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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.
This commit is contained in:
openvela-robot 2014-05-24 11:15:15 +08:00
parent 909a4a960b
commit db600b2cf7
3692 changed files with 72748 additions and 326051 deletions
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38
.gitignore vendored
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@ -3,7 +3,6 @@
*.d
*.def
*.dll
*.dylib
*.exe
*.exp
*.gcda
@ -18,67 +17,44 @@
*.ver
*-example
*-test
*_g
/.config
/.version
/ffmpeg
/ffplay
/ffprobe
/ffserver
/avconv
/avplay
/avprobe
/avserver
/config.*
/coverage.info
/version.h
/doc/*.1
/doc/*.3
/doc/*.html
/doc/*.pod
/doc/config.texi
/doc/avoptions_codec.texi
/doc/avoptions_format.texi
/doc/doxy/html/
/doc/examples/avio_reading
/doc/examples/decoding_encoding
/doc/examples/demuxing_decoding
/doc/examples/avcodec
/doc/examples/filter_audio
/doc/examples/filtering_audio
/doc/examples/filtering_video
/doc/examples/metadata
/doc/examples/muxing
/doc/examples/pc-uninstalled
/doc/examples/remuxing
/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

59
CREDITS
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@ -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currently quite outdated and git serves as a much better tool for determining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.
Dénes 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
Aurélien Jacobs
Steven Johnson
Zdenek Kabelac
Robin Kay
Todd Kirby
Nick Kurshev
Benjamin Larsson
Loïc Le Loarer
Daniel Maas
Mike Melanson
Loren Merritt
Jeff Muizelaar
Michael Niedermayer
François Revol
Peter Ross
Måns Rullgård
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

575
Changelog
View File

@ -2,276 +2,141 @@ Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
version <next>:
- 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 <snmjbm at pacbell.net>)
- ARM optimizations (Lionel Ulmer <lionel.ulmer at free.fr>).
- Windows porting of file converter
- added MJPEG raw format (input/output)
- added MJPEG raw format (input/ouput)
- added JPEG image format support (input/output)

9
INSTALL
View File

@ -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.

70
LICENSE
View File

@ -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.

566
MAINTAINERS
View File

@ -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 <address>) 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 Árpád 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 Clément Bœsch
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* Clément Bœsch
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, Clément Bœsch
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 <khali@linux-fr.org>)
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* Clément Bœsch
jvdec.c Peter Ross
libmodplug.c Clément Bœsch
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
Bœsch Clément 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

135
Makefile
View File

@ -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*

12
README
View File

@ -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.

2
RELEASE
View File

@ -1 +1 @@
2.2.git
11_alpha1

5
arch.mak
View File

@ -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)

2633
avconv.c Normal file
View File

File diff suppressed because it is too large Load Diff

161
ffmpeg.h → avconv.h
View File

@ -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 <stdint.h>
#include <stdio.h>
#include <signal.h>
#if HAVE_PTHREADS
#include <pthread.h>
@ -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 */

13
ffmpeg_dxva2.c → avconv_dxva2.c
View File

@ -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 <d3d9.h>
#include <dxva2api.h>
#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]);
}

447
ffmpeg_filter.c → avconv_filter.c
View File

@ -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 <stdint.h>
#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; i<of->ctx->nb_streams; i++)
if (of->ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
break;
if (i<of->ctx->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;
}

980
ffmpeg_opt.c → avconv_opt.c
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File diff suppressed because it is too large Load Diff

10
ffmpeg_vda.c → avconv_vda.c
View File

@ -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;

10
ffmpeg_vdpau.c → avconv_vdpau.c
View File

@ -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 <X11/Xlib.h>
#include "ffmpeg.h"
#include "avconv.h"
#include "libavcodec/vdpau.h"

3001
avplay.c Normal file
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File diff suppressed because it is too large Load Diff

969
avprobe.c Normal file
View File

@ -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;
}

720
ffserver.c → avserver.c
View File

File diff suppressed because it is too large Load Diff

721
cmdutils.c
View File

File diff suppressed because it is too large Load Diff

119
cmdutils.h
View File

@ -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 <stdint.h>
#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 */

16
cmdutils_common_opts.h
View File

@ -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" },

274
cmdutils_opencl.c
View File

@ -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;
}

93
common.mak
View File

@ -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))

14
compat/aix/math.h
View File

@ -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 */

880
compat/avisynth/avisynth_c.h
View File

@ -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

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compat/avisynth/avisynth_c_25.h
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// 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); }

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compat/avisynth/avxsynth_c.h
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// 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 <stdarg.h>
#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__

85
compat/avisynth/windowsPorts/basicDataTypeConversions.h
View File

@ -1,85 +0,0 @@
#ifndef __DATA_TYPE_CONVERSIONS_H__
#define __DATA_TYPE_CONVERSIONS_H__
#include <stdint.h>
#include <wchar.h>
#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__

77
compat/avisynth/windowsPorts/windows2linux.h
View File

@ -1,77 +0,0 @@
#ifndef __WINDOWS2LINUX_H__
#define __WINDOWS2LINUX_H__
/*
* LINUX SPECIFIC DEFINITIONS
*/
//
// Data types conversions
//
#include <stdlib.h>
#include <string.h>
#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__

8
compat/float/float.h
View File

@ -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
*/

8
compat/float/limits.h
View File

@ -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
*/

11
compat/getopt.c
View File

@ -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);

13
compat/msvcrt/snprintf.c
View File

@ -2,20 +2,20 @@
* C99-compatible snprintf() and vsnprintf() implementations
* Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
*
* 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 <limits.h>
#include <string.h>
#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, ...)

38
compat/msvcrt/snprintf.h
View File

@ -1,38 +0,0 @@
/*
* C99-compatible snprintf() and vsnprintf() implementations
* Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
*
* 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 <stdarg.h>
#include <stdio.h>
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 */

164
compat/os2threads.h
View File

@ -1,164 +0,0 @@
/*
* Copyright (c) 2011 KO Myung-Hun <komh@chollian.net>
*
* 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 <os2.h>
#undef __STRICT_ANSI__ /* for _beginthread() */
#include <stdlib.h>
#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 */

8
compat/plan9/main.c
View File

@ -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
*/

8
compat/strtod.c
View File

@ -2,20 +2,20 @@
* C99-compatible strtod() implementation
* Copyright (c) 2012 Ronald S. Bultje <rsbultje@gmail.com>
*
* 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
*/

14
compat/tms470/math.h
View File

@ -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 <math.h>
@ -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 */

29
compat/va_copy.h
View File

@ -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 <stdarg.h>
#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

26
compat/w32pthreads.h
View File

@ -4,20 +4,20 @@
* Authors: Steven Walters <kemuri9@gmail.com>
* Pegasys Inc. <http://www.pegasys-inc.com>
*
* 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 <windows.h>
#include <process.h>
#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 */

1246
configure vendored
View File

File diff suppressed because it is too large Load Diff

1106
doc/APIchanges
View File

File diff suppressed because it is too large Load Diff

12
doc/Doxyfile
View File

@ -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

143
doc/Makefile
View File

@ -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

84
doc/RELEASE_NOTES
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@ -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.

11
doc/authors.texi
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@ -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.

809
doc/ffmpeg.texi → doc/avconv.texi
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184
doc/avplay.texi Normal file
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@ -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 <duration> 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 <number> 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

141
doc/avprobe.texi Normal file
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@ -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

35
doc/ffserver.conf → doc/avserver.conf
View File

@ -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.
<Feed feed1.ffm>
# 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.
<Stream test1.mpg>
@ -235,7 +236,7 @@ StartSendOnKey
#<Stream test.ogg>
#Feed feed1.ffm
#Metadata title "Stream title"
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
@ -280,10 +281,10 @@ StartSendOnKey
#<Stream file.asf>
#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"
#</Stream>
@ -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
<Redirect index.html>
URL http://www.ffmpeg.org/
URL http://www.libav.org/
</Redirect>

276
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@ -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 <Feed>
section in the configuration file.
For each feed you can have different output streams in various
formats, each one specified by a <Stream> 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
<Stream status.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</Stream>
@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://<linuxbox>: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 <Stream> 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

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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

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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)

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@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

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@ -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 <target>
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<num>
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.

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@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

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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;
}

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@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

419
doc/developer.texi
View File

@ -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<name>/lib<name>.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\+\%#\@@<!$/
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@example
(c-add-style "ffmpeg"
(c-add-style "libav"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
@ -237,7 +275,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
(setq c-default-style "libav")
@end example
@section Development Policy
@ -253,16 +291,20 @@ a gift-style license, the
@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
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.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
All the patches MUST be reviewed in the mailing list before they are
committed.
@item
The Libav coding style should remain consistent. Changes to
conform will be suggested during the review or implemented on commit.
@item
Patches should be generated using @code{git format-patch} or directly sent
using @code{git send-email}.
Please make sure you give the proper credit by setting the correct author
in the commit.
@item
The commit message should have a short first line in the form of
@ -271,12 +313,24 @@ from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
If the patch is a bug fix which should be backported to stable releases,
i.e. a non-API/ABI-breaking bug fix, add @code{CC: libav-stable@@libav.org}
to the bottom of your commit message, and make sure to CC your patch to
this address, too. Some git setups will do this automatically.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
Work in progress patches should be sent to the mailing list with the [WIP]
or the [RFC] tag.
@item
Branches in public personal repos are advised as way to
work on issues collaboratively.
@item
You do not have to over-test things. If it works for you and you think it
should work for others, send it to the mailing list for review.
If you have doubt about portability please state it in the submission so
people with specific hardware could test it.
@item
Do not commit unrelated changes together, split them into self-contained
@ -285,87 +339,44 @@ depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 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

25
doc/devices.texi
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@ -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

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@ -9,4 +9,7 @@ doxygen - <<EOF
@INCLUDE = ${DOXYFILE}
INPUT = $@
EXAMPLE_PATH = ${SRC_PATH}/doc/examples
HTML_HEADER = ${SRC_PATH}/doc/doxy/header.html
HTML_FOOTER = ${SRC_PATH}/doc/doxy/footer.html
HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
EOF

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@ -0,0 +1,9 @@
<footer class="footer pagination-right">
<span class="label label-info">
Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
</span>
</footer>
</div>
</body>
</html>

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@ -0,0 +1,16 @@
<!DOCTYPE html>
<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
<meta http-equiv="X-UA-Compatible" content="IE=9"/>
<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
<!--Header replace -->
</head>
<div class="container">
<!--Header replace -->
<div class="menu">

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@ -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

156
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@ -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

42
doc/examples/Makefile
View File

@ -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)

23
doc/examples/README
View File

@ -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.

339
doc/examples/decoding_encoding.c → doc/examples/avcodec.c
View File

@ -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 <math.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <libavutil/opt.h>
#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>
#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;y<c->height;y++) {
for(x=0;x<c->width;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;y<c->height/2;y++) {
for(x=0;x<c->width/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;i<ysize;i++)
fwrite(buf + i * wrap,1,xsize,f);
fclose(f);
}
static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
{
int len, got_frame;
char buf[1024];
len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
return len;
}
if (got_frame) {
printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
fflush(stdout);
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[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;
}

134
doc/examples/avio_reading.c
View File

@ -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 <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
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;
}

386
doc/examples/demuxing_decoding.c
View File

@ -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 <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
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=<old|new_norefcount|new_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;
}

8
doc/examples/filter_audio.c
View File

@ -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
*/

280
doc/examples/filtering_audio.c
View File

@ -1,280 +0,0 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* 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 <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
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);
}

261
doc/examples/filtering_video.c
View File

@ -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 <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
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);
}

4
doc/examples/metadata.c
View File

@ -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 <stdio.h>
@ -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;
}

606
doc/examples/muxing.c
View File

@ -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 <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
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;
}

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/*
* 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 <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#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;
}

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/*
* 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 <libavutil/timestamp.h>
#include <libavformat/avformat.h>
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;
}

215
doc/examples/resampling_audio.c
View File

@ -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 <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
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;
}

141
doc/examples/scaling_video.c
View File

@ -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 <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
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;
}

95
doc/examples/transcode_aac.c
View File

@ -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) {

597
doc/examples/transcoding.c
View File

@ -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 <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
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 <input file> <output file>\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;
}

314
doc/faq.texi
View File

@ -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?

249
doc/fate.texi
View File

@ -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=<path to the samples directory>'. 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=<path to the 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 <arch>-<os>-<compiler>-<compiler version>. 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}

29
doc/fate_config.sh.template
View File

@ -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'

45
doc/ffmpeg-bitstream-filters.texi
View File

@ -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

42
doc/ffmpeg-codecs.texi
View File

@ -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

42
doc/ffmpeg-devices.texi
View File

@ -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

42
doc/ffmpeg-filters.texi
View File

@ -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

42
doc/ffmpeg-formats.texi
View File

@ -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

42
doc/ffmpeg-protocols.texi
View File

@ -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

44
doc/ffmpeg-resampler.texi
View File

@ -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

43
doc/ffmpeg-scaler.texi
View File

@ -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

42
doc/ffmpeg-utils.texi
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@ -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

47
doc/ffmpeg.txt
View File

@ -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 :
:

293
doc/ffplay.texi
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@ -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 <duration> 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 <number> 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

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