0,0 → 1,157 |
/* |
* 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: 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/internal.h" |
#include "libavutil/opt.h" |
#include "libavutil/mathematics.h" |
#include "libavutil/time.h" |
|
#include "avdevice.h" |
#include "alsa-audio.h" |
|
static av_cold int audio_read_header(AVFormatContext *s1) |
{ |
AlsaData *s = s1->priv_data; |
AVStream *st; |
int ret; |
enum AVCodecID codec_id; |
|
st = avformat_new_stream(s1, NULL); |
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); |
} |
|
/* 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; |
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ |
/* microseconds instead of seconds, MHz instead of Hz */ |
s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate, |
s->period_size, 1.5E-6); |
if (!s->timefilter) |
goto fail; |
|
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; |
int res; |
int64_t dts; |
snd_pcm_sframes_t delay = 0; |
|
if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) { |
return AVERROR(EIO); |
} |
|
while ((res = snd_pcm_readi(s->h, pkt->data, s->period_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); |
} |
ff_timefilter_reset(s->timefilter); |
} |
|
dts = av_gettime(); |
snd_pcm_delay(s->h, &delay); |
dts -= av_rescale(delay + res, 1000000, s->sample_rate); |
pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period); |
s->last_period = res; |
|
pkt->size = res * s->frame_size; |
|
return 0; |
} |
|
static const AVOption options[] = { |
{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, |
{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 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 = { |
.name = "alsa", |
.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"), |
.priv_data_size = sizeof(AlsaData), |
.read_header = audio_read_header, |
.read_packet = audio_read_packet, |
.read_close = ff_alsa_close, |
.flags = AVFMT_NOFILE, |
.priv_class = &alsa_demuxer_class, |
}; |