ffmpeg/libavdevice/alsa_dec.c

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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: input
* @author Luca Abeni ( lucabe72 email it )
* @author Benoit Fouet ( benoit fouet free fr )
* @author Nicolas George ( nicolas george normalesup org )
*
* This avdevice decoder can 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 "libavutil/internal.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavformat/internal.h"
#include "avdevice.h"
#include "alsa.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 */
lavf: replace AVStream.codec with AVStream.codecpar Currently, AVStream contains an embedded AVCodecContext instance, which is used by demuxers to export stream parameters to the caller and by muxers to receive stream parameters from the caller. It is also used internally as the codec context that is passed to parsers. In addition, it is also widely used by the callers as the decoding (when demuxer) or encoding (when muxing) context, though this has been officially discouraged since Libav 11. There are multiple important problems with this approach: - the fields in AVCodecContext are in general one of * stream parameters * codec options * codec state However, it's not clear which ones are which. It is consequently unclear which fields are a demuxer allowed to set or a muxer allowed to read. This leads to erratic behaviour depending on whether decoding or encoding is being performed or not (and whether it uses the AVStream embedded codec context). - various synchronization issues arising from the fact that the same context is used by several different APIs (muxers/demuxers, parsers, bitstream filters and encoders/decoders) simultaneously, with there being no clear rules for who can modify what and the different processes being typically delayed with respect to each other. - avformat_find_stream_info() making it necessary to support opening and closing a single codec context multiple times, thus complicating the semantics of freeing various allocated objects in the codec context. Those problems are resolved by replacing the AVStream embedded codec context with a newly added AVCodecParameters instance, which stores only the stream parameters exported by the demuxers or read by the muxers.
2014-06-18 18:42:52 +00:00
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = codec_id;
st->codecpar->sample_rate = s->sample_rate;
st->codecpar->channels = s->channels;
st->codecpar->frame_size = s->frame_size;
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_packet_unref(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_packet_unref(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 int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
{
return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
}
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,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
};
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,
.get_device_list = audio_get_device_list,
.flags = AVFMT_NOFILE,
.priv_class = &alsa_demuxer_class,
};