mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-09-21 13:56:55 +00:00
9200514ad8
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.
118 lines
3.5 KiB
C
118 lines
3.5 KiB
C
/*
|
|
* 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: 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/internal.h"
|
|
|
|
#include "libavformat/avformat.h"
|
|
|
|
#include "alsa.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->codecpar->sample_rate;
|
|
codec_id = st->codecpar->codec_id;
|
|
res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
|
|
st->codecpar->channels, &codec_id);
|
|
if (sample_rate != st->codecpar->sample_rate) {
|
|
av_log(s1, AV_LOG_ERROR,
|
|
"sample rate %d not available, nearest is %d\n",
|
|
st->codecpar->sample_rate, sample_rate);
|
|
goto fail;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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,
|
|
.flags = AVFMT_NOFILE,
|
|
};
|