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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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* ALSA input and output: input
* @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr )
* @author Nicolas George ( nicolas george normalesup org )
*
* This avdevice decoder allows to capture audio from 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 capture period is set to the lower value available for the device,
* which gives a low latency suitable for real-time capture.
*
* The PTS are an Unix time in microsecond.
*
* Due to a bug in the ALSA library
* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
* decoder does not work with certain ALSA plugins, especially the dsnoop
* plugin.
*/
#include <alsa/asoundlib.h>
#include "libavformat/avformat.h"
#include "libavutil/opt.h"
#include "alsa-audio.h"
static av_cold int audio_read_header(AVFormatContext *s1,
AVFormatParameters *ap)
{
AlsaData *s = s1->priv_data;
AVStream *st;
int ret;
enum CodecID codec_id;
snd_pcm_sw_params_t *sw_params;
st = av_new_stream(s1, 0);
if (!st) {
av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
codec_id = s1->audio_codec_id;
ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
&codec_id);
if (ret < 0) {
return AVERROR(EIO);
}
if (snd_pcm_type(s->h) != SND_PCM_TYPE_HW)
av_log(s1, AV_LOG_WARNING,
"capture with some ALSA plugins, especially dsnoop, "
"may hang.\n");
ret = snd_pcm_sw_params_malloc(&sw_params);
if (ret < 0) {
av_log(s1, AV_LOG_ERROR, "cannot allocate software parameters structure (%s)\n",
snd_strerror(ret));
goto fail;
}
snd_pcm_sw_params_current(s->h, sw_params);
snd_pcm_sw_params_set_tstamp_mode(s->h, sw_params, SND_PCM_TSTAMP_ENABLE);
ret = snd_pcm_sw_params(s->h, sw_params);
snd_pcm_sw_params_free(sw_params);
if (ret < 0) {
av_log(s1, AV_LOG_ERROR, "cannot install ALSA software parameters (%s)\n",
snd_strerror(ret));
goto fail;
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = codec_id;
st->codec->sample_rate = s->sample_rate;
st->codec->channels = s->channels;
av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
fail:
snd_pcm_close(s->h);
return AVERROR(EIO);
}
static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
AlsaData *s = s1->priv_data;
AVStream *st = s1->streams[0];
int res;
snd_htimestamp_t timestamp;
snd_pcm_uframes_t ts_delay;
if (av_new_packet(pkt, s->period_size) < 0) {
return AVERROR(EIO);
}
while ((res = snd_pcm_readi(s->h, pkt->data, pkt->size / s->frame_size)) < 0) {
if (res == -EAGAIN) {
av_free_packet(pkt);
return AVERROR(EAGAIN);
}
if (ff_alsa_xrun_recover(s1, res) < 0) {
av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
snd_strerror(res));
av_free_packet(pkt);
return AVERROR(EIO);
}
}
snd_pcm_htimestamp(s->h, &ts_delay, ×tamp);
ts_delay += res;
pkt->pts = timestamp.tv_sec * 1000000LL
+ (timestamp.tv_nsec * st->codec->sample_rate
- ts_delay * 1000000000LL + st->codec->sample_rate * 500LL)
/ (st->codec->sample_rate * 1000LL);
pkt->size = res * s->frame_size;
return 0;
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(AlsaData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(AlsaData, channels), FF_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass alsa_demuxer_class = {
.class_name = "ALSA demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_alsa_demuxer = {
"alsa",
NULL_IF_CONFIG_SMALL("ALSA audio input"),
sizeof(AlsaData),
NULL,
audio_read_header,
audio_read_packet,
ff_alsa_close,
.flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
};
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