mirror of https://git.ffmpeg.org/ffmpeg.git
109 lines
3.0 KiB
C
109 lines
3.0 KiB
C
/*
|
|
* ALSA input and output
|
|
* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
|
|
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
|
|
*
|
|
* 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
|
|
* ALSA input and output: output
|
|
* @author Luca Abeni ( lucabe72 email it )
|
|
* @author Benoit Fouet ( benoit fouet free fr )
|
|
*
|
|
* This avdevice encoder allows to play audio to an ALSA (Advanced Linux
|
|
* Sound Architecture) device.
|
|
*
|
|
* The filename parameter is the name of an ALSA PCM device capable of
|
|
* capture, for example "default" or "plughw:1"; see the ALSA documentation
|
|
* for naming conventions. The empty string is equivalent to "default".
|
|
*
|
|
* The playback period is set to the lower value available for the device,
|
|
* which gives a low latency suitable for real-time playback.
|
|
*/
|
|
|
|
#include <alsa/asoundlib.h>
|
|
#include "libavformat/avformat.h"
|
|
|
|
#include "alsa-audio.h"
|
|
|
|
static av_cold int audio_write_header(AVFormatContext *s1)
|
|
{
|
|
AlsaData *s = s1->priv_data;
|
|
AVStream *st;
|
|
unsigned int sample_rate;
|
|
enum CodecID codec_id;
|
|
int res;
|
|
|
|
st = s1->streams[0];
|
|
sample_rate = st->codec->sample_rate;
|
|
codec_id = st->codec->codec_id;
|
|
res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
|
|
st->codec->channels, &codec_id);
|
|
if (sample_rate != st->codec->sample_rate) {
|
|
av_log(s1, AV_LOG_ERROR,
|
|
"sample rate %d not available, nearest is %d\n",
|
|
st->codec->sample_rate, sample_rate);
|
|
goto fail;
|
|
}
|
|
|
|
return res;
|
|
|
|
fail:
|
|
snd_pcm_close(s->h);
|
|
return AVERROR(EIO);
|
|
}
|
|
|
|
static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
|
|
{
|
|
AlsaData *s = s1->priv_data;
|
|
int res;
|
|
int size = pkt->size;
|
|
uint8_t *buf = pkt->data;
|
|
|
|
while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) {
|
|
if (res == -EAGAIN) {
|
|
|
|
return AVERROR(EAGAIN);
|
|
}
|
|
|
|
if (ff_alsa_xrun_recover(s1, res) < 0) {
|
|
av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
|
|
snd_strerror(res));
|
|
|
|
return AVERROR(EIO);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVOutputFormat alsa_muxer = {
|
|
"alsa",
|
|
NULL_IF_CONFIG_SMALL("ALSA audio output"),
|
|
"",
|
|
"",
|
|
sizeof(AlsaData),
|
|
DEFAULT_CODEC_ID,
|
|
CODEC_ID_NONE,
|
|
audio_write_header,
|
|
audio_write_packet,
|
|
ff_alsa_close,
|
|
.flags = AVFMT_NOFILE,
|
|
};
|