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/*
* 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 "avdevice.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;
}
av_set_pts_info(st, 64, 1, sample_rate);
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;
size /= s->frame_size;
if (s->reorder_func) {
if (size > s->reorder_buf_size)
if (ff_alsa_extend_reorder_buf(s, size))
return AVERROR(ENOMEM);
s->reorder_func(buf, s->reorder_buf, size);
buf = s->reorder_buf;
}
while ((res = snd_pcm_writei(s->h, buf, 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;
}
static void
audio_get_output_timestamp(AVFormatContext *s1, int stream,
int64_t *dts, int64_t *wall)
{
AlsaData *s = s1->priv_data;
snd_pcm_sframes_t delay = 0;
*wall = av_gettime();
snd_pcm_delay(s->h, &delay);
*dts = s1->streams[0]->cur_dts - delay;
}
AVOutputFormat ff_alsa_muxer = {
.name = "alsa",
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
.priv_data_size = sizeof(AlsaData),
.audio_codec = DEFAULT_CODEC_ID,
.video_codec = CODEC_ID_NONE,
.write_header = audio_write_header,
.write_packet = audio_write_packet,
.write_trailer = ff_alsa_close,
.get_output_timestamp = audio_get_output_timestamp,
.flags = AVFMT_NOFILE,
};
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