0,0 → 1,129 |
/* |
* 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 "libavutil/time.h" |
#include "libavformat/internal.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 AVCodecID 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; |
} |
avpriv_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 = AV_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, |
}; |