -356,80 +253,45 @@ mkfifo temp2.a mkfifo temp2.v mkfifo all.a mkfifo all.v -ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null & -ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null & -ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null & -@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} & +avconv -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null & +avconv -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null & +avconv -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null & +@{ avconv -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} & cat temp1.a temp2.a > all.a & cat temp1.v temp2.v > all.v & -ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \ +avconv -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \ -f yuv4mpegpipe -i all.v \ -y output.flv rm temp[12].[av] all.[av] @end example -@section Using @option{-f lavfi}, audio becomes mono for no apparent reason. +@section -profile option fails when encoding H.264 video with AAC audio -Use @option{-dumpgraph -} to find out exactly where the channel layout is -lost. - -Most likely, it is through @code{auto-inserted aresample}. Try to understand -why the converting filter was needed at that place. - -Just before the output is a likely place, as @option{-f lavfi} currently -only support packed S16. - -Then insert the correct @code{aformat} explicitly in the filtergraph, -specifying the exact format. +@command{avconv} prints an error like @example -aformat=sample_fmts=s16:channel_layouts=stereo +Undefined constant or missing '(' in 'baseline' +Unable to parse option value "baseline" +Error setting option profile to value baseline. @end example -@section Why does FFmpeg not see the subtitles in my VOB file? +Short answer: write @option{-profile:v} instead of @option{-profile}. -VOB and a few other formats do not have a global header that describes -everything present in the file. Instead, applications are supposed to scan -the file to see what it contains. Since VOB files are frequently large, only -the beginning is scanned. If the subtitles happen only later in the file, -they will not be initally detected. +Long answer: this happens because the @option{-profile} option can apply to both +video and audio. Specifically the AAC encoder also defines some profiles, none +of which are named @var{baseline}. -Some applications, including the @code{ffmpeg} command-line tool, can only -work with streams that were detected during the initial scan; streams that -are detected later are ignored. - -The size of the initial scan is controlled by two options: @code{probesize} -(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For -the subtitle stream to be detected, both values must be large enough. - -@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead? - -The @option{-sameq} option meant "same quantizer", and made sense only in a -very limited set of cases. Unfortunately, a lot of people mistook it for -"same quality" and used it in places where it did not make sense: it had -roughly the expected visible effect, but achieved it in a very inefficient -way. - -Each encoder has its own set of options to set the quality-vs-size balance, -use the options for the encoder you are using to set the quality level to a -point acceptable for your tastes. The most common options to do that are -@option{-qscale} and @option{-qmax}, but you should peruse the documentation -of the encoder you chose. +The solution is to apply the @option{-profile} option to the video stream only +by using @url{http://libav.org/avconv.html#Stream-specifiers-1, Stream specifiers}. +Appending @code{:v} to it will do exactly that. @chapter Development -@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat? +@section Are there examples illustrating how to use the Libav libraries, particularly libavcodec and libavformat? -Yes. Check the @file{doc/examples} directory in the source -repository, also available online at: -@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}. - -Examples are also installed by default, usually in -@code{$PREFIX/share/ffmpeg/examples}. - -Also you may read the Developers Guide of the FFmpeg documentation. Alternatively, +Yes. Read the Developers Guide of the Libav documentation. Alternatively, examine the source code for one of the many open source projects that -already incorporate FFmpeg at (@url{projects.html}). +already incorporate Libav at (@url{projects.html}). @section Can you support my C compiler XXX? @@ -440,86 +302,42 @@ with @code{#ifdef}s related to the compiler. @section Is Microsoft Visual C++ supported? Yes. Please see the @uref{platform.html, Microsoft Visual C++} -section in the FFmpeg documentation. +section in the Libav documentation. @section Can you add automake, libtool or autoconf support? No. These tools are too bloated and they complicate the build. -@section Why not rewrite FFmpeg in object-oriented C++? +@section Why not rewrite Libav in object-oriented C++? -FFmpeg is already organized in a highly modular manner and does not need to +Libav is already organized in a highly modular manner and does not need to be rewritten in a formal object language. Further, many of the developers favor straight C; it works for them. For more arguments on this matter, read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}. -@section Why are the ffmpeg programs devoid of debugging symbols? - -The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which -contain full debug information. Those binaries are stripped to create -@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use -the *_g versions. - @section I do not like the LGPL, can I contribute code under the GPL instead? Yes, as long as the code is optional and can easily and cleanly be placed -under #if CONFIG_GPL without breaking anything. So, for example, a new codec +under #if CONFIG_GPL without breaking anything. So for example a new codec or filter would be OK under GPL while a bug fix to LGPL code would not. -@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves. +@section I'm using Libav from within my C++ application but the linker complains about missing symbols which seem to be available. -FFmpeg builds static libraries by default. In static libraries, dependencies -are not handled. That has two consequences. First, you must specify the -libraries in dependency order: @code{-lavdevice} must come before -@code{-lavformat}, @code{-lavutil} must come after everything else, etc. -Second, external libraries that are used in FFmpeg have to be specified too. - -An easy way to get the full list of required libraries in dependency order -is to use @code{pkg-config}. - -@example -c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec) -@end example - -See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for -more details. - -@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available. - -FFmpeg is a pure C project, so to use the libraries within your C++ application +Libav is a pure C project, so to use the libraries within your C++ application you need to explicitly state that you are using a C library. You can do this by -encompassing your FFmpeg includes using @code{extern "C"}. +encompassing your Libav includes using @code{extern "C"}. See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3} @section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope -FFmpeg is a pure C project using C99 math features, in order to enable C++ +Libav is a pure C project using C99 math features, in order to enable C++ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS @section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat? You have to create a custom AVIOContext using @code{avio_alloc_context}, -see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources. - -@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm? - -see @url{http://www.ffmpeg.org/~michael/} - -@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec? - -Even if peculiar since it is network oriented, RTP is a container like any -other. You have to @emph{demux} RTP before feeding the payload to libavcodec. -In this specific case please look at RFC 4629 to see how it should be done. - -@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate. - -@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate -that can accurately represent all timestamps. So no, it is not -wrong if it is larger than the average! -For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate} -will be 150 (it is the least common multiple). -If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}. +see @file{libavformat/aviobuf.c} in Libav and @file{libmpdemux/demux_lavf.c} in MPlayer2 sources. @section Why is @code{make fate} not running all tests? diff --git a/doc/fate.texi b/doc/fate.texi index 4e5cbd7d0f..0185d87de7 100644 --- a/doc/fate.texi +++ b/doc/fate.texi @@ -1,177 +1,83 @@ \input texinfo @c -*- texinfo -*- -@settitle FFmpeg Automated Testing Environment +@settitle FATE Automated Testing Environment @titlepage -@center @titlefont{FFmpeg Automated Testing Environment} +@center @titlefont{FATE Automated Testing Environment} @end titlepage -@node Top @top @contents @chapter Introduction - FATE is an extended regression suite on the client-side and a means -for results aggregation and presentation on the server-side. +FATE provides a regression testsuite embedded within the Libav build system. +It can be run locally and optionally configured to send reports to a web +aggregator and viewer @url{http://fate.libav.org}. - The first part of this document explains how you can use FATE from -your FFmpeg source directory to test your ffmpeg binary. The second -part describes how you can run FATE to submit the results to FFmpeg's -FATE server. +It is advised to run FATE before submitting patches to the current codebase +and provide new tests when submitting patches to add additional features. - In any way you can have a look at the publicly viewable FATE results -by visiting this website: +@chapter Running FATE - @url{http://fate.ffmpeg.org/} +@section Samples and References +In order to run, FATE needs a large amount of data (samples and references) +that is provided separately from the actual source distribution. - This is especially recommended for all people contributing source -code to FFmpeg, as it can be seen if some test on some platform broke -with their recent contribution. This usually happens on the platforms -the developers could not test on. - - The second part of this document describes how you can run FATE to -submit your results to FFmpeg's FATE server. If you want to submit your -results be sure to check that your combination of CPU, OS and compiler -is not already listed on the above mentioned website. - - In the third part you can find a comprehensive listing of FATE makefile -targets and variables. - - -@chapter Using FATE from your FFmpeg source directory - - If you want to run FATE on your machine you need to have the samples -in place. You can get the samples via the build target fate-rsync. -Use this command from the top-level source directory: - -@example -make fate-rsync SAMPLES=fate-suite/ -make fate SAMPLES=fate-suite/ -@end example - - The above commands set the samples location by passing a makefile -variable via command line. It is also possible to set the samples -location at source configuration time by invoking configure with -`--samples='. Afterwards you can -invoke the makefile targets without setting the SAMPLES makefile -variable. This is illustrated by the following commands: - -@example -./configure --samples=fate-suite/ -make fate-rsync -make fate -@end example - - Yet another way to tell FATE about the location of the sample -directory is by making sure the environment variable FATE_SAMPLES -contains the path to your samples directory. This can be achieved -by e.g. putting that variable in your shell profile or by setting -it in your interactive session. - -@example -FATE_SAMPLES=fate-suite/ make fate -@end example - -@float NOTE -Do not put a '~' character in the samples path to indicate a home -directory. Because of shell nuances, this will cause FATE to fail. -@end float +To inform the build system about the testsuite location, pass +@option{--samples=} to @command{configure} or set the +@var{SAMPLES} Make variable or the @var{LIBAV_SAMPLES} environment variable +to a suitable value. To use a custom wrapper to run the test, pass @option{--target-exec} to @command{configure} or set the @var{TARGET_EXEC} Make variable. - -@chapter Submitting the results to the FFmpeg result aggregation server - - To submit your results to the server you should run fate through the -shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs -to be invoked with a configuration file as its first argument. +The dataset is available through @command{rsync}, is possible to fetch +the current sample using the straight rsync command or through a specific +@ref{Makefile target}. @example -tests/fate.sh /path/to/fate_config +# rsync -aL rsync://fate-suite.libav.org/fate-suite/ fate-suite @end example - A configuration file template with comments describing the individual -configuration variables can be found at @file{doc/fate_config.sh.template}. +@example +# make fate-rsync SAMPLES=fate-suite +@end example -@ifhtml - The mentioned configuration template is also available here: -@verbatiminclude fate_config.sh.template -@end ifhtml - Create a configuration that suits your needs, based on the configuration -template. The `slot' configuration variable can be any string that is not -yet used, but it is suggested that you name it adhering to the following -pattern ---. The configuration file -itself will be sourced in a shell script, therefore all shell features may -be used. This enables you to setup the environment as you need it for your -build. +@chapter Manual Run +FATE regression test can be run through @command{make}. +Specific Makefile targets and Makefile variables are available: - For your first test runs the `fate_recv' variable should be empty or -commented out. This will run everything as normal except that it will omit -the submission of the results to the server. The following files should be -present in $workdir as specified in the configuration file: - -@itemize - @item configure.log - @item compile.log - @item test.log - @item report - @item version -@end itemize - - When you have everything working properly you can create an SSH key pair -and send the public key to the FATE server administrator who can be contacted -at the email address @email{fate-admin@@ffmpeg.org}. - - Configure your SSH client to use public key authentication with that key -when connecting to the FATE server. Also do not forget to check the identity -of the server and to accept its host key. This can usually be achieved by -running your SSH client manually and killing it after you accepted the key. -The FATE server's fingerprint is: +@anchor{Makefile target} +@section FATE Makefile targets @table @option -@item RSA - d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51 -@item ECDSA - 76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86 -@end table - - If you have problems connecting to the FATE server, it may help to try out -the @command{ssh} command with one or more @option{-v} options. You should -get detailed output concerning your SSH configuration and the authentication -process. - - The only thing left is to automate the execution of the fate.sh script and -the synchronisation of the samples directory. - - -@chapter FATE makefile targets and variables - -@section Makefile targets - -@table @option -@item fate-rsync -Download/synchronize sample files to the configured samples directory. - @item fate-list -Will list all fate/regression test targets. +List all fate/regression test targets. + +@item fate-rsync +Shortcut to download the fate test samples to the specified testsuite location. @item fate Run the FATE test suite (requires the fate-suite dataset). @end table -@section Makefile variables - +@section FATE Makefile variables @table @option @item V Verbosity level, can be set to 0, 1 or 2. - @itemize - @item 0: show just the test arguments - @item 1: show just the command used in the test - @item 2: show everything - @end itemize + +@table @option +@item 0 +show just the test arguments + +@item 1 +show just the command used in the test + +@item 2 +show everything +@end table @item SAMPLES Specify or override the path to the FATE samples at make time, it has a @@ -186,20 +92,77 @@ Specify which threading strategy test, either @var{slice} or @var{frame}, by default @var{slice+frame} @item CPUFLAGS -Specify CPU flags. +Specify a mask to be applied to autodetected CPU flags. @item TARGET_EXEC Specify or override the wrapper used to run the tests. -The @var{TARGET_EXEC} option provides a way to run FATE wrapped in -@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets -through @command{ssh}. @item GEN Set to @var{1} to generate the missing or mismatched references. @end table -@section Examples +@example + make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate +@end example + +@chapter Automated Tests +In order to automatically testing specific configurations, e.g. multiple +compilers, @command{tests/fate.sh} is provided. + +This shell script builds Libav, runs the regression tests and prepares +a report that can be sent to @url{http://fate.libav.org/} or directly +examined locally. + +@section Testing Profiles +The configuration file passed to @command{fate.sh} is shell scripts as well. + +It must provide at least a @var{slot} identifier, the @var{repo} from +which fetch the sources, the @var{samples} directory, a @var{workdir} with +enough space to build and run all the tests. +Optional submit command @var{fate_recv} and a @var{comment} to describe +the testing profile are available. + +Additional optional parameter to tune the Libav building and reporting process +can be passed. @example -make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate +slot= # some unique identifier +repo=git://git.libav.org/libav.git # the source repository +samples=/path/to/fate/samples +workdir= # directory in which to do all the work +fate_recv="ssh -T fate@@fate.libav.org" # command to submit report +comment= # optional description +build_only= # set to "yes" for a compile-only instance that skips tests + +# the following are optional and map to configure options +arch= +cpu= +cross_prefix= +as= +cc= +ld= +target_os= +sysroot= +target_exec= +target_path= +target_samples= +extra_cflags= +extra_ldflags= +extra_libs= +extra_conf= # extra configure options not covered above + +#make= # name of GNU make if not 'make' +makeopts= # extra options passed to 'make' +#tar= # command to create a tar archive from its arguments on + # stdout, defaults to 'tar c' @end example + +@section Special Instances +The @var{TARGET_EXEC} option provides a way to run FATE wrapped in +@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets +through @command{ssh}. + +@section Submitting Reports +In order to send reports you need to create an @command{ssh} key and send it +to @email{root@@libav.org}. +The current server fingerprint is @var{a4:99:d7:d3:1c:92:0d:56:d6:d5:61:be:01:ae:7d:e6} diff --git a/doc/fate_config.sh.template b/doc/fate_config.sh.template deleted file mode 100644 index 1487c1d2ef..0000000000 --- a/doc/fate_config.sh.template +++ /dev/null @@ -1,29 +0,0 @@ -slot= # some unique identifier -repo=git://source.ffmpeg.org/ffmpeg.git # the source repository -samples= # path to samples directory -workdir= # directory in which to do all the work -#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report -comment= # optional description -build_only= # set to "yes" for a compile-only instance that skips tests - -# the following are optional and map to configure options -arch= -cpu= -cross_prefix= -as= -cc= -ld= -target_os= -sysroot= -target_exec= -target_path= -target_samples= -extra_cflags= -extra_ldflags= -extra_libs= -extra_conf= # extra configure options not covered above - -#make= # name of GNU make if not 'make' -makeopts= # extra options passed to 'make' -#tar= # command to create a tar archive from its arguments on stdout, - # defaults to 'tar c' diff --git a/doc/ffmpeg-bitstream-filters.texi b/doc/ffmpeg-bitstream-filters.texi deleted file mode 100644 index e33e005024..0000000000 --- a/doc/ffmpeg-bitstream-filters.texi +++ /dev/null @@ -1,45 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Bitstream Filters Documentation -@titlepage -@center @titlefont{FFmpeg Bitstream Filters Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -This document describes the bitstream filters provided by the -libavcodec library. - -A bitstream filter operates on the encoded stream data, and performs -bitstream level modifications without performing decoding. - -@c man end DESCRIPTION - -@include bitstream_filters.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libavcodec.html,libavcodec} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-bitstream-filters -@settitle FFmpeg bitstream filters - -@end ignore - -@bye diff --git a/doc/ffmpeg-codecs.texi b/doc/ffmpeg-codecs.texi deleted file mode 100644 index 6f8f5a34d9..0000000000 --- a/doc/ffmpeg-codecs.texi +++ /dev/null @@ -1,42 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Codecs Documentation -@titlepage -@center @titlefont{FFmpeg Codecs Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -This document describes the codecs (decoders and encoders) provided by -the libavcodec library. - -@c man end DESCRIPTION - -@include codecs.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libavcodec.html,libavcodec} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-codecs -@settitle FFmpeg codecs - -@end ignore - -@bye diff --git a/doc/ffmpeg-devices.texi b/doc/ffmpeg-devices.texi deleted file mode 100644 index b44bd7285b..0000000000 --- a/doc/ffmpeg-devices.texi +++ /dev/null @@ -1,42 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Devices Documentation -@titlepage -@center @titlefont{FFmpeg Devices Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -This document describes the input and output devices provided by the -libavdevice library. - -@c man end DESCRIPTION - -@include devices.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libavdevice.html,libavdevice} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-devices -@settitle FFmpeg devices - -@end ignore - -@bye diff --git a/doc/ffmpeg-filters.texi b/doc/ffmpeg-filters.texi deleted file mode 100644 index bb920ce26c..0000000000 --- a/doc/ffmpeg-filters.texi +++ /dev/null @@ -1,42 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Filters Documentation -@titlepage -@center @titlefont{FFmpeg Filters Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -This document describes filters, sources, and sinks provided by the -libavfilter library. - -@c man end DESCRIPTION - -@include filters.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libavfilter.html,libavfilter} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-filters -@settitle FFmpeg filters - -@end ignore - -@bye diff --git a/doc/ffmpeg-formats.texi b/doc/ffmpeg-formats.texi deleted file mode 100644 index e205caabeb..0000000000 --- a/doc/ffmpeg-formats.texi +++ /dev/null @@ -1,42 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Formats Documentation -@titlepage -@center @titlefont{FFmpeg Formats Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -This document describes the supported formats (muxers and demuxers) -provided by the libavformat library. - -@c man end DESCRIPTION - -@include formats.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libavformat.html,libavformat} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-formats -@settitle FFmpeg formats - -@end ignore - -@bye diff --git a/doc/ffmpeg-protocols.texi b/doc/ffmpeg-protocols.texi deleted file mode 100644 index d992e7571a..0000000000 --- a/doc/ffmpeg-protocols.texi +++ /dev/null @@ -1,42 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Protocols Documentation -@titlepage -@center @titlefont{FFmpeg Protocols Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -This document describes the input and output protocols provided by the -libavformat library. - -@c man end DESCRIPTION - -@include protocols.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libavformat.html,libavformat} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-protocols -@settitle FFmpeg protocols - -@end ignore - -@bye diff --git a/doc/ffmpeg-resampler.texi b/doc/ffmpeg-resampler.texi deleted file mode 100644 index 5bef78967b..0000000000 --- a/doc/ffmpeg-resampler.texi +++ /dev/null @@ -1,44 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Resampler Documentation -@titlepage -@center @titlefont{FFmpeg Resampler Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The FFmpeg resampler provides a high-level interface to the -libswresample library audio resampling utilities. In particular it -allows one to perform audio resampling, audio channel layout rematrixing, -and convert audio format and packing layout. - -@c man end DESCRIPTION - -@include resampler.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libswresample.html,libswresample} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-resampler -@settitle FFmpeg Resampler - -@end ignore - -@bye diff --git a/doc/ffmpeg-scaler.texi b/doc/ffmpeg-scaler.texi deleted file mode 100644 index c4ddc0972a..0000000000 --- a/doc/ffmpeg-scaler.texi +++ /dev/null @@ -1,43 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Scaler Documentation -@titlepage -@center @titlefont{FFmpeg Scaler Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The FFmpeg rescaler provides a high-level interface to the libswscale -library image conversion utilities. In particular it allows one to perform -image rescaling and pixel format conversion. - -@c man end DESCRIPTION - -@include scaler.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libswscale.html,libswscale} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-scaler -@settitle FFmpeg video scaling and pixel format converter - -@end ignore - -@bye diff --git a/doc/ffmpeg-utils.texi b/doc/ffmpeg-utils.texi deleted file mode 100644 index 581e2eab81..0000000000 --- a/doc/ffmpeg-utils.texi +++ /dev/null @@ -1,42 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle FFmpeg Utilities Documentation -@titlepage -@center @titlefont{FFmpeg Utilities Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -This document describes some generic features and utilities provided -by the libavutil library. - -@c man end DESCRIPTION - -@include utils.texi - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{libavutil.html,libavutil} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffmpeg-utils -@settitle FFmpeg utilities - -@end ignore - -@bye diff --git a/doc/ffmpeg.txt b/doc/ffmpeg.txt deleted file mode 100644 index a028ca23d2..0000000000 --- a/doc/ffmpeg.txt +++ /dev/null @@ -1,47 +0,0 @@ - : - ffmpeg.c : libav* - ======== : ====== - : - : - --------------------------------:---> AVStream... - InputStream input_streams[] / : - / : - InputFile input_files[] +==========================+ / ^ : - ------> 0 | : st ---:-----------:--/ : : - ^ +------+-----------+-----+ / +--------------------------+ : : - : | :ist_index--:-----:---------/ 1 | : st : | : : - : +------+-----------+-----+ +==========================+ : : - nb_input_files : | :ist_index--:-----:------------------> 2 | : st : | : : - : +------+-----------+-----+ +--------------------------+ : nb_input_streams : - : | :ist_index : | 3 | ... | : : - v +------+-----------+-----+ +--------------------------+ : : - --> 4 | | : : - | +--------------------------+ : : - | 5 | | : : - | +==========================+ v : - | : - | : - | : - | : - --------- --------------------------------:---> AVStream... - \ / : - OutputStream output_streams[] / : - \ / : - +======\======================/======+ ^ : - ------> 0 | : source_index : st-:--- | : : - OutputFile output_files[] / +------------------------------------+ : : - / 1 | : : : | : : - ^ +------+------------+-----+ / +------------------------------------+ : : - : | : ost_index -:-----:------/ 2 | : : : | : : - nb_output_files : +------+------------+-----+ +====================================+ : : - : | : ost_index -:-----|-----------------> 3 | : : : | : : - : +------+------------+-----+ +------------------------------------+ : nb_output_streams : - : | : : | 4 | | : : - : +------+------------+-----+ +------------------------------------+ : : - : | : : | 5 | | : : - v +------+------------+-----+ +------------------------------------+ : : - 6 | | : : - +------------------------------------+ : : - 7 | | : : - +====================================+ v : - : diff --git a/doc/ffplay.texi b/doc/ffplay.texi deleted file mode 100644 index 203085c16d..0000000000 --- a/doc/ffplay.texi +++ /dev/null @@ -1,293 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle ffplay Documentation -@titlepage -@center @titlefont{ffplay Documentation} -@end titlepage - -@top - -@contents - -@chapter Synopsis - -ffplay [@var{options}] [@file{input_file}] - -@chapter Description -@c man begin DESCRIPTION - -FFplay is a very simple and portable media player using the FFmpeg -libraries and the SDL library. It is mostly used as a testbed for the -various FFmpeg APIs. -@c man end - -@chapter Options -@c man begin OPTIONS - -@include fftools-common-opts.texi - -@section Main options - -@table @option -@item -x @var{width} -Force displayed width. -@item -y @var{height} -Force displayed height. -@item -s @var{size} -Set frame size (WxH or abbreviation), needed for videos which do -not contain a header with the frame size like raw YUV. This option -has been deprecated in favor of private options, try -video_size. -@item -an -Disable audio. -@item -vn -Disable video. -@item -ss @var{pos} -Seek to a given position in seconds. -@item -t @var{duration} -play seconds of audio/video -@item -bytes -Seek by bytes. -@item -nodisp -Disable graphical display. -@item -f @var{fmt} -Force format. -@item -window_title @var{title} -Set window title (default is the input filename). -@item -loop @var{number} -Loops movie playback times. 0 means forever. -@item -showmode @var{mode} -Set the show mode to use. -Available values for @var{mode} are: -@table @samp -@item 0, video -show video -@item 1, waves -show audio waves -@item 2, rdft -show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform) -@end table - -Default value is "video", if video is not present or cannot be played -"rdft" is automatically selected. - -You can interactively cycle through the available show modes by -pressing the key @key{w}. - -@item -vf @var{filtergraph} -Create the filtergraph specified by @var{filtergraph} and use it to -filter the video stream. - -@var{filtergraph} is a description of the filtergraph to apply to -the stream, and must have a single video input and a single video -output. In the filtergraph, the input is associated to the label -@code{in}, and the output to the label @code{out}. See the -ffmpeg-filters manual for more information about the filtergraph -syntax. - -You can specify this parameter multiple times and cycle through the specified -filtergraphs along with the show modes by pressing the key @key{w}. - -@item -af @var{filtergraph} -@var{filtergraph} is a description of the filtergraph to apply to -the input audio. -Use the option "-filters" to show all the available filters (including -sources and sinks). - -@item -i @var{input_file} -Read @var{input_file}. -@end table - -@section Advanced options -@table @option -@item -pix_fmt @var{format} -Set pixel format. -This option has been deprecated in favor of private options, try -pixel_format. - -@item -stats -Print several playback statistics, in particular show the stream -duration, the codec parameters, the current position in the stream and -the audio/video synchronisation drift. It is on by default, to -explicitly disable it you need to specify @code{-nostats}. - -@item -bug -Work around bugs. -@item -fast -Non-spec-compliant optimizations. -@item -genpts -Generate pts. -@item -rtp_tcp -Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful -if you are streaming with the RTSP protocol. -@item -sync @var{type} -Set the master clock to audio (@code{type=audio}), video -(@code{type=video}) or external (@code{type=ext}). Default is audio. The -master clock is used to control audio-video synchronization. Most media -players use audio as master clock, but in some cases (streaming or high -quality broadcast) it is necessary to change that. This option is mainly -used for debugging purposes. -@item -threads @var{count} -Set the thread count. -@item -ast @var{audio_stream_number} -Select the desired audio stream number, counting from 0. The number -refers to the list of all the input audio streams. If it is greater -than the number of audio streams minus one, then the last one is -selected, if it is negative the audio playback is disabled. -@item -vst @var{video_stream_number} -Select the desired video stream number, counting from 0. The number -refers to the list of all the input video streams. If it is greater -than the number of video streams minus one, then the last one is -selected, if it is negative the video playback is disabled. -@item -sst @var{subtitle_stream_number} -Select the desired subtitle stream number, counting from 0. The number -refers to the list of all the input subtitle streams. If it is greater -than the number of subtitle streams minus one, then the last one is -selected, if it is negative the subtitle rendering is disabled. -@item -autoexit -Exit when video is done playing. -@item -exitonkeydown -Exit if any key is pressed. -@item -exitonmousedown -Exit if any mouse button is pressed. - -@item -codec:@var{media_specifier} @var{codec_name} -Force a specific decoder implementation for the stream identified by -@var{media_specifier}, which can assume the values @code{a} (audio), -@code{v} (video), and @code{s} subtitle. - -@item -acodec @var{codec_name} -Force a specific audio decoder. - -@item -vcodec @var{codec_name} -Force a specific video decoder. - -@item -scodec @var{codec_name} -Force a specific subtitle decoder. - -@item -autorotate -Automatically rotate the video according to presentation metadata. Set by -default, use -noautorotate to disable. -@end table - -@section While playing - -@table @key -@item q, ESC -Quit. - -@item f -Toggle full screen. - -@item p, SPC -Pause. - -@item a -Cycle audio channel in the current program. - -@item v -Cycle video channel. - -@item t -Cycle subtitle channel in the current program. - -@item c -Cycle program. - -@item w -Cycle video filters or show modes. - -@item s -Step to the next frame. - -Pause if the stream is not already paused, step to the next video -frame, and pause. - -@item left/right -Seek backward/forward 10 seconds. - -@item down/up -Seek backward/forward 1 minute. - -@item page down/page up -Seek to the previous/next chapter. -or if there are no chapters -Seek backward/forward 10 minutes. - -@item mouse click -Seek to percentage in file corresponding to fraction of width. - -@end table - -@c man end - -@include config.texi -@ifset config-all -@set config-readonly -@ifset config-avutil -@include utils.texi -@end ifset -@ifset config-avcodec -@include codecs.texi -@include bitstream_filters.texi -@end ifset -@ifset config-avformat -@include formats.texi -@include protocols.texi -@end ifset -@ifset config-avdevice -@include devices.texi -@end ifset -@ifset config-swresample -@include resampler.texi -@end ifset -@ifset config-swscale -@include scaler.texi -@end ifset -@ifset config-avfilter -@include filters.texi -@end ifset -@end ifset - -@chapter See Also - -@ifhtml -@ifset config-all -@url{ffplay.html,ffplay}, -@end ifset -@ifset config-not-all -@url{ffplay-all.html,ffmpeg-all}, -@end ifset -@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-utils.html,ffmpeg-utils}, -@url{ffmpeg-scaler.html,ffmpeg-scaler}, -@url{ffmpeg-resampler.html,ffmpeg-resampler}, -@url{ffmpeg-codecs.html,ffmpeg-codecs}, -@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, -@url{ffmpeg-formats.html,ffmpeg-formats}, -@url{ffmpeg-devices.html,ffmpeg-devices}, -@url{ffmpeg-protocols.html,ffmpeg-protocols}, -@url{ffmpeg-filters.html,ffmpeg-filters} -@end ifhtml - -@ifnothtml -@ifset config-all -ffplay(1), -@end ifset -@ifset config-not-all -ffplay-all(1), -@end ifset -ffmpeg(1), ffprobe(1), ffserver(1), -ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), -ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), -ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffplay -@settitle FFplay media player - -@end ignore - -@bye diff --git a/doc/ffprobe.texi b/doc/ffprobe.texi deleted file mode 100644 index 8de8956b75..0000000000 --- a/doc/ffprobe.texi +++ /dev/null @@ -1,671 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle ffprobe Documentation -@titlepage -@center @titlefont{ffprobe Documentation} -@end titlepage - -@top - -@contents - -@chapter Synopsis - -ffprobe [@var{options}] [@file{input_file}] - -@chapter Description -@c man begin DESCRIPTION - -ffprobe gathers information from multimedia streams and prints it in -human- and machine-readable fashion. - -For example it can be used to check the format of the container used -by a multimedia stream and the format and type of each media stream -contained in it. - -If a filename is specified in input, ffprobe will try to open and -probe the file content. If the file cannot be opened or recognized as -a multimedia file, a positive exit code is returned. - -ffprobe may be employed both as a standalone application or in -combination with a textual filter, which may perform more -sophisticated processing, e.g. statistical processing or plotting. - -Options are used to list some of the formats supported by ffprobe or -for specifying which information to display, and for setting how -ffprobe will show it. - -ffprobe output is designed to be easily parsable by a textual filter, -and consists of one or more sections of a form defined by the selected -writer, which is specified by the @option{print_format} option. - -Sections may contain other nested sections, and are identified by a -name (which may be shared by other sections), and an unique -name. See the output of @option{sections}. - -Metadata tags stored in the container or in the streams are recognized -and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM" -section. - -@c man end - -@chapter Options -@c man begin OPTIONS - -@include fftools-common-opts.texi - -@section Main options - -@table @option - -@item -f @var{format} -Force format to use. - -@item -unit -Show the unit of the displayed values. - -@item -prefix -Use SI prefixes for the displayed values. -Unless the "-byte_binary_prefix" option is used all the prefixes -are decimal. - -@item -byte_binary_prefix -Force the use of binary prefixes for byte values. - -@item -sexagesimal -Use sexagesimal format HH:MM:SS.MICROSECONDS for time values. - -@item -pretty -Prettify the format of the displayed values, it corresponds to the -options "-unit -prefix -byte_binary_prefix -sexagesimal". - -@item -of, -print_format @var{writer_name}[=@var{writer_options}] -Set the output printing format. - -@var{writer_name} specifies the name of the writer, and -@var{writer_options} specifies the options to be passed to the writer. - -For example for printing the output in JSON format, specify: -@example --print_format json -@end example - -For more details on the available output printing formats, see the -Writers section below. - -@item -sections -Print sections structure and section information, and exit. The output -is not meant to be parsed by a machine. - -@item -select_streams @var{stream_specifier} -Select only the streams specified by @var{stream_specifier}. This -option affects only the options related to streams -(e.g. @code{show_streams}, @code{show_packets}, etc.). - -For example to show only audio streams, you can use the command: -@example -ffprobe -show_streams -select_streams a INPUT -@end example - -To show only video packets belonging to the video stream with index 1: -@example -ffprobe -show_packets -select_streams v:1 INPUT -@end example - -@item -show_data -Show payload data, as a hexadecimal and ASCII dump. Coupled with -@option{-show_packets}, it will dump the packets' data. Coupled with -@option{-show_streams}, it will dump the codec extradata. - -The dump is printed as the "data" field. It may contain newlines. - -@item -show_error -Show information about the error found when trying to probe the input. - -The error information is printed within a section with name "ERROR". - -@item -show_format -Show information about the container format of the input multimedia -stream. - -All the container format information is printed within a section with -name "FORMAT". - -@item -show_format_entry @var{name} -Like @option{-show_format}, but only prints the specified entry of the -container format information, rather than all. This option may be given more -than once, then all specified entries will be shown. - -This option is deprecated, use @code{show_entries} instead. - -@item -show_entries @var{section_entries} -Set list of entries to show. - -Entries are specified according to the following -syntax. @var{section_entries} contains a list of section entries -separated by @code{:}. Each section entry is composed by a section -name (or unique name), optionally followed by a list of entries local -to that section, separated by @code{,}. - -If section name is specified but is followed by no @code{=}, all -entries are printed to output, together with all the contained -sections. Otherwise only the entries specified in the local section -entries list are printed. In particular, if @code{=} is specified but -the list of local entries is empty, then no entries will be shown for -that section. - -Note that the order of specification of the local section entries is -not honored in the output, and the usual display order will be -retained. - -The formal syntax is given by: -@example -@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}] -@var{SECTION_ENTRY} ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]] -@var{SECTION_ENTRIES} ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}] -@end example - -For example, to show only the index and type of each stream, and the PTS -time, duration time, and stream index of the packets, you can specify -the argument: -@example -packet=pts_time,duration_time,stream_index : stream=index,codec_type -@end example - -To show all the entries in the section "format", but only the codec -type in the section "stream", specify the argument: -@example -format : stream=codec_type -@end example - -To show all the tags in the stream and format sections: -@example -format_tags : format_tags -@end example - -To show only the @code{title} tag (if available) in the stream -sections: -@example -stream_tags=title -@end example - -@item -show_packets -Show information about each packet contained in the input multimedia -stream. - -The information for each single packet is printed within a dedicated -section with name "PACKET". - -@item -show_frames -Show information about each frame and subtitle contained in the input -multimedia stream. - -The information for each single frame is printed within a dedicated -section with name "FRAME" or "SUBTITLE". - -@item -show_streams -Show information about each media stream contained in the input -multimedia stream. - -Each media stream information is printed within a dedicated section -with name "STREAM". - -@item -show_programs -Show information about programs and their streams contained in the input -multimedia stream. - -Each media stream information is printed within a dedicated section -with name "PROGRAM_STREAM". - -@item -show_chapters -Show information about chapters stored in the format. - -Each chapter is printed within a dedicated section with name "CHAPTER". - -@item -count_frames -Count the number of frames per stream and report it in the -corresponding stream section. - -@item -count_packets -Count the number of packets per stream and report it in the -corresponding stream section. - -@item -read_intervals @var{read_intervals} - -Read only the specified intervals. @var{read_intervals} must be a -sequence of interval specifications separated by ",". -@command{ffprobe} will seek to the interval starting point, and will -continue reading from that. - -Each interval is specified by two optional parts, separated by "%". - -The first part specifies the interval start position. It is -interpreted as an abolute position, or as a relative offset from the -current position if it is preceded by the "+" character. If this first -part is not specified, no seeking will be performed when reading this -interval. - -The second part specifies the interval end position. It is interpreted -as an absolute position, or as a relative offset from the current -position if it is preceded by the "+" character. If the offset -specification starts with "#", it is interpreted as the number of -packets to read (not including the flushing packets) from the interval -start. If no second part is specified, the program will read until the -end of the input. - -Note that seeking is not accurate, thus the actual interval start -point may be different from the specified position. Also, when an -interval duration is specified, the absolute end time will be computed -by adding the duration to the interval start point found by seeking -the file, rather than to the specified start value. - -The formal syntax is given by: -@example -@var{INTERVAL} ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]] -@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}] -@end example - -A few examples follow. -@itemize -@item -Seek to time 10, read packets until 20 seconds after the found seek -point, then seek to position @code{01:30} (1 minute and thirty -seconds) and read packets until position @code{01:45}. -@example -10%+20,01:30%01:45 -@end example - -@item -Read only 42 packets after seeking to position @code{01:23}: -@example -01:23%+#42 -@end example - -@item -Read only the first 20 seconds from the start: -@example -%+20 -@end example - -@item -Read from the start until position @code{02:30}: -@example -%02:30 -@end example -@end itemize - -@item -show_private_data, -private -Show private data, that is data depending on the format of the -particular shown element. -This option is enabled by default, but you may need to disable it -for specific uses, for example when creating XSD-compliant XML output. - -@item -show_program_version -Show information related to program version. - -Version information is printed within a section with name -"PROGRAM_VERSION". - -@item -show_library_versions -Show information related to library versions. - -Version information for each library is printed within a section with -name "LIBRARY_VERSION". - -@item -show_versions -Show information related to program and library versions. This is the -equivalent of setting both @option{-show_program_version} and -@option{-show_library_versions} options. - -@item -bitexact -Force bitexact output, useful to produce output which is not dependent -on the specific build. - -@item -i @var{input_file} -Read @var{input_file}. - -@end table -@c man end - -@chapter Writers -@c man begin WRITERS - -A writer defines the output format adopted by @command{ffprobe}, and will be -used for printing all the parts of the output. - -A writer may accept one or more arguments, which specify the options -to adopt. The options are specified as a list of @var{key}=@var{value} -pairs, separated by ":". - -All writers support the following options: - -@table @option -@item string_validation, sv -Set string validation mode. - -The following values are accepted. -@table @samp -@item fail -The writer will fail immediately in case an invalid string (UTF-8) -sequence or code point is found in the input. This is especially -useful to validate input metadata. - -@item ignore -Any validation error will be ignored. This will result in possibly -broken output, especially with the json or xml writer. - -@item replace -The writer will substitute invalid UTF-8 sequences or code points with -the string specified with the @option{string_validation_replacement}. -@end table - -Default value is @samp{replace}. - -@item string_validation_replacement, svr -Set replacement string to use in case @option{string_validation} is -set to @samp{replace}. - -In case the option is not specified, the writer will assume the empty -string, that is it will remove the invalid sequences from the input -strings. -@end table - -A description of the currently available writers follows. - -@section default -Default format. - -Print each section in the form: -@example -[SECTION] -key1=val1 -... -keyN=valN -[/SECTION] -@end example - -Metadata tags are printed as a line in the corresponding FORMAT, STREAM or -PROGRAM_STREAM section, and are prefixed by the string "TAG:". - -A description of the accepted options follows. - -@table @option - -@item nokey, nk -If set to 1 specify not to print the key of each field. Default value -is 0. - -@item noprint_wrappers, nw -If set to 1 specify not to print the section header and footer. -Default value is 0. -@end table - -@section compact, csv -Compact and CSV format. - -The @code{csv} writer is equivalent to @code{compact}, but supports -different defaults. - -Each section is printed on a single line. -If no option is specifid, the output has the form: -@example -section|key1=val1| ... |keyN=valN -@end example - -Metadata tags are printed in the corresponding "format" or "stream" -section. A metadata tag key, if printed, is prefixed by the string -"tag:". - -The description of the accepted options follows. - -@table @option - -@item item_sep, s -Specify the character to use for separating fields in the output line. -It must be a single printable character, it is "|" by default ("," for -the @code{csv} writer). - -@item nokey, nk -If set to 1 specify not to print the key of each field. Its default -value is 0 (1 for the @code{csv} writer). - -@item escape, e -Set the escape mode to use, default to "c" ("csv" for the @code{csv} -writer). - -It can assume one of the following values: -@table @option -@item c -Perform C-like escaping. Strings containing a newline ('\n'), carriage -return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping -character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned -escaping, so that a newline is converted to the sequence "\n", a -carriage return to "\r", '\' to "\\" and the separator @var{SEP} is -converted to "\@var{SEP}". - -@item csv -Perform CSV-like escaping, as described in RFC4180. Strings -containing a newline ('\n'), a carriage return ('\r'), a double quote -('"'), or @var{SEP} are enclosed in double-quotes. - -@item none -Perform no escaping. -@end table - -@item print_section, p -Print the section name at the begin of each line if the value is -@code{1}, disable it with value set to @code{0}. Default value is -@code{1}. - -@end table - -@section flat -Flat format. - -A free-form output where each line contains an explicit key=value, such as -"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be -directly embedded in sh scripts as long as the separator character is an -alphanumeric character or an underscore (see @var{sep_char} option). - -The description of the accepted options follows. - -@table @option -@item sep_char, s -Separator character used to separate the chapter, the section name, IDs and -potential tags in the printed field key. - -Default value is '.'. - -@item hierarchical, h -Specify if the section name specification should be hierarchical. If -set to 1, and if there is more than one section in the current -chapter, the section name will be prefixed by the name of the -chapter. A value of 0 will disable this behavior. - -Default value is 1. -@end table - -@section ini -INI format output. - -Print output in an INI based format. - -The following conventions are adopted: - -@itemize -@item -all key and values are UTF-8 -@item -'.' is the subgroup separator -@item -newline, '\t', '\f', '\b' and the following characters are escaped -@item -'\' is the escape character -@item -'#' is the comment indicator -@item -'=' is the key/value separator -@item -':' is not used but usually parsed as key/value separator -@end itemize - -This writer accepts options as a list of @var{key}=@var{value} pairs, -separated by ":". - -The description of the accepted options follows. - -@table @option -@item hierarchical, h -Specify if the section name specification should be hierarchical. If -set to 1, and if there is more than one section in the current -chapter, the section name will be prefixed by the name of the -chapter. A value of 0 will disable this behavior. - -Default value is 1. -@end table - -@section json -JSON based format. - -Each section is printed using JSON notation. - -The description of the accepted options follows. - -@table @option - -@item compact, c -If set to 1 enable compact output, that is each section will be -printed on a single line. Default value is 0. -@end table - -For more information about JSON, see @url{http://www.json.org/}. - -@section xml -XML based format. - -The XML output is described in the XML schema description file -@file{ffprobe.xsd} installed in the FFmpeg datadir. - -An updated version of the schema can be retrieved at the url -@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the -latest schema committed into the FFmpeg development source code tree. - -Note that the output issued will be compliant to the -@file{ffprobe.xsd} schema only when no special global output options -(@option{unit}, @option{prefix}, @option{byte_binary_prefix}, -@option{sexagesimal} etc.) are specified. - -The description of the accepted options follows. - -@table @option - -@item fully_qualified, q -If set to 1 specify if the output should be fully qualified. Default -value is 0. -This is required for generating an XML file which can be validated -through an XSD file. - -@item xsd_compliant, x -If set to 1 perform more checks for ensuring that the output is XSD -compliant. Default value is 0. -This option automatically sets @option{fully_qualified} to 1. -@end table - -For more information about the XML format, see -@url{http://www.w3.org/XML/}. -@c man end WRITERS - -@chapter Timecode -@c man begin TIMECODE - -@command{ffprobe} supports Timecode extraction: - -@itemize - -@item -MPEG1/2 timecode is extracted from the GOP, and is available in the video -stream details (@option{-show_streams}, see @var{timecode}). - -@item -MOV timecode is extracted from tmcd track, so is available in the tmcd -stream metadata (@option{-show_streams}, see @var{TAG:timecode}). - -@item -DV, GXF and AVI timecodes are available in format metadata -(@option{-show_format}, see @var{TAG:timecode}). - -@end itemize -@c man end TIMECODE - -@include config.texi -@ifset config-all -@set config-readonly -@ifset config-avutil -@include utils.texi -@end ifset -@ifset config-avcodec -@include codecs.texi -@include bitstream_filters.texi -@end ifset -@ifset config-avformat -@include formats.texi -@include protocols.texi -@end ifset -@ifset config-avdevice -@include devices.texi -@end ifset -@ifset config-swresample -@include resampler.texi -@end ifset -@ifset config-swscale -@include scaler.texi -@end ifset -@ifset config-avfilter -@include filters.texi -@end ifset -@end ifset - -@chapter See Also - -@ifhtml -@ifset config-all -@url{ffprobe.html,ffprobe}, -@end ifset -@ifset config-not-all -@url{ffprobe-all.html,ffprobe-all}, -@end ifset -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver}, -@url{ffmpeg-utils.html,ffmpeg-utils}, -@url{ffmpeg-scaler.html,ffmpeg-scaler}, -@url{ffmpeg-resampler.html,ffmpeg-resampler}, -@url{ffmpeg-codecs.html,ffmpeg-codecs}, -@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, -@url{ffmpeg-formats.html,ffmpeg-formats}, -@url{ffmpeg-devices.html,ffmpeg-devices}, -@url{ffmpeg-protocols.html,ffmpeg-protocols}, -@url{ffmpeg-filters.html,ffmpeg-filters} -@end ifhtml - -@ifnothtml -@ifset config-all -ffprobe(1), -@end ifset -@ifset config-not-all -ffprobe-all(1), -@end ifset -ffmpeg(1), ffplay(1), ffserver(1), -ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), -ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), -ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffprobe -@settitle ffprobe media prober - -@end ignore - -@bye diff --git a/doc/ffprobe.xsd b/doc/ffprobe.xsd deleted file mode 100644 index 1bc1fb5bb4..0000000000 --- a/doc/ffprobe.xsd +++ /dev/null @@ -1,260 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/doc/ffserver.texi b/doc/ffserver.texi deleted file mode 100644 index fb90dd650d..0000000000 --- a/doc/ffserver.texi +++ /dev/null @@ -1,900 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle ffserver Documentation -@titlepage -@center @titlefont{ffserver Documentation} -@end titlepage - -@top - -@contents - -@chapter Synopsis - -ffserver [@var{options}] - -@chapter Description -@c man begin DESCRIPTION - -@command{ffserver} is a streaming server for both audio and video. -It supports several live feeds, streaming from files and time shifting -on live feeds. You can seek to positions in the past on each live -feed, provided you specify a big enough feed storage. - -@command{ffserver} is configured through a configuration file, which -is read at startup. If not explicitly specified, it will read from -@file{/etc/ffserver.conf}. - -@command{ffserver} receives prerecorded files or FFM streams from some -@command{ffmpeg} instance as input, then streams them over -RTP/RTSP/HTTP. - -An @command{ffserver} instance will listen on some port as specified -in the configuration file. You can launch one or more instances of -@command{ffmpeg} and send one or more FFM streams to the port where -ffserver is expecting to receive them. Alternately, you can make -@command{ffserver} launch such @command{ffmpeg} instances at startup. - -Input streams are called feeds, and each one is specified by a -@code{} section in the configuration file. - -For each feed you can have different output streams in various -formats, each one specified by a @code{} section in the -configuration file. - -@chapter Detailed description - -@command{ffserver} works by forwarding streams encoded by -@command{ffmpeg}, or pre-recorded streams which are read from disk. - -Precisely, @command{ffserver} acts as an HTTP server, accepting POST -requests from @command{ffmpeg} to acquire the stream to publish, and -serving RTSP clients or HTTP clients GET requests with the stream -media content. - -A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to -a port where @command{ffserver} is listening. - -Each feed is identified by a unique name, corresponding to the name -of the resource published on @command{ffserver}, and is configured by -a dedicated @code{Feed} section in the configuration file. - -The feed publish URL is given by: -@example -http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name} -@end example - -where @var{ffserver_ip_address} is the IP address of the machine where -@command{ffserver} is installed, @var{http_port} is the port number of -the HTTP server (configured through the @option{Port} option), and -@var{feed_name} is the name of the corresponding feed defined in the -configuration file. - -Each feed is associated to a file which is stored on disk. This stored -file is used to allow to send pre-recorded data to a player as fast as -possible when new content is added in real-time to the stream. - -A "live-stream" or "stream" is a resource published by -@command{ffserver}, and made accessible through the HTTP protocol to -clients. - -A stream can be connected to a feed, or to a file. In the first case, -the published stream is forwarded from the corresponding feed -generated by a running instance of @command{ffmpeg}, in the second -case the stream is read from a pre-recorded file. - -Each stream is identified by a unique name, corresponding to the name -of the resource served by @command{ffserver}, and is configured by -a dedicated @code{Stream} section in the configuration file. - -The stream access HTTP URL is given by: -@example -http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}] -@end example - -The stream access RTSP URL is given by: -@example -http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}] -@end example - -@var{stream_name} is the name of the corresponding stream defined in -the configuration file. @var{options} is a list of options specified -after the URL which affects how the stream is served by -@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP -and RTSP ports configured with the options @var{Port} and -@var{RTSPPort} respectively. - -In case the stream is associated to a feed, the encoding parameters -must be configured in the stream configuration. They are sent to -@command{ffmpeg} when setting up the encoding. This allows -@command{ffserver} to define the encoding parameters used by -the @command{ffmpeg} encoders. - -The @command{ffmpeg} @option{override_ffserver} commandline option -allows one to override the encoding parameters set by the server. - -Multiple streams can be connected to the same feed. - -For example, you can have a situation described by the following -graph: -@example - _________ __________ - | | | | -ffmpeg 1 -----| feed 1 |-----| stream 1 | - \ |_________|\ |__________| - \ \ - \ \ __________ - \ \ | | - \ \| stream 2 | - \ |__________| - \ - \ _________ __________ - \ | | | | - \| feed 2 |-----| stream 3 | - |_________| |__________| - - _________ __________ - | | | | -ffmpeg 2 -----| feed 3 |-----| stream 4 | - |_________| |__________| - - _________ __________ - | | | | - | file 1 |-----| stream 5 | - |_________| |__________| -@end example - -@anchor{FFM} -@section FFM, FFM2 formats - -FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of -video and audio streams and encoding options, and can store a moving time segment -of an infinite movie or a whole movie. - -FFM is version specific, and there is limited compatibility of FFM files -generated by one version of ffmpeg/ffserver and another version of -ffmpeg/ffserver. It may work but it is not guaranteed to work. - -FFM2 is extensible while maintaining compatibility and should work between -differing versions of tools. FFM2 is the default. - -@section Status stream - -@command{ffserver} supports an HTTP interface which exposes the -current status of the server. - -Simply point your browser to the address of the special status stream -specified in the configuration file. - -For example if you have: -@example - -Format status - -# Only allow local people to get the status -ACL allow localhost -ACL allow 192.168.0.0 192.168.255.255 - -@end example - -then the server will post a page with the status information when -the special stream @file{status.html} is requested. - -@section How do I make it work? - -As a simple test, just run the following two command lines where INPUTFILE -is some file which you can decode with ffmpeg: - -@example -ffserver -f doc/ffserver.conf & -ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm -@end example - -At this point you should be able to go to your Windows machine and fire up -Windows Media Player (WMP). Go to Open URL and enter - -@example - http://:8090/test.asf -@end example - -You should (after a short delay) see video and hear audio. - -WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to -transfer the entire file before starting to play. -The same is true of AVI files. - -@section What happens next? - -You should edit the ffserver.conf file to suit your needs (in terms of -frame rates etc). Then install ffserver and ffmpeg, write a script to start -them up, and off you go. - -@section What else can it do? - -You can replay video from .ffm files that was recorded earlier. -However, there are a number of caveats, including the fact that the -ffserver parameters must match the original parameters used to record the -file. If they do not, then ffserver deletes the file before recording into it. -(Now that I write this, it seems broken). - -You can fiddle with many of the codec choices and encoding parameters, and -there are a bunch more parameters that you cannot control. Post a message -to the mailing list if there are some 'must have' parameters. Look in -ffserver.conf for a list of the currently available controls. - -It will automatically generate the ASX or RAM files that are often used -in browsers. These files are actually redirections to the underlying ASF -or RM file. The reason for this is that the browser often fetches the -entire file before starting up the external viewer. The redirection files -are very small and can be transferred quickly. [The stream itself is -often 'infinite' and thus the browser tries to download it and never -finishes.] - -@section Tips - -* When you connect to a live stream, most players (WMP, RA, etc) want to -buffer a certain number of seconds of material so that they can display the -signal continuously. However, ffserver (by default) starts sending data -in realtime. This means that there is a pause of a few seconds while the -buffering is being done by the player. The good news is that this can be -cured by adding a '?buffer=5' to the end of the URL. This means that the -stream should start 5 seconds in the past -- and so the first 5 seconds -of the stream are sent as fast as the network will allow. It will then -slow down to real time. This noticeably improves the startup experience. - -You can also add a 'Preroll 15' statement into the ffserver.conf that will -add the 15 second prebuffering on all requests that do not otherwise -specify a time. In addition, ffserver will skip frames until a key_frame -is found. This further reduces the startup delay by not transferring data -that will be discarded. - -@section Why does the ?buffer / Preroll stop working after a time? - -It turns out that (on my machine at least) the number of frames successfully -grabbed is marginally less than the number that ought to be grabbed. This -means that the timestamp in the encoded data stream gets behind realtime. -This means that if you say 'Preroll 10', then when the stream gets 10 -or more seconds behind, there is no Preroll left. - -Fixing this requires a change in the internals of how timestamps are -handled. - -@section Does the @code{?date=} stuff work. - -Yes (subject to the limitation outlined above). Also note that whenever you -start ffserver, it deletes the ffm file (if any parameters have changed), -thus wiping out what you had recorded before. - -The format of the @code{?date=xxxxxx} is fairly flexible. You should use one -of the following formats (the 'T' is literal): - -@example -* YYYY-MM-DDTHH:MM:SS (localtime) -* YYYY-MM-DDTHH:MM:SSZ (UTC) -@end example - -You can omit the YYYY-MM-DD, and then it refers to the current day. However -note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this -may be in the future and so is unlikely to be useful. - -You use this by adding the ?date= to the end of the URL for the stream. -For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}. -@c man end - -@chapter Options -@c man begin OPTIONS - -@include fftools-common-opts.texi - -@section Main options - -@table @option -@item -f @var{configfile} -Read configuration file @file{configfile}. If not specified it will -read by default from @file{/etc/ffserver.conf}. - -@item -n -Enable no-launch mode. This option disables all the @code{Launch} -directives within the various @code{} sections. Since -@command{ffserver} will not launch any @command{ffmpeg} instances, you -will have to launch them manually. - -@item -d -Enable debug mode. This option increases log verbosity, and directs -log messages to stdout. When specified, the @option{CustomLog} option -is ignored. -@end table - -@chapter Configuration file syntax - -@command{ffserver} reads a configuration file containing global -options and settings for each stream and feed. - -The configuration file consists of global options and dedicated -sections, which must be introduced by "<@var{SECTION_NAME} -@var{ARGS}>" on a separate line and must be terminated by a line in -the form "". @var{ARGS} is optional. - -Currently the following sections are recognized: @samp{Feed}, -@samp{Stream}, @samp{Redirect}. - -A line starting with @code{#} is ignored and treated as a comment. - -Name of options and sections are case-insensitive. - -@section ACL syntax -An ACL (Access Control List) specifies the address which are allowed -to access a given stream, or to write a given feed. - -It accepts the folling forms -@itemize -@item -Allow/deny access to @var{address}. -@example -ACL ALLOW
-ACL DENY
-@end example - -@item -Allow/deny access to ranges of addresses from @var{first_address} to -@var{last_address}. -@example -ACL ALLOW -ACL DENY -@end example -@end itemize - -You can repeat the ACL allow/deny as often as you like. It is on a per -stream basis. The first match defines the action. If there are no matches, -then the default is the inverse of the last ACL statement. - -Thus 'ACL allow localhost' only allows access from localhost. -'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and -allow everybody else. - -@section Global options -@table @option -@item Port @var{port_number} -@item RTSPPort @var{port_number} - -Set TCP port number on which the HTTP/RTSP server is listening. You -must select a different port from your standard HTTP web server if it -is running on the same computer. - -If not specified, no corresponding server will be created. - -@item BindAddress @var{ip_address} -@item RTSPBindAddress @var{ip_address} -Set address on which the HTTP/RTSP server is bound. Only useful if you -have several network interfaces. - -@item MaxHTTPConnections @var{n} -Set number of simultaneous HTTP connections that can be handled. It -has to be defined @emph{before} the @option{MaxClients} parameter, -since it defines the @option{MaxClients} maximum limit. - -Default value is 2000. - -@item MaxClients @var{n} -Set number of simultaneous requests that can be handled. Since -@command{ffserver} is very fast, it is more likely that you will want -to leave this high and use @option{MaxBandwidth}. - -Default value is 5. - -@item MaxBandwidth @var{kbps} -Set the maximum amount of kbit/sec that you are prepared to consume -when streaming to clients. - -Default value is 1000. - -@item CustomLog @var{filename} -Set access log file (uses standard Apache log file format). '-' is the -standard output. - -If not specified @command{ffserver} will produce no log. - -In case the commandline option @option{-d} is specified this option is -ignored, and the log is written to standard output. - -@item NoDaemon -Set no-daemon mode. This option is currently ignored since now -@command{ffserver} will always work in no-daemon mode, and is -deprecated. -@end table - -@section Feed section - -A Feed section defines a feed provided to @command{ffserver}. - -Each live feed contains one video and/or audio sequence coming from an -@command{ffmpeg} encoder or another @command{ffserver}. This sequence -may be encoded simultaneously with several codecs at several -resolutions. - -A feed instance specification is introduced by a line in the form: -@example - -@end example - -where @var{FEED_FILENAME} specifies the unique name of the FFM stream. - -The following options are recognized within a Feed section. - -@table @option -@item File @var{filename} -@item ReadOnlyFile @var{filename} -Set the path where the feed file is stored on disk. - -If not specified, the @file{/tmp/FEED.ffm} is assumed, where -@var{FEED} is the feed name. - -If @option{ReadOnlyFile} is used the file is marked as read-only and -it will not be deleted or updated. - -@item Truncate -Truncate the feed file, rather than appending to it. By default -@command{ffserver} will append data to the file, until the maximum -file size value is reached (see @option{FileMaxSize} option). - -@item FileMaxSize @var{size} -Set maximum size of the feed file in bytes. 0 means unlimited. The -postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are -recognized. - -Default value is 5M. - -@item Launch @var{args} -Launch an @command{ffmpeg} command when creating @command{ffserver}. - -@var{args} must be a sequence of arguments to be provided to an -@command{ffmpeg} instance. The first provided argument is ignored, and -it is replaced by a path with the same dirname of the @command{ffserver} -instance, followed by the remaining argument and terminated with a -path corresponding to the feed. - -When the launched process exits, @command{ffserver} will launch -another program instance. - -In case you need a more complex @command{ffmpeg} configuration, -e.g. if you need to generate multiple FFM feeds with a single -@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand. - -This option is ignored in case the commandline option @option{-n} is -specified. - -@item ACL @var{spec} -Specify the list of IP address which are allowed or denied to write -the feed. Multiple ACL options can be specified. -@end table - -@section Stream section - -A Stream section defines a stream provided by @command{ffserver}, and -identified by a single name. - -The stream is sent when answering a request containing the stream -name. - -A stream section must be introduced by the line: -@example - -@end example - -where @var{STREAM_NAME} specifies the unique name of the stream. - -The following options are recognized within a Stream section. - -Encoding options are marked with the @emph{encoding} tag, and they are -used to set the encoding parameters, and are mapped to libavcodec -encoding options. Not all encoding options are supported, in -particular it is not possible to set encoder private options. In order -to override the encoding options specified by @command{ffserver}, you -can use the @command{ffmpeg} @option{override_ffserver} commandline -option. - -Only one of the @option{Feed} and @option{File} options should be set. - -@table @option -@item Feed @var{feed_name} -Set the input feed. @var{feed_name} must correspond to an existing -feed defined in a @code{Feed} section. - -When this option is set, encoding options are used to setup the -encoding operated by the remote @command{ffmpeg} process. - -@item File @var{filename} -Set the filename of the pre-recorded input file to stream. - -When this option is set, encoding options are ignored and the input -file content is re-streamed as is. - -@item Format @var{format_name} -Set the format of the output stream. - -Must be the name of a format recognized by FFmpeg. If set to -@samp{status}, it is treated as a status stream. - -@item InputFormat @var{format_name} -Set input format. If not specified, it is automatically guessed. - -@item Preroll @var{n} -Set this to the number of seconds backwards in time to start. Note that -most players will buffer 5-10 seconds of video, and also you need to allow -for a keyframe to appear in the data stream. - -Default value is 0. - -@item StartSendOnKey -Do not send stream until it gets the first key frame. By default -@command{ffserver} will send data immediately. - -@item MaxTime @var{n} -Set the number of seconds to run. This value set the maximum duration -of the stream a client will be able to receive. - -A value of 0 means that no limit is set on the stream duration. - -@item ACL @var{spec} -Set ACL for the stream. - -@item DynamicACL @var{spec} - -@item RTSPOption @var{option} - -@item MulticastAddress @var{address} - -@item MulticastPort @var{port} - -@item MulticastTTL @var{integer} - -@item NoLoop - -@item FaviconURL @var{url} -Set favicon (favourite icon) for the server status page. It is ignored -for regular streams. - -@item Author @var{value} -@item Comment @var{value} -@item Copyright @var{value} -@item Title @var{value} -Set metadata corresponding to the option. All these options are -deprecated in favor of @option{Metadata}. - -@item Metadata @var{key} @var{value} -Set metadata value on the output stream. - -@item NoAudio -@item NoVideo -Suppress audio/video. - -@item AudioCodec @var{codec_name} (@emph{encoding,audio}) -Set audio codec. - -@item AudioBitRate @var{rate} (@emph{encoding,audio}) -Set bitrate for the audio stream in kbits per second. - -@item AudioChannels @var{n} (@emph{encoding,audio}) -Set number of audio channels. - -@item AudioSampleRate @var{n} (@emph{encoding,audio}) -Set sampling frequency for audio. When using low bitrates, you should -lower this frequency to 22050 or 11025. The supported frequencies -depend on the selected audio codec. - -@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio}) -Set generic option for audio stream. - -@item AVPresetAudio @var{preset} (@emph{encoding,audio}) -Set preset for audio stream. - -@item VideoCodec @var{codec_name} (@emph{encoding,video}) -Set video codec. - -@item VideoBitRate @var{n} (@emph{encoding,video}) -Set bitrate for the video stream in kbits per second. - -@item VideoBitRateRange @var{range} (@emph{encoding,video}) -Set video bitrate range. - -A range must be specified in the form @var{minrate}-@var{maxrate}, and -specifies the @option{minrate} and @option{maxrate} encoding options -expressed in kbits per second. - -@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video}) -Set video bitrate tolerance in kbits per second. - -@item PixelFormat @var{pixel_format} (@emph{encoding,video}) -Set video pixel format. - -@item Debug @var{integer} (@emph{encoding,video}) -Set video @option{debug} encoding option. - -@item Strict @var{integer} (@emph{encoding,video}) -Set video @option{strict} encoding option. - -@item VideoBufferSize @var{n} (@emph{encoding,video}) -Set ratecontrol buffer size, expressed in KB. - -@item VideoFrameRate @var{n} (@emph{encoding,video}) -Set number of video frames per second. - -@item VideoSize (@emph{encoding,video}) -Set size of the video frame, must be an abbreviation or in the form -@var{W}x@var{H}. See @ref{video size syntax,,the Video size section -in the ffmpeg-utils(1) manual,ffmpeg-utils}. - -Default value is @code{160x128}. - -@item VideoIntraOnly (@emph{encoding,video}) -Transmit only intra frames (useful for low bitrates, but kills frame rate). - -@item VideoGopSize @var{n} (@emph{encoding,video}) -If non-intra only, an intra frame is transmitted every VideoGopSize -frames. Video synchronization can only begin at an intra frame. - -@item VideoTag @var{tag} (@emph{encoding,video}) -Set video tag. - -@item VideoHighQuality (@emph{encoding,video}) -@item Video4MotionVector (@emph{encoding,video}) - -@item BitExact (@emph{encoding,video}) -Set bitexact encoding flag. - -@item IdctSimple (@emph{encoding,video}) -Set simple IDCT algorithm. - -@item Qscale @var{n} (@emph{encoding,video}) -Enable constant quality encoding, and set video qscale (quantization -scale) value, expressed in @var{n} QP units. - -@item VideoQMin @var{n} (@emph{encoding,video}) -@item VideoQMax @var{n} (@emph{encoding,video}) -Set video qmin/qmax. - -@item VideoQDiff @var{integer} (@emph{encoding,video}) -Set video @option{qdiff} encoding option. - -@item LumiMask @var{float} (@emph{encoding,video}) -@item DarkMask @var{float} (@emph{encoding,video}) -Set @option{lumi_mask}/@option{dark_mask} encoding options. - -@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video}) -Set generic option for video stream. - -@item AVPresetVideo @var{preset} (@emph{encoding,video}) -Set preset for video stream. - -@var{preset} must be the path of a preset file. -@end table - -@subsection Server status stream - -A server status stream is a special stream which is used to show -statistics about the @command{ffserver} operations. - -It must be specified setting the option @option{Format} to -@samp{status}. - -@section Redirect section - -A redirect section specifies where to redirect the requested URL to -another page. - -A redirect section must be introduced by the line: -@example - -@end example - -where @var{NAME} is the name of the page which should be redirected. - -It only accepts the option @option{URL}, which specify the redirection -URL. - -@chapter Stream examples - -@itemize -@item -Multipart JPEG -@example - -Feed feed1.ffm -Format mpjpeg -VideoFrameRate 2 -VideoIntraOnly -NoAudio -Strict -1 - -@end example - -@item -Single JPEG -@example - -Feed feed1.ffm -Format jpeg -VideoFrameRate 2 -VideoIntraOnly -VideoSize 352x240 -NoAudio -Strict -1 - -@end example - -@item -Flash -@example - -Feed feed1.ffm -Format swf -VideoFrameRate 2 -VideoIntraOnly -NoAudio - -@end example - -@item -ASF compatible -@example - -Feed feed1.ffm -Format asf -VideoFrameRate 15 -VideoSize 352x240 -VideoBitRate 256 -VideoBufferSize 40 -VideoGopSize 30 -AudioBitRate 64 -StartSendOnKey - -@end example - -@item -MP3 audio -@example - -Feed feed1.ffm -Format mp2 -AudioCodec mp3 -AudioBitRate 64 -AudioChannels 1 -AudioSampleRate 44100 -NoVideo - -@end example - -@item -Ogg Vorbis audio -@example - -Feed feed1.ffm -Metadata title "Stream title" -AudioBitRate 64 -AudioChannels 2 -AudioSampleRate 44100 -NoVideo - -@end example - -@item -Real with audio only at 32 kbits -@example - -Feed feed1.ffm -Format rm -AudioBitRate 32 -NoVideo - -@end example - -@item -Real with audio and video at 64 kbits -@example - -Feed feed1.ffm -Format rm -AudioBitRate 32 -VideoBitRate 128 -VideoFrameRate 25 -VideoGopSize 25 - -@end example - -@item -For stream coming from a file: you only need to set the input filename -and optionally a new format. - -@example - -File "/usr/local/httpd/htdocs/tlive.rm" -NoAudio - -@end example - -@example - -File "/usr/local/httpd/htdocs/test.asf" -NoAudio -Metadata author "Me" -Metadata copyright "Super MegaCorp" -Metadata title "Test stream from disk" -Metadata comment "Test comment" - -@end example -@end itemize - -@c man end - -@include config.texi -@ifset config-all -@ifset config-avutil -@include utils.texi -@end ifset -@ifset config-avcodec -@include codecs.texi -@include bitstream_filters.texi -@end ifset -@ifset config-avformat -@include formats.texi -@include protocols.texi -@end ifset -@ifset config-avdevice -@include devices.texi -@end ifset -@ifset config-swresample -@include resampler.texi -@end ifset -@ifset config-swscale -@include scaler.texi -@end ifset -@ifset config-avfilter -@include filters.texi -@end ifset -@end ifset - -@chapter See Also - -@ifhtml -@ifset config-all -@url{ffserver.html,ffserver}, -@end ifset -@ifset config-not-all -@url{ffserver-all.html,ffserver-all}, -@end ifset -the @file{doc/ffserver.conf} example, -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, -@url{ffmpeg-utils.html,ffmpeg-utils}, -@url{ffmpeg-scaler.html,ffmpeg-scaler}, -@url{ffmpeg-resampler.html,ffmpeg-resampler}, -@url{ffmpeg-codecs.html,ffmpeg-codecs}, -@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, -@url{ffmpeg-formats.html,ffmpeg-formats}, -@url{ffmpeg-devices.html,ffmpeg-devices}, -@url{ffmpeg-protocols.html,ffmpeg-protocols}, -@url{ffmpeg-filters.html,ffmpeg-filters} -@end ifhtml - -@ifnothtml -@ifset config-all -ffserver(1), -@end ifset -@ifset config-not-all -ffserver-all(1), -@end ifset -the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1), -ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), -ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), -ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename ffserver -@settitle ffserver video server - -@end ignore - -@bye diff --git a/doc/fftools-common-opts.texi b/doc/fftools-common-opts.texi deleted file mode 100644 index 6bc6852d92..0000000000 --- a/doc/fftools-common-opts.texi +++ /dev/null @@ -1,303 +0,0 @@ -All the numerical options, if not specified otherwise, accept a string -representing a number as input, which may be followed by one of the SI -unit prefixes, for example: 'K', 'M', or 'G'. - -If 'i' is appended to the SI unit prefix, the complete prefix will be -interpreted as a unit prefix for binary multiplies, which are based on -powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit -prefix multiplies the value by 8. This allows using, for example: -'KB', 'MiB', 'G' and 'B' as number suffixes. - -Options which do not take arguments are boolean options, and set the -corresponding value to true. They can be set to false by prefixing -the option name with "no". For example using "-nofoo" -will set the boolean option with name "foo" to false. - -@anchor{Stream specifiers} -@section Stream specifiers -Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers -are used to precisely specify which stream(s) a given option belongs to. - -A stream specifier is a string generally appended to the option name and -separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the -@code{a:1} stream specifier, which matches the second audio stream. Therefore, it -would select the ac3 codec for the second audio stream. - -A stream specifier can match several streams, so that the option is applied to all -of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio -streams. - -An empty stream specifier matches all streams. For example, @code{-codec copy} -or @code{-codec: copy} would copy all the streams without reencoding. - -Possible forms of stream specifiers are: -@table @option -@item @var{stream_index} -Matches the stream with this index. E.g. @code{-threads:1 4} would set the -thread count for the second stream to 4. -@item @var{stream_type}[:@var{stream_index}] -@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle, -'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches -stream number @var{stream_index} of this type. Otherwise, it matches all -streams of this type. -@item p:@var{program_id}[:@var{stream_index}] -If @var{stream_index} is given, then it matches the stream with number @var{stream_index} -in the program with the id @var{program_id}. Otherwise, it matches all streams in the -program. -@item #@var{stream_id} or i:@var{stream_id} -Match the stream by stream id (e.g. PID in MPEG-TS container). -@end table - -@section Generic options - -These options are shared amongst the ff* tools. - -@table @option - -@item -L -Show license. - -@item -h, -?, -help, --help [@var{arg}] -Show help. An optional parameter may be specified to print help about a specific -item. If no argument is specified, only basic (non advanced) tool -options are shown. - -Possible values of @var{arg} are: -@table @option -@item long -Print advanced tool options in addition to the basic tool options. - -@item full -Print complete list of options, including shared and private options -for encoders, decoders, demuxers, muxers, filters, etc. - -@item decoder=@var{decoder_name} -Print detailed information about the decoder named @var{decoder_name}. Use the -@option{-decoders} option to get a list of all decoders. - -@item encoder=@var{encoder_name} -Print detailed information about the encoder named @var{encoder_name}. Use the -@option{-encoders} option to get a list of all encoders. - -@item demuxer=@var{demuxer_name} -Print detailed information about the demuxer named @var{demuxer_name}. Use the -@option{-formats} option to get a list of all demuxers and muxers. - -@item muxer=@var{muxer_name} -Print detailed information about the muxer named @var{muxer_name}. Use the -@option{-formats} option to get a list of all muxers and demuxers. - -@item filter=@var{filter_name} -Print detailed information about the filter name @var{filter_name}. Use the -@option{-filters} option to get a list of all filters. -@end table - -@item -version -Show version. - -@item -formats -Show available formats. - -@item -codecs -Show all codecs known to libavcodec. - -Note that the term 'codec' is used throughout this documentation as a shortcut -for what is more correctly called a media bitstream format. - -@item -decoders -Show available decoders. - -@item -encoders -Show all available encoders. - -@item -bsfs -Show available bitstream filters. - -@item -protocols -Show available protocols. - -@item -filters -Show available libavfilter filters. - -@item -pix_fmts -Show available pixel formats. - -@item -sample_fmts -Show available sample formats. - -@item -layouts -Show channel names and standard channel layouts. - -@item -colors -Show recognized color names. - -@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel} -Set the logging level used by the library. -Adding "repeat+" indicates that repeated log output should not be compressed -to the first line and the "Last message repeated n times" line will be -omitted. "repeat" can also be used alone. -If "repeat" is used alone, and with no prior loglevel set, the default -loglevel will be used. If multiple loglevel parameters are given, using -'repeat' will not change the loglevel. -@var{loglevel} is a number or a string containing one of the following values: -@table @samp -@item quiet -Show nothing at all; be silent. -@item panic -Only show fatal errors which could lead the process to crash, such as -and assert failure. This is not currently used for anything. -@item fatal -Only show fatal errors. These are errors after which the process absolutely -cannot continue after. -@item error -Show all errors, including ones which can be recovered from. -@item warning -Show all warnings and errors. Any message related to possibly -incorrect or unexpected events will be shown. -@item info -Show informative messages during processing. This is in addition to -warnings and errors. This is the default value. -@item verbose -Same as @code{info}, except more verbose. -@item debug -Show everything, including debugging information. -@end table - -By default the program logs to stderr, if coloring is supported by the -terminal, colors are used to mark errors and warnings. Log coloring -can be disabled setting the environment variable -@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting -the environment variable @env{AV_LOG_FORCE_COLOR}. -The use of the environment variable @env{NO_COLOR} is deprecated and -will be dropped in a following FFmpeg version. - -@item -report -Dump full command line and console output to a file named -@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current -directory. -This file can be useful for bug reports. -It also implies @code{-loglevel verbose}. - -Setting the environment variable @code{FFREPORT} to any value has the -same effect. If the value is a ':'-separated key=value sequence, these -options will affect the report; options values must be escaped if they -contain special characters or the options delimiter ':' (see the -``Quoting and escaping'' section in the ffmpeg-utils manual). The -following option is recognized: -@table @option -@item file -set the file name to use for the report; @code{%p} is expanded to the name -of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded -to a plain @code{%} -@end table - -Errors in parsing the environment variable are not fatal, and will not -appear in the report. - -@item -hide_banner -Suppress printing banner. - -All FFmpeg tools will normally show a copyright notice, build options -and library versions. This option can be used to suppress printing -this information. - -@item -cpuflags flags (@emph{global}) -Allows setting and clearing cpu flags. This option is intended -for testing. Do not use it unless you know what you're doing. -@example -ffmpeg -cpuflags -sse+mmx ... -ffmpeg -cpuflags mmx ... -ffmpeg -cpuflags 0 ... -@end example -Possible flags for this option are: -@table @samp -@item x86 -@table @samp -@item mmx -@item mmxext -@item sse -@item sse2 -@item sse2slow -@item sse3 -@item sse3slow -@item ssse3 -@item atom -@item sse4.1 -@item sse4.2 -@item avx -@item xop -@item fma4 -@item 3dnow -@item 3dnowext -@item cmov -@end table -@item ARM -@table @samp -@item armv5te -@item armv6 -@item armv6t2 -@item vfp -@item vfpv3 -@item neon -@end table -@item PowerPC -@table @samp -@item altivec -@end table -@item Specific Processors -@table @samp -@item pentium2 -@item pentium3 -@item pentium4 -@item k6 -@item k62 -@item athlon -@item athlonxp -@item k8 -@end table -@end table - -@item -opencl_bench -Benchmark all available OpenCL devices and show the results. This option -is only available when FFmpeg has been compiled with @code{--enable-opencl}. - -@item -opencl_options options (@emph{global}) -Set OpenCL environment options. This option is only available when -FFmpeg has been compiled with @code{--enable-opencl}. - -@var{options} must be a list of @var{key}=@var{value} option pairs -separated by ':'. See the ``OpenCL Options'' section in the -ffmpeg-utils manual for the list of supported options. -@end table - -@section AVOptions - -These options are provided directly by the libavformat, libavdevice and -libavcodec libraries. To see the list of available AVOptions, use the -@option{-help} option. They are separated into two categories: -@table @option -@item generic -These options can be set for any container, codec or device. Generic options -are listed under AVFormatContext options for containers/devices and under -AVCodecContext options for codecs. -@item private -These options are specific to the given container, device or codec. Private -options are listed under their corresponding containers/devices/codecs. -@end table - -For example to write an ID3v2.3 header instead of a default ID3v2.4 to -an MP3 file, use the @option{id3v2_version} private option of the MP3 -muxer: -@example -ffmpeg -i input.flac -id3v2_version 3 out.mp3 -@end example - -All codec AVOptions are per-stream, and thus a stream specifier -should be attached to them. - -Note: the @option{-nooption} syntax cannot be used for boolean -AVOptions, use @option{-option 0}/@option{-option 1}. - -Note: the old undocumented way of specifying per-stream AVOptions by -prepending v/a/s to the options name is now obsolete and will be -removed soon. diff --git a/doc/filter_design.txt b/doc/filter_design.txt deleted file mode 100644 index fca24a94b5..0000000000 --- a/doc/filter_design.txt +++ /dev/null @@ -1,270 +0,0 @@ -Filter design -============= - -This document explains guidelines that should be observed (or ignored with -good reason) when writing filters for libavfilter. - -In this document, the word “frame†indicates either a video frame or a group -of audio samples, as stored in an AVFilterBuffer structure. - - -Format negotiation -================== - - The query_formats method should set, for each input and each output links, - the list of supported formats. - - For video links, that means pixel format. For audio links, that means - channel layout, sample format (the sample packing is implied by the sample - format) and sample rate. - - The lists are not just lists, they are references to shared objects. When - the negotiation mechanism computes the intersection of the formats - supported at each end of a link, all references to both lists are replaced - with a reference to the intersection. And when a single format is - eventually chosen for a link amongst the remaining list, again, all - references to the list are updated. - - That means that if a filter requires that its input and output have the - same format amongst a supported list, all it has to do is use a reference - to the same list of formats. - - query_formats can leave some formats unset and return AVERROR(EAGAIN) to - cause the negotiation mechanism to try again later. That can be used by - filters with complex requirements to use the format negotiated on one link - to set the formats supported on another. - - -Buffer references ownership and permissions -=========================================== - - Principle - --------- - - Audio and video data are voluminous; the buffer and buffer reference - mechanism is intended to avoid, as much as possible, expensive copies of - that data while still allowing the filters to produce correct results. - - The data is stored in buffers represented by AVFilterBuffer structures. - They must not be accessed directly, but through references stored in - AVFilterBufferRef structures. Several references can point to the - same buffer; the buffer is automatically deallocated once all - corresponding references have been destroyed. - - The characteristics of the data (resolution, sample rate, etc.) are - stored in the reference; different references for the same buffer can - show different characteristics. In particular, a video reference can - point to only a part of a video buffer. - - A reference is usually obtained as input to the start_frame or - filter_frame method or requested using the ff_get_video_buffer or - ff_get_audio_buffer functions. A new reference on an existing buffer can - be created with the avfilter_ref_buffer. A reference is destroyed using - the avfilter_unref_bufferp function. - - Reference ownership - ------------------- - - At any time, a reference “belongs†to a particular piece of code, - usually a filter. With a few caveats that will be explained below, only - that piece of code is allowed to access it. It is also responsible for - destroying it, although this is sometimes done automatically (see the - section on link reference fields). - - Here are the (fairly obvious) rules for reference ownership: - - * A reference received by the filter_frame method (or its start_frame - deprecated version) belongs to the corresponding filter. - - Special exception: for video references: the reference may be used - internally for automatic copying and must not be destroyed before - end_frame; it can be given away to ff_start_frame. - - * A reference passed to ff_filter_frame (or the deprecated - ff_start_frame) is given away and must no longer be used. - - * A reference created with avfilter_ref_buffer belongs to the code that - created it. - - * A reference obtained with ff_get_video_buffer or ff_get_audio_buffer - belongs to the code that requested it. - - * A reference given as return value by the get_video_buffer or - get_audio_buffer method is given away and must no longer be used. - - Link reference fields - --------------------- - - The AVFilterLink structure has a few AVFilterBufferRef fields. The - cur_buf and out_buf were used with the deprecated - start_frame/draw_slice/end_frame API and should no longer be used. - src_buf, cur_buf_copy and partial_buf are used by libavfilter internally - and must not be accessed by filters. - - Reference permissions - --------------------- - - The AVFilterBufferRef structure has a perms field that describes what - the code that owns the reference is allowed to do to the buffer data. - Different references for the same buffer can have different permissions. - - For video filters that implement the deprecated - start_frame/draw_slice/end_frame API, the permissions only apply to the - parts of the buffer that have already been covered by the draw_slice - method. - - The value is a binary OR of the following constants: - - * AV_PERM_READ: the owner can read the buffer data; this is essentially - always true and is there for self-documentation. - - * AV_PERM_WRITE: the owner can modify the buffer data. - - * AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data - will not be modified by previous filters. - - * AV_PERM_REUSE: the owner can output the buffer several times, without - modifying the data in between. - - * AV_PERM_REUSE2: the owner can output the buffer several times and - modify the data in between (useless without the WRITE permissions). - - * AV_PERM_ALIGN: the owner can access the data using fast operations - that require data alignment. - - The READ, WRITE and PRESERVE permissions are about sharing the same - buffer between several filters to avoid expensive copies without them - doing conflicting changes on the data. - - The REUSE and REUSE2 permissions are about special memory for direct - rendering. For example a buffer directly allocated in video memory must - not modified once it is displayed on screen, or it will cause tearing; - it will therefore not have the REUSE2 permission. - - The ALIGN permission is about extracting part of the buffer, for - copy-less padding or cropping for example. - - - References received on input pads are guaranteed to have all the - permissions stated in the min_perms field and none of the permissions - stated in the rej_perms. - - References obtained by ff_get_video_buffer and ff_get_audio_buffer are - guaranteed to have at least all the permissions requested as argument. - - References created by avfilter_ref_buffer have the same permissions as - the original reference minus the ones explicitly masked; the mask is - usually ~0 to keep the same permissions. - - Filters should remove permissions on reference they give to output - whenever necessary. It can be automatically done by setting the - rej_perms field on the output pad. - - Here are a few guidelines corresponding to common situations: - - * Filters that modify and forward their frame (like drawtext) need the - WRITE permission. - - * Filters that read their input to produce a new frame on output (like - scale) need the READ permission on input and must request a buffer - with the WRITE permission. - - * Filters that intend to keep a reference after the filtering process - is finished (after filter_frame returns) must have the PRESERVE - permission on it and remove the WRITE permission if they create a new - reference to give it away. - - * Filters that intend to modify a reference they have kept after the end - of the filtering process need the REUSE2 permission and must remove - the PRESERVE permission if they create a new reference to give it - away. - - -Frame scheduling -================ - - The purpose of these rules is to ensure that frames flow in the filter - graph without getting stuck and accumulating somewhere. - - Simple filters that output one frame for each input frame should not have - to worry about it. - - filter_frame - ------------ - - This method is called when a frame is pushed to the filter's input. It - can be called at any time except in a reentrant way. - - If the input frame is enough to produce output, then the filter should - push the output frames on the output link immediately. - - As an exception to the previous rule, if the input frame is enough to - produce several output frames, then the filter needs output only at - least one per link. The additional frames can be left buffered in the - filter; these buffered frames must be flushed immediately if a new input - produces new output. - - (Example: frame rate-doubling filter: filter_frame must (1) flush the - second copy of the previous frame, if it is still there, (2) push the - first copy of the incoming frame, (3) keep the second copy for later.) - - If the input frame is not enough to produce output, the filter must not - call request_frame to get more. It must just process the frame or queue - it. The task of requesting more frames is left to the filter's - request_frame method or the application. - - If a filter has several inputs, the filter must be ready for frames - arriving randomly on any input. Therefore, any filter with several inputs - will most likely require some kind of queuing mechanism. It is perfectly - acceptable to have a limited queue and to drop frames when the inputs - are too unbalanced. - - request_frame - ------------- - - This method is called when a frame is wanted on an output. - - For an input, it should directly call filter_frame on the corresponding - output. - - For a filter, if there are queued frames already ready, one of these - frames should be pushed. If not, the filter should request a frame on - one of its inputs, repeatedly until at least one frame has been pushed. - - Return values: - if request_frame could produce a frame, it should return 0; - if it could not for temporary reasons, it should return AVERROR(EAGAIN); - if it could not because there are no more frames, it should return - AVERROR_EOF. - - The typical implementation of request_frame for a filter with several - inputs will look like that: - - if (frames_queued) { - push_one_frame(); - return 0; - } - while (!frame_pushed) { - input = input_where_a_frame_is_most_needed(); - ret = ff_request_frame(input); - if (ret == AVERROR_EOF) { - process_eof_on_input(); - } else if (ret < 0) { - return ret; - } - } - return 0; - - Note that, except for filters that can have queued frames, request_frame - does not push frames: it requests them to its input, and as a reaction, - the filter_frame method will be called and do the work. - -Legacy API -========== - - Until libavfilter 3.23, the filter_frame method was split: - - - for video filters, it was made of start_frame, draw_slice (that could be - called several times on distinct parts of the frame) and end_frame; - - - for audio filters, it was called filter_samples. diff --git a/doc/filters.texi b/doc/filters.texi index bcaa632cc4..5f9d1f8388 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1,100 +1,3 @@ -@chapter Filtering Introduction -@c man begin FILTERING INTRODUCTION - -Filtering in FFmpeg is enabled through the libavfilter library. - -In libavfilter, a filter can have multiple inputs and multiple -outputs. -To illustrate the sorts of things that are possible, we consider the -following filtergraph. - -@example - [main] -input --> split ---------------------> overlay --> output - | ^ - |[tmp] [flip]| - +-----> crop --> vflip -------+ -@end example - -This filtergraph splits the input stream in two streams, then sends one -stream through the crop filter and the vflip filter, before merging it -back with the other stream by overlaying it on top. You can use the -following command to achieve this: - -@example -ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT -@end example - -The result will be that the top half of the video is mirrored -onto the bottom half of the output video. - -Filters in the same linear chain are separated by commas, and distinct -linear chains of filters are separated by semicolons. In our example, -@var{crop,vflip} are in one linear chain, @var{split} and -@var{overlay} are separately in another. The points where the linear -chains join are labelled by names enclosed in square brackets. In the -example, the split filter generates two outputs that are associated to -the labels @var{[main]} and @var{[tmp]}. - -The stream sent to the second output of @var{split}, labelled as -@var{[tmp]}, is processed through the @var{crop} filter, which crops -away the lower half part of the video, and then vertically flipped. The -@var{overlay} filter takes in input the first unchanged output of the -split filter (which was labelled as @var{[main]}), and overlay on its -lower half the output generated by the @var{crop,vflip} filterchain. - -Some filters take in input a list of parameters: they are specified -after the filter name and an equal sign, and are separated from each other -by a colon. - -There exist so-called @var{source filters} that do not have an -audio/video input, and @var{sink filters} that will not have audio/video -output. - -@c man end FILTERING INTRODUCTION - -@chapter graph2dot -@c man begin GRAPH2DOT - -The @file{graph2dot} program included in the FFmpeg @file{tools} -directory can be used to parse a filtergraph description and issue a -corresponding textual representation in the dot language. - -Invoke the command: -@example -graph2dot -h -@end example - -to see how to use @file{graph2dot}. - -You can then pass the dot description to the @file{dot} program (from -the graphviz suite of programs) and obtain a graphical representation -of the filtergraph. - -For example the sequence of commands: -@example -echo @var{GRAPH_DESCRIPTION} | \ -tools/graph2dot -o graph.tmp && \ -dot -Tpng graph.tmp -o graph.png && \ -display graph.png -@end example - -can be used to create and display an image representing the graph -described by the @var{GRAPH_DESCRIPTION} string. Note that this string must be -a complete self-contained graph, with its inputs and outputs explicitly defined. -For example if your command line is of the form: -@example -ffmpeg -i infile -vf scale=640:360 outfile -@end example -your @var{GRAPH_DESCRIPTION} string will need to be of the form: -@example -nullsrc,scale=640:360,nullsink -@end example -you may also need to set the @var{nullsrc} parameters and add a @var{format} -filter in order to simulate a specific input file. - -@c man end GRAPH2DOT - @chapter Filtergraph description @c man begin FILTERGRAPH DESCRIPTION @@ -116,7 +19,7 @@ output pads is called a "sink". A filtergraph has a textual representation, which is recognized by the @option{-filter}/@option{-vf} and @option{-filter_complex} -options in @command{ffmpeg} and @option{-vf} in @command{ffplay}, and by the +options in @command{avconv} and @option{-vf} in @command{avplay}, and by the @code{avfilter_graph_parse()}/@code{avfilter_graph_parse2()} functions defined in @file{libavfilter/avfilter.h}. @@ -152,12 +55,6 @@ declares three options in this order -- @option{type}, @option{start_frame} and @var{in} is assigned to the option @option{type}, @var{0} to @option{start_frame} and @var{30} to @option{nb_frames}. -@item -A ':'-separated list of mixed direct @var{value} and long @var{key=value} -pairs. The direct @var{value} must precede the @var{key=value} pairs, and -follow the same constraints order of the previous point. The following -@var{key=value} pairs can be set in any preferred order. - @end itemize If the option value itself is a list of items (e.g. the @code{format} filter @@ -172,7 +69,7 @@ terminated when the next special character (belonging to the set The name and arguments of the filter are optionally preceded and followed by a list of link labels. -A link label allows one to name a link and associate it to a filter output +A link label allows to name a link and associate it to a filter output or input pad. The preceding labels @var{in_link_1} ... @var{in_link_N}, are associated to the filter input pads, the following labels @var{out_link_1} ... @var{out_link_M}, are @@ -215,378 +112,21 @@ Here is a BNF description of the filtergraph syntax: @var{FILTERGRAPH} ::= [sws_flags=@var{flags};] @var{FILTERCHAIN} [;@var{FILTERGRAPH}] @end example -@section Notes on filtergraph escaping - -Filtergraph description composition entails several levels of -escaping. See @ref{quoting_and_escaping,,the "Quoting and escaping" -section in the ffmpeg-utils(1) manual,ffmpeg-utils} for more -information about the employed escaping procedure. - -A first level escaping affects the content of each filter option -value, which may contain the special character @code{:} used to -separate values, or one of the escaping characters @code{\'}. - -A second level escaping affects the whole filter description, which -may contain the escaping characters @code{\'} or the special -characters @code{[],;} used by the filtergraph description. - -Finally, when you specify a filtergraph on a shell commandline, you -need to perform a third level escaping for the shell special -characters contained within it. - -For example, consider the following string to be embedded in -the @ref{drawtext} filter description @option{text} value: -@example -this is a 'string': may contain one, or more, special characters -@end example - -This string contains the @code{'} special escaping character, and the -@code{:} special character, so it needs to be escaped in this way: -@example -text=this is a \'string\'\: may contain one, or more, special characters -@end example - -A second level of escaping is required when embedding the filter -description in a filtergraph description, in order to escape all the -filtergraph special characters. Thus the example above becomes: -@example -drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters -@end example -(note that in addition to the @code{\'} escaping special characters, -also @code{,} needs to be escaped). - -Finally an additional level of escaping is needed when writing the -filtergraph description in a shell command, which depends on the -escaping rules of the adopted shell. For example, assuming that -@code{\} is special and needs to be escaped with another @code{\}, the -previous string will finally result in: -@example --vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters" -@end example - -@chapter Timeline editing - -Some filters support a generic @option{enable} option. For the filters -supporting timeline editing, this option can be set to an expression which is -evaluated before sending a frame to the filter. If the evaluation is non-zero, -the filter will be enabled, otherwise the frame will be sent unchanged to the -next filter in the filtergraph. - -The expression accepts the following values: -@table @samp -@item t -timestamp expressed in seconds, NAN if the input timestamp is unknown - -@item n -sequential number of the input frame, starting from 0 - -@item pos -the position in the file of the input frame, NAN if unknown -@end table - -Additionally, these filters support an @option{enable} command that can be used -to re-define the expression. - -Like any other filtering option, the @option{enable} option follows the same -rules. - -For example, to enable a blur filter (@ref{smartblur}) from 10 seconds to 3 -minutes, and a @ref{curves} filter starting at 3 seconds: -@example -smartblur = enable='between(t,10,3*60)', -curves = enable='gte(t,3)' : preset=cross_process -@end example - @c man end FILTERGRAPH DESCRIPTION @chapter Audio Filters @c man begin AUDIO FILTERS -When you configure your FFmpeg build, you can disable any of the -existing filters using @code{--disable-filters}. +When you configure your Libav build, you can disable any of the +existing filters using --disable-filters. The configure output will show the audio filters included in your build. Below is a description of the currently available audio filters. -@section aconvert - -Convert the input audio format to the specified formats. - -@emph{This filter is deprecated. Use @ref{aformat} instead.} - -The filter accepts a string of the form: -"@var{sample_format}:@var{channel_layout}". - -@var{sample_format} specifies the sample format, and can be a string or the -corresponding numeric value defined in @file{libavutil/samplefmt.h}. Use 'p' -suffix for a planar sample format. - -@var{channel_layout} specifies the channel layout, and can be a string -or the corresponding number value defined in @file{libavutil/channel_layout.h}. - -The special parameter "auto", signifies that the filter will -automatically select the output format depending on the output filter. - -@subsection Examples - -@itemize -@item -Convert input to float, planar, stereo: -@example -aconvert=fltp:stereo -@end example - -@item -Convert input to unsigned 8-bit, automatically select out channel layout: -@example -aconvert=u8:auto -@end example -@end itemize - -@section adelay - -Delay one or more audio channels. - -Samples in delayed channel are filled with silence. - -The filter accepts the following option: - -@table @option -@item delays -Set list of delays in milliseconds for each channel separated by '|'. -At least one delay greater than 0 should be provided. -Unused delays will be silently ignored. If number of given delays is -smaller than number of channels all remaining channels will not be delayed. -@end table - -@subsection Examples - -@itemize -@item -Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave -the second channel (and any other channels that may be present) unchanged. -@example -adelay=1500|0|500 -@end example -@end itemize - -@section aecho - -Apply echoing to the input audio. - -Echoes are reflected sound and can occur naturally amongst mountains -(and sometimes large buildings) when talking or shouting; digital echo -effects emulate this behaviour and are often used to help fill out the -sound of a single instrument or vocal. The time difference between the -original signal and the reflection is the @code{delay}, and the -loudness of the reflected signal is the @code{decay}. -Multiple echoes can have different delays and decays. - -A description of the accepted parameters follows. - -@table @option -@item in_gain -Set input gain of reflected signal. Default is @code{0.6}. - -@item out_gain -Set output gain of reflected signal. Default is @code{0.3}. - -@item delays -Set list of time intervals in milliseconds between original signal and reflections -separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}. -Default is @code{1000}. - -@item decays -Set list of loudnesses of reflected signals separated by '|'. -Allowed range for each @code{decay} is @code{(0 - 1.0]}. -Default is @code{0.5}. -@end table - -@subsection Examples - -@itemize -@item -Make it sound as if there are twice as many instruments as are actually playing: -@example -aecho=0.8:0.88:60:0.4 -@end example - -@item -If delay is very short, then it sound like a (metallic) robot playing music: -@example -aecho=0.8:0.88:6:0.4 -@end example - -@item -A longer delay will sound like an open air concert in the mountains: -@example -aecho=0.8:0.9:1000:0.3 -@end example - -@item -Same as above but with one more mountain: -@example -aecho=0.8:0.9:1000|1800:0.3|0.25 -@end example -@end itemize - -@section aeval - -Modify an audio signal according to the specified expressions. - -This filter accepts one or more expressions (one for each channel), -which are evaluated and used to modify a corresponding audio signal. - -It accepts the following parameters: - -@table @option -@item exprs -Set the '|'-separated expressions list for each separate channel. If -the number of input channels is greater than the number of -expressions, the last specified expression is used for the remaining -output channels. - -@item channel_layout, c -Set output channel layout. If not specified, the channel layout is -specified by the number of expressions. If set to @samp{same}, it will -use by default the same input channel layout. -@end table - -Each expression in @var{exprs} can contain the following constants and functions: - -@table @option -@item ch -channel number of the current expression - -@item n -number of the evaluated sample, starting from 0 - -@item s -sample rate - -@item t -time of the evaluated sample expressed in seconds - -@item nb_in_channels -@item nb_out_channels -input and output number of channels - -@item val(CH) -the value of input channel with number @var{CH} -@end table - -Note: this filter is slow. For faster processing you should use a -dedicated filter. - -@subsection Examples - -@itemize -@item -Half volume: -@example -aeval=val(ch)/2:c=same -@end example - -@item -Invert phase of the second channel: -@example -eval=val(0)|-val(1) -@end example -@end itemize - -@section afade - -Apply fade-in/out effect to input audio. - -A description of the accepted parameters follows. - -@table @option -@item type, t -Specify the effect type, can be either @code{in} for fade-in, or -@code{out} for a fade-out effect. Default is @code{in}. - -@item start_sample, ss -Specify the number of the start sample for starting to apply the fade -effect. Default is 0. - -@item nb_samples, ns -Specify the number of samples for which the fade effect has to last. At -the end of the fade-in effect the output audio will have the same -volume as the input audio, at the end of the fade-out transition -the output audio will be silence. Default is 44100. - -@item start_time, st -Specify time for starting to apply the fade effect. Default is 0. -The accepted syntax is: -@example -[-]HH[:MM[:SS[.m...]]] -[-]S+[.m...] -@end example -See also the function @code{av_parse_time()}. -If set this option is used instead of @var{start_sample} one. - -@item duration, d -Specify the duration for which the fade effect has to last. Default is 0. -The accepted syntax is: -@example -[-]HH[:MM[:SS[.m...]]] -[-]S+[.m...] -@end example -See also the function @code{av_parse_time()}. -At the end of the fade-in effect the output audio will have the same -volume as the input audio, at the end of the fade-out transition -the output audio will be silence. -If set this option is used instead of @var{nb_samples} one. - -@item curve -Set curve for fade transition. - -It accepts the following values: -@table @option -@item tri -select triangular, linear slope (default) -@item qsin -select quarter of sine wave -@item hsin -select half of sine wave -@item esin -select exponential sine wave -@item log -select logarithmic -@item par -select inverted parabola -@item qua -select quadratic -@item cub -select cubic -@item squ -select square root -@item cbr -select cubic root -@end table -@end table - -@subsection Examples - -@itemize -@item -Fade in first 15 seconds of audio: -@example -afade=t=in:ss=0:d=15 -@end example - -@item -Fade out last 25 seconds of a 900 seconds audio: -@example -afade=t=out:st=875:d=25 -@end example -@end itemize - -@anchor{aformat} @section aformat -Set output format constraints for the input audio. The framework will +Convert the input audio to one of the specified formats. The framework will negotiate the most appropriate format to minimize conversions. It accepts the following parameters: @@ -601,8 +141,6 @@ A '|'-separated list of requested sample rates. @item channel_layouts A '|'-separated list of requested channel layouts. -See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils} -for the required syntax. @end table If a parameter is omitted, all values are allowed. @@ -612,94 +150,13 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo aformat=sample_fmts=u8|s16:channel_layouts=stereo @end example -@section allpass - -Apply a two-pole all-pass filter with central frequency (in Hz) -@var{frequency}, and filter-width @var{width}. -An all-pass filter changes the audio's frequency to phase relationship -without changing its frequency to amplitude relationship. - -The filter accepts the following options: - -@table @option -@item frequency, f -Set frequency in Hz. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Specify the band-width of a filter in width_type units. -@end table - -@section amerge - -Merge two or more audio streams into a single multi-channel stream. - -The filter accepts the following options: - -@table @option - -@item inputs -Set the number of inputs. Default is 2. - -@end table - -If the channel layouts of the inputs are disjoint, and therefore compatible, -the channel layout of the output will be set accordingly and the channels -will be reordered as necessary. If the channel layouts of the inputs are not -disjoint, the output will have all the channels of the first input then all -the channels of the second input, in that order, and the channel layout of -the output will be the default value corresponding to the total number of -channels. - -For example, if the first input is in 2.1 (FL+FR+LF) and the second input -is FC+BL+BR, then the output will be in 5.1, with the channels in the -following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the -first input, b1 is the first channel of the second input). - -On the other hand, if both input are in stereo, the output channels will be -in the default order: a1, a2, b1, b2, and the channel layout will be -arbitrarily set to 4.0, which may or may not be the expected value. - -All inputs must have the same sample rate, and format. - -If inputs do not have the same duration, the output will stop with the -shortest. - -@subsection Examples - -@itemize -@item -Merge two mono files into a stereo stream: -@example -amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge -@end example - -@item -Multiple merges assuming 1 video stream and 6 audio streams in @file{input.mkv}: -@example -ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv -@end example -@end itemize - @section amix Mixes multiple audio inputs into a single output. For example @example -ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT +avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT @end example will mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds. @@ -735,120 +192,111 @@ stream ends. The default value is 2 seconds. Pass the audio source unchanged to the output. -@section apad +@section asetpts -Pad the end of a audio stream with silence, this can be used together with --shortest to extend audio streams to the same length as the video stream. +Change the PTS (presentation timestamp) of the input audio frames. -@section aphaser -Add a phasing effect to the input audio. - -A phaser filter creates series of peaks and troughs in the frequency spectrum. -The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect. - -A description of the accepted parameters follows. - -@table @option -@item in_gain -Set input gain. Default is 0.4. - -@item out_gain -Set output gain. Default is 0.74 - -@item delay -Set delay in milliseconds. Default is 3.0. - -@item decay -Set decay. Default is 0.4. - -@item speed -Set modulation speed in Hz. Default is 0.5. - -@item type -Set modulation type. Default is triangular. - -It accepts the following values: -@table @samp -@item triangular, t -@item sinusoidal, s -@end table -@end table - -@anchor{aresample} -@section aresample - -Resample the input audio to the specified parameters, using the -libswresample library. If none are specified then the filter will -automatically convert between its input and output. - -This filter is also able to stretch/squeeze the audio data to make it match -the timestamps or to inject silence / cut out audio to make it match the -timestamps, do a combination of both or do neither. - -The filter accepts the syntax -[@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate} -expresses a sample rate and @var{resampler_options} is a list of -@var{key}=@var{value} pairs, separated by ":". See the -ffmpeg-resampler manual for the complete list of supported options. - -@subsection Examples - -@itemize -@item -Resample the input audio to 44100Hz: -@example -aresample=44100 -@end example - -@item -Stretch/squeeze samples to the given timestamps, with a maximum of 1000 -samples per second compensation: -@example -aresample=async=1000 -@end example -@end itemize - -@section asetnsamples - -Set the number of samples per each output audio frame. - -The last output packet may contain a different number of samples, as -the filter will flush all the remaining samples when the input audio -signal its end. - -The filter accepts the following options: +It accepts the following parameters: @table @option -@item nb_out_samples, n -Set the number of frames per each output audio frame. The number is -intended as the number of samples @emph{per each channel}. -Default value is 1024. +@item expr +The expression which is evaluated for each frame to construct its timestamp. -@item pad, p -If set to 1, the filter will pad the last audio frame with zeroes, so -that the last frame will contain the same number of samples as the -previous ones. Default value is 1. @end table -For example, to set the number of per-frame samples to 1234 and -disable padding for the last frame, use: -@example -asetnsamples=n=1234:p=0 -@end example - -@section asetrate - -Set the sample rate without altering the PCM data. -This will result in a change of speed and pitch. - -The filter accepts the following options: +The expression is evaluated through the eval API and can contain the following +constants: @table @option -@item sample_rate, r -Set the output sample rate. Default is 44100 Hz. +@item PTS +the presentation timestamp in input + +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). + +@item N +The number of audio samples passed through the filter so far, starting at 0. + +@item S +The number of audio samples in the current frame. + +@item SR +The audio sample rate. + +@item STARTPTS +The PTS of the first frame. + +@item PREV_INPTS +The previous input PTS. + +@item PREV_OUTPTS +The previous output PTS. + +@item RTCTIME +The wallclock (RTC) time in microseconds. + +@item RTCSTART +The wallclock (RTC) time at the start of the movie in microseconds. + @end table +Some examples: + +@example +# Start counting PTS from zero +asetpts=expr=PTS-STARTPTS + +# Generate timestamps by counting samples +asetpts=expr=N/SR/TB + +# Generate timestamps from a "live source" and rebase onto the current timebase +asetpts='(RTCTIME - RTCSTART) / (TB * 1000000)" +@end example + +@section asettb + +Set the timebase to use for the output frames timestamps. +It is mainly useful for testing timebase configuration. + +This filter accepts the following parameters: + +@table @option + +@item expr +The expression which is evaluated into the output timebase. + +@end table + +The expression can contain the constants @var{PI}, @var{E}, @var{PHI}, @var{AVTB} (the +default timebase), @var{intb} (the input timebase), and @var{sr} (the sample rate, +audio only). + +The default value for the input is @var{intb}. + +Some examples: + +@example +# Set the timebase to 1/25: +settb=1/25 + +# Set the timebase to 1/10: +settb=0.1 + +# Set the timebase to 1001/1000: +settb=1+0.001 + +# Set the timebase to 2*intb: +settb=2*intb + +# Set the default timebase value: +settb=AVTB + +# Set the timebase to twice the sample rate: +asettb=sr*2 +@end example + @section ashowinfo Show a line containing various information for each input audio frame. @@ -870,10 +318,6 @@ depends on the filter input pad, and is usually 1/@var{sample_rate}. @item pts_time The presentation timestamp of the input frame in seconds. -@item pos -position of the frame in the input stream, -1 if this information in -unavailable and/or meaningless (for example in case of synthetic audio) - @item fmt The sample format. @@ -894,94 +338,23 @@ audio, the data is treated as if all the planes were concatenated. A list of Adler-32 checksums for each data plane. @end table -@section astats +@section asplit -Display time domain statistical information about the audio channels. -Statistics are calculated and displayed for each audio channel and, -where applicable, an overall figure is also given. +Split input audio into several identical outputs. -It accepts the following option: -@table @option -@item length -Short window length in seconds, used for peak and trough RMS measurement. -Default is @code{0.05} (50 miliseconds). Allowed range is @code{[0.1 - 10]}. -@end table +It accepts a single parameter, which specifies the number of outputs. If +unspecified, it defaults to 2. -A description of each shown parameter follows: - -@table @option -@item DC offset -Mean amplitude displacement from zero. - -@item Min level -Minimal sample level. - -@item Max level -Maximal sample level. - -@item Peak level dB -@item RMS level dB -Standard peak and RMS level measured in dBFS. - -@item RMS peak dB -@item RMS trough dB -Peak and trough values for RMS level measured over a short window. - -@item Crest factor -Standard ratio of peak to RMS level (note: not in dB). - -@item Flat factor -Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels -(i.e. either @var{Min level} or @var{Max level}). - -@item Peak count -Number of occasions (not the number of samples) that the signal attained either -@var{Min level} or @var{Max level}. -@end table - -@section astreamsync - -Forward two audio streams and control the order the buffers are forwarded. - -The filter accepts the following options: - -@table @option -@item expr, e -Set the expression deciding which stream should be -forwarded next: if the result is negative, the first stream is forwarded; if -the result is positive or zero, the second stream is forwarded. It can use -the following variables: - -@table @var -@item b1 b2 -number of buffers forwarded so far on each stream -@item s1 s2 -number of samples forwarded so far on each stream -@item t1 t2 -current timestamp of each stream -@end table - -The default value is @code{t1-t2}, which means to always forward the stream -that has a smaller timestamp. -@end table - -@subsection Examples - -Stress-test @code{amerge} by randomly sending buffers on the wrong -input, while avoiding too much of a desynchronization: +For example, @example -amovie=file.ogg [a] ; amovie=file.mp3 [b] ; -[a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ; -[a2] [b2] amerge +avconv -i INPUT -filter_complex asplit=5 OUTPUT @end example +will create 5 copies of the input audio. @section asyncts - Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed. -This filter is not built by default, please use @ref{aresample} to do squeezing/stretching. - It accepts the following parameters: @table @option @@ -1008,32 +381,7 @@ with a negative PTS due to encoder delay. @end table -@section atempo - -Adjust audio tempo. - -The filter accepts exactly one parameter, the audio tempo. If not -specified then the filter will assume nominal 1.0 tempo. Tempo must -be in the [0.5, 2.0] range. - -@subsection Examples - -@itemize -@item -Slow down audio to 80% tempo: -@example -atempo=0.8 -@end example - -@item -To speed up audio to 125% tempo: -@example -atempo=1.25 -@end example -@end itemize - @section atrim - Trim the input so that the output contains one continuous subpart of the input. It accepts the following parameters: @@ -1043,7 +391,7 @@ Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp @var{start} will be the first sample in the output. @item end -Specify time of the first audio sample that will be dropped, i.e. the +Timestamp (in seconds) of the first audio sample that will be dropped. I.e. the audio sample immediately preceding the one with the timestamp @var{end} will be the last sample in the output. @@ -1065,10 +413,6 @@ The number of the first sample that should be output. The number of the first sample that should be dropped. @end table -@option{start}, @option{end}, @option{duration} are expressed as time -duration specifications, check the "Time duration" section in the -ffmpeg-utils manual. - Note that the first two sets of the start/end options and the @option{duration} option look at the frame timestamp, while the _sample options simply count the samples that pass through the filter. So start/end_pts and start/end_sample will @@ -1090,122 +434,17 @@ Examples: @item Drop everything except the second minute of input: @example -ffmpeg -i INPUT -af atrim=60:120 +avconv -i INPUT -af atrim=60:120 @end example @item Keep only the first 1000 samples: @example -ffmpeg -i INPUT -af atrim=end_sample=1000 +avconv -i INPUT -af atrim=end_sample=1000 @end example @end itemize -@section bandpass - -Apply a two-pole Butterworth band-pass filter with central -frequency @var{frequency}, and (3dB-point) band-width width. -The @var{csg} option selects a constant skirt gain (peak gain = Q) -instead of the default: constant 0dB peak gain. -The filter roll off at 6dB per octave (20dB per decade). - -The filter accepts the following options: - -@table @option -@item frequency, f -Set the filter's central frequency. Default is @code{3000}. - -@item csg -Constant skirt gain if set to 1. Defaults to 0. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Specify the band-width of a filter in width_type units. -@end table - -@section bandreject - -Apply a two-pole Butterworth band-reject filter with central -frequency @var{frequency}, and (3dB-point) band-width @var{width}. -The filter roll off at 6dB per octave (20dB per decade). - -The filter accepts the following options: - -@table @option -@item frequency, f -Set the filter's central frequency. Default is @code{3000}. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Specify the band-width of a filter in width_type units. -@end table - -@section bass - -Boost or cut the bass (lower) frequencies of the audio using a two-pole -shelving filter with a response similar to that of a standard -hi-fi's tone-controls. This is also known as shelving equalisation (EQ). - -The filter accepts the following options: - -@table @option -@item gain, g -Give the gain at 0 Hz. Its useful range is about -20 -(for a large cut) to +20 (for a large boost). -Beware of clipping when using a positive gain. - -@item frequency, f -Set the filter's central frequency and so can be used -to extend or reduce the frequency range to be boosted or cut. -The default value is @code{100} Hz. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Determine how steep is the filter's shelf transition. -@end table - -@section biquad - -Apply a biquad IIR filter with the given coefficients. -Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2} -are the numerator and denominator coefficients respectively. - @section bs2b Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records. @@ -1236,8 +475,32 @@ Feed level (in Hz). @end table -@section channelmap +@section channelsplit +Split each channel from an input audio stream into a separate output stream. +It accepts the following parameters: +@table @option +@item channel_layout +The channel layout of the input stream. The default is "stereo". +@end table + +For example, assuming a stereo input MP3 file, +@example +avconv -i in.mp3 -filter_complex channelsplit out.mkv +@end example +will create an output Matroska file with two audio streams, one containing only +the left channel and the other the right channel. + +Split a 5.1 WAV file into per-channel files: +@example +avconv -i in.wav -filter_complex +'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]' +-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]' +front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]' +side_right.wav +@end example + +@section channelmap Remap input channels to new locations. It accepts the following parameters: @@ -1260,40 +523,14 @@ output channels, preserving indices. For example, assuming a 5.1+downmix input MOV file, @example -ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav +avconv -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav @end example will create an output WAV file tagged as stereo from the downmix channels of the input. To fix a 5.1 WAV improperly encoded in AAC's native channel order @example -ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav -@end example - -@section channelsplit - -Split each channel from an input audio stream into a separate output stream. - -It accepts the following parameters: -@table @option -@item channel_layout -The channel layout of the input stream. The default is "stereo". -@end table - -For example, assuming a stereo input MP3 file, -@example -ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv -@end example -will create an output Matroska file with two audio streams, one containing only -the left channel and the other the right channel. - -Split a 5.1 WAV file into per-channel files: -@example -ffmpeg -i in.wav -filter_complex -'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]' --map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]' -front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]' -side_right.wav +avconv -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav @end example @section compand @@ -1316,8 +553,7 @@ a typical value for decay is 0.8 seconds. @item points A list of points for the transfer function, specified in dB relative to the maximum possible signal amplitude. Each key points list must be defined using -the following syntax: @code{x0/y0|x1/y1|x2/y2|....} or -@code{x0/y0 x1/y1 x2/y2 ....} +the following syntax: @code{x0/y0|x1/y1|x2/y2|....} The input values must be in strictly increasing order but the transfer function does not have to be monotonically rising. The point @code{0/0} is assumed but @@ -1371,105 +607,7 @@ compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1 @end example @end itemize -@section earwax - -Make audio easier to listen to on headphones. - -This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio -so that when listened to on headphones the stereo image is moved from -inside your head (standard for headphones) to outside and in front of -the listener (standard for speakers). - -Ported from SoX. - -@section equalizer - -Apply a two-pole peaking equalisation (EQ) filter. With this -filter, the signal-level at and around a selected frequency can -be increased or decreased, whilst (unlike bandpass and bandreject -filters) that at all other frequencies is unchanged. - -In order to produce complex equalisation curves, this filter can -be given several times, each with a different central frequency. - -The filter accepts the following options: - -@table @option -@item frequency, f -Set the filter's central frequency in Hz. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Specify the band-width of a filter in width_type units. - -@item gain, g -Set the required gain or attenuation in dB. -Beware of clipping when using a positive gain. -@end table - -@subsection Examples -@itemize -@item -Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz: -@example -equalizer=f=1000:width_type=h:width=200:g=-10 -@end example - -@item -Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2: -@example -equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5 -@end example -@end itemize - -@section highpass - -Apply a high-pass filter with 3dB point frequency. -The filter can be either single-pole, or double-pole (the default). -The filter roll off at 6dB per pole per octave (20dB per pole per decade). - -The filter accepts the following options: - -@table @option -@item frequency, f -Set frequency in Hz. Default is 3000. - -@item poles, p -Set number of poles. Default is 2. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Specify the band-width of a filter in width_type units. -Applies only to double-pole filter. -The default is 0.707q and gives a Butterworth response. -@end table - @section join - Join multiple input streams into one multi-channel stream. It accepts the following parameters: @@ -1496,335 +634,21 @@ and if that fails it picks the first unused input channel. Join 3 inputs (with properly set channel layouts): @example -ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT +avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT @end example Build a 5.1 output from 6 single-channel streams: @example -ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex +avconv -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex 'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE' out @end example -@section ladspa - -Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. - -To enable compilation of this filter you need to configure FFmpeg with -@code{--enable-ladspa}. - -@table @option -@item file, f -Specifies the name of LADSPA plugin library to load. If the environment -variable @env{LADSPA_PATH} is defined, the LADSPA plugin is searched in -each one of the directories specified by the colon separated list in -@env{LADSPA_PATH}, otherwise in the standard LADSPA paths, which are in -this order: @file{HOME/.ladspa/lib/}, @file{/usr/local/lib/ladspa/}, -@file{/usr/lib/ladspa/}. - -@item plugin, p -Specifies the plugin within the library. Some libraries contain only -one plugin, but others contain many of them. If this is not set filter -will list all available plugins within the specified library. - -@item controls, c -Set the '|' separated list of controls which are zero or more floating point -values that determine the behavior of the loaded plugin (for example delay, -threshold or gain). -Controls need to be defined using the following syntax: -c0=@var{value0}|c1=@var{value1}|c2=@var{value2}|..., where -@var{valuei} is the value set on the @var{i}-th control. -If @option{controls} is set to @code{help}, all available controls and -their valid ranges are printed. - -@item sample_rate, s -Specify the sample rate, default to 44100. Only used if plugin have -zero inputs. - -@item nb_samples, n -Set the number of samples per channel per each output frame, default -is 1024. Only used if plugin have zero inputs. - -@item duration, d -Set the minimum duration of the sourced audio. See the function -@code{av_parse_time()} for the accepted format, also check the "Time duration" -section in the ffmpeg-utils manual. -Note that the resulting duration may be greater than the specified duration, -as the generated audio is always cut at the end of a complete frame. -If not specified, or the expressed duration is negative, the audio is -supposed to be generated forever. -Only used if plugin have zero inputs. - -@end table - -@subsection Examples - -@itemize -@item -List all available plugins within amp (LADSPA example plugin) library: -@example -ladspa=file=amp -@end example - -@item -List all available controls and their valid ranges for @code{vcf_notch} -plugin from @code{VCF} library: -@example -ladspa=f=vcf:p=vcf_notch:c=help -@end example - -@item -Simulate low quality audio equipment using @code{Computer Music Toolkit} (CMT) -plugin library: -@example -ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12 -@end example - -@item -Add reverberation to the audio using TAP-plugins -(Tom's Audio Processing plugins): -@example -ladspa=file=tap_reverb:tap_reverb -@end example - -@item -Generate white noise, with 0.2 amplitude: -@example -ladspa=file=cmt:noise_source_white:c=c0=.2 -@end example - -@item -Generate 20 bpm clicks using plugin @code{C* Click - Metronome} from the -@code{C* Audio Plugin Suite} (CAPS) library: -@example -ladspa=file=caps:Click:c=c1=20' -@end example - -@item -Apply @code{C* Eq10X2 - Stereo 10-band equaliser} effect: -@example -ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2 -@end example -@end itemize - -@subsection Commands - -This filter supports the following commands: -@table @option -@item cN -Modify the @var{N}-th control value. - -If the specified value is not valid, it is ignored and prior one is kept. -@end table - -@section lowpass - -Apply a low-pass filter with 3dB point frequency. -The filter can be either single-pole or double-pole (the default). -The filter roll off at 6dB per pole per octave (20dB per pole per decade). - -The filter accepts the following options: - -@table @option -@item frequency, f -Set frequency in Hz. Default is 500. - -@item poles, p -Set number of poles. Default is 2. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Specify the band-width of a filter in width_type units. -Applies only to double-pole filter. -The default is 0.707q and gives a Butterworth response. -@end table - -@section pan - -Mix channels with specific gain levels. The filter accepts the output -channel layout followed by a set of channels definitions. - -This filter is also designed to remap efficiently the channels of an audio -stream. - -The filter accepts parameters of the form: -"@var{l}:@var{outdef}:@var{outdef}:..." - -@table @option -@item l -output channel layout or number of channels - -@item outdef -output channel specification, of the form: -"@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]" - -@item out_name -output channel to define, either a channel name (FL, FR, etc.) or a channel -number (c0, c1, etc.) - -@item gain -multiplicative coefficient for the channel, 1 leaving the volume unchanged - -@item in_name -input channel to use, see out_name for details; it is not possible to mix -named and numbered input channels -@end table - -If the `=' in a channel specification is replaced by `<', then the gains for -that specification will be renormalized so that the total is 1, thus -avoiding clipping noise. - -@subsection Mixing examples - -For example, if you want to down-mix from stereo to mono, but with a bigger -factor for the left channel: -@example -pan=1:c0=0.9*c0+0.1*c1 -@end example - -A customized down-mix to stereo that works automatically for 3-, 4-, 5- and -7-channels surround: -@example -pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR -@end example - -Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system -that should be preferred (see "-ac" option) unless you have very specific -needs. - -@subsection Remapping examples - -The channel remapping will be effective if, and only if: - -@itemize -@item gain coefficients are zeroes or ones, -@item only one input per channel output, -@end itemize - -If all these conditions are satisfied, the filter will notify the user ("Pure -channel mapping detected"), and use an optimized and lossless method to do the -remapping. - -For example, if you have a 5.1 source and want a stereo audio stream by -dropping the extra channels: -@example -pan="stereo: c0=FL : c1=FR" -@end example - -Given the same source, you can also switch front left and front right channels -and keep the input channel layout: -@example -pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5" -@end example - -If the input is a stereo audio stream, you can mute the front left channel (and -still keep the stereo channel layout) with: -@example -pan="stereo:c1=c1" -@end example - -Still with a stereo audio stream input, you can copy the right channel in both -front left and right: -@example -pan="stereo: c0=FR : c1=FR" -@end example - -@section replaygain - -ReplayGain scanner filter. This filter takes an audio stream as an input and -outputs it unchanged. -At end of filtering it displays @code{track_gain} and @code{track_peak}. - @section resample - Convert the audio sample format, sample rate and channel layout. It is -not meant to be used directly. - -@section silencedetect - -Detect silence in an audio stream. - -This filter logs a message when it detects that the input audio volume is less -or equal to a noise tolerance value for a duration greater or equal to the -minimum detected noise duration. - -The printed times and duration are expressed in seconds. - -The filter accepts the following options: - -@table @option -@item duration, d -Set silence duration until notification (default is 2 seconds). - -@item noise, n -Set noise tolerance. Can be specified in dB (in case "dB" is appended to the -specified value) or amplitude ratio. Default is -60dB, or 0.001. -@end table - -@subsection Examples - -@itemize -@item -Detect 5 seconds of silence with -50dB noise tolerance: -@example -silencedetect=n=-50dB:d=5 -@end example - -@item -Complete example with @command{ffmpeg} to detect silence with 0.0001 noise -tolerance in @file{silence.mp3}: -@example -ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null - -@end example -@end itemize - -@section treble - -Boost or cut treble (upper) frequencies of the audio using a two-pole -shelving filter with a response similar to that of a standard -hi-fi's tone-controls. This is also known as shelving equalisation (EQ). - -The filter accepts the following options: - -@table @option -@item gain, g -Give the gain at whichever is the lower of ~22 kHz and the -Nyquist frequency. Its useful range is about -20 (for a large cut) -to +20 (for a large boost). Beware of clipping when using a positive gain. - -@item frequency, f -Set the filter's central frequency and so can be used -to extend or reduce the frequency range to be boosted or cut. -The default value is @code{3000} Hz. - -@item width_type -Set method to specify band-width of filter. -@table @option -@item h -Hz -@item q -Q-Factor -@item o -octave -@item s -slope -@end table - -@item width, w -Determine how steep is the filter's shelf transition. -@end table +not meant to be used directly; it is inserted automatically by libavfilter +whenever conversion is needed. Use the @var{aformat} filter to force a specific +conversion. @section volume @@ -1834,7 +658,7 @@ It accepts the following parameters: @table @option @item volume -Set audio volume expression. +This expresses how the audio volume will be increased or decreased. Output values are clipped to the maximum value. @@ -1843,7 +667,7 @@ The output audio volume is given by the relation: @var{output_volume} = @var{volume} * @var{input_volume} @end example -The default value for @var{volume} is "1.0". +The default value for @var{volume} is 1.0. @item precision This parameter represents the mathematical precision. @@ -1882,65 +706,6 @@ Pre-amplification gain in dB to apply to the selected replaygain gain. Default value for @var{replaygain_preamp} is 0.0. -@item eval -Set when the volume expression is evaluated. - -It accepts the following values: -@table @samp -@item once -only evaluate expression once during the filter initialization, or -when the @samp{volume} command is sent - -@item frame -evaluate expression for each incoming frame -@end table - -Default value is @samp{once}. -@end table - -The volume expression can contain the following parameters. - -@table @option -@item n -frame number (starting at zero) -@item nb_channels -number of channels -@item nb_consumed_samples -number of samples consumed by the filter -@item nb_samples -number of samples in the current frame -@item pos -original frame position in the file -@item pts -frame PTS -@item sample_rate -sample rate -@item startpts -PTS at start of stream -@item startt -time at start of stream -@item t -frame time -@item tb -timestamp timebase -@item volume -last set volume value -@end table - -Note that when @option{eval} is set to @samp{once} only the -@var{sample_rate} and @var{tb} variables are available, all other -variables will evaluate to NAN. - -@subsection Commands - -This filter supports the following commands: -@table @option -@item volume -Modify the volume expression. -The command accepts the same syntax of the corresponding option. - -If the specified expression is not valid, it is kept at its current -value. @item replaygain_noclip Prevent clipping by limiting the gain applied. @@ -1959,67 +724,13 @@ volume=volume=1/2 volume=volume=-6.0206dB @end example -In all the above example the named key for @option{volume} can be -omitted, for example like in: -@example -volume=0.5 -@end example - @item Increase input audio power by 6 decibels using fixed-point precision: @example volume=volume=6dB:precision=fixed @end example - -@item -Fade volume after time 10 with an annihilation period of 5 seconds: -@example -volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame -@end example @end itemize -@section volumedetect - -Detect the volume of the input video. - -The filter has no parameters. The input is not modified. Statistics about -the volume will be printed in the log when the input stream end is reached. - -In particular it will show the mean volume (root mean square), maximum -volume (on a per-sample basis), and the beginning of a histogram of the -registered volume values (from the maximum value to a cumulated 1/1000 of -the samples). - -All volumes are in decibels relative to the maximum PCM value. - -@subsection Examples - -Here is an excerpt of the output: -@example -[Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB -[Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB -[Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6 -[Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62 -[Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286 -[Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042 -[Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551 -[Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609 -[Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409 -@end example - -It means that: -@itemize -@item -The mean square energy is approximately -27 dB, or 10^-2.7. -@item -The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB. -@item -There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc. -@end itemize - -In other words, raising the volume by +4 dB does not cause any clipping, -raising it by +5 dB causes clipping for 6 samples, etc. - @c man end AUDIO FILTERS @chapter Audio Sources @@ -2027,12 +738,39 @@ raising it by +5 dB causes clipping for 6 samples, etc. Below is a description of the currently available audio sources. -@section abuffer +@section anullsrc +The null audio source; it never returns audio frames. It is mainly useful as a +template and for use in analysis / debugging tools. + +It accepts, as an optional parameter, a string of the form +@var{sample_rate}:@var{channel_layout}. + +@var{sample_rate} specifies the sample rate, and defaults to 44100. + +@var{channel_layout} specifies the channel layout, and can be either an +integer or a string representing a channel layout. The default value +of @var{channel_layout} is 3, which corresponds to CH_LAYOUT_STEREO. + +Check the channel_layout_map definition in +@file{libavutil/channel_layout.c} for the mapping between strings and +channel layout values. + +Some examples: +@example +# Set the sample rate to 48000 Hz and the channel layout to CH_LAYOUT_MONO +anullsrc=48000:4 + +# The same as above +anullsrc=48000:mono +@end example + +@section abuffer Buffer audio frames, and make them available to the filter chain. -This source is mainly intended for a programmatic use, in particular -through the interface defined in @file{libavfilter/asrc_abuffer.h}. +This source is not intended to be part of user-supplied graph descriptions; it +is for insertion by calling programs, through the interface defined in +@file{libavfilter/buffersrc.h}. It accepts the following parameters: @table @option @@ -2042,294 +780,18 @@ The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in @var{numerator}/@var{denominator} form. @item sample_rate -The sample rate of the incoming audio buffers. +The audio sample rate. @item sample_fmt -The sample format of the incoming audio buffers. -Either a sample format name or its corresponging integer representation from -the enum AVSampleFormat in @file{libavutil/samplefmt.h} +The name of the sample format, as returned by @code{av_get_sample_fmt_name()}. @item channel_layout -The channel layout of the incoming audio buffers. -Either a channel layout name from channel_layout_map in -@file{libavutil/channel_layout.c} or its corresponding integer representation -from the AV_CH_LAYOUT_* macros in @file{libavutil/channel_layout.h} - -@item channels -The number of channels of the incoming audio buffers. -If both @var{channels} and @var{channel_layout} are specified, then they -must be consistent. - +The channel layout of the audio data, in the form that can be accepted by +@code{av_get_channel_layout()}. @end table -@subsection Examples - -@example -abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo -@end example - -will instruct the source to accept planar 16bit signed stereo at 44100Hz. -Since the sample format with name "s16p" corresponds to the number -6 and the "stereo" channel layout corresponds to the value 0x3, this is -equivalent to: -@example -abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3 -@end example - -@section aevalsrc - -Generate an audio signal specified by an expression. - -This source accepts in input one or more expressions (one for each -channel), which are evaluated and used to generate a corresponding -audio signal. - -This source accepts the following options: - -@table @option -@item exprs -Set the '|'-separated expressions list for each separate channel. In case the -@option{channel_layout} option is not specified, the selected channel layout -depends on the number of provided expressions. Otherwise the last -specified expression is applied to the remaining output channels. - -@item channel_layout, c -Set the channel layout. The number of channels in the specified layout -must be equal to the number of specified expressions. - -@item duration, d -Set the minimum duration of the sourced audio. See the function -@code{av_parse_time()} for the accepted format. -Note that the resulting duration may be greater than the specified -duration, as the generated audio is always cut at the end of a -complete frame. - -If not specified, or the expressed duration is negative, the audio is -supposed to be generated forever. - -@item nb_samples, n -Set the number of samples per channel per each output frame, -default to 1024. - -@item sample_rate, s -Specify the sample rate, default to 44100. -@end table - -Each expression in @var{exprs} can contain the following constants: - -@table @option -@item n -number of the evaluated sample, starting from 0 - -@item t -time of the evaluated sample expressed in seconds, starting from 0 - -@item s -sample rate - -@end table - -@subsection Examples - -@itemize -@item -Generate silence: -@example -aevalsrc=0 -@end example - -@item -Generate a sin signal with frequency of 440 Hz, set sample rate to -8000 Hz: -@example -aevalsrc="sin(440*2*PI*t):s=8000" -@end example - -@item -Generate a two channels signal, specify the channel layout (Front -Center + Back Center) explicitly: -@example -aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC" -@end example - -@item -Generate white noise: -@example -aevalsrc="-2+random(0)" -@end example - -@item -Generate an amplitude modulated signal: -@example -aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)" -@end example - -@item -Generate 2.5 Hz binaural beats on a 360 Hz carrier: -@example -aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)" -@end example - -@end itemize - -@section anullsrc - -The null audio source, return unprocessed audio frames. It is mainly useful -as a template and to be employed in analysis / debugging tools, or as -the source for filters which ignore the input data (for example the sox -synth filter). - -This source accepts the following options: - -@table @option - -@item channel_layout, cl - -Specifies the channel layout, and can be either an integer or a string -representing a channel layout. The default value of @var{channel_layout} -is "stereo". - -Check the channel_layout_map definition in -@file{libavutil/channel_layout.c} for the mapping between strings and -channel layout values. - -@item sample_rate, r -Specifies the sample rate, and defaults to 44100. - -@item nb_samples, n -Set the number of samples per requested frames. - -@end table - -@subsection Examples - -@itemize -@item -Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO. -@example -anullsrc=r=48000:cl=4 -@end example - -@item -Do the same operation with a more obvious syntax: -@example -anullsrc=r=48000:cl=mono -@end example -@end itemize - All the parameters need to be explicitly defined. -@section flite - -Synthesize a voice utterance using the libflite library. - -To enable compilation of this filter you need to configure FFmpeg with -@code{--enable-libflite}. - -Note that the flite library is not thread-safe. - -The filter accepts the following options: - -@table @option - -@item list_voices -If set to 1, list the names of the available voices and exit -immediately. Default value is 0. - -@item nb_samples, n -Set the maximum number of samples per frame. Default value is 512. - -@item textfile -Set the filename containing the text to speak. - -@item text -Set the text to speak. - -@item voice, v -Set the voice to use for the speech synthesis. Default value is -@code{kal}. See also the @var{list_voices} option. -@end table - -@subsection Examples - -@itemize -@item -Read from file @file{speech.txt}, and synthetize the text using the -standard flite voice: -@example -flite=textfile=speech.txt -@end example - -@item -Read the specified text selecting the @code{slt} voice: -@example -flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt -@end example - -@item -Input text to ffmpeg: -@example -ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt -@end example - -@item -Make @file{ffplay} speak the specified text, using @code{flite} and -the @code{lavfi} device: -@example -ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.' -@end example -@end itemize - -For more information about libflite, check: -@url{http://www.speech.cs.cmu.edu/flite/} - -@section sine - -Generate an audio signal made of a sine wave with amplitude 1/8. - -The audio signal is bit-exact. - -The filter accepts the following options: - -@table @option - -@item frequency, f -Set the carrier frequency. Default is 440 Hz. - -@item beep_factor, b -Enable a periodic beep every second with frequency @var{beep_factor} times -the carrier frequency. Default is 0, meaning the beep is disabled. - -@item sample_rate, r -Specify the sample rate, default is 44100. - -@item duration, d -Specify the duration of the generated audio stream. - -@item samples_per_frame -Set the number of samples per output frame, default is 1024. -@end table - -@subsection Examples - -@itemize - -@item -Generate a simple 440 Hz sine wave: -@example -sine -@end example - -@item -Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds: -@example -sine=220:4:d=5 -sine=f=220:b=4:d=5 -sine=frequency=220:beep_factor=4:duration=5 -@end example - -@end itemize - @c man end AUDIO SOURCES @chapter Audio Sinks @@ -2337,135 +799,31 @@ sine=frequency=220:beep_factor=4:duration=5 Below is a description of the currently available audio sinks. -@section abuffersink - -Buffer audio frames, and make them available to the end of filter chain. - -This sink is mainly intended for programmatic use, in particular -through the interface defined in @file{libavfilter/buffersink.h} -or the options system. - -It accepts a pointer to an AVABufferSinkContext structure, which -defines the incoming buffers' formats, to be passed as the opaque -parameter to @code{avfilter_init_filter} for initialization. @section anullsink Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a template and for use in analysis / debugging tools. +@section abuffersink +This sink is intended for programmatic use. Frames that arrive on this sink can +be retrieved by the calling program, using the interface defined in +@file{libavfilter/buffersink.h}. + +It does not accept any parameters. + @c man end AUDIO SINKS @chapter Video Filters @c man begin VIDEO FILTERS -When you configure your FFmpeg build, you can disable any of the -existing filters using @code{--disable-filters}. +When you configure your Libav build, you can disable any of the +existing filters using --disable-filters. The configure output will show the video filters included in your build. Below is a description of the currently available video filters. -@section alphaextract - -Extract the alpha component from the input as a grayscale video. This -is especially useful with the @var{alphamerge} filter. - -@section alphamerge - -Add or replace the alpha component of the primary input with the -grayscale value of a second input. This is intended for use with -@var{alphaextract} to allow the transmission or storage of frame -sequences that have alpha in a format that doesn't support an alpha -channel. - -For example, to reconstruct full frames from a normal YUV-encoded video -and a separate video created with @var{alphaextract}, you might use: -@example -movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out] -@end example - -Since this filter is designed for reconstruction, it operates on frame -sequences without considering timestamps, and terminates when either -input reaches end of stream. This will cause problems if your encoding -pipeline drops frames. If you're trying to apply an image as an -overlay to a video stream, consider the @var{overlay} filter instead. - -@section ass - -Same as the @ref{subtitles} filter, except that it doesn't require libavcodec -and libavformat to work. On the other hand, it is limited to ASS (Advanced -Substation Alpha) subtitles files. - -@section bbox - -Compute the bounding box for the non-black pixels in the input frame -luminance plane. - -This filter computes the bounding box containing all the pixels with a -luminance value greater than the minimum allowed value. -The parameters describing the bounding box are printed on the filter -log. - -The filter accepts the following option: - -@table @option -@item min_val -Set the minimal luminance value. Default is @code{16}. -@end table - -@section blackdetect - -Detect video intervals that are (almost) completely black. Can be -useful to detect chapter transitions, commercials, or invalid -recordings. Output lines contains the time for the start, end and -duration of the detected black interval expressed in seconds. - -In order to display the output lines, you need to set the loglevel at -least to the AV_LOG_INFO value. - -The filter accepts the following options: - -@table @option -@item black_min_duration, d -Set the minimum detected black duration expressed in seconds. It must -be a non-negative floating point number. - -Default value is 2.0. - -@item picture_black_ratio_th, pic_th -Set the threshold for considering a picture "black". -Express the minimum value for the ratio: -@example -@var{nb_black_pixels} / @var{nb_pixels} -@end example - -for which a picture is considered black. -Default value is 0.98. - -@item pixel_black_th, pix_th -Set the threshold for considering a pixel "black". - -The threshold expresses the maximum pixel luminance value for which a -pixel is considered "black". The provided value is scaled according to -the following equation: -@example -@var{absolute_threshold} = @var{luminance_minimum_value} + @var{pixel_black_th} * @var{luminance_range_size} -@end example - -@var{luminance_range_size} and @var{luminance_minimum_value} depend on -the input video format, the range is [0-255] for YUV full-range -formats and [16-235] for YUV non full-range formats. - -Default value is 0.10. -@end table - -The following example sets the maximum pixel threshold to the minimum -value, and detects only black intervals of 2 or more seconds: -@example -blackdetect=d=2:pix_th=0.00 -@end example - @section blackframe Detect frames that are (almost) completely black. Can be useful to @@ -2482,150 +840,13 @@ It accepts the following parameters: @item amount The percentage of the pixels that have to be below the threshold; it defaults to -@code{98}. +98. -@item threshold, thresh -The threshold below which a pixel value is considered black; it defaults to -@code{32}. +@item threshold +The threshold below which a pixel value is considered black; it defaults to 32. @end table -@section blend - -Blend two video frames into each other. - -It takes two input streams and outputs one stream, the first input is the -"top" layer and second input is "bottom" layer. -Output terminates when shortest input terminates. - -A description of the accepted options follows. - -@table @option -@item c0_mode -@item c1_mode -@item c2_mode -@item c3_mode -@item all_mode -Set blend mode for specific pixel component or all pixel components in case -of @var{all_mode}. Default value is @code{normal}. - -Available values for component modes are: -@table @samp -@item addition -@item and -@item average -@item burn -@item darken -@item difference -@item divide -@item dodge -@item exclusion -@item hardlight -@item lighten -@item multiply -@item negation -@item normal -@item or -@item overlay -@item phoenix -@item pinlight -@item reflect -@item screen -@item softlight -@item subtract -@item vividlight -@item xor -@end table - -@item c0_opacity -@item c1_opacity -@item c2_opacity -@item c3_opacity -@item all_opacity -Set blend opacity for specific pixel component or all pixel components in case -of @var{all_opacity}. Only used in combination with pixel component blend modes. - -@item c0_expr -@item c1_expr -@item c2_expr -@item c3_expr -@item all_expr -Set blend expression for specific pixel component or all pixel components in case -of @var{all_expr}. Note that related mode options will be ignored if those are set. - -The expressions can use the following variables: - -@table @option -@item N -The sequential number of the filtered frame, starting from @code{0}. - -@item X -@item Y -the coordinates of the current sample - -@item W -@item H -the width and height of currently filtered plane - -@item SW -@item SH -Width and height scale depending on the currently filtered plane. It is the -ratio between the corresponding luma plane number of pixels and the current -plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and -@code{0.5,0.5} for chroma planes. - -@item T -Time of the current frame, expressed in seconds. - -@item TOP, A -Value of pixel component at current location for first video frame (top layer). - -@item BOTTOM, B -Value of pixel component at current location for second video frame (bottom layer). -@end table - -@item shortest -Force termination when the shortest input terminates. Default is @code{0}. -@item repeatlast -Continue applying the last bottom frame after the end of the stream. A value of -@code{0} disable the filter after the last frame of the bottom layer is reached. -Default is @code{1}. -@end table - -@subsection Examples - -@itemize -@item -Apply transition from bottom layer to top layer in first 10 seconds: -@example -blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))' -@end example - -@item -Apply 1x1 checkerboard effect: -@example -blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)' -@end example - -@item -Apply uncover left effect: -@example -blend=all_expr='if(gte(N*SW+X,W),A,B)' -@end example - -@item -Apply uncover down effect: -@example -blend=all_expr='if(gte(Y-N*SH,0),A,B)' -@end example - -@item -Apply uncover up-left effect: -@example -blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)' -@end example -@end itemize - @section boxblur Apply a boxblur algorithm to the input video. @@ -2634,77 +855,58 @@ It accepts the following parameters: @table @option -@item luma_radius, lr -@item luma_power, lp -@item chroma_radius, cr -@item chroma_power, cp -@item alpha_radius, ar -@item alpha_power, ap +@item luma_radius +@item luma_power +@item chroma_radius +@item chroma_power +@item alpha_radius +@item alpha_power @end table -A description of the accepted options follows. +The chroma and alpha parameters are optional. If not specified, they default +to the corresponding values set for @var{luma_radius} and +@var{luma_power}. +@var{luma_radius}, @var{chroma_radius}, and @var{alpha_radius} represent +the radius in pixels of the box used for blurring the corresponding +input plane. They are expressions, and can contain the following +constants: @table @option -@item luma_radius, lr -@item chroma_radius, cr -@item alpha_radius, ar -Set an expression for the box radius in pixels used for blurring the -corresponding input plane. - -The radius value must be a non-negative number, and must not be -greater than the value of the expression @code{min(w,h)/2} for the -luma and alpha planes, and of @code{min(cw,ch)/2} for the chroma -planes. - -Default value for @option{luma_radius} is "2". If not specified, -@option{chroma_radius} and @option{alpha_radius} default to the -corresponding value set for @option{luma_radius}. - -The expressions can contain the following constants: -@table @option -@item w -@item h +@item w, h The input width and height in pixels. -@item cw -@item ch +@item cw, ch The input chroma image width and height in pixels. -@item hsub -@item vsub +@item hsub, vsub The horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p", @var{hsub} is 2 and @var{vsub} is 1. @end table -@item luma_power, lp -@item chroma_power, cp -@item alpha_power, ap -Specify how many times the boxblur filter is applied to the -corresponding plane. +The radius must be a non-negative number, and must not be greater than +the value of the expression @code{min(w,h)/2} for the luma and alpha planes, +and of @code{min(cw,ch)/2} for the chroma planes. -Default value for @option{luma_power} is 2. If not specified, -@option{chroma_power} and @option{alpha_power} default to the -corresponding value set for @option{luma_power}. +@var{luma_power}, @var{chroma_power}, and @var{alpha_power} represent +how many times the boxblur filter is applied to the corresponding +plane. -A value of 0 will disable the effect. -@end table - -@subsection Examples +Some examples: @itemize + @item Apply a boxblur filter with the luma, chroma, and alpha radii set to 2: @example boxblur=luma_radius=2:luma_power=1 -boxblur=2:1 @end example @item Set the luma radius to 2, and alpha and chroma radius to 0: @example -boxblur=2:1:cr=0:ar=0 +boxblur=2:1:0:0:0:0 @end example @item @@ -2712,141 +914,9 @@ Set the luma and chroma radii to a fraction of the video dimension: @example boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1 @end example + @end itemize -@section colorbalance -Modify intensity of primary colors (red, green and blue) of input frames. - -The filter allows an input frame to be adjusted in the shadows, midtones or highlights -regions for the red-cyan, green-magenta or blue-yellow balance. - -A positive adjustment value shifts the balance towards the primary color, a negative -value towards the complementary color. - -The filter accepts the following options: - -@table @option -@item rs -@item gs -@item bs -Adjust red, green and blue shadows (darkest pixels). - -@item rm -@item gm -@item bm -Adjust red, green and blue midtones (medium pixels). - -@item rh -@item gh -@item bh -Adjust red, green and blue highlights (brightest pixels). - -Allowed ranges for options are @code{[-1.0, 1.0]}. Defaults are @code{0}. -@end table - -@subsection Examples - -@itemize -@item -Add red color cast to shadows: -@example -colorbalance=rs=.3 -@end example -@end itemize - -@section colorchannelmixer - -Adjust video input frames by re-mixing color channels. - -This filter modifies a color channel by adding the values associated to -the other channels of the same pixels. For example if the value to -modify is red, the output value will be: -@example -@var{red}=@var{red}*@var{rr} + @var{blue}*@var{rb} + @var{green}*@var{rg} + @var{alpha}*@var{ra} -@end example - -The filter accepts the following options: - -@table @option -@item rr -@item rg -@item rb -@item ra -Adjust contribution of input red, green, blue and alpha channels for output red channel. -Default is @code{1} for @var{rr}, and @code{0} for @var{rg}, @var{rb} and @var{ra}. - -@item gr -@item gg -@item gb -@item ga -Adjust contribution of input red, green, blue and alpha channels for output green channel. -Default is @code{1} for @var{gg}, and @code{0} for @var{gr}, @var{gb} and @var{ga}. - -@item br -@item bg -@item bb -@item ba -Adjust contribution of input red, green, blue and alpha channels for output blue channel. -Default is @code{1} for @var{bb}, and @code{0} for @var{br}, @var{bg} and @var{ba}. - -@item ar -@item ag -@item ab -@item aa -Adjust contribution of input red, green, blue and alpha channels for output alpha channel. -Default is @code{1} for @var{aa}, and @code{0} for @var{ar}, @var{ag} and @var{ab}. - -Allowed ranges for options are @code{[-2.0, 2.0]}. -@end table - -@subsection Examples - -@itemize -@item -Convert source to grayscale: -@example -colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3 -@end example -@item -Simulate sepia tones: -@example -colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131 -@end example -@end itemize - -@section colormatrix - -Convert color matrix. - -The filter accepts the following options: - -@table @option -@item src -@item dst -Specify the source and destination color matrix. Both values must be -specified. - -The accepted values are: -@table @samp -@item bt709 -BT.709 - -@item bt601 -BT.601 - -@item smpte240m -SMPTE-240M - -@item fcc -FCC -@end table -@end table - -For example to convert from BT.601 to SMPTE-240M, use the command: -@example -colormatrix=bt601:smpte240m -@end example - @section copy Copy the input source unchanged to the output. This is mainly useful for @@ -2859,82 +929,60 @@ Crop the input video to given dimensions. It accepts the following parameters: @table @option -@item w, out_w -The width of the output video. It defaults to @code{iw}. -This expression is evaluated only once during the filter -configuration. -@item h, out_h -The height of the output video. It defaults to @code{ih}. -This expression is evaluated only once during the filter -configuration. +@item out_w +The width of the output video. + +@item out_h +The height of the output video. @item x The horizontal position, in the input video, of the left edge of the output -video. It defaults to @code{(in_w-out_w)/2}. -This expression is evaluated per-frame. +video. @item y The vertical position, in the input video, of the top edge of the output video. -It defaults to @code{(in_h-out_h)/2}. -This expression is evaluated per-frame. -@item keep_aspect -If set to 1 will force the output display aspect ratio -to be the same of the input, by changing the output sample aspect -ratio. It defaults to 0. @end table -The @var{out_w}, @var{out_h}, @var{x}, @var{y} parameters are -expressions containing the following constants: +The parameters are expressions containing the following constants: @table @option -@item x -@item y +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). + +@item x, y The computed values for @var{x} and @var{y}. They are evaluated for each new frame. -@item in_w -@item in_h +@item in_w, in_h The input width and height. -@item iw -@item ih +@item iw, ih These are the same as @var{in_w} and @var{in_h}. -@item out_w -@item out_h +@item out_w, out_h The output (cropped) width and height. -@item ow -@item oh +@item ow, oh These are the same as @var{out_w} and @var{out_h}. -@item a -same as @var{iw} / @var{ih} - -@item sar -input sample aspect ratio - -@item dar -input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar} - -@item hsub -@item vsub -horizontal and vertical chroma subsample values. For example for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. - @item n The number of the input frame, starting from 0. -@item pos -the position in the file of the input frame, NAN if unknown - @item t The timestamp expressed in seconds. It's NAN if the input timestamp is unknown. @end table +The @var{out_w} and @var{out_h} parameters specify the expressions for +the width and height of the output (cropped) video. They are only +evaluated during the configuration of the filter. + +The default value of @var{out_w} is "in_w", and the default value of +@var{out_h} is "in_h". + The expression for @var{out_w} may depend on the value of @var{out_h}, and the expression for @var{out_h} may depend on @var{out_w}, but they cannot depend on @var{x} and @var{y}, as @var{x} and @var{y} are @@ -2945,87 +993,48 @@ position of the top-left corner of the output (non-cropped) area. They are evaluated for each frame. If the evaluated value is not valid, it is approximated to the nearest valid value. +The default value of @var{x} is "(in_w-out_w)/2", and the default +value for @var{y} is "(in_h-out_h)/2", which set the cropped area at +the center of the input image. + The expression for @var{x} may depend on @var{y}, and the expression for @var{y} may depend on @var{x}. -@subsection Examples - -@itemize -@item -Crop area with size 100x100 at position (12,34). +Some examples: @example -crop=100:100:12:34 -@end example +# Crop the central input area with size 100x100 +crop=out_w=100:out_h=100 -Using named options, the example above becomes: -@example -crop=w=100:h=100:x=12:y=34 -@end example +# Crop the central input area with size 2/3 of the input video +"crop=out_w=2/3*in_w:out_h=2/3*in_h" -@item -Crop the central input area with size 100x100: -@example -crop=100:100 -@end example - -@item -Crop the central input area with size 2/3 of the input video: -@example -crop=2/3*in_w:2/3*in_h -@end example - -@item -Crop the input video central square: -@example +# Crop the input video central square crop=out_w=in_h -crop=in_h -@end example -@item -Delimit the rectangle with the top-left corner placed at position -100:100 and the right-bottom corner corresponding to the right-bottom -corner of the input image. -@example -crop=in_w-100:in_h-100:100:100 -@end example +# Delimit the rectangle with the top-left corner placed at position +# 100:100 and the right-bottom corner corresponding to the right-bottom +# corner of the input image +crop=out_w=in_w-100:out_h=in_h-100:x=100:y=100 -@item -Crop 10 pixels from the left and right borders, and 20 pixels from -the top and bottom borders -@example -crop=in_w-2*10:in_h-2*20 -@end example +# Crop 10 pixels from the left and right borders, and 20 pixels from +# the top and bottom borders +"crop=out_w=in_w-2*10:out_h=in_h-2*20" -@item -Keep only the bottom right quarter of the input image: -@example -crop=in_w/2:in_h/2:in_w/2:in_h/2 -@end example +# Keep only the bottom right quarter of the input image +"crop=out_w=in_w/2:out_h=in_h/2:x=in_w/2:y=in_h/2" -@item -Crop height for getting Greek harmony: -@example -crop=in_w:1/PHI*in_w -@end example +# Crop height for getting Greek harmony +"crop=out_w=in_w:out_h=1/PHI*in_w" -@item -Appply trembling effect: -@example -crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7) -@end example +# Trembling effect +"crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)" -@item -Apply erratic camera effect depending on timestamp: -@example -crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)" -@end example +# Erratic camera effect depending on timestamp +"crop=out_w=in_w/2:out_h=in_h/2:x=(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):y=(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)" -@item -Set x depending on the value of y: -@example -crop=in_w/2:in_h/2:y:10+10*sin(n/10) +# Set x depending on the value of y +"crop=in_w/2:in_h/2:y:10+10*sin(n/10)" @end example -@end itemize @section cropdetect @@ -3040,9 +1049,8 @@ It accepts the following parameters: @table @option @item limit -Set higher black value threshold, which can be optionally specified -from nothing (0) to everything (255). An intensity value greater -to the set value is considered non-black. It defaults to 24. +The threshold, an optional parameter between nothing (0) and +everything (255). It defaults to 24. @item round The value which the width/height should be divisible by. It defaults to @@ -3050,255 +1058,16 @@ The value which the width/height should be divisible by. It defaults to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs. -@item reset_count, reset -Set the counter that determines after how many frames cropdetect will -reset the previously detected largest video area and start over to -detect the current optimal crop area. Default value is 0. +@item reset +A counter that determines how many frames cropdetect will reset +the previously detected largest video area after. It will then start over +and detect the current optimal crop area. It defaults to 0. This can be useful when channel logos distort the video area. 0 indicates 'never reset', and returns the largest area encountered during playback. @end table -@anchor{curves} -@section curves - -Apply color adjustments using curves. - -This filter is similar to the Adobe Photoshop and GIMP curves tools. Each -component (red, green and blue) has its values defined by @var{N} key points -tied from each other using a smooth curve. The x-axis represents the pixel -values from the input frame, and the y-axis the new pixel values to be set for -the output frame. - -By default, a component curve is defined by the two points @var{(0;0)} and -@var{(1;1)}. This creates a straight line where each original pixel value is -"adjusted" to its own value, which means no change to the image. - -The filter allows you to redefine these two points and add some more. A new -curve (using a natural cubic spline interpolation) will be define to pass -smoothly through all these new coordinates. The new defined points needs to be -strictly increasing over the x-axis, and their @var{x} and @var{y} values must -be in the @var{[0;1]} interval. If the computed curves happened to go outside -the vector spaces, the values will be clipped accordingly. - -If there is no key point defined in @code{x=0}, the filter will automatically -insert a @var{(0;0)} point. In the same way, if there is no key point defined -in @code{x=1}, the filter will automatically insert a @var{(1;1)} point. - -The filter accepts the following options: - -@table @option -@item preset -Select one of the available color presets. This option can be used in addition -to the @option{r}, @option{g}, @option{b} parameters; in this case, the later -options takes priority on the preset values. -Available presets are: -@table @samp -@item none -@item color_negative -@item cross_process -@item darker -@item increase_contrast -@item lighter -@item linear_contrast -@item medium_contrast -@item negative -@item strong_contrast -@item vintage -@end table -Default is @code{none}. -@item master, m -Set the master key points. These points will define a second pass mapping. It -is sometimes called a "luminance" or "value" mapping. It can be used with -@option{r}, @option{g}, @option{b} or @option{all} since it acts like a -post-processing LUT. -@item red, r -Set the key points for the red component. -@item green, g -Set the key points for the green component. -@item blue, b -Set the key points for the blue component. -@item all -Set the key points for all components (not including master). -Can be used in addition to the other key points component -options. In this case, the unset component(s) will fallback on this -@option{all} setting. -@item psfile -Specify a Photoshop curves file (@code{.asv}) to import the settings from. -@end table - -To avoid some filtergraph syntax conflicts, each key points list need to be -defined using the following syntax: @code{x0/y0 x1/y1 x2/y2 ...}. - -@subsection Examples - -@itemize -@item -Increase slightly the middle level of blue: -@example -curves=blue='0.5/0.58' -@end example - -@item -Vintage effect: -@example -curves=r='0/0.11 .42/.51 1/0.95':g='0.50/0.48':b='0/0.22 .49/.44 1/0.8' -@end example -Here we obtain the following coordinates for each components: -@table @var -@item red -@code{(0;0.11) (0.42;0.51) (1;0.95)} -@item green -@code{(0;0) (0.50;0.48) (1;1)} -@item blue -@code{(0;0.22) (0.49;0.44) (1;0.80)} -@end table - -@item -The previous example can also be achieved with the associated built-in preset: -@example -curves=preset=vintage -@end example - -@item -Or simply: -@example -curves=vintage -@end example - -@item -Use a Photoshop preset and redefine the points of the green component: -@example -curves=psfile='MyCurvesPresets/purple.asv':green='0.45/0.53' -@end example -@end itemize - -@section dctdnoiz - -Denoise frames using 2D DCT (frequency domain filtering). - -This filter is not designed for real time and can be extremely slow. - -The filter accepts the following options: - -@table @option -@item sigma, s -Set the noise sigma constant. - -This @var{sigma} defines a hard threshold of @code{3 * sigma}; every DCT -coefficient (absolute value) below this threshold with be dropped. - -If you need a more advanced filtering, see @option{expr}. - -Default is @code{0}. - -@item overlap -Set number overlapping pixels for each block. Each block is of size -@code{16x16}. Since the filter can be slow, you may want to reduce this value, -at the cost of a less effective filter and the risk of various artefacts. - -If the overlapping value doesn't allow to process the whole input width or -height, a warning will be displayed and according borders won't be denoised. - -Default value is @code{15}. - -@item expr, e -Set the coefficient factor expression. - -For each coefficient of a DCT block, this expression will be evaluated as a -multiplier value for the coefficient. - -If this is option is set, the @option{sigma} option will be ignored. - -The absolute value of the coefficient can be accessed through the @var{c} -variable. -@end table - -@subsection Examples - -Apply a denoise with a @option{sigma} of @code{4.5}: -@example -dctdnoiz=4.5 -@end example - -The same operation can be achieved using the expression system: -@example -dctdnoiz=e='gte(c, 4.5*3)' -@end example - -@anchor{decimate} -@section decimate - -Drop duplicated frames at regular intervals. - -The filter accepts the following options: - -@table @option -@item cycle -Set the number of frames from which one will be dropped. Setting this to -@var{N} means one frame in every batch of @var{N} frames will be dropped. -Default is @code{5}. - -@item dupthresh -Set the threshold for duplicate detection. If the difference metric for a frame -is less than or equal to this value, then it is declared as duplicate. Default -is @code{1.1} - -@item scthresh -Set scene change threshold. Default is @code{15}. - -@item blockx -@item blocky -Set the size of the x and y-axis blocks used during metric calculations. -Larger blocks give better noise suppression, but also give worse detection of -small movements. Must be a power of two. Default is @code{32}. - -@item ppsrc -Mark main input as a pre-processed input and activate clean source input -stream. This allows the input to be pre-processed with various filters to help -the metrics calculation while keeping the frame selection lossless. When set to -@code{1}, the first stream is for the pre-processed input, and the second -stream is the clean source from where the kept frames are chosen. Default is -@code{0}. - -@item chroma -Set whether or not chroma is considered in the metric calculations. Default is -@code{1}. -@end table - -@section dejudder - -Remove judder produced by partially interlaced telecined content. - -Judder can be introduced, for instance, by @ref{pullup} filter. If the original -source was partially telecined content then the output of @code{pullup,dejudder} -will have a variable frame rate. May change the recorded frame rate of the -container. Aside from that change, this filter will not affect constant frame -rate video. - -The option available in this filter is: -@table @option - -@item cycle -Specify the length of the window over which the judder repeats. - -Accepts any interger greater than 1. Useful values are: -@table @samp - -@item 4 -If the original was telecined from 24 to 30 fps (Film to NTSC). - -@item 5 -If the original was telecined from 25 to 30 fps (PAL to NTSC). - -@item 20 -If a mixture of the two. -@end table - -The default is @samp{4}. -@end table - @section delogo Suppress a TV station logo by a simple interpolation of the surrounding @@ -3308,13 +1077,11 @@ pixels. Just set a rectangle covering the logo and watch it disappear It accepts the following parameters: @table @option -@item x -@item y +@item x, y Specify the top left corner coordinates of the logo. They must be specified. -@item w -@item h +@item w, h Specify the width and height of the logo to clear. They must be specified. @@ -3324,19 +1091,15 @@ Specify the thickness of the fuzzy edge of the rectangle (added to @item show When set to 1, a green rectangle is drawn on the screen to simplify -finding the right @var{x}, @var{y}, @var{w}, and @var{h} parameters. -The default value is 0. - -The rectangle is drawn on the outermost pixels which will be (partly) -replaced with interpolated values. The values of the next pixels -immediately outside this rectangle in each direction will be used to -compute the interpolated pixel values inside the rectangle. +finding the right @var{x}, @var{y}, @var{w}, @var{h} parameters, and +@var{band} is set to 4. The default value is 0. @end table -@subsection Examples +An example: @itemize + @item Set a rectangle covering the area with top left corner coordinates 0,0 and size 100x77, and a band of size 10: @@ -3346,86 +1109,6 @@ delogo=x=0:y=0:w=100:h=77:band=10 @end itemize -@section deshake - -Attempt to fix small changes in horizontal and/or vertical shift. This -filter helps remove camera shake from hand-holding a camera, bumping a -tripod, moving on a vehicle, etc. - -The filter accepts the following options: - -@table @option - -@item x -@item y -@item w -@item h -Specify a rectangular area where to limit the search for motion -vectors. -If desired the search for motion vectors can be limited to a -rectangular area of the frame defined by its top left corner, width -and height. These parameters have the same meaning as the drawbox -filter which can be used to visualise the position of the bounding -box. - -This is useful when simultaneous movement of subjects within the frame -might be confused for camera motion by the motion vector search. - -If any or all of @var{x}, @var{y}, @var{w} and @var{h} are set to -1 -then the full frame is used. This allows later options to be set -without specifying the bounding box for the motion vector search. - -Default - search the whole frame. - -@item rx -@item ry -Specify the maximum extent of movement in x and y directions in the -range 0-64 pixels. Default 16. - -@item edge -Specify how to generate pixels to fill blanks at the edge of the -frame. Available values are: -@table @samp -@item blank, 0 -Fill zeroes at blank locations -@item original, 1 -Original image at blank locations -@item clamp, 2 -Extruded edge value at blank locations -@item mirror, 3 -Mirrored edge at blank locations -@end table -Default value is @samp{mirror}. - -@item blocksize -Specify the blocksize to use for motion search. Range 4-128 pixels, -default 8. - -@item contrast -Specify the contrast threshold for blocks. Only blocks with more than -the specified contrast (difference between darkest and lightest -pixels) will be considered. Range 1-255, default 125. - -@item search -Specify the search strategy. Available values are: -@table @samp -@item exhaustive, 0 -Set exhaustive search -@item less, 1 -Set less exhaustive search. -@end table -Default value is @samp{exhaustive}. - -@item filename -If set then a detailed log of the motion search is written to the -specified file. - -@item opencl -If set to 1, specify using OpenCL capabilities, only available if -FFmpeg was configured with @code{--enable-opencl}. Default value is 0. - -@end table - @section drawbox Draw a colored box on the input image. @@ -3433,228 +1116,45 @@ Draw a colored box on the input image. It accepts the following parameters: @table @option -@item x -@item y -The expressions which specify the top left corner coordinates of the box. It defaults to 0. -@item width, w -@item height, h -The expressions which specify the width and height of the box; if 0 they are interpreted as +@item x, y +Specify the top left corner coordinates of the box. It defaults to 0. + +@item width, height +Specify the width and height of the box; if 0 they are interpreted as the input width and height. It defaults to 0. -@item color, c -Specify the color of the box to write. For the general syntax of this option, -check the "Color" section in the ffmpeg-utils manual. If the special -value @code{invert} is used, the box edge color is the same as the -video with inverted luma. - -@item thickness, t -The expression which sets the thickness of the box edge. Default value is @code{3}. - -See below for the list of accepted constants. +@item color +Specify the color of the box to write. It can be the name of a color +(case insensitive match) or a 0xRRGGBB[AA] sequence. @end table -The parameters for @var{x}, @var{y}, @var{w} and @var{h} and @var{t} are expressions containing the -following constants: - -@table @option -@item dar -The input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar}. - -@item hsub -@item vsub -horizontal and vertical chroma subsample values. For example for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. - -@item in_h, ih -@item in_w, iw -The input width and height. - -@item sar -The input sample aspect ratio. - -@item x -@item y -The x and y offset coordinates where the box is drawn. - -@item w -@item h -The width and height of the drawn box. - -@item t -The thickness of the drawn box. - -These constants allow the @var{x}, @var{y}, @var{w}, @var{h} and @var{t} expressions to refer to -each other, so you may for example specify @code{y=x/dar} or @code{h=w/dar}. - -@end table - -@subsection Examples - -@itemize -@item -Draw a black box around the edge of the input image: +Some examples: @example +# Draw a black box around the edge of the input image drawbox + +# Draw a box with color red and an opacity of 50% +drawbox=x=10:y=20:width=200:height=60:color=red@@0.5" @end example -@item -Draw a box with color red and an opacity of 50%: -@example -drawbox=10:20:200:60:red@@0.5 -@end example - -The previous example can be specified as: -@example -drawbox=x=10:y=20:w=200:h=60:color=red@@0.5 -@end example - -@item -Fill the box with pink color: -@example -drawbox=x=10:y=10:w=100:h=100:color=pink@@0.5:t=max -@end example - -@item -Draw a 2-pixel red 2.40:1 mask: -@example -drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red -@end example -@end itemize - -@section drawgrid - -Draw a grid on the input image. - -It accepts the following parameters: - -@table @option -@item x -@item y -The expressions which specify the coordinates of some point of grid intersection (meant to configure offset). Both default to 0. - -@item width, w -@item height, h -The expressions which specify the width and height of the grid cell, if 0 they are interpreted as the -input width and height, respectively, minus @code{thickness}, so image gets -framed. Default to 0. - -@item color, c -Specify the color of the grid. For the general syntax of this option, -check the "Color" section in the ffmpeg-utils manual. If the special -value @code{invert} is used, the grid color is the same as the -video with inverted luma. - -@item thickness, t -The expression which sets the thickness of the grid line. Default value is @code{1}. - -See below for the list of accepted constants. -@end table - -The parameters for @var{x}, @var{y}, @var{w} and @var{h} and @var{t} are expressions containing the -following constants: - -@table @option -@item dar -The input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar}. - -@item hsub -@item vsub -horizontal and vertical chroma subsample values. For example for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. - -@item in_h, ih -@item in_w, iw -The input grid cell width and height. - -@item sar -The input sample aspect ratio. - -@item x -@item y -The x and y coordinates of some point of grid intersection (meant to configure offset). - -@item w -@item h -The width and height of the drawn cell. - -@item t -The thickness of the drawn cell. - -These constants allow the @var{x}, @var{y}, @var{w}, @var{h} and @var{t} expressions to refer to -each other, so you may for example specify @code{y=x/dar} or @code{h=w/dar}. - -@end table - -@subsection Examples - -@itemize -@item -Draw a grid with cell 100x100 pixels, thickness 2 pixels, with color red and an opacity of 50%: -@example -drawgrid=width=100:height=100:thickness=2:color=red@@0.5 -@end example - -@item -Draw a white 3x3 grid with an opacity of 50%: -@example -drawgrid=w=iw/3:h=ih/3:t=2:c=white@@0.5 -@end example -@end itemize - -@anchor{drawtext} @section drawtext Draw a text string or text from a specified file on top of a video, using the libfreetype library. -To enable compilation of this filter, you need to configure FFmpeg with +To enable compilation of this filter, you need to configure Libav with @code{--enable-libfreetype}. To enable default font fallback and the @var{font} option you need to -configure FFmpeg with @code{--enable-libfontconfig}. +configure Libav with @code{--enable-libfontconfig}. -@subsection Syntax +The filter also recognizes strftime() sequences in the provided text +and expands them accordingly. Check the documentation of strftime(). It accepts the following parameters: @table @option -@item box -Used to draw a box around text using the background color. -The value must be either 1 (enable) or 0 (disable). -The default value of @var{box} is 0. - -@item boxcolor -The color to be used for drawing box around text. For the syntax of this -option, check the "Color" section in the ffmpeg-utils manual. - -The default value of @var{boxcolor} is "white". - -@item borderw -Set the width of the border to be drawn around the text using @var{bordercolor}. -The default value of @var{borderw} is 0. - -@item bordercolor -Set the color to be used for drawing border around text. For the syntax of this -option, check the "Color" section in the ffmpeg-utils manual. - -The default value of @var{bordercolor} is "black". - -@item expansion -Select how the @var{text} is expanded. Can be either @code{none}, -@code{strftime} (deprecated) or -@code{normal} (default). See the @ref{drawtext_expansion, Text expansion} section -below for details. - -@item fix_bounds -If true, check and fix text coords to avoid clipping. - -@item fontcolor -The color to be used for drawing fonts. For the syntax of this option, check -the "Color" section in the ffmpeg-utils manual. - -The default value of @var{fontcolor} is "black". - @item font The font family to be used for drawing text. By default Sans. @@ -3662,10 +1162,85 @@ The font family to be used for drawing text. By default Sans. The font file to be used for drawing text. The path must be included. This parameter is mandatory if the fontconfig support is disabled. +@item text +The text string to be drawn. The text must be a sequence of UTF-8 +encoded characters. +This parameter is mandatory if no file is specified with the parameter +@var{textfile}. + +@item textfile +A text file containing text to be drawn. The text must be a sequence +of UTF-8 encoded characters. + +This parameter is mandatory if no text string is specified with the +parameter @var{text}. + +If both text and textfile are specified, an error is thrown. + +@item x, y +The offsets where text will be drawn within the video frame. +It is relative to the top/left border of the output image. +They accept expressions similar to the @ref{overlay} filter: +@table @option + +@item x, y +The computed values for @var{x} and @var{y}. They are evaluated for +each new frame. + +@item main_w, main_h +The main input width and height. + +@item W, H +These are the same as @var{main_w} and @var{main_h}. + +@item text_w, text_h +The rendered text's width and height. + +@item w, h +These are the same as @var{text_w} and @var{text_h}. + +@item n +The number of frames processed, starting from 0. + +@item t +The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown. + +@end table + +The default value of @var{x} and @var{y} is 0. + @item fontsize The font size to be used for drawing text. The default value of @var{fontsize} is 16. +@item fontcolor +The color to be used for drawing fonts. +It is either a string (e.g. "red"), or in 0xRRGGBB[AA] format +(e.g. "0xff000033"), possibly followed by an alpha specifier. +The default value of @var{fontcolor} is "black". + +@item boxcolor +The color to be used for drawing box around text. +It is either a string (e.g. "yellow") or in 0xRRGGBB[AA] format +(e.g. "0xff00ff"), possibly followed by an alpha specifier. +The default value of @var{boxcolor} is "white". + +@item box +Used to draw a box around text using the background color. +The value must be either 1 (enable) or 0 (disable). +The default value of @var{box} is 0. + +@item shadowx, shadowy +The x and y offsets for the text shadow position with respect to the +position of the text. They can be either positive or negative +values. The default value for both is "0". + +@item shadowcolor +The color to be used for drawing a shadow behind the drawn text. It +can be a color name (e.g. "yellow") or a string in the 0xRRGGBB[AA] +form (e.g. "0xff00ff"), possibly followed by an alpha specifier. +The default value of @var{shadowcolor} is "black". + @item ft_load_flags The flags to be used for loading the fonts. @@ -3687,404 +1262,47 @@ a combination of the following values: @item monochrome @item linear_design @item no_autohint +@item end table @end table -Default value is "default". +Default value is "render". For more information consult the documentation for the FT_LOAD_* libfreetype flags. -@item shadowcolor -The color to be used for drawing a shadow behind the drawn text. For the -syntax of this option, check the "Color" section in the ffmpeg-utils manual. - -The default value of @var{shadowcolor} is "black". - -@item shadowx -@item shadowy -The x and y offsets for the text shadow position with respect to the -position of the text. They can be either positive or negative -values. The default value for both is "0". - -@item start_number -The starting frame number for the n/frame_num variable. The default value -is "0". - @item tabsize The size in number of spaces to use for rendering the tab. Default value is 4. -@item timecode -Set the initial timecode representation in "hh:mm:ss[:;.]ff" -format. It can be used with or without text parameter. @var{timecode_rate} -option must be specified. - -@item timecode_rate, rate, r -Set the timecode frame rate (timecode only). - -@item text -The text string to be drawn. The text must be a sequence of UTF-8 -encoded characters. -This parameter is mandatory if no file is specified with the parameter -@var{textfile}. - -@item textfile -A text file containing text to be drawn. The text must be a sequence -of UTF-8 encoded characters. - -This parameter is mandatory if no text string is specified with the -parameter @var{text}. - -If both @var{text} and @var{textfile} are specified, an error is thrown. - -@item reload -If set to 1, the @var{textfile} will be reloaded before each frame. -Be sure to update it atomically, or it may be read partially, or even fail. - -@item x -@item y -The expressions which specify the offsets where text will be drawn -within the video frame. They are relative to the top/left border of the -output image. - -The default value of @var{x} and @var{y} is "0". - -See below for the list of accepted constants and functions. +@item fix_bounds +If true, check and fix text coords to avoid clipping. @end table -The parameters for @var{x} and @var{y} are expressions containing the -following constants and functions: - -@table @option -@item dar -input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar} - -@item hsub -@item vsub -horizontal and vertical chroma subsample values. For example for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. - -@item line_h, lh -the height of each text line - -@item main_h, h, H -the input height - -@item main_w, w, W -the input width - -@item max_glyph_a, ascent -the maximum distance from the baseline to the highest/upper grid -coordinate used to place a glyph outline point, for all the rendered -glyphs. -It is a positive value, due to the grid's orientation with the Y axis -upwards. - -@item max_glyph_d, descent -the maximum distance from the baseline to the lowest grid coordinate -used to place a glyph outline point, for all the rendered glyphs. -This is a negative value, due to the grid's orientation, with the Y axis -upwards. - -@item max_glyph_h -maximum glyph height, that is the maximum height for all the glyphs -contained in the rendered text, it is equivalent to @var{ascent} - -@var{descent}. - -@item max_glyph_w -maximum glyph width, that is the maximum width for all the glyphs -contained in the rendered text - -@item n -the number of input frame, starting from 0 - -@item rand(min, max) -return a random number included between @var{min} and @var{max} - -@item sar -The input sample aspect ratio. - -@item t -timestamp expressed in seconds, NAN if the input timestamp is unknown - -@item text_h, th -the height of the rendered text - -@item text_w, tw -the width of the rendered text - -@item x -@item y -the x and y offset coordinates where the text is drawn. - -These parameters allow the @var{x} and @var{y} expressions to refer -each other, so you can for example specify @code{y=x/dar}. -@end table - -@anchor{drawtext_expansion} -@subsection Text expansion - -If @option{expansion} is set to @code{strftime}, -the filter recognizes strftime() sequences in the provided text and -expands them accordingly. Check the documentation of strftime(). This -feature is deprecated. - -If @option{expansion} is set to @code{none}, the text is printed verbatim. - -If @option{expansion} is set to @code{normal} (which is the default), -the following expansion mechanism is used. - -The backslash character '\', followed by any character, always expands to -the second character. - -Sequence of the form @code{%@{...@}} are expanded. The text between the -braces is a function name, possibly followed by arguments separated by ':'. -If the arguments contain special characters or delimiters (':' or '@}'), -they should be escaped. - -Note that they probably must also be escaped as the value for the -@option{text} option in the filter argument string and as the filter -argument in the filtergraph description, and possibly also for the shell, -that makes up to four levels of escaping; using a text file avoids these -problems. - -The following functions are available: - -@table @command - -@item expr, e -The expression evaluation result. - -It must take one argument specifying the expression to be evaluated, -which accepts the same constants and functions as the @var{x} and -@var{y} values. Note that not all constants should be used, for -example the text size is not known when evaluating the expression, so -the constants @var{text_w} and @var{text_h} will have an undefined -value. - -@item gmtime -The time at which the filter is running, expressed in UTC. -It can accept an argument: a strftime() format string. - -@item localtime -The time at which the filter is running, expressed in the local time zone. -It can accept an argument: a strftime() format string. - -@item metadata -Frame metadata. It must take one argument specifying metadata key. - -@item n, frame_num -The frame number, starting from 0. - -@item pict_type -A 1 character description of the current picture type. - -@item pts -The timestamp of the current frame. -It can take up to two arguments. - -The first argument is the format of the timestamp; it defaults to @code{flt} -for seconds as a decimal number with microsecond accuracy; @code{hms} stands -for a formatted @var{[-]HH:MM:SS.mmm} timestamp with millisecond accuracy. - -The second argument is an offset added to the timestamp. - -@end table - -@subsection Examples - -@itemize -@item -Draw "Test Text" with font FreeSerif, using the default values for the -optional parameters. - +For example the command: @example drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'" @end example -@item -Draw 'Test Text' with font FreeSerif of size 24 at position x=100 -and y=50 (counting from the top-left corner of the screen), text is -yellow with a red box around it. Both the text and the box have an -opacity of 20%. +will draw "Test Text" with font FreeSerif, using the default values +for the optional parameters. +The command: @example drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\ x=100: y=50: fontsize=24: fontcolor=yellow@@0.2: box=1: boxcolor=red@@0.2" @end example +will draw 'Test Text' with font FreeSerif of size 24 at position x=100 +and y=50 (counting from the top-left corner of the screen), text is +yellow with a red box around it. Both the text and the box have an +opacity of 20%. + Note that the double quotes are not necessary if spaces are not used within the parameter list. -@item -Show the text at the center of the video frame: -@example -drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h-line_h)/2" -@end example - -@item -Show a text line sliding from right to left in the last row of the video -frame. The file @file{LONG_LINE} is assumed to contain a single line -with no newlines. -@example -drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t" -@end example - -@item -Show the content of file @file{CREDITS} off the bottom of the frame and scroll up. -@example -drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t" -@end example - -@item -Draw a single green letter "g", at the center of the input video. -The glyph baseline is placed at half screen height. -@example -drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent" -@end example - -@item -Show text for 1 second every 3 seconds: -@example -drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'" -@end example - -@item -Use fontconfig to set the font. Note that the colons need to be escaped. -@example -drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg' -@end example - -@item -Print the date of a real-time encoding (see strftime(3)): -@example -drawtext='fontfile=FreeSans.ttf:text=%@{localtime:%a %b %d %Y@}' -@end example - -@end itemize - For more information about libfreetype, check: @url{http://www.freetype.org/}. -For more information about fontconfig, check: -@url{http://freedesktop.org/software/fontconfig/fontconfig-user.html}. - -@section edgedetect - -Detect and draw edges. The filter uses the Canny Edge Detection algorithm. - -The filter accepts the following options: - -@table @option -@item low -@item high -Set low and high threshold values used by the Canny thresholding -algorithm. - -The high threshold selects the "strong" edge pixels, which are then -connected through 8-connectivity with the "weak" edge pixels selected -by the low threshold. - -@var{low} and @var{high} threshold values must be chosen in the range -[0,1], and @var{low} should be lesser or equal to @var{high}. - -Default value for @var{low} is @code{20/255}, and default value for @var{high} -is @code{50/255}. - -@item mode -Define the drawing mode. - -@table @samp -@item wires -Draw white/gray wires on black background. - -@item colormix -Mix the colors to create a paint/cartoon effect. -@end table - -Default value is @var{wires}. -@end table - -@subsection Examples - -@itemize -@item -Standard edge detection with custom values for the hysteresis thresholding: -@example -edgedetect=low=0.1:high=0.4 -@end example - -@item -Painting effect without thresholding: -@example -edgedetect=mode=colormix:high=0 -@end example -@end itemize - -@section extractplanes - -Extract color channel components from input video stream into -separate grayscale video streams. - -The filter accepts the following option: - -@table @option -@item planes -Set plane(s) to extract. - -Available values for planes are: -@table @samp -@item y -@item u -@item v -@item a -@item r -@item g -@item b -@end table - -Choosing planes not available in the input will result in an error. -That means you cannot select @code{r}, @code{g}, @code{b} planes -with @code{y}, @code{u}, @code{v} planes at same time. -@end table - -@subsection Examples - -@itemize -@item -Extract luma, u and v color channel component from input video frame -into 3 grayscale outputs: -@example -ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi -@end example -@end itemize - -@section elbg - -Apply a posterize effect using the ELBG (Enhanced LBG) algorithm. - -For each input image, the filter will compute the optimal mapping from -the input to the output given the codebook length, that is the number -of distinct output colors. - -This filter accepts the following options. - -@table @option -@item codebook_length, l -Set codebook length. The value must be a positive integer, and -represents the number of distinct output colors. Default value is 256. - -@item nb_steps, n -Set the maximum number of iterations to apply for computing the optimal -mapping. The higher the value the better the result and the higher the -computation time. Default value is 1. - -@item seed, s -Set a random seed, must be an integer included between 0 and -UINT32_MAX. If not specified, or if explicitly set to -1, the filter -will try to use a good random seed on a best effort basis. -@end table - @section fade Apply a fade-in/out effect to the input video. @@ -4092,424 +1310,34 @@ Apply a fade-in/out effect to the input video. It accepts the following parameters: @table @option -@item type, t + +@item type The effect type can be either "in" for a fade-in, or "out" for a fade-out effect. -Default is @code{in}. -@item start_frame, s -Specify the number of the frame to start applying the fade -effect at. Default is 0. +@item start_frame +The number of the frame to start applying the fade effect at. -@item nb_frames, n +@item nb_frames The number of frames that the fade effect lasts. At the end of the fade-in effect, the output video will have the same intensity as the input video. -At the end of the fade-out transition, the output video will be filled with the -selected @option{color}. -Default is 25. +At the end of the fade-out transition, the output video will be completely black. -@item alpha -If set to 1, fade only alpha channel, if one exists on the input. -Default value is 0. - -@item start_time, st -Specify the timestamp (in seconds) of the frame to start to apply the fade -effect. If both start_frame and start_time are specified, the fade will start at -whichever comes last. Default is 0. - -@item duration, d -The number of seconds for which the fade effect has to last. At the end of the -fade-in effect the output video will have the same intensity as the input video, -at the end of the fade-out transition the output video will be filled with the -selected @option{color}. -If both duration and nb_frames are specified, duration is used. Default is 0. - -@item color, c -Specify the color of the fade. Default is "black". @end table -@subsection Examples - -@itemize -@item -Fade in the first 30 frames of video: +Some examples: @example -fade=in:0:30 -@end example +# Fade in the first 30 frames of video +fade=type=in:nb_frames=30 -The command above is equivalent to: -@example -fade=t=in:s=0:n=30 -@end example - -@item -Fade out the last 45 frames of a 200-frame video: -@example -fade=out:155:45 +# Fade out the last 45 frames of a 200-frame video fade=type=out:start_frame=155:nb_frames=45 -@end example -@item -Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video: -@example -fade=in:0:25, fade=out:975:25 -@end example +# Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video +fade=type=in:start_frame=0:nb_frames=25, fade=type=out:start_frame=975:nb_frames=25 -@item -Make the first 5 frames yellow, then fade in from frame 5-24: -@example -fade=in:5:20:color=yellow -@end example - -@item -Fade in alpha over first 25 frames of video: -@example -fade=in:0:25:alpha=1 -@end example - -@item -Make the first 5.5 seconds black, then fade in for 0.5 seconds: -@example -fade=t=in:st=5.5:d=0.5 -@end example - -@end itemize - -@section field - -Extract a single field from an interlaced image using stride -arithmetic to avoid wasting CPU time. The output frames are marked as -non-interlaced. - -The filter accepts the following options: - -@table @option -@item type -Specify whether to extract the top (if the value is @code{0} or -@code{top}) or the bottom field (if the value is @code{1} or -@code{bottom}). -@end table - -@section fieldmatch - -Field matching filter for inverse telecine. It is meant to reconstruct the -progressive frames from a telecined stream. The filter does not drop duplicated -frames, so to achieve a complete inverse telecine @code{fieldmatch} needs to be -followed by a decimation filter such as @ref{decimate} in the filtergraph. - -The separation of the field matching and the decimation is notably motivated by -the possibility of inserting a de-interlacing filter fallback between the two. -If the source has mixed telecined and real interlaced content, -@code{fieldmatch} will not be able to match fields for the interlaced parts. -But these remaining combed frames will be marked as interlaced, and thus can be -de-interlaced by a later filter such as @ref{yadif} before decimation. - -In addition to the various configuration options, @code{fieldmatch} can take an -optional second stream, activated through the @option{ppsrc} option. If -enabled, the frames reconstruction will be based on the fields and frames from -this second stream. This allows the first input to be pre-processed in order to -help the various algorithms of the filter, while keeping the output lossless -(assuming the fields are matched properly). Typically, a field-aware denoiser, -or brightness/contrast adjustments can help. - -Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth project) -and VIVTC/VFM (VapourSynth project). The later is a light clone of TFM from -which @code{fieldmatch} is based on. While the semantic and usage are very -close, some behaviour and options names can differ. - -The filter accepts the following options: - -@table @option -@item order -Specify the assumed field order of the input stream. Available values are: - -@table @samp -@item auto -Auto detect parity (use FFmpeg's internal parity value). -@item bff -Assume bottom field first. -@item tff -Assume top field first. -@end table - -Note that it is sometimes recommended not to trust the parity announced by the -stream. - -Default value is @var{auto}. - -@item mode -Set the matching mode or strategy to use. @option{pc} mode is the safest in the -sense that it won't risk creating jerkiness due to duplicate frames when -possible, but if there are bad edits or blended fields it will end up -outputting combed frames when a good match might actually exist. On the other -hand, @option{pcn_ub} mode is the most risky in terms of creating jerkiness, -but will almost always find a good frame if there is one. The other values are -all somewhere in between @option{pc} and @option{pcn_ub} in terms of risking -jerkiness and creating duplicate frames versus finding good matches in sections -with bad edits, orphaned fields, blended fields, etc. - -More details about p/c/n/u/b are available in @ref{p/c/n/u/b meaning} section. - -Available values are: - -@table @samp -@item pc -2-way matching (p/c) -@item pc_n -2-way matching, and trying 3rd match if still combed (p/c + n) -@item pc_u -2-way matching, and trying 3rd match (same order) if still combed (p/c + u) -@item pc_n_ub -2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if -still combed (p/c + n + u/b) -@item pcn -3-way matching (p/c/n) -@item pcn_ub -3-way matching, and trying 4th/5th matches if all 3 of the original matches are -detected as combed (p/c/n + u/b) -@end table - -The parenthesis at the end indicate the matches that would be used for that -mode assuming @option{order}=@var{tff} (and @option{field} on @var{auto} or -@var{top}). - -In terms of speed @option{pc} mode is by far the fastest and @option{pcn_ub} is -the slowest. - -Default value is @var{pc_n}. - -@item ppsrc -Mark the main input stream as a pre-processed input, and enable the secondary -input stream as the clean source to pick the fields from. See the filter -introduction for more details. It is similar to the @option{clip2} feature from -VFM/TFM. - -Default value is @code{0} (disabled). - -@item field -Set the field to match from. It is recommended to set this to the same value as -@option{order} unless you experience matching failures with that setting. In -certain circumstances changing the field that is used to match from can have a -large impact on matching performance. Available values are: - -@table @samp -@item auto -Automatic (same value as @option{order}). -@item bottom -Match from the bottom field. -@item top -Match from the top field. -@end table - -Default value is @var{auto}. - -@item mchroma -Set whether or not chroma is included during the match comparisons. In most -cases it is recommended to leave this enabled. You should set this to @code{0} -only if your clip has bad chroma problems such as heavy rainbowing or other -artifacts. Setting this to @code{0} could also be used to speed things up at -the cost of some accuracy. - -Default value is @code{1}. - -@item y0 -@item y1 -These define an exclusion band which excludes the lines between @option{y0} and -@option{y1} from being included in the field matching decision. An exclusion -band can be used to ignore subtitles, a logo, or other things that may -interfere with the matching. @option{y0} sets the starting scan line and -@option{y1} sets the ending line; all lines in between @option{y0} and -@option{y1} (including @option{y0} and @option{y1}) will be ignored. Setting -@option{y0} and @option{y1} to the same value will disable the feature. -@option{y0} and @option{y1} defaults to @code{0}. - -@item scthresh -Set the scene change detection threshold as a percentage of maximum change on -the luma plane. Good values are in the @code{[8.0, 14.0]} range. Scene change -detection is only relevant in case @option{combmatch}=@var{sc}. The range for -@option{scthresh} is @code{[0.0, 100.0]}. - -Default value is @code{12.0}. - -@item combmatch -When @option{combatch} is not @var{none}, @code{fieldmatch} will take into -account the combed scores of matches when deciding what match to use as the -final match. Available values are: - -@table @samp -@item none -No final matching based on combed scores. -@item sc -Combed scores are only used when a scene change is detected. -@item full -Use combed scores all the time. -@end table - -Default is @var{sc}. - -@item combdbg -Force @code{fieldmatch} to calculate the combed metrics for certain matches and -print them. This setting is known as @option{micout} in TFM/VFM vocabulary. -Available values are: - -@table @samp -@item none -No forced calculation. -@item pcn -Force p/c/n calculations. -@item pcnub -Force p/c/n/u/b calculations. -@end table - -Default value is @var{none}. - -@item cthresh -This is the area combing threshold used for combed frame detection. This -essentially controls how "strong" or "visible" combing must be to be detected. -Larger values mean combing must be more visible and smaller values mean combing -can be less visible or strong and still be detected. Valid settings are from -@code{-1} (every pixel will be detected as combed) to @code{255} (no pixel will -be detected as combed). This is basically a pixel difference value. A good -range is @code{[8, 12]}. - -Default value is @code{9}. - -@item chroma -Sets whether or not chroma is considered in the combed frame decision. Only -disable this if your source has chroma problems (rainbowing, etc.) that are -causing problems for the combed frame detection with chroma enabled. Actually, -using @option{chroma}=@var{0} is usually more reliable, except for the case -where there is chroma only combing in the source. - -Default value is @code{0}. - -@item blockx -@item blocky -Respectively set the x-axis and y-axis size of the window used during combed -frame detection. This has to do with the size of the area in which -@option{combpel} pixels are required to be detected as combed for a frame to be -declared combed. See the @option{combpel} parameter description for more info. -Possible values are any number that is a power of 2 starting at 4 and going up -to 512. - -Default value is @code{16}. - -@item combpel -The number of combed pixels inside any of the @option{blocky} by -@option{blockx} size blocks on the frame for the frame to be detected as -combed. While @option{cthresh} controls how "visible" the combing must be, this -setting controls "how much" combing there must be in any localized area (a -window defined by the @option{blockx} and @option{blocky} settings) on the -frame. Minimum value is @code{0} and maximum is @code{blocky x blockx} (at -which point no frames will ever be detected as combed). This setting is known -as @option{MI} in TFM/VFM vocabulary. - -Default value is @code{80}. -@end table - -@anchor{p/c/n/u/b meaning} -@subsection p/c/n/u/b meaning - -@subsubsection p/c/n - -We assume the following telecined stream: - -@example -Top fields: 1 2 2 3 4 -Bottom fields: 1 2 3 4 4 -@end example - -The numbers correspond to the progressive frame the fields relate to. Here, the -first two frames are progressive, the 3rd and 4th are combed, and so on. - -When @code{fieldmatch} is configured to run a matching from bottom -(@option{field}=@var{bottom}) this is how this input stream get transformed: - -@example -Input stream: - T 1 2 2 3 4 - B 1 2 3 4 4 <-- matching reference - -Matches: c c n n c - -Output stream: - T 1 2 3 4 4 - B 1 2 3 4 4 -@end example - -As a result of the field matching, we can see that some frames get duplicated. -To perform a complete inverse telecine, you need to rely on a decimation filter -after this operation. See for instance the @ref{decimate} filter. - -The same operation now matching from top fields (@option{field}=@var{top}) -looks like this: - -@example -Input stream: - T 1 2 2 3 4 <-- matching reference - B 1 2 3 4 4 - -Matches: c c p p c - -Output stream: - T 1 2 2 3 4 - B 1 2 2 3 4 -@end example - -In these examples, we can see what @var{p}, @var{c} and @var{n} mean; -basically, they refer to the frame and field of the opposite parity: - -@itemize -@item @var{p} matches the field of the opposite parity in the previous frame -@item @var{c} matches the field of the opposite parity in the current frame -@item @var{n} matches the field of the opposite parity in the next frame -@end itemize - -@subsubsection u/b - -The @var{u} and @var{b} matching are a bit special in the sense that they match -from the opposite parity flag. In the following examples, we assume that we are -currently matching the 2nd frame (Top:2, bottom:2). According to the match, a -'x' is placed above and below each matched fields. - -With bottom matching (@option{field}=@var{bottom}): -@example -Match: c p n b u - - x x x x x - Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2 - Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3 - x x x x x - -Output frames: - 2 1 2 2 2 - 2 2 2 1 3 -@end example - -With top matching (@option{field}=@var{top}): -@example -Match: c p n b u - - x x x x x - Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2 - Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3 - x x x x x - -Output frames: - 2 2 2 1 2 - 2 1 3 2 2 -@end example - -@subsection Examples - -Simple IVTC of a top field first telecined stream: -@example -fieldmatch=order=tff:combmatch=none, decimate -@end example - -Advanced IVTC, with fallback on @ref{yadif} for still combed frames: -@example -fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate +# Make the first 5 frames black, then fade in from frame 5-24 +fade=type=in:start_frame=5:nb_frames=20 @end example @section fieldorder @@ -4525,7 +1353,7 @@ The output field order. Valid values are @var{tff} for top field first or @var{b for bottom field first. @end table -The default value is @samp{tff}. +The default value is "tff". The transformation is done by shifting the picture content up or down by one line, and filling the remaining line with appropriate picture content. @@ -4540,7 +1368,7 @@ which is bottom field first. For example: @example -ffmpeg -i in.vob -vf "fieldorder=bff" out.dv +./avconv -i in.vob -vf "fieldorder=order=bff" out.dv @end example @section fifo @@ -4552,7 +1380,6 @@ framework. It does not take parameters. -@anchor{format} @section format Convert the input video to one of the specified pixel formats. @@ -4568,50 +1395,26 @@ A '|'-separated list of pixel format names, such as @end table -@subsection Examples - -@itemize -@item -Convert the input video to the @var{yuv420p} format +Some examples: @example +# Convert the input video to the "yuv420p" format format=pix_fmts=yuv420p -@end example -Convert the input video to any of the formats in the list -@example +# Convert the input video to any of the formats in the list format=pix_fmts=yuv420p|yuv444p|yuv410p @end example -@end itemize @anchor{fps} @section fps -Convert the video to specified constant frame rate by duplicating or dropping +Convert the video to specified constant framerate by duplicating or dropping frames as necessary. It accepts the following parameters: @table @option @item fps -The desired output frame rate. The default is @code{25}. - -@item round -Rounding method. - -Possible values are: -@table @option -@item zero -zero round towards 0 -@item inf -round away from 0 -@item down -round towards -infinity -@item up -round towards +infinity -@item near -round to nearest -@end table -The default is @code{near}. +The desired output framerate. @item start_time Assume the first PTS should be the given value, in seconds. This allows for @@ -4623,27 +1426,6 @@ frames with a negative PTS. @end table -Alternatively, the options can be specified as a flat string: -@var{fps}[:@var{round}]. - -See also the @ref{setpts} filter. - -@subsection Examples - -@itemize -@item -A typical usage in order to set the fps to 25: -@example -fps=fps=25 -@end example - -@item -Sets the fps to 24, using abbreviation and rounding method to round to nearest: -@example -fps=fps=film:round=near -@end example -@end itemize - @section framepack Pack two different video streams into a stereoscopic video, setting proper @@ -4683,30 +1465,19 @@ Some examples: @example # Convert left and right views into a frame-sequential video -ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT +avconv -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT # Convert views into a side-by-side video with the same output resolution as the input -ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT +avconv -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT @end example -@section framestep - -Select one frame every N-th frame. - -This filter accepts the following option: -@table @option -@item step -Select frame after every @code{step} frames. -Allowed values are positive integers higher than 0. Default value is @code{1}. -@end table - @anchor{frei0r} @section frei0r Apply a frei0r effect to the input video. To enable the compilation of this filter, you need to install the frei0r -header and configure FFmpeg with @code{--enable-frei0r}. +header and configure Libav with --enable-frei0r. It accepts the following parameters: @@ -4728,167 +1499,35 @@ A '|'-separated list of parameters to pass to the frei0r effect. A frei0r effect parameter can be a boolean (its value is either "y" or "n"), a double, a color (specified as @var{R}/@var{G}/@var{B}, where @var{R}, @var{G}, and @var{B} are floating point -numbers between 0.0 and 1.0, inclusive) or by a color description specified in the "Color" -section in the ffmpeg-utils manual), a position (specified as @var{X}/@var{Y}, where +numbers between 0.0 and 1.0, inclusive) or by an @code{av_parse_color()} color +description), a position (specified as @var{X}/@var{Y}, where @var{X} and @var{Y} are floating point numbers) and/or a string. The number and types of parameters depend on the loaded effect. If an effect parameter is not specified, the default value is set. -@subsection Examples - -@itemize -@item -Apply the distort0r effect, setting the first two double parameters: +Some examples: @example +# Apply the distort0r effect, setting the first two double parameters frei0r=filter_name=distort0r:filter_params=0.5|0.01 -@end example -@item -Apply the colordistance effect, taking a color as the first parameter: -@example +# Apply the colordistance effect, taking a color as the first parameter frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233 -@end example -@item -Apply the perspective effect, specifying the top left and top right image -positions: -@example +# Apply the perspective effect, specifying the top left and top right +# image positions frei0r=perspective:0.2/0.2|0.8/0.2 @end example -@end itemize For more information, see -@url{http://frei0r.dyne.org} - -@section geq - -The filter accepts the following options: - -@table @option -@item lum_expr, lum -Set the luminance expression. -@item cb_expr, cb -Set the chrominance blue expression. -@item cr_expr, cr -Set the chrominance red expression. -@item alpha_expr, a -Set the alpha expression. -@item red_expr, r -Set the red expression. -@item green_expr, g -Set the green expression. -@item blue_expr, b -Set the blue expression. -@end table - -The colorspace is selected according to the specified options. If one -of the @option{lum_expr}, @option{cb_expr}, or @option{cr_expr} -options is specified, the filter will automatically select a YCbCr -colorspace. If one of the @option{red_expr}, @option{green_expr}, or -@option{blue_expr} options is specified, it will select an RGB -colorspace. - -If one of the chrominance expression is not defined, it falls back on the other -one. If no alpha expression is specified it will evaluate to opaque value. -If none of chrominance expressions are specified, they will evaluate -to the luminance expression. - -The expressions can use the following variables and functions: - -@table @option -@item N -The sequential number of the filtered frame, starting from @code{0}. - -@item X -@item Y -The coordinates of the current sample. - -@item W -@item H -The width and height of the image. - -@item SW -@item SH -Width and height scale depending on the currently filtered plane. It is the -ratio between the corresponding luma plane number of pixels and the current -plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and -@code{0.5,0.5} for chroma planes. - -@item T -Time of the current frame, expressed in seconds. - -@item p(x, y) -Return the value of the pixel at location (@var{x},@var{y}) of the current -plane. - -@item lum(x, y) -Return the value of the pixel at location (@var{x},@var{y}) of the luminance -plane. - -@item cb(x, y) -Return the value of the pixel at location (@var{x},@var{y}) of the -blue-difference chroma plane. Return 0 if there is no such plane. - -@item cr(x, y) -Return the value of the pixel at location (@var{x},@var{y}) of the -red-difference chroma plane. Return 0 if there is no such plane. - -@item r(x, y) -@item g(x, y) -@item b(x, y) -Return the value of the pixel at location (@var{x},@var{y}) of the -red/green/blue component. Return 0 if there is no such component. - -@item alpha(x, y) -Return the value of the pixel at location (@var{x},@var{y}) of the alpha -plane. Return 0 if there is no such plane. -@end table - -For functions, if @var{x} and @var{y} are outside the area, the value will be -automatically clipped to the closer edge. - -@subsection Examples - -@itemize -@item -Flip the image horizontally: -@example -geq=p(W-X\,Y) -@end example - -@item -Generate a bidimensional sine wave, with angle @code{PI/3} and a -wavelength of 100 pixels: -@example -geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128 -@end example - -@item -Generate a fancy enigmatic moving light: -@example -nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128 -@end example - -@item -Generate a quick emboss effect: -@example -format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2' -@end example - -@item -Modify RGB components depending on pixel position: -@example -geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)' -@end example -@end itemize +@url{http://piksel.org/frei0r} @section gradfun Fix the banding artifacts that are sometimes introduced into nearly flat -regions by truncation to 8bit color depth. +regions by truncation to 8bit colordepth. Interpolate the gradients that should go where the bands are, and dither them. @@ -4914,264 +1553,23 @@ values will be clipped to the valid range. @end table -Alternatively, the options can be specified as a flat string: -@var{strength}[:@var{radius}] - -@subsection Examples - -@itemize -@item -Apply the filter with a @code{3.5} strength and radius of @code{8}: @example -gradfun=3.5:8 -@end example +# Default parameters +gradfun=strength=1.2:radius=16 -@item -Specify radius, omitting the strength (which will fall-back to the default -value): -@example -gradfun=radius=8 -@end example - -@end itemize - -@anchor{haldclut} -@section haldclut - -Apply a Hald CLUT to a video stream. - -First input is the video stream to process, and second one is the Hald CLUT. -The Hald CLUT input can be a simple picture or a complete video stream. - -The filter accepts the following options: - -@table @option -@item shortest -Force termination when the shortest input terminates. Default is @code{0}. -@item repeatlast -Continue applying the last CLUT after the end of the stream. A value of -@code{0} disable the filter after the last frame of the CLUT is reached. -Default is @code{1}. -@end table - -@code{haldclut} also has the same interpolation options as @ref{lut3d} (both -filters share the same internals). - -More information about the Hald CLUT can be found on Eskil Steenberg's website -(Hald CLUT author) at @url{http://www.quelsolaar.com/technology/clut.html}. - -@subsection Workflow examples - -@subsubsection Hald CLUT video stream - -Generate an identity Hald CLUT stream altered with various effects: -@example -ffmpeg -f lavfi -i @ref{haldclutsrc}=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut -@end example - -Note: make sure you use a lossless codec. - -Then use it with @code{haldclut} to apply it on some random stream: -@example -ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv -@end example - -The Hald CLUT will be applied to the 10 first seconds (duration of -@file{clut.nut}), then the latest picture of that CLUT stream will be applied -to the remaining frames of the @code{mandelbrot} stream. - -@subsubsection Hald CLUT with preview - -A Hald CLUT is supposed to be a squared image of @code{Level*Level*Level} by -@code{Level*Level*Level} pixels. For a given Hald CLUT, FFmpeg will select the -biggest possible square starting at the top left of the picture. The remaining -padding pixels (bottom or right) will be ignored. This area can be used to add -a preview of the Hald CLUT. - -Typically, the following generated Hald CLUT will be supported by the -@code{haldclut} filter: - -@example -ffmpeg -f lavfi -i @ref{haldclutsrc}=8 -vf " - pad=iw+320 [padded_clut]; - smptebars=s=320x256, split [a][b]; - [padded_clut][a] overlay=W-320:h, curves=color_negative [main]; - [main][b] overlay=W-320" -frames:v 1 clut.png -@end example - -It contains the original and a preview of the effect of the CLUT: SMPTE color -bars are displayed on the right-top, and below the same color bars processed by -the color changes. - -Then, the effect of this Hald CLUT can be visualized with: -@example -ffplay input.mkv -vf "movie=clut.png, [in] haldclut" +# Omitting the radius +gradfun=1.2 @end example @section hflip Flip the input video horizontally. -For example, to horizontally flip the input video with @command{ffmpeg}: +For example, to horizontally flip the input video with @command{avconv}: @example -ffmpeg -i in.avi -vf "hflip" out.avi +avconv -i in.avi -vf "hflip" out.avi @end example -@section histeq -This filter applies a global color histogram equalization on a -per-frame basis. - -It can be used to correct video that has a compressed range of pixel -intensities. The filter redistributes the pixel intensities to -equalize their distribution across the intensity range. It may be -viewed as an "automatically adjusting contrast filter". This filter is -useful only for correcting degraded or poorly captured source -video. - -The filter accepts the following options: - -@table @option -@item strength -Determine the amount of equalization to be applied. As the strength -is reduced, the distribution of pixel intensities more-and-more -approaches that of the input frame. The value must be a float number -in the range [0,1] and defaults to 0.200. - -@item intensity -Set the maximum intensity that can generated and scale the output -values appropriately. The strength should be set as desired and then -the intensity can be limited if needed to avoid washing-out. The value -must be a float number in the range [0,1] and defaults to 0.210. - -@item antibanding -Set the antibanding level. If enabled the filter will randomly vary -the luminance of output pixels by a small amount to avoid banding of -the histogram. Possible values are @code{none}, @code{weak} or -@code{strong}. It defaults to @code{none}. -@end table - -@section histogram - -Compute and draw a color distribution histogram for the input video. - -The computed histogram is a representation of the color component -distribution in an image. - -The filter accepts the following options: - -@table @option -@item mode -Set histogram mode. - -It accepts the following values: -@table @samp -@item levels -Standard histogram that displays the color components distribution in an -image. Displays color graph for each color component. Shows distribution of -the Y, U, V, A or R, G, B components, depending on input format, in the -current frame. Below each graph a color component scale meter is shown. - -@item color -Displays chroma values (U/V color placement) in a two dimensional -graph (which is called a vectorscope). The brighter a pixel in the -vectorscope, the more pixels of the input frame correspond to that pixel -(i.e., more pixels have this chroma value). The V component is displayed on -the horizontal (X) axis, with the leftmost side being V = 0 and the rightmost -side being V = 255. The U component is displayed on the vertical (Y) axis, -with the top representing U = 0 and the bottom representing U = 255. - -The position of a white pixel in the graph corresponds to the chroma value of -a pixel of the input clip. The graph can therefore be used to read the hue -(color flavor) and the saturation (the dominance of the hue in the color). As -the hue of a color changes, it moves around the square. At the center of the -square the saturation is zero, which means that the corresponding pixel has no -color. If the amount of a specific color is increased (while leaving the other -colors unchanged) the saturation increases, and the indicator moves towards -the edge of the square. - -@item color2 -Chroma values in vectorscope, similar as @code{color} but actual chroma values -are displayed. - -@item waveform -Per row/column color component graph. In row mode, the graph on the left side -represents color component value 0 and the right side represents value = 255. -In column mode, the top side represents color component value = 0 and bottom -side represents value = 255. -@end table -Default value is @code{levels}. - -@item level_height -Set height of level in @code{levels}. Default value is @code{200}. -Allowed range is [50, 2048]. - -@item scale_height -Set height of color scale in @code{levels}. Default value is @code{12}. -Allowed range is [0, 40]. - -@item step -Set step for @code{waveform} mode. Smaller values are useful to find out how -many values of the same luminance are distributed across input rows/columns. -Default value is @code{10}. Allowed range is [1, 255]. - -@item waveform_mode -Set mode for @code{waveform}. Can be either @code{row}, or @code{column}. -Default is @code{row}. - -@item waveform_mirror -Set mirroring mode for @code{waveform}. @code{0} means unmirrored, @code{1} -means mirrored. In mirrored mode, higher values will be represented on the left -side for @code{row} mode and at the top for @code{column} mode. Default is -@code{0} (unmirrored). - -@item display_mode -Set display mode for @code{waveform} and @code{levels}. -It accepts the following values: -@table @samp -@item parade -Display separate graph for the color components side by side in -@code{row} waveform mode or one below the other in @code{column} waveform mode -for @code{waveform} histogram mode. For @code{levels} histogram mode, -per color component graphs are placed below each other. - -Using this display mode in @code{waveform} histogram mode makes it easy to -spot color casts in the highlights and shadows of an image, by comparing the -contours of the top and the bottom graphs of each waveform. Since whites, -grays, and blacks are characterized by exactly equal amounts of red, green, -and blue, neutral areas of the picture should display three waveforms of -roughly equal width/height. If not, the correction is easy to perform by -making level adjustments the three waveforms. - -@item overlay -Presents information identical to that in the @code{parade}, except -that the graphs representing color components are superimposed directly -over one another. - -This display mode in @code{waveform} histogram mode makes it easier to spot -relative differences or similarities in overlapping areas of the color -components that are supposed to be identical, such as neutral whites, grays, -or blacks. -@end table -Default is @code{parade}. - -@item levels_mode -Set mode for @code{levels}. Can be either @code{linear}, or @code{logarithmic}. -Default is @code{linear}. -@end table - -@subsection Examples - -@itemize - -@item -Calculate and draw histogram: -@example -ffplay -i input -vf histogram -@end example - -@end itemize - -@anchor{hqdn3d} @section hqdn3d This is a high precision/quality 3d denoise filter. It aims to reduce @@ -5198,167 +1596,6 @@ A floating point number which specifies chroma temporal strength. It defaults to @var{luma_tmp}*@var{chroma_spatial}/@var{luma_spatial}. @end table -@section hue - -Modify the hue and/or the saturation of the input. - -It accepts the following parameters: - -@table @option -@item h -Specify the hue angle as a number of degrees. It accepts an expression, -and defaults to "0". - -@item s -Specify the saturation in the [-10,10] range. It accepts an expression and -defaults to "1". - -@item H -Specify the hue angle as a number of radians. It accepts an -expression, and defaults to "0". - -@item b -Specify the brightness in the [-10,10] range. It accepts an expression and -defaults to "0". -@end table - -@option{h} and @option{H} are mutually exclusive, and can't be -specified at the same time. - -The @option{b}, @option{h}, @option{H} and @option{s} option values are -expressions containing the following constants: - -@table @option -@item n -frame count of the input frame starting from 0 - -@item pts -presentation timestamp of the input frame expressed in time base units - -@item r -frame rate of the input video, NAN if the input frame rate is unknown - -@item t -timestamp expressed in seconds, NAN if the input timestamp is unknown - -@item tb -time base of the input video -@end table - -@subsection Examples - -@itemize -@item -Set the hue to 90 degrees and the saturation to 1.0: -@example -hue=h=90:s=1 -@end example - -@item -Same command but expressing the hue in radians: -@example -hue=H=PI/2:s=1 -@end example - -@item -Rotate hue and make the saturation swing between 0 -and 2 over a period of 1 second: -@example -hue="H=2*PI*t: s=sin(2*PI*t)+1" -@end example - -@item -Apply a 3 seconds saturation fade-in effect starting at 0: -@example -hue="s=min(t/3\,1)" -@end example - -The general fade-in expression can be written as: -@example -hue="s=min(0\, max((t-START)/DURATION\, 1))" -@end example - -@item -Apply a 3 seconds saturation fade-out effect starting at 5 seconds: -@example -hue="s=max(0\, min(1\, (8-t)/3))" -@end example - -The general fade-out expression can be written as: -@example -hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))" -@end example - -@end itemize - -@subsection Commands - -This filter supports the following commands: -@table @option -@item b -@item s -@item h -@item H -Modify the hue and/or the saturation and/or brightness of the input video. -The command accepts the same syntax of the corresponding option. - -If the specified expression is not valid, it is kept at its current -value. -@end table - -@section idet - -Detect video interlacing type. - -This filter tries to detect if the input is interlaced or progressive, -top or bottom field first. - -The filter accepts the following options: - -@table @option -@item intl_thres -Set interlacing threshold. -@item prog_thres -Set progressive threshold. -@end table - -@section il - -Deinterleave or interleave fields. - -This filter allows one to process interlaced images fields without -deinterlacing them. Deinterleaving splits the input frame into 2 -fields (so called half pictures). Odd lines are moved to the top -half of the output image, even lines to the bottom half. -You can process (filter) them independently and then re-interleave them. - -The filter accepts the following options: - -@table @option -@item luma_mode, l -@item chroma_mode, c -@item alpha_mode, a -Available values for @var{luma_mode}, @var{chroma_mode} and -@var{alpha_mode} are: - -@table @samp -@item none -Do nothing. - -@item deinterleave, d -Deinterleave fields, placing one above the other. - -@item interleave, i -Interleave fields. Reverse the effect of deinterleaving. -@end table -Default value is @code{none}. - -@item luma_swap, ls -@item chroma_swap, cs -@item alpha_swap, as -Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is @code{0}. -@end table - @section interlace Simple interlacing filter from progressive contents. This interleaves upper (or @@ -5386,95 +1623,6 @@ This determines whether the interlaced frame is taken from the even (tff - default) or odd (bff) lines of the progressive frame. @end table -@section kerndeint - -Deinterlace input video by applying Donald Graft's adaptive kernel -deinterling. Work on interlaced parts of a video to produce -progressive frames. - -The description of the accepted parameters follows. - -@table @option -@item thresh -Set the threshold which affects the filter's tolerance when -determining if a pixel line must be processed. It must be an integer -in the range [0,255] and defaults to 10. A value of 0 will result in -applying the process on every pixels. - -@item map -Paint pixels exceeding the threshold value to white if set to 1. -Default is 0. - -@item order -Set the fields order. Swap fields if set to 1, leave fields alone if -0. Default is 0. - -@item sharp -Enable additional sharpening if set to 1. Default is 0. - -@item twoway -Enable twoway sharpening if set to 1. Default is 0. -@end table - -@subsection Examples - -@itemize -@item -Apply default values: -@example -kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0 -@end example - -@item -Enable additional sharpening: -@example -kerndeint=sharp=1 -@end example - -@item -Paint processed pixels in white: -@example -kerndeint=map=1 -@end example -@end itemize - -@anchor{lut3d} -@section lut3d - -Apply a 3D LUT to an input video. - -The filter accepts the following options: - -@table @option -@item file -Set the 3D LUT file name. - -Currently supported formats: -@table @samp -@item 3dl -AfterEffects -@item cube -Iridas -@item dat -DaVinci -@item m3d -Pandora -@end table -@item interp -Select interpolation mode. - -Available values are: - -@table @samp -@item nearest -Use values from the nearest defined point. -@item trilinear -Interpolate values using the 8 points defining a cube. -@item tetrahedral -Interpolate values using a tetrahedron. -@end table -@end table - @section lut, lutrgb, lutyuv Compute a look-up table for binding each pixel component input value @@ -5485,30 +1633,19 @@ to an RGB input video. These filters accept the following parameters: @table @option -@item c0 -set first pixel component expression -@item c1 -set second pixel component expression -@item c2 -set third pixel component expression -@item c3 -set fourth pixel component expression, corresponds to the alpha component +@item @var{c0} (first pixel component) +@item @var{c1} (second pixel component) +@item @var{c2} (third pixel component) +@item @var{c3} (fourth pixel component, corresponds to the alpha component) -@item r -set red component expression -@item g -set green component expression -@item b -set blue component expression -@item a -alpha component expression +@item @var{r} (red component) +@item @var{g} (green component) +@item @var{b} (blue component) +@item @var{a} (alpha component) -@item y -set Y/luminance component expression -@item u -set U/Cb component expression -@item v -set V/Cr component expression +@item @var{y} (Y/luminance component) +@item @var{u} (U/Cb component) +@item @var{v} (V/Cr component) @end table Each of them specifies the expression to use for computing the lookup table for @@ -5523,8 +1660,11 @@ The @var{lut} filter requires either YUV or RGB pixel formats in input, The expressions can contain the following constants and functions: @table @option -@item w -@item h +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). + +@item w, h The input width and height. @item val @@ -5558,253 +1698,35 @@ expression All expressions default to "val". -@subsection Examples - -@itemize -@item -Negate input video: +Some examples: @example +# Negate input video lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val" lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val" -@end example -The above is the same as: -@example +# The above is the same as lutrgb="r=negval:g=negval:b=negval" lutyuv="y=negval:u=negval:v=negval" -@end example -@item -Negate luminance: -@example -lutyuv=y=negval -@end example +# Negate luminance +lutyuv=negval -@item -Remove chroma components, turning the video into a graytone image: -@example +# Remove chroma components, turning the video into a graytone image lutyuv="u=128:v=128" -@end example -@item -Apply a luma burning effect: -@example +# Apply a luma burning effect lutyuv="y=2*val" -@end example -@item -Remove green and blue components: -@example +# Remove green and blue components lutrgb="g=0:b=0" -@end example -@item -Set a constant alpha channel value on input: -@example +# Set a constant alpha channel value on input format=rgba,lutrgb=a="maxval-minval/2" -@end example -@item -Correct luminance gamma by a factor of 0.5: -@example +# Correct luminance gamma by a factor of 0.5 lutyuv=y=gammaval(0.5) @end example -@item -Discard least significant bits of luma: -@example -lutyuv=y='bitand(val, 128+64+32)' -@end example -@end itemize - -@section mergeplanes - -Merge color channel components from several video streams. - -The filter accepts up to 4 input streams, and merge selected input -planes to the output video. - -This filter accepts the following options: -@table @option -@item mapping -Set input to output plane mapping. Default is @code{0}. - -The mappings is specified as a bitmap. It should be specified as a -hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the -mapping for the first plane of the output stream. 'A' sets the number of -the input stream to use (from 0 to 3), and 'a' the plane number of the -corresponding input to use (from 0 to 3). The rest of the mappings is -similar, 'Bb' describes the mapping for the output stream second -plane, 'Cc' describes the mapping for the output stream third plane and -'Dd' describes the mapping for the output stream fourth plane. - -@item format -Set output pixel format. Default is @code{yuva444p}. -@end table - -@subsection Examples - -@itemize -@item -Merge three gray video streams of same width and height into single video stream: -@example -[a0][a1][a2]mergeplanes=0x001020:yuv444p -@end example - -@item -Merge 1st yuv444p stream and 2nd gray video stream into yuva444p video stream: -@example -[a0][a1]mergeplanes=0x00010210:yuva444p -@end example - -@item -Swap Y and A plane in yuva444p stream: -@example -format=yuva444p,mergeplanes=0x03010200:yuva444p -@end example - -@item -Swap U and V plane in yuv420p stream: -@example -format=yuv420p,mergeplanes=0x000201:yuv420p -@end example - -@item -Cast a rgb24 clip to yuv444p: -@example -format=rgb24,mergeplanes=0x000102:yuv444p -@end example -@end itemize - -@section mcdeint - -Apply motion-compensation deinterlacing. - -It needs one field per frame as input and must thus be used together -with yadif=1/3 or equivalent. - -This filter accepts the following options: -@table @option -@item mode -Set the deinterlacing mode. - -It accepts one of the following values: -@table @samp -@item fast -@item medium -@item slow -use iterative motion estimation -@item extra_slow -like @samp{slow}, but use multiple reference frames. -@end table -Default value is @samp{fast}. - -@item parity -Set the picture field parity assumed for the input video. It must be -one of the following values: - -@table @samp -@item 0, tff -assume top field first -@item 1, bff -assume bottom field first -@end table - -Default value is @samp{bff}. - -@item qp -Set per-block quantization parameter (QP) used by the internal -encoder. - -Higher values should result in a smoother motion vector field but less -optimal individual vectors. Default value is 1. -@end table - -@section mp - -Apply an MPlayer filter to the input video. - -This filter provides a wrapper around some of the filters of -MPlayer/MEncoder. - -This wrapper is considered experimental. Some of the wrapped filters -may not work properly and we may drop support for them, as they will -be implemented natively into FFmpeg. Thus you should avoid -depending on them when writing portable scripts. - -The filter accepts the parameters: -@var{filter_name}[:=]@var{filter_params} - -@var{filter_name} is the name of a supported MPlayer filter, -@var{filter_params} is a string containing the parameters accepted by -the named filter. - -The list of the currently supported filters follows: -@table @var -@item eq2 -@item eq -@item fspp -@item ilpack -@item pp7 -@item softpulldown -@item uspp -@end table - -The parameter syntax and behavior for the listed filters are the same -of the corresponding MPlayer filters. For detailed instructions check -the "VIDEO FILTERS" section in the MPlayer manual. - -@subsection Examples - -@itemize -@item -Adjust gamma, brightness, contrast: -@example -mp=eq2=1.0:2:0.5 -@end example -@end itemize - -See also mplayer(1), @url{http://www.mplayerhq.hu/}. - -@section mpdecimate - -Drop frames that do not differ greatly from the previous frame in -order to reduce frame rate. - -The main use of this filter is for very-low-bitrate encoding -(e.g. streaming over dialup modem), but it could in theory be used for -fixing movies that were inverse-telecined incorrectly. - -A description of the accepted options follows. - -@table @option -@item max -Set the maximum number of consecutive frames which can be dropped (if -positive), or the minimum interval between dropped frames (if -negative). If the value is 0, the frame is dropped unregarding the -number of previous sequentially dropped frames. - -Default value is 0. - -@item hi -@item lo -@item frac -Set the dropping threshold values. - -Values for @option{hi} and @option{lo} are for 8x8 pixel blocks and -represent actual pixel value differences, so a threshold of 64 -corresponds to 1 unit of difference for each pixel, or the same spread -out differently over the block. - -A frame is a candidate for dropping if no 8x8 blocks differ by more -than a threshold of @option{hi}, and if no more than @option{frac} blocks (1 -meaning the whole image) differ by more than a threshold of @option{lo}. - -Default value for @option{hi} is 64*12, default value for @option{lo} is -64*5, and default value for @option{frac} is 0.33. -@end table - - @section negate Negate input video. @@ -5826,71 +1748,15 @@ apix_fmts=yuv420p|monow|rgb24". @end table -@subsection Examples - -@itemize -@item -Force libavfilter to use a format different from @var{yuv420p} for the -input to the vflip filter: +Some examples: @example +# Force libavfilter to use a format different from "yuv420p" for the +# input to the vflip filter noformat=pix_fmts=yuv420p,vflip -@end example -@item -Convert the input video to any of the formats not contained in the list: -@example +# Convert the input video to any of the formats not contained in the list noformat=yuv420p|yuv444p|yuv410p @end example -@end itemize - -@section noise - -Add noise on video input frame. - -The filter accepts the following options: - -@table @option -@item all_seed -@item c0_seed -@item c1_seed -@item c2_seed -@item c3_seed -Set noise seed for specific pixel component or all pixel components in case -of @var{all_seed}. Default value is @code{123457}. - -@item all_strength, alls -@item c0_strength, c0s -@item c1_strength, c1s -@item c2_strength, c2s -@item c3_strength, c3s -Set noise strength for specific pixel component or all pixel components in case -@var{all_strength}. Default value is @code{0}. Allowed range is [0, 100]. - -@item all_flags, allf -@item c0_flags, c0f -@item c1_flags, c1f -@item c2_flags, c2f -@item c3_flags, c3f -Set pixel component flags or set flags for all components if @var{all_flags}. -Available values for component flags are: -@table @samp -@item a -averaged temporal noise (smoother) -@item p -mix random noise with a (semi)regular pattern -@item t -temporal noise (noise pattern changes between frames) -@item u -uniform noise (gaussian otherwise) -@end table -@end table - -@subsection Examples - -Add temporal and uniform noise to input video: -@example -noise=alls=20:allf=t+u -@end example @section null @@ -5901,7 +1767,7 @@ Pass the video source unchanged to the output. Apply a video transform using libopencv. To enable this filter, install the libopencv library and headers and -configure FFmpeg with @code{--enable-libopencv}. +configure Libav with --enable-libopencv. It accepts the following parameters: @@ -6010,15 +1876,30 @@ video on which the second input is overlayed. It accepts the following parameters: -A description of the accepted options follows. +@table @option + +@item x +The horizontal position of the left edge of the overlaid video on the main video. + +@item y +The vertical position of the top edge of the overlaid video on the main video. + +@end table + +The parameters are expressions containing the following parameters: @table @option -@item x -@item y -Set the expression for the x and y coordinates of the overlayed video -on the main video. Default value is "0" for both expressions. In case -the expression is invalid, it is set to a huge value (meaning that the -overlay will not be displayed within the output visible area). +@item main_w, main_h +The main input width and height. + +@item W, H +These are the same as @var{main_w} and @var{main_h}. + +@item overlay_w, overlay_h +The overlay input width and height. + +@item w, h +These are the same as @var{overlay_w} and @var{overlay_h}. @item eof_action The action to take when EOF is encountered on the secondary input; it accepts @@ -6033,229 +1914,40 @@ End both streams. Pass the main input through. @end table -@item eval -Set when the expressions for @option{x}, and @option{y} are evaluated. - -It accepts the following values: -@table @samp -@item init -only evaluate expressions once during the filter initialization or -when a command is processed - -@item frame -evaluate expressions for each incoming frame @end table -Default value is @samp{frame}. - -@item shortest -If set to 1, force the output to terminate when the shortest input -terminates. Default value is 0. - -@item format -Set the format for the output video. - -It accepts the following values: -@table @samp -@item yuv420 -force YUV420 output - -@item yuv422 -force YUV422 output - -@item yuv444 -force YUV444 output - -@item rgb -force RGB output -@end table - -Default value is @samp{yuv420}. - -@item rgb @emph{(deprecated)} -If set to 1, force the filter to accept inputs in the RGB -color space. Default value is 0. This option is deprecated, use -@option{format} instead. - -@item repeatlast -If set to 1, force the filter to draw the last overlay frame over the -main input until the end of the stream. A value of 0 disables this -behavior. Default value is 1. -@end table - -The @option{x}, and @option{y} expressions can contain the following -parameters. - -@table @option -@item main_w, W -@item main_h, H -The main input width and height. - -@item overlay_w, w -@item overlay_h, h -The overlay input width and height. - -@item x -@item y -The computed values for @var{x} and @var{y}. They are evaluated for -each new frame. - -@item hsub -@item vsub -horizontal and vertical chroma subsample values of the output -format. For example for the pixel format "yuv422p" @var{hsub} is 2 and -@var{vsub} is 1. - -@item n -the number of input frame, starting from 0 - -@item pos -the position in the file of the input frame, NAN if unknown - -@item t -The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown. - -@end table - -Note that the @var{n}, @var{pos}, @var{t} variables are available only -when evaluation is done @emph{per frame}, and will evaluate to NAN -when @option{eval} is set to @samp{init}. - Be aware that frames are taken from each input video in timestamp -order, hence, if their initial timestamps differ, it is a good idea +order, hence, if their initial timestamps differ, it is a a good idea to pass the two inputs through a @var{setpts=PTS-STARTPTS} filter to have them begin in the same zero timestamp, as the example for the @var{movie} filter does. -You can chain together more overlays but you should test the -efficiency of such approach. - -@subsection Commands - -This filter supports the following commands: -@table @option -@item x -@item y -Modify the x and y of the overlay input. -The command accepts the same syntax of the corresponding option. - -If the specified expression is not valid, it is kept at its current -value. -@end table - -@subsection Examples - -@itemize -@item -Draw the overlay at 10 pixels from the bottom right corner of the main -video: -@example -overlay=main_w-overlay_w-10:main_h-overlay_h-10 -@end example - -Using named options the example above becomes: +Some examples: @example +# Draw the overlay at 10 pixels from the bottom right +# corner of the main video overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10 -@end example -@item -Insert a transparent PNG logo in the bottom left corner of the input, -using the @command{ffmpeg} tool with the @code{-filter_complex} option: -@example -ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output -@end example +# Insert a transparent PNG logo in the bottom left corner of the input +avconv -i input -i logo -filter_complex 'overlay=x=10:y=main_h-overlay_h-10' output -@item -Insert 2 different transparent PNG logos (second logo on bottom -right corner) using the @command{ffmpeg} tool: -@example -ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output -@end example +# Insert 2 different transparent PNG logos (second logo on bottom +# right corner) +avconv -i input -i logo1 -i logo2 -filter_complex +'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output -@item -Add a transparent color layer on top of the main video; @code{WxH} -must specify the size of the main input to the overlay filter: -@example -color=color=red@@.3:size=WxH [over]; [in][over] overlay [out] -@end example +# Add a transparent color layer on top of the main video; +# WxH specifies the size of the main input to the overlay filter +color=red@.3:WxH [over]; [in][over] overlay [out] -@item -Play an original video and a filtered version (here with the deshake -filter) side by side using the @command{ffplay} tool: -@example -ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w' -@end example - -The above command is the same as: -@example -ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w' -@end example - -@item -Make a sliding overlay appearing from the left to the right top part of the -screen starting since time 2: -@example -overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0 -@end example - -@item -Compose output by putting two input videos side to side: -@example -ffmpeg -i left.avi -i right.avi -filter_complex " -nullsrc=size=200x100 [background]; -[0:v] setpts=PTS-STARTPTS, scale=100x100 [left]; -[1:v] setpts=PTS-STARTPTS, scale=100x100 [right]; -[background][left] overlay=shortest=1 [background+left]; -[background+left][right] overlay=shortest=1:x=100 [left+right] -" -@end example - -@item -Mask 10-20 seconds of a video by applying the delogo filter to a section -@example -ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k +# Mask 10-20 seconds of a video by applying the delogo filter to a section +avconv -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]' masked.avi @end example -@item -Chain several overlays in cascade: -@example -nullsrc=s=200x200 [bg]; -testsrc=s=100x100, split=4 [in0][in1][in2][in3]; -[in0] lutrgb=r=0, [bg] overlay=0:0 [mid0]; -[in1] lutrgb=g=0, [mid0] overlay=100:0 [mid1]; -[in2] lutrgb=b=0, [mid1] overlay=0:100 [mid2]; -[in3] null, [mid2] overlay=100:100 [out0] -@end example - -@end itemize - -@section owdenoise - -Apply Overcomplete Wavelet denoiser. - -The filter accepts the following options: - -@table @option -@item depth -Set depth. - -Larger depth values will denoise lower frequency components more, but -slow down filtering. - -Must be an int in the range 8-16, default is @code{8}. - -@item luma_strength, ls -Set luma strength. - -Must be a double value in the range 0-1000, default is @code{1.0}. - -@item chroma_strength, cs -Set chroma strength. - -Must be a double value in the range 0-1000, default is @code{1.0}. -@end table +You can chain together more overlays but the efficiency of such +approach is yet to be tested. @section pad @@ -6265,19 +1957,19 @@ provided @var{x}, @var{y} coordinates. It accepts the following parameters: @table @option -@item width, w -@item height, h -Specify an expression for the size of the output image with the -paddings added. If the value for @var{width} or @var{height} is 0, the -corresponding input size is used for the output. +@item width, height + +Specify the size of the output image with the paddings added. If the +value for @var{width} or @var{height} is 0, the corresponding input size +is used for the output. The @var{width} expression can reference the value set by the @var{height} expression, and vice versa. The default value of @var{width} and @var{height} is 0. -@item x -@item y +@item x, y + Specify the offsets to place the input image at within the padded area, with respect to the top/left border of the output image. @@ -6287,210 +1979,71 @@ expression, and vice versa. The default value of @var{x} and @var{y} is 0. @item color -Specify the color of the padded area. For the syntax of this option, -check the "Color" section in the ffmpeg-utils manual. + +Specify the color of the padded area. It can be the name of a color +(case insensitive match) or an 0xRRGGBB[AA] sequence. The default value of @var{color} is "black". + @end table -The value for the @var{width}, @var{height}, @var{x}, and @var{y} -options are expressions containing the following constants: +The parameters @var{width}, @var{height}, @var{x}, and @var{y} are +expressions containing the following constants: @table @option -@item in_w -@item in_h +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). + +@item in_w, in_h The input video width and height. -@item iw -@item ih +@item iw, ih These are the same as @var{in_w} and @var{in_h}. -@item out_w -@item out_h +@item out_w, out_h The output width and height (the size of the padded area), as specified by the @var{width} and @var{height} expressions. -@item ow -@item oh +@item ow, oh These are the same as @var{out_w} and @var{out_h}. -@item x -@item y +@item x, y The x and y offsets as specified by the @var{x} and @var{y} expressions, or NAN if not yet specified. @item a -same as @var{iw} / @var{ih} +The input display aspect ratio, same as @var{iw} / @var{ih}. -@item sar -input sample aspect ratio - -@item dar -input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar} - -@item hsub -@item vsub +@item hsub, vsub The horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. @end table -@subsection Examples +Some examples: -@itemize -@item -Add paddings with the color "violet" to the input video. The output video -size is 640x480, and the top-left corner of the input video is placed at -column 0, row 40 -@example -pad=640:480:0:40:violet -@end example - -The example above is equivalent to the following command: @example +# Add paddings with the color "violet" to the input video. The output video +# size is 640x480, and the top-left corner of the input video is placed at +# column 0, row 40 pad=width=640:height=480:x=0:y=40:color=violet -@end example -@item -Pad the input to get an output with dimensions increased by 3/2, -and put the input video at the center of the padded area: -@example +# Pad the input to get an output with dimensions increased by 3/2, +# and put the input video at the center of the padded area pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" -@end example -@item -Pad the input to get a squared output with size equal to the maximum -value between the input width and height, and put the input video at -the center of the padded area: -@example +# Pad the input to get a squared output with size equal to the maximum +# value between the input width and height, and put the input video at +# the center of the padded area pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" -@end example -@item -Pad the input to get a final w/h ratio of 16:9: -@example +# Pad the input to get a final w/h ratio of 16:9 pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" -@end example -@item -In case of anamorphic video, in order to set the output display aspect -correctly, it is necessary to use @var{sar} in the expression, -according to the relation: -@example -(ih * X / ih) * sar = output_dar -X = output_dar / sar -@end example - -Thus the previous example needs to be modified to: -@example -pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2" -@end example - -@item -Double the output size and put the input video in the bottom-right -corner of the output padded area: -@example +# Double the output size and put the input video in the bottom-right +# corner of the output padded area pad="2*iw:2*ih:ow-iw:oh-ih" @end example -@end itemize - -@section perspective - -Correct perspective of video not recorded perpendicular to the screen. - -A description of the accepted parameters follows. - -@table @option -@item x0 -@item y0 -@item x1 -@item y1 -@item x2 -@item y2 -@item x3 -@item y3 -Set coordinates expression for top left, top right, bottom left and bottom right corners. -Default values are @code{0:0:W:0:0:H:W:H} with which perspective will remain unchanged. - -The expressions can use the following variables: - -@table @option -@item W -@item H -the width and height of video frame. -@end table - -@item interpolation -Set interpolation for perspective correction. - -It accepts the following values: -@table @samp -@item linear -@item cubic -@end table - -Default value is @samp{linear}. -@end table - -@section phase - -Delay interlaced video by one field time so that the field order changes. - -The intended use is to fix PAL movies that have been captured with the -opposite field order to the film-to-video transfer. - -A description of the accepted parameters follows. - -@table @option -@item mode -Set phase mode. - -It accepts the following values: -@table @samp -@item t -Capture field order top-first, transfer bottom-first. -Filter will delay the bottom field. - -@item b -Capture field order bottom-first, transfer top-first. -Filter will delay the top field. - -@item p -Capture and transfer with the same field order. This mode only exists -for the documentation of the other options to refer to, but if you -actually select it, the filter will faithfully do nothing. - -@item a -Capture field order determined automatically by field flags, transfer -opposite. -Filter selects among @samp{t} and @samp{b} modes on a frame by frame -basis using field flags. If no field information is available, -then this works just like @samp{u}. - -@item u -Capture unknown or varying, transfer opposite. -Filter selects among @samp{t} and @samp{b} on a frame by frame basis by -analyzing the images and selecting the alternative that produces best -match between the fields. - -@item T -Capture top-first, transfer unknown or varying. -Filter selects among @samp{t} and @samp{p} using image analysis. - -@item B -Capture bottom-first, transfer unknown or varying. -Filter selects among @samp{b} and @samp{p} using image analysis. - -@item A -Capture determined by field flags, transfer unknown or varying. -Filter selects among @samp{t}, @samp{b} and @samp{p} using field flags and -image analysis. If no field information is available, then this works just -like @samp{U}. This is the default mode. - -@item U -Both capture and transfer unknown or varying. -Filter selects among @samp{t}, @samp{b} and @samp{p} using image analysis only. -@end table -@end table @section pixdesctest @@ -6504,857 +2057,24 @@ format=monow, pixdesctest can be used to test the monowhite pixel format descriptor definition. -@section pp - -Enable the specified chain of postprocessing subfilters using libpostproc. This -library should be automatically selected with a GPL build (@code{--enable-gpl}). -Subfilters must be separated by '/' and can be disabled by prepending a '-'. -Each subfilter and some options have a short and a long name that can be used -interchangeably, i.e. dr/dering are the same. - -The filters accept the following options: - -@table @option -@item subfilters -Set postprocessing subfilters string. -@end table - -All subfilters share common options to determine their scope: - -@table @option -@item a/autoq -Honor the quality commands for this subfilter. - -@item c/chrom -Do chrominance filtering, too (default). - -@item y/nochrom -Do luminance filtering only (no chrominance). - -@item n/noluma -Do chrominance filtering only (no luminance). -@end table - -These options can be appended after the subfilter name, separated by a '|'. - -Available subfilters are: - -@table @option -@item hb/hdeblock[|difference[|flatness]] -Horizontal deblocking filter -@table @option -@item difference -Difference factor where higher values mean more deblocking (default: @code{32}). -@item flatness -Flatness threshold where lower values mean more deblocking (default: @code{39}). -@end table - -@item vb/vdeblock[|difference[|flatness]] -Vertical deblocking filter -@table @option -@item difference -Difference factor where higher values mean more deblocking (default: @code{32}). -@item flatness -Flatness threshold where lower values mean more deblocking (default: @code{39}). -@end table - -@item ha/hadeblock[|difference[|flatness]] -Accurate horizontal deblocking filter -@table @option -@item difference -Difference factor where higher values mean more deblocking (default: @code{32}). -@item flatness -Flatness threshold where lower values mean more deblocking (default: @code{39}). -@end table - -@item va/vadeblock[|difference[|flatness]] -Accurate vertical deblocking filter -@table @option -@item difference -Difference factor where higher values mean more deblocking (default: @code{32}). -@item flatness -Flatness threshold where lower values mean more deblocking (default: @code{39}). -@end table -@end table - -The horizontal and vertical deblocking filters share the difference and -flatness values so you cannot set different horizontal and vertical -thresholds. - -@table @option -@item h1/x1hdeblock -Experimental horizontal deblocking filter - -@item v1/x1vdeblock -Experimental vertical deblocking filter - -@item dr/dering -Deringing filter - -@item tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise reducer -@table @option -@item threshold1 -larger -> stronger filtering -@item threshold2 -larger -> stronger filtering -@item threshold3 -larger -> stronger filtering -@end table - -@item al/autolevels[:f/fullyrange], automatic brightness / contrast correction -@table @option -@item f/fullyrange -Stretch luminance to @code{0-255}. -@end table - -@item lb/linblenddeint -Linear blend deinterlacing filter that deinterlaces the given block by -filtering all lines with a @code{(1 2 1)} filter. - -@item li/linipoldeint -Linear interpolating deinterlacing filter that deinterlaces the given block by -linearly interpolating every second line. - -@item ci/cubicipoldeint -Cubic interpolating deinterlacing filter deinterlaces the given block by -cubically interpolating every second line. - -@item md/mediandeint -Median deinterlacing filter that deinterlaces the given block by applying a -median filter to every second line. - -@item fd/ffmpegdeint -FFmpeg deinterlacing filter that deinterlaces the given block by filtering every -second line with a @code{(-1 4 2 4 -1)} filter. - -@item l5/lowpass5 -Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given -block by filtering all lines with a @code{(-1 2 6 2 -1)} filter. - -@item fq/forceQuant[|quantizer] -Overrides the quantizer table from the input with the constant quantizer you -specify. -@table @option -@item quantizer -Quantizer to use -@end table - -@item de/default -Default pp filter combination (@code{hb|a,vb|a,dr|a}) - -@item fa/fast -Fast pp filter combination (@code{h1|a,v1|a,dr|a}) - -@item ac -High quality pp filter combination (@code{ha|a|128|7,va|a,dr|a}) -@end table - -@subsection Examples - -@itemize -@item -Apply horizontal and vertical deblocking, deringing and automatic -brightness/contrast: -@example -pp=hb/vb/dr/al -@end example - -@item -Apply default filters without brightness/contrast correction: -@example -pp=de/-al -@end example - -@item -Apply default filters and temporal denoiser: -@example -pp=default/tmpnoise|1|2|3 -@end example - -@item -Apply deblocking on luminance only, and switch vertical deblocking on or off -automatically depending on available CPU time: -@example -pp=hb|y/vb|a -@end example -@end itemize - -@section psnr - -Obtain the average, maximum and minimum PSNR (Peak Signal to Noise -Ratio) between two input videos. - -This filter takes in input two input videos, the first input is -considered the "main" source and is passed unchanged to the -output. The second input is used as a "reference" video for computing -the PSNR. - -Both video inputs must have the same resolution and pixel format for -this filter to work correctly. Also it assumes that both inputs -have the same number of frames, which are compared one by one. - -The obtained average PSNR is printed through the logging system. - -The filter stores the accumulated MSE (mean squared error) of each -frame, and at the end of the processing it is averaged across all frames -equally, and the following formula is applied to obtain the PSNR: - -@example -PSNR = 10*log10(MAX^2/MSE) -@end example - -Where MAX is the average of the maximum values of each component of the -image. - -The description of the accepted parameters follows. - -@table @option -@item stats_file, f -If specified the filter will use the named file to save the PSNR of -each individual frame. -@end table - -The file printed if @var{stats_file} is selected, contains a sequence of -key/value pairs of the form @var{key}:@var{value} for each compared -couple of frames. - -A description of each shown parameter follows: - -@table @option -@item n -sequential number of the input frame, starting from 1 - -@item mse_avg -Mean Square Error pixel-by-pixel average difference of the compared -frames, averaged over all the image components. - -@item mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a -Mean Square Error pixel-by-pixel average difference of the compared -frames for the component specified by the suffix. - -@item psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a -Peak Signal to Noise ratio of the compared frames for the component -specified by the suffix. -@end table - -For example: -@example -movie=ref_movie.mpg, setpts=PTS-STARTPTS [main]; -[main][ref] psnr="stats_file=stats.log" [out] -@end example - -On this example the input file being processed is compared with the -reference file @file{ref_movie.mpg}. The PSNR of each individual frame -is stored in @file{stats.log}. - -@anchor{pullup} -@section pullup - -Pulldown reversal (inverse telecine) filter, capable of handling mixed -hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps progressive -content. - -The pullup filter is designed to take advantage of future context in making -its decisions. This filter is stateless in the sense that it does not lock -onto a pattern to follow, but it instead looks forward to the following -fields in order to identify matches and rebuild progressive frames. - -To produce content with an even framerate, insert the fps filter after -pullup, use @code{fps=24000/1001} if the input frame rate is 29.97fps, -@code{fps=24} for 30fps and the (rare) telecined 25fps input. - -The filter accepts the following options: - -@table @option -@item jl -@item jr -@item jt -@item jb -These options set the amount of "junk" to ignore at the left, right, top, and -bottom of the image, respectively. Left and right are in units of 8 pixels, -while top and bottom are in units of 2 lines. -The default is 8 pixels on each side. - -@item sb -Set the strict breaks. Setting this option to 1 will reduce the chances of -filter generating an occasional mismatched frame, but it may also cause an -excessive number of frames to be dropped during high motion sequences. -Conversely, setting it to -1 will make filter match fields more easily. -This may help processing of video where there is slight blurring between -the fields, but may also cause there to be interlaced frames in the output. -Default value is @code{0}. - -@item mp -Set the metric plane to use. It accepts the following values: -@table @samp -@item l -Use luma plane. - -@item u -Use chroma blue plane. - -@item v -Use chroma red plane. -@end table - -This option may be set to use chroma plane instead of the default luma plane -for doing filter's computations. This may improve accuracy on very clean -source material, but more likely will decrease accuracy, especially if there -is chroma noise (rainbow effect) or any grayscale video. -The main purpose of setting @option{mp} to a chroma plane is to reduce CPU -load and make pullup usable in realtime on slow machines. -@end table - -For best results (without duplicated frames in the output file) it is -necessary to change the output frame rate. For example, to inverse -telecine NTSC input: -@example -ffmpeg -i input -vf pullup -r 24000/1001 ... -@end example - -@section removelogo - -Suppress a TV station logo, using an image file to determine which -pixels comprise the logo. It works by filling in the pixels that -comprise the logo with neighboring pixels. - -The filter accepts the following options: - -@table @option -@item filename, f -Set the filter bitmap file, which can be any image format supported by -libavformat. The width and height of the image file must match those of the -video stream being processed. -@end table - -Pixels in the provided bitmap image with a value of zero are not -considered part of the logo, non-zero pixels are considered part of -the logo. If you use white (255) for the logo and black (0) for the -rest, you will be safe. For making the filter bitmap, it is -recommended to take a screen capture of a black frame with the logo -visible, and then using a threshold filter followed by the erode -filter once or twice. - -If needed, little splotches can be fixed manually. Remember that if -logo pixels are not covered, the filter quality will be much -reduced. Marking too many pixels as part of the logo does not hurt as -much, but it will increase the amount of blurring needed to cover over -the image and will destroy more information than necessary, and extra -pixels will slow things down on a large logo. - -@section rotate - -Rotate video by an arbitrary angle expressed in radians. - -The filter accepts the following options: - -A description of the optional parameters follows. -@table @option -@item angle, a -Set an expression for the angle by which to rotate the input video -clockwise, expressed as a number of radians. A negative value will -result in a counter-clockwise rotation. By default it is set to "0". - -This expression is evaluated for each frame. - -@item out_w, ow -Set the output width expression, default value is "iw". -This expression is evaluated just once during configuration. - -@item out_h, oh -Set the output height expression, default value is "ih". -This expression is evaluated just once during configuration. - -@item bilinear -Enable bilinear interpolation if set to 1, a value of 0 disables -it. Default value is 1. - -@item fillcolor, c -Set the color used to fill the output area not covered by the rotated -image. For the generalsyntax of this option, check the "Color" section in the -ffmpeg-utils manual. If the special value "none" is selected then no -background is printed (useful for example if the background is never shown). - -Default value is "black". -@end table - -The expressions for the angle and the output size can contain the -following constants and functions: - -@table @option -@item n -sequential number of the input frame, starting from 0. It is always NAN -before the first frame is filtered. - -@item t -time in seconds of the input frame, it is set to 0 when the filter is -configured. It is always NAN before the first frame is filtered. - -@item hsub -@item vsub -horizontal and vertical chroma subsample values. For example for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. - -@item in_w, iw -@item in_h, ih -the input video width and height - -@item out_w, ow -@item out_h, oh -the output width and height, that is the size of the padded area as -specified by the @var{width} and @var{height} expressions - -@item rotw(a) -@item roth(a) -the minimal width/height required for completely containing the input -video rotated by @var{a} radians. - -These are only available when computing the @option{out_w} and -@option{out_h} expressions. -@end table - -@subsection Examples - -@itemize -@item -Rotate the input by PI/6 radians clockwise: -@example -rotate=PI/6 -@end example - -@item -Rotate the input by PI/6 radians counter-clockwise: -@example -rotate=-PI/6 -@end example - -@item -Rotate the input by 45 degrees clockwise: -@example -rotate=45*PI/180 -@end example - -@item -Apply a constant rotation with period T, starting from an angle of PI/3: -@example -rotate=PI/3+2*PI*t/T -@end example - -@item -Make the input video rotation oscillating with a period of T -seconds and an amplitude of A radians: -@example -rotate=A*sin(2*PI/T*t) -@end example - -@item -Rotate the video, output size is chosen so that the whole rotating -input video is always completely contained in the output: -@example -rotate='2*PI*t:ow=hypot(iw,ih):oh=ow' -@end example - -@item -Rotate the video, reduce the output size so that no background is ever -shown: -@example -rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none -@end example -@end itemize - -@subsection Commands - -The filter supports the following commands: - -@table @option -@item a, angle -Set the angle expression. -The command accepts the same syntax of the corresponding option. - -If the specified expression is not valid, it is kept at its current -value. -@end table - -@section sab - -Apply Shape Adaptive Blur. - -The filter accepts the following options: - -@table @option -@item luma_radius, lr -Set luma blur filter strength, must be a value in range 0.1-4.0, default -value is 1.0. A greater value will result in a more blurred image, and -in slower processing. - -@item luma_pre_filter_radius, lpfr -Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default -value is 1.0. - -@item luma_strength, ls -Set luma maximum difference between pixels to still be considered, must -be a value in the 0.1-100.0 range, default value is 1.0. - -@item chroma_radius, cr -Set chroma blur filter strength, must be a value in range 0.1-4.0. A -greater value will result in a more blurred image, and in slower -processing. - -@item chroma_pre_filter_radius, cpfr -Set chroma pre-filter radius, must be a value in the 0.1-2.0 range. - -@item chroma_strength, cs -Set chroma maximum difference between pixels to still be considered, -must be a value in the 0.1-100.0 range. -@end table - -Each chroma option value, if not explicitly specified, is set to the -corresponding luma option value. - @anchor{scale} @section scale -Scale (resize) the input video, using the libswscale library. - -The scale filter forces the output display aspect ratio to be the same -of the input, by changing the output sample aspect ratio. - -If the input image format is different from the format requested by -the next filter, the scale filter will convert the input to the -requested format. - -@subsection Options -The filter accepts the following options, or any of the options -supported by the libswscale scaler. - -See @ref{scaler_options,,the ffmpeg-scaler manual,ffmpeg-scaler} for -the complete list of scaler options. - -@table @option -@item width, w -@item height, h -Set the output video dimension expression. Default value is the input -dimension. - -If the value is 0, the input width is used for the output. - -If one of the values is -1, the scale filter will use a value that -maintains the aspect ratio of the input image, calculated from the -other specified dimension. If both of them are -1, the input size is -used - -If one of the values is -n with n > 1, the scale filter will also use a value -that maintains the aspect ratio of the input image, calculated from the other -specified dimension. After that it will, however, make sure that the calculated -dimension is divisible by n and adjust the value if necessary. - -See below for the list of accepted constants for use in the dimension -expression. - -@item interl -Set the interlacing mode. It accepts the following values: - -@table @samp -@item 1 -Force interlaced aware scaling. - -@item 0 -Do not apply interlaced scaling. - -@item -1 -Select interlaced aware scaling depending on whether the source frames -are flagged as interlaced or not. -@end table - -Default value is @samp{0}. - -@item flags -Set libswscale scaling flags. See -@ref{sws_flags,,the ffmpeg-scaler manual,ffmpeg-scaler} for the -complete list of values. If not explicitly specified the filter applies -the default flags. - -@item size, s -Set the video size. For the syntax of this option, check the "Video size" -section in the ffmpeg-utils manual. - -@item in_color_matrix -@item out_color_matrix -Set in/output YCbCr color space type. - -This allows the autodetected value to be overridden as well as allows forcing -a specific value used for the output and encoder. - -If not specified, the color space type depends on the pixel format. - -Possible values: - -@table @samp -@item auto -Choose automatically. - -@item bt709 -Format conforming to International Telecommunication Union (ITU) -Recommendation BT.709. - -@item fcc -Set color space conforming to the United States Federal Communications -Commission (FCC) Code of Federal Regulations (CFR) Title 47 (2003) 73.682 (a). - -@item bt601 -Set color space conforming to: - -@itemize -@item -ITU Radiocommunication Sector (ITU-R) Recommendation BT.601 - -@item -ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G - -@item -Society of Motion Picture and Television Engineers (SMPTE) ST 170:2004 - -@end itemize - -@item smpte240m -Set color space conforming to SMPTE ST 240:1999. -@end table - -@item in_range -@item out_range -Set in/output YCbCr sample range. - -This allows the autodetected value to be overridden as well as allows forcing -a specific value used for the output and encoder. If not specified, the -range depends on the pixel format. Possible values: - -@table @samp -@item auto -Choose automatically. - -@item jpeg/full/pc -Set full range (0-255 in case of 8-bit luma). - -@item mpeg/tv -Set "MPEG" range (16-235 in case of 8-bit luma). -@end table - -@item force_original_aspect_ratio -Enable decreasing or increasing output video width or height if necessary to -keep the original aspect ratio. Possible values: - -@table @samp -@item disable -Scale the video as specified and disable this feature. - -@item decrease -The output video dimensions will automatically be decreased if needed. - -@item increase -The output video dimensions will automatically be increased if needed. - -@end table - -One useful instance of this option is that when you know a specific device's -maximum allowed resolution, you can use this to limit the output video to -that, while retaining the aspect ratio. For example, device A allows -1280x720 playback, and your video is 1920x800. Using this option (set it to -decrease) and specifying 1280x720 to the command line makes the output -1280x533. - -Please note that this is a different thing than specifying -1 for @option{w} -or @option{h}, you still need to specify the output resolution for this option -to work. - -@end table - -The values of the @option{w} and @option{h} options are expressions -containing the following constants: - -@table @var -@item in_w -@item in_h -The input width and height - -@item iw -@item ih -These are the same as @var{in_w} and @var{in_h}. - -@item out_w -@item out_h -The output (scaled) width and height - -@item ow -@item oh -These are the same as @var{out_w} and @var{out_h} - -@item a -The same as @var{iw} / @var{ih} - -@item sar -input sample aspect ratio - -@item dar -The input display aspect ratio. Calculated from @code{(iw / ih) * sar}. - -@item hsub -@item vsub -horizontal and vertical input chroma subsample values. For example for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. - -@item ohsub -@item ovsub -horizontal and vertical output chroma subsample values. For example for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. -@end table - -@subsection Examples - -@itemize -@item -Scale the input video to a size of 200x100 -@example -scale=w=200:h=100 -@end example - -This is equivalent to: -@example -scale=200:100 -@end example - -or: -@example -scale=200x100 -@end example - -@item -Specify a size abbreviation for the output size: -@example -scale=qcif -@end example - -which can also be written as: -@example -scale=size=qcif -@end example - -@item -Scale the input to 2x: -@example -scale=w=2*iw:h=2*ih -@end example - -@item -The above is the same as: -@example -scale=2*in_w:2*in_h -@end example - -@item -Scale the input to 2x with forced interlaced scaling: -@example -scale=2*iw:2*ih:interl=1 -@end example - -@item -Scale the input to half size: -@example -scale=w=iw/2:h=ih/2 -@end example - -@item -Increase the width, and set the height to the same size: -@example -scale=3/2*iw:ow -@end example - -@item -Seek Greek harmony: -@example -scale=iw:1/PHI*iw -scale=ih*PHI:ih -@end example - -@item -Increase the height, and set the width to 3/2 of the height: -@example -scale=w=3/2*oh:h=3/5*ih -@end example - -@item -Increase the size, making the size a multiple of the chroma -subsample values: -@example -scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" -@end example - -@item -Increase the width to a maximum of 500 pixels, -keeping the same aspect ratio as the input: -@example -scale=w='min(500\, iw*3/2):h=-1' -@end example -@end itemize - -@section separatefields - -The @code{separatefields} takes a frame-based video input and splits -each frame into its components fields, producing a new half height clip -with twice the frame rate and twice the frame count. - -This filter use field-dominance information in frame to decide which -of each pair of fields to place first in the output. -If it gets it wrong use @ref{setfield} filter before @code{separatefields} filter. - -@section setdar, setsar - -The @code{setdar} filter sets the Display Aspect Ratio for the filter -output video. - -This is done by changing the specified Sample (aka Pixel) Aspect -Ratio, according to the following equation: -@example -@var{DAR} = @var{HORIZONTAL_RESOLUTION} / @var{VERTICAL_RESOLUTION} * @var{SAR} -@end example - -Keep in mind that the @code{setdar} filter does not modify the pixel -dimensions of the video frame. Also, the display aspect ratio set by -this filter may be changed by later filters in the filterchain, -e.g. in case of scaling or if another "setdar" or a "setsar" filter is -applied. - -The @code{setsar} filter sets the Sample (aka Pixel) Aspect Ratio for -the filter output video. - -Note that as a consequence of the application of this filter, the -output display aspect ratio will change according to the equation -above. - -Keep in mind that the sample aspect ratio set by the @code{setsar} -filter may be changed by later filters in the filterchain, e.g. if -another "setsar" or a "setdar" filter is applied. +Scale the input video and/or convert the image format. It accepts the following parameters: @table @option -@item r, ratio, dar (@code{setdar} only), sar (@code{setsar} only) -Set the aspect ratio used by the filter. -The parameter can be a floating point number string, an expression, or -a string of the form @var{num}:@var{den}, where @var{num} and -@var{den} are the numerator and denominator of the aspect ratio. If -the parameter is not specified, it is assumed the value "0". -In case the form "@var{num}:@var{den}" is used, the @code{:} character -should be escaped. +@item w +The output video width. -@item max -Set the maximum integer value to use for expressing numerator and -denominator when reducing the expressed aspect ratio to a rational. -Default value is @code{100}. +@item h +The output video height. @end table -The parameter @var{sar} is an expression containing +The parameters @var{w} and @var{h} are expressions containing the following constants: @table @option @@ -7362,2299 +2082,98 @@ the following constants: These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio). -@item w, h +@item in_w, in_h The input width and height. +@item iw, ih +These are the same as @var{in_w} and @var{in_h}. + +@item out_w, out_h +The output (cropped) width and height. + +@item ow, oh +These are the same as @var{out_w} and @var{out_h}. + @item a -These are the same as @var{w} / @var{h}. +This is the same as @var{iw} / @var{ih}. @item sar -The input sample aspect ratio. +input sample aspect ratio @item dar -The input display aspect ratio. It is the same as -(@var{w} / @var{h}) * @var{sar}. +The input display aspect ratio; it is the same as +(@var{iw} / @var{ih}) * @var{sar}. @item hsub, vsub -Horizontal and vertical chroma subsample values. For example, for the +The horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. @end table -@subsection Examples +If the input image format is different from the format requested by +the next filter, the scale filter will convert the input to the +requested format. -@itemize +If the value for @var{w} or @var{h} is 0, the respective input +size is used for the output. -@item -To change the display aspect ratio to 16:9, specify one of the following: -@example -setdar=dar=1.77777 -setdar=dar=16/9 -setdar=dar=1.77777 -@end example +If the value for @var{w} or @var{h} is -1, the scale filter will use, for the +respective output size, a value that maintains the aspect ratio of the input +image. -@item -To change the sample aspect ratio to 10:11, specify: -@example -setsar=sar=10/11 -@end example - -@item -To set a display aspect ratio of 16:9, and specify a maximum integer value of -1000 in the aspect ratio reduction, use the command: -@example -setdar=ratio=16/9:max=1000 -@end example - -@end itemize - -@anchor{setfield} -@section setfield - -Force field for the output video frame. - -The @code{setfield} filter marks the interlace type field for the -output frames. It does not change the input frame, but only sets the -corresponding property, which affects how the frame is treated by -following filters (e.g. @code{fieldorder} or @code{yadif}). - -The filter accepts the following options: - -@table @option - -@item mode -Available values are: - -@table @samp -@item auto -Keep the same field property. - -@item bff -Mark the frame as bottom-field-first. - -@item tff -Mark the frame as top-field-first. - -@item prog -Mark the frame as progressive. -@end table -@end table - -@section showinfo - -Show a line containing various information for each input video frame. -The input video is not modified. - -The shown line contains a sequence of key/value pairs of the form -@var{key}:@var{value}. - -It accepts the following parameters: - -@table @option -@item n -The (sequential) number of the input frame, starting from 0. - -@item pts -The Presentation TimeStamp of the input frame, expressed as a number of -time base units. The time base unit depends on the filter input pad. - -@item pts_time -The Presentation TimeStamp of the input frame, expressed as a number of -seconds. - -@item pos -The position of the frame in the input stream, or -1 if this information is -unavailable and/or meaningless (for example in case of synthetic video). - -@item fmt -The pixel format name. - -@item sar -The sample aspect ratio of the input frame, expressed in the form -@var{num}/@var{den}. - -@item s -The size of the input frame. For the syntax of this option, check the "Video size" -section in the ffmpeg-utils manual. - -@item i -The type of interlaced mode ("P" for "progressive", "T" for top field first, "B" -for bottom field first). - -@item iskey -This is 1 if the frame is a key frame, 0 otherwise. - -@item type -The picture type of the input frame ("I" for an I-frame, "P" for a -P-frame, "B" for a B-frame, or "?" for an unknown type). -Also refer to the documentation of the @code{AVPictureType} enum and of -the @code{av_get_picture_type_char} function defined in -@file{libavutil/avutil.h}. - -@item checksum -The Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame. - -@item plane_checksum -The Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, -expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3}]". -@end table - -@section shuffleplanes - -Reorder and/or duplicate video planes. - -It accepts the following parameters: - -@table @option - -@item map0 -The index of the input plane to be used as the first output plane. - -@item map1 -The index of the input plane to be used as the second output plane. - -@item map2 -The index of the input plane to be used as the third output plane. - -@item map3 -The index of the input plane to be used as the fourth output plane. - -@end table - -The first plane has the index 0. The default is to keep the input unchanged. - -Swap the second and third planes of the input: -@example -ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT -@end example - -@anchor{smartblur} -@section smartblur - -Blur the input video without impacting the outlines. - -It accepts the following options: - -@table @option -@item luma_radius, lr -Set the luma radius. The option value must be a float number in -the range [0.1,5.0] that specifies the variance of the gaussian filter -used to blur the image (slower if larger). Default value is 1.0. - -@item luma_strength, ls -Set the luma strength. The option value must be a float number -in the range [-1.0,1.0] that configures the blurring. A value included -in [0.0,1.0] will blur the image whereas a value included in -[-1.0,0.0] will sharpen the image. Default value is 1.0. - -@item luma_threshold, lt -Set the luma threshold used as a coefficient to determine -whether a pixel should be blurred or not. The option value must be an -integer in the range [-30,30]. A value of 0 will filter all the image, -a value included in [0,30] will filter flat areas and a value included -in [-30,0] will filter edges. Default value is 0. - -@item chroma_radius, cr -Set the chroma radius. The option value must be a float number in -the range [0.1,5.0] that specifies the variance of the gaussian filter -used to blur the image (slower if larger). Default value is 1.0. - -@item chroma_strength, cs -Set the chroma strength. The option value must be a float number -in the range [-1.0,1.0] that configures the blurring. A value included -in [0.0,1.0] will blur the image whereas a value included in -[-1.0,0.0] will sharpen the image. Default value is 1.0. - -@item chroma_threshold, ct -Set the chroma threshold used as a coefficient to determine -whether a pixel should be blurred or not. The option value must be an -integer in the range [-30,30]. A value of 0 will filter all the image, -a value included in [0,30] will filter flat areas and a value included -in [-30,0] will filter edges. Default value is 0. -@end table - -If a chroma option is not explicitly set, the corresponding luma value -is set. - -@section stereo3d - -Convert between different stereoscopic image formats. - -The filters accept the following options: - -@table @option -@item in -Set stereoscopic image format of input. - -Available values for input image formats are: -@table @samp -@item sbsl -side by side parallel (left eye left, right eye right) - -@item sbsr -side by side crosseye (right eye left, left eye right) - -@item sbs2l -side by side parallel with half width resolution -(left eye left, right eye right) - -@item sbs2r -side by side crosseye with half width resolution -(right eye left, left eye right) - -@item abl -above-below (left eye above, right eye below) - -@item abr -above-below (right eye above, left eye below) - -@item ab2l -above-below with half height resolution -(left eye above, right eye below) - -@item ab2r -above-below with half height resolution -(right eye above, left eye below) - -@item al -alternating frames (left eye first, right eye second) - -@item ar -alternating frames (right eye first, left eye second) - -Default value is @samp{sbsl}. -@end table - -@item out -Set stereoscopic image format of output. - -Available values for output image formats are all the input formats as well as: -@table @samp -@item arbg -anaglyph red/blue gray -(red filter on left eye, blue filter on right eye) - -@item argg -anaglyph red/green gray -(red filter on left eye, green filter on right eye) - -@item arcg -anaglyph red/cyan gray -(red filter on left eye, cyan filter on right eye) - -@item arch -anaglyph red/cyan half colored -(red filter on left eye, cyan filter on right eye) - -@item arcc -anaglyph red/cyan color -(red filter on left eye, cyan filter on right eye) - -@item arcd -anaglyph red/cyan color optimized with the least squares projection of dubois -(red filter on left eye, cyan filter on right eye) - -@item agmg -anaglyph green/magenta gray -(green filter on left eye, magenta filter on right eye) - -@item agmh -anaglyph green/magenta half colored -(green filter on left eye, magenta filter on right eye) - -@item agmc -anaglyph green/magenta colored -(green filter on left eye, magenta filter on right eye) - -@item agmd -anaglyph green/magenta color optimized with the least squares projection of dubois -(green filter on left eye, magenta filter on right eye) - -@item aybg -anaglyph yellow/blue gray -(yellow filter on left eye, blue filter on right eye) - -@item aybh -anaglyph yellow/blue half colored -(yellow filter on left eye, blue filter on right eye) - -@item aybc -anaglyph yellow/blue colored -(yellow filter on left eye, blue filter on right eye) - -@item aybd -anaglyph yellow/blue color optimized with the least squares projection of dubois -(yellow filter on left eye, blue filter on right eye) - -@item irl -interleaved rows (left eye has top row, right eye starts on next row) - -@item irr -interleaved rows (right eye has top row, left eye starts on next row) - -@item ml -mono output (left eye only) - -@item mr -mono output (right eye only) -@end table - -Default value is @samp{arcd}. -@end table - -@subsection Examples - -@itemize -@item -Convert input video from side by side parallel to anaglyph yellow/blue dubois: -@example -stereo3d=sbsl:aybd -@end example - -@item -Convert input video from above bellow (left eye above, right eye below) to side by side crosseye. -@example -stereo3d=abl:sbsr -@end example -@end itemize - -@section spp - -Apply a simple postprocessing filter that compresses and decompresses the image -at several (or - in the case of @option{quality} level @code{6} - all) shifts -and average the results. - -The filter accepts the following options: - -@table @option -@item quality -Set quality. This option defines the number of levels for averaging. It accepts -an integer in the range 0-6. If set to @code{0}, the filter will have no -effect. A value of @code{6} means the higher quality. For each increment of -that value the speed drops by a factor of approximately 2. Default value is -@code{3}. - -@item qp -Force a constant quantization parameter. If not set, the filter will use the QP -from the video stream (if available). - -@item mode -Set thresholding mode. Available modes are: - -@table @samp -@item hard -Set hard thresholding (default). -@item soft -Set soft thresholding (better de-ringing effect, but likely blurrier). -@end table - -@item use_bframe_qp -Enable the use of the QP from the B-Frames if set to @code{1}. Using this -option may cause flicker since the B-Frames have often larger QP. Default is -@code{0} (not enabled). -@end table - -@anchor{subtitles} -@section subtitles - -Draw subtitles on top of input video using the libass library. - -To enable compilation of this filter you need to configure FFmpeg with -@code{--enable-libass}. This filter also requires a build with libavcodec and -libavformat to convert the passed subtitles file to ASS (Advanced Substation -Alpha) subtitles format. - -The filter accepts the following options: - -@table @option -@item filename, f -Set the filename of the subtitle file to read. It must be specified. - -@item original_size -Specify the size of the original video, the video for which the ASS file -was composed. For the syntax of this option, check the "Video size" section in -the ffmpeg-utils manual. Due to a misdesign in ASS aspect ratio arithmetic, -this is necessary to correctly scale the fonts if the aspect ratio has been -changed. - -@item charenc -Set subtitles input character encoding. @code{subtitles} filter only. Only -useful if not UTF-8. - -@item stream_index, si -Set subtitles stream index. @code{subtitles} filter only. -@end table - -If the first key is not specified, it is assumed that the first value -specifies the @option{filename}. - -For example, to render the file @file{sub.srt} on top of the input -video, use the command: -@example -subtitles=sub.srt -@end example - -which is equivalent to: -@example -subtitles=filename=sub.srt -@end example - -To render the default subtitles stream from file @file{video.mkv}, use: -@example -subtitles=video.mkv -@end example - -To render the second subtitles stream from that file, use: -@example -subtitles=video.mkv:si=1 -@end example - -@section super2xsai - -Scale the input by 2x and smooth using the Super2xSaI (Scale and -Interpolate) pixel art scaling algorithm. - -Useful for enlarging pixel art images without reducing sharpness. - -@section swapuv -Swap U & V plane. - -@section telecine - -Apply telecine process to the video. - -This filter accepts the following options: - -@table @option -@item first_field -@table @samp -@item top, t -top field first -@item bottom, b -bottom field first -The default value is @code{top}. -@end table - -@item pattern -A string of numbers representing the pulldown pattern you wish to apply. -The default value is @code{23}. -@end table - -@example -Some typical patterns: - -NTSC output (30i): -27.5p: 32222 -24p: 23 (classic) -24p: 2332 (preferred) -20p: 33 -18p: 334 -16p: 3444 - -PAL output (25i): -27.5p: 12222 -24p: 222222222223 ("Euro pulldown") -16.67p: 33 -16p: 33333334 -@end example - -@section thumbnail -Select the most representative frame in a given sequence of consecutive frames. - -The filter accepts the following options: - -@table @option -@item n -Set the frames batch size to analyze; in a set of @var{n} frames, the filter -will pick one of them, and then handle the next batch of @var{n} frames until -the end. Default is @code{100}. -@end table - -Since the filter keeps track of the whole frames sequence, a bigger @var{n} -value will result in a higher memory usage, so a high value is not recommended. - -@subsection Examples - -@itemize -@item -Extract one picture each 50 frames: -@example -thumbnail=50 -@end example - -@item -Complete example of a thumbnail creation with @command{ffmpeg}: -@example -ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png -@end example -@end itemize - -@section tile - -Tile several successive frames together. - -The filter accepts the following options: - -@table @option - -@item layout -Set the grid size (i.e. the number of lines and columns). For the syntax of -this option, check the "Video size" section in the ffmpeg-utils manual. - -@item nb_frames -Set the maximum number of frames to render in the given area. It must be less -than or equal to @var{w}x@var{h}. The default value is @code{0}, meaning all -the area will be used. - -@item margin -Set the outer border margin in pixels. - -@item padding -Set the inner border thickness (i.e. the number of pixels between frames). For -more advanced padding options (such as having different values for the edges), -refer to the pad video filter. - -@item color -Specify the color of the unused areaFor the syntax of this option, check the -"Color" section in the ffmpeg-utils manual. The default value of @var{color} -is "black". -@end table - -@subsection Examples - -@itemize -@item -Produce 8x8 PNG tiles of all keyframes (@option{-skip_frame nokey}) in a movie: -@example -ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png -@end example -The @option{-vsync 0} is necessary to prevent @command{ffmpeg} from -duplicating each output frame to accommodate the originally detected frame -rate. - -@item -Display @code{5} pictures in an area of @code{3x2} frames, -with @code{7} pixels between them, and @code{2} pixels of initial margin, using -mixed flat and named options: -@example -tile=3x2:nb_frames=5:padding=7:margin=2 -@end example -@end itemize - -@section tinterlace - -Perform various types of temporal field interlacing. - -Frames are counted starting from 1, so the first input frame is -considered odd. - -The filter accepts the following options: - -@table @option - -@item mode -Specify the mode of the interlacing. This option can also be specified -as a value alone. See below for a list of values for this option. - -Available values are: - -@table @samp -@item merge, 0 -Move odd frames into the upper field, even into the lower field, -generating a double height frame at half frame rate. - -@item drop_odd, 1 -Only output even frames, odd frames are dropped, generating a frame with -unchanged height at half frame rate. - -@item drop_even, 2 -Only output odd frames, even frames are dropped, generating a frame with -unchanged height at half frame rate. - -@item pad, 3 -Expand each frame to full height, but pad alternate lines with black, -generating a frame with double height at the same input frame rate. - -@item interleave_top, 4 -Interleave the upper field from odd frames with the lower field from -even frames, generating a frame with unchanged height at half frame rate. - -@item interleave_bottom, 5 -Interleave the lower field from odd frames with the upper field from -even frames, generating a frame with unchanged height at half frame rate. - -@item interlacex2, 6 -Double frame rate with unchanged height. Frames are inserted each -containing the second temporal field from the previous input frame and -the first temporal field from the next input frame. This mode relies on -the top_field_first flag. Useful for interlaced video displays with no -field synchronisation. -@end table - -Numeric values are deprecated but are accepted for backward -compatibility reasons. - -Default mode is @code{merge}. - -@item flags -Specify flags influencing the filter process. - -Available value for @var{flags} is: - -@table @option -@item low_pass_filter, vlfp -Enable vertical low-pass filtering in the filter. -Vertical low-pass filtering is required when creating an interlaced -destination from a progressive source which contains high-frequency -vertical detail. Filtering will reduce interlace 'twitter' and Moire -patterning. - -Vertical low-pass filtering can only be enabled for @option{mode} -@var{interleave_top} and @var{interleave_bottom}. - -@end table -@end table - -@section transpose - -Transpose rows with columns in the input video and optionally flip it. - -It accepts the following parameters: - -@table @option - -@item dir -Specify the transposition direction. - -Can assume the following values: -@table @samp -@item 0, 4, cclock_flip -Rotate by 90 degrees counterclockwise and vertically flip (default), that is: -@example -L.R L.l -. . -> . . -l.r R.r -@end example - -@item 1, 5, clock -Rotate by 90 degrees clockwise, that is: -@example -L.R l.L -. . -> . . -l.r r.R -@end example - -@item 2, 6, cclock -Rotate by 90 degrees counterclockwise, that is: -@example -L.R R.r -. . -> . . -l.r L.l -@end example - -@item 3, 7, clock_flip -Rotate by 90 degrees clockwise and vertically flip, that is: -@example -L.R r.R -. . -> . . -l.r l.L -@end example -@end table - -For values between 4-7, the transposition is only done if the input -video geometry is portrait and not landscape. These values are -deprecated, the @code{passthrough} option should be used instead. - -Numerical values are deprecated, and should be dropped in favor of -symbolic constants. - -@item passthrough -Do not apply the transposition if the input geometry matches the one -specified by the specified value. It accepts the following values: -@table @samp -@item none -Always apply transposition. -@item portrait -Preserve portrait geometry (when @var{height} >= @var{width}). -@item landscape -Preserve landscape geometry (when @var{width} >= @var{height}). -@end table - -Default value is @code{none}. -@end table - -For example to rotate by 90 degrees clockwise and preserve portrait -layout: -@example -transpose=dir=1:passthrough=portrait -@end example - -The command above can also be specified as: -@example -transpose=1:portrait -@end example - -@section trim -Trim the input so that the output contains one continuous subpart of the input. - -It accepts the following parameters: -@table @option -@item start -Specify the time of the start of the kept section, i.e. the frame with the -timestamp @var{start} will be the first frame in the output. - -@item end -Specify the time of the first frame that will be dropped, i.e. the frame -immediately preceding the one with the timestamp @var{end} will be the last -frame in the output. - -@item start_pts -This is the same as @var{start}, except this option sets the start timestamp -in timebase units instead of seconds. - -@item end_pts -This is the same as @var{end}, except this option sets the end timestamp -in timebase units instead of seconds. - -@item duration -The maximum duration of the output in seconds. - -@item start_frame -The number of the first frame that should be passed to the output. - -@item end_frame -The number of the first frame that should be dropped. -@end table - -@option{start}, @option{end}, @option{duration} are expressed as time -duration specifications, check the "Time duration" section in the -ffmpeg-utils manual. - -Note that the first two sets of the start/end options and the @option{duration} -option look at the frame timestamp, while the _frame variants simply count the -frames that pass through the filter. Also note that this filter does not modify -the timestamps. If you wish for the output timestamps to start at zero, insert a -setpts filter after the trim filter. - -If multiple start or end options are set, this filter tries to be greedy and -keep all the frames that match at least one of the specified constraints. To keep -only the part that matches all the constraints at once, chain multiple trim -filters. - -The defaults are such that all the input is kept. So it is possible to set e.g. -just the end values to keep everything before the specified time. - -Examples: -@itemize -@item -Drop everything except the second minute of input: -@example -ffmpeg -i INPUT -vf trim=60:120 -@end example - -@item -Keep only the first second: -@example -ffmpeg -i INPUT -vf trim=duration=1 -@end example - -@end itemize - - -@section unsharp - -Sharpen or blur the input video. - -It accepts the following parameters: - -@table @option -@item luma_msize_x, lx -Set the luma matrix horizontal size. It must be an odd integer between -3 and 63. The default value is 5. - -@item luma_msize_y, ly -Set the luma matrix vertical size. It must be an odd integer between 3 -and 63. The default value is 5. - -@item luma_amount, la -Set the luma effect strength. It must be a floating point number, reasonable -values lay between -1.5 and 1.5. - -Negative values will blur the input video, while positive values will -sharpen it, a value of zero will disable the effect. - -Default value is 1.0. - -@item chroma_msize_x, cx -Set the chroma matrix horizontal size. It must be an odd integer -between 3 and 63. The default value is 5. - -@item chroma_msize_y, cy -Set the chroma matrix vertical size. It must be an odd integer -between 3 and 63. The default value is 5. - -@item chroma_amount, ca -Set the chroma effect strength. It must be a floating point number, reasonable -values lay between -1.5 and 1.5. - -Negative values will blur the input video, while positive values will -sharpen it, a value of zero will disable the effect. - -Default value is 0.0. - -@item opencl -If set to 1, specify using OpenCL capabilities, only available if -FFmpeg was configured with @code{--enable-opencl}. Default value is 0. - -@end table - -All parameters are optional and default to the equivalent of the -string '5:5:1.0:5:5:0.0'. - -@subsection Examples - -@itemize -@item -Apply strong luma sharpen effect: -@example -unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5 -@end example - -@item -Apply a strong blur of both luma and chroma parameters: -@example -unsharp=7:7:-2:7:7:-2 -@end example -@end itemize - -@anchor{vidstabdetect} -@section vidstabdetect - -Analyze video stabilization/deshaking. Perform pass 1 of 2, see -@ref{vidstabtransform} for pass 2. - -This filter generates a file with relative translation and rotation -transform information about subsequent frames, which is then used by -the @ref{vidstabtransform} filter. - -To enable compilation of this filter you need to configure FFmpeg with -@code{--enable-libvidstab}. - -This filter accepts the following options: - -@table @option -@item result -Set the path to the file used to write the transforms information. -Default value is @file{transforms.trf}. - -@item shakiness -Set how shaky the video is and how quick the camera is. It accepts an -integer in the range 1-10, a value of 1 means little shakiness, a -value of 10 means strong shakiness. Default value is 5. - -@item accuracy -Set the accuracy of the detection process. It must be a value in the -range 1-15. A value of 1 means low accuracy, a value of 15 means high -accuracy. Default value is 15. - -@item stepsize -Set stepsize of the search process. The region around minimum is -scanned with 1 pixel resolution. Default value is 6. - -@item mincontrast -Set minimum contrast. Below this value a local measurement field is -discarded. Must be a floating point value in the range 0-1. Default -value is 0.3. - -@item tripod -Set reference frame number for tripod mode. - -If enabled, the motion of the frames is compared to a reference frame -in the filtered stream, identified by the specified number. The idea -is to compensate all movements in a more-or-less static scene and keep -the camera view absolutely still. - -If set to 0, it is disabled. The frames are counted starting from 1. - -@item show -Show fields and transforms in the resulting frames. It accepts an -integer in the range 0-2. Default value is 0, which disables any -visualization. -@end table - -@subsection Examples - -@itemize -@item -Use default values: -@example -vidstabdetect -@end example - -@item -Analyze strongly shaky movie and put the results in file -@file{mytransforms.trf}: -@example -vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf" -@end example - -@item -Visualize the result of internal transformations in the resulting -video: -@example -vidstabdetect=show=1 -@end example - -@item -Analyze a video with medium shakiness using @command{ffmpeg}: -@example -ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi -@end example -@end itemize - -@anchor{vidstabtransform} -@section vidstabtransform - -Video stabilization/deshaking: pass 2 of 2, -see @ref{vidstabdetect} for pass 1. - -Read a file with transform information for each frame and -apply/compensate them. Together with the @ref{vidstabdetect} -filter this can be used to deshake videos. See also -@url{http://public.hronopik.de/vid.stab}. It is important to also use -the unsharp filter, see below. - -To enable compilation of this filter you need to configure FFmpeg with -@code{--enable-libvidstab}. - -@subsection Options - -@table @option -@item input -Set path to the file used to read the transforms. Default value is -@file{transforms.trf}). - -@item smoothing -Set the number of frames (value*2 + 1) used for lowpass filtering the -camera movements. Default value is 10. - -For example a number of 10 means that 21 frames are used (10 in the -past and 10 in the future) to smoothen the motion in the video. A -larger values leads to a smoother video, but limits the acceleration -of the camera (pan/tilt movements). 0 is a special case where a -static camera is simulated. - -@item optalgo -Set the camera path optimization algorithm. - -Accepted values are: -@table @samp -@item gauss -gaussian kernel low-pass filter on camera motion (default) -@item avg -averaging on transformations -@end table - -@item maxshift -Set maximal number of pixels to translate frames. Default value is -1, -meaning no limit. - -@item maxangle -Set maximal angle in radians (degree*PI/180) to rotate frames. Default -value is -1, meaning no limit. - -@item crop -Specify how to deal with borders that may be visible due to movement -compensation. - -Available values are: -@table @samp -@item keep -keep image information from previous frame (default) -@item black -fill the border black -@end table - -@item invert -Invert transforms if set to 1. Default value is 0. - -@item relative -Consider transforms as relative to previsou frame if set to 1, -absolute if set to 0. Default value is 0. - -@item zoom -Set percentage to zoom. A positive value will result in a zoom-in -effect, a negative value in a zoom-out effect. Default value is 0 (no -zoom). - -@item optzoom -Set optimal zooming to avoid borders. - -Accepted values are: -@table @samp -@item 0 -disabled -@item 1 -optimal static zoom value is determined (only very strong movements -will lead to visible borders) (default) -@item 2 -optimal adaptive zoom value is determined (no borders will be -visible), see @option{zoomspeed} -@end table - -Note that the value given at zoom is added to the one calculated here. - -@item zoomspeed -Set percent to zoom maximally each frame (enabled when -@option{optzoom} is set to 2). Range is from 0 to 5, default value is -0.25. - -@item interpol -Specify type of interpolation. - -Available values are: -@table @samp -@item no -no interpolation -@item linear -linear only horizontal -@item bilinear -linear in both directions (default) -@item bicubic -cubic in both directions (slow) -@end table - -@item tripod -Enable virtual tripod mode if set to 1, which is equivalent to -@code{relative=0:smoothing=0}. Default value is 0. - -Use also @code{tripod} option of @ref{vidstabdetect}. - -@item debug -Increase log verbosity if set to 1. Also the detected global motions -are written to the temporary file @file{global_motions.trf}. Default -value is 0. -@end table - -@subsection Examples - -@itemize -@item -Use @command{ffmpeg} for a typical stabilization with default values: -@example -ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg -@end example - -Note the use of the unsharp filter which is always recommended. - -@item -Zoom in a bit more and load transform data from a given file: -@example -vidstabtransform=zoom=5:input="mytransforms.trf" -@end example - -@item -Smoothen the video even more: -@example -vidstabtransform=smoothing=30 -@end example -@end itemize - -@section vflip - -Flip the input video vertically. - -For example, to vertically flip a video with @command{ffmpeg}: -@example -ffmpeg -i in.avi -vf "vflip" out.avi -@end example - -@section vignette - -Make or reverse a natural vignetting effect. - -The filter accepts the following options: - -@table @option -@item angle, a -Set lens angle expression as a number of radians. - -The value is clipped in the @code{[0,PI/2]} range. - -Default value: @code{"PI/5"} - -@item x0 -@item y0 -Set center coordinates expressions. Respectively @code{"w/2"} and @code{"h/2"} -by default. - -@item mode -Set forward/backward mode. - -Available modes are: -@table @samp -@item forward -The larger the distance from the central point, the darker the image becomes. - -@item backward -The larger the distance from the central point, the brighter the image becomes. -This can be used to reverse a vignette effect, though there is no automatic -detection to extract the lens @option{angle} and other settings (yet). It can -also be used to create a burning effect. -@end table - -Default value is @samp{forward}. - -@item eval -Set evaluation mode for the expressions (@option{angle}, @option{x0}, @option{y0}). - -It accepts the following values: -@table @samp -@item init -Evaluate expressions only once during the filter initialization. - -@item frame -Evaluate expressions for each incoming frame. This is way slower than the -@samp{init} mode since it requires all the scalers to be re-computed, but it -allows advanced dynamic expressions. -@end table - -Default value is @samp{init}. - -@item dither -Set dithering to reduce the circular banding effects. Default is @code{1} -(enabled). - -@item aspect -Set vignette aspect. This setting allows one to adjust the shape of the vignette. -Setting this value to the SAR of the input will make a rectangular vignetting -following the dimensions of the video. - -Default is @code{1/1}. -@end table - -@subsection Expressions - -The @option{alpha}, @option{x0} and @option{y0} expressions can contain the -following parameters. - -@table @option -@item w -@item h -input width and height - -@item n -the number of input frame, starting from 0 - -@item pts -the PTS (Presentation TimeStamp) time of the filtered video frame, expressed in -@var{TB} units, NAN if undefined - -@item r -frame rate of the input video, NAN if the input frame rate is unknown - -@item t -the PTS (Presentation TimeStamp) of the filtered video frame, -expressed in seconds, NAN if undefined - -@item tb -time base of the input video -@end table - - -@subsection Examples - -@itemize -@item -Apply simple strong vignetting effect: -@example -vignette=PI/4 -@end example - -@item -Make a flickering vignetting: -@example -vignette='PI/4+random(1)*PI/50':eval=frame -@end example - -@end itemize - -@section w3fdif - -Deinterlace the input video ("w3fdif" stands for "Weston 3 Field -Deinterlacing Filter"). - -Based on the process described by Martin Weston for BBC R&D, and -implemented based on the de-interlace algorithm written by Jim -Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter -uses filter coefficients calculated by BBC R&D. - -There are two sets of filter coefficients, so called "simple": -and "complex". Which set of filter coefficients is used can -be set by passing an optional parameter: - -@table @option -@item filter -Set the interlacing filter coefficients. Accepts one of the following values: - -@table @samp -@item simple -Simple filter coefficient set. -@item complex -More-complex filter coefficient set. -@end table -Default value is @samp{complex}. - -@item deint -Specify which frames to deinterlace. Accept one of the following values: - -@table @samp -@item all -Deinterlace all frames, -@item interlaced -Only deinterlace frames marked as interlaced. -@end table - -Default value is @samp{all}. -@end table - -@anchor{yadif} -@section yadif - -Deinterlace the input video ("yadif" means "yet another deinterlacing -filter"). - -It accepts the following parameters: - - -@table @option - -@item mode -The interlacing mode to adopt. It accepts one of the following values: - -@table @option -@item 0, send_frame -Output one frame for each frame. -@item 1, send_field -Output one frame for each field. -@item 2, send_frame_nospatial -Like @code{send_frame}, but it skips the spatial interlacing check. -@item 3, send_field_nospatial -Like @code{send_field}, but it skips the spatial interlacing check. -@end table - -The default value is @code{send_frame}. - -@item parity -The picture field parity assumed for the input interlaced video. It accepts one -of the following values: - -@table @option -@item 0, tff -Assume the top field is first. -@item 1, bff -Assume the bottom field is first. -@item -1, auto -Enable automatic detection of field parity. -@end table - -The default value is @code{auto}. -If the interlacing is unknown or the decoder does not export this information, -top field first will be assumed. - -@item deint -Specify which frames to deinterlace. Accept one of the following -values: - -@table @option -@item 0, all -Deinterlace all frames. -@item 1, interlaced -Only deinterlace frames marked as interlaced. -@end table - -The default value is @code{all}. -@end table - -@c man end VIDEO FILTERS - -@chapter Video Sources -@c man begin VIDEO SOURCES - -Below is a description of the currently available video sources. - -@section buffer - -Buffer video frames, and make them available to the filter chain. - -This source is mainly intended for a programmatic use, in particular -through the interface defined in @file{libavfilter/vsrc_buffer.h}. - -It accepts the following parameters: - -@table @option - -@item video_size -Specify the size (width and height) of the buffered video frames. For the -syntax of this option, check the "Video size" section in the ffmpeg-utils -manual. - -@item width -The input video width. - -@item height -The input video height. - -@item pix_fmt -A string representing the pixel format of the buffered video frames. -It may be a number corresponding to a pixel format, or a pixel format -name. - -@item time_base -Specify the timebase assumed by the timestamps of the buffered frames. - -@item frame_rate -Specify the frame rate expected for the video stream. - -@item pixel_aspect, sar -The sample (pixel) aspect ratio of the input video. - -@item sws_param -Specify the optional parameters to be used for the scale filter which -is automatically inserted when an input change is detected in the -input size or format. -@end table - -For example: -@example -buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1 -@end example - -will instruct the source to accept video frames with size 320x240 and -with format "yuv410p", assuming 1/24 as the timestamps timebase and -square pixels (1:1 sample aspect ratio). -Since the pixel format with name "yuv410p" corresponds to the number 6 -(check the enum AVPixelFormat definition in @file{libavutil/pixfmt.h}), -this example corresponds to: -@example -buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1 -@end example - -Alternatively, the options can be specified as a flat string, but this -syntax is deprecated: - -@var{width}:@var{height}:@var{pix_fmt}:@var{time_base.num}:@var{time_base.den}:@var{pixel_aspect.num}:@var{pixel_aspect.den}[:@var{sws_param}] - -@section cellauto - -Create a pattern generated by an elementary cellular automaton. - -The initial state of the cellular automaton can be defined through the -@option{filename}, and @option{pattern} options. If such options are -not specified an initial state is created randomly. - -At each new frame a new row in the video is filled with the result of -the cellular automaton next generation. The behavior when the whole -frame is filled is defined by the @option{scroll} option. - -This source accepts the following options: - -@table @option -@item filename, f -Read the initial cellular automaton state, i.e. the starting row, from -the specified file. -In the file, each non-whitespace character is considered an alive -cell, a newline will terminate the row, and further characters in the -file will be ignored. - -@item pattern, p -Read the initial cellular automaton state, i.e. the starting row, from -the specified string. - -Each non-whitespace character in the string is considered an alive -cell, a newline will terminate the row, and further characters in the -string will be ignored. - -@item rate, r -Set the video rate, that is the number of frames generated per second. -Default is 25. - -@item random_fill_ratio, ratio -Set the random fill ratio for the initial cellular automaton row. It -is a floating point number value ranging from 0 to 1, defaults to -1/PHI. - -This option is ignored when a file or a pattern is specified. - -@item random_seed, seed -Set the seed for filling randomly the initial row, must be an integer -included between 0 and UINT32_MAX. If not specified, or if explicitly -set to -1, the filter will try to use a good random seed on a best -effort basis. - -@item rule -Set the cellular automaton rule, it is a number ranging from 0 to 255. -Default value is 110. - -@item size, s -Set the size of the output video. For the syntax of this option, check -the "Video size" section in the ffmpeg-utils manual. - -If @option{filename} or @option{pattern} is specified, the size is set -by default to the width of the specified initial state row, and the -height is set to @var{width} * PHI. - -If @option{size} is set, it must contain the width of the specified -pattern string, and the specified pattern will be centered in the -larger row. - -If a filename or a pattern string is not specified, the size value -defaults to "320x518" (used for a randomly generated initial state). - -@item scroll -If set to 1, scroll the output upward when all the rows in the output -have been already filled. If set to 0, the new generated row will be -written over the top row just after the bottom row is filled. -Defaults to 1. - -@item start_full, full -If set to 1, completely fill the output with generated rows before -outputting the first frame. -This is the default behavior, for disabling set the value to 0. - -@item stitch -If set to 1, stitch the left and right row edges together. -This is the default behavior, for disabling set the value to 0. -@end table - -@subsection Examples - -@itemize -@item -Read the initial state from @file{pattern}, and specify an output of -size 200x400. -@example -cellauto=f=pattern:s=200x400 -@end example - -@item -Generate a random initial row with a width of 200 cells, with a fill -ratio of 2/3: -@example -cellauto=ratio=2/3:s=200x200 -@end example - -@item -Create a pattern generated by rule 18 starting by a single alive cell -centered on an initial row with width 100: -@example -cellauto=p=@@:s=100x400:full=0:rule=18 -@end example - -@item -Specify a more elaborated initial pattern: -@example -cellauto=p='@@@@ @@ @@@@':s=100x400:full=0:rule=18 -@end example - -@end itemize - -@section mandelbrot - -Generate a Mandelbrot set fractal, and progressively zoom towards the -point specified with @var{start_x} and @var{start_y}. - -This source accepts the following options: - -@table @option - -@item end_pts -Set the terminal pts value. Default value is 400. - -@item end_scale -Set the terminal scale value. -Must be a floating point value. Default value is 0.3. - -@item inner -Set the inner coloring mode, that is the algorithm used to draw the -Mandelbrot fractal internal region. - -It shall assume one of the following values: -@table @option -@item black -Set black mode. -@item convergence -Show time until convergence. -@item mincol -Set color based on point closest to the origin of the iterations. -@item period -Set period mode. -@end table - -Default value is @var{mincol}. - -@item bailout -Set the bailout value. Default value is 10.0. - -@item maxiter -Set the maximum of iterations performed by the rendering -algorithm. Default value is 7189. - -@item outer -Set outer coloring mode. -It shall assume one of following values: -@table @option -@item iteration_count -Set iteration cound mode. -@item normalized_iteration_count -set normalized iteration count mode. -@end table -Default value is @var{normalized_iteration_count}. - -@item rate, r -Set frame rate, expressed as number of frames per second. Default -value is "25". - -@item size, s -Set frame size. For the syntax of this option, check the "Video -size" section in the ffmpeg-utils manual. Default value is "640x480". - -@item start_scale -Set the initial scale value. Default value is 3.0. - -@item start_x -Set the initial x position. Must be a floating point value between --100 and 100. Default value is -0.743643887037158704752191506114774. - -@item start_y -Set the initial y position. Must be a floating point value between --100 and 100. Default value is -0.131825904205311970493132056385139. -@end table - -@section mptestsrc - -Generate various test patterns, as generated by the MPlayer test filter. - -The size of the generated video is fixed, and is 256x256. -This source is useful in particular for testing encoding features. - -This source accepts the following options: - -@table @option - -@item rate, r -Specify the frame rate of the sourced video, as the number of frames -generated per second. It has to be a string in the format -@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a floating point -number or a valid video frame rate abbreviation. The default value is -"25". - -@item duration, d -Set the video duration of the sourced video. The accepted syntax is: -@example -[-]HH:MM:SS[.m...] -[-]S+[.m...] -@end example -See also the function @code{av_parse_time()}. - -If not specified, or the expressed duration is negative, the video is -supposed to be generated forever. - -@item test, t - -Set the number or the name of the test to perform. Supported tests are: -@table @option -@item dc_luma -@item dc_chroma -@item freq_luma -@item freq_chroma -@item amp_luma -@item amp_chroma -@item cbp -@item mv -@item ring1 -@item ring2 -@item all - -@end table - -Default value is "all", which will cycle through the list of all tests. -@end table +The default value of @var{w} and @var{h} is 0. Some examples: @example -testsrc=t=dc_luma +# Scale the input video to a size of 200x100 +scale=w=200:h=100 + +# Scale the input to 2x +scale=w=2*iw:h=2*ih +# The above is the same as +scale=2*in_w:2*in_h + +# Scale the input to half the original size +scale=w=iw/2:h=ih/2 + +# Increase the width, and set the height to the same size +scale=3/2*iw:ow + +# Seek Greek harmony +scale=iw:1/PHI*iw +scale=ih*PHI:ih + +# Increase the height, and set the width to 3/2 of the height +scale=w=3/2*oh:h=3/5*ih + +# Increase the size, making the size a multiple of the chroma +scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" + +# Increase the width to a maximum of 500 pixels, +# keeping the same aspect ratio as the input +scale=w='min(500\, iw*3/2):h=-1' @end example -will generate a "dc_luma" test pattern. - -@section frei0r_src - -Provide a frei0r source. - -To enable compilation of this filter you need to install the frei0r -header and configure FFmpeg with @code{--enable-frei0r}. - -This source accepts the following parameters: - -@table @option - -@item size -The size of the video to generate. For the syntax of this option, check the -"Video size" section in the ffmpeg-utils manual. - -@item framerate -The framerate of the generated video. It may be a string of the form -@var{num}/@var{den} or a frame rate abbreviation. - -@item filter_name -The name to the frei0r source to load. For more information regarding frei0r and -how to set the parameters, read the @ref{frei0r} section in the video filters -documentation. - -@item filter_params -A '|'-separated list of parameters to pass to the frei0r source. - -@end table - -For example, to generate a frei0r partik0l source with size 200x200 -and frame rate 10 which is overlayed on the overlay filter main input: -@example -frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay -@end example - -@section life - -Generate a life pattern. - -This source is based on a generalization of John Conway's life game. - -The sourced input represents a life grid, each pixel represents a cell -which can be in one of two possible states, alive or dead. Every cell -interacts with its eight neighbours, which are the cells that are -horizontally, vertically, or diagonally adjacent. - -At each interaction the grid evolves according to the adopted rule, -which specifies the number of neighbor alive cells which will make a -cell stay alive or born. The @option{rule} option allows one to specify -the rule to adopt. - -This source accepts the following options: - -@table @option -@item filename, f -Set the file from which to read the initial grid state. In the file, -each non-whitespace character is considered an alive cell, and newline -is used to delimit the end of each row. - -If this option is not specified, the initial grid is generated -randomly. - -@item rate, r -Set the video rate, that is the number of frames generated per second. -Default is 25. - -@item random_fill_ratio, ratio -Set the random fill ratio for the initial random grid. It is a -floating point number value ranging from 0 to 1, defaults to 1/PHI. -It is ignored when a file is specified. - -@item random_seed, seed -Set the seed for filling the initial random grid, must be an integer -included between 0 and UINT32_MAX. If not specified, or if explicitly -set to -1, the filter will try to use a good random seed on a best -effort basis. - -@item rule -Set the life rule. - -A rule can be specified with a code of the kind "S@var{NS}/B@var{NB}", -where @var{NS} and @var{NB} are sequences of numbers in the range 0-8, -@var{NS} specifies the number of alive neighbor cells which make a -live cell stay alive, and @var{NB} the number of alive neighbor cells -which make a dead cell to become alive (i.e. to "born"). -"s" and "b" can be used in place of "S" and "B", respectively. - -Alternatively a rule can be specified by an 18-bits integer. The 9 -high order bits are used to encode the next cell state if it is alive -for each number of neighbor alive cells, the low order bits specify -the rule for "borning" new cells. Higher order bits encode for an -higher number of neighbor cells. -For example the number 6153 = @code{(12<<9)+9} specifies a stay alive -rule of 12 and a born rule of 9, which corresponds to "S23/B03". - -Default value is "S23/B3", which is the original Conway's game of life -rule, and will keep a cell alive if it has 2 or 3 neighbor alive -cells, and will born a new cell if there are three alive cells around -a dead cell. - -@item size, s -Set the size of the output video. For the syntax of this option, check the -"Video size" section in the ffmpeg-utils manual. - -If @option{filename} is specified, the size is set by default to the -same size of the input file. If @option{size} is set, it must contain -the size specified in the input file, and the initial grid defined in -that file is centered in the larger resulting area. - -If a filename is not specified, the size value defaults to "320x240" -(used for a randomly generated initial grid). - -@item stitch -If set to 1, stitch the left and right grid edges together, and the -top and bottom edges also. Defaults to 1. - -@item mold -Set cell mold speed. If set, a dead cell will go from @option{death_color} to -@option{mold_color} with a step of @option{mold}. @option{mold} can have a -value from 0 to 255. - -@item life_color -Set the color of living (or new born) cells. - -@item death_color -Set the color of dead cells. If @option{mold} is set, this is the first color -used to represent a dead cell. - -@item mold_color -Set mold color, for definitely dead and moldy cells. - -For the syntax of these 3 color options, check the "Color" section in the -ffmpeg-utils manual. -@end table - -@subsection Examples - -@itemize -@item -Read a grid from @file{pattern}, and center it on a grid of size -300x300 pixels: -@example -life=f=pattern:s=300x300 -@end example - -@item -Generate a random grid of size 200x200, with a fill ratio of 2/3: -@example -life=ratio=2/3:s=200x200 -@end example - -@item -Specify a custom rule for evolving a randomly generated grid: -@example -life=rule=S14/B34 -@end example - -@item -Full example with slow death effect (mold) using @command{ffplay}: -@example -ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16 -@end example -@end itemize - -@anchor{color} -@anchor{haldclutsrc} -@anchor{nullsrc} -@anchor{rgbtestsrc} -@anchor{smptebars} -@anchor{smptehdbars} -@anchor{testsrc} -@section color, haldclutsrc, nullsrc, rgbtestsrc, smptebars, smptehdbars, testsrc - -The @code{color} source provides an uniformly colored input. - -The @code{haldclutsrc} source provides an identity Hald CLUT. See also -@ref{haldclut} filter. - -The @code{nullsrc} source returns unprocessed video frames. It is -mainly useful to be employed in analysis / debugging tools, or as the -source for filters which ignore the input data. - -The @code{rgbtestsrc} source generates an RGB test pattern useful for -detecting RGB vs BGR issues. You should see a red, green and blue -stripe from top to bottom. - -The @code{smptebars} source generates a color bars pattern, based on -the SMPTE Engineering Guideline EG 1-1990. - -The @code{smptehdbars} source generates a color bars pattern, based on -the SMPTE RP 219-2002. - -The @code{testsrc} source generates a test video pattern, showing a -color pattern, a scrolling gradient and a timestamp. This is mainly -intended for testing purposes. - -The sources accept the following parameters: - -@table @option - -@item color, c -Specify the color of the source, only available in the @code{color} -source. For the syntax of this option, check the "Color" section in the -ffmpeg-utils manual. - -@item level -Specify the level of the Hald CLUT, only available in the @code{haldclutsrc} -source. A level of @code{N} generates a picture of @code{N*N*N} by @code{N*N*N} -pixels to be used as identity matrix for 3D lookup tables. Each component is -coded on a @code{1/(N*N)} scale. - -@item size, s -Specify the size of the sourced video. For the syntax of this option, check the -"Video size" section in the ffmpeg-utils manual. The default value is -"320x240". - -This option is not available with the @code{haldclutsrc} filter. - -@item rate, r -Specify the frame rate of the sourced video, as the number of frames -generated per second. It has to be a string in the format -@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a floating point -number or a valid video frame rate abbreviation. The default value is -"25". - -@item sar -Set the sample aspect ratio of the sourced video. - -@item duration, d -Set the video duration of the sourced video. The accepted syntax is: -@example -[-]HH[:MM[:SS[.m...]]] -[-]S+[.m...] -@end example -Also see the the @code{av_parse_time()} function. - -If not specified, or the expressed duration is negative, the video is -supposed to be generated forever. - -@item decimals, n -Set the number of decimals to show in the timestamp, only available in the -@code{testsrc} source. - -The displayed timestamp value will correspond to the original -timestamp value multiplied by the power of 10 of the specified -value. Default value is 0. -@end table - -For example the following: -@example -testsrc=duration=5.3:size=qcif:rate=10 -@end example - -will generate a video with a duration of 5.3 seconds, with size -176x144 and a frame rate of 10 frames per second. - -The following graph description will generate a red source -with an opacity of 0.2, with size "qcif" and a frame rate of 10 -frames per second. -@example -color=c=red@@0.2:s=qcif:r=10 -@end example - -If the input content is to be ignored, @code{nullsrc} can be used. The -following command generates noise in the luminance plane by employing -the @code{geq} filter: -@example -nullsrc=s=256x256, geq=random(1)*255:128:128 -@end example - -@subsection Commands - -The @code{color} source supports the following commands: - -@table @option -@item c, color -Set the color of the created image. Accepts the same syntax of the -corresponding @option{color} option. -@end table - -@c man end VIDEO SOURCES - -@chapter Video Sinks -@c man begin VIDEO SINKS - -Below is a description of the currently available video sinks. - -@section buffersink - -Buffer video frames, and make them available to the end of the filter -graph. - -This sink is mainly intended for programmatic use, in particular -through the interface defined in @file{libavfilter/buffersink.h} -or the options system. - -It accepts a pointer to an AVBufferSinkContext structure, which -defines the incoming buffers' formats, to be passed as the opaque -parameter to @code{avfilter_init_filter} for initialization. - -@section nullsink - -Null video sink: do absolutely nothing with the input video. It is -mainly useful as a template and for use in analysis / debugging -tools. - -@c man end VIDEO SINKS - -@chapter Multimedia Filters -@c man begin MULTIMEDIA FILTERS - -Below is a description of the currently available multimedia filters. - -@section avectorscope - -Convert input audio to a video output, representing the audio vector -scope. - -The filter is used to measure the difference between channels of stereo -audio stream. A monoaural signal, consisting of identical left and right -signal, results in straight vertical line. Any stereo separation is visible -as a deviation from this line, creating a Lissajous figure. -If the straight (or deviation from it) but horizontal line appears this -indicates that the left and right channels are out of phase. - -The filter accepts the following options: - -@table @option -@item mode, m -Set the vectorscope mode. - -Available values are: -@table @samp -@item lissajous -Lissajous rotated by 45 degrees. - -@item lissajous_xy -Same as above but not rotated. -@end table - -Default value is @samp{lissajous}. - -@item size, s -Set the video size for the output. For the syntax of this option, check the "Video size" -section in the ffmpeg-utils manual. Default value is @code{400x400}. - -@item rate, r -Set the output frame rate. Default value is @code{25}. - -@item rc -@item gc -@item bc -Specify the red, green and blue contrast. Default values are @code{40}, @code{160} and @code{80}. -Allowed range is @code{[0, 255]}. - -@item rf -@item gf -@item bf -Specify the red, green and blue fade. Default values are @code{15}, @code{10} and @code{5}. -Allowed range is @code{[0, 255]}. - -@item zoom -Set the zoom factor. Default value is @code{1}. Allowed range is @code{[1, 10]}. -@end table - -@subsection Examples - -@itemize -@item -Complete example using @command{ffplay}: -@example -ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1]; - [a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]' -@end example -@end itemize - -@section concat - -Concatenate audio and video streams, joining them together one after the -other. - -The filter works on segments of synchronized video and audio streams. All -segments must have the same number of streams of each type, and that will -also be the number of streams at output. - -The filter accepts the following options: - -@table @option - -@item n -Set the number of segments. Default is 2. - -@item v -Set the number of output video streams, that is also the number of video -streams in each segment. Default is 1. - -@item a -Set the number of output audio streams, that is also the number of audio -streams in each segment. Default is 0. - -@item unsafe -Activate unsafe mode: do not fail if segments have a different format. - -@end table - -The filter has @var{v}+@var{a} outputs: first @var{v} video outputs, then -@var{a} audio outputs. - -There are @var{n}x(@var{v}+@var{a}) inputs: first the inputs for the first -segment, in the same order as the outputs, then the inputs for the second -segment, etc. - -Related streams do not always have exactly the same duration, for various -reasons including codec frame size or sloppy authoring. For that reason, -related synchronized streams (e.g. a video and its audio track) should be -concatenated at once. The concat filter will use the duration of the longest -stream in each segment (except the last one), and if necessary pad shorter -audio streams with silence. - -For this filter to work correctly, all segments must start at timestamp 0. - -All corresponding streams must have the same parameters in all segments; the -filtering system will automatically select a common pixel format for video -streams, and a common sample format, sample rate and channel layout for -audio streams, but other settings, such as resolution, must be converted -explicitly by the user. - -Different frame rates are acceptable but will result in variable frame rate -at output; be sure to configure the output file to handle it. - -@subsection Examples - -@itemize -@item -Concatenate an opening, an episode and an ending, all in bilingual version -(video in stream 0, audio in streams 1 and 2): -@example -ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \ - '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2] - concat=n=3:v=1:a=2 [v] [a1] [a2]' \ - -map '[v]' -map '[a1]' -map '[a2]' output.mkv -@end example - -@item -Concatenate two parts, handling audio and video separately, using the -(a)movie sources, and adjusting the resolution: -@example -movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ; -movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ; -[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa] -@end example -Note that a desync will happen at the stitch if the audio and video streams -do not have exactly the same duration in the first file. - -@end itemize - -@section ebur128 - -EBU R128 scanner filter. This filter takes an audio stream as input and outputs -it unchanged. By default, it logs a message at a frequency of 10Hz with the -Momentary loudness (identified by @code{M}), Short-term loudness (@code{S}), -Integrated loudness (@code{I}) and Loudness Range (@code{LRA}). - -The filter also has a video output (see the @var{video} option) with a real -time graph to observe the loudness evolution. The graphic contains the logged -message mentioned above, so it is not printed anymore when this option is set, -unless the verbose logging is set. The main graphing area contains the -short-term loudness (3 seconds of analysis), and the gauge on the right is for -the momentary loudness (400 milliseconds). - -More information about the Loudness Recommendation EBU R128 on -@url{http://tech.ebu.ch/loudness}. - -The filter accepts the following options: - -@table @option - -@item video -Activate the video output. The audio stream is passed unchanged whether this -option is set or no. The video stream will be the first output stream if -activated. Default is @code{0}. - -@item size -Set the video size. This option is for video only. For the syntax of this -option, check the "Video size" section in the ffmpeg-utils manual. Default -and minimum resolution is @code{640x480}. - -@item meter -Set the EBU scale meter. Default is @code{9}. Common values are @code{9} and -@code{18}, respectively for EBU scale meter +9 and EBU scale meter +18. Any -other integer value between this range is allowed. - -@item metadata -Set metadata injection. If set to @code{1}, the audio input will be segmented -into 100ms output frames, each of them containing various loudness information -in metadata. All the metadata keys are prefixed with @code{lavfi.r128.}. - -Default is @code{0}. - -@item framelog -Force the frame logging level. - -Available values are: -@table @samp -@item info -information logging level -@item verbose -verbose logging level -@end table - -By default, the logging level is set to @var{info}. If the @option{video} or -the @option{metadata} options are set, it switches to @var{verbose}. - -@item peak -Set peak mode(s). - -Available modes can be cumulated (the option is a @code{flag} type). Possible -values are: -@table @samp -@item none -Disable any peak mode (default). -@item sample -Enable sample-peak mode. - -Simple peak mode looking for the higher sample value. It logs a message -for sample-peak (identified by @code{SPK}). -@item true -Enable true-peak mode. - -If enabled, the peak lookup is done on an over-sampled version of the input -stream for better peak accuracy. It logs a message for true-peak. -(identified by @code{TPK}) and true-peak per frame (identified by @code{FTPK}). -This mode requires a build with @code{libswresample}. -@end table - -@end table - -@subsection Examples - -@itemize -@item -Real-time graph using @command{ffplay}, with a EBU scale meter +18: -@example -ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]" -@end example - -@item -Run an analysis with @command{ffmpeg}: -@example -ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null - -@end example -@end itemize - -@section interleave, ainterleave - -Temporally interleave frames from several inputs. - -@code{interleave} works with video inputs, @code{ainterleave} with audio. - -These filters read frames from several inputs and send the oldest -queued frame to the output. - -Input streams must have a well defined, monotonically increasing frame -timestamp values. - -In order to submit one frame to output, these filters need to enqueue -at least one frame for each input, so they cannot work in case one -input is not yet terminated and will not receive incoming frames. - -For example consider the case when one input is a @code{select} filter -which always drop input frames. The @code{interleave} filter will keep -reading from that input, but it will never be able to send new frames -to output until the input will send an end-of-stream signal. - -Also, depending on inputs synchronization, the filters will drop -frames in case one input receives more frames than the other ones, and -the queue is already filled. - -These filters accept the following options: - -@table @option -@item nb_inputs, n -Set the number of different inputs, it is 2 by default. -@end table - -@subsection Examples - -@itemize -@item -Interleave frames belonging to different streams using @command{ffmpeg}: -@example -ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi -@end example - -@item -Add flickering blur effect: -@example -select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave -@end example -@end itemize - -@section perms, aperms - -Set read/write permissions for the output frames. - -These filters are mainly aimed at developers to test direct path in the -following filter in the filtergraph. - -The filters accept the following options: - -@table @option -@item mode -Select the permissions mode. - -It accepts the following values: -@table @samp -@item none -Do nothing. This is the default. -@item ro -Set all the output frames read-only. -@item rw -Set all the output frames directly writable. -@item toggle -Make the frame read-only if writable, and writable if read-only. -@item random -Set each output frame read-only or writable randomly. -@end table - -@item seed -Set the seed for the @var{random} mode, must be an integer included between -@code{0} and @code{UINT32_MAX}. If not specified, or if explicitly set to -@code{-1}, the filter will try to use a good random seed on a best effort -basis. -@end table - -Note: in case of auto-inserted filter between the permission filter and the -following one, the permission might not be received as expected in that -following filter. Inserting a @ref{format} or @ref{aformat} filter before the -perms/aperms filter can avoid this problem. - -@section select, aselect - +@section select Select frames to pass in output. -This filter accepts the following options: +It accepts the following parameters: @table @option -@item expr, e -Set expression, which is evaluated for each input frame. +@item expr +An expression, which is evaluated for each input frame. If the expression is +evaluated to a non-zero value, the frame is selected and passed to the output, +otherwise it is discarded. -If the expression is evaluated to zero, the frame is discarded. - -If the evaluation result is negative or NaN, the frame is sent to the -first output; otherwise it is sent to the output with index -@code{ceil(val)-1}, assuming that the input index starts from 0. - -For example a value of @code{1.2} corresponds to the output with index -@code{ceil(1.2)-1 = 2-1 = 1}, that is the second output. - -@item outputs, n -Set the number of outputs. The output to which to send the selected -frame is based on the result of the evaluation. Default value is 1. @end table The expression can contain the following constants: @table @option +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). + @item n The (sequential) number of the filtered frame, starting from 0. @@ -9690,7 +2209,7 @@ The PTS of the first video frame in the video. It's NAN if undefined. @item start_t The time of the first video frame in the video. It's NAN if undefined. -@item pict_type @emph{(video only)} +@item pict_type The type of the filtered frame. It can assume one of the following values: @table @option @@ -9703,7 +2222,7 @@ values: @item BI @end table -@item interlace_type @emph{(video only)} +@item interlace_type The frame interlace type. It can assume one of the following values: @table @option @item PROGRESSIVE @@ -9714,242 +2233,104 @@ The frame is top-field-first. The frame is bottom-field-first. @end table -@item consumed_sample_n @emph{(audio only)} -the number of selected samples before the current frame - -@item samples_n @emph{(audio only)} -the number of samples in the current frame - -@item sample_rate @emph{(audio only)} -the input sample rate - @item key This is 1 if the filtered frame is a key-frame, 0 otherwise. -@item pos -the position in the file of the filtered frame, -1 if the information -is not available (e.g. for synthetic video) - -@item scene @emph{(video only)} -value between 0 and 1 to indicate a new scene; a low value reflects a low -probability for the current frame to introduce a new scene, while a higher -value means the current frame is more likely to be one (see the example below) - @end table The default value of the select expression is "1". -@subsection Examples +Some examples: -@itemize -@item -Select all frames in input: @example +# Select all the frames in input select -@end example -The example above is the same as: -@example -select=1 -@end example +# The above is the same as +select=expr=1 -@item -Skip all frames: -@example -select=0 -@end example +# Skip all frames +select=expr=0 -@item -Select only I-frames: -@example -select='eq(pict_type\,I)' -@end example +# Select only I-frames +select='expr=eq(pict_type\,I)' -@item -Select one frame every 100: -@example +# Select one frame per 100 select='not(mod(n\,100))' -@end example -@item -Select only frames contained in the 10-20 time interval: -@example -select=between(t\,10\,20) -@end example +# Select only frames contained in the 10-20 time interval +select='gte(t\,10)*lte(t\,20)' -@item -Select only I frames contained in the 10-20 time interval: -@example -select=between(t\,10\,20)*eq(pict_type\,I) -@end example +# Select only I frames contained in the 10-20 time interval +select='gte(t\,10)*lte(t\,20)*eq(pict_type\,I)' -@item -Select frames with a minimum distance of 10 seconds: -@example +# Select frames with a minimum distance of 10 seconds select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)' @end example -@item -Use aselect to select only audio frames with samples number > 100: -@example -aselect='gt(samples_n\,100)' -@end example +@anchor{setdar} +@section setdar -@item -Create a mosaic of the first scenes: -@example -ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png -@end example +Set the Display Aspect Ratio for the filter output video. -Comparing @var{scene} against a value between 0.3 and 0.5 is generally a sane -choice. +This is done by changing the specified Sample (aka Pixel) Aspect +Ratio, according to the following equation: +@math{DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR} -@item -Send even and odd frames to separate outputs, and compose them: -@example -select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h -@end example -@end itemize +Keep in mind that this filter does not modify the pixel dimensions of +the video frame. Also, the display aspect ratio set by this filter may +be changed by later filters in the filterchain, e.g. in case of +scaling or if another "setdar" or a "setsar" filter is applied. -@section sendcmd, asendcmd +It accepts the following parameters: -Send commands to filters in the filtergraph. - -These filters read commands to be sent to other filters in the -filtergraph. - -@code{sendcmd} must be inserted between two video filters, -@code{asendcmd} must be inserted between two audio filters, but apart -from that they act the same way. - -The specification of commands can be provided in the filter arguments -with the @var{commands} option, or in a file specified by the -@var{filename} option. - -These filters accept the following options: @table @option -@item commands, c -Set the commands to be read and sent to the other filters. -@item filename, f -Set the filename of the commands to be read and sent to the other -filters. + +@item dar +The output display aspect ratio. + @end table -@subsection Commands syntax +The parameter @var{dar} is an expression containing +the following constants: -A commands description consists of a sequence of interval -specifications, comprising a list of commands to be executed when a -particular event related to that interval occurs. The occurring event -is typically the current frame time entering or leaving a given time -interval. - -An interval is specified by the following syntax: -@example -@var{START}[-@var{END}] @var{COMMANDS}; -@end example - -The time interval is specified by the @var{START} and @var{END} times. -@var{END} is optional and defaults to the maximum time. - -The current frame time is considered within the specified interval if -it is included in the interval [@var{START}, @var{END}), that is when -the time is greater or equal to @var{START} and is lesser than -@var{END}. - -@var{COMMANDS} consists of a sequence of one or more command -specifications, separated by ",", relating to that interval. The -syntax of a command specification is given by: -@example -[@var{FLAGS}] @var{TARGET} @var{COMMAND} @var{ARG} -@end example - -@var{FLAGS} is optional and specifies the type of events relating to -the time interval which enable sending the specified command, and must -be a non-null sequence of identifier flags separated by "+" or "|" and -enclosed between "[" and "]". - -The following flags are recognized: @table @option -@item enter -The command is sent when the current frame timestamp enters the -specified interval. In other words, the command is sent when the -previous frame timestamp was not in the given interval, and the -current is. +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). -@item leave -The command is sent when the current frame timestamp leaves the -specified interval. In other words, the command is sent when the -previous frame timestamp was in the given interval, and the -current is not. +@item w, h +The input width and height. + +@item a +This is the same as @var{w} / @var{h}. + +@item sar +The input sample aspect ratio. + +@item dar +The input display aspect ratio. It is the same as +(@var{w} / @var{h}) * @var{sar}. + +@item hsub, vsub +The horizontal and vertical chroma subsample values. For example, for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. @end table -If @var{FLAGS} is not specified, a default value of @code{[enter]} is -assumed. - -@var{TARGET} specifies the target of the command, usually the name of -the filter class or a specific filter instance name. - -@var{COMMAND} specifies the name of the command for the target filter. - -@var{ARG} is optional and specifies the optional list of argument for -the given @var{COMMAND}. - -Between one interval specification and another, whitespaces, or -sequences of characters starting with @code{#} until the end of line, -are ignored and can be used to annotate comments. - -A simplified BNF description of the commands specification syntax -follows: +To change the display aspect ratio to 16:9, specify: @example -@var{COMMAND_FLAG} ::= "enter" | "leave" -@var{COMMAND_FLAGS} ::= @var{COMMAND_FLAG} [(+|"|")@var{COMMAND_FLAG}] -@var{COMMAND} ::= ["[" @var{COMMAND_FLAGS} "]"] @var{TARGET} @var{COMMAND} [@var{ARG}] -@var{COMMANDS} ::= @var{COMMAND} [,@var{COMMANDS}] -@var{INTERVAL} ::= @var{START}[-@var{END}] @var{COMMANDS} -@var{INTERVALS} ::= @var{INTERVAL}[;@var{INTERVALS}] +setdar=dar=16/9 +# The above is equivalent to +setdar=dar=1.77777 @end example -@subsection Examples +Also see the the @ref{setsar} filter documentation. -@itemize -@item -Specify audio tempo change at second 4: -@example -asendcmd=c='4.0 atempo tempo 1.5',atempo -@end example +@section setpts -@item -Specify a list of drawtext and hue commands in a file. -@example -# show text in the interval 5-10 -5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world', - [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text='; +Change the PTS (presentation timestamp) of the input video frames. -# desaturate the image in the interval 15-20 -15.0-20.0 [enter] hue s 0, - [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor', - [leave] hue s 1, - [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color'; - -# apply an exponential saturation fade-out effect, starting from time 25 -25 [enter] hue s exp(25-t) -@end example - -A filtergraph allowing to read and process the above command list -stored in a file @file{test.cmd}, can be specified with: -@example -sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue -@end example -@end itemize - -@anchor{setpts} -@section setpts, asetpts - -Change the PTS (presentation timestamp) of the input frames. - -@code{setpts} works on video frames, @code{asetpts} on audio frames. - -This filter accepts the following options: +It accepts the following parameters: @table @option @@ -9962,57 +2343,30 @@ The expression is evaluated through the eval API and can contain the following constants: @table @option -@item FRAME_RATE -frame rate, only defined for constant frame-rate video - @item PTS -The presentation timestamp in input +The presentation timestamp in input. + +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). @item N -The count of the input frame for video or the number of consumed samples, -not including the current frame for audio, starting from 0. - -@item NB_CONSUMED_SAMPLES -The number of consumed samples, not including the current frame (only -audio) - -@item NB_SAMPLES, S -The number of samples in the current frame (only audio) - -@item SAMPLE_RATE, SR -The audio sample rate. +The count of the input frame, starting from 0. @item STARTPTS -The PTS of the first frame. - -@item STARTT -the time in seconds of the first frame +The PTS of the first video frame. @item INTERLACED State whether the current frame is interlaced. -@item T -the time in seconds of the current frame - -@item POS -original position in the file of the frame, or undefined if undefined -for the current frame - @item PREV_INPTS The previous input PTS. -@item PREV_INT -previous input time in seconds - @item PREV_OUTPTS The previous output PTS. -@item PREV_OUTT -previous output time in seconds - @item RTCTIME -The wallclock (RTC) time in microseconds.. This is deprecated, use time(0) -instead. +The wallclock (RTC) time in microseconds. @item RTCSTART The wallclock (RTC) time at the start of the movie in microseconds. @@ -10022,60 +2376,83 @@ The timebase of the input timestamps. @end table -@subsection Examples +Some examples: -@itemize -@item -Start counting PTS from zero @example -setpts=PTS-STARTPTS -@end example +# Start counting the PTS from zero +setpts=expr=PTS-STARTPTS -@item -Apply fast motion effect: -@example -setpts=0.5*PTS -@end example +# Fast motion +setpts=expr=0.5*PTS -@item -Apply slow motion effect: -@example +# Slow motion setpts=2.0*PTS -@end example -@item -Set fixed rate of 25 frames per second: -@example +# Fixed rate 25 fps setpts=N/(25*TB) -@end example -@item -Set fixed rate 25 fps with some jitter: -@example +# Fixed rate 25 fps with some jitter setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))' + +# Generate timestamps from a "live source" and rebase onto the current timebase +setpts='(RTCTIME - RTCSTART) / (TB * 1000000)" @end example -@item -Apply an offset of 10 seconds to the input PTS: +@anchor{setsar} +@section setsar + +Set the Sample (aka Pixel) Aspect Ratio for the filter output video. + +Note that as a consequence of the application of this filter, the +output display aspect ratio will change according to the following +equation: +@math{DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR} + +Keep in mind that the sample aspect ratio set by this filter may be +changed by later filters in the filterchain, e.g. if another "setsar" +or a "setdar" filter is applied. + +It accepts the following parameters: + +@table @option + +@item sar +The output sample aspect ratio. + +@end table + +The parameter @var{sar} is an expression containing +the following constants: + +@table @option +@item E, PI, PHI +These are approximated values for the mathematical constants e +(Euler's number), pi (Greek pi), and phi (the golden ratio). + +@item w, h +The input width and height. + +@item a +These are the same as @var{w} / @var{h}. + +@item sar +The input sample aspect ratio. + +@item dar +The input display aspect ratio. It is the same as +(@var{w} / @var{h}) * @var{sar}. + +@item hsub, vsub +Horizontal and vertical chroma subsample values. For example, for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. +@end table + +To change the sample aspect ratio to 10:11, specify: @example -setpts=PTS+10/TB +setsar=sar=10/11 @end example -@item -Generate timestamps from a "live source" and rebase onto the current timebase: -@example -setpts='(RTCTIME - RTCSTART) / (TB * 1000000)' -@end example - -@item -Generate timestamps by counting samples: -@example -asetpts=N/SR/TB -@end example - -@end itemize - -@section settb, asettb +@section settb Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration. @@ -10084,339 +2461,452 @@ It accepts the following parameters: @table @option -@item expr, tb +@item expr The expression which is evaluated into the output timebase. @end table -The value for @option{tb} is an arithmetic expression representing a -rational. The expression can contain the constants "AVTB" (the default -timebase), "intb" (the input timebase) and "sr" (the sample rate, -audio only). Default value is "intb". +The expression can contain the constants "PI", "E", "PHI", "AVTB" (the +default timebase), and "intb" (the input timebase). -@subsection Examples +The default value for the input is "intb". + +Some examples: -@itemize -@item -Set the timebase to 1/25: @example +# Set the timebase to 1/25 settb=expr=1/25 -@end example -@item -Set the timebase to 1/10: -@example +# Set the timebase to 1/10 settb=expr=0.1 -@end example -@item -Set the timebase to 1001/1000: -@example +# Set the timebase to 1001/1000 settb=1+0.001 -@end example -@item -Set the timebase to 2*intb: -@example +#Set the timebase to 2*intb settb=2*intb -@end example -@item -Set the default timebase value: -@example +#Set the default timebase value settb=AVTB @end example -@end itemize -@section showspectrum +@section showinfo -Convert input audio to a video output, representing the audio frequency -spectrum. +Show a line containing various information for each input video frame. +The input video is not modified. -The filter accepts the following options: - -@table @option -@item size, s -Specify the video size for the output. For the syntax of this option, check -the "Video size" section in the ffmpeg-utils manual. Default value is -@code{640x512}. - -@item slide -Specify if the spectrum should slide along the window. Default value is -@code{0}. - -@item mode -Specify display mode. - -It accepts the following values: -@table @samp -@item combined -all channels are displayed in the same row -@item separate -all channels are displayed in separate rows -@end table - -Default value is @samp{combined}. - -@item color -Specify display color mode. - -It accepts the following values: -@table @samp -@item channel -each channel is displayed in a separate color -@item intensity -each channel is is displayed using the same color scheme -@end table - -Default value is @samp{channel}. - -@item scale -Specify scale used for calculating intensity color values. - -It accepts the following values: -@table @samp -@item lin -linear -@item sqrt -square root, default -@item cbrt -cubic root -@item log -logarithmic -@end table - -Default value is @samp{sqrt}. - -@item saturation -Set saturation modifier for displayed colors. Negative values provide -alternative color scheme. @code{0} is no saturation at all. -Saturation must be in [-10.0, 10.0] range. -Default value is @code{1}. - -@item win_func -Set window function. - -It accepts the following values: -@table @samp -@item none -No samples pre-processing (do not expect this to be faster) -@item hann -Hann window -@item hamming -Hamming window -@item blackman -Blackman window -@end table - -Default value is @code{hann}. -@end table - -The usage is very similar to the showwaves filter; see the examples in that -section. - -@subsection Examples - -@itemize -@item -Large window with logarithmic color scaling: -@example -showspectrum=s=1280x480:scale=log -@end example - -@item -Complete example for a colored and sliding spectrum per channel using @command{ffplay}: -@example -ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1]; - [a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]' -@end example -@end itemize - -@section showwaves - -Convert input audio to a video output, representing the samples waves. - -The filter accepts the following options: - -@table @option -@item size, s -Specify the video size for the output. For the syntax of this option, check -the "Video size" section in the ffmpeg-utils manual. Default value -is "600x240". - -@item mode -Set display mode. - -Available values are: -@table @samp -@item point -Draw a point for each sample. - -@item line -Draw a vertical line for each sample. -@end table - -Default value is @code{point}. - -@item n -Set the number of samples which are printed on the same column. A -larger value will decrease the frame rate. Must be a positive -integer. This option can be set only if the value for @var{rate} -is not explicitly specified. - -@item rate, r -Set the (approximate) output frame rate. This is done by setting the -option @var{n}. Default value is "25". - -@end table - -@subsection Examples - -@itemize -@item -Output the input file audio and the corresponding video representation -at the same time: -@example -amovie=a.mp3,asplit[out0],showwaves[out1] -@end example - -@item -Create a synthetic signal and show it with showwaves, forcing a -frame rate of 30 frames per second: -@example -aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1] -@end example -@end itemize - -@section split, asplit - -Split input into several identical outputs. - -@code{asplit} works with audio input, @code{split} with video. - -The filter accepts a single parameter which specifies the number of outputs. If -unspecified, it defaults to 2. - -@subsection Examples - -@itemize -@item -Create two separate outputs from the same input: -@example -[in] split [out0][out1] -@end example - -@item -To create 3 or more outputs, you need to specify the number of -outputs, like in: -@example -[in] asplit=3 [out0][out1][out2] -@end example - -@item -Create two separate outputs from the same input, one cropped and -one padded: -@example -[in] split [splitout1][splitout2]; -[splitout1] crop=100:100:0:0 [cropout]; -[splitout2] pad=200:200:100:100 [padout]; -@end example - -@item -Create 5 copies of the input audio with @command{ffmpeg}: -@example -ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT -@end example -@end itemize - -@section zmq, azmq - -Receive commands sent through a libzmq client, and forward them to -filters in the filtergraph. - -@code{zmq} and @code{azmq} work as a pass-through filters. @code{zmq} -must be inserted between two video filters, @code{azmq} between two -audio filters. - -To enable these filters you need to install the libzmq library and -headers and configure FFmpeg with @code{--enable-libzmq}. - -For more information about libzmq see: -@url{http://www.zeromq.org/} - -The @code{zmq} and @code{azmq} filters work as a libzmq server, which -receives messages sent through a network interface defined by the -@option{bind_address} option. - -The received message must be in the form: -@example -@var{TARGET} @var{COMMAND} [@var{ARG}] -@end example - -@var{TARGET} specifies the target of the command, usually the name of -the filter class or a specific filter instance name. - -@var{COMMAND} specifies the name of the command for the target filter. - -@var{ARG} is optional and specifies the optional argument list for the -given @var{COMMAND}. - -Upon reception, the message is processed and the corresponding command -is injected into the filtergraph. Depending on the result, the filter -will send a reply to the client, adopting the format: -@example -@var{ERROR_CODE} @var{ERROR_REASON} -@var{MESSAGE} -@end example - -@var{MESSAGE} is optional. - -@subsection Examples - -Look at @file{tools/zmqsend} for an example of a zmq client which can -be used to send commands processed by these filters. - -Consider the following filtergraph generated by @command{ffplay} -@example -ffplay -dumpgraph 1 -f lavfi " -color=s=100x100:c=red [l]; -color=s=100x100:c=blue [r]; -nullsrc=s=200x100, zmq [bg]; -[bg][l] overlay [bg+l]; -[bg+l][r] overlay=x=100 " -@end example - -To change the color of the left side of the video, the following -command can be used: -@example -echo Parsed_color_0 c yellow | tools/zmqsend -@end example - -To change the right side: -@example -echo Parsed_color_1 c pink | tools/zmqsend -@end example - -@c man end MULTIMEDIA FILTERS - -@chapter Multimedia Sources -@c man begin MULTIMEDIA SOURCES - -Below is a description of the currently available multimedia sources. - -@section amovie - -This is the same as @ref{movie} source, except it selects an audio -stream by default. - -@anchor{movie} -@section movie - -Read audio and/or video stream(s) from a movie container. +The shown line contains a sequence of key/value pairs of the form +@var{key}:@var{value}. It accepts the following parameters: @table @option +@item n +The (sequential) number of the input frame, starting from 0. + +@item pts +The Presentation TimeStamp of the input frame, expressed as a number of +time base units. The time base unit depends on the filter input pad. + +@item pts_time +The Presentation TimeStamp of the input frame, expressed as a number of +seconds. + +@item pos +The position of the frame in the input stream, or -1 if this information is +unavailable and/or meaningless (for example in case of synthetic video). + +@item fmt +The pixel format name. + +@item sar +The sample aspect ratio of the input frame, expressed in the form +@var{num}/@var{den}. + +@item s +The size of the input frame, expressed in the form +@var{width}x@var{height}. + +@item i +The type of interlaced mode ("P" for "progressive", "T" for top field first, "B" +for bottom field first). + +@item iskey +This is 1 if the frame is a key frame, 0 otherwise. + +@item type +The picture type of the input frame ("I" for an I-frame, "P" for a +P-frame, "B" for a B-frame, or "?" for an unknown type). +Also refer to the documentation of the @code{AVPictureType} enum and of +the @code{av_get_picture_type_char} function defined in +@file{libavutil/avutil.h}. + +@item checksum +The Adler-32 checksum of all the planes of the input frame. + +@item plane_checksum +The Adler-32 checksum of each plane of the input frame, expressed in the form +"[@var{c0} @var{c1} @var{c2} @var{c3}]". +@end table + +@section shuffleplanes + +Reorder and/or duplicate video planes. + +It accepts the following parameters: + +@table @option + +@item map0 +The index of the input plane to be used as the first output plane. + +@item map1 +The index of the input plane to be used as the second output plane. + +@item map2 +The index of the input plane to be used as the third output plane. + +@item map3 +The index of the input plane to be used as the fourth output plane. + +@end table + +The first plane has the index 0. The default is to keep the input unchanged. + +Swap the second and third planes of the input: +@example +avconv -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT +@end example + +@section split + +Split input video into several identical outputs. + +It accepts a single parameter, which specifies the number of outputs. If +unspecified, it defaults to 2. + +Create 5 copies of the input video: +@example +avconv -i INPUT -filter_complex split=5 OUTPUT +@end example + +@section transpose + +Transpose rows with columns in the input video and optionally flip it. + +It accepts the following parameters: + +@table @option + +@item dir +The direction of the transpose. + +@end table + +The direction can assume the following values: + +@table @samp +@item cclock_flip +Rotate by 90 degrees counterclockwise and vertically flip (default), that is: +@example +L.R L.l +. . -> . . +l.r R.r +@end example + +@item clock +Rotate by 90 degrees clockwise, that is: +@example +L.R l.L +. . -> . . +l.r r.R +@end example + +@item cclock +Rotate by 90 degrees counterclockwise, that is: +@example +L.R R.r +. . -> . . +l.r L.l +@end example + +@item clock_flip +Rotate by 90 degrees clockwise and vertically flip, that is: +@example +L.R r.R +. . -> . . +l.r l.L +@end example +@end table + +@section trim +Trim the input so that the output contains one continuous subpart of the input. + +It accepts the following parameters: +@table @option +@item start +The timestamp (in seconds) of the start of the kept section. The frame with the +timestamp @var{start} will be the first frame in the output. + +@item end +The timestamp (in seconds) of the first frame that will be dropped. The frame +immediately preceding the one with the timestamp @var{end} will be the last +frame in the output. + +@item start_pts +This is the same as @var{start}, except this option sets the start timestamp +in timebase units instead of seconds. + +@item end_pts +This is the same as @var{end}, except this option sets the end timestamp +in timebase units instead of seconds. + +@item duration +The maximum duration of the output in seconds. + +@item start_frame +The number of the first frame that should be passed to the output. + +@item end_frame +The number of the first frame that should be dropped. +@end table + +Note that the first two sets of the start/end options and the @option{duration} +option look at the frame timestamp, while the _frame variants simply count the +frames that pass through the filter. Also note that this filter does not modify +the timestamps. If you wish for the output timestamps to start at zero, insert a +setpts filter after the trim filter. + +If multiple start or end options are set, this filter tries to be greedy and +keep all the frames that match at least one of the specified constraints. To keep +only the part that matches all the constraints at once, chain multiple trim +filters. + +The defaults are such that all the input is kept. So it is possible to set e.g. +just the end values to keep everything before the specified time. + +Examples: +@itemize +@item +Drop everything except the second minute of input: +@example +avconv -i INPUT -vf trim=60:120 +@end example + +@item +Keep only the first second: +@example +avconv -i INPUT -vf trim=duration=1 +@end example + +@end itemize +@section unsharp + +Sharpen or blur the input video. + +It accepts the following parameters: + +@table @option + +@item luma_msize_x +Set the luma matrix horizontal size. It must be an integer between 3 +and 13. The default value is 5. + +@item luma_msize_y +Set the luma matrix vertical size. It must be an integer between 3 +and 13. The default value is 5. + +@item luma_amount +Set the luma effect strength. It must be a floating point number between -2.0 +and 5.0. The default value is 1.0. + +@item chroma_msize_x +Set the chroma matrix horizontal size. It must be an integer between 3 +and 13. The default value is 5. + +@item chroma_msize_y +Set the chroma matrix vertical size. It must be an integer between 3 +and 13. The default value is 5. + +@item chroma_amount +Set the chroma effect strength. It must be a floating point number between -2.0 +and 5.0. The default value is 0.0. + +@end table + +Negative values for the amount will blur the input video, while positive +values will sharpen. All parameters are optional and default to the +equivalent of the string '5:5:1.0:5:5:0.0'. + +@example +# Strong luma sharpen effect parameters +unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5 + +# A strong blur of both luma and chroma parameters +unsharp=7:7:-2:7:7:-2 + +# Use the default values with @command{avconv} +./avconv -i in.avi -vf "unsharp" out.mp4 +@end example + +@section vflip + +Flip the input video vertically. + +@example +./avconv -i in.avi -vf "vflip" out.avi +@end example + +@section yadif + +Deinterlace the input video ("yadif" means "yet another deinterlacing +filter"). + +It accepts the following parameters: + +@table @option + +@item mode +The interlacing mode to adopt. It accepts one of the following values: + +@table @option +@item 0 +Output one frame for each frame. +@item 1 +Output one frame for each field. +@item 2 +Like 0, but it skips the spatial interlacing check. +@item 3 +Like 1, but it skips the spatial interlacing check. +@end table + +The default value is 0. + +@item parity +The picture field parity assumed for the input interlaced video. It accepts one +of the following values: + +@table @option +@item 0 +Assume the top field is first. +@item 1 +Assume the bottom field is first. +@item -1 +Enable automatic detection of field parity. +@end table + +The default value is -1. +If the interlacing is unknown or the decoder does not export this information, +top field first will be assumed. + +@item auto +Whether the deinterlacer should trust the interlaced flag and only deinterlace +frames marked as interlaced. + +@table @option +@item 0 +Deinterlace all frames. +@item 1 +Only deinterlace frames marked as interlaced. +@end table + +The default value is 0. + +@end table + +@c man end VIDEO FILTERS + +@chapter Video Sources +@c man begin VIDEO SOURCES + +Below is a description of the currently available video sources. + +@section buffer + +Buffer video frames, and make them available to the filter chain. + +This source is mainly intended for a programmatic use, in particular +through the interface defined in @file{libavfilter/vsrc_buffer.h}. + +It accepts the following parameters: + +@table @option + +@item width +The input video width. + +@item height +The input video height. + +@item pix_fmt +The name of the input video pixel format. + +@item time_base +The time base used for input timestamps. + +@item sar +The sample (pixel) aspect ratio of the input video. + +@end table + +For example: +@example +buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1 +@end example + +will instruct the source to accept video frames with size 320x240 and +with format "yuv410p", assuming 1/24 as the timestamps timebase and +square pixels (1:1 sample aspect ratio). + +@section color + +Provide an uniformly colored input. + +It accepts the following parameters: + +@table @option + +@item color +Specify the color of the source. It can be the name of a color (case +insensitive match) or a 0xRRGGBB[AA] sequence, possibly followed by an +alpha specifier. The default value is "black". + +@item size +Specify the size of the sourced video, it may be a string of the form +@var{width}x@var{height}, or the name of a size abbreviation. The +default value is "320x240". + +@item framerate +Specify the frame rate of the sourced video, as the number of frames +generated per second. It has to be a string in the format +@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a floating point +number or a valid video frame rate abbreviation. The default value is +"25". + +@end table + +The following graph description will generate a red source +with an opacity of 0.2, with size "qcif" and a frame rate of 10 +frames per second, which will be overlayed over the source connected +to the pad with identifier "in": + +@example +"color=red@@0.2:qcif:10 [color]; [in][color] overlay [out]" +@end example + +@section movie + +Read a video stream from a movie container. + +Note that this source is a hack that bypasses the standard input path. It can be +useful in applications that do not support arbitrary filter graphs, but its use +is discouraged in those that do. It should never be used with +@command{avconv}; the @option{-filter_complex} option fully replaces it. + +It accepts the following parameters: + +@table @option + @item filename The name of the resource to read (not necessarily a file; it can also be a device or a stream accessed through some protocol). @@ -10432,27 +2922,11 @@ starting from this seek point. The parameter is evaluated with @code{av_strtod}, so the numerical value may be suffixed by an IS postfix. The default value is "0". -@item streams, s -Specifies the streams to read. Several streams can be specified, -separated by "+". The source will then have as many outputs, in the -same order. The syntax is explained in the ``Stream specifiers'' -section in the ffmpeg manual. Two special names, "dv" and "da" specify -respectively the default (best suited) video and audio stream. Default -is "dv", or "da" if the filter is called as "amovie". - @item stream_index, si Specifies the index of the video stream to read. If the value is -1, the most suitable video stream will be automatically selected. The default -value is "-1". Deprecated. If the filter is called "amovie", it will select -audio instead of video. +value is "-1". -@item loop -Specifies how many times to read the stream in sequence. -If the value is less than 1, the stream will be read again and again. -Default value is "1". - -Note that when the movie is looped the source timestamps are not -changed, so it will generate non monotonically increasing timestamps. @end table It allows overlaying a second video on top of the main input of @@ -10463,34 +2937,141 @@ input -----------> deltapts0 --> overlay --> output | movie --> scale--> deltapts1 -------+ @end example -@subsection Examples -@itemize -@item -Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it -on top of the input labelled "in": +Some examples: @example -movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over]; -[in] setpts=PTS-STARTPTS [main]; -[main][over] overlay=16:16 [out] +# Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it +# on top of the input labelled "in" +movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [movie]; +[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] + +# Read from a video4linux2 device, and overlay it on top of the input +# labelled "in" +movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie]; +[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] + @end example -@item -Read from a video4linux2 device, and overlay it on top of the input -labelled "in": +@section nullsrc + +Null video source: never return images. It is mainly useful as a +template and to be employed in analysis / debugging tools. + +It accepts a string of the form +@var{width}:@var{height}:@var{timebase} as an optional parameter. + +@var{width} and @var{height} specify the size of the configured +source. The default values of @var{width} and @var{height} are +respectively 352 and 288 (corresponding to the CIF size format). + +@var{timebase} specifies an arithmetic expression representing a +timebase. The expression can contain the constants "PI", "E", "PHI", and +"AVTB" (the default timebase), and defaults to the value "AVTB". + +@section frei0r_src + +Provide a frei0r source. + +To enable compilation of this filter you need to install the frei0r +header and configure Libav with --enable-frei0r. + +This source accepts the following parameters: + +@table @option + +@item size +The size of the video to generate. It may be a string of the form +@var{width}x@var{height} or a frame size abbreviation. + +@item framerate +The framerate of the generated video. It may be a string of the form +@var{num}/@var{den} or a frame rate abbreviation. + +@item filter_name +The name to the frei0r source to load. For more information regarding frei0r and +how to set the parameters, read the @ref{frei0r} section in the video filters +documentation. + +@item filter_params +A '|'-separated list of parameters to pass to the frei0r source. + +@end table + +An example: @example -movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over]; -[in] setpts=PTS-STARTPTS [main]; -[main][over] overlay=16:16 [out] +# Generate a frei0r partik0l source with size 200x200 and framerate 10 +# which is overlayed on the overlay filter main input +frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay @end example -@item -Read the first video stream and the audio stream with id 0x81 from -dvd.vob; the video is connected to the pad named "video" and the audio is -connected to the pad named "audio": -@example -movie=dvd.vob:s=v:0+#0x81 [video] [audio] -@end example -@end itemize +@section rgbtestsrc, testsrc -@c man end MULTIMEDIA SOURCES +The @code{rgbtestsrc} source generates an RGB test pattern useful for +detecting RGB vs BGR issues. You should see a red, green and blue +stripe from top to bottom. + +The @code{testsrc} source generates a test video pattern, showing a +color pattern, a scrolling gradient and a timestamp. This is mainly +intended for testing purposes. + +The sources accept the following parameters: + +@table @option + +@item size, s +Specify the size of the sourced video, it may be a string of the form +@var{width}x@var{height}, or the name of a size abbreviation. The +default value is "320x240". + +@item rate, r +Specify the frame rate of the sourced video, as the number of frames +generated per second. It has to be a string in the format +@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a floating point +number or a valid video frame rate abbreviation. The default value is +"25". + +@item sar +Set the sample aspect ratio of the sourced video. + +@item duration +Set the video duration of the sourced video. The accepted syntax is: +@example +[-]HH[:MM[:SS[.m...]]] +[-]S+[.m...] +@end example +Also see the the @code{av_parse_time()} function. + +If not specified, or the expressed duration is negative, the video is +supposed to be generated forever. +@end table + +For example the following: +@example +testsrc=duration=5.3:size=qcif:rate=10 +@end example + +will generate a video with a duration of 5.3 seconds, with size +176x144 and a framerate of 10 frames per second. + +@c man end VIDEO SOURCES + +@chapter Video Sinks +@c man begin VIDEO SINKS + +Below is a description of the currently available video sinks. + +@section buffersink + +Buffer video frames, and make them available to the end of the filter +graph. + +This sink is intended for programmatic use through the interface defined in +@file{libavfilter/buffersink.h}. + +@section nullsink + +Null video sink: do absolutely nothing with the input video. It is +mainly useful as a template and for use in analysis / debugging +tools. + +@c man end VIDEO SINKS diff --git a/doc/formats.texi b/doc/formats.texi deleted file mode 100644 index 027510eb6d..0000000000 --- a/doc/formats.texi +++ /dev/null @@ -1,212 +0,0 @@ -@chapter Format Options -@c man begin FORMAT OPTIONS - -The libavformat library provides some generic global options, which -can be set on all the muxers and demuxers. In addition each muxer or -demuxer may support so-called private options, which are specific for -that component. - -Options may be set by specifying -@var{option} @var{value} in the -FFmpeg tools, or by setting the value explicitly in the -@code{AVFormatContext} options or using the @file{libavutil/opt.h} API -for programmatic use. - -The list of supported options follows: - -@table @option -@item avioflags @var{flags} (@emph{input/output}) -Possible values: -@table @samp -@item direct -Reduce buffering. -@end table - -@item probesize @var{integer} (@emph{input}) -Set probing size in bytes, i.e. the size of the data to analyze to get -stream information. A higher value will allow to detect more -information in case it is dispersed into the stream, but will increase -latency. Must be an integer not lesser than 32. It is 5000000 by default. - -@item packetsize @var{integer} (@emph{output}) -Set packet size. - -@item fflags @var{flags} (@emph{input/output}) -Set format flags. - -Possible values: -@table @samp -@item ignidx -Ignore index. -@item genpts -Generate PTS. -@item nofillin -Do not fill in missing values that can be exactly calculated. -@item noparse -Disable AVParsers, this needs @code{+nofillin} too. -@item igndts -Ignore DTS. -@item discardcorrupt -Discard corrupted frames. -@item sortdts -Try to interleave output packets by DTS. -@item keepside -Do not merge side data. -@item latm -Enable RTP MP4A-LATM payload. -@item nobuffer -Reduce the latency introduced by optional buffering -@end table - -@item seek2any @var{integer} (@emph{input}) -Allow seeking to non-keyframes on demuxer level when supported if set to 1. -Default is 0. - -@item analyzeduration @var{integer} (@emph{input}) -Specify how many microseconds are analyzed to probe the input. A -higher value will allow to detect more accurate information, but will -increase latency. It defaults to 5,000,000 microseconds = 5 seconds. - -@item cryptokey @var{hexadecimal string} (@emph{input}) -Set decryption key. - -@item indexmem @var{integer} (@emph{input}) -Set max memory used for timestamp index (per stream). - -@item rtbufsize @var{integer} (@emph{input}) -Set max memory used for buffering real-time frames. - -@item fdebug @var{flags} (@emph{input/output}) -Print specific debug info. - -Possible values: -@table @samp -@item ts -@end table - -@item max_delay @var{integer} (@emph{input/output}) -Set maximum muxing or demuxing delay in microseconds. - -@item fpsprobesize @var{integer} (@emph{input}) -Set number of frames used to probe fps. - -@item audio_preload @var{integer} (@emph{output}) -Set microseconds by which audio packets should be interleaved earlier. - -@item chunk_duration @var{integer} (@emph{output}) -Set microseconds for each chunk. - -@item chunk_size @var{integer} (@emph{output}) -Set size in bytes for each chunk. - -@item err_detect, f_err_detect @var{flags} (@emph{input}) -Set error detection flags. @code{f_err_detect} is deprecated and -should be used only via the @command{ffmpeg} tool. - -Possible values: -@table @samp -@item crccheck -Verify embedded CRCs. -@item bitstream -Detect bitstream specification deviations. -@item buffer -Detect improper bitstream length. -@item explode -Abort decoding on minor error detection. -@item careful -Consider things that violate the spec and have not been seen in the -wild as errors. -@item compliant -Consider all spec non compliancies as errors. -@item aggressive -Consider things that a sane encoder should not do as an error. -@end table - -@item use_wallclock_as_timestamps @var{integer} (@emph{input}) -Use wallclock as timestamps. - -@item avoid_negative_ts @var{integer} (@emph{output}) - -Possible values: -@table @samp -@item make_non_negative -Shift timestamps to make them non-negative. -Also note that this affects only leading negative timestamps, and not -non-monotonic negative timestamps. -@item make_zero -Shift timestamps so that the first timestamp is 0. -@item auto (default) -Enables shifting when required by the target format. -@item disabled -Disables shifting of timestamp. -@end table - -When shifting is enabled, all output timestamps are shifted by the -same amount. Audio, video, and subtitles desynching and relative -timestamp differences are preserved compared to how they would have -been without shifting. - -@item skip_initial_bytes @var{integer} (@emph{input}) -Set number of bytes to skip before reading header and frames if set to 1. -Default is 0. - -@item correct_ts_overflow @var{integer} (@emph{input}) -Correct single timestamp overflows if set to 1. Default is 1. - -@item flush_packets @var{integer} (@emph{output}) -Flush the underlying I/O stream after each packet. Default 1 enables it, and -has the effect of reducing the latency; 0 disables it and may slightly -increase performance in some cases. - -@item output_ts_offset @var{offset} (@emph{output}) -Set the output time offset. - -@var{offset} must be a time duration specification, -see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. - -The offset is added by the muxer to the output timestamps. - -Specifying a positive offset means that the corresponding streams are -delayed bt the time duration specified in @var{offset}. Default value -is @code{0} (meaning that no offset is applied). -@end table - -@c man end FORMAT OPTIONS - -@anchor{Format stream specifiers} -@section Format stream specifiers - -Format stream specifiers allow selection of one or more streams that -match specific properties. - -Possible forms of stream specifiers are: -@table @option -@item @var{stream_index} -Matches the stream with this index. - -@item @var{stream_type}[:@var{stream_index}] -@var{stream_type} is one of following: 'v' for video, 'a' for audio, -'s' for subtitle, 'd' for data, and 't' for attachments. If -@var{stream_index} is given, then it matches the stream number -@var{stream_index} of this type. Otherwise, it matches all streams of -this type. - -@item p:@var{program_id}[:@var{stream_index}] -If @var{stream_index} is given, then it matches the stream with number -@var{stream_index} in the program with the id -@var{program_id}. Otherwise, it matches all streams in the program. - -@item #@var{stream_id} -Matches the stream by a format-specific ID. -@end table - -The exact semantics of stream specifiers is defined by the -@code{avformat_match_stream_specifier()} function declared in the -@file{libavformat/avformat.h} header. - -@ifclear config-writeonly -@include demuxers.texi -@end ifclear -@ifclear config-readonly -@include muxers.texi -@end ifclear -@include metadata.texi diff --git a/doc/general.texi b/doc/general.texi index 86569a20e7..3b7979683e 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -11,20 +11,12 @@ @chapter External libraries -FFmpeg can be hooked up with a number of external libraries to add support +Libav can be hooked up with a number of external libraries to add support for more formats. None of them are used by default, their use has to be explicitly requested by passing the appropriate flags to @command{./configure}. -@section OpenJPEG - -FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to -@url{http://www.openjpeg.org/} to get the libraries and follow the installation -instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to -@file{./configure}. - - -@section OpenCORE, VisualOn, and Fraunhofer libraries +@section OpenCORE and VisualOn libraries Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer libraries provide encoders for a number of audio codecs. @@ -32,19 +24,14 @@ libraries provide encoders for a number of audio codecs. @float NOTE OpenCORE and VisualOn libraries are under the Apache License 2.0 (see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is -incompatible to the LGPL version 2.1 and GPL version 2. You have to -upgrade FFmpeg's license to LGPL version 3 (or if you have enabled -GPL components, GPL version 3) by passing @code{--enable-version3} to configure in -order to use it. - -The Fraunhofer AAC library is licensed under a license incompatible to the GPL -and is not known to be compatible to the LGPL. Therefore, you have to pass -@code{--enable-nonfree} to configure to use it. +incompatible with the LGPL version 2.1 and GPL version 2. You have to +upgrade Libav's license to LGPL version 3 (or if you have enabled +GPL components, GPL version 3) to use it. @end float @subsection OpenCORE AMR -FFmpeg can make use of the OpenCORE libraries for AMR-NB +Libav can make use of the OpenCORE libraries for AMR-NB decoding/encoding and AMR-WB decoding. Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the @@ -54,7 +41,7 @@ Then pass @code{--enable-libopencore-amrnb} and/or @subsection VisualOn AAC encoder library -FFmpeg can make use of the VisualOn AACenc library for AAC encoding. +Libav can make use of the VisualOn AACenc library for AAC encoding. Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for installing the library. @@ -62,7 +49,7 @@ Then pass @code{--enable-libvo-aacenc} to configure to enable it. @subsection VisualOn AMR-WB encoder library -FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding. +Libav can make use of the VisualOn AMR-WBenc library for AMR-WB encoding. Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for installing the library. @@ -70,7 +57,7 @@ Then pass @code{--enable-libvo-amrwbenc} to configure to enable it. @subsection Fraunhofer AAC library -FFmpeg can make use of the Fraunhofer AAC library for AAC encoding. +Libav can make use of the Fraunhofer AAC library for AAC encoding. Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for installing the library. @@ -78,7 +65,7 @@ Then pass @code{--enable-libfdk-aac} to configure to enable it. @section LAME -FFmpeg can make use of the LAME library for MP3 encoding. +Libav can make use of the LAME library for MP3 encoding. Go to @url{http://lame.sourceforge.net/} and follow the instructions for installing the library. @@ -86,7 +73,7 @@ Then pass @code{--enable-libmp3lame} to configure to enable it. @section TwoLAME -FFmpeg can make use of the TwoLAME library for MP2 encoding. +Libav can make use of the TwoLAME library for MP2 encoding. Go to @url{http://www.twolame.org/} and follow the instructions for installing the library. @@ -94,7 +81,7 @@ Then pass @code{--enable-libtwolame} to configure to enable it. @section libvpx -FFmpeg can make use of the libvpx library for VP8/VP9 encoding. +Libav can make use of the libvpx library for VP8 encoding. Go to @url{http://www.webmproject.org/} and follow the instructions for installing the library. Then pass @code{--enable-libvpx} to configure to @@ -102,7 +89,7 @@ enable it. @section libwavpack -FFmpeg can make use of the libwavpack library for WavPack encoding. +Libav can make use of the libwavpack library for WavPack encoding. Go to @url{http://www.wavpack.com/} and follow the instructions for installing the library. Then pass @code{--enable-libwavpack} to configure to @@ -110,7 +97,7 @@ enable it. @section x264 -FFmpeg can make use of the x264 library for H.264 encoding. +Libav can make use of the x264 library for H.264 encoding. Go to @url{http://www.videolan.org/developers/x264.html} and follow the instructions for installing the library. Then pass @code{--enable-libx264} to @@ -119,12 +106,12 @@ configure to enable it. @float NOTE x264 is under the GNU Public License Version 2 or later (see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for -details), you must upgrade FFmpeg's license to GPL in order to use it. +details), you must upgrade Libav's license to GPL in order to use it. @end float @section x265 -FFmpeg can make use of the x265 library for HEVC encoding. +Libav can make use of the x265 library for HEVC encoding. Go to @url{http://x265.org/developers.html} and follow the instructions for installing the library. Then pass @code{--enable-libx265} to configure @@ -133,64 +120,61 @@ to enable it. @float note x265 is under the GNU Public License Version 2 or later (see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for -details), you must upgrade FFmpeg's license to GPL in order to use it. +details), you must upgrade Libav's license to GPL in order to use it. @end float @section libilbc iLBC is a narrowband speech codec that has been made freely available by Google as part of the WebRTC project. libilbc is a packaging friendly -copy of the iLBC codec. FFmpeg can make use of the libilbc library for +copy of the iLBC codec. Libav can make use of the libilbc library for iLBC encoding and decoding. Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for installing the library. Then pass @code{--enable-libilbc} to configure to enable it. -@section libzvbi - -libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB -teletext pages and DVB teletext subtitles. - -Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for -installing the library. Then pass @code{--enable-libzvbi} to configure to -enable it. - -@float NOTE -libzvbi is licensed under the GNU General Public License Version 2 or later -(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for details), -you must upgrade FFmpeg's license to GPL in order to use it. -@end float - @section AviSynth -FFmpeg can read AviSynth scripts as input. To enable support, pass -@code{--enable-avisynth} to configure. The correct headers are -included in compat/avisynth/, which allows the user to enable support -without needing to search for these headers themselves. +Libav can read AviSynth scripts as input. To enable support you need a +suitable @file{avisynth_c.h} header to compile against. The header in +classic AviSynth's CVS repository is not compatible as it has not been +updated to support AviSynth 2.6. AviSynth 2.5 is not supported by Libav. +Once you have the appropriate header, pass @code{--enable-avisynth} to +configure to enable AviSynth support. For Windows, supported AviSynth variants are -@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and +@url{http://avisynth.nl, AviSynth 2.6} for 32-bit builds and @url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds. +The necessary @file{avisynth_c.h} header is the variant in the @file{extras/} +directory of the @url{https://www.videolan.org/developers/x264.html, x264} +source tree. For convenience, this header is also available from a +@url{https://github.com/qyot27/avisynth_headers, temporary repository} +along with an installation routine. For Linux and OS X, the supported AviSynth variant is @url{https://github.com/avxsynth/avxsynth, AvxSynth}. +@file{avxsynth_c.h} is installed as part of the normal +build routine, as illustrated on +@url{https://github.com/avxsynth/avxsynth/wiki/System-Setup, AvxSynth's wiki}. +(the instructions for compiling its prerequisites are outdated, as FFMS 2.18 +or higher is now needed; the list of dependencies to be downloaded from the +repositories is still the same, though). @float NOTE -AviSynth and AvxSynth are loaded dynamically. Distributors can build FFmpeg +AviSynth and AvxSynth are loaded dynamically. Distributors can build Libav with @code{--enable-avisynth}, and the binaries will work regardless of the end user having AviSynth or AvxSynth installed - they'll only need to be installed to use AviSynth scripts (obviously). @end float - -@chapter Supported File Formats, Codecs or Features +@chapter Supported File Formats and Codecs You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list. @section File Formats -FFmpeg supports the following file formats through the @code{libavformat} +Libav supports the following file formats through the @code{libavformat} library: @multitable @columnfractions .4 .1 .1 .4 @@ -198,8 +182,6 @@ library: @item 4xm @tab @tab X @tab 4X Technologies format, used in some games. @item 8088flex TMV @tab @tab X -@item ACT Voice @tab @tab X - @tab contains G.729 audio @item Adobe Filmstrip @tab X @tab X @item Audio IFF (AIFF) @tab X @tab X @item American Laser Games MM @tab @tab X @@ -208,33 +190,21 @@ library: @item Amazing Studio Packed Animation File @tab @tab X @tab Multimedia format used in game Heart Of Darkness. @item Apple HTTP Live Streaming @tab @tab X -@item Artworx Data Format @tab @tab X -@item ADP @tab @tab X - @tab Audio format used on the Nintendo Gamecube. -@item AFC @tab @tab X - @tab Audio format used on the Nintendo Gamecube. @item ASF @tab X @tab X -@item AST @tab X @tab X - @tab Audio format used on the Nintendo Wii. @item AVI @tab X @tab X @item AviSynth @tab @tab X -@item AVR @tab @tab X - @tab Audio format used on Mac. @item AVS @tab @tab X @tab Multimedia format used by the Creature Shock game. @item Beam Software SIFF @tab @tab X @tab Audio and video format used in some games by Beam Software. @item Bethesda Softworks VID @tab @tab X @tab Used in some games from Bethesda Softworks. -@item Binary text @tab @tab X @item Bink @tab @tab X @tab Multimedia format used by many games. @item Bitmap Brothers JV @tab @tab X @tab Used in Z and Z95 games. @item Brute Force & Ignorance @tab @tab X @tab Used in the game Flash Traffic: City of Angels. -@item BRSTM @tab @tab X - @tab Audio format used on the Nintendo Wii. @item BWF @tab X @tab X @item CRI ADX @tab X @tab X @tab Audio-only format used in console video games. @@ -245,10 +215,9 @@ library: @tab Multimedia format used by Delphine Software games. @item CD+G @tab @tab X @tab Video format used by CD+G karaoke disks -@item Phantom Cine @tab @tab X @item Commodore CDXL @tab @tab X @tab Amiga CD video format -@item Core Audio Format @tab X @tab X +@item Core Audio Format @tab @tab X @tab Apple Core Audio Format @item CRC testing format @tab X @tab @item Creative Voice @tab X @tab X @@ -259,7 +228,6 @@ library: @item Deluxe Paint Animation @tab @tab X @item DFA @tab @tab X @tab This format is used in Chronomaster game -@item DSD Stream File (DSF) @tab @tab X @item DV video @tab X @tab X @item DXA @tab @tab X @tab This format is used in the non-Windows version of the Feeble Files @@ -267,8 +235,7 @@ library: @item Electronic Arts cdata @tab @tab X @item Electronic Arts Multimedia @tab @tab X @tab Used in various EA games; files have extensions like WVE and UV2. -@item Ensoniq Paris Audio File @tab @tab X -@item FFM (FFserver live feed) @tab X @tab X +@item FFM (AVserver live feed) @tab X @tab X @item Flash (SWF) @tab X @tab X @item Flash 9 (AVM2) @tab X @tab X @tab Only embedded audio is decoded. @@ -279,21 +246,15 @@ library: @item framecrc testing format @tab X @tab @item FunCom ISS @tab @tab X @tab Audio format used in various games from FunCom like The Longest Journey. -@item G.723.1 @tab X @tab X -@item G.729 BIT @tab X @tab X -@item G.729 raw @tab @tab X -@item GIF Animation @tab X @tab X +@item GIF Animation @tab X @tab @item GXF @tab X @tab X @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley playout servers. @item HNM @tab @tab X @tab Only version 4 supported, used in some games from Cryo Interactive -@item iCEDraw File @tab @tab X -@item ICO @tab X @tab X - @tab Microsoft Windows ICO @item id Quake II CIN video @tab @tab X @item id RoQ @tab X @tab X - @tab Used in Quake III, Jedi Knight 2 and other computer games. + @tab Used in Quake III, Jedi Knight 2, other computer games. @item IEC61937 encapsulation @tab X @tab X @item IFF @tab @tab X @tab Interchange File Format @@ -304,16 +265,11 @@ library: @tab A format generated by IndigoVision 8000 video server. @item IVF (On2) @tab X @tab X @tab A format used by libvpx -@item IRCAM @tab X @tab X @item LATM @tab X @tab X @item LMLM4 @tab @tab X @tab Used by Linux Media Labs MPEG-4 PCI boards -@item LOAS @tab @tab X - @tab contains LATM multiplexed AAC audio -@item LVF @tab @tab X @item LXF @tab @tab X @tab VR native stream format, used by Leitch/Harris' video servers. -@item Magic Lantern Video (MLV) @tab @tab X @item Matroska @tab X @tab X @item Matroska audio @tab X @tab @item FFmpeg metadata @tab X @tab X @@ -321,9 +277,6 @@ library: @item MAXIS XA @tab @tab X @tab Used in Sim City 3000; file extension .xa. @item MD Studio @tab @tab X -@item Metal Gear Solid: The Twin Snakes @tab @tab X -@item Megalux Frame @tab @tab X - @tab Used by Megalux Ultimate Paint @item Mobotix .mxg @tab @tab X @item Monkey's Audio @tab @tab X @item Motion Pixels MVI @tab @tab X @@ -353,7 +306,6 @@ library: @tab SMPTE 386M, D-10/IMX Mapping. @item NC camera feed @tab @tab X @tab NC (AVIP NC4600) camera streams -@item NIST SPeech HEader REsources @tab @tab X @item NTT TwinVQ (VQF) @tab @tab X @tab Nippon Telegraph and Telephone Corporation TwinVQ. @item Nullsoft Streaming Video @tab @tab X @@ -362,7 +314,6 @@ library: @tab NUT Open Container Format @item Ogg @tab X @tab X @item Playstation Portable PMP @tab @tab X -@item Portable Voice Format @tab @tab X @item TechnoTrend PVA @tab @tab X @tab Used by TechnoTrend DVB PCI boards. @item QCP @tab @tab X @@ -373,7 +324,6 @@ library: @item raw Dirac @tab X @tab X @item raw DNxHD @tab X @tab X @item raw DTS @tab X @tab X -@item raw DTS-HD @tab @tab X @item raw E-AC-3 @tab X @tab X @item raw FLAC @tab X @tab X @item raw GSM @tab @tab X @@ -394,7 +344,7 @@ library: @item raw Shorten @tab @tab X @item raw TAK @tab @tab X @item raw TrueHD @tab X @tab X -@item raw VC-1 @tab X @tab X +@item raw VC-1 @tab @tab X @item raw PCM A-law @tab X @tab X @item raw PCM mu-law @tab X @tab X @item raw PCM signed 8 bit @tab X @tab X @@ -420,19 +370,16 @@ library: @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio. @item RealMedia @tab X @tab X @item Redirector @tab @tab X -@item RedSpark @tab @tab X @item Renderware TeXture Dictionary @tab @tab X @item RL2 @tab @tab X @tab Audio and video format used in some games by Entertainment Software Partners. @item RPL/ARMovie @tab @tab X @item Lego Mindstorms RSO @tab X @tab X -@item RSD @tab @tab X @item RTMP @tab X @tab X @tab Output is performed by publishing stream to RTMP server @item RTP @tab X @tab X @item RTSP @tab X @tab X @item SAP @tab X @tab X -@item SBG @tab @tab X @item SDP @tab @tab X @item Sega FILM/CPK @tab @tab X @tab Used in many Sega Saturn console games. @@ -445,12 +392,12 @@ library: @tab Multimedia format used by many games. @item SMJPEG @tab X @tab X @tab Used in certain Loki game ports. -@item Smush @tab @tab X +@item Smush @tab Multimedia format used in some LucasArts games. @item Sony OpenMG (OMA) @tab X @tab X @tab Audio format used in Sony Sonic Stage and Sony Vegas. @item Sony PlayStation STR @tab @tab X -@item Sony Wave64 (W64) @tab X @tab X +@item Sony Wave64 (W64) @tab @tab X @item SoX native format @tab X @tab X @item SUN AU format @tab X @tab X @item Text files @tab @tab X @@ -460,11 +407,10 @@ library: @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback. @item True Audio @tab @tab X @item VC-1 test bitstream @tab X @tab X -@item Vivo @tab @tab X @item WAV @tab X @tab X -@item WavPack @tab X @tab X +@item WavPack @tab @tab X @item WebM @tab X @tab X -@item Windows Televison (WTV) @tab X @tab X +@item Windows Televison (WTV) @tab @tab X @item Wing Commander III movie @tab @tab X @tab Multimedia format used in Origin's Wing Commander III computer game. @item Westwood Studios audio @tab @tab X @@ -475,16 +421,16 @@ library: @tab Microsoft video container used in Xbox games. @item xWMA @tab @tab X @tab Microsoft audio container used by XAudio 2. -@item eXtended BINary text (XBIN) @tab @tab X @item YUV4MPEG pipe @tab X @tab X @item Psygnosis YOP @tab @tab X +@item ZeroCodec Lossless Video @tab @tab X @end multitable @code{X} means that encoding (resp. decoding) is supported. @section Image Formats -FFmpeg can read and write images for each frame of a video sequence. The +Libav can read and write images for each frame of a video sequence. The following image formats are supported: @multitable @columnfractions .4 .1 .1 .4 @@ -494,6 +440,7 @@ following image formats are supported: @item Alias PIX @tab X @tab X @tab Alias/Wavefront PIX image format @item animated GIF @tab X @tab X + @tab Only uncompressed GIFs are generated. @item BMP @tab X @tab X @tab Microsoft BMP image @item BRender PIX @tab @tab X @@ -504,7 +451,8 @@ following image formats are supported: @tab OpenEXR @item JPEG @tab X @tab X @tab Progressive JPEG is not supported. -@item JPEG 2000 @tab X @tab X +@item JPEG 2000 @tab E @tab X + @tab encoding supported through external library libopenjpeg @item JPEG-LS @tab X @tab X @item LJPEG @tab X @tab @tab Lossless JPEG @@ -521,6 +469,7 @@ following image formats are supported: @item PIC @tab @tab X @tab Pictor/PC Paint @item PNG @tab X @tab X + @tab 2/4 bpp not supported yet @item PPM @tab X @tab X @tab Portable PixelMap image @item PTX @tab @tab X @@ -537,8 +486,6 @@ following image formats are supported: @tab WebP image format, encoding supported through external library libwebp @item XBM @tab X @tab X @tab X BitMap image format -@item XFace @tab X @tab X - @tab X-Face image format @item XWD @tab X @tab X @tab X Window Dump image format @end multitable @@ -554,12 +501,14 @@ following image formats are supported: @item 4X Movie @tab @tab X @tab Used in certain computer games. @item 8088flex TMV @tab @tab X +@item 8SVX exponential @tab @tab X +@item 8SVX fibonacci @tab @tab X @item A64 multicolor @tab X @tab @tab Creates video suitable to be played on a commodore 64 (multicolor mode). @item Amazing Studio PAF Video @tab @tab X @item American Laser Games MM @tab @tab X @tab Used in games like Mad Dog McCree. -@item AMV Video @tab X @tab X +@item AMV Video @tab @tab X @tab Used in Chinese MP3 players. @item ANSI/ASCII art @tab @tab X @item Apple Intermediate Codec @tab @tab X @@ -580,18 +529,13 @@ following image formats are supported: @item Autodesk Animator Flic video @tab @tab X @item Autodesk RLE @tab @tab X @tab fourcc: AASC -@item Avid 1:1 10-bit RGB Packer @tab X @tab X - @tab fourcc: AVrp @item AVS (Audio Video Standard) video @tab @tab X @tab Video encoding used by the Creature Shock game. -@item AYUV @tab X @tab X - @tab Microsoft uncompressed packed 4:4:4:4 @item Beam Software VB @tab @tab X @item Bethesda VID video @tab @tab X @tab Used in some games from Bethesda Softworks. @item Bink Video @tab @tab X @item Bitmap Brothers JV video @tab @tab X -@item y41p Brooktree uncompressed 4:1:1 12-bit @tab X @tab X @item Brute Force & Ignorance @tab @tab X @tab Used in the game Flash Traffic: City of Angels. @item C93 video @tab @tab X @@ -611,11 +555,10 @@ following image formats are supported: @item Cinepak @tab @tab X @item Cirrus Logic AccuPak @tab X @tab X @tab fourcc: CLJR -@item CPiA Video Format @tab @tab X @item Creative YUV (CYUV) @tab @tab X @item DFA @tab @tab X @tab Codec used in Chronomaster game. -@item Dirac @tab E @tab X +@item Dirac @tab E @tab E @tab supported through external library libschroedinger @item Deluxe Paint Animation @tab @tab X @item DNxHD @tab X @tab X @@ -637,10 +580,10 @@ following image formats are supported: @item Escape 124 @tab @tab X @item Escape 130 @tab @tab X @item FFmpeg video codec #1 @tab X @tab X - @tab lossless codec (fourcc: FFV1) + @tab experimental lossless codec (fourcc: FFV1) @item Flash Screen Video v1 @tab X @tab X @tab fourcc: FSV1 -@item Flash Screen Video v2 @tab X @tab X +@item Flash Screen Video v2 @tab @tab X @item Flash Video (FLV) @tab X @tab X @tab Sorenson H.263 used in Flash @item Forward Uncompressed @tab @tab X @@ -676,7 +619,6 @@ following image formats are supported: @tab Used in the game Cyberia from Interplay. @item Interplay MVE video @tab @tab X @tab Used in Interplay .MVE files. -@item J2K @tab X @tab X @item Karl Morton's video codec @tab @tab X @tab Codec used in Worms games. @item Kega Game Video (KGV1) @tab @tab X @@ -685,8 +627,8 @@ following image formats are supported: @item LCL (LossLess Codec Library) MSZH @tab @tab X @item LCL (LossLess Codec Library) ZLIB @tab E @tab E @item LOCO @tab @tab X -@item LucasArts SANM/Smush @tab @tab X - @tab Used in LucasArts games / SMUSH animations. +@item LucasArts SANM @tab @tab X + @tab Used in LucasArts SMUSH animations. @item lossless MJPEG @tab X @tab X @item Microsoft ATC Screen @tab @tab X @tab Also known as Microsoft Screen 3. @@ -726,11 +668,9 @@ following image formats are supported: @item VP8 @tab E @tab X @tab fourcc: VP80, encoding supported through external library libvpx @item VP9 @tab E @tab X - @tab encoding supported through external library libvpx -@item Pinnacle TARGA CineWave YUV16 @tab @tab X - @tab fourcc: Y216 -@item Prores @tab @tab X - @tab fourcc: apch,apcn,apcs,apco + @tab Encoding supported through external library libvpx +@item planar RGB @tab @tab X + @tab fourcc: 8BPS @item Q-team QPEG @tab @tab X @tab fourccs: QPEG, Q1.0, Q1.1 @item QuickTime 8BPS video @tab @tab X @@ -740,8 +680,8 @@ following image formats are supported: @tab fourcc: 'smc ' @item QuickTime video (RPZA) @tab @tab X @tab fourcc: rpza -@item R10K AJA Kona 10-bit RGB Codec @tab X @tab X -@item R210 Quicktime Uncompressed RGB 10-bit @tab X @tab X +@item R10K AJA Kona 10-bit RGB Codec @tab @tab X +@item R210 Quicktime Uncompressed RGB 10-bit @tab @tab X @item Raw Video @tab X @tab X @item RealVideo 1.0 @tab X @tab X @item RealVideo 2.0 @tab X @tab X @@ -760,8 +700,6 @@ following image formats are supported: @item Smacker video @tab @tab X @tab Video encoding used in Smacker. @item SMPTE VC-1 @tab @tab X -@item Snow @tab X @tab X - @tab experimental wavelet codec (fourcc: SNOW) @item Sony PlayStation MDEC (Motion DECoder) @tab @tab X @item Sorenson Vector Quantizer 1 @tab X @tab X @tab fourcc: SVQ1 @@ -779,8 +717,6 @@ following image formats are supported: @tab Codec used in DOS CD-ROM FlashBack game. @item Ut Video @tab X @tab X @item v210 QuickTime uncompressed 4:2:2 10-bit @tab X @tab X -@item v308 QuickTime uncompressed 4:4:4 @tab X @tab X -@item v408 QuickTime uncompressed 4:4:4:4 @tab X @tab X @item v410 QuickTime uncompressed 4:4:4 10-bit @tab X @tab X @item VBLE Lossless Codec @tab @tab X @item VMware Screen Codec / VMware Video @tab @tab X @@ -799,9 +735,6 @@ following image formats are supported: @item WMV7 @tab X @tab X @item YAMAHA SMAF @tab X @tab X @item Psygnosis YOP Video @tab @tab X -@item yuv4 @tab X @tab X - @tab libquicktime uncompressed packed 4:2:0 -@item ZeroCodec Lossless Video @tab @tab X @item ZLIB @tab X @tab X @tab part of LCL, encoder experimental @item Zip Motion Blocks Video @tab X @tab X @@ -816,13 +749,10 @@ following image formats are supported: @multitable @columnfractions .4 .1 .1 .4 @item Name @tab Encoding @tab Decoding @tab Comments -@item 8SVX exponential @tab @tab X -@item 8SVX fibonacci @tab @tab X -@item AAC+ @tab E @tab X - @tab encoding supported through external library libaacplus +@item 8SVX audio @tab @tab X @item AAC @tab E @tab X @tab encoding supported through external library libfaac and libvo-aacenc -@item AC-3 @tab IX @tab IX +@item AC-3 @tab IX @tab X @item ADPCM 4X Movie @tab @tab X @item ADPCM CDROM XA @tab @tab X @item ADPCM Creative Technology @tab @tab X @@ -848,21 +778,19 @@ following image formats are supported: @item ADPCM IMA Westwood @tab @tab X @item ADPCM ISS IMA @tab @tab X @tab Used in FunCom games. -@item ADPCM IMA Dialogic @tab @tab X @item ADPCM IMA Duck DK3 @tab @tab X @tab Used in some Sega Saturn console games. @item ADPCM IMA Duck DK4 @tab @tab X @tab Used in some Sega Saturn console games. -@item ADPCM IMA Radical @tab @tab X @item ADPCM Microsoft @tab X @tab X @item ADPCM MS IMA @tab X @tab X -@item ADPCM Nintendo Gamecube AFC @tab @tab X -@item ADPCM Nintendo Gamecube DTK @tab @tab X @item ADPCM Nintendo Gamecube THP @tab @tab X @item ADPCM QT IMA @tab X @tab X @item ADPCM SEGA CRI ADX @tab X @tab X @tab Used in Sega Dreamcast games. @item ADPCM Shockwave Flash @tab X @tab X +@item ADPCM SMJPEG IMA @tab @tab X + @tab Used in certain Loki game ports. @item ADPCM Sound Blaster Pro 2-bit @tab @tab X @item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X @item ADPCM Sound Blaster Pro 4-bit @tab @tab X @@ -883,16 +811,14 @@ following image formats are supported: @item ATRAC3+ @tab @tab X @item Bink Audio @tab @tab X @tab Used in Bink and Smacker files in many games. -@item CELT @tab @tab E - @tab decoding supported through external library libcelt @item Delphine Software International CIN audio @tab @tab X @tab Codec used in Delphine Software International games. @item Discworld II BMV Audio @tab @tab X @item COOK @tab @tab X @tab All versions except 5.1 are supported. -@item DCA (DTS Coherent Acoustics) @tab X @tab X +@item DCA (DTS Coherent Acoustics) @tab @tab X @item DPCM id RoQ @tab X @tab X - @tab Used in Quake III, Jedi Knight 2 and other computer games. + @tab Used in Quake III, Jedi Knight 2, other computer games. @item DPCM Interplay @tab @tab X @tab Used in various Interplay computer games. @item DPCM Sierra Online @tab @tab X @@ -900,17 +826,11 @@ following image formats are supported: @item DPCM Sol @tab @tab X @item DPCM Xan @tab @tab X @tab Used in Origin's Wing Commander IV AVI files. -@item DSD (Direct Stream Digitial), least significant bit first @tab @tab X -@item DSD (Direct Stream Digitial), most significant bit first @tab @tab X -@item DSD (Direct Stream Digitial), least significant bit first, planar @tab @tab X -@item DSD (Direct Stream Digitial), most significant bit first, planar @tab @tab X @item DSP Group TrueSpeech @tab @tab X @item DV audio @tab @tab X @item Enhanced AC-3 @tab X @tab X -@item EVRC (Enhanced Variable Rate Codec) @tab @tab X @item FLAC (Free Lossless Audio Codec) @tab X @tab IX -@item G.723.1 @tab X @tab X -@item G.729 @tab @tab X +@item G.723.1 @tab @tab X @item GSM @tab E @tab X @tab encoding supported through external library libgsm @item GSM Microsoft variant @tab E @tab X @@ -924,8 +844,9 @@ following image formats are supported: @item MLP (Meridian Lossless Packing) @tab @tab X @tab Used in DVD-Audio discs. @item Monkey's Audio @tab @tab X + @tab Only versions 3.97-3.99 are supported. @item MP1 (MPEG audio layer 1) @tab @tab IX -@item MP2 (MPEG audio layer 2) @tab IX @tab IX +@item MP2 (MPEG audio layer 2) @tab IE @tab IX @tab encoding supported also through external library TwoLAME @item MP3 (MPEG audio layer 3) @tab E @tab IX @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported @@ -938,11 +859,9 @@ following image formats are supported: @tab supported through external library libopus @item PCM A-law @tab X @tab X @item PCM mu-law @tab X @tab X -@item PCM signed 8-bit planar @tab X @tab X -@item PCM signed 16-bit big-endian planar @tab X @tab X -@item PCM signed 16-bit little-endian planar @tab X @tab X -@item PCM signed 24-bit little-endian planar @tab X @tab X -@item PCM signed 32-bit little-endian planar @tab X @tab X +@item PCM signed 16-bit little-endian planar @tab @tab X +@item PCM signed 24-bit little-endian planar @tab @tab X +@item PCM signed 32-bit little-endian planar @tab @tab X @item PCM 32-bit floating point big-endian @tab X @tab X @item PCM 32-bit floating point little-endian @tab X @tab X @item PCM 64-bit floating point big-endian @tab X @tab X @@ -979,24 +898,19 @@ following image formats are supported: @item Sierra VMD audio @tab @tab X @tab Used in Sierra VMD files. @item Smacker audio @tab @tab X -@item SMPTE 302M AES3 audio @tab X @tab X -@item Sonic @tab X @tab X - @tab experimental codec -@item Sonic lossless @tab X @tab X - @tab experimental codec +@item SMPTE 302M AES3 audio @tab @tab X @item Speex @tab E @tab E @tab supported through external library libspeex @item TAK (Tom's lossless Audio Kompressor) @tab @tab X -@item True Audio (TTA) @tab X @tab X +@item True Audio (TTA) @tab @tab X @item TrueHD @tab @tab X @tab Used in HD-DVD and Blu-Ray discs. @item TwinVQ (VQF flavor) @tab @tab X -@item VIMA @tab @tab X - @tab Used in LucasArts SMUSH animations. @item Vorbis @tab E @tab X @tab A native but very primitive encoder exists. @item Voxware MetaSound @tab @tab X -@item WavPack @tab X @tab X +@item WavPack @tab E @tab X + @tab supported through external library libwavpack @item Westwood Audio (SND1) @tab @tab X @item Windows Media Audio 1 @tab X @tab X @item Windows Media Audio 2 @tab X @tab X @@ -1016,40 +930,21 @@ performance on systems without hardware floating point support). @multitable @columnfractions .4 .1 .1 .1 .1 @item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding -@item 3GPP Timed Text @tab @tab @tab X @tab X -@item AQTitle @tab @tab X @tab @tab X -@item DVB @tab X @tab X @tab X @tab X -@item DVB teletext @tab @tab X @tab @tab E -@item DVD @tab X @tab X @tab X @tab X -@item JACOsub @tab X @tab X @tab @tab X -@item MicroDVD @tab X @tab X @tab @tab X -@item MPL2 @tab @tab X @tab @tab X -@item MPsub (MPlayer) @tab @tab X @tab @tab X -@item PGS @tab @tab @tab @tab X -@item PJS (Phoenix) @tab @tab X @tab @tab X -@item RealText @tab @tab X @tab @tab X -@item SAMI @tab @tab X @tab @tab X -@item SSA/ASS @tab X @tab X @tab X @tab X -@item SubRip (SRT) @tab X @tab X @tab X @tab X -@item SubViewer v1 @tab @tab X @tab @tab X -@item SubViewer @tab @tab X @tab @tab X -@item TED Talks captions @tab @tab X @tab @tab X -@item VobSub (IDX+SUB) @tab @tab X @tab @tab X -@item VPlayer @tab @tab X @tab @tab X -@item WebVTT @tab X @tab X @tab X @tab X -@item XSUB @tab @tab @tab X @tab X +@item SSA/ASS @tab X @tab X @tab X @tab X +@item DVB @tab X @tab X @tab X @tab X +@item DVD @tab X @tab X @tab X @tab X +@item PGS @tab @tab @tab @tab X +@item SubRip (SRT) @tab X @tab X @tab @tab X +@item XSUB @tab @tab @tab X @tab X @end multitable @code{X} means that the feature is supported. -@code{E} means that support is provided through an external library. - @section Network Protocols @multitable @columnfractions .4 .1 @item Name @tab Support @item file @tab X -@item FTP @tab X @item Gopher @tab X @item HLS @tab X @item HTTP @tab X @@ -1065,7 +960,6 @@ performance on systems without hardware floating point support). @item RTMPTS @tab X @item RTP @tab X @item SCTP @tab X -@item SFTP @tab E @item TCP @tab X @item TLS @tab X @item UDP @tab X @@ -1082,36 +976,18 @@ performance on systems without hardware floating point support). @item Name @tab Input @tab Output @item ALSA @tab X @tab X @item BKTR @tab X @tab -@item caca @tab @tab X @item DV1394 @tab X @tab -@item Lavfi virtual device @tab X @tab -@item Linux framebuffer @tab X @tab X +@item Linux framebuffer @tab X @tab @item JACK @tab X @tab @item LIBCDIO @tab X @item LIBDC1394 @tab X @tab -@item OpenAL @tab X -@item OpenGL @tab @tab X @item OSS @tab X @tab X -@item PulseAudio @tab X @tab X -@item SDL @tab @tab X -@item Video4Linux2 @tab X @tab X +@item Pulseaudio @tab X @tab +@item Video4Linux2 @tab X @tab @item VfW capture @tab X @tab @item X11 grabbing @tab X @tab -@item Win32 grabbing @tab X @tab @end multitable @code{X} means that input/output is supported. -@section Timecode - -@multitable @columnfractions .4 .1 .1 -@item Codec/format @tab Read @tab Write -@item AVI @tab X @tab X -@item DV @tab X @tab X -@item GXF @tab X @tab X -@item MOV @tab X @tab X -@item MPEG1/2 @tab X @tab X -@item MXF @tab X @tab X -@end multitable - @bye diff --git a/doc/git-howto.texi b/doc/git-howto.texi index 750662a458..5a8e2a3823 100644 --- a/doc/git-howto.texi +++ b/doc/git-howto.texi @@ -1,9 +1,9 @@ \input texinfo @c -*- texinfo -*- -@settitle Using git to develop FFmpeg +@settitle Using git to develop Libav @titlepage -@center @titlefont{Using git to develop FFmpeg} +@center @titlefont{Using git to develop Libav} @end titlepage @top @@ -39,7 +39,7 @@ For more information about the Git project, visit the Consult these resources whenever you have problems, they are quite exhaustive. -What follows now is a basic introduction to Git and some FFmpeg-specific +What follows now is a basic introduction to Git and some Libav-specific guidelines to ease the contribution to the project @chapter Basics Usage @@ -53,16 +53,16 @@ Most distribution and operating system provide a package for it. @section Cloning the source tree @example -git clone git://source.ffmpeg.org/ffmpeg +git clone git://git.libav.org/libav.git @end example -This will put the FFmpeg sources into the directory @var{}. +This will put the Libav sources into the directory @var{}. @example -git clone git@@source.ffmpeg.org:ffmpeg +git clone git@@git.libav.org:libav.git @end example -This will put the FFmpeg sources into the directory @var{} and let +This will put the Libav sources into the directory @var{} and let you push back your changes to the remote repository. Make sure that you do not have Windows line endings in your checkouts, @@ -85,7 +85,7 @@ can be remote. By default the master branch tracks the branch master in the remote origin. @float IMPORTANT -@command{--rebase} (see below) is recommended. +Since merge commits are forbidden @command{--rebase} (see below) is recommended. @end float @section Rebasing your local branches @@ -96,7 +96,7 @@ git pull --rebase fetches the changes from the main repository and replays your local commits over it. This is required to keep all your local changes at the top of -FFmpeg's master tree. The master tree will reject pushes with merge commits. +Libav's master tree. The master tree will reject pushes with merge commits. @section Adding/removing files/directories @@ -127,7 +127,7 @@ git log @end example You may also use the graphical tools like gitview or gitk or the web -interface available at http://source.ffmpeg.org/ +interface available at http://git.libav.org/ @section Checking source tree status @@ -261,7 +261,7 @@ git commit @chapter Git configuration In order to simplify a few workflows, it is advisable to configure both -your personal Git installation and your local FFmpeg repository. +your personal Git installation and your local Libav repository. @section Personal Git installation @@ -276,15 +276,15 @@ and @command{git format-patch} detect renames: @section Repository configuration In order to have @command{git send-email} automatically send patches -to the ffmpeg-devel mailing list, add the following stanza -to @file{/path/to/ffmpeg/repository/.git/config}: +to the libav-devel mailing list, add the following stanza +to @file{/path/to/libav/repository/.git/config}: @example [sendemail] - to = ffmpeg-devel@@ffmpeg.org + to = libav-devel@@libav.org @end example -@chapter FFmpeg specific +@chapter Libav specific @section Reverting broken commits @@ -299,7 +299,7 @@ the current branch history. git commit --amend @end example -allows one to amend the last commit details quickly. +allows to amend the last commit details quickly. @example git rebase -i origin/master @@ -381,35 +381,60 @@ proper order. This list tries to be exhaustive. In case you are just pushing a typo in a comment, some of the steps may be unnecessary. Apply your common sense, but if in doubt, err on the side of caution. -First, make sure that the commits and branches you are going to push -match what you want pushed and that nothing is missing, extraneous or -wrong. You can see what will be pushed by running the git push command -with --dry-run first. And then inspecting the commits listed with -@command{git log -p 1234567..987654}. The @command{git status} command -may help in finding local changes that have been forgotten to be added. +First make sure your Git repository is on a branch that is a direct +descendant of the Libav master branch, which is the only one from which +pushing to Libav is possible. Then run the following command: -Next let the code pass through a full run of our testsuite. +@itemize +@item @command{git log --patch --stat origin/master..} + +to make sure that only the commits you want to push are pending, that +the log messages of the commits are correct and descriptive and contain +no cruft from @command{git am} and to doublecheck that the commits you +want to push really only contain the changes they are supposed to contain. + +@item @command{git status} + +to ensure no local changes still need to be committed and that no local +changes may have thrown off the results of your testing. +@end itemize + +Next let the code pass through a full run of our testsuite. Before you do, +the command @command{make fate-rsync} will update the test samples. Changes +to the samples set are not very common and commits depending on samples +changes are delayed for at least 24 hours to allow the new samples to +propagate, so updating it once per day is sufficient. Now execute @itemize @item @command{make distclean} -@item @command{/path/to/ffmpeg/configure} +@item @command{/path/to/libav/configure} @item @command{make check} -@item if fate fails due to missing samples run @command{make fate-rsync} and retry @end itemize -Make sure all your changes have been checked before pushing them, the -testsuite only checks against regressions and that only to some extend. It does -obviously not check newly added features/code to be working unless you have -added a test for that (which is recommended). +While the test suite covers a wide range of possible problems, it is not +a panacea. Do not hesitate to perform any other tests necessary to convince +yourself that the changes you are about to push actually work as expected. Also note that every single commit should pass the test suite, not just -the result of a series of patches. +the result of a series of patches. So if you have a series of commits +to push, run the test suite on every single commit. + +Give other developers a reasonable amount of time to look at and review +patches before you push them. Not everybody is online 24/7, but may wish +to look at and comment on a patch nonetheless. The time you leave depends +on the urgency and complexity of the patch. Use your common sense to pick +a timeframe that allows everybody that you think may wish to comment +and/or should comment on the change an opportunity to see it. + +Finally, after pushing, mark all patches as committed on +@url{http://patches.libav.org/,patchwork}. +Sometimes this is not automatically done when a patch has been +slightly modified from the version on the mailing list. +Also update previous incarnations of the patches you push so that +patchwork is not cluttered with cruft. -Once everything passed, push the changes to your public ffmpeg clone and post a -merge request to ffmpeg-devel. You can also push them directly but this is not -recommended. @chapter Server Issues -Contact the project admins @email{root@@ffmpeg.org} if you have technical +Contact the project admins @email{git@@libav.org} if you have technical problems with the GIT server. diff --git a/doc/git-howto.txt b/doc/git-howto.txt new file mode 100644 index 0000000000..036b567084 --- /dev/null +++ b/doc/git-howto.txt @@ -0,0 +1,272 @@ + +About Git write access: +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +Before everything else, you should know how to use GIT properly. +Luckily Git comes with excellent documentation. + + git --help + man git + +shows you the available subcommands, + + git --help + man git- + +shows information about the subcommand . + +The most comprehensive manual is the website Git Reference + +http://gitref.org/ + +For more information about the Git project, visit + +http://git-scm.com/ + +Consult these resources whenever you have problems, they are quite exhaustive. + +You do not need a special username or password. +All you need is to provide a ssh public key to the Git server admin. + +What follows now is a basic introduction to Git and some Libav-specific +guidelines. Read it at least once, if you are granted commit privileges to the +Libav project you are expected to be familiar with these rules. + + + +I. BASICS: +========== + +0. Get GIT: + + You can get git from http://git-scm.com/ + + +1. Cloning the source tree: + + git clone git://git.libav.org/libav.git + + This will put the Libav sources into the directory . + + git clone git@git.libav.org:libav.git + + This will put the Libav sources into the directory and let + you push back your changes to the remote repository. + + +2. Updating the source tree to the latest revision: + + git pull (--ff-only) + + pulls in the latest changes from the tracked branch. The tracked branch + can be remote. By default the master branch tracks the branch master in + the remote origin. + Caveat: Since merge commits are forbidden at least for the initial + months of git --ff-only or --rebase (see below) are recommended. + --ff-only will fail and not create merge commits if your branch + has diverged (has a different history) from the tracked branch. + +2.a Rebasing your local branches: + + git pull --rebase + + fetches the changes from the main repository and replays your local commits + over it. This is required to keep all your local changes at the top of + Libav's master tree. The master tree will reject pushes with merge commits. + + +3. Adding/removing files/directories: + + git add [-A] + git rm [-r] + + GIT needs to get notified of all changes you make to your working + directory that makes files appear or disappear. + Line moves across files are automatically tracked. + + +4. Showing modifications: + + git diff + + will show all local modifications in your working directory as unified diff. + + +5. Inspecting the changelog: + + git log + + You may also use the graphical tools like gitview or gitk or the web + interface available at http://git.libav.org/ + +6. Checking source tree status: + + git status + + detects all the changes you made and lists what actions will be taken in case + of a commit (additions, modifications, deletions, etc.). + + +7. Committing: + + git diff --check + + to double check your changes before committing them to avoid trouble later + on. All experienced developers do this on each and every commit, no matter + how small. + Every one of them has been saved from looking like a fool by this many times. + It's very easy for stray debug output or cosmetic modifications to slip in, + please avoid problems through this extra level of scrutiny. + + For cosmetics-only commits you should get (almost) empty output from + + git diff -w -b + + Also check the output of + + git status + + to make sure you don't have untracked files or deletions. + + git add [-i|-p|-A] + + Make sure you have told git your name and email address, e.g. by running + git config --global user.name "My Name" + git config --global user.email my@email.invalid + (--global to set the global configuration for all your git checkouts). + + Git will select the changes to the files for commit. Optionally you can use + the interactive or the patch mode to select hunk by hunk what should be + added to the commit. + + git commit + + Git will commit the selected changes to your current local branch. + + You will be prompted for a log message in an editor, which is either + set in your personal configuration file through + + git config core.editor + + or set by one of the following environment variables: + GIT_EDITOR, VISUAL or EDITOR. + + Log messages should be concise but descriptive. Explain why you made a change, + what you did will be obvious from the changes themselves most of the time. + Saying just "bug fix" or "10l" is bad. Remember that people of varying skill + levels look at and educate themselves while reading through your code. Don't + include filenames in log messages, Git provides that information. + + Possibly make the commit message have a terse, descriptive first line, an + empty line and then a full description. The first line will be used to name + the patch by git format-patch. + + +8. Renaming/moving/copying files or contents of files: + + Git automatically tracks such changes, making those normal commits. + + mv/cp path/file otherpath/otherfile + + git add [-A] . + + git commit + + Do not move, rename or copy files of which you are not the maintainer without + discussing it on the mailing list first! + +9. Reverting broken commits + + git revert + + git revert will generate a revert commit. This will not make the faulty + commit disappear from the history. + + git reset + + git reset will uncommit the changes till rewriting the current + branch history. + + git commit --amend + + allows to amend the last commit details quickly. + + git rebase -i origin/master + + will replay local commits over the main repository allowing to edit, + merge or remove some of them in the process. + + Note that the reset, commit --amend and rebase rewrite history, so you + should use them ONLY on your local or topic branches. + + The main repository will reject those changes. + +10. Preparing a patchset. + + git format-patch [-o directory] + + will generate a set of patches for each commit between and + current HEAD. E.g. + + git format-patch origin/master + + will generate patches for all commits on current branch which are not + present in upstream. + A useful shortcut is also + + git format-patch -n + + which will generate patches from last n commits. + By default the patches are created in the current directory. + +11. Sending patches for review + + git send-email + + will send the patches created by git format-patch or directly generates + them. All the email fields can be configured in the global/local + configuration or overridden by command line. + Note that this tool must often be installed separately (e.g. git-email + package on Debian-based distros). + +12. Pushing changes to remote trees + + git push + + Will push the changes to the default remote (origin). + Git will prevent you from pushing changes if the local and remote trees are + out of sync. Refer to 2 and 2.a to sync the local tree. + + git remote add + + Will add additional remote with a name reference, it is useful if you want + to push your local branch for review on a remote host. + + git push + + Will push the changes to the remote repository. Omitting refspec makes git + push update all the remote branches matching the local ones. + +13. Finding a specific svn revision + + Since version 1.7.1 git supports ':/foo' syntax for specifying commits + based on a regular expression. see man gitrevisions + + git show :/'as revision 23456' + + will show the svn changeset r23456. With older git versions searching in + the git log output is the easiest option (especially if a pager with + search capabilities is used). + This commit can be checked out with + + git checkout -b svn_23456 :/'as revision 23456' + + or for git < 1.7.1 with + + git checkout -b svn_23456 $SHA1 + + where $SHA1 is the commit SHA1 from the 'git log' output. + + +Contact the project admins if you have technical +problems with the GIT server. diff --git a/doc/indevs.texi b/doc/indevs.texi index 7b94155380..8ff64a78ed 100644 --- a/doc/indevs.texi +++ b/doc/indevs.texi @@ -1,10 +1,10 @@ @chapter Input Devices @c man begin INPUT DEVICES -Input devices are configured elements in FFmpeg which allow to access +Input devices are configured elements in Libav which allow to access the data coming from a multimedia device attached to your system. -When you configure your FFmpeg build, all the supported input devices +When you configure your Libav build, all the supported input devices are enabled by default. You can list all available ones using the configure option "--list-indevs". @@ -13,7 +13,7 @@ You can disable all the input devices using the configure option option "--enable-indev=@var{INDEV}", or you can disable a particular input device using the option "--disable-indev=@var{INDEV}". -The option "-formats" of the ff* tools will display the list of +The option "-formats" of the av* tools will display the list of supported input devices (amongst the demuxers). A description of the currently available input devices follows. @@ -42,157 +42,19 @@ specify card number or identifier, device number and subdevice number To see the list of cards currently recognized by your system check the files @file{/proc/asound/cards} and @file{/proc/asound/devices}. -For example to capture with @command{ffmpeg} from an ALSA device with +For example to capture with @command{avconv} from an ALSA device with card id 0, you may run the command: @example -ffmpeg -f alsa -i hw:0 alsaout.wav +avconv -f alsa -i hw:0 alsaout.wav @end example For more information see: @url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html} -@section avfoundation - -AVFoundation input device. - -AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS. -The older QTKit framework has been marked deprecated since OSX version 10.7. - -The filename passed as input is parsed to contain either a device name or index. -The device index can also be given by using -video_device_index. -A given device index will override any given device name. -If the desired device consists of numbers only, use -video_device_index to identify it. -The default device will be chosen if an empty string or the device name "default" is given. -The available devices can be enumerated by using -list_devices. - -@example -ffmpeg -f avfoundation -i "0" out.mpg -@end example - -@example -ffmpeg -f avfoundation -video_device_index 0 -i "" out.mpg -@end example - -@example -ffmpeg -f avfoundation -i "default" out.mpg -@end example - -@example -ffmpeg -f avfoundation -list_devices true -i "" -@end example - @section bktr BSD video input device. -@section dshow - -Windows DirectShow input device. - -DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. -Currently only audio and video devices are supported. - -Multiple devices may be opened as separate inputs, but they may also be -opened on the same input, which should improve synchronism between them. - -The input name should be in the format: - -@example -@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}] -@end example - -where @var{TYPE} can be either @var{audio} or @var{video}, -and @var{NAME} is the device's name. - -@subsection Options - -If no options are specified, the device's defaults are used. -If the device does not support the requested options, it will -fail to open. - -@table @option - -@item video_size -Set the video size in the captured video. - -@item framerate -Set the frame rate in the captured video. - -@item sample_rate -Set the sample rate (in Hz) of the captured audio. - -@item sample_size -Set the sample size (in bits) of the captured audio. - -@item channels -Set the number of channels in the captured audio. - -@item list_devices -If set to @option{true}, print a list of devices and exit. - -@item list_options -If set to @option{true}, print a list of selected device's options -and exit. - -@item video_device_number -Set video device number for devices with same name (starts at 0, -defaults to 0). - -@item audio_device_number -Set audio device number for devices with same name (starts at 0, -defaults to 0). - -@item pixel_format -Select pixel format to be used by DirectShow. This may only be set when -the video codec is not set or set to rawvideo. - -@item audio_buffer_size -Set audio device buffer size in milliseconds (which can directly -impact latency, depending on the device). -Defaults to using the audio device's -default buffer size (typically some multiple of 500ms). -Setting this value too low can degrade performance. -See also -@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx} - -@end table - -@subsection Examples - -@itemize - -@item -Print the list of DirectShow supported devices and exit: -@example -$ ffmpeg -list_devices true -f dshow -i dummy -@end example - -@item -Open video device @var{Camera}: -@example -$ ffmpeg -f dshow -i video="Camera" -@end example - -@item -Open second video device with name @var{Camera}: -@example -$ ffmpeg -f dshow -video_device_number 1 -i video="Camera" -@end example - -@item -Open video device @var{Camera} and audio device @var{Microphone}: -@example -$ ffmpeg -f dshow -i video="Camera":audio="Microphone" -@end example - -@item -Print the list of supported options in selected device and exit: -@example -$ ffmpeg -list_options true -f dshow -i video="Camera" -@end example - -@end itemize - @section dv1394 Linux DV 1394 input device. @@ -210,153 +72,18 @@ For more detailed information read the file Documentation/fb/framebuffer.txt included in the Linux source tree. To record from the framebuffer device @file{/dev/fb0} with -@command{ffmpeg}: +@command{avconv}: @example -ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi +avconv -f fbdev -r 10 -i /dev/fb0 out.avi @end example You can take a single screenshot image with the command: @example -ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg +avconv -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg @end example See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1). -@section gdigrab - -Win32 GDI-based screen capture device. - -This device allows you to capture a region of the display on Windows. - -There are two options for the input filename: -@example -desktop -@end example -or -@example -title=@var{window_title} -@end example - -The first option will capture the entire desktop, or a fixed region of the -desktop. The second option will instead capture the contents of a single -window, regardless of its position on the screen. - -For example, to grab the entire desktop using @command{ffmpeg}: -@example -ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg -@end example - -Grab a 640x480 region at position @code{10,20}: -@example -ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg -@end example - -Grab the contents of the window named "Calculator" -@example -ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg -@end example - -@subsection Options - -@table @option -@item draw_mouse -Specify whether to draw the mouse pointer. Use the value @code{0} to -not draw the pointer. Default value is @code{1}. - -@item framerate -Set the grabbing frame rate. Default value is @code{ntsc}, -corresponding to a frame rate of @code{30000/1001}. - -@item show_region -Show grabbed region on screen. - -If @var{show_region} is specified with @code{1}, then the grabbing -region will be indicated on screen. With this option, it is easy to -know what is being grabbed if only a portion of the screen is grabbed. - -Note that @var{show_region} is incompatible with grabbing the contents -of a single window. - -For example: -@example -ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg -@end example - -@item video_size -Set the video frame size. The default is to capture the full screen if @file{desktop} is selected, or the full window size if @file{title=@var{window_title}} is selected. - -@item offset_x -When capturing a region with @var{video_size}, set the distance from the left edge of the screen or desktop. - -Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned to the left of your primary monitor, you will need to use a negative @var{offset_x} value to move the region to that monitor. - -@item offset_y -When capturing a region with @var{video_size}, set the distance from the top edge of the screen or desktop. - -Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned above your primary monitor, you will need to use a negative @var{offset_y} value to move the region to that monitor. - -@end table - -@section iec61883 - -FireWire DV/HDV input device using libiec61883. - -To enable this input device, you need libiec61883, libraw1394 and -libavc1394 installed on your system. Use the configure option -@code{--enable-libiec61883} to compile with the device enabled. - -The iec61883 capture device supports capturing from a video device -connected via IEEE1394 (FireWire), using libiec61883 and the new Linux -FireWire stack (juju). This is the default DV/HDV input method in Linux -Kernel 2.6.37 and later, since the old FireWire stack was removed. - -Specify the FireWire port to be used as input file, or "auto" -to choose the first port connected. - -@subsection Options - -@table @option - -@item dvtype -Override autodetection of DV/HDV. This should only be used if auto -detection does not work, or if usage of a different device type -should be prohibited. Treating a DV device as HDV (or vice versa) will -not work and result in undefined behavior. -The values @option{auto}, @option{dv} and @option{hdv} are supported. - -@item dvbuffer -Set maxiumum size of buffer for incoming data, in frames. For DV, this -is an exact value. For HDV, it is not frame exact, since HDV does -not have a fixed frame size. - -@item dvguid -Select the capture device by specifying it's GUID. Capturing will only -be performed from the specified device and fails if no device with the -given GUID is found. This is useful to select the input if multiple -devices are connected at the same time. -Look at /sys/bus/firewire/devices to find out the GUIDs. - -@end table - -@subsection Examples - -@itemize - -@item -Grab and show the input of a FireWire DV/HDV device. -@example -ffplay -f iec61883 -i auto -@end example - -@item -Grab and record the input of a FireWire DV/HDV device, -using a packet buffer of 100000 packets if the source is HDV. -@example -ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg -@end example - -@end itemize - @section jack JACK input device. @@ -368,24 +95,24 @@ A JACK input device creates one or more JACK writable clients, one for each audio channel, with name @var{client_name}:input_@var{N}, where @var{client_name} is the name provided by the application, and @var{N} is a number which identifies the channel. -Each writable client will send the acquired data to the FFmpeg input +Each writable client will send the acquired data to the Libav input device. Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients. -To connect or disconnect JACK clients you can use the @command{jack_connect} -and @command{jack_disconnect} programs, or do it through a graphical interface, -for example with @command{qjackctl}. +To connect or disconnect JACK clients you can use the +@file{jack_connect} and @file{jack_disconnect} programs, or do it +through a graphical interface, for example with @file{qjackctl}. To list the JACK clients and their properties you can invoke the command -@command{jack_lsp}. +@file{jack_lsp}. Follows an example which shows how to capture a JACK readable client -with @command{ffmpeg}. +with @command{avconv}. @example -# Create a JACK writable client with name "ffmpeg". -$ ffmpeg -f jack -i ffmpeg -y out.wav +# Create a JACK writable client with name "libav". +$ avconv -f jack -i libav -y out.wav # Start the sample jack_metro readable client. $ jack_metro -b 120 -d 0.2 -f 4000 @@ -396,181 +123,20 @@ system:capture_1 system:capture_2 system:playback_1 system:playback_2 -ffmpeg:input_1 +libav:input_1 metro:120_bpm -# Connect metro to the ffmpeg writable client. -$ jack_connect metro:120_bpm ffmpeg:input_1 +# Connect metro to the avconv writable client. +$ jack_connect metro:120_bpm libav:input_1 @end example For more information read: @url{http://jackaudio.org/} -@section lavfi - -Libavfilter input virtual device. - -This input device reads data from the open output pads of a libavfilter -filtergraph. - -For each filtergraph open output, the input device will create a -corresponding stream which is mapped to the generated output. Currently -only video data is supported. The filtergraph is specified through the -option @option{graph}. - -@subsection Options - -@table @option - -@item graph -Specify the filtergraph to use as input. Each video open output must be -labelled by a unique string of the form "out@var{N}", where @var{N} is a -number starting from 0 corresponding to the mapped input stream -generated by the device. -The first unlabelled output is automatically assigned to the "out0" -label, but all the others need to be specified explicitly. - -If not specified defaults to the filename specified for the input -device. - -@item graph_file -Set the filename of the filtergraph to be read and sent to the other -filters. Syntax of the filtergraph is the same as the one specified by -the option @var{graph}. - -@end table - -@subsection Examples - -@itemize -@item -Create a color video stream and play it back with @command{ffplay}: -@example -ffplay -f lavfi -graph "color=c=pink [out0]" dummy -@end example - -@item -As the previous example, but use filename for specifying the graph -description, and omit the "out0" label: -@example -ffplay -f lavfi color=c=pink -@end example - -@item -Create three different video test filtered sources and play them: -@example -ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3 -@end example - -@item -Read an audio stream from a file using the amovie source and play it -back with @command{ffplay}: -@example -ffplay -f lavfi "amovie=test.wav" -@end example - -@item -Read an audio stream and a video stream and play it back with -@command{ffplay}: -@example -ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]" -@end example - -@end itemize - @section libdc1394 IIDC1394 input device, based on libdc1394 and libraw1394. -@section openal - -The OpenAL input device provides audio capture on all systems with a -working OpenAL 1.1 implementation. - -To enable this input device during configuration, you need OpenAL -headers and libraries installed on your system, and need to configure -FFmpeg with @code{--enable-openal}. - -OpenAL headers and libraries should be provided as part of your OpenAL -implementation, or as an additional download (an SDK). Depending on your -installation you may need to specify additional flags via the -@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build -system to locate the OpenAL headers and libraries. - -An incomplete list of OpenAL implementations follows: - -@table @strong -@item Creative -The official Windows implementation, providing hardware acceleration -with supported devices and software fallback. -See @url{http://openal.org/}. -@item OpenAL Soft -Portable, open source (LGPL) software implementation. Includes -backends for the most common sound APIs on the Windows, Linux, -Solaris, and BSD operating systems. -See @url{http://kcat.strangesoft.net/openal.html}. -@item Apple -OpenAL is part of Core Audio, the official Mac OS X Audio interface. -See @url{http://developer.apple.com/technologies/mac/audio-and-video.html} -@end table - -This device allows one to capture from an audio input device handled -through OpenAL. - -You need to specify the name of the device to capture in the provided -filename. If the empty string is provided, the device will -automatically select the default device. You can get the list of the -supported devices by using the option @var{list_devices}. - -@subsection Options - -@table @option - -@item channels -Set the number of channels in the captured audio. Only the values -@option{1} (monaural) and @option{2} (stereo) are currently supported. -Defaults to @option{2}. - -@item sample_size -Set the sample size (in bits) of the captured audio. Only the values -@option{8} and @option{16} are currently supported. Defaults to -@option{16}. - -@item sample_rate -Set the sample rate (in Hz) of the captured audio. -Defaults to @option{44.1k}. - -@item list_devices -If set to @option{true}, print a list of devices and exit. -Defaults to @option{false}. - -@end table - -@subsection Examples - -Print the list of OpenAL supported devices and exit: -@example -$ ffmpeg -list_devices true -f openal -i dummy out.ogg -@end example - -Capture from the OpenAL device @file{DR-BT101 via PulseAudio}: -@example -$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg -@end example - -Capture from the default device (note the empty string '' as filename): -@example -$ ffmpeg -f openal -i '' out.ogg -@end example - -Capture from two devices simultaneously, writing to two different files, -within the same @command{ffmpeg} command: -@example -$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg -@end example -Note: not all OpenAL implementations support multiple simultaneous capture - -try the latest OpenAL Soft if the above does not work. - @section oss Open Sound System input device. @@ -579,10 +145,10 @@ The filename to provide to the input device is the device node representing the OSS input device, and is usually set to @file{/dev/dsp}. -For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the +For example to grab from @file{/dev/dsp} using @command{avconv} use the command: @example -ffmpeg -f oss -i /dev/dsp /tmp/oss.wav +avconv -f oss -i /dev/dsp /tmp/oss.wav @end example For more information about OSS see: @@ -590,78 +156,86 @@ For more information about OSS see: @section pulse -PulseAudio input device. +pulseaudio input device. -To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}. +To enable this input device during configuration you need libpulse-simple +installed in your system. The filename to provide to the input device is a source device or the string "default" -To list the PulseAudio source devices and their properties you can invoke -the command @command{pactl list sources}. +To list the pulse source devices and their properties you can invoke +the command @file{pactl list sources}. -More information about PulseAudio can be found on @url{http://www.pulseaudio.org}. +@example +avconv -f pulse -i default /tmp/pulse.wav +@end example -@subsection Options -@table @option -@item server -Connect to a specific PulseAudio server, specified by an IP address. -Default server is used when not provided. +@subsection @var{server} AVOption -@item name -Specify the application name PulseAudio will use when showing active clients, -by default it is the @code{LIBAVFORMAT_IDENT} string. +The syntax is: +@example +-server @var{server name} +@end example -@item stream_name -Specify the stream name PulseAudio will use when showing active streams, -by default it is "record". +Connects to a specific server. + +@subsection @var{name} AVOption + +The syntax is: +@example +-name @var{application name} +@end example + +Specify the application name pulse will use when showing active clients, +by default it is "libav" + +@subsection @var{stream_name} AVOption + +The syntax is: +@example +-stream_name @var{stream name} +@end example + +Specify the stream name pulse will use when showing active streams, +by default it is "record" + +@subsection @var{sample_rate} AVOption + +The syntax is: +@example +-sample_rate @var{samplerate} +@end example -@item sample_rate Specify the samplerate in Hz, by default 48kHz is used. -@item channels +@subsection @var{channels} AVOption + +The syntax is: +@example +-channels @var{N} +@end example + Specify the channels in use, by default 2 (stereo) is set. -@item frame_size -Specify the number of bytes per frame, by default it is set to 1024. +@subsection @var{frame_size} AVOption -@item fragment_size -Specify the minimal buffering fragment in PulseAudio, it will affect the +The syntax is: +@example +-frame_size @var{bytes} +@end example + +Specify the number of byte per frame, by default it is set to 1024. + +@subsection @var{fragment_size} AVOption + +The syntax is: +@example +-fragment_size @var{bytes} +@end example + +Specify the minimal buffering fragment in pulseaudio, it will affect the audio latency. By default it is unset. -@end table - -@subsection Examples -Record a stream from default device: -@example -ffmpeg -f pulse -i default /tmp/pulse.wav -@end example - -@section qtkit - -QTKit input device. - -The filename passed as input is parsed to contain either a device name or index. -The device index can also be given by using -video_device_index. -A given device index will override any given device name. -If the desired device consists of numbers only, use -video_device_index to identify it. -The default device will be chosen if an empty string or the device name "default" is given. -The available devices can be enumerated by using -list_devices. - -@example -ffmpeg -f qtkit -i "0" out.mpg -@end example - -@example -ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg -@end example - -@example -ffmpeg -f qtkit -i "default" out.mpg -@end example - -@example -ffmpeg -f qtkit -list_devices true -i "" -@end example @section sndio @@ -674,22 +248,16 @@ The filename to provide to the input device is the device node representing the sndio input device, and is usually set to @file{/dev/audio0}. -For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the +For example to grab from @file{/dev/audio0} using @command{avconv} use the command: @example -ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav +avconv -f sndio -i /dev/audio0 /tmp/oss.wav @end example -@section video4linux2, v4l2 +@section video4linux2 Video4Linux2 input video device. -"v4l2" can be used as alias for "video4linux2". - -If FFmpeg is built with v4l-utils support (by using the -@code{--enable-libv4l2} configure option), it is possible to use it with the -@code{-use_libv4l2} input device option. - The name of the device to grab is a file device node, usually Linux systems tend to automatically create such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the @@ -697,107 +265,20 @@ kind @file{/dev/video@var{N}}, where @var{N} is a number associated to the device. Video4Linux2 devices usually support a limited set of -@var{width}x@var{height} sizes and frame rates. You can check which are +@var{width}x@var{height} sizes and framerates. You can check which are supported using @command{-list_formats all} for Video4Linux2 devices. -Some devices, like TV cards, support one or more standards. It is possible -to list all the supported standards using @command{-list_standards all}. -The time base for the timestamps is 1 microsecond. Depending on the kernel -version and configuration, the timestamps may be derived from the real time -clock (origin at the Unix Epoch) or the monotonic clock (origin usually at -boot time, unaffected by NTP or manual changes to the clock). The -@option{-timestamps abs} or @option{-ts abs} option can be used to force -conversion into the real time clock. +Some usage examples of the video4linux2 devices with avconv and avplay: -Some usage examples of the video4linux2 device with @command{ffmpeg} -and @command{ffplay}: -@itemize -@item -Grab and show the input of a video4linux2 device: @example -ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0 +# Grab and show the input of a video4linux2 device. +avplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0 + +# Grab and record the input of a video4linux2 device, leave the +framerate and size as previously set. +avconv -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg @end example -@item -Grab and record the input of a video4linux2 device, leave the -frame rate and size as previously set: -@example -ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg -@end example -@end itemize - -For more information about Video4Linux, check @url{http://linuxtv.org/}. - -@subsection Options - -@table @option -@item standard -Set the standard. Must be the name of a supported standard. To get a -list of the supported standards, use the @option{list_standards} -option. - -@item channel -Set the input channel number. Default to -1, which means using the -previously selected channel. - -@item video_size -Set the video frame size. The argument must be a string in the form -@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation. - -@item pixel_format -Select the pixel format (only valid for raw video input). - -@item input_format -Set the preferred pixel format (for raw video) or a codec name. -This option allows one to select the input format, when several are -available. - -@item framerate -Set the preferred video frame rate. - -@item list_formats -List available formats (supported pixel formats, codecs, and frame -sizes) and exit. - -Available values are: -@table @samp -@item all -Show all available (compressed and non-compressed) formats. - -@item raw -Show only raw video (non-compressed) formats. - -@item compressed -Show only compressed formats. -@end table - -@item list_standards -List supported standards and exit. - -Available values are: -@table @samp -@item all -Show all supported standards. -@end table - -@item timestamps, ts -Set type of timestamps for grabbed frames. - -Available values are: -@table @samp -@item default -Use timestamps from the kernel. - -@item abs -Use absolute timestamps (wall clock). - -@item mono2abs -Force conversion from monotonic to absolute timestamps. -@end table - -Default value is @code{default}. -@end table - @section vfwcap VfW (Video for Windows) capture input device. @@ -810,7 +291,7 @@ other filename will be interpreted as device number 0. X11 video input device. -This device allows one to capture a region of an X11 display. +This device allows to capture a region of an X11 display. The filename passed as input has the syntax: @example @@ -828,30 +309,24 @@ default to 0. Check the X11 documentation (e.g. man X) for more detailed information. -Use the @command{dpyinfo} program for getting basic information about the +Use the @file{dpyinfo} program for getting basic information about the properties of your X11 display (e.g. grep for "name" or "dimensions"). -For example to grab from @file{:0.0} using @command{ffmpeg}: +For example to grab from @file{:0.0} using @command{avconv}: @example -ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg +avconv -f x11grab -r 25 -s cif -i :0.0 out.mpg + +# Grab at position 10,20. +avconv -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg @end example -Grab at position @code{10,20}: +@subsection @var{follow_mouse} AVOption + +The syntax is: @example -ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg +-follow_mouse centered|@var{PIXELS} @end example -@subsection Options - -@table @option -@item draw_mouse -Specify whether to draw the mouse pointer. A value of @code{0} specify -not to draw the pointer. Default value is @code{1}. - -@item follow_mouse -Make the grabbed area follow the mouse. The argument can be -@code{centered} or a number of pixels @var{PIXELS}. - When it is specified with "centered", the grabbing region follows the mouse pointer and keeps the pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches within @var{PIXELS} (greater than @@ -859,37 +334,29 @@ zero) to the edge of region. For example: @example -ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg +avconv -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg + +# Follows only when the mouse pointer reaches within 100 pixels to edge +avconv -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg @end example -To follow only when the mouse pointer reaches within 100 pixels to edge: +@subsection @var{show_region} AVOption + +The syntax is: @example -ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg +-show_region 1 @end example -@item framerate -Set the grabbing frame rate. Default value is @code{ntsc}, -corresponding to a frame rate of @code{30000/1001}. - -@item show_region -Show grabbed region on screen. - -If @var{show_region} is specified with @code{1}, then the grabbing -region will be indicated on screen. With this option, it is easy to -know what is being grabbed if only a portion of the screen is grabbed. +If @var{show_region} AVOption is specified with @var{1}, then the grabbing +region will be indicated on screen. With this option, it's easy to know what is +being grabbed if only a portion of the screen is grabbed. For example: @example -ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg -@end example +avconv -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg -With @var{follow_mouse}: -@example -ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg +# With follow_mouse +avconv -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg @end example -@item video_size -Set the video frame size. Default value is @code{vga}. -@end table - @c man end INPUT DEVICES diff --git a/doc/issue_tracker.txt b/doc/issue_tracker.txt deleted file mode 100644 index 33b3535dcd..0000000000 --- a/doc/issue_tracker.txt +++ /dev/null @@ -1,194 +0,0 @@ -FFmpeg's bug/feature request tracker manual -================================================= - -NOTE: This is a draft. - -Overview: ---------- - -FFmpeg uses Trac for tracking issues, new issues and changes to -existing issues can be done through a web interface. - -Issues can be different kinds of things we want to keep track of -but that do not belong into the source tree itself. This includes -bug reports, feature requests and license violations. We -might add more items to this list in the future, so feel free to -propose a new `type of issue' on the ffmpeg-devel mailing list if -you feel it is worth tracking. - -It is possible to subscribe to individual issues by adding yourself to the -Cc list or to subscribe to the ffmpeg-trac mailing list which receives -a mail for every change to every issue. -(the above does all work already after light testing) - -The subscription URL for the ffmpeg-trac list is: -http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac -The URL of the webinterface of the tracker is: -http(s)://trac.ffmpeg.org - -Type: ------ -art - Artwork such as photos, music, banners, and logos. - -bug / defect - An error, flaw, mistake, failure, or fault in FFmpeg or libav* that - prevents it from behaving as intended. - -feature request / enhancement - Request of support for encoding or decoding of a new codec, container - or variant. - Request of support for more, less or plain different output or behavior - where the current implementation cannot be considered wrong. - -license violation - ticket to keep track of (L)GPL violations of ffmpeg by others - -sponsoring request - Developer requests for hardware, software, specifications, money, - refunds, etc. - -Priority: ---------- -critical - Bugs about data loss and security issues. - No feature request can be critical. - -important - Bugs which make FFmpeg unusable for a significant number of users. - Examples here might be completely broken MPEG-4 decoding or a build issue - on Linux. - While broken 4xm decoding or a broken OS/2 build would not be important, - the separation to normal is somewhat fuzzy. - For feature requests this priority would be used for things many people - want. - Regressions also should be marked as important, regressions are bugs that - don't exist in a past revision or another branch. - -normal - - -minor - Bugs about things like spelling errors, "mp2" instead of - "mp3" being shown and such. - Feature requests about things few people want or which do not make a big - difference. - -wish - Something that is desirable to have but that there is no urgency at - all to implement, e.g. something completely cosmetic like a website - restyle or a personalized doxy template or the FFmpeg logo. - This priority is not valid for bugs. - - -Status: -------- -new - initial state - -open - intermediate states - -closed - final state - - -Analyzed flag: --------------- -Bugs which have been analyzed and where it is understood what causes them -and which exact chain of events triggers them. This analysis should be -available as a message in the bug report. -Note, do not change the status to analyzed without also providing a clear -and understandable analysis. -This state implicates that the bug either has been reproduced or that -reproduction is not needed as the bug is already understood. - - -Type/Status: ----------- -*/new - Initial state of new bugs and feature requests submitted by - users. - -*/open - Issues which have been briefly looked at and which did not look outright - invalid. - This implicates that no real more detailed state applies yet. Conversely, - the more detailed states below implicate that the issue has been briefly - looked at. - -*/closed/duplicate - Bugs or feature requests which are duplicates. - Note, if you mark something as duplicate, do not forget setting the - superseder so bug reports are properly linked. - -*/closed/invalid - Bugs caused by user errors, random ineligible or otherwise nonsense stuff. - -*/closed/needs_more_info - Issues for which some information has been requested by the developers, - but which has not been provided by anyone within reasonable time. - - -bug/closed/fixed - Bugs which have to the best of our knowledge been fixed. - -bug/closed/wontfix - Bugs which we will not fix. Possible reasons include legality, high - complexity for the sake of supporting obscure corner cases, speed loss - for similarly esoteric purposes, et cetera. - This also means that we would reject a patch. - If we are just too lazy to fix a bug then the correct state is open - and unassigned. Closed means that the case is closed which is not - the case if we are just waiting for a patch. - -bug/closed/works_for_me - Bugs for which sufficient information was provided to reproduce but - reproduction failed - that is the code seems to work correctly to the - best of our knowledge. - -feature_request/closed/fixed - Feature requests which have been implemented. - -feature_request/closed/wontfix - Feature requests which will not be implemented. The reasons here could - be legal, philosophical or others. - -Note2, if you provide the requested info do not forget to remove the -needs_more_info resolution. - -Component: ----------- - -avcodec - issues in libavcodec/* - -avformat - issues in libavformat/* - -avutil - issues in libavutil/* - -regression test - issues in tests/* - -ffmpeg - issues in or related to ffmpeg.c - -ffplay - issues in or related to ffplay.c - -ffprobe - issues in or related to ffprobe.c - -ffserver - issues in or related to ffserver.c - -build system - issues in or related to configure/Makefile - -regression - bugs which were not present in a past revision - -trac - issues related to our issue tracker diff --git a/doc/libavcodec.texi b/doc/libavcodec.texi deleted file mode 100644 index 618f9f6b74..0000000000 --- a/doc/libavcodec.texi +++ /dev/null @@ -1,48 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle Libavcodec Documentation -@titlepage -@center @titlefont{Libavcodec Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The libavcodec library provides a generic encoding/decoding framework -and contains multiple decoders and encoders for audio, video and -subtitle streams, and several bitstream filters. - -The shared architecture provides various services ranging from bit -stream I/O to DSP optimizations, and makes it suitable for -implementing robust and fast codecs as well as for experimentation. - -@c man end DESCRIPTION - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters}, -@url{libavutil.html,libavutil} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), -libavutil(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename libavcodec -@settitle media streams decoding and encoding library - -@end ignore - -@bye diff --git a/doc/libavdevice.texi b/doc/libavdevice.texi deleted file mode 100644 index d5f790b675..0000000000 --- a/doc/libavdevice.texi +++ /dev/null @@ -1,45 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle Libavdevice Documentation -@titlepage -@center @titlefont{Libavdevice Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The libavdevice library provides a generic framework for grabbing from -and rendering to many common multimedia input/output devices, and -supports several input and output devices, including Video4Linux2, -VfW, DShow, and ALSA. - -@c man end DESCRIPTION - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-devices.html,ffmpeg-devices}, -@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-devices(1), -libavutil(3), libavcodec(3), libavformat(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename libavdevice -@settitle multimedia device handling library - -@end ignore - -@bye diff --git a/doc/libavfilter.texi b/doc/libavfilter.texi index 4f82944d36..84bad29dab 100644 --- a/doc/libavfilter.texi +++ b/doc/libavfilter.texi @@ -9,36 +9,84 @@ @contents -@chapter Description -@c man begin DESCRIPTION +@chapter Introduction -The libavfilter library provides a generic audio/video filtering -framework containing several filters, sources and sinks. +Libavfilter is the filtering API of Libav. It replaces 'vhooks', and +started as a Google Summer of Code project. -@c man end DESCRIPTION +Note that there may still be serious bugs in the code and its API +and ABI should not be considered stable yet! -@chapter See Also +@chapter Tutorial -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-filters.html,ffmpeg-filters}, -@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample}, -@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice} -@end ifhtml +In libavfilter, it is possible for filters to have multiple inputs and +multiple outputs. +To illustrate the sorts of things that are possible, we can +use a complex filter graph. For example, the following one: -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-filters(1), -libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3) -@end ifnothtml +@example +input --> split --> fifo -----------------------> overlay --> output + | ^ + | | + +------> fifo --> crop --> vflip --------+ +@end example -@include authors.texi +splits the stream in two streams, then sends one stream through the crop filter +and the vflip filter, before merging it back with the other stream by +overlaying it on top. You can use the following command to achieve this: -@ignore +@example +./avconv -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output +@end example -@setfilename libavfilter -@settitle multimedia filtering library +The result will be that the top half of the video is mirrored +onto the bottom half of the output video. -@end ignore +Video filters are loaded using the @var{-vf} option passed to +avconv or to avplay. Filters in the same linear chain are separated by +commas. In our example, @var{split}, @var{fifo}, and @var{overlay} are in one +linear chain, and @var{fifo}, @var{crop}, and @var{vflip} are in another. The +points where the linear chains join are labeled by names enclosed in square +brackets. In our example, they join at @var{[T1]} and @var{[T2]}. The magic +labels @var{[in]} and @var{[out]} are the points where video is input +and output. + +Some filters take a list of parameters: they are specified +after the filter name and an equal sign, and are separated +by a semicolon. + +There are so-called @var{source filters} that do not take video +input, and we expect that some @var{sink filters} will +not have video output, at some point in the future. + +@chapter graph2dot + +The @file{graph2dot} program included in the Libav @file{tools} +directory can be used to parse a filter graph description and issue a +corresponding textual representation in the dot language. + +Invoke the command: +@example +graph2dot -h +@end example + +to see how to use @file{graph2dot}. + +You can then pass the dot description to the @file{dot} program (from +the graphviz suite of programs) and obtain a graphical representation +of the filter graph. + +For example the sequence of commands: +@example +echo @var{GRAPH_DESCRIPTION} | \ +tools/graph2dot -o graph.tmp && \ +dot -Tpng graph.tmp -o graph.png && \ +display graph.png +@end example + +can be used to create and display an image representing the graph +described by the @var{GRAPH_DESCRIPTION} string. + +@include filters.texi @bye diff --git a/doc/libavformat.texi b/doc/libavformat.texi deleted file mode 100644 index 85e49cb952..0000000000 --- a/doc/libavformat.texi +++ /dev/null @@ -1,48 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle Libavformat Documentation -@titlepage -@center @titlefont{Libavformat Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The libavformat library provides a generic framework for multiplexing -and demultiplexing (muxing and demuxing) audio, video and subtitle -streams. It encompasses multiple muxers and demuxers for multimedia -container formats. - -It also supports several input and output protocols to access a media -resource. - -@c man end DESCRIPTION - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols}, -@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-formats(1), ffmpeg-protocols(1), -libavutil(3), libavcodec(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename libavformat -@settitle multimedia muxing and demuxing library - -@end ignore - -@bye diff --git a/doc/libavutil.texi b/doc/libavutil.texi deleted file mode 100644 index 5ec7e84dfc..0000000000 --- a/doc/libavutil.texi +++ /dev/null @@ -1,62 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle Libavutil Documentation -@titlepage -@center @titlefont{Libavutil Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The libavutil library is a utility library to aid portable -multimedia programming. It contains safe portable string functions, -random number generators, data structures, additional mathematics -functions, cryptography and multimedia related functionality (like -enumerations for pixel and sample formats). It is not a library for -code needed by both libavcodec and libavformat. - -The goals for this library is to be: - -@table @strong -@item Modular -It should have few interdependencies and the possibility of disabling individual -parts during @command{./configure}. - -@item Small -Both sources and objects should be small. - -@item Efficient -It should have low CPU and memory usage. - -@item Useful -It should avoid useless features that almost no one needs. -@end table - -@c man end DESCRIPTION - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-utils.html,ffmpeg-utils} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-utils(1) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename libavutil -@settitle multimedia-biased utility library - -@end ignore - -@bye diff --git a/doc/libswresample.texi b/doc/libswresample.texi deleted file mode 100644 index 383e53779a..0000000000 --- a/doc/libswresample.texi +++ /dev/null @@ -1,70 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle Libswresample Documentation -@titlepage -@center @titlefont{Libswresample Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The libswresample library performs highly optimized audio resampling, -rematrixing and sample format conversion operations. - -Specifically, this library performs the following conversions: - -@itemize -@item -@emph{Resampling}: is the process of changing the audio rate, for -example from a high sample rate of 44100Hz to 8000Hz. Audio -conversion from high to low sample rate is a lossy process. Several -resampling options and algorithms are available. - -@item -@emph{Format conversion}: is the process of converting the type of -samples, for example from 16-bit signed samples to unsigned 8-bit or -float samples. It also handles packing conversion, when passing from -packed layout (all samples belonging to distinct channels interleaved -in the same buffer), to planar layout (all samples belonging to the -same channel stored in a dedicated buffer or "plane"). - -@item -@emph{Rematrixing}: is the process of changing the channel layout, for -example from stereo to mono. When the input channels cannot be mapped -to the output streams, the process is lossy, since it involves -different gain factors and mixing. -@end itemize - -Various other audio conversions (e.g. stretching and padding) are -enabled through dedicated options. - -@c man end DESCRIPTION - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-resampler.html,ffmpeg-resampler}, -@url{libavutil.html,libavutil} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-resampler(1), -libavutil(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename libswresample -@settitle audio resampling library - -@end ignore - -@bye diff --git a/doc/libswscale.texi b/doc/libswscale.texi deleted file mode 100644 index 818e98880b..0000000000 --- a/doc/libswscale.texi +++ /dev/null @@ -1,63 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle Libswscale Documentation -@titlepage -@center @titlefont{Libswscale Documentation} -@end titlepage - -@top - -@contents - -@chapter Description -@c man begin DESCRIPTION - -The libswscale library performs highly optimized image scaling and -colorspace and pixel format conversion operations. - -Specifically, this library performs the following conversions: - -@itemize -@item -@emph{Rescaling}: is the process of changing the video size. Several -rescaling options and algorithms are available. This is usually a -lossy process. - -@item -@emph{Pixel format conversion}: is the process of converting the image -format and colorspace of the image, for example from planar YUV420P to -RGB24 packed. It also handles packing conversion, that is converts -from packed layout (all pixels belonging to distinct planes -interleaved in the same buffer), to planar layout (all samples -belonging to the same plane stored in a dedicated buffer or "plane"). - -This is usually a lossy process in case the source and destination -colorspaces differ. -@end itemize - -@c man end DESCRIPTION - -@chapter See Also - -@ifhtml -@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, -@url{ffmpeg-scaler.html,ffmpeg-scaler}, -@url{libavutil.html,libavutil} -@end ifhtml - -@ifnothtml -ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), -ffmpeg-scaler(1), -libavutil(3) -@end ifnothtml - -@include authors.texi - -@ignore - -@setfilename libswscale -@settitle video scaling and pixel format conversion library - -@end ignore - -@bye diff --git a/doc/metadata.texi b/doc/metadata.texi index b7fc789e2a..cfaf491c2d 100644 --- a/doc/metadata.texi +++ b/doc/metadata.texi @@ -1,7 +1,7 @@ @chapter Metadata @c man begin METADATA -FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded +Libav is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then load it back using the metadata muxer/demuxer. The file format is as follows: @@ -53,7 +53,7 @@ A ffmetadata file might look like this: ;FFMETADATA1 title=bike\\shed ;this is a comment -artist=FFmpeg troll team +artist=Libav troll team [CHAPTER] TIMEBASE=1/1000 @@ -65,20 +65,4 @@ title=chapter \#1 title=multi\ line @end example - -By using the ffmetadata muxer and demuxer it is possible to extract -metadata from an input file to an ffmetadata file, and then transcode -the file into an output file with the edited ffmetadata file. - -Extracting an ffmetadata file with @file{ffmpeg} goes as follows: -@example -ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE -@end example - -Reinserting edited metadata information from the FFMETADATAFILE file can -be done as: -@example -ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT -@end example - @c man end METADATA diff --git a/doc/mips.txt b/doc/mips.txt deleted file mode 100644 index 8c6779f67d..0000000000 --- a/doc/mips.txt +++ /dev/null @@ -1,75 +0,0 @@ -MIPS optimizations info -=============================================== - -MIPS optimizations of codecs are targeting MIPS 74k family of -CPUs. Some of these optimizations are relying more on properties of -this architecture and some are relying less (and can be used on most -MIPS architectures without degradation in performance). - -Along with FFMPEG copyright notice, there is MIPS copyright notice in -all the files that are created by people from MIPS Technologies. - -Example of copyright notice: -=============================================== -/* - * Copyright (c) 2012 - * MIPS Technologies, Inc., California. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions - * are met: - * 1. Redistributions of source code must retain the above copyright - * notice, this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright - * notice, this list of conditions and the following disclaimer in the - * documentation and/or other materials provided with the distribution. - * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its - * contributors may be used to endorse or promote products derived from - * this software without specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND - * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE - * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE - * ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE - * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL - * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS - * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) - * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT - * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY - * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF - * SUCH DAMAGE. - * - * Author: Author Name (author_name@@mips.com) - */ - -Files that have MIPS copyright notice in them: -=============================================== -* libavutil/mips/ - float_dsp_mips.c - libm_mips.h -* libavcodec/ - fft_fixed_32.c - fft_init_table.c - fft_table.h - mdct_fixed_32.c -* libavcodec/mips/ - aaccoder_mips.c - aacpsy_mips.h - ac3dsp_mips.c - acelp_filters_mips.c - acelp_vectors_mips.c - amrwbdec_mips.c - amrwbdec_mips.h - celp_filters_mips.c - celp_math_mips.c - compute_antialias_fixed.h - compute_antialias_float.h - lsp_mips.h - dsputil_mips.c - fft_mips.c - fft_table.h - fft_init_table.c - fmtconvert_mips.c - iirfilter_mips.c - mpegaudiodsp_mips_fixed.c - mpegaudiodsp_mips_float.c diff --git a/doc/multithreading.txt b/doc/multithreading.txt index 2b992fcbc5..9b27b108c6 100644 --- a/doc/multithreading.txt +++ b/doc/multithreading.txt @@ -1,7 +1,7 @@ -FFmpeg multithreading methods +Libav multithreading methods ============================================== -FFmpeg provides two methods for multithreading codecs. +Libav provides two methods for multithreading codecs. Slice threading decodes multiple parts of a frame at the same time, using AVCodecContext execute() and execute2(). diff --git a/doc/muxers.texi b/doc/muxers.texi index b9e5d4bf6f..e1c0383eac 100644 --- a/doc/muxers.texi +++ b/doc/muxers.texi @@ -1,10 +1,10 @@ @chapter Muxers @c man begin MUXERS -Muxers are configured elements in FFmpeg which allow writing +Muxers are configured elements in Libav which allow writing multimedia streams to a particular type of file. -When you configure your FFmpeg build, all the supported muxers +When you configure your Libav build, all the supported muxers are enabled by default. You can list all available muxers using the configure option @code{--list-muxers}. @@ -13,30 +13,11 @@ You can disable all the muxers with the configure option with the options @code{--enable-muxer=@var{MUXER}} / @code{--disable-muxer=@var{MUXER}}. -The option @code{-formats} of the ff* tools will display the list of +The option @code{-formats} of the av* tools will display the list of enabled muxers. A description of some of the currently available muxers follows. -@anchor{aiff} -@section aiff - -Audio Interchange File Format muxer. - -@subsection Options - -It accepts the following options: - -@table @option -@item write_id3v2 -Enable ID3v2 tags writing when set to 1. Default is 0 (disabled). - -@item id3v2_version -Select ID3v2 version to write. Currently only version 3 and 4 (aka. -ID3v2.3 and ID3v2.4) are supported. The default is version 4. - -@end table - @anchor{crc} @section crc @@ -51,231 +32,93 @@ The output of the muxer consists of a single line of the form: CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames. -See also the @ref{framecrc} muxer. - -@subsection Examples - For example to compute the CRC of the input, and store it in the file @file{out.crc}: @example -ffmpeg -i INPUT -f crc out.crc +avconv -i INPUT -f crc out.crc @end example You can print the CRC to stdout with the command: @example -ffmpeg -i INPUT -f crc - +avconv -i INPUT -f crc - @end example -You can select the output format of each frame with @command{ffmpeg} by +You can select the output format of each frame with @command{avconv} by specifying the audio and video codec and format. For example to compute the CRC of the input audio converted to PCM unsigned 8-bit and the input video converted to MPEG-2 video, use the command: @example -ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - +avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - @end example +See also the @ref{framecrc} muxer. + @anchor{framecrc} @section framecrc -Per-packet CRC (Cyclic Redundancy Check) testing format. +Per-frame CRC (Cyclic Redundancy Check) testing format. -This muxer computes and prints the Adler-32 CRC for each audio -and video packet. By default audio frames are converted to signed +This muxer computes and prints the Adler-32 CRC for each decoded audio +and video frame. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC. The output of the muxer consists of a line for each audio and video -packet of the form: +frame of the form: @var{stream_index}, @var{frame_dts}, +@var{frame_size}, 0x@var{CRC}, where @var{CRC} is a hexadecimal +number 0-padded to 8 digits containing the CRC of the decoded frame. + +For example to compute the CRC of each decoded frame in the input, and +store it in the file @file{out.crc}: @example -@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC} +avconv -i INPUT -f framecrc out.crc @end example -@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the -CRC of the packet. - -@subsection Examples - -For example to compute the CRC of the audio and video frames in -@file{INPUT}, converted to raw audio and video packets, and store it -in the file @file{out.crc}: +You can print the CRC of each decoded frame to stdout with the command: @example -ffmpeg -i INPUT -f framecrc out.crc +avconv -i INPUT -f framecrc - @end example -To print the information to stdout, use the command: -@example -ffmpeg -i INPUT -f framecrc - -@end example - -With @command{ffmpeg}, you can select the output format to which the -audio and video frames are encoded before computing the CRC for each -packet by specifying the audio and video codec. For example, to +You can select the output format of each frame with @command{avconv} by +specifying the audio and video codec and format. For example, to compute the CRC of each decoded input audio frame converted to PCM unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the command: @example -ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - +avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - @end example See also the @ref{crc} muxer. -@anchor{framemd5} -@section framemd5 - -Per-packet MD5 testing format. - -This muxer computes and prints the MD5 hash for each audio -and video packet. By default audio frames are converted to signed -16-bit raw audio and video frames to raw video before computing the -hash. - -The output of the muxer consists of a line for each audio and video -packet of the form: -@example -@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5} -@end example - -@var{MD5} is a hexadecimal number representing the computed MD5 hash -for the packet. - -@subsection Examples - -For example to compute the MD5 of the audio and video frames in -@file{INPUT}, converted to raw audio and video packets, and store it -in the file @file{out.md5}: -@example -ffmpeg -i INPUT -f framemd5 out.md5 -@end example - -To print the information to stdout, use the command: -@example -ffmpeg -i INPUT -f framemd5 - -@end example - -See also the @ref{md5} muxer. - -@anchor{gif} -@section gif - -Animated GIF muxer. - -It accepts the following options: - -@table @option -@item loop -Set the number of times to loop the output. Use @code{-1} for no loop, @code{0} -for looping indefinitely (default). - -@item final_delay -Force the delay (expressed in centiseconds) after the last frame. Each frame -ends with a delay until the next frame. The default is @code{-1}, which is a -special value to tell the muxer to re-use the previous delay. In case of a -loop, you might want to customize this value to mark a pause for instance. -@end table - -For example, to encode a gif looping 10 times, with a 5 seconds delay between -the loops: -@example -ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif -@end example - -Note 1: if you wish to extract the frames in separate GIF files, you need to -force the @ref{image2} muxer: -@example -ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif" -@end example - -Note 2: the GIF format has a very small time base: the delay between two frames -can not be smaller than one centi second. - @anchor{hls} @section hls Apple HTTP Live Streaming muxer that segments MPEG-TS according to -the HTTP Live Streaming (HLS) specification. +the HTTP Live Streaming specification. It creates a playlist file and numbered segment files. The output filename specifies the playlist filename; the segment filenames receive the same basename as the playlist, a sequential number and a .ts extension. -For example, to convert an input file with @command{ffmpeg}: @example -ffmpeg -i in.nut out.m3u8 +avconv -i in.nut out.m3u8 @end example -See also the @ref{segment} muxer, which provides a more generic and -flexible implementation of a segmenter, and can be used to perform HLS -segmentation. - -@subsection Options - -This muxer supports the following options: - @table @option -@item hls_time @var{seconds} -Set the segment length in seconds. Default value is 2. - -@item hls_list_size @var{size} -Set the maximum number of playlist entries. If set to 0 the list file -will contain all the segments. Default value is 5. - -@item hls_wrap @var{wrap} -Set the number after which the segment filename number (the number -specified in each segment file) wraps. If set to 0 the number will be -never wrapped. Default value is 0. - -This option is useful to avoid to fill the disk with many segment -files, and limits the maximum number of segment files written to disk -to @var{wrap}. - -@item start_number @var{number} -Start the playlist sequence number from @var{number}. Default value is -0. - -@item hls_base_url @var{baseurl} +@item -hls_time @var{seconds} +Set the segment length in seconds. +@item -hls_list_size @var{size} +Set the maximum number of playlist entries. +@item -hls_wrap @var{wrap} +Set the number after which index wraps. +@item -start_number @var{number} +Start the sequence from @var{number}. +@item -hls_base_url @var{baseurl} Append @var{baseurl} to every entry in the playlist. Useful to generate playlists with absolute paths. - -Note that the playlist sequence number must be unique for each segment -and it is not to be confused with the segment filename sequence number -which can be cyclic, for example if the @option{wrap} option is -specified. @end table -@anchor{ico} -@section ico - -ICO file muxer. - -Microsoft's icon file format (ICO) has some strict limitations that should be noted: - -@itemize -@item -Size cannot exceed 256 pixels in any dimension - -@item -Only BMP and PNG images can be stored - -@item -If a BMP image is used, it must be one of the following pixel formats: -@example -BMP Bit Depth FFmpeg Pixel Format -1bit pal8 -4bit pal8 -8bit pal8 -16bit rgb555le -24bit bgr24 -32bit bgra -@end example - -@item -If a BMP image is used, it must use the BITMAPINFOHEADER DIB header - -@item -If a PNG image is used, it must use the rgba pixel format -@end itemize - @anchor{image2} @section image2 @@ -306,91 +149,64 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg}, etc. -@subsection Examples - -The following example shows how to use @command{ffmpeg} for creating a +The following example shows how to use @command{avconv} for creating a sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ..., taking one image every second from the input video: @example -ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' +avconv -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' @end example -Note that with @command{ffmpeg}, if the format is not specified with the +Note that with @command{avconv}, if the format is not specified with the @code{-f} option and the output filename specifies an image file format, the image2 muxer is automatically selected, so the previous command can be written as: @example -ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg' +avconv -i in.avi -vsync 1 -r 1 'img-%03d.jpeg' @end example Note also that the pattern must not necessarily contain "%d" or "%0@var{N}d", for example to create a single image file @file{img.jpeg} from the input video you can employ the command: @example -ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg +avconv -i in.avi -f image2 -frames:v 1 img.jpeg @end example -The @option{strftime} option allows you to expand the filename with -date and time information. Check the documentation of -the @code{strftime()} function for the syntax. - -For example to generate image files from the @code{strftime()} -"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command -can be used: -@example -ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg" -@end example - -@subsection Options - @table @option -@item start_number -Start the sequence from the specified number. Default value is 1. Must -be a non-negative number. +@item -start_number @var{number} +Start the sequence from @var{number}. -@item update -If set to 1, the filename will always be interpreted as just a -filename, not a pattern, and the corresponding file will be continuously -overwritten with new images. Default value is 0. +@item -update @var{number} +If @var{number} is nonzero, the filename will always be interpreted as just a +filename, not a pattern, and this file will be continuously overwritten with new +images. -@item strftime -If set to 1, expand the filename with date and time information from -@code{strftime()}. Default value is 0. @end table -The image muxer supports the .Y.U.V image file format. This format is -special in that that each image frame consists of three files, for -each of the YUV420P components. To read or write this image file format, -specify the name of the '.Y' file. The muxer will automatically open the -'.U' and '.V' files as required. - @section matroska Matroska container muxer. This muxer implements the matroska and webm container specs. -@subsection Metadata - The recognized metadata settings in this muxer are: @table @option -@item title -Set title name provided to a single track. -@item language -Specify the language of the track in the Matroska languages form. +@item title=@var{title name} +Name provided to a single track +@end table -The language can be either the 3 letters bibliographic ISO-639-2 (ISO -639-2/B) form (like "fre" for French), or a language code mixed with a -country code for specialities in languages (like "fre-ca" for Canadian -French). +@table @option -@item stereo_mode -Set stereo 3D video layout of two views in a single video track. +@item language=@var{language name} +Specifies the language of the track in the Matroska languages form +@end table -The following values are recognized: -@table @samp +@table @option + +@item STEREO_MODE=@var{mode} +Stereo 3D video layout of two views in a single video track +@table @option @item mono video is not stereo @item left_right @@ -426,14 +242,13 @@ Both eyes laced in one Block, Right-eye view is first For example a 3D WebM clip can be created using the following command line: @example -ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm +avconv -i sample_left_right_clip.mpg -an -c:v libvpx -metadata STEREO_MODE=left_right -y stereo_clip.webm @end example -@subsection Options - This muxer supports the following options: @table @option + @item reserve_index_space By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space @@ -448,44 +263,15 @@ for most use cases should be about 50kB per hour of video. Note that cues are only written if the output is seekable and this option will have no effect if it is not. + @end table -@anchor{md5} -@section md5 - -MD5 testing format. - -This muxer computes and prints the MD5 hash of all the input audio -and video frames. By default audio frames are converted to signed -16-bit raw audio and video frames to raw video before computing the -hash. - -The output of the muxer consists of a single line of the form: -MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing -the computed MD5 hash. - -For example to compute the MD5 hash of the input converted to raw -audio and video, and store it in the file @file{out.md5}: -@example -ffmpeg -i INPUT -f md5 out.md5 -@end example - -You can print the MD5 to stdout with the command: -@example -ffmpeg -i INPUT -f md5 - -@end example - -See also the @ref{framemd5} muxer. - @section mov, mp4, ismv -MOV/MP4/ISMV (Smooth Streaming) muxer. - The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file has all the metadata about all packets stored in one location (written at the end of the file, it can be moved to the start for -better playback by adding @var{faststart} to the @var{movflags}, or -using the @command{qt-faststart} tool). A fragmented +better playback using the @command{qt-faststart} tool). A fragmented file consists of a number of fragments, where packets and metadata about these packets are stored together. Writing a fragmented file has the advantage that the file is decodable even if the @@ -495,15 +281,10 @@ very long files (since writing normal MOV/MP4 files stores info about every single packet in memory until the file is closed). The downside is that it is less compatible with other applications. -@subsection Options - Fragmentation is enabled by setting one of the AVOptions that define how to cut the file into fragments: @table @option -@item -moov_size @var{bytes} -Reserves space for the moov atom at the beginning of the file instead of placing the -moov atom at the end. If the space reserved is insufficient, muxing will fail. @item -movflags frag_keyframe Start a new fragment at each video keyframe. @item -frag_duration @var{duration} @@ -514,7 +295,7 @@ Create fragments that contain up to @var{size} bytes of payload data. Allow the caller to manually choose when to cut fragments, by calling @code{av_write_frame(ctx, NULL)} to write a fragment with the packets written so far. (This is only useful with other -applications integrating libavformat, not from @command{ffmpeg}.) +applications integrating libavformat, not from @command{avconv}.) @item -min_frag_duration @var{duration} Don't create fragments that are shorter than @var{duration} microseconds long. @end table @@ -549,16 +330,12 @@ This option is implicitly set when writing ismv (Smooth Streaming) files. Run a second pass moving the index (moov atom) to the beginning of the file. This operation can take a while, and will not work in various situations such as fragmented output, thus it is not enabled by default. -@item -movflags rtphint -Add RTP hinting tracks to the output file. @end table -@subsection Example - Smooth Streaming content can be pushed in real time to a publishing point on IIS with this muxer. Example: @example -ffmpeg -re @var{} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) +avconv -re @var{} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) @end example @section mp3 @@ -590,19 +367,18 @@ Examples: Write an mp3 with an ID3v2.3 header and an ID3v1 footer: @example -ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 +avconv -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 @end example -To attach a picture to an mp3 file select both the audio and the picture stream -with @code{map}: +Attach a picture to an mp3: @example -ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1 --metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 +avconv -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover" +-metadata:s:v comment="Cover (Front)" out.mp3 @end example Write a "clean" MP3 without any extra features: @example -ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3 +avconv -i input.wav -write_xing 0 -id3v2_version 0 out.mp3 @end example @section mpegts @@ -611,13 +387,6 @@ MPEG transport stream muxer. This muxer implements ISO 13818-1 and part of ETSI EN 300 468. -The recognized metadata settings in mpegts muxer are @code{service_provider} -and @code{service_name}. If they are not set the default for -@code{service_provider} is "FFmpeg" and the default for -@code{service_name} is "Service01". - -@subsection Options - The muxer options are: @table @option @@ -634,46 +403,15 @@ Set the service_id (default 0x0001) also known as program in DVB. Set the first PID for PMT (default 0x1000, max 0x1f00). @item -mpegts_start_pid @var{number} Set the first PID for data packets (default 0x0100, max 0x0f00). -@item -mpegts_m2ts_mode @var{number} -Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode. -@item -muxrate @var{number} -Set muxrate. -@item -pes_payload_size @var{number} -Set minimum PES packet payload in bytes. -@item -mpegts_flags @var{flags} -Set flags (see below). -@item -mpegts_copyts @var{number} -Preserve original timestamps, if value is set to 1. Default value is -1, which -results in shifting timestamps so that they start from 0. -@item -tables_version @var{number} -Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively). -This option allows updating stream structure so that standard consumer may -detect the change. To do so, reopen output AVFormatContext (in case of API -usage) or restart ffmpeg instance, cyclically changing tables_version value: -@example -ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111 -ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111 -... -ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111 -ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111 -ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111 -... -@end example @end table -Option mpegts_flags may take a set of such flags: - -@table @option -@item resend_headers -Reemit PAT/PMT before writing the next packet. -@item latm -Use LATM packetization for AAC. -@end table - -@subsection Example +The recognized metadata settings in mpegts muxer are @code{service_provider} +and @code{service_name}. If they are not set the default for +@code{service_provider} is "Libav" and the default for +@code{service_name} is "Service01". @example -ffmpeg -i file.mpg -c copy \ +avconv -i file.mpg -c copy \ -mpegts_original_network_id 0x1122 \ -mpegts_transport_stream_id 0x3344 \ -mpegts_service_id 0x5566 \ @@ -691,19 +429,19 @@ Null muxer. This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes. -For example to benchmark decoding with @command{ffmpeg} you can use the +For example to benchmark decoding with @command{avconv} you can use the command: @example -ffmpeg -benchmark -i INPUT -f null out.null +avconv -benchmark -i INPUT -f null out.null @end example Note that the above command does not read or write the @file{out.null} -file, but specifying the output file is required by the @command{ffmpeg} +file, but specifying the output file is required by the @command{avconv} syntax. Alternatively you can write the command as: @example -ffmpeg -benchmark -i INPUT -f null - +avconv -benchmark -i INPUT -f null - @end example @section ogg @@ -721,342 +459,38 @@ situations, giving a small seek granularity at the cost of additional container overhead. @end table -@anchor{segment} -@section segment, stream_segment, ssegment +@section segment Basic stream segmenter. -This muxer outputs streams to a number of separate files of nearly +The segmenter muxer outputs streams to a number of separate files of nearly fixed duration. Output filename pattern can be set in a fashion similar to @ref{image2}. -@code{stream_segment} is a variant of the muxer used to write to -streaming output formats, i.e. which do not require global headers, -and is recommended for outputting e.g. to MPEG transport stream segments. -@code{ssegment} is a shorter alias for @code{stream_segment}. - -Every segment starts with a keyframe of the selected reference stream, -which is set through the @option{reference_stream} option. - -Note that if you want accurate splitting for a video file, you need to -make the input key frames correspond to the exact splitting times -expected by the segmenter, or the segment muxer will start the new -segment with the key frame found next after the specified start -time. - +Every segment starts with a video keyframe, if a video stream is present. The segment muxer works best with a single constant frame rate video. -Optionally it can generate a list of the created segments, by setting -the option @var{segment_list}. The list type is specified by the -@var{segment_list_type} option. The entry filenames in the segment -list are set by default to the basename of the corresponding segment -files. - -See also the @ref{hls} muxer, which provides a more specific -implementation for HLS segmentation. - -@subsection Options - -The segment muxer supports the following options: +Optionally it can generate a flat list of the created segments, one segment +per line. @table @option -@item reference_stream @var{specifier} -Set the reference stream, as specified by the string @var{specifier}. -If @var{specifier} is set to @code{auto}, the reference is chosen -automatically. Otherwise it must be a stream specifier (see the ``Stream -specifiers'' chapter in the ffmpeg manual) which specifies the -reference stream. The default value is @code{auto}. - @item segment_format @var{format} Override the inner container format, by default it is guessed by the filename extension. - +@item segment_time @var{t} +Set segment duration to @var{t} seconds. @item segment_list @var{name} -Generate also a listfile named @var{name}. If not specified no -listfile is generated. - -@item segment_list_flags @var{flags} -Set flags affecting the segment list generation. - -It currently supports the following flags: -@table @samp -@item cache -Allow caching (only affects M3U8 list files). - -@item live -Allow live-friendly file generation. -@end table - +Generate also a listfile named @var{name}. @item segment_list_size @var{size} -Update the list file so that it contains at most the last @var{size} -segments. If 0 the list file will contain all the segments. Default -value is 0. - +Overwrite the listfile once it reaches @var{size} entries. @item segment_list_entry_prefix @var{prefix} Prepend @var{prefix} to each entry. Useful to generate absolute paths. -By default no prefix is applied. - -@item segment_list_type @var{type} -Specify the format for the segment list file. - -The following values are recognized: -@table @samp -@item flat -Generate a flat list for the created segments, one segment per line. - -@item csv, ext -Generate a list for the created segments, one segment per line, -each line matching the format (comma-separated values): -@example -@var{segment_filename},@var{segment_start_time},@var{segment_end_time} -@end example - -@var{segment_filename} is the name of the output file generated by the -muxer according to the provided pattern. CSV escaping (according to -RFC4180) is applied if required. - -@var{segment_start_time} and @var{segment_end_time} specify -the segment start and end time expressed in seconds. - -A list file with the suffix @code{".csv"} or @code{".ext"} will -auto-select this format. - -@samp{ext} is deprecated in favor or @samp{csv}. - -@item ffconcat -Generate an ffconcat file for the created segments. The resulting file -can be read using the FFmpeg @ref{concat} demuxer. - -A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will -auto-select this format. - -@item m3u8 -Generate an extended M3U8 file, version 3, compliant with -@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}. - -A list file with the suffix @code{".m3u8"} will auto-select this format. -@end table - -If not specified the type is guessed from the list file name suffix. - -@item segment_time @var{time} -Set segment duration to @var{time}, the value must be a duration -specification. Default value is "2". See also the -@option{segment_times} option. - -Note that splitting may not be accurate, unless you force the -reference stream key-frames at the given time. See the introductory -notice and the examples below. - -@item segment_time_delta @var{delta} -Specify the accuracy time when selecting the start time for a -segment, expressed as a duration specification. Default value is "0". - -When delta is specified a key-frame will start a new segment if its -PTS satisfies the relation: -@example -PTS >= start_time - time_delta -@end example - -This option is useful when splitting video content, which is always -split at GOP boundaries, in case a key frame is found just before the -specified split time. - -In particular may be used in combination with the @file{ffmpeg} option -@var{force_key_frames}. The key frame times specified by -@var{force_key_frames} may not be set accurately because of rounding -issues, with the consequence that a key frame time may result set just -before the specified time. For constant frame rate videos a value of -1/(2*@var{frame_rate}) should address the worst case mismatch between -the specified time and the time set by @var{force_key_frames}. - -@item segment_times @var{times} -Specify a list of split points. @var{times} contains a list of comma -separated duration specifications, in increasing order. See also -the @option{segment_time} option. - -@item segment_frames @var{frames} -Specify a list of split video frame numbers. @var{frames} contains a -list of comma separated integer numbers, in increasing order. - -This option specifies to start a new segment whenever a reference -stream key frame is found and the sequential number (starting from 0) -of the frame is greater or equal to the next value in the list. - @item segment_wrap @var{limit} Wrap around segment index once it reaches @var{limit}. - -@item segment_start_number @var{number} -Set the sequence number of the first segment. Defaults to @code{0}. - -@item reset_timestamps @var{1|0} -Reset timestamps at the begin of each segment, so that each segment -will start with near-zero timestamps. It is meant to ease the playback -of the generated segments. May not work with some combinations of -muxers/codecs. It is set to @code{0} by default. - -@item initial_offset @var{offset} -Specify timestamp offset to apply to the output packet timestamps. The -argument must be a time duration specification, and defaults to 0. @end table -@subsection Examples - -@itemize -@item -To remux the content of file @file{in.mkv} to a list of segments -@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of -generated segments to @file{out.list}: @example -ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut +avconv -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut @end example -@item -As the example above, but segment the input file according to the split -points specified by the @var{segment_times} option: -@example -ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut -@end example - -@item -As the example above, but use the @command{ffmpeg} @option{force_key_frames} -option to force key frames in the input at the specified location, together -with the segment option @option{segment_time_delta} to account for -possible roundings operated when setting key frame times. -@example -ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \ --f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut -@end example -In order to force key frames on the input file, transcoding is -required. - -@item -Segment the input file by splitting the input file according to the -frame numbers sequence specified with the @option{segment_frames} option: -@example -ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut -@end example - -@item -To convert the @file{in.mkv} to TS segments using the @code{libx264} -and @code{libfaac} encoders: -@example -ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts -@end example - -@item -Segment the input file, and create an M3U8 live playlist (can be used -as live HLS source): -@example -ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \ --segment_list_flags +live -segment_time 10 out%03d.mkv -@end example -@end itemize - -@section smoothstreaming - -Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server. - -@table @option -@item window_size -Specify the number of fragments kept in the manifest. Default 0 (keep all). - -@item extra_window_size -Specify the number of fragments kept outside of the manifest before removing from disk. Default 5. - -@item lookahead_count -Specify the number of lookahead fragments. Default 2. - -@item min_frag_duration -Specify the minimum fragment duration (in microseconds). Default 5000000. - -@item remove_at_exit -Specify whether to remove all fragments when finished. Default 0 (do not remove). - -@end table - -@section tee - -The tee muxer can be used to write the same data to several files or any -other kind of muxer. It can be used, for example, to both stream a video to -the network and save it to disk at the same time. - -It is different from specifying several outputs to the @command{ffmpeg} -command-line tool because the audio and video data will be encoded only once -with the tee muxer; encoding can be a very expensive process. It is not -useful when using the libavformat API directly because it is then possible -to feed the same packets to several muxers directly. - -The slave outputs are specified in the file name given to the muxer, -separated by '|'. If any of the slave name contains the '|' separator, -leading or trailing spaces or any special character, it must be -escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping" -section in the ffmpeg-utils(1) manual,ffmpeg-utils}). - -Muxer options can be specified for each slave by prepending them as a list of -@var{key}=@var{value} pairs separated by ':', between square brackets. If -the options values contain a special character or the ':' separator, they -must be escaped; note that this is a second level escaping. - -The following special options are also recognized: -@table @option -@item f -Specify the format name. Useful if it cannot be guessed from the -output name suffix. - -@item bsfs[/@var{spec}] -Specify a list of bitstream filters to apply to the specified -output. - -It is possible to specify to which streams a given bitstream filter -applies, by appending a stream specifier to the option separated by -@code{/}. @var{spec} must be a stream specifier (see @ref{Format -stream specifiers}). If the stream specifier is not specified, the -bitstream filters will be applied to all streams in the output. - -Several bitstream filters can be specified, separated by ",". - -@item select -Select the streams that should be mapped to the slave output, -specified by a stream specifier. If not specified, this defaults to -all the input streams. -@end table - -@subsection Examples - -@itemize -@item -Encode something and both archive it in a WebM file and stream it -as MPEG-TS over UDP (the streams need to be explicitly mapped): -@example -ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a - "archive-20121107.mkv|[f=mpegts]udp://10.0.0.0:1234/" -@end example - -@item -Use @command{ffmpeg} to encode the input, and send the output -to three different destinations. The @code{dump_extra} bitstream -filter is used to add extradata information to all the output video -keyframes packets, as requested by the MPEG-TS format. The select -option is applied to @file{out.aac} in order to make it contain only -audio packets. -@example -ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental - -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac" -@end example - -@item -As below, but select only stream @code{a:1} for the audio output. Note -that a second level escaping must be performed, as ":" is a special -character used to separate options. -@example -ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental - -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac" -@end example -@end itemize - -Note: some codecs may need different options depending on the output format; -the auto-detection of this can not work with the tee muxer. The main example -is the @option{global_header} flag. - @c man end MUXERS diff --git a/doc/nut.texi b/doc/nut.texi index 0026a121b5..39a22ff3ea 100644 --- a/doc/nut.texi +++ b/doc/nut.texi @@ -17,10 +17,6 @@ subtitle and user-defined streams in a simple, yet efficient, way. It was created by a group of FFmpeg and MPlayer developers in 2003 and was finalized in 2008. -The official nut specification is at svn://svn.mplayerhq.hu/nut -In case of any differences between this text and the official specification, -the official specification shall prevail. - @chapter Container-specific codec tags @section Generic raw YUVA formats @@ -113,7 +109,6 @@ PFD[32] would for example be signed 32 bit little-endian IEEE float @item RV20 @tab RealVideo 2.0 @item RV30 @tab RealVideo 3.0 @item RV40 @tab RealVideo 4.0 -@item SNOW @tab FFmpeg Snow @item SVQ1 @tab Sorenson Video 1 @item SVQ3 @tab Sorenson Video 3 @item theo @tab Xiph Theora diff --git a/doc/optimization.txt b/doc/optimization.txt index cce472f4ca..b51183fa34 100644 --- a/doc/optimization.txt +++ b/doc/optimization.txt @@ -17,15 +17,15 @@ Understanding these overoptimized functions: As many functions tend to be a bit difficult to understand because of optimizations, it can be hard to optimize them further, or write architecture-specific versions. It is recommended to look at older -revisions of the interesting files (web frontends for the various FFmpeg -branches are listed at http://ffmpeg.org/download.html). +revisions of the interesting files (web frontends for the various Libav +branches are listed at http://libav.org/download.html). Alternatively, look into the other architecture-specific versions in the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly comprehend the instructions, it could help understanding the functions and how they can be optimized. NOTE: If you still don't understand some function, ask at our mailing list!!! -(http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel) +(https://lists.libav.org/mailman/listinfo/libav-devel) When is an optimization justified? @@ -198,7 +198,7 @@ Inline asm vs. external asm --------------------------- Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc) and external asm (.s or .asm files, handled by an assembler such as yasm/nasm) -are accepted in FFmpeg. Which one to use differs per specific case. +are accepted in Libav. Which one to use differs per specific case. - if your code is intended to be inlined in a C function, inline asm is always better, because external asm cannot be inlined diff --git a/doc/outdevs.texi b/doc/outdevs.texi index 488a1ad362..dd7bd6475d 100644 --- a/doc/outdevs.texi +++ b/doc/outdevs.texi @@ -1,10 +1,10 @@ @chapter Output Devices @c man begin OUTPUT DEVICES -Output devices are configured elements in FFmpeg that can write +Output devices are configured elements in Libav which allow to write multimedia data to an output device attached to your system. -When you configure your FFmpeg build, all the supported output devices +When you configure your Libav build, all the supported output devices are enabled by default. You can list all available ones using the configure option "--list-outdevs". @@ -13,7 +13,7 @@ You can disable all the output devices using the configure option option "--enable-outdev=@var{OUTDEV}", or you can disable a particular input device using the option "--disable-outdev=@var{OUTDEV}". -The option "-formats" of the ff* tools will display the list of +The option "-formats" of the av* tools will display the list of enabled output devices (amongst the muxers). A description of the currently available output devices follows. @@ -22,429 +22,12 @@ A description of the currently available output devices follows. ALSA (Advanced Linux Sound Architecture) output device. -@subsection Examples - -@itemize -@item -Play a file on default ALSA device: -@example -ffmpeg -i INPUT -f alsa default -@end example - -@item -Play a file on soundcard 1, audio device 7: -@example -ffmpeg -i INPUT -f alsa hw:1,7 -@end example -@end itemize - -@section caca - -CACA output device. - -This output device allows one to show a video stream in CACA window. -Only one CACA window is allowed per application, so you can -have only one instance of this output device in an application. - -To enable this output device you need to configure FFmpeg with -@code{--enable-libcaca}. -libcaca is a graphics library that outputs text instead of pixels. - -For more information about libcaca, check: -@url{http://caca.zoy.org/wiki/libcaca} - -@subsection Options - -@table @option - -@item window_title -Set the CACA window title, if not specified default to the filename -specified for the output device. - -@item window_size -Set the CACA window size, can be a string of the form -@var{width}x@var{height} or a video size abbreviation. -If not specified it defaults to the size of the input video. - -@item driver -Set display driver. - -@item algorithm -Set dithering algorithm. Dithering is necessary -because the picture being rendered has usually far more colours than -the available palette. -The accepted values are listed with @code{-list_dither algorithms}. - -@item antialias -Set antialias method. Antialiasing smoothens the rendered -image and avoids the commonly seen staircase effect. -The accepted values are listed with @code{-list_dither antialiases}. - -@item charset -Set which characters are going to be used when rendering text. -The accepted values are listed with @code{-list_dither charsets}. - -@item color -Set color to be used when rendering text. -The accepted values are listed with @code{-list_dither colors}. - -@item list_drivers -If set to @option{true}, print a list of available drivers and exit. - -@item list_dither -List available dither options related to the argument. -The argument must be one of @code{algorithms}, @code{antialiases}, -@code{charsets}, @code{colors}. -@end table - -@subsection Examples - -@itemize -@item -The following command shows the @command{ffmpeg} output is an -CACA window, forcing its size to 80x25: -@example -ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca - -@end example - -@item -Show the list of available drivers and exit: -@example -ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true - -@end example - -@item -Show the list of available dither colors and exit: -@example -ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors - -@end example -@end itemize - -@section decklink - -The decklink output device provides playback capabilities for Blackmagic -DeckLink devices. - -To enable this output device, you need the Blackmagic DeckLink SDK and you -need to configure with the appropriate @code{--extra-cflags} -and @code{--extra-ldflags}. -On Windows, you need to run the IDL files through @command{widl}. - -DeckLink is very picky about the formats it supports. Pixel format is always -uyvy422, framerate and video size must be determined for your device with -@command{-list_formats 1}. Audio sample rate is always 48 kHz. - -@subsection Options - -@table @option - -@item list_devices -If set to @option{true}, print a list of devices and exit. -Defaults to @option{false}. - -@item list_formats -If set to @option{true}, print a list of supported formats and exit. -Defaults to @option{false}. - -@item preroll -Amount of time to preroll video in seconds. -Defaults to @option{0.5}. - -@end table - -@subsection Examples - -@itemize - -@item -List output devices: -@example -ffmpeg -i test.avi -f decklink -list_devices 1 dummy -@end example - -@item -List supported formats: -@example -ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor' -@end example - -@item -Play video clip: -@example -ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor' -@end example - -@item -Play video clip with non-standard framerate or video size: -@example -ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor' -@end example - -@end itemize - -@section fbdev - -Linux framebuffer output device. - -The Linux framebuffer is a graphic hardware-independent abstraction -layer to show graphics on a computer monitor, typically on the -console. It is accessed through a file device node, usually -@file{/dev/fb0}. - -For more detailed information read the file -@file{Documentation/fb/framebuffer.txt} included in the Linux source tree. - -@subsection Options -@table @option - -@item xoffset -@item yoffset -Set x/y coordinate of top left corner. Default is 0. -@end table - -@subsection Examples -Play a file on framebuffer device @file{/dev/fb0}. -Required pixel format depends on current framebuffer settings. -@example -ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0 -@end example - -See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1). - -@section opengl -OpenGL output device. - -To enable this output device you need to configure FFmpeg with @code{--enable-opengl}. - -This output device allows one to render to OpenGL context. -Context may be provided by application or default SDL window is created. - -When device renders to external context, application must implement handlers for following messages: -@code{AV_DEV_TO_APP_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread. -@code{AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER} - make OpenGL context current. -@code{AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER} - swap buffers. -@code{AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER} - destroy OpenGL context. -Application is also required to inform a device about current resolution by sending @code{AV_APP_TO_DEV_WINDOW_SIZE} message. - -@subsection Options -@table @option - -@item background -Set background color. Black is a default. -@item no_window -Disables default SDL window when set to non-zero value. -Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set. -@item window_title -Set the SDL window title, if not specified default to the filename specified for the output device. -Ignored when @option{no_window} is set. -@item window_size -Set preferred window size, can be a string of the form widthxheight or a video size abbreviation. -If not specified it defaults to the size of the input video, downscaled according to the aspect ratio. -Mostly usable when @option{no_window} is not set. - -@end table - -@subsection Examples -Play a file on SDL window using OpenGL rendering: -@example -ffmpeg -i INPUT -f opengl "window title" -@end example - @section oss OSS (Open Sound System) output device. -@section pulse - -PulseAudio output device. - -To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}. - -More information about PulseAudio can be found on @url{http://www.pulseaudio.org} - -@subsection Options -@table @option - -@item server -Connect to a specific PulseAudio server, specified by an IP address. -Default server is used when not provided. - -@item name -Specify the application name PulseAudio will use when showing active clients, -by default it is the @code{LIBAVFORMAT_IDENT} string. - -@item stream_name -Specify the stream name PulseAudio will use when showing active streams, -by default it is set to the specified output name. - -@item device -Specify the device to use. Default device is used when not provided. -List of output devices can be obtained with command @command{pactl list sinks}. - -@item buffer_size -@item buffer_duration -Control the size and duration of the PulseAudio buffer. A small buffer -gives more control, but requires more frequent updates. - -@option{buffer_size} specifies size in bytes while -@option{buffer_duration} specifies duration in milliseconds. - -When both options are provided then the highest value is used -(duration is recalculated to bytes using stream parameters). If they -are set to 0 (which is default), the device will use the default -PulseAudio duration value. By default PulseAudio set buffer duration -to around 2 seconds. - -@item prebuf -Specify pre-buffering size in bytes. The server does not start with -playback before at least @option{prebuf} bytes are available in the -buffer. By default this option is initialized to the same value as -@option{buffer_size} or @option{buffer_duration} (whichever is bigger). - -@item minreq -Specify minimum request size in bytes. The server does not request less -than @option{minreq} bytes from the client, instead waits until the buffer -is free enough to request more bytes at once. It is recommended to not set -this option, which will initialize this to a value that is deemed sensible -by the server. - -@end table - -@subsection Examples -Play a file on default device on default server: -@example -ffmpeg -i INPUT -f pulse "stream name" -@end example - -@section sdl - -SDL (Simple DirectMedia Layer) output device. - -This output device allows one to show a video stream in an SDL -window. Only one SDL window is allowed per application, so you can -have only one instance of this output device in an application. - -To enable this output device you need libsdl installed on your system -when configuring your build. - -For more information about SDL, check: -@url{http://www.libsdl.org/} - -@subsection Options - -@table @option - -@item window_title -Set the SDL window title, if not specified default to the filename -specified for the output device. - -@item icon_title -Set the name of the iconified SDL window, if not specified it is set -to the same value of @var{window_title}. - -@item window_size -Set the SDL window size, can be a string of the form -@var{width}x@var{height} or a video size abbreviation. -If not specified it defaults to the size of the input video, -downscaled according to the aspect ratio. - -@item window_fullscreen -Set fullscreen mode when non-zero value is provided. -Default value is zero. -@end table - -@subsection Interactive commands - -The window created by the device can be controlled through the -following interactive commands. - -@table @key -@item q, ESC -Quit the device immediately. -@end table - -@subsection Examples - -The following command shows the @command{ffmpeg} output is an -SDL window, forcing its size to the qcif format: -@example -ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output" -@end example - @section sndio sndio audio output device. -@section xv - -XV (XVideo) output device. - -This output device allows one to show a video stream in a X Window System -window. - -@subsection Options - -@table @option -@item display_name -Specify the hardware display name, which determines the display and -communications domain to be used. - -The display name or DISPLAY environment variable can be a string in -the format @var{hostname}[:@var{number}[.@var{screen_number}]]. - -@var{hostname} specifies the name of the host machine on which the -display is physically attached. @var{number} specifies the number of -the display server on that host machine. @var{screen_number} specifies -the screen to be used on that server. - -If unspecified, it defaults to the value of the DISPLAY environment -variable. - -For example, @code{dual-headed:0.1} would specify screen 1 of display -0 on the machine named ``dual-headed''. - -Check the X11 specification for more detailed information about the -display name format. - -@item window_id -When set to non-zero value then device doesn't create new window, -but uses existing one with provided @var{window_id}. By default -this options is set to zero and device creates its own window. - -@item window_size -Set the created window size, can be a string of the form -@var{width}x@var{height} or a video size abbreviation. If not -specified it defaults to the size of the input video. -Ignored when @var{window_id} is set. - -@item window_x -@item window_y -Set the X and Y window offsets for the created window. They are both -set to 0 by default. The values may be ignored by the window manager. -Ignored when @var{window_id} is set. - -@item window_title -Set the window title, if not specified default to the filename -specified for the output device. Ignored when @var{window_id} is set. -@end table - -For more information about XVideo see @url{http://www.x.org/}. - -@subsection Examples - -@itemize -@item -Decode, display and encode video input with @command{ffmpeg} at the -same time: -@example -ffmpeg -i INPUT OUTPUT -f xv display -@end example - -@item -Decode and display the input video to multiple X11 windows: -@example -ffmpeg -i INPUT -f xv normal -vf negate -f xv negated -@end example -@end itemize - @c man end OUTPUT DEVICES diff --git a/doc/platform.texi b/doc/platform.texi index c999383acb..45ec2754d9 100644 --- a/doc/platform.texi +++ b/doc/platform.texi @@ -1,8 +1,8 @@ \input texinfo @c -*- texinfo -*- -@settitle Platform Specific Information +@settitle Platform Specific information @titlepage -@center @titlefont{Platform Specific Information} +@center @titlefont{Platform Specific information} @end titlepage @top @@ -11,7 +11,7 @@ @chapter Unix-like -Some parts of FFmpeg cannot be built with version 2.15 of the GNU +Some parts of Libav cannot be built with version 2.15 of the GNU assembler which is still provided by a few AMD64 distributions. To make sure your compiler really uses the required version of gas after a binutils upgrade, run: @@ -26,12 +26,12 @@ to configure. @section BSD -BSD make will not build FFmpeg, you need to install and use GNU Make +BSD make will not build Libav, you need to install and use GNU Make (@command{gmake}). @section (Open)Solaris -GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}), +GNU Make is required to build Libav, so you have to invoke (@command{gmake}), standard Solaris Make will not work. When building with a non-c99 front-end (gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o} or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options @@ -45,22 +45,20 @@ bash ./configure @end example @anchor{Darwin} -@section Darwin (Mac OS X, iPhone) +@section Darwin (OS X, iPhone) The toolchain provided with Xcode is sufficient to build the basic unacelerated code. -Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from -@url{https://github.com/FFmpeg/gas-preprocessor} or -@url{https://github.com/yuvi/gas-preprocessor}(currently outdated) to build the optimized +OS X on PowerPC or ARM (iPhone) requires a preprocessor from +@url{git://git.libav.org/gas-preprocessor.git} to build the optimized assembler functions. Put the Perl script somewhere -in your PATH, FFmpeg's configure will pick it up automatically. +in your PATH, Libav's configure will pick it up automatically. -Mac OS X on amd64 and x86 requires @command{yasm} to build most of the -optimized assembler functions. @uref{http://www.finkproject.org/, Fink}, -@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix}, -@uref{https://mxcl.github.com/homebrew/, Homebrew} -or @uref{http://www.macports.org, MacPorts} can easily provide it. +OS X on AMD64 and x86 requires @command{yasm} to build most of the +optimized assembler functions @url{http://mxcl.github.com/homebrew/, Homebrew}, +@url{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix} +or @url{http://www.macports.org, MacPorts} can easily provide it. @chapter DOS @@ -71,18 +69,15 @@ Using a cross-compiler is preferred for various reasons. @chapter OS/2 -For information about compiling FFmpeg on OS/2 see +For information about compiling Libav on OS/2 see @url{http://www.edm2.com/index.php/FFmpeg}. @chapter Windows -To get help and instructions for building FFmpeg under Windows, check out -the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}. - @section Native Windows compilation using MinGW or MinGW-w64 -FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64 +Libav can be built to run natively on Windows using the MinGW or MinGW-w64 toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from @url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}. You can find detailed installation instructions in the download section and @@ -98,18 +93,17 @@ speed up is close to non-existent for normal one-off builds and is only noticeable when running make for a second time (for example during @code{make install}). -@item In order to compile FFplay, you must have the MinGW development library +@item In order to compile AVplay, you must have the MinGW development library of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed. -@item By using @code{./configure --enable-shared} when configuring FFmpeg, -you can build the FFmpeg libraries (e.g. libavutil, libavcodec, -libavformat) as DLLs. +@item By using @code{./configure --enable-shared} when configuring Libav, +you can build all libraries as DLLs. @end itemize @section Microsoft Visual C++ or Intel C++ Compiler for Windows -FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility +Libav can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility and wrapper, or with MSVC 2013 and ICL natively. You will need the following prerequisites: @@ -175,15 +169,15 @@ follow step 3, or compilation will fail. @enumerate @item Grab the @uref{http://zlib.net/, zlib sources}. @item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since -this is how FFmpeg is built as well. +this is how Libav is built as well. @item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets -erroneously included when building FFmpeg. +erroneously included when building Libav. @item Run @code{nmake -f win32/Makefile.msc}. @item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC can see. @end enumerate -@item FFmpeg has been tested with the following on i686 and x86_64: +@item Libav has been tested with the following on i686 and x86_64: @itemize @item Visual Studio 2010 Pro and Express @item Visual Studio 2012 Pro and Express @@ -195,7 +189,7 @@ Anything else is not officially supported. @end itemize -@subsection Linking to FFmpeg with Microsoft Visual C++ +@subsection Linking to Libav with Microsoft Visual C++ If you plan to link with MSVC-built static libraries, you will need to make sure you have @code{Runtime Library} set to @@ -247,14 +241,14 @@ Replace @code{foo-version} and @code{foo} with the respective library names. You must use the MinGW cross compilation tools available at @url{http://www.mingw.org/}. -Then configure FFmpeg with the following options: +Then configure Libav with the following options: @example ./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc- @end example (you can change the cross-prefix according to the prefix chosen for the MinGW tools). -Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}. +Then you can easily test Libav with @uref{http://www.winehq.com/, Wine}. @section Compilation under Cygwin @@ -272,7 +266,7 @@ In order to run FATE you will also need the following "Utils" packages: bc, diffutils @end example -If you want to build FFmpeg with additional libraries, download Cygwin +If you want to build Libav with additional libraries, download Cygwin "Devel" packages for Ogg and Vorbis from any Cygwin packages repository: @example libogg-devel, libvorbis-devel @@ -282,7 +276,7 @@ These library packages are only available from @uref{http://sourceware.org/cygwinports/, Cygwin Ports}: @example -yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel, +yasm, libSDL-devel, libfaac-devel, libgsm-devel, libmp3lame-devel, libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel @end example @@ -314,7 +308,7 @@ and for a build with shared libraries @chapter Plan 9 The native @uref{http://plan9.bell-labs.com/plan9/, Plan 9} compiler -does not implement all the C99 features needed by FFmpeg so the gcc +does not implement all the C99 features needed by Libav so the gcc port must be used. Furthermore, a few items missing from the C library and shell environment need to be fixed. @@ -330,7 +324,7 @@ utility by setting @code{pkgpath} to @item Missing/broken @code{head} and @code{printf} commands -Replacements adequate for building FFmpeg can be found in the +Replacements adequate for building Libav can be found in the @code{compat/plan9} directory. Place these somewhere they will be found by the shell. These are not full implementations of the commands and are @emph{not} suitable for general use. @@ -368,7 +362,7 @@ build system of this library. @item FPU exceptions enabled by default Unlike most other systems, Plan 9 enables FPU exceptions by default. -These must be disabled before calling any FFmpeg functions. While the +These must be disabled before calling any Libav functions. While the included tools will do this automatically, other users of the libraries must do it themselves. diff --git a/doc/print_options.c b/doc/print_options.c index ec8d839939..aa75a00ba2 100644 --- a/doc/print_options.c +++ b/doc/print_options.c @@ -1,20 +1,20 @@ /* * Copyright (c) 2012 Anton Khirnov * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -113,8 +113,6 @@ int main(int argc, char **argv) if (argc < 2) print_usage(); - printf("@c DO NOT EDIT THIS FILE!\n" - "@c It was generated by print_options.\n\n"); if (!strcmp(argv[1], "format")) show_format_opts(); else if (!strcmp(argv[1], "codec")) diff --git a/doc/protocols.texi b/doc/protocols.texi index 133512a70d..1501dab60c 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -1,10 +1,10 @@ @chapter Protocols @c man begin PROTOCOLS -Protocols are configured elements in FFmpeg that enable access to -resources that require specific protocols. +Protocols are configured elements in Libav which allow to access +resources which require the use of a particular protocol. -When you configure your FFmpeg build, all the supported protocols are +When you configure your Libav build, all the supported protocols are enabled by default. You can list all available ones using the configure option "--list-protocols". @@ -14,51 +14,11 @@ option "--enable-protocol=@var{PROTOCOL}", or you can disable a particular protocol using the option "--disable-protocol=@var{PROTOCOL}". -The option "-protocols" of the ff* tools will display the list of +The option "-protocols" of the av* tools will display the list of supported protocols. A description of the currently available protocols follows. -@section bluray - -Read BluRay playlist. - -The accepted options are: -@table @option - -@item angle -BluRay angle - -@item chapter -Start chapter (1...N) - -@item playlist -Playlist to read (BDMV/PLAYLIST/?????.mpls) - -@end table - -Examples: - -Read longest playlist from BluRay mounted to /mnt/bluray: -@example -bluray:/mnt/bluray -@end example - -Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: -@example --playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray -@end example - -@section cache - -Caching wrapper for input stream. - -Cache the input stream to temporary file. It brings seeking capability to live streams. - -@example -cache:@var{URL} -@end example - @section concat Physical concatenation protocol. @@ -76,114 +36,30 @@ resource to be concatenated, each one possibly specifying a distinct protocol. For example to read a sequence of files @file{split1.mpeg}, -@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the +@file{split2.mpeg}, @file{split3.mpeg} with @command{avplay} use the command: @example -ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg +avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg @end example Note that you may need to escape the character "|" which is special for many shells. -@section crypto - -AES-encrypted stream reading protocol. - -The accepted options are: -@table @option -@item key -Set the AES decryption key binary block from given hexadecimal representation. - -@item iv -Set the AES decryption initialization vector binary block from given hexadecimal representation. -@end table - -Accepted URL formats: -@example -crypto:@var{URL} -crypto+@var{URL} -@end example - -@section data - -Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}. - -For example, to convert a GIF file given inline with @command{ffmpeg}: -@example -ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png -@end example - @section file File access protocol. -Allow to read from or write to a file. +Allow to read from or read to a file. -A file URL can have the form: -@example -file:@var{filename} -@end example - -where @var{filename} is the path of the file to read. - -An URL that does not have a protocol prefix will be assumed to be a -file URL. Depending on the build, an URL that looks like a Windows -path with the drive letter at the beginning will also be assumed to be -a file URL (usually not the case in builds for unix-like systems). - -For example to read from a file @file{input.mpeg} with @command{ffmpeg} +For example to read from a file @file{input.mpeg} with @command{avconv} use the command: @example -ffmpeg -i file:input.mpeg output.mpeg +avconv -i file:input.mpeg output.mpeg @end example -This protocol accepts the following options: - -@table @option -@item truncate -Truncate existing files on write, if set to 1. A value of 0 prevents -truncating. Default value is 1. - -@item blocksize -Set I/O operation maximum block size, in bytes. Default value is -@code{INT_MAX}, which results in not limiting the requested block size. -Setting this value reasonably low improves user termination request reaction -time, which is valuable for files on slow medium. -@end table - -@section ftp - -FTP (File Transfer Protocol). - -Allow to read from or write to remote resources using FTP protocol. - -Following syntax is required. -@example -ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg -@end example - -This protocol accepts the following options. - -@table @option -@item timeout -Set timeout of socket I/O operations used by the underlying low level -operation. By default it is set to -1, which means that the timeout is -not specified. - -@item ftp-anonymous-password -Password used when login as anonymous user. Typically an e-mail address -should be used. - -@item ftp-write-seekable -Control seekability of connection during encoding. If set to 1 the -resource is supposed to be seekable, if set to 0 it is assumed not -to be seekable. Default value is 0. -@end table - -NOTE: Protocol can be used as output, but it is recommended to not do -it, unless special care is taken (tests, customized server configuration -etc.). Different FTP servers behave in different way during seek -operation. ff* tools may produce incomplete content due to server limitations. +The av* tools default to the file protocol, that is a resource +specified with the name "FILE.mpeg" is interpreted as the URL +"file:FILE.mpeg". @section gopher @@ -216,12 +92,6 @@ HTTP (Hyper Text Transfer Protocol). This protocol accepts the following options: @table @option -@item seekable -Control seekability of connection. If set to 1 the resource is -supposed to be seekable, if set to 0 it is assumed not to be seekable, -if set to -1 it will try to autodetect if it is seekable. Default -value is -1. - @item chunked_post If set to 1 use chunked Transfer-Encoding for posts, default is 1. @@ -238,15 +108,9 @@ Use persistent connections if set to 1, default is 0. @item post_data Set custom HTTP post data. -@item user-agent @item user_agent -Override the User-Agent header. If not specified the protocol will use a -string describing the libavformat build. ("Lavf/") - -@item timeout -Set timeout of socket I/O operations used by the underlying low level -operation. By default it is set to -1, which means that the timeout is -not specified. +Override the User-Agent header. If not specified a string of the form +"Lavf/" will be used. @item mime_type Export the MIME type. @@ -267,11 +131,6 @@ contains the last non-empty metadata packet sent by the server. It should be polled in regular intervals by applications interested in mid-stream metadata updates. -@item cookies -Set the cookies to be sent in future requests. The format of each cookie is the -same as the value of a Set-Cookie HTTP response field. Multiple cookies can be -delimited by a newline character. - @item offset Set initial byte offset. @@ -279,20 +138,6 @@ Set initial byte offset. Try to limit the request to bytes preceding this offset. @end table -@subsection HTTP Cookies - -Some HTTP requests will be denied unless cookie values are passed in with the -request. The @option{cookies} option allows these cookies to be specified. At -the very least, each cookie must specify a value along with a path and domain. -HTTP requests that match both the domain and path will automatically include the -cookie value in the HTTP Cookie header field. Multiple cookies can be delimited -by a newline. - -The required syntax to play a stream specifying a cookie is: -@example -ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8 -@end example - @section mmst MMS (Microsoft Media Server) protocol over TCP. @@ -317,10 +162,10 @@ be used to test muxers without writing an actual file. Some examples follow. @example # Write the MD5 hash of the encoded AVI file to the file output.avi.md5. -ffmpeg -i input.flv -f avi -y md5:output.avi.md5 +avconv -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. -ffmpeg -i input.flv -f avi -y md5: +avconv -i input.flv -f avi -y md5: @end example Note that some formats (typically MOV) require the output protocol to @@ -342,30 +187,20 @@ pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number} is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. -For example to read from stdin with @command{ffmpeg}: +For example to read from stdin with @command{avconv}: @example -cat test.wav | ffmpeg -i pipe:0 +cat test.wav | avconv -i pipe:0 # ...this is the same as... -cat test.wav | ffmpeg -i pipe: +cat test.wav | avconv -i pipe: @end example -For writing to stdout with @command{ffmpeg}: +For writing to stdout with @command{avconv}: @example -ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi +avconv -i test.wav -f avi pipe:1 | cat > test.avi # ...this is the same as... -ffmpeg -i test.wav -f avi pipe: | cat > test.avi +avconv -i test.wav -f avi pipe: | cat > test.avi @end example -This protocol accepts the following options: - -@table @option -@item blocksize -Set I/O operation maximum block size, in bytes. Default value is -@code{INT_MAX}, which results in not limiting the requested block size. -Setting this value reasonably low improves user termination request reaction -time, which is valuable if data transmission is slow. -@end table - Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol. @@ -485,16 +320,16 @@ URL of the target stream. Defaults to proto://host[:port]/app. @end table -For example to read with @command{ffplay} a multimedia resource named +For example to read with @command{avplay} a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver": @example -ffplay rtmp://myserver/vod/sample +avplay rtmp://myserver/vod/sample @end example To publish to a password protected server, passing the playpath and app names separately: @example -ffmpeg -re -i -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/ +avconv -re -i -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/ @end example @section rtmpe @@ -537,42 +372,6 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls. -@section libssh - -Secure File Transfer Protocol via libssh - -Allow to read from or write to remote resources using SFTP protocol. - -Following syntax is required. - -@example -sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg -@end example - -This protocol accepts the following options. - -@table @option -@item timeout -Set timeout of socket I/O operations used by the underlying low level -operation. By default it is set to -1, which means that the timeout -is not specified. - -@item truncate -Truncate existing files on write, if set to 1. A value of 0 prevents -truncating. Default value is 1. - -@item private_key -Specify the path of the file containing private key to use during authorization. -By default libssh searches for keys in the @file{~/.ssh/} directory. - -@end table - -Example: Play a file stored on remote server. - -@example -ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg -@end example - @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte Real-Time Messaging Protocol and its variants supported through @@ -603,87 +402,22 @@ meaning as specified for the RTMP native protocol. See the librtmp manual page (man 3 librtmp) for more information. For example, to stream a file in real-time to an RTMP server using -@command{ffmpeg}: +@command{avconv}: @example -ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream +avconv -re -i myfile -f flv rtmp://myserver/live/mystream @end example -To play the same stream using @command{ffplay}: +To play the same stream using @command{avplay}: @example -ffplay "rtmp://myserver/live/mystream live=1" +avplay "rtmp://myserver/live/mystream live=1" @end example @section rtp -Real-time Transport Protocol. - -The required syntax for an RTP URL is: -rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...] - -@var{port} specifies the RTP port to use. - -The following URL options are supported: - -@table @option - -@item ttl=@var{n} -Set the TTL (Time-To-Live) value (for multicast only). - -@item rtcpport=@var{n} -Set the remote RTCP port to @var{n}. - -@item localrtpport=@var{n} -Set the local RTP port to @var{n}. - -@item localrtcpport=@var{n}' -Set the local RTCP port to @var{n}. - -@item pkt_size=@var{n} -Set max packet size (in bytes) to @var{n}. - -@item connect=0|1 -Do a @code{connect()} on the UDP socket (if set to 1) or not (if set -to 0). - -@item sources=@var{ip}[,@var{ip}] -List allowed source IP addresses. - -@item block=@var{ip}[,@var{ip}] -List disallowed (blocked) source IP addresses. - -@item write_to_source=0|1 -Send packets to the source address of the latest received packet (if -set to 1) or to a default remote address (if set to 0). - -@item localport=@var{n} -Set the local RTP port to @var{n}. - -This is a deprecated option. Instead, @option{localrtpport} should be -used. - -@end table - -Important notes: - -@enumerate - -@item -If @option{rtcpport} is not set the RTCP port will be set to the RTP -port value plus 1. - -@item -If @option{localrtpport} (the local RTP port) is not set any available -port will be used for the local RTP and RTCP ports. - -@item -If @option{localrtcpport} (the local RTCP port) is not set it will be -set to the the local RTP port value plus 1. -@end enumerate +Real-Time Protocol. @section rtsp -Real-Time Streaming Protocol. - RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with @@ -691,29 +425,21 @@ data transferred over RDT). The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's -@uref{https://github.com/revmischa/rtsp-server, RTSP server}). +@uref{http://github.com/revmischa/rtsp-server, RTSP server}). The required syntax for a RTSP url is: @example rtsp://@var{hostname}[:@var{port}]/@var{path} @end example -Options can be set on the @command{ffmpeg}/@command{ffplay} command -line, or set in code via @code{AVOption}s or in -@code{avformat_open_input}. +The following options (set on the @command{avconv}/@command{avplay} command +line, or set in code via @code{AVOption}s or in @code{avformat_open_input}), +are supported: -The following options are supported. +Flags for @code{rtsp_transport}: @table @option -@item initial_pause -Do not start playing the stream immediately if set to 1. Default value -is 0. -@item rtsp_transport -Set RTSP trasport protocols. - -It accepts the following values: -@table @samp @item udp Use UDP as lower transport protocol. @@ -731,56 +457,15 @@ passing proxies. Multiple lower transport protocols may be specified, in that case they are tried one at a time (if the setup of one fails, the next one is tried). -For the muxer, only the @samp{tcp} and @samp{udp} options are supported. +For the muxer, only the @code{tcp} and @code{udp} options are supported. -@item rtsp_flags -Set RTSP flags. +Flags for @code{rtsp_flags}: -The following values are accepted: -@table @samp +@table @option @item filter_src Accept packets only from negotiated peer address and port. @item listen Act as a server, listening for an incoming connection. -@item prefer_tcp -Try TCP for RTP transport first, if TCP is available as RTSP RTP transport. -@end table - -Default value is @samp{none}. - -@item allowed_media_types -Set media types to accept from the server. - -The following flags are accepted: -@table @samp -@item video -@item audio -@item data -@end table - -By default it accepts all media types. - -@item min_port -Set minimum local UDP port. Default value is 5000. - -@item max_port -Set maximum local UDP port. Default value is 65000. - -@item timeout -Set maximum timeout (in seconds) to wait for incoming connections. - -A value of -1 mean infinite (default). This option implies the -@option{rtsp_flags} set to @samp{listen}. - -@item reorder_queue_size -Set number of packets to buffer for handling of reordered packets. - -@item stimeout -Set socket TCP I/O timeout in micro seconds. - -@item user-agent -Override User-Agent header. If not specified, it default to the -libavformat identifier string. @end table When receiving data over UDP, the demuxer tries to reorder received packets @@ -788,41 +473,36 @@ When receiving data over UDP, the demuxer tries to reorder received packets can be disabled by setting the maximum demuxing delay to zero (via the @code{max_delay} field of AVFormatContext). -When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the +When watching multi-bitrate Real-RTSP streams with @command{avplay}, the streams to display can be chosen with @code{-vst} @var{n} and @code{-ast} @var{n} for video and audio respectively, and can be switched on the fly by pressing @code{v} and @code{a}. -@subsection Examples +Example command lines: -The following examples all make use of the @command{ffplay} and -@command{ffmpeg} tools. +To watch a stream over UDP, with a max reordering delay of 0.5 seconds: -@itemize -@item -Watch a stream over UDP, with a max reordering delay of 0.5 seconds: @example -ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 +avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 @end example -@item -Watch a stream tunneled over HTTP: +To watch a stream tunneled over HTTP: + @example -ffplay -rtsp_transport http rtsp://server/video.mp4 +avplay -rtsp_transport http rtsp://server/video.mp4 @end example -@item -Send a stream in realtime to a RTSP server, for others to watch: +To send a stream in realtime to a RTSP server, for others to watch: + @example -ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp +avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp @end example -@item -Receive a stream in realtime: +To receive a stream in realtime: + @example -ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} +avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} @end example -@end itemize @section sap @@ -873,19 +553,19 @@ Example command lines follow. To broadcast a stream on the local subnet, for watching in VLC: @example -ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1 +avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1 @end example -Similarly, for watching in @command{ffplay}: +Similarly, for watching in avplay: @example -ffmpeg -re -i @var{input} -f sap sap://224.0.0.255 +avconv -re -i @var{input} -f sap sap://224.0.0.255 @end example -And for watching in @command{ffplay}, over IPv6: +And for watching in avplay, over IPv6: @example -ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4] +avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4] @end example @subsection Demuxer @@ -907,125 +587,43 @@ Example command lines follow. To play back the first stream announced on the normal SAP multicast address: @example -ffplay sap:// +avplay sap:// @end example To play back the first stream announced on one the default IPv6 SAP multicast address: @example -ffplay sap://[ff0e::2:7ffe] +avplay sap://[ff0e::2:7ffe] @end example -@section sctp - -Stream Control Transmission Protocol. - -The accepted URL syntax is: -@example -sctp://@var{host}:@var{port}[?@var{options}] -@end example - -The protocol accepts the following options: -@table @option -@item listen -If set to any value, listen for an incoming connection. Outgoing connection is done by default. - -@item max_streams -Set the maximum number of streams. By default no limit is set. -@end table - -@section srtp - -Secure Real-time Transport Protocol. - -The accepted options are: -@table @option -@item srtp_in_suite -@item srtp_out_suite -Select input and output encoding suites. - -Supported values: -@table @samp -@item AES_CM_128_HMAC_SHA1_80 -@item SRTP_AES128_CM_HMAC_SHA1_80 -@item AES_CM_128_HMAC_SHA1_32 -@item SRTP_AES128_CM_HMAC_SHA1_32 -@end table - -@item srtp_in_params -@item srtp_out_params -Set input and output encoding parameters, which are expressed by a -base64-encoded representation of a binary block. The first 16 bytes of -this binary block are used as master key, the following 14 bytes are -used as master salt. -@end table - -@section subfile - -Virtually extract a segment of a file or another stream. -The underlying stream must be seekable. - -Accepted options: -@table @option -@item start -Start offset of the extracted segment, in bytes. -@item end -End offset of the extracted segment, in bytes. -@end table - -Examples: - -Extract a chapter from a DVD VOB file (start and end sectors obtained -externally and multiplied by 2048): -@example -subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB -@end example - -Play an AVI file directly from a TAR archive: -subfile,,start,183241728,end,366490624,,:archive.tar - @section tcp -Transmission Control Protocol. +Trasmission Control Protocol. The required syntax for a TCP url is: @example tcp://@var{hostname}:@var{port}[?@var{options}] @end example -@var{options} contains a list of &-separated options of the form -@var{key}=@var{val}. - -The list of supported options follows. - @table @option -@item listen=@var{1|0} -Listen for an incoming connection. Default value is 0. -@item timeout=@var{microseconds} -Set raise error timeout, expressed in microseconds. +@item listen +Listen for an incoming connection -This option is only relevant in read mode: if no data arrived in more -than this time interval, raise error. - -@item listen_timeout=@var{microseconds} -Set listen timeout, expressed in microseconds. -@end table - -The following example shows how to setup a listening TCP connection -with @command{ffmpeg}, which is then accessed with @command{ffplay}: @example -ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen -ffplay tcp://@var{hostname}:@var{port} +avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen +avplay tcp://@var{hostname}:@var{port} @end example +@end table + @section tls Transport Layer Security (TLS) / Secure Sockets Layer (SSL) -The required syntax for a TLS/SSL url is: +The required syntax for a TLS url is: @example -tls://@var{hostname}:@var{port}[?@var{options}] +tls://@var{hostname}:@var{port} @end example The following parameters can be set via command line options @@ -1033,12 +631,11 @@ The following parameters can be set via command line options @table @option -@item ca_file, cafile=@var{filename} +@item ca_file A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS library contains a default this might not need to be specified for verification to work, but not all libraries and setups have defaults built in. -The file must be in OpenSSL PEM format. @item tls_verify=@var{1|0} If enabled, try to verify the peer that we are communicating with. @@ -1051,13 +648,13 @@ the host name is validated as well.) This is disabled by default since it requires a CA database to be provided by the caller in many cases. -@item cert_file, cert=@var{filename} +@item cert_file A file containing a certificate to use in the handshake with the peer. (When operating as server, in listen mode, this is more often required by the peer, while client certificates only are mandated in certain setups.) -@item key_file, key=@var{filename} +@item key_file A file containing the private key for the certificate. @item listen=@var{1|0} @@ -1066,46 +663,25 @@ the server role in the handshake instead of the client role. @end table -Example command lines: - -To create a TLS/SSL server that serves an input stream. - -@example -ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key} -@end example - -To play back a stream from the TLS/SSL server using @command{ffplay}: - -@example -ffplay tls://@var{hostname}:@var{port} -@end example - @section udp User Datagram Protocol. -The required syntax for an UDP URL is: +The required syntax for a UDP url is: @example udp://@var{hostname}:@var{port}[?@var{options}] @end example @var{options} contains a list of &-separated options of the form @var{key}=@var{val}. - -In case threading is enabled on the system, a circular buffer is used -to store the incoming data, which allows one to reduce loss of data due to -UDP socket buffer overruns. The @var{fifo_size} and -@var{overrun_nonfatal} options are related to this buffer. - -The list of supported options follows. +Follow the list of supported options. @table @option + @item buffer_size=@var{size} -Set the UDP maximum socket buffer size in bytes. This is used to set either -the receive or send buffer size, depending on what the socket is used for. -Default is 64KB. See also @var{fifo_size}. +set the UDP buffer size in bytes @item localport=@var{port} -Override the local UDP port to bind with. +override the local UDP port to bind with @item localaddr=@var{addr} Choose the local IP address. This is useful e.g. if sending multicast @@ -1113,13 +689,13 @@ and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface. @item pkt_size=@var{size} -Set the size in bytes of UDP packets. +set the size in bytes of UDP packets @item reuse=@var{1|0} -Explicitly allow or disallow reusing UDP sockets. +explicitly allow or disallow reusing UDP sockets @item ttl=@var{ttl} -Set the time to live value (for multicast only). +set the time to live value (for multicast only) @item connect=@var{1|0} Initialize the UDP socket with @code{connect()}. In this case, the @@ -1139,50 +715,24 @@ specified sender IP addresses. @item block=@var{address}[,@var{address}] Ignore packets sent to the multicast group from the specified sender IP addresses. - -@item fifo_size=@var{units} -Set the UDP receiving circular buffer size, expressed as a number of -packets with size of 188 bytes. If not specified defaults to 7*4096. - -@item overrun_nonfatal=@var{1|0} -Survive in case of UDP receiving circular buffer overrun. Default -value is 0. - -@item timeout=@var{microseconds} -Set raise error timeout, expressed in microseconds. - -This option is only relevant in read mode: if no data arrived in more -than this time interval, raise error. - -@item broadcast=@var{1|0} -Explicitly allow or disallow UDP broadcasting. - -Note that broadcasting may not work properly on networks having -a broadcast storm protection. @end table -@subsection Examples +Some usage examples of the udp protocol with @command{avconv} follow. -@itemize -@item -Use @command{ffmpeg} to stream over UDP to a remote endpoint: +To stream over UDP to a remote endpoint: @example -ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port} +avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port} @end example -@item -Use @command{ffmpeg} to stream in mpegts format over UDP using 188 -sized UDP packets, using a large input buffer: +To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer: @example -ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535 +avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535 @end example -@item -Use @command{ffmpeg} to receive over UDP from a remote endpoint: +To receive over UDP from a remote endpoint: @example -ffmpeg -i udp://[@var{multicast-address}]:@var{port} ... +avconv -i udp://[@var{multicast-address}]:@var{port} @end example -@end itemize @section unix diff --git a/doc/resampler.texi b/doc/resampler.texi deleted file mode 100644 index f9eef03f20..0000000000 --- a/doc/resampler.texi +++ /dev/null @@ -1,232 +0,0 @@ -@chapter Resampler Options -@c man begin RESAMPLER OPTIONS - -The audio resampler supports the following named options. - -Options may be set by specifying -@var{option} @var{value} in the -FFmpeg tools, @var{option}=@var{value} for the aresample filter, -by setting the value explicitly in the -@code{SwrContext} options or using the @file{libavutil/opt.h} API for -programmatic use. - -@table @option - -@item ich, in_channel_count -Set the number of input channels. Default value is 0. Setting this -value is not mandatory if the corresponding channel layout -@option{in_channel_layout} is set. - -@item och, out_channel_count -Set the number of output channels. Default value is 0. Setting this -value is not mandatory if the corresponding channel layout -@option{out_channel_layout} is set. - -@item uch, used_channel_count -Set the number of used input channels. Default value is 0. This option is -only used for special remapping. - -@item isr, in_sample_rate -Set the input sample rate. Default value is 0. - -@item osr, out_sample_rate -Set the output sample rate. Default value is 0. - -@item isf, in_sample_fmt -Specify the input sample format. It is set by default to @code{none}. - -@item osf, out_sample_fmt -Specify the output sample format. It is set by default to @code{none}. - -@item tsf, internal_sample_fmt -Set the internal sample format. Default value is @code{none}. -This will automatically be chosen when it is not explicitly set. - -@item icl, in_channel_layout -@item ocl, out_channel_layout -Set the input/output channel layout. - -See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils} -for the required syntax. - -@item clev, center_mix_level -Set the center mix level. It is a value expressed in deciBel, and must be -in the interval [-32,32]. - -@item slev, surround_mix_level -Set the surround mix level. It is a value expressed in deciBel, and must -be in the interval [-32,32]. - -@item lfe_mix_level -Set LFE mix into non LFE level. It is used when there is a LFE input but no -LFE output. It is a value expressed in deciBel, and must -be in the interval [-32,32]. - -@item rmvol, rematrix_volume -Set rematrix volume. Default value is 1.0. - -@item rematrix_maxval -Set maximum output value for rematrixing. -This can be used to prevent clipping vs. preventing volumn reduction -A value of 1.0 prevents cliping. - -@item flags, swr_flags -Set flags used by the converter. Default value is 0. - -It supports the following individual flags: -@table @option -@item res -force resampling, this flag forces resampling to be used even when the -input and output sample rates match. -@end table - -@item dither_scale -Set the dither scale. Default value is 1. - -@item dither_method -Set dither method. Default value is 0. - -Supported values: -@table @samp -@item rectangular -select rectangular dither -@item triangular -select triangular dither -@item triangular_hp -select triangular dither with high pass -@item lipshitz -select lipshitz noise shaping dither -@item shibata -select shibata noise shaping dither -@item low_shibata -select low shibata noise shaping dither -@item high_shibata -select high shibata noise shaping dither -@item f_weighted -select f-weighted noise shaping dither -@item modified_e_weighted -select modified-e-weighted noise shaping dither -@item improved_e_weighted -select improved-e-weighted noise shaping dither - -@end table - -@item resampler -Set resampling engine. Default value is swr. - -Supported values: -@table @samp -@item swr -select the native SW Resampler; filter options precision and cheby are not -applicable in this case. -@item soxr -select the SoX Resampler (where available); compensation, and filter options -filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this -case. -@end table - -@item filter_size -For swr only, set resampling filter size, default value is 32. - -@item phase_shift -For swr only, set resampling phase shift, default value is 10, and must be in -the interval [0,30]. - -@item linear_interp -Use Linear Interpolation if set to 1, default value is 0. - -@item cutoff -Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float -value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr -(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz). - -@item precision -For soxr only, the precision in bits to which the resampled signal will be -calculated. The default value of 20 (which, with suitable dithering, is -appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a -value of 28 gives SoX's 'Very High Quality'. - -@item cheby -For soxr only, selects passband rolloff none (Chebyshev) & higher-precision -approximation for 'irrational' ratios. Default value is 0. - -@item async -For swr only, simple 1 parameter audio sync to timestamps using stretching, -squeezing, filling and trimming. Setting this to 1 will enable filling and -trimming, larger values represent the maximum amount in samples that the data -may be stretched or squeezed for each second. -Default value is 0, thus no compensation is applied to make the samples match -the audio timestamps. - -@item first_pts -For swr only, assume the first pts should be this value. The time unit is 1 / sample rate. -This allows for padding/trimming at the start of stream. By default, no -assumption is made about the first frame's expected pts, so no padding or -trimming is done. For example, this could be set to 0 to pad the beginning with -silence if an audio stream starts after the video stream or to trim any samples -with a negative pts due to encoder delay. - -@item min_comp -For swr only, set the minimum difference between timestamps and audio data (in -seconds) to trigger stretching/squeezing/filling or trimming of the -data to make it match the timestamps. The default is that -stretching/squeezing/filling and trimming is disabled -(@option{min_comp} = @code{FLT_MAX}). - -@item min_hard_comp -For swr only, set the minimum difference between timestamps and audio data (in -seconds) to trigger adding/dropping samples to make it match the -timestamps. This option effectively is a threshold to select between -hard (trim/fill) and soft (squeeze/stretch) compensation. Note that -all compensation is by default disabled through @option{min_comp}. -The default is 0.1. - -@item comp_duration -For swr only, set duration (in seconds) over which data is stretched/squeezed -to make it match the timestamps. Must be a non-negative double float value, -default value is 1.0. - -@item max_soft_comp -For swr only, set maximum factor by which data is stretched/squeezed to make it -match the timestamps. Must be a non-negative double float value, default value -is 0. - -@item matrix_encoding -Select matrixed stereo encoding. - -It accepts the following values: -@table @samp -@item none -select none -@item dolby -select Dolby -@item dplii -select Dolby Pro Logic II -@end table - -Default value is @code{none}. - -@item filter_type -For swr only, select resampling filter type. This only affects resampling -operations. - -It accepts the following values: -@table @samp -@item cubic -select cubic -@item blackman_nuttall -select Blackman Nuttall Windowed Sinc -@item kaiser -select Kaiser Windowed Sinc -@end table - -@item kaiser_beta -For swr only, set Kaiser Window Beta value. Must be an integer in the -interval [2,16], default value is 9. - -@item output_sample_bits -For swr only, set number of used output sample bits for dithering. Must be an integer in the -interval [0,64], default value is 0, which means it's not used. - -@end table - -@c man end RESAMPLER OPTIONS diff --git a/doc/scaler.texi b/doc/scaler.texi deleted file mode 100644 index 23d6393883..0000000000 --- a/doc/scaler.texi +++ /dev/null @@ -1,127 +0,0 @@ -@anchor{scaler_options} -@chapter Scaler Options -@c man begin SCALER OPTIONS - -The video scaler supports the following named options. - -Options may be set by specifying -@var{option} @var{value} in the -FFmpeg tools. For programmatic use, they can be set explicitly in the -@code{SwsContext} options or through the @file{libavutil/opt.h} API. - -@table @option - -@anchor{sws_flags} -@item sws_flags -Set the scaler flags. This is also used to set the scaling -algorithm. Only a single algorithm should be selected. - -It accepts the following values: -@table @samp -@item fast_bilinear -Select fast bilinear scaling algorithm. - -@item bilinear -Select bilinear scaling algorithm. - -@item bicubic -Select bicubic scaling algorithm. - -@item experimental -Select experimental scaling algorithm. - -@item neighbor -Select nearest neighbor rescaling algorithm. - -@item area -Select averaging area rescaling algorithm. - -@item bicublin -Select bicubic scaling algorithm for the luma component, bilinear for -chroma components. - -@item gauss -Select Gaussian rescaling algorithm. - -@item sinc -Select sinc rescaling algorithm. - -@item lanczos -Select lanczos rescaling algorithm. - -@item spline -Select natural bicubic spline rescaling algorithm. - -@item print_info -Enable printing/debug logging. - -@item accurate_rnd -Enable accurate rounding. - -@item full_chroma_int -Enable full chroma interpolation. - -@item full_chroma_inp -Select full chroma input. - -@item bitexact -Enable bitexact output. -@end table - -@item srcw -Set source width. - -@item srch -Set source height. - -@item dstw -Set destination width. - -@item dsth -Set destination height. - -@item src_format -Set source pixel format (must be expressed as an integer). - -@item dst_format -Set destination pixel format (must be expressed as an integer). - -@item src_range -Select source range. - -@item dst_range -Select destination range. - -@item param0, param1 -Set scaling algorithm parameters. The specified values are specific of -some scaling algorithms and ignored by others. The specified values -are floating point number values. - -@item sws_dither -Set the dithering algorithm. Accepts one of the following -values. Default value is @samp{auto}. - -@table @samp -@item auto -automatic choice - -@item none -no dithering - -@item bayer -bayer dither - -@item ed -error diffusion dither - -@item a_dither -arithmetic dither, based using addition - -@item x_dither -arithmetic dither, based using xor (more random/less apparent patterning that -a_dither). - -@end table - -@end table - -@c man end SCALER OPTIONS diff --git a/doc/snow.txt b/doc/snow.txt deleted file mode 100644 index 080a33b38f..0000000000 --- a/doc/snow.txt +++ /dev/null @@ -1,638 +0,0 @@ -============================================= -Snow Video Codec Specification Draft 20080110 -============================================= - -Introduction: -============= -This specification describes the Snow bitstream syntax and semantics as -well as the formal Snow decoding process. - -The decoding process is described precisely and any compliant decoder -MUST produce the exact same output for a spec-conformant Snow stream. -For encoding, though, any process which generates a stream compliant to -the syntactical and semantic requirements and which is decodable by -the process described in this spec shall be considered a conformant -Snow encoder. - -Definitions: -============ - -MUST the specific part must be done to conform to this standard -SHOULD it is recommended to be done that way, but not strictly required - -ilog2(x) is the rounded down logarithm of x with basis 2 -ilog2(0) = 0 - -Type definitions: -================= - -b 1-bit range coded -u unsigned scalar value range coded -s signed scalar value range coded - - -Bitstream syntax: -================= - -frame: - header - prediction - residual - -header: - keyframe b MID_STATE - if(keyframe || always_reset) - reset_contexts - if(keyframe){ - version u header_state - always_reset b header_state - temporal_decomposition_type u header_state - temporal_decomposition_count u header_state - spatial_decomposition_count u header_state - colorspace_type u header_state - if (nb_planes > 2) { - chroma_h_shift u header_state - chroma_v_shift u header_state - } - spatial_scalability b header_state - max_ref_frames-1 u header_state - qlogs - } - if(!keyframe){ - update_mc b header_state - if(update_mc){ - for(plane=0; planehtaps/2; i; i--) - |hcoeff[i]| u header_state - } - } - update_qlogs b header_state - if(update_qlogs){ - spatial_decomposition_count u header_state - qlogs - } - } - - spatial_decomposition_type s header_state - qlog s header_state - mv_scale s header_state - qbias s header_state - block_max_depth s header_state - -qlogs: - for(plane=0; planelevel + 2*top->level + topleft->level + topright->level - leaf b block_state[4 + s_context] - } - if(level==max_block_depth || leaf){ - intra b block_state[1 + left->intra + top->intra] - if(intra){ - y_diff s block_state[32] - cb_diff s block_state[64] - cr_diff s block_state[96] - }else{ - ref_context= ilog2(2*left->ref) + ilog2(2*top->ref) - if(ref_frames > 1) - ref u block_state[128 + 1024 + 32*ref_context] - mx_context= ilog2(2*abs(left->mx - top->mx)) - my_context= ilog2(2*abs(left->my - top->my)) - mvx_diff s block_state[128 + 32*(mx_context + 16*!!ref)] - mvy_diff s block_state[128 + 32*(my_context + 16*!!ref)] - } - }else{ - block(level+1) - block(level+1) - block(level+1) - block(level+1) - } - } - - -residual: - residual2(luma) - if (nb_planes > 2) { - residual2(chroma_cr) - residual2(chroma_cb) - } - -residual2: - for(level=0; level0 and <10 - -hcoeff - half pel interpolation filter coefficients, hcoeff[0] are the 2 middle - coefficients [1] are the next outer ones and so on, resulting in a filter - like: ...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... - the sign of the coefficients is not explicitly stored but alternates - after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... - hcoeff[0] is not explicitly stored but found by subtracting the sum - of all stored coefficients with signs from 32 - hcoeff[0]= 32 - hcoeff[1] - hcoeff[2] - ... - a good choice for hcoeff and htaps is - htaps= 6 - hcoeff={40,-10,2} - an alternative which requires more computations at both encoder and - decoder side and may or may not be better is - htaps= 8 - hcoeff={42,-14,6,-2} - - -ref_frames - minimum of the number of available reference frames and max_ref_frames - for example the first frame after a key frame always has ref_frames=1 - -spatial_decomposition_type - wavelet type - 0 is a 9/7 symmetric compact integer wavelet - 1 is a 5/3 symmetric compact integer wavelet - others are reserved - stored as delta from last, last is reset to 0 if always_reset || keyframe - -qlog - quality (logarthmic quantizer scale) - stored as delta from last, last is reset to 0 if always_reset || keyframe - -mv_scale - stored as delta from last, last is reset to 0 if always_reset || keyframe - FIXME check that everything works fine if this changes between frames - -qbias - dequantization bias - stored as delta from last, last is reset to 0 if always_reset || keyframe - -block_max_depth - maximum depth of the block tree - stored as delta from last, last is reset to 0 if always_reset || keyframe - -quant_table - quantiztation table - - -Highlevel bitstream structure: -============================= - -------------------------------------------- -| Header | - -------------------------------------------- -| ------------------------------------ | -| | Block0 | | -| | split? | | -| | yes no | | -| | ......... intra? | | -| | : Block01 : yes no | | -| | : Block02 : ....... .......... | | -| | : Block03 : : y DC : : ref index: | | -| | : Block04 : : cb DC : : motion x : | | -| | ......... : cr DC : : motion y : | | -| | ....... .......... | | -| ------------------------------------ | -| ------------------------------------ | -| | Block1 | | -| ... | - -------------------------------------------- -| ------------ ------------ ------------ | -|| Y subbands | | Cb subbands| | Cr subbands|| -|| --- --- | | --- --- | | --- --- || -|| |LL0||HL0| | | |LL0||HL0| | | |LL0||HL0| || -|| --- --- | | --- --- | | --- --- || -|| --- --- | | --- --- | | --- --- || -|| |LH0||HH0| | | |LH0||HH0| | | |LH0||HH0| || -|| --- --- | | --- --- | | --- --- || -|| --- --- | | --- --- | | --- --- || -|| |HL1||LH1| | | |HL1||LH1| | | |HL1||LH1| || -|| --- --- | | --- --- | | --- --- || -|| --- --- | | --- --- | | --- --- || -|| |HH1||HL2| | | |HH1||HL2| | | |HH1||HL2| || -|| ... | | ... | | ... || -| ------------ ------------ ------------ | - -------------------------------------------- - -Decoding process: -================= - - ------------ - | | - | Subbands | - ------------ | | - | | ------------ - | Intra DC | | - | | LL0 subband prediction - ------------ | - \ Dequantizaton - ------------------- \ | -| Reference frames | \ IDWT -| ------- ------- | Motion \ | -||Frame 0| |Frame 1|| Compensation . OBMC v ------- -| ------- ------- | --------------. \------> + --->|Frame n|-->output -| ------- ------- | ------- -||Frame 2| |Frame 3||<----------------------------------/ -| ... | - ------------------- - - -Range Coder: -============ - -Binary Range Coder: -------------------- -The implemented range coder is an adapted version based upon "Range encoding: -an algorithm for removing redundancy from a digitised message." by G. N. N. -Martin. -The symbols encoded by the Snow range coder are bits (0|1). The -associated probabilities are not fix but change depending on the symbol mix -seen so far. - - -bit seen | new state ----------+----------------------------------------------- - 0 | 256 - state_transition_table[256 - old_state]; - 1 | state_transition_table[ old_state]; - -state_transition_table = { - 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, - 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, - 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, - 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, - 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, - 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, -104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, -119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, -134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, -150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, -165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, -180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, -195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, -210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, -226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, -241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0}; - -FIXME - - -Range Coding of integers: -------------------------- -FIXME - - -Neighboring Blocks: -=================== -left and top are set to the respective blocks unless they are outside of -the image in which case they are set to the Null block - -top-left is set to the top left block unless it is outside of the image in -which case it is set to the left block - -if this block has no larger parent block or it is at the left side of its -parent block and the top right block is not outside of the image then the -top right block is used for top-right else the top-left block is used - -Null block -y,cb,cr are 128 -level, ref, mx and my are 0 - - -Motion Vector Prediction: -========================= -1. the motion vectors of all the neighboring blocks are scaled to -compensate for the difference of reference frames - -scaled_mv= (mv * (256 * (current_reference+1) / (mv.reference+1)) + 128)>>8 - -2. the median of the scaled left, top and top-right vectors is used as -motion vector prediction - -3. the used motion vector is the sum of the predictor and - (mvx_diff, mvy_diff)*mv_scale - - -Intra DC Predicton: -====================== -the luma and chroma values of the left block are used as predictors - -the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff -to reverse this in the decoder apply the following: -block[y][x].dc[0] = block[y][x-1].dc[0] + y_diff; -block[y][x].dc[1] = block[y][x-1].dc[1] + cb_diff; -block[y][x].dc[2] = block[y][x-1].dc[2] + cr_diff; -block[*][-1].dc[*]= 128; - - -Motion Compensation: -==================== - -Halfpel interpolation: ----------------------- -halfpel interpolation is done by convolution with the halfpel filter stored -in the header: - -horizontal halfpel samples are found by -H1[y][x] = hcoeff[0]*(F[y][x ] + F[y][x+1]) - + hcoeff[1]*(F[y][x-1] + F[y][x+2]) - + hcoeff[2]*(F[y][x-2] + F[y][x+3]) - + ... -h1[y][x] = (H1[y][x] + 32)>>6; - -vertical halfpel samples are found by -H2[y][x] = hcoeff[0]*(F[y ][x] + F[y+1][x]) - + hcoeff[1]*(F[y-1][x] + F[y+2][x]) - + ... -h2[y][x] = (H2[y][x] + 32)>>6; - -vertical+horizontal halfpel samples are found by -H3[y][x] = hcoeff[0]*(H2[y][x ] + H2[y][x+1]) - + hcoeff[1]*(H2[y][x-1] + H2[y][x+2]) - + ... -H3[y][x] = hcoeff[0]*(H1[y ][x] + H1[y+1][x]) - + hcoeff[1]*(H1[y+1][x] + H1[y+2][x]) - + ... -h3[y][x] = (H3[y][x] + 2048)>>12; - - - F H1 F - | | | - | | | - | | | - F H1 F - | | | - | | | - | | | - F-------F-------F-> H1<-F-------F-------F - v v v - H2 H3 H2 - ^ ^ ^ - F-------F-------F-> H1<-F-------F-------F - | | | - | | | - | | | - F H1 F - | | | - | | | - | | | - F H1 F - - -unavailable fullpel samples (outside the picture for example) shall be equal -to the closest available fullpel sample - - -Smaller pel interpolation: --------------------------- -if diag_mc is set then points which lie on a line between 2 vertically, -horiziontally or diagonally adjacent halfpel points shall be interpolated -linearls with rounding to nearest and halfway values rounded up. -points which lie on 2 diagonals at the same time should only use the one -diagonal not containing the fullpel point - - - - F-->O---q---O<--h1->O---q---O<--F - v \ / v \ / v - O O O O O O O - | / | \ | - q q q q q - | / | \ | - O O O O O O O - ^ / \ ^ / \ ^ - h2-->O---q---O<--h3->O---q---O<--h2 - v \ / v \ / v - O O O O O O O - | \ | / | - q q q q q - | \ | / | - O O O O O O O - ^ / \ ^ / \ ^ - F-->O---q---O<--h1->O---q---O<--F - - - -the remaining points shall be bilinearly interpolated from the -up to 4 surrounding halfpel and fullpel points, again rounding should be to -nearest and halfway values rounded up - -compliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chroma -interpolation at least - - -Overlapped block motion compensation: -------------------------------------- -FIXME - -LL band prediction: -=================== -Each sample in the LL0 subband is predicted by the median of the left, top and -left+top-topleft samples, samples outside the subband shall be considered to -be 0. To reverse this prediction in the decoder apply the following. -for(y=0; y|-------+-------|->|-------+-------|->| L1 | H1 |->... -| | | | | | | | | | | | -| LH1 | HH1 | | LH1 | HH1 | | LH1 | HH1 | | | | -| | | | | | | | | | | | - --------------- --------------- --------------- --------------- - - -1D Filter: ----------- -1. interleave the samples of the low and high frequency subbands like -s={L0, H0, L1, H1, L2, H2, L3, H3, ... } -note, this can end with a L or a H, the number of elements shall be w -s[-1] shall be considered equivalent to s[1 ] -s[w ] shall be considered equivalent to s[w-2] - -2. perform the lifting steps in order as described below - -5/3 Integer filter: -1. s[i] -= (s[i-1] + s[i+1] + 2)>>2; for all even i < w -2. s[i] += (s[i-1] + s[i+1] )>>1; for all odd i < w - -\ | /|\ | /|\ | /|\ | /|\ - \|/ | \|/ | \|/ | \|/ | - + | + | + | + | -1/4 - /|\ | /|\ | /|\ | /|\ | -/ | \|/ | \|/ | \|/ | \|/ - | + | + | + | + +1/2 - - -Snow's 9/7 Integer filter: -1. s[i] -= (3*(s[i-1] + s[i+1]) + 4)>>3; for all even i < w -2. s[i] -= s[i-1] + s[i+1] ; for all odd i < w -3. s[i] += ( s[i-1] + s[i+1] + 4*s[i] + 8)>>4; for all even i < w -4. s[i] += (3*(s[i-1] + s[i+1]) )>>1; for all odd i < w - -\ | /|\ | /|\ | /|\ | /|\ - \|/ | \|/ | \|/ | \|/ | - + | + | + | + | -3/8 - /|\ | /|\ | /|\ | /|\ | -/ | \|/ | \|/ | \|/ | \|/ - (| + (| + (| + (| + -1 -\ + /|\ + /|\ + /|\ + /|\ +1/4 - \|/ | \|/ | \|/ | \|/ | - + | + | + | + | +1/16 - /|\ | /|\ | /|\ | /|\ | -/ | \|/ | \|/ | \|/ | \|/ - | + | + | + | + +3/2 - -optimization tips: -following are exactly identical -(3a)>>1 == a + (a>>1) -(a + 4b + 8)>>4 == ((a>>2) + b + 2)>>2 - -16bit implementation note: -The IDWT can be implemented with 16bits, but this requires some care to -prevent overflows, the following list, lists the minimum number of bits needed -for some terms -1. lifting step -A= s[i-1] + s[i+1] 16bit -3*A + 4 18bit -A + (A>>1) + 2 17bit - -3. lifting step -s[i-1] + s[i+1] 17bit - -4. lifiting step -3*(s[i-1] + s[i+1]) 17bit - - -TODO: -===== -Important: -finetune initial contexts -flip wavelet? -try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients -try the MV length as context for coding the residual coefficients -use extradata for stuff which is in the keyframes now? -the MV median predictor is patented IIRC -implement per picture halfpel interpolation -try different range coder state transition tables for different contexts - -Not Important: -compare the 6 tap and 8 tap hpel filters (psnr/bitrate and subjective quality) -spatial_scalability b vs u (!= 0 breaks syntax anyway so we can add a u later) - - -Credits: -======== -Michael Niedermayer -Loren Merritt - - -Copyright: -========== -GPL + GFDL + whatever is needed to make this a RFC diff --git a/doc/soc.txt b/doc/soc.txt new file mode 100644 index 0000000000..89728b5201 --- /dev/null +++ b/doc/soc.txt @@ -0,0 +1,24 @@ +Google Summer of Code and similar project guidelines + +Summer of Code is a project by Google in which students are paid to implement +some nice new features for various participating open source projects ... + +This text is a collection of things to take care of for the next soc as +it's a little late for this year's soc (2006). + +The Goal: +Our goal in respect to soc is and must be of course exactly one thing and +that is to improve Libav, to reach this goal, code must +* conform to the development policy and patch submission guidelines +* must improve Libav somehow (faster, smaller, "better", + more codecs supported, fewer bugs, cleaner, ...) + +for mentors and other developers to help students to reach that goal it is +essential that changes to their codebase are publicly visible, clean and +easy reviewable that again leads us to: +* use of a revision control system like git +* separation of cosmetic from non-cosmetic changes (this is almost entirely + ignored by mentors and students in soc 2006 which might lead to a surprise + when the code will be reviewed at the end before a possible inclusion in + Libav, individual changes were generally not reviewable due to cosmetics). +* frequent commits, so that comments can be provided early diff --git a/doc/swresample.txt b/doc/swresample.txt deleted file mode 100644 index 2d192a394e..0000000000 --- a/doc/swresample.txt +++ /dev/null @@ -1,46 +0,0 @@ - The official guide to swresample for confused developers. - ========================================================= - -Current (simplified) Architecture: ---------------------------------- - Input - v - __________________/|\___________ - / | \ - / input sample format convert v - / | ___________/ - | |/ - | v - | ___________/|\___________ _____________ - | / | \ | | - | Rematrix | resample <---->| Buffers | - | \___________ | ___________/ |_____________| - v \|/ -Special Converter v - v ___________/|\___________ _____________ - | / | \ | | - | Rematrix | resample <---->| Buffers | - | \___________ | ___________/ |_____________| - | \|/ - | v - | |\___________ - \ | \ - \ output sample format convert v - \_________________ | ___________/ - \|/ - v - Output - -Planar/Packed conversion is done when needed during sample format conversion. -Every step can be skipped without memcpy when it is not needed. -Either Resampling and Rematrixing can be performed first depending on which -way it is faster. -The Buffers are needed for resampling due to resamplng being a process that -requires future and past data, it thus also introduces inevitably a delay when -used. -Internally 32bit float and 16bit int is supported currently, other formats can -easily be added. -Externally all sample formats in packed and planar configuration are supported -It's also trivial to add special converters for common cases. -If only sample format and/or packed/planar conversion is needed, it -is performed from input to output directly in a single pass with no intermediates. diff --git a/doc/t2h.init b/doc/t2h.init index e7760f4527..a42637ae0a 100644 --- a/doc/t2h.init +++ b/doc/t2h.init @@ -1,6 +1,6 @@ # no horiz rules between sections -$end_section = \&FFmpeg_end_section; -sub FFmpeg_end_section($$) +$end_section = \&Libav_end_section; +sub Libav_end_section($$) { } @@ -8,16 +8,132 @@ $EXTRA_HEAD = ' '; -$CSS_LINES = $ENV{"FFMPEG_CSS"} || < +$CSS_LINES = $ENV{"LIBAV_CSS"} || < + + EOT -my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <
'; $SMALL_RULE = ''; $BODYTEXT = ''; -$print_page_foot = \&FFmpeg_print_page_foot; -sub FFmpeg_print_page_foot($$) +$print_page_foot = \&Libav_print_page_foot; +sub Libav_print_page_foot($$) { my $fh = shift; my $program_string = defined &T2H_DEFAULT_program_string ? T2H_DEFAULT_program_string() : program_string(); print $fh '
' . "\n"; print $fh '' . $program_string; - print $fh "
\n"; + print $fh "\n"; } -$float = \&FFmpeg_float; +$float = \&Libav_float; -sub FFmpeg_float($$$$) +sub Libav_float($$$$) { my $text = shift; my $float = shift; @@ -65,8 +181,8 @@ sub FFmpeg_float($$$$) return '
' . "$label\n" . $text . '
'; } -$print_page_head = \&FFmpeg_print_page_head; -sub FFmpeg_print_page_head($$) +$print_page_head = \&Libav_print_page_head; +sub Libav_print_page_head($$) { my $fh = shift; my $longtitle = "$Texi2HTML::THISDOC{'fulltitle_no_texi'}"; @@ -79,7 +195,7 @@ sub FFmpeg_print_page_head($$) my $encoding = ''; $encoding = "" if (defined($ENCODING) and ($ENCODING ne '')); $longtitle =~ s/Documentation.*//g; - $longtitle = "FFmpeg documentation : " . $longtitle; + $longtitle = "Libav documentation : " . $longtitle; print $fh < @@ -93,6 +209,8 @@ $Texi2HTML::THISDOC{program_authors} $description + + $encoding $CSS_LINES @@ -100,9 +218,6 @@ $TEMPLATE_HEADER EOT } -# declare encoding in header -$IN_ENCODING = $ENCODING = "utf-8"; - # no navigation elements $SECTION_NAVIGATION = 0; # the same for texi2html 5.0 diff --git a/doc/texi2pod.pl b/doc/texi2pod.pl old mode 100644 new mode 100755 index 94e011e42b..54d57172d5 --- a/doc/texi2pod.pl +++ b/doc/texi2pod.pl @@ -27,9 +27,9 @@ use warnings; $output = 0; $skipping = 0; -%chapters = (); -@chapters_sequence = (); -$chapter = ""; +%sects = (); +@sects_sequence = (); +$section = ""; @icstack = (); @endwstack = (); @skstack = (); @@ -116,24 +116,18 @@ INF: while(<$inf>) { die "cannot open $1: $!\n"; }; - /^\@chapter\s+([A-Za-z ]+)/ and do { - # close old chapter - $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name); - - # start new chapter - $chapter_name = $1, push (@chapters_sequence, $chapter_name) unless $skipping; - $chapters{$chapter_name} = "" unless exists $chapters{$chapter_name}; - $chapter = ""; - $output = 1; + # Look for blocks surrounded by @c man begin SECTION ... @c man end. + # This really oughta be @ifman ... @end ifman and the like, but such + # would require rev'ing all other Texinfo translators. + /^\@c\s+man\s+begin\s+([A-Za-z ]+)/ and $sect = $1, push (@sects_sequence, $sect), $output = 1, next; + /^\@c\s+man\s+end/ and do { + $sects{$sect} = "" unless exists $sects{$sect}; + $sects{$sect} .= postprocess($section); + $section = ""; + $output = 0; next; }; - /^\@bye/ and do { - # close old chapter - $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name); - last INF; - }; - # handle variables /^\@set\s+([a-zA-Z0-9_-]+)\s*(.*)$/ and do { $defs{$1} = $2; @@ -156,14 +150,14 @@ INF: while(<$inf>) { # Ignore @end foo, where foo is not an operation which may # cause us to skip, if we are presently skipping. my $ended = $1; - next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/; + next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex)$/; die "\@end $ended without \@$ended at line $.\n" unless defined $endw; die "\@$endw ended by \@end $ended at line $.\n" unless $ended eq $endw; $endw = pop @endwstack; - if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) { + if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex)$/) { $skipping = pop @skstack; next; } elsif ($ended =~ /^(?:example|smallexample|display)$/) { @@ -196,11 +190,11 @@ INF: while(<$inf>) { next; }; - /^\@(ignore|menu|iftex|ifhtml|ifnothtml)\b/ and do { + /^\@(ignore|menu|iftex)\b/ and do { push @endwstack, $endw; push @skstack, $skipping; $endw = $1; - $skipping = $endw !~ /ifnothtml/; + $skipping = 1; next; }; @@ -217,6 +211,7 @@ INF: while(<$inf>) { s/\@TeX\{\}/TeX/g; s/\@pounds\{\}/\#/g; s/\@minus(?:\{\})?/-/g; + s/\\,/,/g; # Now the ones that have to be replaced by special escapes # (which will be turned back into text by unmunge()) @@ -274,7 +269,7 @@ INF: while(<$inf>) { push @icstack, $ic; $endw = $1; $ic = $2; - $ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env|command)/B/; + $ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env)/B/; $ic =~ s/\@(?:code|kbd)/C/; $ic =~ s/\@(?:dfn|var|emph|cite|i)/I/; $ic =~ s/\@(?:file)/F/; @@ -309,7 +304,7 @@ INF: while(<$inf>) { $columns =~ s/\@tab//; $_ = $columns; - $chapter =~ s/$//; + $section =~ s/$//; }; /^\@itemx?\s*(.+)?$/ and do { @@ -323,7 +318,7 @@ INF: while(<$inf>) { } }; - $chapter .= $shift.$_."\n"; + $section .= $shift.$_."\n"; } # End of current file. close($inf); @@ -332,18 +327,18 @@ $inf = pop @instack; die "No filename or title\n" unless defined $fn && defined $tl; -$chapters{NAME} = "$fn \- $tl\n"; -$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES}; +$sects{NAME} = "$fn \- $tl\n"; +$sects{FOOTNOTES} .= "=back\n" if exists $sects{FOOTNOTES}; -# always use utf8 -print "=encoding utf8\n\n"; - -unshift @chapters_sequence, "NAME"; -for $chapter (@chapters_sequence) { - if (exists $chapters{$chapter}) { - $head = uc($chapter); +unshift @sects_sequence, "NAME"; +for $sect (@sects_sequence) { + # always use utf8 + print "=encoding utf8\n"; + if(exists $sects{$sect}) { + $head = $sect; + $head =~ s/SEEALSO/SEE ALSO/; print "=head1 $head\n\n"; - print scalar unmunge ($chapters{$chapter}); + print scalar unmunge ($sects{$sect}); print "\n"; } } @@ -388,8 +383,7 @@ sub postprocess s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g; s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g; s/;\s+\@pxref\{(?:[^\}]*)\}//g; - s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/B<$1>/g; - s/\@ref\{([^\}]*)\}/B<$1>/g; + s/\@ref\{([^\}]*)\}/$1/g; s/\@noindent\s*//g; s/\@refill//g; s/\@gol//g; @@ -398,7 +392,7 @@ sub postprocess # @uref can take one, two, or three arguments, with different # semantics each time. @url and @email are just like @uref with # one argument, for our purposes. - s/\@(?:uref|url|email)\{([^\},]*),?[^\}]*\}/<B<$1>>/g; + s/\@(?:uref|url|email)\{([^\},]*)\}/<B<$1>>/g; s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g; s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g; @@ -442,13 +436,13 @@ sub unmunge sub add_footnote { - unless (exists $chapters{FOOTNOTES}) { - $chapters{FOOTNOTES} = "\n=over 4\n\n"; + unless (exists $sects{FOOTNOTES}) { + $sects{FOOTNOTES} = "\n=over 4\n\n"; } - $chapters{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++; - $chapters{FOOTNOTES} .= $_[0]; - $chapters{FOOTNOTES} .= "\n\n"; + $sects{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++; + $sects{FOOTNOTES} .= $_[0]; + $sects{FOOTNOTES} .= "\n\n"; } # stolen from Symbol.pm diff --git a/doc/texidep.pl b/doc/texidep.pl deleted file mode 100644 index 099690378e..0000000000 --- a/doc/texidep.pl +++ /dev/null @@ -1,32 +0,0 @@ -#! /usr/bin/env perl - -# This script will print the dependency of a Texinfo file to stdout. -# texidep.pl - -use warnings; -use strict; - -die unless @ARGV == 3; - -my ($src_path, $root, $target) = @ARGV; - -sub print_deps { - my ($file, $deps) = @_; - $deps->{$file} = 1; - - open(my $fh, "<", "$file") or die "Cannot open file '$file': $!"; - while (<$fh>) { - if (my ($i) = /^\@(?:verbatim)?include\s+(\S+)/) { - die "Circular dependency found in file $root\n" if exists $deps->{"doc/$1"}; - print "$target: doc/$1\n"; - - # skip looking for config.texi dependencies, since it has - # none, and is not located in the source tree - if ("$1" ne "config.texi") { - print_deps("$src_path/doc/$1", {%$deps}); - } - } - } -} - -print_deps($root, {}); diff --git a/doc/utils.texi b/doc/utils.texi deleted file mode 100644 index 5abfb0ccc6..0000000000 --- a/doc/utils.texi +++ /dev/null @@ -1,1064 +0,0 @@ -@chapter Syntax -@c man begin SYNTAX - -This section documents the syntax and formats employed by the FFmpeg -libraries and tools. - -@anchor{quoting_and_escaping} -@section Quoting and escaping - -FFmpeg adopts the following quoting and escaping mechanism, unless -explicitly specified. The following rules are applied: - -@itemize -@item -@code{'} and @code{\} are special characters (respectively used for -quoting and escaping). In addition to them, there might be other -special characters depending on the specific syntax where the escaping -and quoting are employed. - -@item -A special character is escaped by prefixing it with a '\'. - -@item -All characters enclosed between '' are included literally in the -parsed string. The quote character @code{'} itself cannot be quoted, -so you may need to close the quote and escape it. - -@item -Leading and trailing whitespaces, unless escaped or quoted, are -removed from the parsed string. -@end itemize - -Note that you may need to add a second level of escaping when using -the command line or a script, which depends on the syntax of the -adopted shell language. - -The function @code{av_get_token} defined in -@file{libavutil/avstring.h} can be used to parse a token quoted or -escaped according to the rules defined above. - -The tool @file{tools/ffescape} in the FFmpeg source tree can be used -to automatically quote or escape a string in a script. - -@subsection Examples - -@itemize -@item -Escape the string @code{Crime d'Amour} containing the @code{'} special -character: -@example -Crime d\'Amour -@end example - -@item -The string above contains a quote, so the @code{'} needs to be escaped -when quoting it: -@example -'Crime d'\''Amour' -@end example - -@item -Include leading or trailing whitespaces using quoting: -@example -' this string starts and ends with whitespaces ' -@end example - -@item -Escaping and quoting can be mixed together: -@example -' The string '\'string\'' is a string ' -@end example - -@item -To include a literal @code{\} you can use either escaping or quoting: -@example -'c:\foo' can be written as c:\\foo -@end example -@end itemize - -@anchor{date syntax} -@section Date - -The accepted syntax is: -@example -[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z] -now -@end example - -If the value is "now" it takes the current time. - -Time is local time unless Z is appended, in which case it is -interpreted as UTC. -If the year-month-day part is not specified it takes the current -year-month-day. - -@anchor{time duration syntax} -@section Time duration - -There are two accepted syntaxes for expressing time duration. - -@example -[-][@var{HH}:]@var{MM}:@var{SS}[.@var{m}...] -@end example - -@var{HH} expresses the number of hours, @var{MM} the number of minutes -for a maximum of 2 digits, and @var{SS} the number of seconds for a -maximum of 2 digits. The @var{m} at the end expresses decimal value for -@var{SS}. - -@emph{or} - -@example -[-]@var{S}+[.@var{m}...] -@end example - -@var{S} expresses the number of seconds, with the optional decimal part -@var{m}. - -In both expressions, the optional @samp{-} indicates negative duration. - -@subsection Examples - -The following examples are all valid time duration: - -@table @samp -@item 55 -55 seconds - -@item 12:03:45 -12 hours, 03 minutes and 45 seconds - -@item 23.189 -23.189 seconds -@end table - -@anchor{video size syntax} -@section Video size -Specify the size of the sourced video, it may be a string of the form -@var{width}x@var{height}, or the name of a size abbreviation. - -The following abbreviations are recognized: -@table @samp -@item ntsc -720x480 -@item pal -720x576 -@item qntsc -352x240 -@item qpal -352x288 -@item sntsc -640x480 -@item spal -768x576 -@item film -352x240 -@item ntsc-film -352x240 -@item sqcif -128x96 -@item qcif -176x144 -@item cif -352x288 -@item 4cif -704x576 -@item 16cif -1408x1152 -@item qqvga -160x120 -@item qvga -320x240 -@item vga -640x480 -@item svga -800x600 -@item xga -1024x768 -@item uxga -1600x1200 -@item qxga -2048x1536 -@item sxga -1280x1024 -@item qsxga -2560x2048 -@item hsxga -5120x4096 -@item wvga -852x480 -@item wxga -1366x768 -@item wsxga -1600x1024 -@item wuxga -1920x1200 -@item woxga -2560x1600 -@item wqsxga -3200x2048 -@item wquxga -3840x2400 -@item whsxga -6400x4096 -@item whuxga -7680x4800 -@item cga -320x200 -@item ega -640x350 -@item hd480 -852x480 -@item hd720 -1280x720 -@item hd1080 -1920x1080 -@item 2k -2048x1080 -@item 2kflat -1998x1080 -@item 2kscope -2048x858 -@item 4k -4096x2160 -@item 4kflat -3996x2160 -@item 4kscope -4096x1716 -@item nhd -640x360 -@item hqvga -240x160 -@item wqvga -400x240 -@item fwqvga -432x240 -@item hvga -480x320 -@item qhd -960x540 -@end table - -@anchor{video rate syntax} -@section Video rate - -Specify the frame rate of a video, expressed as the number of frames -generated per second. It has to be a string in the format -@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float -number or a valid video frame rate abbreviation. - -The following abbreviations are recognized: -@table @samp -@item ntsc -30000/1001 -@item pal -25/1 -@item qntsc -30000/1001 -@item qpal -25/1 -@item sntsc -30000/1001 -@item spal -25/1 -@item film -24/1 -@item ntsc-film -24000/1001 -@end table - -@anchor{ratio syntax} -@section Ratio - -A ratio can be expressed as an expression, or in the form -@var{numerator}:@var{denominator}. - -Note that a ratio with infinite (1/0) or negative value is -considered valid, so you should check on the returned value if you -want to exclude those values. - -The undefined value can be expressed using the "0:0" string. - -@anchor{color syntax} -@section Color - -It can be the name of a color as defined below (case insensitive match) or a -@code{[0x|#]RRGGBB[AA]} sequence, possibly followed by @@ and a string -representing the alpha component. - -The alpha component may be a string composed by "0x" followed by an -hexadecimal number or a decimal number between 0.0 and 1.0, which -represents the opacity value (@samp{0x00} or @samp{0.0} means completely -transparent, @samp{0xff} or @samp{1.0} completely opaque). If the alpha -component is not specified then @samp{0xff} is assumed. - -The string @samp{random} will result in a random color. - -The following names of colors are recognized: -@table @samp -@item AliceBlue -0xF0F8FF -@item AntiqueWhite -0xFAEBD7 -@item Aqua -0x00FFFF -@item Aquamarine -0x7FFFD4 -@item Azure -0xF0FFFF -@item Beige -0xF5F5DC -@item Bisque -0xFFE4C4 -@item Black -0x000000 -@item BlanchedAlmond -0xFFEBCD -@item Blue -0x0000FF -@item BlueViolet -0x8A2BE2 -@item Brown -0xA52A2A -@item BurlyWood -0xDEB887 -@item CadetBlue -0x5F9EA0 -@item Chartreuse -0x7FFF00 -@item Chocolate -0xD2691E -@item Coral -0xFF7F50 -@item CornflowerBlue -0x6495ED -@item Cornsilk -0xFFF8DC -@item Crimson -0xDC143C -@item Cyan -0x00FFFF -@item DarkBlue -0x00008B -@item DarkCyan -0x008B8B -@item DarkGoldenRod -0xB8860B -@item DarkGray -0xA9A9A9 -@item DarkGreen -0x006400 -@item DarkKhaki -0xBDB76B -@item DarkMagenta -0x8B008B -@item DarkOliveGreen -0x556B2F -@item Darkorange -0xFF8C00 -@item DarkOrchid -0x9932CC -@item DarkRed -0x8B0000 -@item DarkSalmon -0xE9967A -@item DarkSeaGreen -0x8FBC8F -@item DarkSlateBlue -0x483D8B -@item DarkSlateGray -0x2F4F4F -@item DarkTurquoise -0x00CED1 -@item DarkViolet -0x9400D3 -@item DeepPink -0xFF1493 -@item DeepSkyBlue -0x00BFFF -@item DimGray -0x696969 -@item DodgerBlue -0x1E90FF -@item FireBrick -0xB22222 -@item FloralWhite -0xFFFAF0 -@item ForestGreen -0x228B22 -@item Fuchsia -0xFF00FF -@item Gainsboro -0xDCDCDC -@item GhostWhite -0xF8F8FF -@item Gold -0xFFD700 -@item GoldenRod -0xDAA520 -@item Gray -0x808080 -@item Green -0x008000 -@item GreenYellow -0xADFF2F -@item HoneyDew -0xF0FFF0 -@item HotPink -0xFF69B4 -@item IndianRed -0xCD5C5C -@item Indigo -0x4B0082 -@item Ivory -0xFFFFF0 -@item Khaki -0xF0E68C -@item Lavender -0xE6E6FA -@item LavenderBlush -0xFFF0F5 -@item LawnGreen -0x7CFC00 -@item LemonChiffon -0xFFFACD -@item LightBlue -0xADD8E6 -@item LightCoral -0xF08080 -@item LightCyan -0xE0FFFF -@item LightGoldenRodYellow -0xFAFAD2 -@item LightGreen -0x90EE90 -@item LightGrey -0xD3D3D3 -@item LightPink -0xFFB6C1 -@item LightSalmon -0xFFA07A -@item LightSeaGreen -0x20B2AA -@item LightSkyBlue -0x87CEFA -@item LightSlateGray -0x778899 -@item LightSteelBlue -0xB0C4DE -@item LightYellow -0xFFFFE0 -@item Lime -0x00FF00 -@item LimeGreen -0x32CD32 -@item Linen -0xFAF0E6 -@item Magenta -0xFF00FF -@item Maroon -0x800000 -@item MediumAquaMarine -0x66CDAA -@item MediumBlue -0x0000CD -@item MediumOrchid -0xBA55D3 -@item MediumPurple -0x9370D8 -@item MediumSeaGreen -0x3CB371 -@item MediumSlateBlue -0x7B68EE -@item MediumSpringGreen -0x00FA9A -@item MediumTurquoise -0x48D1CC -@item MediumVioletRed -0xC71585 -@item MidnightBlue -0x191970 -@item MintCream -0xF5FFFA -@item MistyRose -0xFFE4E1 -@item Moccasin -0xFFE4B5 -@item NavajoWhite -0xFFDEAD -@item Navy -0x000080 -@item OldLace -0xFDF5E6 -@item Olive -0x808000 -@item OliveDrab -0x6B8E23 -@item Orange -0xFFA500 -@item OrangeRed -0xFF4500 -@item Orchid -0xDA70D6 -@item PaleGoldenRod -0xEEE8AA -@item PaleGreen -0x98FB98 -@item PaleTurquoise -0xAFEEEE -@item PaleVioletRed -0xD87093 -@item PapayaWhip -0xFFEFD5 -@item PeachPuff -0xFFDAB9 -@item Peru -0xCD853F -@item Pink -0xFFC0CB -@item Plum -0xDDA0DD -@item PowderBlue -0xB0E0E6 -@item Purple -0x800080 -@item Red -0xFF0000 -@item RosyBrown -0xBC8F8F -@item RoyalBlue -0x4169E1 -@item SaddleBrown -0x8B4513 -@item Salmon -0xFA8072 -@item SandyBrown -0xF4A460 -@item SeaGreen -0x2E8B57 -@item SeaShell -0xFFF5EE -@item Sienna -0xA0522D -@item Silver -0xC0C0C0 -@item SkyBlue -0x87CEEB -@item SlateBlue -0x6A5ACD -@item SlateGray -0x708090 -@item Snow -0xFFFAFA -@item SpringGreen -0x00FF7F -@item SteelBlue -0x4682B4 -@item Tan -0xD2B48C -@item Teal -0x008080 -@item Thistle -0xD8BFD8 -@item Tomato -0xFF6347 -@item Turquoise -0x40E0D0 -@item Violet -0xEE82EE -@item Wheat -0xF5DEB3 -@item White -0xFFFFFF -@item WhiteSmoke -0xF5F5F5 -@item Yellow -0xFFFF00 -@item YellowGreen -0x9ACD32 -@end table - -@anchor{channel layout syntax} -@section Channel Layout - -A channel layout specifies the spatial disposition of the channels in -a multi-channel audio stream. To specify a channel layout, FFmpeg -makes use of a special syntax. - -Individual channels are identified by an id, as given by the table -below: -@table @samp -@item FL -front left -@item FR -front right -@item FC -front center -@item LFE -low frequency -@item BL -back left -@item BR -back right -@item FLC -front left-of-center -@item FRC -front right-of-center -@item BC -back center -@item SL -side left -@item SR -side right -@item TC -top center -@item TFL -top front left -@item TFC -top front center -@item TFR -top front right -@item TBL -top back left -@item TBC -top back center -@item TBR -top back right -@item DL -downmix left -@item DR -downmix right -@item WL -wide left -@item WR -wide right -@item SDL -surround direct left -@item SDR -surround direct right -@item LFE2 -low frequency 2 -@end table - -Standard channel layout compositions can be specified by using the -following identifiers: -@table @samp -@item mono -FC -@item stereo -FL+FR -@item 2.1 -FL+FR+LFE -@item 3.0 -FL+FR+FC -@item 3.0(back) -FL+FR+BC -@item 4.0 -FL+FR+FC+BC -@item quad -FL+FR+BL+BR -@item quad(side) -FL+FR+SL+SR -@item 3.1 -FL+FR+FC+LFE -@item 5.0 -FL+FR+FC+BL+BR -@item 5.0(side) -FL+FR+FC+SL+SR -@item 4.1 -FL+FR+FC+LFE+BC -@item 5.1 -FL+FR+FC+LFE+BL+BR -@item 5.1(side) -FL+FR+FC+LFE+SL+SR -@item 6.0 -FL+FR+FC+BC+SL+SR -@item 6.0(front) -FL+FR+FLC+FRC+SL+SR -@item hexagonal -FL+FR+FC+BL+BR+BC -@item 6.1 -FL+FR+FC+LFE+BC+SL+SR -@item 6.1 -FL+FR+FC+LFE+BL+BR+BC -@item 6.1(front) -FL+FR+LFE+FLC+FRC+SL+SR -@item 7.0 -FL+FR+FC+BL+BR+SL+SR -@item 7.0(front) -FL+FR+FC+FLC+FRC+SL+SR -@item 7.1 -FL+FR+FC+LFE+BL+BR+SL+SR -@item 7.1(wide) -FL+FR+FC+LFE+BL+BR+FLC+FRC -@item 7.1(wide-side) -FL+FR+FC+LFE+FLC+FRC+SL+SR -@item octagonal -FL+FR+FC+BL+BR+BC+SL+SR -@item downmix -DL+DR -@end table - -A custom channel layout can be specified as a sequence of terms, separated by -'+' or '|'. Each term can be: -@itemize -@item -the name of a standard channel layout (e.g. @samp{mono}, -@samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.) - -@item -the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.) - -@item -a number of channels, in decimal, optionally followed by 'c', yielding -the default channel layout for that number of channels (see the -function @code{av_get_default_channel_layout}) - -@item -a channel layout mask, in hexadecimal starting with "0x" (see the -@code{AV_CH_*} macros in @file{libavutil/channel_layout.h}. -@end itemize - -Starting from libavutil version 53 the trailing character "c" to -specify a number of channels will be required, while a channel layout -mask could also be specified as a decimal number (if and only if not -followed by "c"). - -See also the function @code{av_get_channel_layout} defined in -@file{libavutil/channel_layout.h}. -@c man end SYNTAX - -@chapter Expression Evaluation -@c man begin EXPRESSION EVALUATION - -When evaluating an arithmetic expression, FFmpeg uses an internal -formula evaluator, implemented through the @file{libavutil/eval.h} -interface. - -An expression may contain unary, binary operators, constants, and -functions. - -Two expressions @var{expr1} and @var{expr2} can be combined to form -another expression "@var{expr1};@var{expr2}". -@var{expr1} and @var{expr2} are evaluated in turn, and the new -expression evaluates to the value of @var{expr2}. - -The following binary operators are available: @code{+}, @code{-}, -@code{*}, @code{/}, @code{^}. - -The following unary operators are available: @code{+}, @code{-}. - -The following functions are available: -@table @option -@item abs(x) -Compute absolute value of @var{x}. - -@item acos(x) -Compute arccosine of @var{x}. - -@item asin(x) -Compute arcsine of @var{x}. - -@item atan(x) -Compute arctangent of @var{x}. - -@item between(x, min, max) -Return 1 if @var{x} is greater than or equal to @var{min} and lesser than or -equal to @var{max}, 0 otherwise. - -@item bitand(x, y) -@item bitor(x, y) -Compute bitwise and/or operation on @var{x} and @var{y}. - -The results of the evaluation of @var{x} and @var{y} are converted to -integers before executing the bitwise operation. - -Note that both the conversion to integer and the conversion back to -floating point can lose precision. Beware of unexpected results for -large numbers (usually 2^53 and larger). - -@item ceil(expr) -Round the value of expression @var{expr} upwards to the nearest -integer. For example, "ceil(1.5)" is "2.0". - -@item cos(x) -Compute cosine of @var{x}. - -@item cosh(x) -Compute hyperbolic cosine of @var{x}. - -@item eq(x, y) -Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise. - -@item exp(x) -Compute exponential of @var{x} (with base @code{e}, the Euler's number). - -@item floor(expr) -Round the value of expression @var{expr} downwards to the nearest -integer. For example, "floor(-1.5)" is "-2.0". - -@item gauss(x) -Compute Gauss function of @var{x}, corresponding to -@code{exp(-x*x/2) / sqrt(2*PI)}. - -@item gcd(x, y) -Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and -@var{y} are 0 or either or both are less than zero then behavior is undefined. - -@item gt(x, y) -Return 1 if @var{x} is greater than @var{y}, 0 otherwise. - -@item gte(x, y) -Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise. - -@item hypot(x, y) -This function is similar to the C function with the same name; it returns -"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a -right triangle with sides of length @var{x} and @var{y}, or the distance of the -point (@var{x}, @var{y}) from the origin. - -@item if(x, y) -Evaluate @var{x}, and if the result is non-zero return the result of -the evaluation of @var{y}, return 0 otherwise. - -@item if(x, y, z) -Evaluate @var{x}, and if the result is non-zero return the evaluation -result of @var{y}, otherwise the evaluation result of @var{z}. - -@item ifnot(x, y) -Evaluate @var{x}, and if the result is zero return the result of the -evaluation of @var{y}, return 0 otherwise. - -@item ifnot(x, y, z) -Evaluate @var{x}, and if the result is zero return the evaluation -result of @var{y}, otherwise the evaluation result of @var{z}. - -@item isinf(x) -Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise. - -@item isnan(x) -Return 1.0 if @var{x} is NAN, 0.0 otherwise. - -@item ld(var) -Allow to load the value of the internal variable with number -@var{var}, which was previously stored with st(@var{var}, @var{expr}). -The function returns the loaded value. - -@item log(x) -Compute natural logarithm of @var{x}. - -@item lt(x, y) -Return 1 if @var{x} is lesser than @var{y}, 0 otherwise. - -@item lte(x, y) -Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise. - -@item max(x, y) -Return the maximum between @var{x} and @var{y}. - -@item min(x, y) -Return the maximum between @var{x} and @var{y}. - -@item mod(x, y) -Compute the remainder of division of @var{x} by @var{y}. - -@item not(expr) -Return 1.0 if @var{expr} is zero, 0.0 otherwise. - -@item pow(x, y) -Compute the power of @var{x} elevated @var{y}, it is equivalent to -"(@var{x})^(@var{y})". - -@item print(t) -@item print(t, l) -Print the value of expression @var{t} with loglevel @var{l}. If -@var{l} is not specified then a default log level is used. -Returns the value of the expression printed. - -Prints t with loglevel l - -@item random(x) -Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the -internal variable which will be used to save the seed/state. - -@item root(expr, max) -Find an input value for which the function represented by @var{expr} -with argument @var{ld(0)} is 0 in the interval 0..@var{max}. - -The expression in @var{expr} must denote a continuous function or the -result is undefined. - -@var{ld(0)} is used to represent the function input value, which means -that the given expression will be evaluated multiple times with -various input values that the expression can access through -@code{ld(0)}. When the expression evaluates to 0 then the -corresponding input value will be returned. - -@item sin(x) -Compute sine of @var{x}. - -@item sinh(x) -Compute hyperbolic sine of @var{x}. - -@item sqrt(expr) -Compute the square root of @var{expr}. This is equivalent to -"(@var{expr})^.5". - -@item squish(x) -Compute expression @code{1/(1 + exp(4*x))}. - -@item st(var, expr) -Allow to store the value of the expression @var{expr} in an internal -variable. @var{var} specifies the number of the variable where to -store the value, and it is a value ranging from 0 to 9. The function -returns the value stored in the internal variable. -Note, Variables are currently not shared between expressions. - -@item tan(x) -Compute tangent of @var{x}. - -@item tanh(x) -Compute hyperbolic tangent of @var{x}. - -@item taylor(expr, x) -@item taylor(expr, x, id) -Evaluate a Taylor series at @var{x}, given an expression representing -the @code{ld(id)}-th derivative of a function at 0. - -When the series does not converge the result is undefined. - -@var{ld(id)} is used to represent the derivative order in @var{expr}, -which means that the given expression will be evaluated multiple times -with various input values that the expression can access through -@code{ld(id)}. If @var{id} is not specified then 0 is assumed. - -Note, when you have the derivatives at y instead of 0, -@code{taylor(expr, x-y)} can be used. - -@item time(0) -Return the current (wallclock) time in seconds. - -@item trunc(expr) -Round the value of expression @var{expr} towards zero to the nearest -integer. For example, "trunc(-1.5)" is "-1.0". - -@item while(cond, expr) -Evaluate expression @var{expr} while the expression @var{cond} is -non-zero, and returns the value of the last @var{expr} evaluation, or -NAN if @var{cond} was always false. -@end table - -The following constants are available: -@table @option -@item PI -area of the unit disc, approximately 3.14 -@item E -exp(1) (Euler's number), approximately 2.718 -@item PHI -golden ratio (1+sqrt(5))/2, approximately 1.618 -@end table - -Assuming that an expression is considered "true" if it has a non-zero -value, note that: - -@code{*} works like AND - -@code{+} works like OR - -For example the construct: -@example -if (A AND B) then C -@end example -is equivalent to: -@example -if(A*B, C) -@end example - -In your C code, you can extend the list of unary and binary functions, -and define recognized constants, so that they are available for your -expressions. - -The evaluator also recognizes the International System unit prefixes. -If 'i' is appended after the prefix, binary prefixes are used, which -are based on powers of 1024 instead of powers of 1000. -The 'B' postfix multiplies the value by 8, and can be appended after a -unit prefix or used alone. This allows using for example 'KB', 'MiB', -'G' and 'B' as number postfix. - -The list of available International System prefixes follows, with -indication of the corresponding powers of 10 and of 2. -@table @option -@item y -10^-24 / 2^-80 -@item z -10^-21 / 2^-70 -@item a -10^-18 / 2^-60 -@item f -10^-15 / 2^-50 -@item p -10^-12 / 2^-40 -@item n -10^-9 / 2^-30 -@item u -10^-6 / 2^-20 -@item m -10^-3 / 2^-10 -@item c -10^-2 -@item d -10^-1 -@item h -10^2 -@item k -10^3 / 2^10 -@item K -10^3 / 2^10 -@item M -10^6 / 2^20 -@item G -10^9 / 2^30 -@item T -10^12 / 2^40 -@item P -10^15 / 2^40 -@item E -10^18 / 2^50 -@item Z -10^21 / 2^60 -@item Y -10^24 / 2^70 -@end table - -@c man end - -@chapter OpenCL Options -@c man begin OPENCL OPTIONS - -When FFmpeg is configured with @code{--enable-opencl}, it is possible -to set the options for the global OpenCL context. - -The list of supported options follows: - -@table @option -@item build_options -Set build options used to compile the registered kernels. - -See reference "OpenCL Specification Version: 1.2 chapter 5.6.4". - -@item platform_idx -Select the index of the platform to run OpenCL code. - -The specified index must be one of the indexes in the device list -which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}. - -@item device_idx -Select the index of the device used to run OpenCL code. - -The specified index must be one of the indexes in the device list which -can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}. - -@end table - -@c man end OPENCL OPTIONS diff --git a/doc/viterbi.txt b/doc/viterbi.txt new file mode 100644 index 0000000000..97825462cc --- /dev/null +++ b/doc/viterbi.txt @@ -0,0 +1,109 @@ +This is a quick description of the viterbi aka dynamic programing +algorthm. + +Its reason for existence is that wikipedia has become very poor on +describing algorithms in a way that makes it useable for understanding +them or anything else actually. It tends now to describe the very same +algorithm under 50 different names and pages with few understandable +by even people who fully understand the algorithm and the theory behind. + +Problem description: (that is what it can solve) +assume we have a 2d table, or you could call it a graph or matrix if you +prefer + + O O O O O O O + + O O O O O O O + + O O O O O O O + + O O O O O O O + + +That table has edges connecting points from each column to the next column +and each edge has a score like: (only some edge and scores shown to keep it +readable) + + + O--5--O-----O-----O-----O-----O + 2 / 7 / \ / \ / \ / + \ / \ / \ / \ / \ / + O7-/--O--/--O--/--O--/--O--/--O + \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/ + /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\ + O3-/--O--/--O--/--O--/--O--/--O + / \ / \ / \ / \ / \ + 1 \ 9 \ / \ / \ / \ + O--2--O--1--O--5--O--3--O--8--O + + + +Our goal is to find a path from left to right through it which +minimizes the sum of the score of all edges. +(and of course left/right is just a convention here it could be top down too) +Similarly the minimum could be the maximum by just fliping the sign, +Example of a path with scores: + + O O O O O O O + +>---O. O O .O-2-O O O + 5. .7 . + O O-1-O O O 8 O O + . + O O O O O O-1-O---> (sum here is 24) + + +The viterbi algorthm now solves this simply column by column +For the previous column each point has a best path and a associated +score: + + O-----5 O + \ + \ + O \ 1 O + \/ + /\ + O / 2 O + / + / + O-----2 O + + +To move one column forward we just need to find the best path and associated +scores for the next column +here are some edges we could choose from: + + + O-----5--3--O + \ \8 + \ \ + O \ 1--9--O + \/ \3 + /\ \ + O / 2--1--O + / \2 + / \ + O-----2--4--O + +Finding the new best paths and scores for each point of our new column is +trivial given we know the previous column best paths and scores: + + O-----0-----8 + \ + \ + O \ 0----10 + \/ + /\ + O / 0-----3 + / \ + / \ + O 0 4 + + +the viterbi algorthm continues exactly like this column for column until the +end and then just picks the path with the best score (above that would be the +one with score 3) + + +Author: Michael niedermayer +Copyright LGPL diff --git a/doc/writing_filters.txt b/doc/writing_filters.txt deleted file mode 100644 index c7923e884f..0000000000 --- a/doc/writing_filters.txt +++ /dev/null @@ -1,424 +0,0 @@ -This document is a tutorial/initiation for writing simple filters in -libavfilter. - -Foreword: just like everything else in FFmpeg, libavfilter is monolithic, which -means that it is highly recommended that you submit your filters to the FFmpeg -development mailing-list and make sure it is applied. Otherwise, your filter is -likely to have a very short lifetime due to more a less regular internal API -changes, and a limited distribution, review, and testing. - -Bootstrap -========= - -Let's say you want to write a new simple video filter called "foobar" which -takes one frame in input, changes the pixels in whatever fashion you fancy, and -outputs the modified frame. The most simple way of doing this is to take a -similar filter. We'll pick edgedetect, but any other should do. You can look -for others using the `./ffmpeg -v 0 -filters|grep ' V->V '` command. - - - cp libavfilter/vf_{edgedetect,foobar}.c - - sed -i s/edgedetect/foobar/g -i libavfilter/vf_foobar.c - - sed -i s/EdgeDetect/Foobar/g -i libavfilter/vf_foobar.c - - edit libavfilter/Makefile, and add an entry for "foobar" following the - pattern of the other filters. - - edit libavfilter/allfilters.c, and add an entry for "foobar" following the - pattern of the other filters. - - ./configure ... - - make -j ffmpeg - - ./ffmpeg -i tests/lena.pnm -vf foobar foobar.png - -If everything went right, you should get a foobar.png with Lena edge-detected. - -That's it, your new playground is ready. - -Some little details about what's going on: -libavfilter/allfilters.c:avfilter_register_all() is called at runtime to create -a list of the available filters, but it's important to know that this file is -also parsed by the configure script, which in turn will define variables for -the build system and the C: - - --- after running configure --- - - $ grep FOOBAR config.mak - CONFIG_FOOBAR_FILTER=yes - $ grep FOOBAR config.h - #define CONFIG_FOOBAR_FILTER 1 - -CONFIG_FOOBAR_FILTER=yes from the config.mak is later used to enable the filter in -libavfilter/Makefile and CONFIG_FOOBAR_FILTER=1 from the config.h will be used -for registering the filter in libavfilter/allfilters.c. - -Filter code layout -================== - -You now need some theory about the general code layout of a filter. Open your -libavfilter/vf_foobar.c. This section will detail the important parts of the -code you need to understand before messing with it. - -Copyright ---------- - -First chunk is the copyright. Most filters are LGPL, and we are assuming -vf_foobar is as well. We are also assuming vf_foobar is not an edge detector -filter, so you can update the boilerplate with your credits. - -Doxy ----- - -Next chunk is the Doxygen about the file. See http://ffmpeg.org/doxygen/trunk/. -Detail here what the filter is, does, and add some references if you feel like -it. - -Context -------- - -Skip the headers and scroll down to the definition of FoobarContext. This is -your local state context. It is already filled with 0 when you get it so do not -worry about uninitialized read into this context. This is where you put every -"global" information you need, typically the variable storing the user options. -You'll notice the first field "const AVClass *class"; it's the only field you -need to keep assuming you have a context. There are some magic you don't care -about around this field, just let it be (in first position) for now. - -Options -------- - -Then comes the options array. This is what will define the user accessible -options. For example, -vf foobar=mode=colormix:high=0.4:low=0.1. Most options -have the following pattern: - name, description, offset, type, default value, minimum value, maximum value, flags - - - name is the option name, keep it simple, lowercase - - description are short, in lowercase, without period, and describe what they - do, for example "set the foo of the bar" - - offset is the offset of the field in your local context, see the OFFSET() - macro; the option parser will use that information to fill the fields - according to the user input - - type is any of AV_OPT_TYPE_* defined in libavutil/opt.h - - default value is an union where you pick the appropriate type; "{.dbl=0.3}", - "{.i64=0x234}", "{.str=NULL}", ... - - min and max values define the range of available values, inclusive - - flags are AVOption generic flags. See AV_OPT_FLAG_* definitions - -In doubt, just look at the other AVOption definitions all around the codebase, -there are tons of examples. - -Class ------ - -AVFILTER_DEFINE_CLASS(foobar) will define a unique foobar_class with some kind -of signature referencing the options, etc. which will be referenced in the -definition of the AVFilter. - -Filter definition ------------------ - -At the end of the file, you will find foobar_inputs, foobar_outputs and -the AVFilter ff_vf_foobar. Don't forget to update the AVFilter.description with -a description of what the filter does, starting with a capitalized letter and -ending with a period. You'd better drop the AVFilter.flags entry for now, and -re-add them later depending on the capabilities of your filter. - -Callbacks ---------- - -Let's now study the common callbacks. Before going into details, note that all -these callbacks are explained in details in libavfilter/avfilter.h, so in -doubt, refer to the doxy in that file. - -init() -~~~~~~ - -First one to be called is init(). It's flagged as cold because not called -often. Look for "cold" on -http://gcc.gnu.org/onlinedocs/gcc/Function-Attributes.html for more -information. - -As the name suggests, init() is where you eventually initialize and allocate -your buffers, pre-compute your data, etc. Note that at this point, your local -context already has the user options initialized, but you still haven't any -clue about the kind of data input you will get, so this function is often -mainly used to sanitize the user options. - -Some init()s will also define the number of inputs or outputs dynamically -according to the user options. A good example of this is the split filter, but -we won't cover this here since vf_foobar is just a simple 1:1 filter. - -uninit() -~~~~~~~~ - -Similarly, there is the uninit() callback, doing what the name suggest. Free -everything you allocated here. - -query_formats() -~~~~~~~~~~~~~~~ - -This is following the init() and is used for the format negotiation, basically -where you say what pixel format(s) (gray, rgb 32, yuv 4:2:0, ...) you accept -for your inputs, and what you can output. All pixel formats are defined in -libavutil/pixfmt.h. If you don't change the pixel format between the input and -the output, you just have to define a pixel formats array and call -ff_set_common_formats(). For more complex negotiation, you can refer to other -filters such as vf_scale. - -config_props() -~~~~~~~~~~~~~~ - -This callback is not necessary, but you will probably have one or more -config_props() anyway. It's not a callback for the filter itself but for its -inputs or outputs (they're called "pads" - AVFilterPad - in libavfilter's -lexicon). - -Inside the input config_props(), you are at a point where you know which pixel -format has been picked after query_formats(), and more information such as the -video width and height (inlink->{w,h}). So if you need to update your internal -context state depending on your input you can do it here. In edgedetect you can -see that this callback is used to allocate buffers depending on these -information. They will be destroyed in uninit(). - -Inside the output config_props(), you can define what you want to change in the -output. Typically, if your filter is going to double the size of the video, you -will update outlink->w and outlink->h. - -filter_frame() -~~~~~~~~~~~~~~ - -This is the callback you are waiting from the beginning: it is where you -process the received frames. Along with the frame, you get the input link from -where the frame comes from. - - static int filter_frame(AVFilterLink *inlink, AVFrame *in) { ... } - -You can get the filter context through that input link: - - AVFilterContext *ctx = inlink->dst; - -Then access your internal state context: - - FoobarContext *foobar = ctx->priv; - -And also the output link where you will send your frame when you are done: - - AVFilterLink *outlink = ctx->outputs[0]; - -Here, we are picking the first output. You can have several, but in our case we -only have one since we are in a 1:1 input-output situation. - -If you want to define a simple pass-through filter, you can just do: - - return ff_filter_frame(outlink, in); - -But of course, you probably want to change the data of that frame. - -This can be done by accessing frame->data[] and frame->linesize[]. Important -note here: the width does NOT match the linesize. The linesize is always -greater or equal to the width. The padding created should not be changed or -even read. Typically, keep in mind that a previous filter in your chain might -have altered the frame dimension but not the linesize. Imagine a crop filter -that halves the video size: the linesizes won't be changed, just the width. - - <-------------- linesize ------------------------> - +-------------------------------+----------------+ ^ - | | | | - | | | | - | picture | padding | | height - | | | | - | | | | - +-------------------------------+----------------+ v - <----------- width -------------> - -Before modifying the "in" frame, you have to make sure it is writable, or get a -new one. Multiple scenarios are possible here depending on the kind of -processing you are doing. - -Let's say you want to change one pixel depending on multiple pixels (typically -the surrounding ones) of the input. In that case, you can't do an in-place -processing of the input so you will need to allocate a new frame, with the same -properties as the input one, and send that new frame to the next filter: - - AVFrame *out = ff_get_video_buffer(outlink, outlink->w, outlink->h); - if (!out) { - av_frame_free(&in); - return AVERROR(ENOMEM); - } - av_frame_copy_props(out, in); - - // out->data[...] = foobar(in->data[...]) - - av_frame_free(&in); - return ff_filter_frame(outlink, out); - -In-place processing -~~~~~~~~~~~~~~~~~~~ - -If you can just alter the input frame, you probably just want to do that -instead: - - av_frame_make_writable(in); - // in->data[...] = foobar(in->data[...]) - return ff_filter_frame(outlink, in); - -You may wonder why a frame might not be writable. The answer is that for -example a previous filter might still own the frame data: imagine a filter -prior to yours in the filtergraph that needs to cache the frame. You must not -alter that frame, otherwise it will make that previous filter buggy. This is -where av_frame_make_writable() helps (it won't have any effect if the frame -already is writable). - -The problem with using av_frame_make_writable() is that in the worst case it -will copy the whole input frame before you change it all over again with your -filter: if the frame is not writable, av_frame_make_writable() will allocate -new buffers, and copy the input frame data. You don't want that, and you can -avoid it by just allocating a new buffer if necessary, and process from in to -out in your filter, saving the memcpy. Generally, this is done following this -scheme: - - int direct = 0; - AVFrame *out; - - if (av_frame_is_writable(in)) { - direct = 1; - out = in; - } else { - out = ff_get_video_buffer(outlink, outlink->w, outlink->h); - if (!out) { - av_frame_free(&in); - return AVERROR(ENOMEM); - } - av_frame_copy_props(out, in); - } - - // out->data[...] = foobar(in->data[...]) - - if (!direct) - av_frame_free(&in); - return ff_filter_frame(outlink, out); - -Of course, this will only work if you can do in-place processing. To test if -your filter handles well the permissions, you can use the perms filter. For -example with: - - -vf perms=random,foobar - -Make sure no automatic pixel conversion is inserted between perms and foobar, -otherwise the frames permissions might change again and the test will be -meaningless: add av_log(0,0,"direct=%d\n",direct) in your code to check that. -You can avoid the issue with something like: - - -vf format=rgb24,perms=random,foobar - -...assuming your filter accepts rgb24 of course. This will make sure the -necessary conversion is inserted before the perms filter. - -Timeline -~~~~~~~~ - -Adding timeline support -(http://ffmpeg.org/ffmpeg-filters.html#Timeline-editing) is often an easy -feature to add. In the most simple case, you just have to add -AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC to the AVFilter.flags. You can typically -do this when your filter does not need to save the previous context frames, or -basically if your filter just alter whatever goes in and doesn't need -previous/future information. See for instance commit 86cb986ce that adds -timeline support to the fieldorder filter. - -In some cases, you might need to reset your context somehow. This is handled by -the AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL flag which is used if the filter -must not process the frames but still wants to keep track of the frames going -through (to keep them in cache for when it's enabled again). See for example -commit 69d72140a that adds timeline support to the phase filter. - -Threading -~~~~~~~~~ - -libavfilter does not yet support frame threading, but you can add slice -threading to your filters. - -Let's say the foobar filter has the following frame processing function: - - dst = out->data[0]; - src = in ->data[0]; - - for (y = 0; y < inlink->h; y++) { - for (x = 0; x < inlink->w; x++) - dst[x] = foobar(src[x]); - dst += out->linesize[0]; - src += in ->linesize[0]; - } - -The first thing is to make this function work into slices. The new code will -look like this: - - for (y = slice_start; y < slice_end; y++) { - for (x = 0; x < inlink->w; x++) - dst[x] = foobar(src[x]); - dst += out->linesize[0]; - src += in ->linesize[0]; - } - -The source and destination pointers, and slice_start/slice_end will be defined -according to the number of jobs. Generally, it looks like this: - - const int slice_start = (in->height * jobnr ) / nb_jobs; - const int slice_end = (in->height * (jobnr+1)) / nb_jobs; - uint8_t *dst = out->data[0] + slice_start * out->linesize[0]; - const uint8_t *src = in->data[0] + slice_start * in->linesize[0]; - -This new code will be isolated in a new filter_slice(): - - static int filter_slice(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { ... } - -Note that we need our input and output frame to define slice_{start,end} and -dst/src, which are not available in that callback. They will be transmitted -through the opaque void *arg. You have to define a structure which contains -everything you need: - - typedef struct ThreadData { - AVFrame *in, *out; - } ThreadData; - -If you need some more information from your local context, put them here. - -In you filter_slice function, you access it like that: - - const ThreadData *td = arg; - -Then in your filter_frame() callback, you need to call the threading -distributor with something like this: - - ThreadData td; - - // ... - - td.in = in; - td.out = out; - ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ctx->graph->nb_threads)); - - // ... - - return ff_filter_frame(outlink, out); - -Last step is to add AVFILTER_FLAG_SLICE_THREADS flag to AVFilter.flags. - -For more example of slice threading additions, you can try to run git log -p ---grep 'slice threading' libavfilter/ - -Finalization -~~~~~~~~~~~~ - -When your awesome filter is finished, you have a few more steps before you're -done: - - - write its documentation in doc/filters.texi, and test the output with make - doc/ffmpeg-filters.html. - - add a FATE test, generally by adding an entry in - tests/fate/filter-video.mak, add running make fate-filter-foobar GEN=1 to - generate the data. - - add an entry in the Changelog - - edit libavfilter/version.h and increase LIBAVFILTER_VERSION_MINOR by one - (and reset LIBAVFILTER_VERSION_MICRO to 100) - - git add ... && git commit -m "avfilter: add foobar filter." && git format-patch -1 - -When all of this is done, you can submit your patch to the ffmpeg-devel -mailing-list for review. If you need any help, feel free to come on our IRC -channel, #ffmpeg-devel on irc.freenode.net. diff --git a/ffmpeg.c b/ffmpeg.c deleted file mode 100644 index a0e2be294f..0000000000 --- a/ffmpeg.c +++ /dev/null @@ -1,3771 +0,0 @@ -/* - * Copyright (c) 2000-2003 Fabrice Bellard - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/** - * @file - * multimedia converter based on the FFmpeg libraries - */ - -#include "config.h" -#include -#include -#include -#include -#include -#include -#include - -#if HAVE_ISATTY -#if HAVE_IO_H -#include -#endif -#if HAVE_UNISTD_H -#include -#endif -#endif - -#include "libavformat/avformat.h" -#include "libavdevice/avdevice.h" -#include "libswresample/swresample.h" -#include "libavutil/opt.h" -#include "libavutil/channel_layout.h" -#include "libavutil/parseutils.h" -#include "libavutil/samplefmt.h" -#include "libavutil/fifo.h" -#include "libavutil/intreadwrite.h" -#include "libavutil/dict.h" -#include "libavutil/mathematics.h" -#include "libavutil/pixdesc.h" -#include "libavutil/avstring.h" -#include "libavutil/libm.h" -#include "libavutil/imgutils.h" -#include "libavutil/timestamp.h" -#include "libavutil/bprint.h" -#include "libavutil/time.h" -#include "libavutil/threadmessage.h" -#include "libavformat/os_support.h" - -#include "libavformat/ffm.h" // not public API - -# include "libavfilter/avcodec.h" -# include "libavfilter/avfilter.h" -# include "libavfilter/buffersrc.h" -# include "libavfilter/buffersink.h" - -#if HAVE_SYS_RESOURCE_H -#include -#include -#include -#elif HAVE_GETPROCESSTIMES -#include -#endif -#if HAVE_GETPROCESSMEMORYINFO -#include -#include -#endif - -#if HAVE_SYS_SELECT_H -#include -#endif - -#if HAVE_TERMIOS_H -#include -#include -#include -#include -#elif HAVE_KBHIT -#include -#endif - -#if HAVE_PTHREADS -#include -#endif - -#include - -#include "ffmpeg.h" -#include "cmdutils.h" - -#include "libavutil/avassert.h" - -const char program_name[] = "ffmpeg"; -const int program_birth_year = 2000; - -static FILE *vstats_file; - -const char *const forced_keyframes_const_names[] = { - "n", - "n_forced", - "prev_forced_n", - "prev_forced_t", - "t", - NULL -}; - -static void do_video_stats(OutputStream *ost, int frame_size); -static int64_t getutime(void); -static int64_t getmaxrss(void); - -static int run_as_daemon = 0; -static int nb_frames_dup = 0; -static int nb_frames_drop = 0; -static int64_t decode_error_stat[2]; - -static int current_time; -AVIOContext *progress_avio = NULL; - -static uint8_t *subtitle_out; - -#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass" - -InputStream **input_streams = NULL; -int nb_input_streams = 0; -InputFile **input_files = NULL; -int nb_input_files = 0; - -OutputStream **output_streams = NULL; -int nb_output_streams = 0; -OutputFile **output_files = NULL; -int nb_output_files = 0; - -FilterGraph **filtergraphs; -int nb_filtergraphs; - -#if HAVE_TERMIOS_H - -/* init terminal so that we can grab keys */ -static struct termios oldtty; -static int restore_tty; -#endif - -static void free_input_threads(void); - - -/* sub2video hack: - Convert subtitles to video with alpha to insert them in filter graphs. - This is a temporary solution until libavfilter gets real subtitles support. - */ - -static int sub2video_get_blank_frame(InputStream *ist) -{ - int ret; - AVFrame *frame = ist->sub2video.frame; - - av_frame_unref(frame); - ist->sub2video.frame->width = ist->sub2video.w; - ist->sub2video.frame->height = ist->sub2video.h; - ist->sub2video.frame->format = AV_PIX_FMT_RGB32; - if ((ret = av_frame_get_buffer(frame, 32)) < 0) - return ret; - memset(frame->data[0], 0, frame->height * frame->linesize[0]); - return 0; -} - -static void sub2video_copy_rect(uint8_t *dst, int dst_linesize, int w, int h, - AVSubtitleRect *r) -{ - uint32_t *pal, *dst2; - uint8_t *src, *src2; - int x, y; - - if (r->type != SUBTITLE_BITMAP) { - av_log(NULL, AV_LOG_WARNING, "sub2video: non-bitmap subtitle\n"); - return; - } - if (r->x < 0 || r->x + r->w > w || r->y < 0 || r->y + r->h > h) { - av_log(NULL, AV_LOG_WARNING, "sub2video: rectangle overflowing\n"); - return; - } - - dst += r->y * dst_linesize + r->x * 4; - src = r->pict.data[0]; - pal = (uint32_t *)r->pict.data[1]; - for (y = 0; y < r->h; y++) { - dst2 = (uint32_t *)dst; - src2 = src; - for (x = 0; x < r->w; x++) - *(dst2++) = pal[*(src2++)]; - dst += dst_linesize; - src += r->pict.linesize[0]; - } -} - -static void sub2video_push_ref(InputStream *ist, int64_t pts) -{ - AVFrame *frame = ist->sub2video.frame; - int i; - - av_assert1(frame->data[0]); - ist->sub2video.last_pts = frame->pts = pts; - for (i = 0; i < ist->nb_filters; i++) - av_buffersrc_add_frame_flags(ist->filters[i]->filter, frame, - AV_BUFFERSRC_FLAG_KEEP_REF | - AV_BUFFERSRC_FLAG_PUSH); -} - -static void sub2video_update(InputStream *ist, AVSubtitle *sub) -{ - int w = ist->sub2video.w, h = ist->sub2video.h; - AVFrame *frame = ist->sub2video.frame; - int8_t *dst; - int dst_linesize; - int num_rects, i; - int64_t pts, end_pts; - - if (!frame) - return; - if (sub) { - pts = av_rescale_q(sub->pts + sub->start_display_time * 1000, - AV_TIME_BASE_Q, ist->st->time_base); - end_pts = av_rescale_q(sub->pts + sub->end_display_time * 1000, - AV_TIME_BASE_Q, ist->st->time_base); - num_rects = sub->num_rects; - } else { - pts = ist->sub2video.end_pts; - end_pts = INT64_MAX; - num_rects = 0; - } - if (sub2video_get_blank_frame(ist) < 0) { - av_log(ist->st->codec, AV_LOG_ERROR, - "Impossible to get a blank canvas.\n"); - return; - } - dst = frame->data [0]; - dst_linesize = frame->linesize[0]; - for (i = 0; i < num_rects; i++) - sub2video_copy_rect(dst, dst_linesize, w, h, sub->rects[i]); - sub2video_push_ref(ist, pts); - ist->sub2video.end_pts = end_pts; -} - -static void sub2video_heartbeat(InputStream *ist, int64_t pts) -{ - InputFile *infile = input_files[ist->file_index]; - int i, j, nb_reqs; - int64_t pts2; - - /* When a frame is read from a file, examine all sub2video streams in - the same file and send the sub2video frame again. Otherwise, decoded - video frames could be accumulating in the filter graph while a filter - (possibly overlay) is desperately waiting for a subtitle frame. */ - for (i = 0; i < infile->nb_streams; i++) { - InputStream *ist2 = input_streams[infile->ist_index + i]; - if (!ist2->sub2video.frame) - continue; - /* subtitles seem to be usually muxed ahead of other streams; - if not, substracting a larger time here is necessary */ - pts2 = av_rescale_q(pts, ist->st->time_base, ist2->st->time_base) - 1; - /* do not send the heartbeat frame if the subtitle is already ahead */ - if (pts2 <= ist2->sub2video.last_pts) - continue; - if (pts2 >= ist2->sub2video.end_pts || !ist2->sub2video.frame->data[0]) - sub2video_update(ist2, NULL); - for (j = 0, nb_reqs = 0; j < ist2->nb_filters; j++) - nb_reqs += av_buffersrc_get_nb_failed_requests(ist2->filters[j]->filter); - if (nb_reqs) - sub2video_push_ref(ist2, pts2); - } -} - -static void sub2video_flush(InputStream *ist) -{ - int i; - - if (ist->sub2video.end_pts < INT64_MAX) - sub2video_update(ist, NULL); - for (i = 0; i < ist->nb_filters; i++) - av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0); -} - -/* end of sub2video hack */ - -static void term_exit_sigsafe(void) -{ -#if HAVE_TERMIOS_H - if(restore_tty) - tcsetattr (0, TCSANOW, &oldtty); -#endif -} - -void term_exit(void) -{ - av_log(NULL, AV_LOG_QUIET, "%s", ""); - term_exit_sigsafe(); -} - -static volatile int received_sigterm = 0; -static volatile int received_nb_signals = 0; -static volatile int transcode_init_done = 0; -static int main_return_code = 0; - -static void -sigterm_handler(int sig) -{ - received_sigterm = sig; - received_nb_signals++; - term_exit_sigsafe(); - if(received_nb_signals > 3) - exit(123); -} - -void term_init(void) -{ -#if HAVE_TERMIOS_H - if(!run_as_daemon){ - struct termios tty; - int istty = 1; -#if HAVE_ISATTY - istty = isatty(0) && isatty(2); -#endif - if (istty && tcgetattr (0, &tty) == 0) { - oldtty = tty; - restore_tty = 1; - - tty.c_iflag &= ~(IGNBRK|BRKINT|PARMRK|ISTRIP - |INLCR|IGNCR|ICRNL|IXON); - tty.c_oflag |= OPOST; - tty.c_lflag &= ~(ECHO|ECHONL|ICANON|IEXTEN); - tty.c_cflag &= ~(CSIZE|PARENB); - tty.c_cflag |= CS8; - tty.c_cc[VMIN] = 1; - tty.c_cc[VTIME] = 0; - - tcsetattr (0, TCSANOW, &tty); - } - signal(SIGQUIT, sigterm_handler); /* Quit (POSIX). */ - } -#endif - avformat_network_deinit(); - - signal(SIGINT , sigterm_handler); /* Interrupt (ANSI). */ - signal(SIGTERM, sigterm_handler); /* Termination (ANSI). */ -#ifdef SIGXCPU - signal(SIGXCPU, sigterm_handler); -#endif -} - -/* read a key without blocking */ -static int read_key(void) -{ - unsigned char ch; -#if HAVE_TERMIOS_H - int n = 1; - struct timeval tv; - fd_set rfds; - - FD_ZERO(&rfds); - FD_SET(0, &rfds); - tv.tv_sec = 0; - tv.tv_usec = 0; - n = select(1, &rfds, NULL, NULL, &tv); - if (n > 0) { - n = read(0, &ch, 1); - if (n == 1) - return ch; - - return n; - } -#elif HAVE_KBHIT -# if HAVE_PEEKNAMEDPIPE - static int is_pipe; - static HANDLE input_handle; - DWORD dw, nchars; - if(!input_handle){ - input_handle = GetStdHandle(STD_INPUT_HANDLE); - is_pipe = !GetConsoleMode(input_handle, &dw); - } - - if (stdin->_cnt > 0) { - read(0, &ch, 1); - return ch; - } - if (is_pipe) { - /* When running under a GUI, you will end here. */ - if (!PeekNamedPipe(input_handle, NULL, 0, NULL, &nchars, NULL)) { - // input pipe may have been closed by the program that ran ffmpeg - return -1; - } - //Read it - if(nchars != 0) { - read(0, &ch, 1); - return ch; - }else{ - return -1; - } - } -# endif - if(kbhit()) - return(getch()); -#endif - return -1; -} - -static int decode_interrupt_cb(void *ctx) -{ - return received_nb_signals > transcode_init_done; -} - -const AVIOInterruptCB int_cb = { decode_interrupt_cb, NULL }; - -static void ffmpeg_cleanup(int ret) -{ - int i, j; - - if (do_benchmark) { - int maxrss = getmaxrss() / 1024; - printf("bench: maxrss=%ikB\n", maxrss); - } - - for (i = 0; i < nb_filtergraphs; i++) { - FilterGraph *fg = filtergraphs[i]; - avfilter_graph_free(&fg->graph); - for (j = 0; j < fg->nb_inputs; j++) { - av_freep(&fg->inputs[j]->name); - av_freep(&fg->inputs[j]); - } - av_freep(&fg->inputs); - for (j = 0; j < fg->nb_outputs; j++) { - av_freep(&fg->outputs[j]->name); - av_freep(&fg->outputs[j]); - } - av_freep(&fg->outputs); - av_freep(&fg->graph_desc); - - av_freep(&filtergraphs[i]); - } - av_freep(&filtergraphs); - - av_freep(&subtitle_out); - - /* close files */ - for (i = 0; i < nb_output_files; i++) { - OutputFile *of = output_files[i]; - AVFormatContext *s = of->ctx; - if (s && s->oformat && !(s->oformat->flags & AVFMT_NOFILE) && s->pb) - avio_close(s->pb); - avformat_free_context(s); - av_dict_free(&of->opts); - - av_freep(&output_files[i]); - } - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - AVBitStreamFilterContext *bsfc = ost->bitstream_filters; - while (bsfc) { - AVBitStreamFilterContext *next = bsfc->next; - av_bitstream_filter_close(bsfc); - bsfc = next; - } - ost->bitstream_filters = NULL; - av_frame_free(&ost->filtered_frame); - - av_parser_close(ost->parser); - - av_freep(&ost->forced_keyframes); - av_expr_free(ost->forced_keyframes_pexpr); - av_freep(&ost->avfilter); - av_freep(&ost->logfile_prefix); - - av_freep(&output_streams[i]); - } -#if HAVE_PTHREADS - free_input_threads(); -#endif - for (i = 0; i < nb_input_files; i++) { - avformat_close_input(&input_files[i]->ctx); - av_freep(&input_files[i]); - } - for (i = 0; i < nb_input_streams; i++) { - InputStream *ist = input_streams[i]; - - av_frame_free(&ist->decoded_frame); - av_frame_free(&ist->filter_frame); - av_dict_free(&ist->decoder_opts); - avsubtitle_free(&ist->prev_sub.subtitle); - av_frame_free(&ist->sub2video.frame); - av_freep(&ist->filters); - av_freep(&ist->hwaccel_device); - - av_freep(&input_streams[i]); - } - - if (vstats_file) - fclose(vstats_file); - av_free(vstats_filename); - - av_freep(&input_streams); - av_freep(&input_files); - av_freep(&output_streams); - av_freep(&output_files); - - uninit_opts(); - - avformat_network_deinit(); - - if (received_sigterm) { - av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n", - (int) received_sigterm); - } else if (ret && transcode_init_done) { - av_log(NULL, AV_LOG_INFO, "Conversion failed!\n"); - } - term_exit(); -} - -void assert_avoptions(AVDictionary *m) -{ - AVDictionaryEntry *t; - if ((t = av_dict_get(m, "", NULL, AV_DICT_IGNORE_SUFFIX))) { - av_log(NULL, AV_LOG_FATAL, "Option %s not found.\n", t->key); - exit_program(1); - } -} - -static void abort_codec_experimental(AVCodec *c, int encoder) -{ - exit_program(1); -} - -static void update_benchmark(const char *fmt, ...) -{ - if (do_benchmark_all) { - int64_t t = getutime(); - va_list va; - char buf[1024]; - - if (fmt) { - va_start(va, fmt); - vsnprintf(buf, sizeof(buf), fmt, va); - va_end(va); - printf("bench: %8"PRIu64" %s \n", t - current_time, buf); - } - current_time = t; - } -} - -static void close_all_output_streams(OutputStream *ost, OSTFinished this_stream, OSTFinished others) -{ - int i; - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost2 = output_streams[i]; - ost2->finished |= ost == ost2 ? this_stream : others; - } -} - -static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost) -{ - AVBitStreamFilterContext *bsfc = ost->bitstream_filters; - AVCodecContext *avctx = ost->st->codec; - int ret; - - if ((avctx->codec_type == AVMEDIA_TYPE_VIDEO && video_sync_method == VSYNC_DROP) || - (avctx->codec_type == AVMEDIA_TYPE_AUDIO && audio_sync_method < 0)) - pkt->pts = pkt->dts = AV_NOPTS_VALUE; - - /* - * Audio encoders may split the packets -- #frames in != #packets out. - * But there is no reordering, so we can limit the number of output packets - * by simply dropping them here. - * Counting encoded video frames needs to be done separately because of - * reordering, see do_video_out() - */ - if (!(avctx->codec_type == AVMEDIA_TYPE_VIDEO && avctx->codec)) { - if (ost->frame_number >= ost->max_frames) { - av_free_packet(pkt); - return; - } - ost->frame_number++; - } - - if (bsfc) - av_packet_split_side_data(pkt); - - while (bsfc) { - AVPacket new_pkt = *pkt; - int a = av_bitstream_filter_filter(bsfc, avctx, NULL, - &new_pkt.data, &new_pkt.size, - pkt->data, pkt->size, - pkt->flags & AV_PKT_FLAG_KEY); - if(a == 0 && new_pkt.data != pkt->data && new_pkt.destruct) { - uint8_t *t = av_malloc(new_pkt.size + FF_INPUT_BUFFER_PADDING_SIZE); //the new should be a subset of the old so cannot overflow - if(t) { - memcpy(t, new_pkt.data, new_pkt.size); - memset(t + new_pkt.size, 0, FF_INPUT_BUFFER_PADDING_SIZE); - new_pkt.data = t; - new_pkt.buf = NULL; - a = 1; - } else - a = AVERROR(ENOMEM); - } - if (a > 0) { - av_free_packet(pkt); - new_pkt.buf = av_buffer_create(new_pkt.data, new_pkt.size, - av_buffer_default_free, NULL, 0); - if (!new_pkt.buf) - exit_program(1); - } else if (a < 0) { - av_log(NULL, AV_LOG_ERROR, "Failed to open bitstream filter %s for stream %d with codec %s", - bsfc->filter->name, pkt->stream_index, - avctx->codec ? avctx->codec->name : "copy"); - print_error("", a); - if (exit_on_error) - exit_program(1); - } - *pkt = new_pkt; - - bsfc = bsfc->next; - } - - if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS) && - (avctx->codec_type == AVMEDIA_TYPE_AUDIO || avctx->codec_type == AVMEDIA_TYPE_VIDEO) && - pkt->dts != AV_NOPTS_VALUE && - ost->last_mux_dts != AV_NOPTS_VALUE) { - int64_t max = ost->last_mux_dts + !(s->oformat->flags & AVFMT_TS_NONSTRICT); - if (pkt->dts < max) { - int loglevel = max - pkt->dts > 2 || avctx->codec_type == AVMEDIA_TYPE_VIDEO ? AV_LOG_WARNING : AV_LOG_DEBUG; - av_log(s, loglevel, "Non-monotonous DTS in output stream " - "%d:%d; previous: %"PRId64", current: %"PRId64"; ", - ost->file_index, ost->st->index, ost->last_mux_dts, pkt->dts); - if (exit_on_error) { - av_log(NULL, AV_LOG_FATAL, "aborting.\n"); - exit_program(1); - } - av_log(s, loglevel, "changing to %"PRId64". This may result " - "in incorrect timestamps in the output file.\n", - max); - if(pkt->pts >= pkt->dts) - pkt->pts = FFMAX(pkt->pts, max); - pkt->dts = max; - } - } - ost->last_mux_dts = pkt->dts; - - ost->data_size += pkt->size; - ost->packets_written++; - - pkt->stream_index = ost->index; - - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "muxer <- type:%s " - "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s size:%d\n", - av_get_media_type_string(ost->st->codec->codec_type), - av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, &ost->st->time_base), - av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, &ost->st->time_base), - pkt->size - ); - } - - ret = av_interleaved_write_frame(s, pkt); - if (ret < 0) { - print_error("av_interleaved_write_frame()", ret); - main_return_code = 1; - close_all_output_streams(ost, MUXER_FINISHED | ENCODER_FINISHED, ENCODER_FINISHED); - } - av_free_packet(pkt); -} - -static void close_output_stream(OutputStream *ost) -{ - OutputFile *of = output_files[ost->file_index]; - - ost->finished |= ENCODER_FINISHED; - if (of->shortest) { - int64_t end = av_rescale_q(ost->sync_opts - ost->first_pts, ost->st->codec->time_base, AV_TIME_BASE_Q); - of->recording_time = FFMIN(of->recording_time, end); - } -} - -static int check_recording_time(OutputStream *ost) -{ - OutputFile *of = output_files[ost->file_index]; - - if (of->recording_time != INT64_MAX && - av_compare_ts(ost->sync_opts - ost->first_pts, ost->st->codec->time_base, of->recording_time, - AV_TIME_BASE_Q) >= 0) { - close_output_stream(ost); - return 0; - } - return 1; -} - -static void do_audio_out(AVFormatContext *s, OutputStream *ost, - AVFrame *frame) -{ - AVCodecContext *enc = ost->st->codec; - AVPacket pkt; - int got_packet = 0; - - av_init_packet(&pkt); - pkt.data = NULL; - pkt.size = 0; - - if (!check_recording_time(ost)) - return; - - if (frame->pts == AV_NOPTS_VALUE || audio_sync_method < 0) - frame->pts = ost->sync_opts; - ost->sync_opts = frame->pts + frame->nb_samples; - ost->samples_encoded += frame->nb_samples; - ost->frames_encoded++; - - av_assert0(pkt.size || !pkt.data); - update_benchmark(NULL); - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "encoder <- type:audio " - "frame_pts:%s frame_pts_time:%s time_base:%d/%d\n", - av_ts2str(frame->pts), av_ts2timestr(frame->pts, &enc->time_base), - enc->time_base.num, enc->time_base.den); - } - - if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) { - av_log(NULL, AV_LOG_FATAL, "Audio encoding failed (avcodec_encode_audio2)\n"); - exit_program(1); - } - update_benchmark("encode_audio %d.%d", ost->file_index, ost->index); - - if (got_packet) { - if (pkt.pts != AV_NOPTS_VALUE) - pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base); - if (pkt.dts != AV_NOPTS_VALUE) - pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base); - if (pkt.duration > 0) - pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base); - - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "encoder -> type:audio " - "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s\n", - av_ts2str(pkt.pts), av_ts2timestr(pkt.pts, &ost->st->time_base), - av_ts2str(pkt.dts), av_ts2timestr(pkt.dts, &ost->st->time_base)); - } - - write_frame(s, &pkt, ost); - } -} - -static void do_subtitle_out(AVFormatContext *s, - OutputStream *ost, - InputStream *ist, - AVSubtitle *sub) -{ - int subtitle_out_max_size = 1024 * 1024; - int subtitle_out_size, nb, i; - AVCodecContext *enc; - AVPacket pkt; - int64_t pts; - - if (sub->pts == AV_NOPTS_VALUE) { - av_log(NULL, AV_LOG_ERROR, "Subtitle packets must have a pts\n"); - if (exit_on_error) - exit_program(1); - return; - } - - enc = ost->st->codec; - - if (!subtitle_out) { - subtitle_out = av_malloc(subtitle_out_max_size); - } - - /* Note: DVB subtitle need one packet to draw them and one other - packet to clear them */ - /* XXX: signal it in the codec context ? */ - if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE) - nb = 2; - else - nb = 1; - - /* shift timestamp to honor -ss and make check_recording_time() work with -t */ - pts = sub->pts; - if (output_files[ost->file_index]->start_time != AV_NOPTS_VALUE) - pts -= output_files[ost->file_index]->start_time; - for (i = 0; i < nb; i++) { - ost->sync_opts = av_rescale_q(pts, AV_TIME_BASE_Q, enc->time_base); - if (!check_recording_time(ost)) - return; - - sub->pts = pts; - // start_display_time is required to be 0 - sub->pts += av_rescale_q(sub->start_display_time, (AVRational){ 1, 1000 }, AV_TIME_BASE_Q); - sub->end_display_time -= sub->start_display_time; - sub->start_display_time = 0; - if (i == 1) - sub->num_rects = 0; - - ost->frames_encoded++; - - subtitle_out_size = avcodec_encode_subtitle(enc, subtitle_out, - subtitle_out_max_size, sub); - if (subtitle_out_size < 0) { - av_log(NULL, AV_LOG_FATAL, "Subtitle encoding failed\n"); - exit_program(1); - } - - av_init_packet(&pkt); - pkt.data = subtitle_out; - pkt.size = subtitle_out_size; - pkt.pts = av_rescale_q(sub->pts, AV_TIME_BASE_Q, ost->st->time_base); - pkt.duration = av_rescale_q(sub->end_display_time, (AVRational){ 1, 1000 }, ost->st->time_base); - if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE) { - /* XXX: the pts correction is handled here. Maybe handling - it in the codec would be better */ - if (i == 0) - pkt.pts += 90 * sub->start_display_time; - else - pkt.pts += 90 * sub->end_display_time; - } - pkt.dts = pkt.pts; - write_frame(s, &pkt, ost); - } -} - -static void do_video_out(AVFormatContext *s, - OutputStream *ost, - AVFrame *in_picture) -{ - int ret, format_video_sync; - AVPacket pkt; - AVCodecContext *enc = ost->st->codec; - int nb_frames, i; - double sync_ipts, delta; - double duration = 0; - int frame_size = 0; - InputStream *ist = NULL; - - if (ost->source_index >= 0) - ist = input_streams[ost->source_index]; - - if(ist && ist->st->start_time != AV_NOPTS_VALUE && ist->st->first_dts != AV_NOPTS_VALUE && ost->frame_rate.num) - duration = 1/(av_q2d(ost->frame_rate) * av_q2d(enc->time_base)); - - sync_ipts = in_picture->pts; - delta = sync_ipts - ost->sync_opts + duration; - - /* by default, we output a single frame */ - nb_frames = 1; - - format_video_sync = video_sync_method; - if (format_video_sync == VSYNC_AUTO) { - if(!strcmp(s->oformat->name, "avi")) { - format_video_sync = VSYNC_VFR; - } else - format_video_sync = (s->oformat->flags & AVFMT_VARIABLE_FPS) ? ((s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH : VSYNC_VFR) : VSYNC_CFR; - if ( ist - && format_video_sync == VSYNC_CFR - && input_files[ist->file_index]->ctx->nb_streams == 1 - && input_files[ist->file_index]->input_ts_offset == 0) { - format_video_sync = VSYNC_VSCFR; - } - if (format_video_sync == VSYNC_CFR && copy_ts) { - format_video_sync = VSYNC_VSCFR; - } - } - - switch (format_video_sync) { - case VSYNC_VSCFR: - if (ost->frame_number == 0 && delta - duration >= 0.5) { - av_log(NULL, AV_LOG_DEBUG, "Not duplicating %d initial frames\n", (int)lrintf(delta - duration)); - delta = duration; - ost->sync_opts = lrint(sync_ipts); - } - case VSYNC_CFR: - // FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c - if (delta < -1.1) - nb_frames = 0; - else if (delta > 1.1) - nb_frames = lrintf(delta); - break; - case VSYNC_VFR: - if (delta <= -0.6) - nb_frames = 0; - else if (delta > 0.6) - ost->sync_opts = lrint(sync_ipts); - break; - case VSYNC_DROP: - case VSYNC_PASSTHROUGH: - ost->sync_opts = lrint(sync_ipts); - break; - default: - av_assert0(0); - } - - nb_frames = FFMIN(nb_frames, ost->max_frames - ost->frame_number); - if (nb_frames == 0) { - nb_frames_drop++; - av_log(NULL, AV_LOG_VERBOSE, - "*** dropping frame %d from stream %d at ts %"PRId64"\n", - ost->frame_number, ost->st->index, in_picture->pts); - return; - } else if (nb_frames > 1) { - if (nb_frames > dts_error_threshold * 30) { - av_log(NULL, AV_LOG_ERROR, "%d frame duplication too large, skipping\n", nb_frames - 1); - nb_frames_drop++; - return; - } - nb_frames_dup += nb_frames - 1; - av_log(NULL, AV_LOG_VERBOSE, "*** %d dup!\n", nb_frames - 1); - } - - /* duplicates frame if needed */ - for (i = 0; i < nb_frames; i++) { - av_init_packet(&pkt); - pkt.data = NULL; - pkt.size = 0; - - in_picture->pts = ost->sync_opts; - -#if 1 - if (!check_recording_time(ost)) -#else - if (ost->frame_number >= ost->max_frames) -#endif - return; - - if (s->oformat->flags & AVFMT_RAWPICTURE && - enc->codec->id == AV_CODEC_ID_RAWVIDEO) { - /* raw pictures are written as AVPicture structure to - avoid any copies. We support temporarily the older - method. */ - enc->coded_frame->interlaced_frame = in_picture->interlaced_frame; - enc->coded_frame->top_field_first = in_picture->top_field_first; - if (enc->coded_frame->interlaced_frame) - enc->field_order = enc->coded_frame->top_field_first ? AV_FIELD_TB:AV_FIELD_BT; - else - enc->field_order = AV_FIELD_PROGRESSIVE; - pkt.data = (uint8_t *)in_picture; - pkt.size = sizeof(AVPicture); - pkt.pts = av_rescale_q(in_picture->pts, enc->time_base, ost->st->time_base); - pkt.flags |= AV_PKT_FLAG_KEY; - - write_frame(s, &pkt, ost); - } else { - int got_packet, forced_keyframe = 0; - double pts_time; - - if (ost->st->codec->flags & (CODEC_FLAG_INTERLACED_DCT|CODEC_FLAG_INTERLACED_ME) && - ost->top_field_first >= 0) - in_picture->top_field_first = !!ost->top_field_first; - - if (in_picture->interlaced_frame) { - if (enc->codec->id == AV_CODEC_ID_MJPEG) - enc->field_order = in_picture->top_field_first ? AV_FIELD_TT:AV_FIELD_BB; - else - enc->field_order = in_picture->top_field_first ? AV_FIELD_TB:AV_FIELD_BT; - } else - enc->field_order = AV_FIELD_PROGRESSIVE; - - in_picture->quality = ost->st->codec->global_quality; - if (!enc->me_threshold) - in_picture->pict_type = 0; - - pts_time = in_picture->pts != AV_NOPTS_VALUE ? - in_picture->pts * av_q2d(enc->time_base) : NAN; - if (ost->forced_kf_index < ost->forced_kf_count && - in_picture->pts >= ost->forced_kf_pts[ost->forced_kf_index]) { - ost->forced_kf_index++; - forced_keyframe = 1; - } else if (ost->forced_keyframes_pexpr) { - double res; - ost->forced_keyframes_expr_const_values[FKF_T] = pts_time; - res = av_expr_eval(ost->forced_keyframes_pexpr, - ost->forced_keyframes_expr_const_values, NULL); - av_dlog(NULL, "force_key_frame: n:%f n_forced:%f prev_forced_n:%f t:%f prev_forced_t:%f -> res:%f\n", - ost->forced_keyframes_expr_const_values[FKF_N], - ost->forced_keyframes_expr_const_values[FKF_N_FORCED], - ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N], - ost->forced_keyframes_expr_const_values[FKF_T], - ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T], - res); - if (res) { - forced_keyframe = 1; - ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N] = - ost->forced_keyframes_expr_const_values[FKF_N]; - ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T] = - ost->forced_keyframes_expr_const_values[FKF_T]; - ost->forced_keyframes_expr_const_values[FKF_N_FORCED] += 1; - } - - ost->forced_keyframes_expr_const_values[FKF_N] += 1; - } - - if (forced_keyframe) { - in_picture->pict_type = AV_PICTURE_TYPE_I; - av_log(NULL, AV_LOG_DEBUG, "Forced keyframe at time %f\n", pts_time); - } - - update_benchmark(NULL); - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "encoder <- type:video " - "frame_pts:%s frame_pts_time:%s time_base:%d/%d\n", - av_ts2str(in_picture->pts), av_ts2timestr(in_picture->pts, &enc->time_base), - enc->time_base.num, enc->time_base.den); - } - - ost->frames_encoded++; - - ret = avcodec_encode_video2(enc, &pkt, in_picture, &got_packet); - update_benchmark("encode_video %d.%d", ost->file_index, ost->index); - if (ret < 0) { - av_log(NULL, AV_LOG_FATAL, "Video encoding failed\n"); - exit_program(1); - } - - if (got_packet) { - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "encoder -> type:video " - "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s\n", - av_ts2str(pkt.pts), av_ts2timestr(pkt.pts, &enc->time_base), - av_ts2str(pkt.dts), av_ts2timestr(pkt.dts, &enc->time_base)); - } - - if (pkt.pts == AV_NOPTS_VALUE && !(enc->codec->capabilities & CODEC_CAP_DELAY)) - pkt.pts = ost->sync_opts; - - if (pkt.pts != AV_NOPTS_VALUE) - pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base); - if (pkt.dts != AV_NOPTS_VALUE) - pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base); - - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "encoder -> type:video " - "pkt_pts:%s pkt_pts_time:%s pkt_dts:%s pkt_dts_time:%s\n", - av_ts2str(pkt.pts), av_ts2timestr(pkt.pts, &ost->st->time_base), - av_ts2str(pkt.dts), av_ts2timestr(pkt.dts, &ost->st->time_base)); - } - - frame_size = pkt.size; - write_frame(s, &pkt, ost); - - /* if two pass, output log */ - if (ost->logfile && enc->stats_out) { - fprintf(ost->logfile, "%s", enc->stats_out); - } - } - } - ost->sync_opts++; - /* - * For video, number of frames in == number of packets out. - * But there may be reordering, so we can't throw away frames on encoder - * flush, we need to limit them here, before they go into encoder. - */ - ost->frame_number++; - - if (vstats_filename && frame_size) - do_video_stats(ost, frame_size); - } -} - -static double psnr(double d) -{ - return -10.0 * log(d) / log(10.0); -} - -static void do_video_stats(OutputStream *ost, int frame_size) -{ - AVCodecContext *enc; - int frame_number; - double ti1, bitrate, avg_bitrate; - - /* this is executed just the first time do_video_stats is called */ - if (!vstats_file) { - vstats_file = fopen(vstats_filename, "w"); - if (!vstats_file) { - perror("fopen"); - exit_program(1); - } - } - - enc = ost->st->codec; - if (enc->codec_type == AVMEDIA_TYPE_VIDEO) { - frame_number = ost->st->nb_frames; - fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number, enc->coded_frame->quality / (float)FF_QP2LAMBDA); - if (enc->flags&CODEC_FLAG_PSNR) - fprintf(vstats_file, "PSNR= %6.2f ", psnr(enc->coded_frame->error[0] / (enc->width * enc->height * 255.0 * 255.0))); - - fprintf(vstats_file,"f_size= %6d ", frame_size); - /* compute pts value */ - ti1 = av_stream_get_end_pts(ost->st) * av_q2d(ost->st->time_base); - if (ti1 < 0.01) - ti1 = 0.01; - - bitrate = (frame_size * 8) / av_q2d(enc->time_base) / 1000.0; - avg_bitrate = (double)(ost->data_size * 8) / ti1 / 1000.0; - fprintf(vstats_file, "s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s ", - (double)ost->data_size / 1024, ti1, bitrate, avg_bitrate); - fprintf(vstats_file, "type= %c\n", av_get_picture_type_char(enc->coded_frame->pict_type)); - } -} - -/** - * Get and encode new output from any of the filtergraphs, without causing - * activity. - * - * @return 0 for success, <0 for severe errors - */ -static int reap_filters(void) -{ - AVFrame *filtered_frame = NULL; - int i; - int64_t frame_pts; - - /* Reap all buffers present in the buffer sinks */ - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - OutputFile *of = output_files[ost->file_index]; - AVFilterContext *filter; - AVCodecContext *enc = ost->st->codec; - int ret = 0; - - if (!ost->filter) - continue; - filter = ost->filter->filter; - - if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) { - return AVERROR(ENOMEM); - } - filtered_frame = ost->filtered_frame; - - while (1) { - ret = av_buffersink_get_frame_flags(filter, filtered_frame, - AV_BUFFERSINK_FLAG_NO_REQUEST); - if (ret < 0) { - if (ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) { - av_log(NULL, AV_LOG_WARNING, - "Error in av_buffersink_get_frame_flags(): %s\n", av_err2str(ret)); - } - break; - } - if (ost->finished) { - av_frame_unref(filtered_frame); - continue; - } - frame_pts = AV_NOPTS_VALUE; - if (filtered_frame->pts != AV_NOPTS_VALUE) { - int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time; - filtered_frame->pts = frame_pts = - av_rescale_q(filtered_frame->pts, filter->inputs[0]->time_base, enc->time_base) - - av_rescale_q(start_time, AV_TIME_BASE_Q, enc->time_base); - } - //if (ost->source_index >= 0) - // *filtered_frame= *input_streams[ost->source_index]->decoded_frame; //for me_threshold - - switch (filter->inputs[0]->type) { - case AVMEDIA_TYPE_VIDEO: - filtered_frame->pts = frame_pts; - if (!ost->frame_aspect_ratio.num) - enc->sample_aspect_ratio = filtered_frame->sample_aspect_ratio; - - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "filter -> pts:%s pts_time:%s time_base:%d/%d\n", - av_ts2str(filtered_frame->pts), av_ts2timestr(filtered_frame->pts, &enc->time_base), - enc->time_base.num, enc->time_base.den); - } - - do_video_out(of->ctx, ost, filtered_frame); - break; - case AVMEDIA_TYPE_AUDIO: - filtered_frame->pts = frame_pts; - if (!(enc->codec->capabilities & CODEC_CAP_PARAM_CHANGE) && - enc->channels != av_frame_get_channels(filtered_frame)) { - av_log(NULL, AV_LOG_ERROR, - "Audio filter graph output is not normalized and encoder does not support parameter changes\n"); - break; - } - do_audio_out(of->ctx, ost, filtered_frame); - break; - default: - // TODO support subtitle filters - av_assert0(0); - } - - av_frame_unref(filtered_frame); - } - } - - return 0; -} - -static void print_final_stats(int64_t total_size) -{ - uint64_t video_size = 0, audio_size = 0, extra_size = 0, other_size = 0; - uint64_t subtitle_size = 0; - uint64_t data_size = 0; - float percent = -1.0; - int i, j; - - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - switch (ost->st->codec->codec_type) { - case AVMEDIA_TYPE_VIDEO: video_size += ost->data_size; break; - case AVMEDIA_TYPE_AUDIO: audio_size += ost->data_size; break; - case AVMEDIA_TYPE_SUBTITLE: subtitle_size += ost->data_size; break; - default: other_size += ost->data_size; break; - } - extra_size += ost->st->codec->extradata_size; - data_size += ost->data_size; - } - - if (data_size && total_size >= data_size) - percent = 100.0 * (total_size - data_size) / data_size; - - av_log(NULL, AV_LOG_INFO, "\n"); - av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB subtitle:%1.0fkB other streams:%1.0fkB global headers:%1.0fkB muxing overhead: ", - video_size / 1024.0, - audio_size / 1024.0, - subtitle_size / 1024.0, - other_size / 1024.0, - extra_size / 1024.0); - if (percent >= 0.0) - av_log(NULL, AV_LOG_INFO, "%f%%", percent); - else - av_log(NULL, AV_LOG_INFO, "unknown"); - av_log(NULL, AV_LOG_INFO, "\n"); - - /* print verbose per-stream stats */ - for (i = 0; i < nb_input_files; i++) { - InputFile *f = input_files[i]; - uint64_t total_packets = 0, total_size = 0; - - av_log(NULL, AV_LOG_VERBOSE, "Input file #%d (%s):\n", - i, f->ctx->filename); - - for (j = 0; j < f->nb_streams; j++) { - InputStream *ist = input_streams[f->ist_index + j]; - enum AVMediaType type = ist->st->codec->codec_type; - - total_size += ist->data_size; - total_packets += ist->nb_packets; - - av_log(NULL, AV_LOG_VERBOSE, " Input stream #%d:%d (%s): ", - i, j, media_type_string(type)); - av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" packets read (%"PRIu64" bytes); ", - ist->nb_packets, ist->data_size); - - if (ist->decoding_needed) { - av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" frames decoded", - ist->frames_decoded); - if (type == AVMEDIA_TYPE_AUDIO) - av_log(NULL, AV_LOG_VERBOSE, " (%"PRIu64" samples)", ist->samples_decoded); - av_log(NULL, AV_LOG_VERBOSE, "; "); - } - - av_log(NULL, AV_LOG_VERBOSE, "\n"); - } - - av_log(NULL, AV_LOG_VERBOSE, " Total: %"PRIu64" packets (%"PRIu64" bytes) demuxed\n", - total_packets, total_size); - } - - for (i = 0; i < nb_output_files; i++) { - OutputFile *of = output_files[i]; - uint64_t total_packets = 0, total_size = 0; - - av_log(NULL, AV_LOG_VERBOSE, "Output file #%d (%s):\n", - i, of->ctx->filename); - - for (j = 0; j < of->ctx->nb_streams; j++) { - OutputStream *ost = output_streams[of->ost_index + j]; - enum AVMediaType type = ost->st->codec->codec_type; - - total_size += ost->data_size; - total_packets += ost->packets_written; - - av_log(NULL, AV_LOG_VERBOSE, " Output stream #%d:%d (%s): ", - i, j, media_type_string(type)); - if (ost->encoding_needed) { - av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" frames encoded", - ost->frames_encoded); - if (type == AVMEDIA_TYPE_AUDIO) - av_log(NULL, AV_LOG_VERBOSE, " (%"PRIu64" samples)", ost->samples_encoded); - av_log(NULL, AV_LOG_VERBOSE, "; "); - } - - av_log(NULL, AV_LOG_VERBOSE, "%"PRIu64" packets muxed (%"PRIu64" bytes); ", - ost->packets_written, ost->data_size); - - av_log(NULL, AV_LOG_VERBOSE, "\n"); - } - - av_log(NULL, AV_LOG_VERBOSE, " Total: %"PRIu64" packets (%"PRIu64" bytes) muxed\n", - total_packets, total_size); - } - if(video_size + data_size + audio_size + subtitle_size + extra_size == 0){ - av_log(NULL, AV_LOG_WARNING, "Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)\n"); - } -} - -static void print_report(int is_last_report, int64_t timer_start, int64_t cur_time) -{ - char buf[1024]; - AVBPrint buf_script; - OutputStream *ost; - AVFormatContext *oc; - int64_t total_size; - AVCodecContext *enc; - int frame_number, vid, i; - double bitrate; - int64_t pts = INT64_MIN; - static int64_t last_time = -1; - static int qp_histogram[52]; - int hours, mins, secs, us; - - if (!print_stats && !is_last_report && !progress_avio) - return; - - if (!is_last_report) { - if (last_time == -1) { - last_time = cur_time; - return; - } - if ((cur_time - last_time) < 500000) - return; - last_time = cur_time; - } - - - oc = output_files[0]->ctx; - - total_size = avio_size(oc->pb); - if (total_size <= 0) // FIXME improve avio_size() so it works with non seekable output too - total_size = avio_tell(oc->pb); - - buf[0] = '\0'; - vid = 0; - av_bprint_init(&buf_script, 0, 1); - for (i = 0; i < nb_output_streams; i++) { - float q = -1; - ost = output_streams[i]; - enc = ost->st->codec; - if (!ost->stream_copy && enc->coded_frame) - q = enc->coded_frame->quality / (float)FF_QP2LAMBDA; - if (vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) { - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "q=%2.1f ", q); - av_bprintf(&buf_script, "stream_%d_%d_q=%.1f\n", - ost->file_index, ost->index, q); - } - if (!vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) { - float fps, t = (cur_time-timer_start) / 1000000.0; - - frame_number = ost->frame_number; - fps = t > 1 ? frame_number / t : 0; - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "frame=%5d fps=%3.*f q=%3.1f ", - frame_number, fps < 9.95, fps, q); - av_bprintf(&buf_script, "frame=%d\n", frame_number); - av_bprintf(&buf_script, "fps=%.1f\n", fps); - av_bprintf(&buf_script, "stream_%d_%d_q=%.1f\n", - ost->file_index, ost->index, q); - if (is_last_report) - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "L"); - if (qp_hist) { - int j; - int qp = lrintf(q); - if (qp >= 0 && qp < FF_ARRAY_ELEMS(qp_histogram)) - qp_histogram[qp]++; - for (j = 0; j < 32; j++) - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%X", (int)lrintf(log2(qp_histogram[j] + 1))); - } - if ((enc->flags&CODEC_FLAG_PSNR) && (enc->coded_frame || is_last_report)) { - int j; - double error, error_sum = 0; - double scale, scale_sum = 0; - double p; - char type[3] = { 'Y','U','V' }; - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "PSNR="); - for (j = 0; j < 3; j++) { - if (is_last_report) { - error = enc->error[j]; - scale = enc->width * enc->height * 255.0 * 255.0 * frame_number; - } else { - error = enc->coded_frame->error[j]; - scale = enc->width * enc->height * 255.0 * 255.0; - } - if (j) - scale /= 4; - error_sum += error; - scale_sum += scale; - p = psnr(error / scale); - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%c:%2.2f ", type[j], p); - av_bprintf(&buf_script, "stream_%d_%d_psnr_%c=%2.2f\n", - ost->file_index, ost->index, type[j] | 32, p); - } - p = psnr(error_sum / scale_sum); - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "*:%2.2f ", psnr(error_sum / scale_sum)); - av_bprintf(&buf_script, "stream_%d_%d_psnr_all=%2.2f\n", - ost->file_index, ost->index, p); - } - vid = 1; - } - /* compute min output value */ - if (av_stream_get_end_pts(ost->st) != AV_NOPTS_VALUE) - pts = FFMAX(pts, av_rescale_q(av_stream_get_end_pts(ost->st), - ost->st->time_base, AV_TIME_BASE_Q)); - } - - secs = pts / AV_TIME_BASE; - us = pts % AV_TIME_BASE; - mins = secs / 60; - secs %= 60; - hours = mins / 60; - mins %= 60; - - bitrate = pts && total_size >= 0 ? total_size * 8 / (pts / 1000.0) : -1; - - if (total_size < 0) snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), - "size=N/A time="); - else snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), - "size=%8.0fkB time=", total_size / 1024.0); - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), - "%02d:%02d:%02d.%02d ", hours, mins, secs, - (100 * us) / AV_TIME_BASE); - if (bitrate < 0) snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), - "bitrate=N/A"); - else snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), - "bitrate=%6.1fkbits/s", bitrate); - if (total_size < 0) av_bprintf(&buf_script, "total_size=N/A\n"); - else av_bprintf(&buf_script, "total_size=%"PRId64"\n", total_size); - av_bprintf(&buf_script, "out_time_ms=%"PRId64"\n", pts); - av_bprintf(&buf_script, "out_time=%02d:%02d:%02d.%06d\n", - hours, mins, secs, us); - - if (nb_frames_dup || nb_frames_drop) - snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), " dup=%d drop=%d", - nb_frames_dup, nb_frames_drop); - av_bprintf(&buf_script, "dup_frames=%d\n", nb_frames_dup); - av_bprintf(&buf_script, "drop_frames=%d\n", nb_frames_drop); - - if (print_stats || is_last_report) { - if (print_stats==1 && AV_LOG_INFO > av_log_get_level()) { - fprintf(stderr, "%s \r", buf); - } else - av_log(NULL, AV_LOG_INFO, "%s \r", buf); - - fflush(stderr); - } - - if (progress_avio) { - av_bprintf(&buf_script, "progress=%s\n", - is_last_report ? "end" : "continue"); - avio_write(progress_avio, buf_script.str, - FFMIN(buf_script.len, buf_script.size - 1)); - avio_flush(progress_avio); - av_bprint_finalize(&buf_script, NULL); - if (is_last_report) { - avio_close(progress_avio); - progress_avio = NULL; - } - } - - if (is_last_report) - print_final_stats(total_size); -} - -static void flush_encoders(void) -{ - int i, ret; - - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - AVCodecContext *enc = ost->st->codec; - AVFormatContext *os = output_files[ost->file_index]->ctx; - int stop_encoding = 0; - - if (!ost->encoding_needed) - continue; - - if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <= 1) - continue; - if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE) && enc->codec->id == AV_CODEC_ID_RAWVIDEO) - continue; - - for (;;) { - int (*encode)(AVCodecContext*, AVPacket*, const AVFrame*, int*) = NULL; - const char *desc; - - switch (ost->st->codec->codec_type) { - case AVMEDIA_TYPE_AUDIO: - encode = avcodec_encode_audio2; - desc = "Audio"; - break; - case AVMEDIA_TYPE_VIDEO: - encode = avcodec_encode_video2; - desc = "Video"; - break; - default: - stop_encoding = 1; - } - - if (encode) { - AVPacket pkt; - int pkt_size; - int got_packet; - av_init_packet(&pkt); - pkt.data = NULL; - pkt.size = 0; - - update_benchmark(NULL); - ret = encode(enc, &pkt, NULL, &got_packet); - update_benchmark("flush %s %d.%d", desc, ost->file_index, ost->index); - if (ret < 0) { - av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc); - exit_program(1); - } - if (ost->logfile && enc->stats_out) { - fprintf(ost->logfile, "%s", enc->stats_out); - } - if (!got_packet) { - stop_encoding = 1; - break; - } - if (ost->finished & MUXER_FINISHED) { - av_free_packet(&pkt); - continue; - } - if (pkt.pts != AV_NOPTS_VALUE) - pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base); - if (pkt.dts != AV_NOPTS_VALUE) - pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base); - if (pkt.duration > 0) - pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base); - pkt_size = pkt.size; - write_frame(os, &pkt, ost); - if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && vstats_filename) { - do_video_stats(ost, pkt_size); - } - } - - if (stop_encoding) - break; - } - } -} - -/* - * Check whether a packet from ist should be written into ost at this time - */ -static int check_output_constraints(InputStream *ist, OutputStream *ost) -{ - OutputFile *of = output_files[ost->file_index]; - int ist_index = input_files[ist->file_index]->ist_index + ist->st->index; - - if (ost->source_index != ist_index) - return 0; - - if (ost->finished) - return 0; - - if (of->start_time != AV_NOPTS_VALUE && ist->pts < of->start_time) - return 0; - - return 1; -} - -static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *pkt) -{ - OutputFile *of = output_files[ost->file_index]; - InputFile *f = input_files [ist->file_index]; - int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time; - int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->st->time_base); - int64_t ist_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ist->st->time_base); - AVPicture pict; - AVPacket opkt; - - av_init_packet(&opkt); - - if ((!ost->frame_number && !(pkt->flags & AV_PKT_FLAG_KEY)) && - !ost->copy_initial_nonkeyframes) - return; - - if (pkt->pts == AV_NOPTS_VALUE) { - if (!ost->frame_number && ist->pts < start_time && - !ost->copy_prior_start) - return; - } else { - if (!ost->frame_number && pkt->pts < ist_tb_start_time && - !ost->copy_prior_start) - return; - } - - if (of->recording_time != INT64_MAX && - ist->pts >= of->recording_time + start_time) { - close_output_stream(ost); - return; - } - - if (f->recording_time != INT64_MAX) { - start_time = f->ctx->start_time; - if (f->start_time != AV_NOPTS_VALUE) - start_time += f->start_time; - if (ist->pts >= f->recording_time + start_time) { - close_output_stream(ost); - return; - } - } - - /* force the input stream PTS */ - if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) - ost->sync_opts++; - - if (pkt->pts != AV_NOPTS_VALUE) - opkt.pts = av_rescale_q(pkt->pts, ist->st->time_base, ost->st->time_base) - ost_tb_start_time; - else - opkt.pts = AV_NOPTS_VALUE; - - if (pkt->dts == AV_NOPTS_VALUE) - opkt.dts = av_rescale_q(ist->dts, AV_TIME_BASE_Q, ost->st->time_base); - else - opkt.dts = av_rescale_q(pkt->dts, ist->st->time_base, ost->st->time_base); - opkt.dts -= ost_tb_start_time; - - if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && pkt->dts != AV_NOPTS_VALUE) { - int duration = av_get_audio_frame_duration(ist->st->codec, pkt->size); - if(!duration) - duration = ist->st->codec->frame_size; - opkt.dts = opkt.pts = av_rescale_delta(ist->st->time_base, pkt->dts, - (AVRational){1, ist->st->codec->sample_rate}, duration, &ist->filter_in_rescale_delta_last, - ost->st->time_base) - ost_tb_start_time; - } - - opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->st->time_base); - opkt.flags = pkt->flags; - - // FIXME remove the following 2 lines they shall be replaced by the bitstream filters - if ( ost->st->codec->codec_id != AV_CODEC_ID_H264 - && ost->st->codec->codec_id != AV_CODEC_ID_MPEG1VIDEO - && ost->st->codec->codec_id != AV_CODEC_ID_MPEG2VIDEO - && ost->st->codec->codec_id != AV_CODEC_ID_VC1 - ) { - if (av_parser_change(ost->parser, ost->st->codec, - &opkt.data, &opkt.size, - pkt->data, pkt->size, - pkt->flags & AV_PKT_FLAG_KEY)) { - opkt.buf = av_buffer_create(opkt.data, opkt.size, av_buffer_default_free, NULL, 0); - if (!opkt.buf) - exit_program(1); - } - } else { - opkt.data = pkt->data; - opkt.size = pkt->size; - } - av_copy_packet_side_data(&opkt, pkt); - - if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) { - /* store AVPicture in AVPacket, as expected by the output format */ - avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height); - opkt.data = (uint8_t *)&pict; - opkt.size = sizeof(AVPicture); - opkt.flags |= AV_PKT_FLAG_KEY; - } - - write_frame(of->ctx, &opkt, ost); -} - -int guess_input_channel_layout(InputStream *ist) -{ - AVCodecContext *dec = ist->st->codec; - - if (!dec->channel_layout) { - char layout_name[256]; - - if (dec->channels > ist->guess_layout_max) - return 0; - dec->channel_layout = av_get_default_channel_layout(dec->channels); - if (!dec->channel_layout) - return 0; - av_get_channel_layout_string(layout_name, sizeof(layout_name), - dec->channels, dec->channel_layout); - av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Input Stream " - "#%d.%d : %s\n", ist->file_index, ist->st->index, layout_name); - } - return 1; -} - -static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output) -{ - AVFrame *decoded_frame, *f; - AVCodecContext *avctx = ist->st->codec; - int i, ret, err = 0, resample_changed; - AVRational decoded_frame_tb; - - if (!ist->decoded_frame && !(ist->decoded_frame = av_frame_alloc())) - return AVERROR(ENOMEM); - if (!ist->filter_frame && !(ist->filter_frame = av_frame_alloc())) - return AVERROR(ENOMEM); - decoded_frame = ist->decoded_frame; - - update_benchmark(NULL); - ret = avcodec_decode_audio4(avctx, decoded_frame, got_output, pkt); - update_benchmark("decode_audio %d.%d", ist->file_index, ist->st->index); - - if (ret >= 0 && avctx->sample_rate <= 0) { - av_log(avctx, AV_LOG_ERROR, "Sample rate %d invalid\n", avctx->sample_rate); - ret = AVERROR_INVALIDDATA; - } - - if (*got_output || ret<0 || pkt->size) - decode_error_stat[ret<0] ++; - - if (!*got_output || ret < 0) { - if (!pkt->size) { - for (i = 0; i < ist->nb_filters; i++) -#if 1 - av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0); -#else - av_buffersrc_add_frame(ist->filters[i]->filter, NULL); -#endif - } - return ret; - } - - ist->samples_decoded += decoded_frame->nb_samples; - ist->frames_decoded++; - -#if 1 - /* increment next_dts to use for the case where the input stream does not - have timestamps or there are multiple frames in the packet */ - ist->next_pts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) / - avctx->sample_rate; - ist->next_dts += ((int64_t)AV_TIME_BASE * decoded_frame->nb_samples) / - avctx->sample_rate; -#endif - - resample_changed = ist->resample_sample_fmt != decoded_frame->format || - ist->resample_channels != avctx->channels || - ist->resample_channel_layout != decoded_frame->channel_layout || - ist->resample_sample_rate != decoded_frame->sample_rate; - if (resample_changed) { - char layout1[64], layout2[64]; - - if (!guess_input_channel_layout(ist)) { - av_log(NULL, AV_LOG_FATAL, "Unable to find default channel " - "layout for Input Stream #%d.%d\n", ist->file_index, - ist->st->index); - exit_program(1); - } - decoded_frame->channel_layout = avctx->channel_layout; - - av_get_channel_layout_string(layout1, sizeof(layout1), ist->resample_channels, - ist->resample_channel_layout); - av_get_channel_layout_string(layout2, sizeof(layout2), avctx->channels, - decoded_frame->channel_layout); - - av_log(NULL, AV_LOG_INFO, - "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d chl:%s to rate:%d fmt:%s ch:%d chl:%s\n", - ist->file_index, ist->st->index, - ist->resample_sample_rate, av_get_sample_fmt_name(ist->resample_sample_fmt), - ist->resample_channels, layout1, - decoded_frame->sample_rate, av_get_sample_fmt_name(decoded_frame->format), - avctx->channels, layout2); - - ist->resample_sample_fmt = decoded_frame->format; - ist->resample_sample_rate = decoded_frame->sample_rate; - ist->resample_channel_layout = decoded_frame->channel_layout; - ist->resample_channels = avctx->channels; - - for (i = 0; i < nb_filtergraphs; i++) - if (ist_in_filtergraph(filtergraphs[i], ist)) { - FilterGraph *fg = filtergraphs[i]; - int j; - if (configure_filtergraph(fg) < 0) { - av_log(NULL, AV_LOG_FATAL, "Error reinitializing filters!\n"); - exit_program(1); - } - for (j = 0; j < fg->nb_outputs; j++) { - OutputStream *ost = fg->outputs[j]->ost; - if (ost->enc->type == AVMEDIA_TYPE_AUDIO && - !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) - av_buffersink_set_frame_size(ost->filter->filter, - ost->st->codec->frame_size); - } - } - } - - /* if the decoder provides a pts, use it instead of the last packet pts. - the decoder could be delaying output by a packet or more. */ - if (decoded_frame->pts != AV_NOPTS_VALUE) { - ist->dts = ist->next_dts = ist->pts = ist->next_pts = av_rescale_q(decoded_frame->pts, avctx->time_base, AV_TIME_BASE_Q); - decoded_frame_tb = avctx->time_base; - } else if (decoded_frame->pkt_pts != AV_NOPTS_VALUE) { - decoded_frame->pts = decoded_frame->pkt_pts; - decoded_frame_tb = ist->st->time_base; - } else if (pkt->pts != AV_NOPTS_VALUE) { - decoded_frame->pts = pkt->pts; - decoded_frame_tb = ist->st->time_base; - }else { - decoded_frame->pts = ist->dts; - decoded_frame_tb = AV_TIME_BASE_Q; - } - pkt->pts = AV_NOPTS_VALUE; - if (decoded_frame->pts != AV_NOPTS_VALUE) - decoded_frame->pts = av_rescale_delta(decoded_frame_tb, decoded_frame->pts, - (AVRational){1, ist->st->codec->sample_rate}, decoded_frame->nb_samples, &ist->filter_in_rescale_delta_last, - (AVRational){1, ist->st->codec->sample_rate}); - for (i = 0; i < ist->nb_filters; i++) { - if (i < ist->nb_filters - 1) { - f = ist->filter_frame; - err = av_frame_ref(f, decoded_frame); - if (err < 0) - break; - } else - f = decoded_frame; - err = av_buffersrc_add_frame_flags(ist->filters[i]->filter, f, - AV_BUFFERSRC_FLAG_PUSH); - if (err == AVERROR_EOF) - err = 0; /* ignore */ - if (err < 0) - break; - } - decoded_frame->pts = AV_NOPTS_VALUE; - - av_frame_unref(ist->filter_frame); - av_frame_unref(decoded_frame); - return err < 0 ? err : ret; -} - -static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output) -{ - AVFrame *decoded_frame, *f; - int i, ret = 0, err = 0, resample_changed; - int64_t best_effort_timestamp; - AVRational *frame_sample_aspect; - - if (!ist->decoded_frame && !(ist->decoded_frame = av_frame_alloc())) - return AVERROR(ENOMEM); - if (!ist->filter_frame && !(ist->filter_frame = av_frame_alloc())) - return AVERROR(ENOMEM); - decoded_frame = ist->decoded_frame; - pkt->dts = av_rescale_q(ist->dts, AV_TIME_BASE_Q, ist->st->time_base); - - update_benchmark(NULL); - ret = avcodec_decode_video2(ist->st->codec, - decoded_frame, got_output, pkt); - update_benchmark("decode_video %d.%d", ist->file_index, ist->st->index); - - if (*got_output || ret<0 || pkt->size) - decode_error_stat[ret<0] ++; - - if (!*got_output || ret < 0) { - if (!pkt->size) { - for (i = 0; i < ist->nb_filters; i++) -#if 1 - av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0); -#else - av_buffersrc_add_frame(ist->filters[i]->filter, NULL); -#endif - } - return ret; - } - - if(ist->top_field_first>=0) - decoded_frame->top_field_first = ist->top_field_first; - - ist->frames_decoded++; - - if (ist->hwaccel_retrieve_data && decoded_frame->format == ist->hwaccel_pix_fmt) { - err = ist->hwaccel_retrieve_data(ist->st->codec, decoded_frame); - if (err < 0) - goto fail; - } - ist->hwaccel_retrieved_pix_fmt = decoded_frame->format; - - best_effort_timestamp= av_frame_get_best_effort_timestamp(decoded_frame); - if(best_effort_timestamp != AV_NOPTS_VALUE) - ist->next_pts = ist->pts = av_rescale_q(decoded_frame->pts = best_effort_timestamp, ist->st->time_base, AV_TIME_BASE_Q); - - if (debug_ts) { - av_log(NULL, AV_LOG_INFO, "decoder -> ist_index:%d type:video " - "frame_pts:%s frame_pts_time:%s best_effort_ts:%"PRId64" best_effort_ts_time:%s keyframe:%d frame_type:%d time_base:%d/%d\n", - ist->st->index, av_ts2str(decoded_frame->pts), - av_ts2timestr(decoded_frame->pts, &ist->st->time_base), - best_effort_timestamp, - av_ts2timestr(best_effort_timestamp, &ist->st->time_base), - decoded_frame->key_frame, decoded_frame->pict_type, - ist->st->time_base.num, ist->st->time_base.den); - } - - pkt->size = 0; - - if (ist->st->sample_aspect_ratio.num) - decoded_frame->sample_aspect_ratio = ist->st->sample_aspect_ratio; - - resample_changed = ist->resample_width != decoded_frame->width || - ist->resample_height != decoded_frame->height || - ist->resample_pix_fmt != decoded_frame->format; - if (resample_changed) { - av_log(NULL, AV_LOG_INFO, - "Input stream #%d:%d frame changed from size:%dx%d fmt:%s to size:%dx%d fmt:%s\n", - ist->file_index, ist->st->index, - ist->resample_width, ist->resample_height, av_get_pix_fmt_name(ist->resample_pix_fmt), - decoded_frame->width, decoded_frame->height, av_get_pix_fmt_name(decoded_frame->format)); - - ist->resample_width = decoded_frame->width; - ist->resample_height = decoded_frame->height; - ist->resample_pix_fmt = decoded_frame->format; - - for (i = 0; i < nb_filtergraphs; i++) { - if (ist_in_filtergraph(filtergraphs[i], ist) && ist->reinit_filters && - configure_filtergraph(filtergraphs[i]) < 0) { - av_log(NULL, AV_LOG_FATAL, "Error reinitializing filters!\n"); - exit_program(1); - } - } - } - - frame_sample_aspect= av_opt_ptr(avcodec_get_frame_class(), decoded_frame, "sample_aspect_ratio"); - for (i = 0; i < ist->nb_filters; i++) { - if (!frame_sample_aspect->num) - *frame_sample_aspect = ist->st->sample_aspect_ratio; - - if (i < ist->nb_filters - 1) { - f = ist->filter_frame; - err = av_frame_ref(f, decoded_frame); - if (err < 0) - break; - } else - f = decoded_frame; - ret = av_buffersrc_add_frame_flags(ist->filters[i]->filter, f, AV_BUFFERSRC_FLAG_PUSH); - if (ret == AVERROR_EOF) { - ret = 0; /* ignore */ - } else if (ret < 0) { - av_log(NULL, AV_LOG_FATAL, - "Failed to inject frame into filter network: %s\n", av_err2str(ret)); - exit_program(1); - } - } - -fail: - av_frame_unref(ist->filter_frame); - av_frame_unref(decoded_frame); - return err < 0 ? err : ret; -} - -static int transcode_subtitles(InputStream *ist, AVPacket *pkt, int *got_output) -{ - AVSubtitle subtitle; - int i, ret = avcodec_decode_subtitle2(ist->st->codec, - &subtitle, got_output, pkt); - - if (*got_output || ret<0 || pkt->size) - decode_error_stat[ret<0] ++; - - if (ret < 0 || !*got_output) { - if (!pkt->size) - sub2video_flush(ist); - return ret; - } - - if (ist->fix_sub_duration) { - int end = 1; - if (ist->prev_sub.got_output) { - end = av_rescale(subtitle.pts - ist->prev_sub.subtitle.pts, - 1000, AV_TIME_BASE); - if (end < ist->prev_sub.subtitle.end_display_time) { - av_log(ist->st->codec, AV_LOG_DEBUG, - "Subtitle duration reduced from %d to %d%s\n", - ist->prev_sub.subtitle.end_display_time, end, - end <= 0 ? ", dropping it" : ""); - ist->prev_sub.subtitle.end_display_time = end; - } - } - FFSWAP(int, *got_output, ist->prev_sub.got_output); - FFSWAP(int, ret, ist->prev_sub.ret); - FFSWAP(AVSubtitle, subtitle, ist->prev_sub.subtitle); - if (end <= 0) - goto out; - } - - if (!*got_output) - return ret; - - sub2video_update(ist, &subtitle); - - if (!subtitle.num_rects) - goto out; - - ist->frames_decoded++; - - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - - if (!check_output_constraints(ist, ost) || !ost->encoding_needed - || ost->enc->type != AVMEDIA_TYPE_SUBTITLE) - continue; - - do_subtitle_out(output_files[ost->file_index]->ctx, ost, ist, &subtitle); - } - -out: - avsubtitle_free(&subtitle); - return ret; -} - -/* pkt = NULL means EOF (needed to flush decoder buffers) */ -static int output_packet(InputStream *ist, const AVPacket *pkt) -{ - int ret = 0, i; - int got_output = 0; - - AVPacket avpkt; - if (!ist->saw_first_ts) { - ist->dts = ist->st->avg_frame_rate.num ? - ist->st->codec->has_b_frames * AV_TIME_BASE / av_q2d(ist->st->avg_frame_rate) : 0; - ist->pts = 0; - if (pkt != NULL && pkt->pts != AV_NOPTS_VALUE && !ist->decoding_needed) { - ist->dts += av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q); - ist->pts = ist->dts; //unused but better to set it to a value thats not totally wrong - } - ist->saw_first_ts = 1; - } - - if (ist->next_dts == AV_NOPTS_VALUE) - ist->next_dts = ist->dts; - if (ist->next_pts == AV_NOPTS_VALUE) - ist->next_pts = ist->pts; - - if (pkt == NULL) { - /* EOF handling */ - av_init_packet(&avpkt); - avpkt.data = NULL; - avpkt.size = 0; - goto handle_eof; - } else { - avpkt = *pkt; - } - - if (pkt->dts != AV_NOPTS_VALUE) { - ist->next_dts = ist->dts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q); - if (ist->st->codec->codec_type != AVMEDIA_TYPE_VIDEO || !ist->decoding_needed) - ist->next_pts = ist->pts = ist->dts; - } - - // while we have more to decode or while the decoder did output something on EOF - while (ist->decoding_needed && (avpkt.size > 0 || (!pkt && got_output))) { - int duration; - handle_eof: - - ist->pts = ist->next_pts; - ist->dts = ist->next_dts; - - if (avpkt.size && avpkt.size != pkt->size && - !(ist->dec->capabilities & CODEC_CAP_SUBFRAMES)) { - av_log(NULL, ist->showed_multi_packet_warning ? AV_LOG_VERBOSE : AV_LOG_WARNING, - "Multiple frames in a packet from stream %d\n", pkt->stream_index); - ist->showed_multi_packet_warning = 1; - } - - switch (ist->st->codec->codec_type) { - case AVMEDIA_TYPE_AUDIO: - ret = decode_audio (ist, &avpkt, &got_output); - break; - case AVMEDIA_TYPE_VIDEO: - ret = decode_video (ist, &avpkt, &got_output); - if (avpkt.duration) { - duration = av_rescale_q(avpkt.duration, ist->st->time_base, AV_TIME_BASE_Q); - } else if(ist->st->codec->time_base.num != 0 && ist->st->codec->time_base.den != 0) { - int ticks= ist->st->parser ? ist->st->parser->repeat_pict+1 : ist->st->codec->ticks_per_frame; - duration = ((int64_t)AV_TIME_BASE * - ist->st->codec->time_base.num * ticks) / - ist->st->codec->time_base.den; - } else - duration = 0; - - if(ist->dts != AV_NOPTS_VALUE && duration) { - ist->next_dts += duration; - }else - ist->next_dts = AV_NOPTS_VALUE; - - if (got_output) - ist->next_pts += duration; //FIXME the duration is not correct in some cases - break; - case AVMEDIA_TYPE_SUBTITLE: - ret = transcode_subtitles(ist, &avpkt, &got_output); - break; - default: - return -1; - } - - if (ret < 0) - return ret; - - avpkt.dts= - avpkt.pts= AV_NOPTS_VALUE; - - // touch data and size only if not EOF - if (pkt) { - if(ist->st->codec->codec_type != AVMEDIA_TYPE_AUDIO) - ret = avpkt.size; - avpkt.data += ret; - avpkt.size -= ret; - } - if (!got_output) { - continue; - } - } - - /* handle stream copy */ - if (!ist->decoding_needed) { - ist->dts = ist->next_dts; - switch (ist->st->codec->codec_type) { - case AVMEDIA_TYPE_AUDIO: - ist->next_dts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) / - ist->st->codec->sample_rate; - break; - case AVMEDIA_TYPE_VIDEO: - if (ist->framerate.num) { - // TODO: Remove work-around for c99-to-c89 issue 7 - AVRational time_base_q = AV_TIME_BASE_Q; - int64_t next_dts = av_rescale_q(ist->next_dts, time_base_q, av_inv_q(ist->framerate)); - ist->next_dts = av_rescale_q(next_dts + 1, av_inv_q(ist->framerate), time_base_q); - } else if (pkt->duration) { - ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q); - } else if(ist->st->codec->time_base.num != 0) { - int ticks= ist->st->parser ? ist->st->parser->repeat_pict + 1 : ist->st->codec->ticks_per_frame; - ist->next_dts += ((int64_t)AV_TIME_BASE * - ist->st->codec->time_base.num * ticks) / - ist->st->codec->time_base.den; - } - break; - } - ist->pts = ist->dts; - ist->next_pts = ist->next_dts; - } - for (i = 0; pkt && i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - - if (!check_output_constraints(ist, ost) || ost->encoding_needed) - continue; - - do_streamcopy(ist, ost, pkt); - } - - return 0; -} - -static void print_sdp(void) -{ - char sdp[16384]; - int i; - AVFormatContext **avc = av_malloc_array(nb_output_files, sizeof(*avc)); - - if (!avc) - exit_program(1); - for (i = 0; i < nb_output_files; i++) - avc[i] = output_files[i]->ctx; - - av_sdp_create(avc, nb_output_files, sdp, sizeof(sdp)); - printf("SDP:\n%s\n", sdp); - fflush(stdout); - av_freep(&avc); -} - -static const HWAccel *get_hwaccel(enum AVPixelFormat pix_fmt) -{ - int i; - for (i = 0; hwaccels[i].name; i++) - if (hwaccels[i].pix_fmt == pix_fmt) - return &hwaccels[i]; - return NULL; -} - -static enum AVPixelFormat get_format(AVCodecContext *s, const enum AVPixelFormat *pix_fmts) -{ - InputStream *ist = s->opaque; - const enum AVPixelFormat *p; - int ret; - - for (p = pix_fmts; *p != -1; p++) { - const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(*p); - const HWAccel *hwaccel; - - if (!(desc->flags & AV_PIX_FMT_FLAG_HWACCEL)) - break; - - hwaccel = get_hwaccel(*p); - if (!hwaccel || - (ist->active_hwaccel_id && ist->active_hwaccel_id != hwaccel->id) || - (ist->hwaccel_id != HWACCEL_AUTO && ist->hwaccel_id != hwaccel->id)) - continue; - - ret = hwaccel->init(s); - if (ret < 0) { - if (ist->hwaccel_id == hwaccel->id) { - av_log(NULL, AV_LOG_FATAL, - "%s hwaccel requested for input stream #%d:%d, " - "but cannot be initialized.\n", hwaccel->name, - ist->file_index, ist->st->index); - exit_program(1); - } - continue; - } - ist->active_hwaccel_id = hwaccel->id; - ist->hwaccel_pix_fmt = *p; - break; - } - - return *p; -} - -static int get_buffer(AVCodecContext *s, AVFrame *frame, int flags) -{ - InputStream *ist = s->opaque; - - if (ist->hwaccel_get_buffer && frame->format == ist->hwaccel_pix_fmt) - return ist->hwaccel_get_buffer(s, frame, flags); - - return avcodec_default_get_buffer2(s, frame, flags); -} - -static int init_input_stream(int ist_index, char *error, int error_len) -{ - int ret; - InputStream *ist = input_streams[ist_index]; - - if (ist->decoding_needed) { - AVCodec *codec = ist->dec; - if (!codec) { - snprintf(error, error_len, "Decoder (codec %s) not found for input stream #%d:%d", - avcodec_get_name(ist->st->codec->codec_id), ist->file_index, ist->st->index); - return AVERROR(EINVAL); - } - - ist->st->codec->opaque = ist; - ist->st->codec->get_format = get_format; - ist->st->codec->get_buffer2 = get_buffer; - ist->st->codec->thread_safe_callbacks = 1; - - av_opt_set_int(ist->st->codec, "refcounted_frames", 1, 0); - - if (!av_dict_get(ist->decoder_opts, "threads", NULL, 0)) - av_dict_set(&ist->decoder_opts, "threads", "auto", 0); - if ((ret = avcodec_open2(ist->st->codec, codec, &ist->decoder_opts)) < 0) { - if (ret == AVERROR_EXPERIMENTAL) - abort_codec_experimental(codec, 0); - - snprintf(error, error_len, - "Error while opening decoder for input stream " - "#%d:%d : %s", - ist->file_index, ist->st->index, av_err2str(ret)); - return ret; - } - assert_avoptions(ist->decoder_opts); - } - - ist->next_pts = AV_NOPTS_VALUE; - ist->next_dts = AV_NOPTS_VALUE; - - return 0; -} - -static InputStream *get_input_stream(OutputStream *ost) -{ - if (ost->source_index >= 0) - return input_streams[ost->source_index]; - return NULL; -} - -static int compare_int64(const void *a, const void *b) -{ - int64_t va = *(int64_t *)a, vb = *(int64_t *)b; - return va < vb ? -1 : va > vb ? +1 : 0; -} - -static void parse_forced_key_frames(char *kf, OutputStream *ost, - AVCodecContext *avctx) -{ - char *p; - int n = 1, i, size, index = 0; - int64_t t, *pts; - - for (p = kf; *p; p++) - if (*p == ',') - n++; - size = n; - pts = av_malloc_array(size, sizeof(*pts)); - if (!pts) { - av_log(NULL, AV_LOG_FATAL, "Could not allocate forced key frames array.\n"); - exit_program(1); - } - - p = kf; - for (i = 0; i < n; i++) { - char *next = strchr(p, ','); - - if (next) - *next++ = 0; - - if (!memcmp(p, "chapters", 8)) { - - AVFormatContext *avf = output_files[ost->file_index]->ctx; - int j; - - if (avf->nb_chapters > INT_MAX - size || - !(pts = av_realloc_f(pts, size += avf->nb_chapters - 1, - sizeof(*pts)))) { - av_log(NULL, AV_LOG_FATAL, - "Could not allocate forced key frames array.\n"); - exit_program(1); - } - t = p[8] ? parse_time_or_die("force_key_frames", p + 8, 1) : 0; - t = av_rescale_q(t, AV_TIME_BASE_Q, avctx->time_base); - - for (j = 0; j < avf->nb_chapters; j++) { - AVChapter *c = avf->chapters[j]; - av_assert1(index < size); - pts[index++] = av_rescale_q(c->start, c->time_base, - avctx->time_base) + t; - } - - } else { - - t = parse_time_or_die("force_key_frames", p, 1); - av_assert1(index < size); - pts[index++] = av_rescale_q(t, AV_TIME_BASE_Q, avctx->time_base); - - } - - p = next; - } - - av_assert0(index == size); - qsort(pts, size, sizeof(*pts), compare_int64); - ost->forced_kf_count = size; - ost->forced_kf_pts = pts; -} - -static void report_new_stream(int input_index, AVPacket *pkt) -{ - InputFile *file = input_files[input_index]; - AVStream *st = file->ctx->streams[pkt->stream_index]; - - if (pkt->stream_index < file->nb_streams_warn) - return; - av_log(file->ctx, AV_LOG_WARNING, - "New %s stream %d:%d at pos:%"PRId64" and DTS:%ss\n", - av_get_media_type_string(st->codec->codec_type), - input_index, pkt->stream_index, - pkt->pos, av_ts2timestr(pkt->dts, &st->time_base)); - file->nb_streams_warn = pkt->stream_index + 1; -} - -static void set_encoder_id(OutputFile *of, OutputStream *ost) -{ - AVDictionaryEntry *e; - - uint8_t *encoder_string; - int encoder_string_len; - int format_flags = 0; - int codec_flags = 0; - - if (av_dict_get(ost->st->metadata, "encoder", NULL, 0)) - return; - - e = av_dict_get(of->opts, "fflags", NULL, 0); - if (e) { - const AVOption *o = av_opt_find(of->ctx, "fflags", NULL, 0, 0); - if (!o) - return; - av_opt_eval_flags(of->ctx, o, e->value, &format_flags); - } - e = av_dict_get(ost->encoder_opts, "flags", NULL, 0); - if (e) { - const AVOption *o = av_opt_find(ost->st->codec, "flags", NULL, 0, 0); - if (!o) - return; - av_opt_eval_flags(ost->st->codec, o, e->value, &codec_flags); - } - - encoder_string_len = sizeof(LIBAVCODEC_IDENT) + strlen(ost->enc->name) + 2; - encoder_string = av_mallocz(encoder_string_len); - if (!encoder_string) - exit_program(1); - - if (!(format_flags & AVFMT_FLAG_BITEXACT) && !(codec_flags & CODEC_FLAG_BITEXACT)) - av_strlcpy(encoder_string, LIBAVCODEC_IDENT " ", encoder_string_len); - else - av_strlcpy(encoder_string, "Lavc ", encoder_string_len); - av_strlcat(encoder_string, ost->enc->name, encoder_string_len); - av_dict_set(&ost->st->metadata, "encoder", encoder_string, - AV_DICT_DONT_STRDUP_VAL | AV_DICT_DONT_OVERWRITE); -} - -static int transcode_init(void) -{ - int ret = 0, i, j, k; - AVFormatContext *oc; - OutputStream *ost; - InputStream *ist; - char error[1024]; - int want_sdp = 1; - - for (i = 0; i < nb_filtergraphs; i++) { - FilterGraph *fg = filtergraphs[i]; - for (j = 0; j < fg->nb_outputs; j++) { - OutputFilter *ofilter = fg->outputs[j]; - if (!ofilter->ost || ofilter->ost->source_index >= 0) - continue; - if (fg->nb_inputs != 1) - continue; - for (k = nb_input_streams-1; k >= 0 ; k--) - if (fg->inputs[0]->ist == input_streams[k]) - break; - ofilter->ost->source_index = k; - } - } - - /* init framerate emulation */ - for (i = 0; i < nb_input_files; i++) { - InputFile *ifile = input_files[i]; - if (ifile->rate_emu) - for (j = 0; j < ifile->nb_streams; j++) - input_streams[j + ifile->ist_index]->start = av_gettime_relative(); - } - - /* output stream init */ - for (i = 0; i < nb_output_files; i++) { - oc = output_files[i]->ctx; - if (!oc->nb_streams && !(oc->oformat->flags & AVFMT_NOSTREAMS)) { - av_dump_format(oc, i, oc->filename, 1); - av_log(NULL, AV_LOG_ERROR, "Output file #%d does not contain any stream\n", i); - return AVERROR(EINVAL); - } - } - - /* init complex filtergraphs */ - for (i = 0; i < nb_filtergraphs; i++) - if ((ret = avfilter_graph_config(filtergraphs[i]->graph, NULL)) < 0) - return ret; - - /* for each output stream, we compute the right encoding parameters */ - for (i = 0; i < nb_output_streams; i++) { - AVCodecContext *enc_ctx; - AVCodecContext *dec_ctx = NULL; - ost = output_streams[i]; - oc = output_files[ost->file_index]->ctx; - ist = get_input_stream(ost); - - if (ost->attachment_filename) - continue; - - enc_ctx = ost->st->codec; - - if (ist) { - dec_ctx = ist->st->codec; - - ost->st->disposition = ist->st->disposition; - enc_ctx->bits_per_raw_sample = dec_ctx->bits_per_raw_sample; - enc_ctx->chroma_sample_location = dec_ctx->chroma_sample_location; - } else { - for (j=0; jnb_streams; j++) { - AVStream *st = oc->streams[j]; - if (st != ost->st && st->codec->codec_type == enc_ctx->codec_type) - break; - } - if (j == oc->nb_streams) - if (enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO || enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO) - ost->st->disposition = AV_DISPOSITION_DEFAULT; - } - - if (ost->stream_copy) { - AVRational sar; - uint64_t extra_size; - - av_assert0(ist && !ost->filter); - - extra_size = (uint64_t)dec_ctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE; - - if (extra_size > INT_MAX) { - return AVERROR(EINVAL); - } - - /* if stream_copy is selected, no need to decode or encode */ - enc_ctx->codec_id = dec_ctx->codec_id; - enc_ctx->codec_type = dec_ctx->codec_type; - - if (!enc_ctx->codec_tag) { - unsigned int codec_tag; - if (!oc->oformat->codec_tag || - av_codec_get_id (oc->oformat->codec_tag, dec_ctx->codec_tag) == enc_ctx->codec_id || - !av_codec_get_tag2(oc->oformat->codec_tag, dec_ctx->codec_id, &codec_tag)) - enc_ctx->codec_tag = dec_ctx->codec_tag; - } - - enc_ctx->bit_rate = dec_ctx->bit_rate; - enc_ctx->rc_max_rate = dec_ctx->rc_max_rate; - enc_ctx->rc_buffer_size = dec_ctx->rc_buffer_size; - enc_ctx->field_order = dec_ctx->field_order; - enc_ctx->extradata = av_mallocz(extra_size); - if (!enc_ctx->extradata) { - return AVERROR(ENOMEM); - } - memcpy(enc_ctx->extradata, dec_ctx->extradata, dec_ctx->extradata_size); - enc_ctx->extradata_size= dec_ctx->extradata_size; - enc_ctx->bits_per_coded_sample = dec_ctx->bits_per_coded_sample; - - enc_ctx->time_base = ist->st->time_base; - /* - * Avi is a special case here because it supports variable fps but - * having the fps and timebase differe significantly adds quite some - * overhead - */ - if(!strcmp(oc->oformat->name, "avi")) { - if ( copy_tb<0 && av_q2d(ist->st->r_frame_rate) >= av_q2d(ist->st->avg_frame_rate) - && 0.5/av_q2d(ist->st->r_frame_rate) > av_q2d(ist->st->time_base) - && 0.5/av_q2d(ist->st->r_frame_rate) > av_q2d(dec_ctx->time_base) - && av_q2d(ist->st->time_base) < 1.0/500 && av_q2d(dec_ctx->time_base) < 1.0/500 - || copy_tb==2){ - enc_ctx->time_base.num = ist->st->r_frame_rate.den; - enc_ctx->time_base.den = 2*ist->st->r_frame_rate.num; - enc_ctx->ticks_per_frame = 2; - } else if ( copy_tb<0 && av_q2d(dec_ctx->time_base)*dec_ctx->ticks_per_frame > 2*av_q2d(ist->st->time_base) - && av_q2d(ist->st->time_base) < 1.0/500 - || copy_tb==0){ - enc_ctx->time_base = dec_ctx->time_base; - enc_ctx->time_base.num *= dec_ctx->ticks_per_frame; - enc_ctx->time_base.den *= 2; - enc_ctx->ticks_per_frame = 2; - } - } else if(!(oc->oformat->flags & AVFMT_VARIABLE_FPS) - && strcmp(oc->oformat->name, "mov") && strcmp(oc->oformat->name, "mp4") && strcmp(oc->oformat->name, "3gp") - && strcmp(oc->oformat->name, "3g2") && strcmp(oc->oformat->name, "psp") && strcmp(oc->oformat->name, "ipod") - && strcmp(oc->oformat->name, "f4v") - ) { - if( copy_tb<0 && dec_ctx->time_base.den - && av_q2d(dec_ctx->time_base)*dec_ctx->ticks_per_frame > av_q2d(ist->st->time_base) - && av_q2d(ist->st->time_base) < 1.0/500 - || copy_tb==0){ - enc_ctx->time_base = dec_ctx->time_base; - enc_ctx->time_base.num *= dec_ctx->ticks_per_frame; - } - } - if ( enc_ctx->codec_tag == AV_RL32("tmcd") - && dec_ctx->time_base.num < dec_ctx->time_base.den - && dec_ctx->time_base.num > 0 - && 121LL*dec_ctx->time_base.num > dec_ctx->time_base.den) { - enc_ctx->time_base = dec_ctx->time_base; - } - - if (ist && !ost->frame_rate.num) - ost->frame_rate = ist->framerate; - if(ost->frame_rate.num) - enc_ctx->time_base = av_inv_q(ost->frame_rate); - - av_reduce(&enc_ctx->time_base.num, &enc_ctx->time_base.den, - enc_ctx->time_base.num, enc_ctx->time_base.den, INT_MAX); - - ost->parser = av_parser_init(enc_ctx->codec_id); - - switch (enc_ctx->codec_type) { - case AVMEDIA_TYPE_AUDIO: - if (audio_volume != 256) { - av_log(NULL, AV_LOG_FATAL, "-acodec copy and -vol are incompatible (frames are not decoded)\n"); - exit_program(1); - } - enc_ctx->channel_layout = dec_ctx->channel_layout; - enc_ctx->sample_rate = dec_ctx->sample_rate; - enc_ctx->channels = dec_ctx->channels; - enc_ctx->frame_size = dec_ctx->frame_size; - enc_ctx->audio_service_type = dec_ctx->audio_service_type; - enc_ctx->block_align = dec_ctx->block_align; - enc_ctx->delay = dec_ctx->delay; - if((enc_ctx->block_align == 1 || enc_ctx->block_align == 1152 || enc_ctx->block_align == 576) && enc_ctx->codec_id == AV_CODEC_ID_MP3) - enc_ctx->block_align= 0; - if(enc_ctx->codec_id == AV_CODEC_ID_AC3) - enc_ctx->block_align= 0; - break; - case AVMEDIA_TYPE_VIDEO: - enc_ctx->pix_fmt = dec_ctx->pix_fmt; - enc_ctx->width = dec_ctx->width; - enc_ctx->height = dec_ctx->height; - enc_ctx->has_b_frames = dec_ctx->has_b_frames; - if (ost->frame_aspect_ratio.num) { // overridden by the -aspect cli option - sar = - av_mul_q(ost->frame_aspect_ratio, - (AVRational){ enc_ctx->height, enc_ctx->width }); - av_log(NULL, AV_LOG_WARNING, "Overriding aspect ratio " - "with stream copy may produce invalid files\n"); - } - else if (ist->st->sample_aspect_ratio.num) - sar = ist->st->sample_aspect_ratio; - else - sar = dec_ctx->sample_aspect_ratio; - ost->st->sample_aspect_ratio = enc_ctx->sample_aspect_ratio = sar; - ost->st->avg_frame_rate = ist->st->avg_frame_rate; - break; - case AVMEDIA_TYPE_SUBTITLE: - enc_ctx->width = dec_ctx->width; - enc_ctx->height = dec_ctx->height; - break; - case AVMEDIA_TYPE_DATA: - case AVMEDIA_TYPE_ATTACHMENT: - break; - default: - abort(); - } - } else { - if (!ost->enc) - ost->enc = avcodec_find_encoder(enc_ctx->codec_id); - if (!ost->enc) { - /* should only happen when a default codec is not present. */ - snprintf(error, sizeof(error), "Encoder (codec %s) not found for output stream #%d:%d", - avcodec_get_name(ost->st->codec->codec_id), ost->file_index, ost->index); - ret = AVERROR(EINVAL); - goto dump_format; - } - - if (ist) - ist->decoding_needed++; - ost->encoding_needed = 1; - - set_encoder_id(output_files[ost->file_index], ost); - - if (!ost->filter && - (enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO || - enc_ctx->codec_type == AVMEDIA_TYPE_AUDIO)) { - FilterGraph *fg; - fg = init_simple_filtergraph(ist, ost); - if (configure_filtergraph(fg)) { - av_log(NULL, AV_LOG_FATAL, "Error opening filters!\n"); - exit_program(1); - } - } - - if (enc_ctx->codec_type == AVMEDIA_TYPE_VIDEO) { - if (ost->filter && !ost->frame_rate.num) - ost->frame_rate = av_buffersink_get_frame_rate(ost->filter->filter); - if (ist && !ost->frame_rate.num) - ost->frame_rate = ist->framerate; - if (ist && !ost->frame_rate.num) - ost->frame_rate = ist->st->r_frame_rate; - if (ist && !ost->frame_rate.num) { - ost->frame_rate = (AVRational){25, 1}; - av_log(NULL, AV_LOG_WARNING, - "No information " - "about the input framerate is available. Falling " - "back to a default value of 25fps for output stream #%d:%d. Use the -r option " - "if you want a different framerate.\n", - ost->file_index, ost->index); - } -// ost->frame_rate = ist->st->avg_frame_rate.num ? ist->st->avg_frame_rate : (AVRational){25, 1}; - if (ost->enc && ost->enc->supported_framerates && !ost->force_fps) { - int idx = av_find_nearest_q_idx(ost->frame_rate, ost->enc->supported_framerates); - ost->frame_rate = ost->enc->supported_framerates[idx]; - } - if (enc_ctx->codec_id == AV_CODEC_ID_MPEG4) { - av_reduce(&ost->frame_rate.num, &ost->frame_rate.den, - ost->frame_rate.num, ost->frame_rate.den, 65535); - } - } - - switch (enc_ctx->codec_type) { - case AVMEDIA_TYPE_AUDIO: - enc_ctx->sample_fmt = ost->filter->filter->inputs[0]->format; - enc_ctx->sample_rate = ost->filter->filter->inputs[0]->sample_rate; - enc_ctx->channel_layout = ost->filter->filter->inputs[0]->channel_layout; - enc_ctx->channels = avfilter_link_get_channels(ost->filter->filter->inputs[0]); - enc_ctx->time_base = (AVRational){ 1, enc_ctx->sample_rate }; - break; - case AVMEDIA_TYPE_VIDEO: - enc_ctx->time_base = av_inv_q(ost->frame_rate); - if (ost->filter && !(enc_ctx->time_base.num && enc_ctx->time_base.den)) - enc_ctx->time_base = ost->filter->filter->inputs[0]->time_base; - if ( av_q2d(enc_ctx->time_base) < 0.001 && video_sync_method != VSYNC_PASSTHROUGH - && (video_sync_method == VSYNC_CFR || video_sync_method == VSYNC_VSCFR || (video_sync_method == VSYNC_AUTO && !(oc->oformat->flags & AVFMT_VARIABLE_FPS)))){ - av_log(oc, AV_LOG_WARNING, "Frame rate very high for a muxer not efficiently supporting it.\n" - "Please consider specifying a lower framerate, a different muxer or -vsync 2\n"); - } - for (j = 0; j < ost->forced_kf_count; j++) - ost->forced_kf_pts[j] = av_rescale_q(ost->forced_kf_pts[j], - AV_TIME_BASE_Q, - enc_ctx->time_base); - - enc_ctx->width = ost->filter->filter->inputs[0]->w; - enc_ctx->height = ost->filter->filter->inputs[0]->h; - enc_ctx->sample_aspect_ratio = ost->st->sample_aspect_ratio = - ost->frame_aspect_ratio.num ? // overridden by the -aspect cli option - av_mul_q(ost->frame_aspect_ratio, (AVRational){ enc_ctx->height, enc_ctx->width }) : - ost->filter->filter->inputs[0]->sample_aspect_ratio; - if (!strncmp(ost->enc->name, "libx264", 7) && - enc_ctx->pix_fmt == AV_PIX_FMT_NONE && - ost->filter->filter->inputs[0]->format != AV_PIX_FMT_YUV420P) - av_log(NULL, AV_LOG_WARNING, - "No pixel format specified, %s for H.264 encoding chosen.\n" - "Use -pix_fmt yuv420p for compatibility with outdated media players.\n", - av_get_pix_fmt_name(ost->filter->filter->inputs[0]->format)); - if (!strncmp(ost->enc->name, "mpeg2video", 10) && - enc_ctx->pix_fmt == AV_PIX_FMT_NONE && - ost->filter->filter->inputs[0]->format != AV_PIX_FMT_YUV420P) - av_log(NULL, AV_LOG_WARNING, - "No pixel format specified, %s for MPEG-2 encoding chosen.\n" - "Use -pix_fmt yuv420p for compatibility with outdated media players.\n", - av_get_pix_fmt_name(ost->filter->filter->inputs[0]->format)); - enc_ctx->pix_fmt = ost->filter->filter->inputs[0]->format; - - ost->st->avg_frame_rate = ost->frame_rate; - - if (!dec_ctx || - enc_ctx->width != dec_ctx->width || - enc_ctx->height != dec_ctx->height || - enc_ctx->pix_fmt != dec_ctx->pix_fmt) { - enc_ctx->bits_per_raw_sample = frame_bits_per_raw_sample; - } - - if (ost->forced_keyframes) { - if (!strncmp(ost->forced_keyframes, "expr:", 5)) { - ret = av_expr_parse(&ost->forced_keyframes_pexpr, ost->forced_keyframes+5, - forced_keyframes_const_names, NULL, NULL, NULL, NULL, 0, NULL); - if (ret < 0) { - av_log(NULL, AV_LOG_ERROR, - "Invalid force_key_frames expression '%s'\n", ost->forced_keyframes+5); - return ret; - } - ost->forced_keyframes_expr_const_values[FKF_N] = 0; - ost->forced_keyframes_expr_const_values[FKF_N_FORCED] = 0; - ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N] = NAN; - ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_T] = NAN; - } else { - parse_forced_key_frames(ost->forced_keyframes, ost, ost->st->codec); - } - } - break; - case AVMEDIA_TYPE_SUBTITLE: - enc_ctx->time_base = (AVRational){1, 1000}; - if (!enc_ctx->width) { - enc_ctx->width = input_streams[ost->source_index]->st->codec->width; - enc_ctx->height = input_streams[ost->source_index]->st->codec->height; - } - break; - default: - abort(); - break; - } - /* two pass mode */ - if (enc_ctx->flags & (CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2)) { - char logfilename[1024]; - FILE *f; - - snprintf(logfilename, sizeof(logfilename), "%s-%d.log", - ost->logfile_prefix ? ost->logfile_prefix : - DEFAULT_PASS_LOGFILENAME_PREFIX, - i); - if (!strcmp(ost->enc->name, "libx264")) { - av_dict_set(&ost->encoder_opts, "stats", logfilename, AV_DICT_DONT_OVERWRITE); - } else { - if (enc_ctx->flags & CODEC_FLAG_PASS2) { - char *logbuffer; - size_t logbuffer_size; - if (cmdutils_read_file(logfilename, &logbuffer, &logbuffer_size) < 0) { - av_log(NULL, AV_LOG_FATAL, "Error reading log file '%s' for pass-2 encoding\n", - logfilename); - exit_program(1); - } - enc_ctx->stats_in = logbuffer; - } - if (enc_ctx->flags & CODEC_FLAG_PASS1) { - f = av_fopen_utf8(logfilename, "wb"); - if (!f) { - av_log(NULL, AV_LOG_FATAL, "Cannot write log file '%s' for pass-1 encoding: %s\n", - logfilename, strerror(errno)); - exit_program(1); - } - ost->logfile = f; - } - } - } - } - } - - /* open each encoder */ - for (i = 0; i < nb_output_streams; i++) { - ost = output_streams[i]; - if (ost->encoding_needed) { - AVCodec *codec = ost->enc; - AVCodecContext *dec = NULL; - - if ((ist = get_input_stream(ost))) - dec = ist->st->codec; - if (dec && dec->subtitle_header) { - /* ASS code assumes this buffer is null terminated so add extra byte. */ - ost->st->codec->subtitle_header = av_mallocz(dec->subtitle_header_size + 1); - if (!ost->st->codec->subtitle_header) { - ret = AVERROR(ENOMEM); - goto dump_format; - } - memcpy(ost->st->codec->subtitle_header, dec->subtitle_header, dec->subtitle_header_size); - ost->st->codec->subtitle_header_size = dec->subtitle_header_size; - } - if (!av_dict_get(ost->encoder_opts, "threads", NULL, 0)) - av_dict_set(&ost->encoder_opts, "threads", "auto", 0); - if ((ret = avcodec_open2(ost->st->codec, codec, &ost->encoder_opts)) < 0) { - if (ret == AVERROR_EXPERIMENTAL) - abort_codec_experimental(codec, 1); - snprintf(error, sizeof(error), "Error while opening encoder for output stream #%d:%d - maybe incorrect parameters such as bit_rate, rate, width or height", - ost->file_index, ost->index); - goto dump_format; - } - if (ost->enc->type == AVMEDIA_TYPE_AUDIO && - !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)) - av_buffersink_set_frame_size(ost->filter->filter, - ost->st->codec->frame_size); - assert_avoptions(ost->encoder_opts); - if (ost->st->codec->bit_rate && ost->st->codec->bit_rate < 1000) - av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low." - " It takes bits/s as argument, not kbits/s\n"); - } else { - av_opt_set_dict(ost->st->codec, &ost->encoder_opts); - } - } - - /* init input streams */ - for (i = 0; i < nb_input_streams; i++) - if ((ret = init_input_stream(i, error, sizeof(error))) < 0) { - for (i = 0; i < nb_output_streams; i++) { - ost = output_streams[i]; - avcodec_close(ost->st->codec); - } - goto dump_format; - } - - /* discard unused programs */ - for (i = 0; i < nb_input_files; i++) { - InputFile *ifile = input_files[i]; - for (j = 0; j < ifile->ctx->nb_programs; j++) { - AVProgram *p = ifile->ctx->programs[j]; - int discard = AVDISCARD_ALL; - - for (k = 0; k < p->nb_stream_indexes; k++) - if (!input_streams[ifile->ist_index + p->stream_index[k]]->discard) { - discard = AVDISCARD_DEFAULT; - break; - } - p->discard = discard; - } - } - - /* open files and write file headers */ - for (i = 0; i < nb_output_files; i++) { - oc = output_files[i]->ctx; - oc->interrupt_callback = int_cb; - if ((ret = avformat_write_header(oc, &output_files[i]->opts)) < 0) { - snprintf(error, sizeof(error), - "Could not write header for output file #%d " - "(incorrect codec parameters ?): %s", - i, av_err2str(ret)); - ret = AVERROR(EINVAL); - goto dump_format; - } -// assert_avoptions(output_files[i]->opts); - if (strcmp(oc->oformat->name, "rtp")) { - want_sdp = 0; - } - } - - dump_format: - /* dump the file output parameters - cannot be done before in case - of stream copy */ - for (i = 0; i < nb_output_files; i++) { - av_dump_format(output_files[i]->ctx, i, output_files[i]->ctx->filename, 1); - } - - /* dump the stream mapping */ - av_log(NULL, AV_LOG_INFO, "Stream mapping:\n"); - for (i = 0; i < nb_input_streams; i++) { - ist = input_streams[i]; - - for (j = 0; j < ist->nb_filters; j++) { - if (ist->filters[j]->graph->graph_desc) { - av_log(NULL, AV_LOG_INFO, " Stream #%d:%d (%s) -> %s", - ist->file_index, ist->st->index, ist->dec ? ist->dec->name : "?", - ist->filters[j]->name); - if (nb_filtergraphs > 1) - av_log(NULL, AV_LOG_INFO, " (graph %d)", ist->filters[j]->graph->index); - av_log(NULL, AV_LOG_INFO, "\n"); - } - } - } - - for (i = 0; i < nb_output_streams; i++) { - ost = output_streams[i]; - - if (ost->attachment_filename) { - /* an attached file */ - av_log(NULL, AV_LOG_INFO, " File %s -> Stream #%d:%d\n", - ost->attachment_filename, ost->file_index, ost->index); - continue; - } - - if (ost->filter && ost->filter->graph->graph_desc) { - /* output from a complex graph */ - av_log(NULL, AV_LOG_INFO, " %s", ost->filter->name); - if (nb_filtergraphs > 1) - av_log(NULL, AV_LOG_INFO, " (graph %d)", ost->filter->graph->index); - - av_log(NULL, AV_LOG_INFO, " -> Stream #%d:%d (%s)\n", ost->file_index, - ost->index, ost->enc ? ost->enc->name : "?"); - continue; - } - - av_log(NULL, AV_LOG_INFO, " Stream #%d:%d -> #%d:%d", - input_streams[ost->source_index]->file_index, - input_streams[ost->source_index]->st->index, - ost->file_index, - ost->index); - if (ost->sync_ist != input_streams[ost->source_index]) - av_log(NULL, AV_LOG_INFO, " [sync #%d:%d]", - ost->sync_ist->file_index, - ost->sync_ist->st->index); - if (ost->stream_copy) - av_log(NULL, AV_LOG_INFO, " (copy)"); - else - av_log(NULL, AV_LOG_INFO, " (%s -> %s)", input_streams[ost->source_index]->dec ? - input_streams[ost->source_index]->dec->name : "?", - ost->enc ? ost->enc->name : "?"); - av_log(NULL, AV_LOG_INFO, "\n"); - } - - if (ret) { - av_log(NULL, AV_LOG_ERROR, "%s\n", error); - return ret; - } - - if (want_sdp) { - print_sdp(); - } - - transcode_init_done = 1; - - return 0; -} - -/* Return 1 if there remain streams where more output is wanted, 0 otherwise. */ -static int need_output(void) -{ - int i; - - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - OutputFile *of = output_files[ost->file_index]; - AVFormatContext *os = output_files[ost->file_index]->ctx; - - if (ost->finished || - (os->pb && avio_tell(os->pb) >= of->limit_filesize)) - continue; - if (ost->frame_number >= ost->max_frames) { - int j; - for (j = 0; j < of->ctx->nb_streams; j++) - close_output_stream(output_streams[of->ost_index + j]); - continue; - } - - return 1; - } - - return 0; -} - -/** - * Select the output stream to process. - * - * @return selected output stream, or NULL if none available - */ -static OutputStream *choose_output(void) -{ - int i; - int64_t opts_min = INT64_MAX; - OutputStream *ost_min = NULL; - - for (i = 0; i < nb_output_streams; i++) { - OutputStream *ost = output_streams[i]; - int64_t opts = av_rescale_q(ost->st->cur_dts, ost->st->time_base, - AV_TIME_BASE_Q); - if (!ost->unavailable && !ost->finished && opts < opts_min) { - opts_min = opts; - ost_min = ost; - } - } - return ost_min; -} - -static int check_keyboard_interaction(int64_t cur_time) -{ - int i, ret, key; - static int64_t last_time; - if (received_nb_signals) - return AVERROR_EXIT; - /* read_key() returns 0 on EOF */ - if(cur_time - last_time >= 100000 && !run_as_daemon){ - key = read_key(); - last_time = cur_time; - }else - key = -1; - if (key == 'q') - return AVERROR_EXIT; - if (key == '+') av_log_set_level(av_log_get_level()+10); - if (key == '-') av_log_set_level(av_log_get_level()-10); - if (key == 's') qp_hist ^= 1; - if (key == 'h'){ - if (do_hex_dump){ - do_hex_dump = do_pkt_dump = 0; - } else if(do_pkt_dump){ - do_hex_dump = 1; - } else - do_pkt_dump = 1; - av_log_set_level(AV_LOG_DEBUG); - } - if (key == 'c' || key == 'C'){ - char buf[4096], target[64], command[256], arg[256] = {0}; - double time; - int k, n = 0; - fprintf(stderr, "\nEnter command: |all