mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-02 03:06:28 +02:00
b404ab9e74
* qatar/master: mov: Don't av_malloc(0). avconv: only allocate 1 AVFrame per input stream avconv: fix memleaks due to not freeing the AVFrame for audio h264-fate: remove -strict 1 except where necessary (mr4/5-tandberg). misc Doxygen markup improvements doxygen: eliminate Qt-style doxygen syntax g722: Add a regression test for muxing/demuxing in wav g722: Change bits per sample to 4 g722dec: Signal skipping the lower bits via AVOptions instead of bits_per_coded_sample api-example: update to use avcodec_decode_audio4() avplay: use avcodec_decode_audio4() avplay: use a separate buffer for playing silence avformat: use avcodec_decode_audio4() in avformat_find_stream_info() avconv: use avcodec_decode_audio4() instead of avcodec_decode_audio3() mov: Allow empty stts atom. doc: document preferred Doxygen syntax and make patcheck detect it Conflicts: avconv.c ffplay.c libavcodec/mlpdec.c libavcodec/version.h libavformat/mov.c tests/codec-regression.sh tests/fate/h264.mak Merged-by: Michael Niedermayer <michaelni@gmx.at>
192 lines
6.4 KiB
C
192 lines
6.4 KiB
C
/*
|
|
* Pulseaudio input
|
|
* Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
|
|
*
|
|
* 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
|
|
* PulseAudio input using the simple API.
|
|
* @author Luca Barbato <lu_zero@gentoo.org>
|
|
*/
|
|
|
|
#include <pulse/simple.h>
|
|
#include <pulse/rtclock.h>
|
|
#include <pulse/error.h>
|
|
|
|
#include "libavformat/avformat.h"
|
|
#include "libavformat/internal.h"
|
|
#include "libavutil/opt.h"
|
|
|
|
#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
|
|
|
|
typedef struct PulseData {
|
|
AVClass *class;
|
|
char *server;
|
|
char *name;
|
|
char *stream_name;
|
|
int sample_rate;
|
|
int channels;
|
|
int frame_size;
|
|
int fragment_size;
|
|
pa_simple *s;
|
|
int64_t pts;
|
|
int64_t frame_duration;
|
|
} PulseData;
|
|
|
|
static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
|
|
switch (codec_id) {
|
|
case CODEC_ID_PCM_U8: return PA_SAMPLE_U8;
|
|
case CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW;
|
|
case CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
|
|
case CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
|
|
case CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
|
|
case CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
|
|
case CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
|
|
case CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
|
|
case CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
|
|
case CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
|
|
case CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
|
|
default: return PA_SAMPLE_INVALID;
|
|
}
|
|
}
|
|
|
|
static av_cold int pulse_read_header(AVFormatContext *s,
|
|
AVFormatParameters *ap)
|
|
{
|
|
PulseData *pd = s->priv_data;
|
|
AVStream *st;
|
|
char *device = NULL;
|
|
int ret;
|
|
enum CodecID codec_id =
|
|
s->audio_codec_id == CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
|
|
const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
|
|
pd->sample_rate,
|
|
pd->channels };
|
|
|
|
pa_buffer_attr attr = { -1 };
|
|
|
|
st = avformat_new_stream(s, NULL);
|
|
|
|
if (!st) {
|
|
av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
attr.fragsize = pd->fragment_size;
|
|
|
|
if (strcmp(s->filename, "default"))
|
|
device = s->filename;
|
|
|
|
pd->s = pa_simple_new(pd->server, pd->name,
|
|
PA_STREAM_RECORD,
|
|
device, pd->stream_name, &ss,
|
|
NULL, &attr, &ret);
|
|
|
|
if (!pd->s) {
|
|
av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
|
|
pa_strerror(ret));
|
|
return AVERROR(EIO);
|
|
}
|
|
/* take real parameters */
|
|
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
|
|
st->codec->codec_id = codec_id;
|
|
st->codec->sample_rate = pd->sample_rate;
|
|
st->codec->channels = pd->channels;
|
|
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
|
|
|
|
pd->pts = AV_NOPTS_VALUE;
|
|
pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
|
|
(pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
PulseData *pd = s->priv_data;
|
|
int res;
|
|
pa_usec_t latency;
|
|
|
|
if (av_new_packet(pkt, pd->frame_size) < 0) {
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
|
|
av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
|
|
pa_strerror(res));
|
|
av_free_packet(pkt);
|
|
return AVERROR(EIO);
|
|
}
|
|
|
|
if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
|
|
av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
|
|
pa_strerror(res));
|
|
return AVERROR(EIO);
|
|
}
|
|
|
|
if (pd->pts == AV_NOPTS_VALUE) {
|
|
pd->pts = -latency;
|
|
}
|
|
|
|
pkt->pts = pd->pts;
|
|
|
|
pd->pts += pd->frame_duration;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int pulse_close(AVFormatContext *s)
|
|
{
|
|
PulseData *pd = s->priv_data;
|
|
pa_simple_free(pd->s);
|
|
return 0;
|
|
}
|
|
|
|
#define OFFSET(a) offsetof(PulseData, a)
|
|
#define D AV_OPT_FLAG_DECODING_PARAM
|
|
|
|
static const AVOption options[] = {
|
|
{ "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
|
|
{ "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
|
|
{ "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
|
|
{ "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D },
|
|
{ "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
|
|
{ "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D },
|
|
{ "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.dbl = -1}, -1, INT_MAX, D },
|
|
{ NULL },
|
|
};
|
|
|
|
static const AVClass pulse_demuxer_class = {
|
|
.class_name = "Pulse demuxer",
|
|
.item_name = av_default_item_name,
|
|
.option = options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVInputFormat ff_pulse_demuxer = {
|
|
.name = "pulse",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
|
|
.priv_data_size = sizeof(PulseData),
|
|
.read_header = pulse_read_header,
|
|
.read_packet = pulse_read_packet,
|
|
.read_close = pulse_close,
|
|
.flags = AVFMT_NOFILE,
|
|
.priv_class = &pulse_demuxer_class,
|
|
};
|