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FFmpeg/libavdevice/alsa_enc.c
Andreas Rheinhardt 0c6e5f321b avformat/avformat: Avoid including codec.h, frame.h
AVCodec is only ever used as an incomplete type (i.e. via a pointer
to an AVCodec) in avformat.h and it is not really part of the core
of avformat.h or libavformat; almost none of our internal users
make use of it (and none make use of hwcontext.h, which is implicitly
included). So switch to use struct AVCodec, but continue to include
codec.h for external users for compatibility.

Also, do the same for AVFrame and frame.h, which is implicitly included
by codec.h (via lavu/hwcontext.h).

Also, remove an unnecessary inclusion of <time.h>.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-09-07 00:30:08 +02:00

184 lines
5.7 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 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 can 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/frame.h"
#include "libavutil/internal.h"
#include "libavutil/time.h"
#include "libavformat/internal.h"
#include "libavformat/mux.h"
#include "avdevice.h"
#include "alsa.h"
static av_cold int audio_write_header(AVFormatContext *s1)
{
AlsaData *s = s1->priv_data;
AVStream *st = NULL;
unsigned int sample_rate;
enum AVCodecID codec_id;
int res;
if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
return AVERROR(EINVAL);
}
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->ch_layout.nb_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;
}
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;
const uint8_t *buf = pkt->data;
size /= s->frame_size;
if (pkt->dts != AV_NOPTS_VALUE)
s->timestamp = pkt->dts;
s->timestamp += pkt->duration ? pkt->duration : 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 int audio_write_frame(AVFormatContext *s1, int stream_index,
AVFrame **frame, unsigned flags)
{
AlsaData *s = s1->priv_data;
AVPacket pkt;
/* ff_alsa_open() should have accepted only supported formats */
if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ?
AVERROR(EINVAL) : 0;
/* set only used fields */
pkt.data = (*frame)->data[0];
pkt.size = (*frame)->nb_samples * s->frame_size;
pkt.dts = (*frame)->pkt_dts;
#if FF_API_PKT_DURATION
FF_DISABLE_DEPRECATION_WARNINGS
if ((*frame)->pkt_duration)
pkt.duration = (*frame)->pkt_duration;
else
FF_ENABLE_DEPRECATION_WARNINGS
#endif
pkt.duration = (*frame)->duration;
return audio_write_packet(s1, &pkt);
}
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 = s->timestamp - delay;
}
static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
{
return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
}
static const AVClass alsa_muxer_class = {
.class_name = "ALSA outdev",
.item_name = av_default_item_name,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
};
const FFOutputFormat ff_alsa_muxer = {
.p.name = "alsa",
.p.long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
.priv_data_size = sizeof(AlsaData),
.p.audio_codec = DEFAULT_CODEC_ID,
.p.video_codec = AV_CODEC_ID_NONE,
.write_header = audio_write_header,
.write_packet = audio_write_packet,
.write_trailer = ff_alsa_close,
.write_uncoded_frame = audio_write_frame,
.get_device_list = audio_get_device_list,
.get_output_timestamp = audio_get_output_timestamp,
.p.flags = AVFMT_NOFILE,
.p.priv_class = &alsa_muxer_class,
};