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https://github.com/librempeg/librempeg
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419bddd366
This avoids symbol redefinitions problems, for example avoids the "free" symbol to be redefined before system headers actually using it are included, thus breaking compilation. In particular this change allows to build FFmpeg with salsa. Patch by matthieu castet <$surname.mat?hieu@free fr>. Originally committed as revision 20665 to svn://svn.ffmpeg.org/ffmpeg/trunk
109 lines
3.1 KiB
C
109 lines
3.1 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 libavdevice/alsa-audio-enc.c
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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 allows to 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 "libavformat/avformat.h"
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#include "alsa-audio.h"
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static av_cold int audio_write_header(AVFormatContext *s1)
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{
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AlsaData *s = s1->priv_data;
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AVStream *st;
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unsigned int sample_rate;
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enum CodecID codec_id;
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int res;
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st = s1->streams[0];
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sample_rate = st->codec->sample_rate;
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codec_id = st->codec->codec_id;
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res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
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st->codec->channels, &codec_id);
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if (sample_rate != st->codec->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->codec->sample_rate, sample_rate);
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goto fail;
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}
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return res;
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fail:
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snd_pcm_close(s->h);
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return AVERROR(EIO);
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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 *s = s1->priv_data;
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int res;
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int size = pkt->size;
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uint8_t *buf = pkt->data;
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while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) {
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if (res == -EAGAIN) {
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return AVERROR(EAGAIN);
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}
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if (ff_alsa_xrun_recover(s1, res) < 0) {
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av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
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snd_strerror(res));
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return AVERROR(EIO);
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}
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}
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return 0;
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}
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AVOutputFormat alsa_muxer = {
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"alsa",
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NULL_IF_CONFIG_SMALL("ALSA audio output"),
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"",
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"",
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sizeof(AlsaData),
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DEFAULT_CODEC_ID,
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CODEC_ID_NONE,
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audio_write_header,
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audio_write_packet,
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ff_alsa_close,
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.flags = AVFMT_NOFILE,
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};
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