343 lines
11 KiB
C
343 lines
11 KiB
C
/*
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* ALSA input and output
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* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
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* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* ALSA input and output: output
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* @author Luca Abeni ( lucabe72 email it )
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* @author Benoit Fouet ( benoit fouet free fr )
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*
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* This avdevice encoder can play audio to an ALSA (Advanced Linux
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* Sound Architecture) device.
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*
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* The filename parameter is the name of an ALSA PCM device capable of
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* capture, for example "default" or "plughw:1"; see the ALSA documentation
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* for naming conventions. The empty string is equivalent to "default".
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*
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* The playback period is set to the lower value available for the device,
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* which gives a low latency suitable for real-time playback.
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*/
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#include <alsa/asoundlib.h>
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#include <poll.h>
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#include "alsa.h"
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#include "avdevice.h"
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#include "libavcodec/bsf.h"
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#include "libavformat/internal.h"
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#include "libavformat/mux.h"
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#include "libavutil/frame.h"
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#include "libavutil/internal.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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static int audio_capbility_query_ranges(struct AVOptionRanges **ranges, void *obj,
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const char *key, int flags)
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{
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return ff_audio_capbility_query_ranges(ranges, obj, key, flags, true);
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}
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static const AVClass alsa_cap_class = {
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.class_name = "ALSA outdev capbility",
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.item_name = av_default_item_name,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
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.query_ranges = audio_capbility_query_ranges,
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};
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static av_cold int audio_write_header(AVFormatContext *s1)
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{
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AlsaData *priv = s1->priv_data;
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AVStream *st = NULL;
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unsigned int sample_rate;
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enum AVCodecID codec_id;
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int res;
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if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
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av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
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return AVERROR(EINVAL);
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}
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st = s1->streams[0];
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sample_rate = st->codecpar->sample_rate;
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codec_id = st->codecpar->codec_id;
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res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
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st->codecpar->ch_layout.nb_channels, &codec_id);
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if (sample_rate != st->codecpar->sample_rate) {
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av_log(s1, AV_LOG_ERROR,
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"sample rate %d not available, nearest is %d\n",
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st->codecpar->sample_rate, sample_rate);
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goto fail;
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}
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avpriv_set_pts_info(st, 64, 1, sample_rate);
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priv->running = 1;
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priv->timestamp = 0;
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return res;
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fail:
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snd_pcm_close(priv->h);
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return AVERROR(EIO);
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}
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static int audio_write_trailer(struct AVFormatContext *s1)
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{
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FFStream *const sti = ffstream(s1->streams[0]);
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AlsaData *priv = s1->priv_data;
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ff_alsa_close(s1);
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priv->timestamp = 0;
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sti->cur_dts = 0;
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priv->running = 0;
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if (sti->bsfc) {
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av_bsf_flush(sti->bsfc);
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av_bsf_free(&sti->bsfc);
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sti->bitstream_checked = 0;
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}
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return 0;
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}
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static int audio_write_lastpacket(AVFormatContext *s1)
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{
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AlsaData *priv = s1->priv_data;
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int ret;
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ret = snd_pcm_writei(priv->h, priv->lastpkt->data, priv->lastpkt->size / priv->frame_size);
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av_log(s1, AV_LOG_TRACE, "audio_write_lastpacket->snd_pcm_writei(%p, %p, %d %d)\n", priv->h, priv->lastpkt->data, priv->lastpkt->size / priv->frame_size, ret);
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if (ret < 0) {
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if (ff_alsa_xrun_recover(s1, ret) < 0)
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return AVERROR(EIO);
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ret = snd_pcm_writei(priv->h, priv->lastpkt->data, priv->lastpkt->size / priv->frame_size);
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if (ret < 0)
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return AVERROR(EAGAIN);
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}
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priv->timestamp += ret;
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priv->lastpkt->data += ret * priv->frame_size;
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priv->lastpkt->size -= ret * priv->frame_size;
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if (priv->lastpkt->size)
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return AVERROR(EAGAIN);
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av_packet_free(&priv->lastpkt);
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return 0;
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}
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static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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AlsaData *priv = s1->priv_data;
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int res;
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int size = pkt->size;
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const uint8_t *buf = pkt->data;
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size /= priv->frame_size;
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if (!priv->running) {
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if (priv->lastpkt)
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av_packet_free(&priv->lastpkt);
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return AVERROR_EOF;
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}
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if (priv->reorder_func) {
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if (size > priv->reorder_buf_size)
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if (ff_alsa_extend_reorder_buf(priv, size))
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return AVERROR(ENOMEM);
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priv->reorder_func(buf, priv->reorder_buf, size);
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buf = priv->reorder_buf;
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}
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if (snd_pcm_state(priv->h) == SND_PCM_STATE_PAUSED)
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{
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priv->resume_min -= size;
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if (priv->resume_min <= 0)
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snd_pcm_pause(priv->h, 0);
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}
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if (priv->lastpkt)
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return audio_write_lastpacket(s1);
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res = snd_pcm_writei(priv->h, buf, size);
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av_log(s1, AV_LOG_TRACE, "audio_write_packet->snd_pcm_writei(%p, %p, %d %d)\n", priv->h, buf, size, res);
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if (res < 0) {
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if (ff_alsa_xrun_recover(s1, res) < 0)
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return AVERROR(EIO);
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res = snd_pcm_writei(priv->h, buf, size);
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if (res < 0)
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return AVERROR(EAGAIN);
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}
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priv->timestamp += res;
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if (res != size) {
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priv->lastpkt = av_packet_clone(pkt);
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priv->lastpkt->data += res * priv->frame_size;
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priv->lastpkt->size -= res * priv->frame_size;
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return AVERROR(EAGAIN);
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}
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return 0;
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}
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static int audio_write_frame(AVFormatContext *s1, int stream_index,
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AVFrame **frame, unsigned flags)
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{
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AlsaData *s = s1->priv_data;
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AVPacket pkt;
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/* ff_alsa_open() should have accepted only supported formats */
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if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
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return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ?
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AVERROR(EINVAL) : 0;
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/* set only used fields */
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pkt.data = (*frame)->data[0];
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pkt.size = (*frame)->nb_samples * s->frame_size;
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pkt.dts = (*frame)->pkt_dts;
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pkt.duration = (*frame)->duration;
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return audio_write_packet(s1, &pkt);
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}
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static int audio_control_message(struct AVFormatContext *s1, int type,
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void *data, size_t data_size)
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{
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AlsaData *priv = s1->priv_data;
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snd_pcm_t *pcm = priv->h;
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snd_pcm_sw_params_t *sw_params;
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switch (type) {
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case AV_APP_TO_DEV_GET_POLLFD: {
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struct pollfd *poll = data;
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int ret;
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if (!data || data_size < sizeof(struct pollfd))
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return AVERROR(EINVAL);
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if (!pcm || !priv->running)
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return 0;
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if (snd_pcm_state(pcm) == SND_PCM_STATE_PAUSED)
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return 0;
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ret = snd_pcm_poll_descriptors(pcm, poll, 1);
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if (ret < 0)
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return 0;
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return 1;
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}
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case AV_APP_TO_DEV_POLL_AVAILABLE: {
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enum AVDevToAppMessageType t;
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snd_pcm_sframes_t avail;
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if (!pcm)
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return 0;
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if (priv->running) {
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avail = snd_pcm_avail_update(pcm);
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if (avail == -EPIPE) {
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snd_pcm_pause(pcm, 1);
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snd_pcm_sw_params_alloca(&sw_params);
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snd_pcm_sw_params_current(priv->h, sw_params);
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priv->resume_min = sw_params->avail_min;
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}
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t = AV_DEV_TO_APP_BUFFER_WRITABLE;
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} else {
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t = AV_DEV_TO_APP_BUFFER_DRAINED;
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}
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avdevice_dev_to_app_control_message(s1, t, NULL, 0);
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return 0;
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}
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case AV_APP_TO_DEV_START: {
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snd_pcm_start(pcm);
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avdevice_dev_to_app_control_message(s1, AV_DEV_TO_APP_STATE_CHANGED, &type, 0);
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return 0;
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}
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case AV_APP_TO_DEV_STOP: {
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priv->running = 0;
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snd_pcm_close(pcm);
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avdevice_dev_to_app_control_message(s1, AV_DEV_TO_APP_STATE_CHANGED, &type, 0);
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return 0;
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}
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case AV_APP_TO_DEV_PAUSE:
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return snd_pcm_pause(pcm, 1);
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case AV_APP_TO_DEV_PLAY:
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return snd_pcm_pause(pcm, 0);
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case AV_APP_TO_DEV_DUMP:
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snprintf(data, data_size, "%s|%d %d", "alsa", priv->running, priv->running ? snd_pcm_state(pcm) : -1);
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return 0;
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}
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return AVERROR(ENOSYS);
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}
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static void
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audio_get_output_timestamp(AVFormatContext *s1, int stream,
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int64_t *dts, int64_t *wall)
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{
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AlsaData *s = s1->priv_data;
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snd_pcm_sframes_t delay = 0;
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*wall = av_gettime();
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snd_pcm_delay(s->h, &delay);
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*dts = s->timestamp - delay;
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}
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static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
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{
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return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
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}
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#define OFFSET(x) offsetof(AlsaData, x)
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#define FLAGS AV_OPT_FLAG_ENCODING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
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static const AVOption options[] = {
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{ "periods", "", OFFSET(periods), AV_OPT_TYPE_INT, {.i64 = 4}, 0, INT_MAX, FLAGS},
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{ "period_time", "", OFFSET(period_time), AV_OPT_TYPE_INT, {.i64 = 20}, 0, INT_MAX, FLAGS},
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{ NULL },
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};
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static const AVClass alsa_muxer_class = {
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.class_name = "ALSA outdev",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
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.query_ranges = audio_capbility_query_ranges,
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};
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const FFOutputFormat ff_alsa_muxer = {
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.p.name = "alsa",
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.p.long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
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.priv_data_size = sizeof(AlsaData),
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.p.audio_codec = DEFAULT_CODEC_ID,
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.p.video_codec = AV_CODEC_ID_NONE,
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.write_header = audio_write_header,
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.write_packet = audio_write_packet,
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.write_trailer = audio_write_trailer,
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.write_uncoded_frame = audio_write_frame,
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.control_message = audio_control_message,
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.get_device_list = audio_get_device_list,
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.get_output_timestamp = audio_get_output_timestamp,
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.p.flags = AVFMT_NOFILE | AVFMT_TS_NONSTRICT | AVFMT_NOTIMESTAMPS,
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.p.priv_class = &alsa_muxer_class,
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};
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