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pulse: introduce pulseaudio input

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It currently use the simple api and is using the latency information
provided only to offset the stream start.
This commit is contained in:
Luca Barbato 2011-10-19 11:23:20 -07:00
parent f0eeff708a
commit f5b0938169
4 changed files with 198 additions and 0 deletions
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4
configure vendored
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@ -940,6 +940,7 @@ CONFIG_LIST="
libopencore_amrwb
libopencv
libopenjpeg
libpulse
librtmp
libschroedinger
libspeex
@ -1458,6 +1459,7 @@ libcdio_indev_deps="libcdio"
libdc1394_indev_deps="libdc1394"
oss_indev_deps_any="soundcard_h sys_soundcard_h"
oss_outdev_deps_any="soundcard_h sys_soundcard_h"
pulse_indev_deps="libpulse"
sndio_indev_deps="sndio_h"
sndio_outdev_deps="sndio_h"
v4l_indev_deps="linux_videodev_h"
@ -2894,6 +2896,7 @@ enabled libopencore_amrnb && require libopencore_amrnb opencore-amrnb/interf_de
enabled libopencore_amrwb && require libopencore_amrwb opencore-amrwb/dec_if.h D_IF_init -lopencore-amrwb
enabled libopencv && require_pkg_config opencv opencv/cv.h cvCreateImageHeader
enabled libopenjpeg && require libopenjpeg openjpeg.h opj_version -lopenjpeg
enabled libpulse && require_pkg_config libpulse-simple pulse/simple.h pa_simple_new
enabled librtmp && require_pkg_config librtmp librtmp/rtmp.h RTMP_Socket
enabled libschroedinger && require_pkg_config schroedinger-1.0 schroedinger/schro.h schro_init
enabled libspeex && require libspeex speex/speex.h speex_decoder_init -lspeex
@ -3179,6 +3182,7 @@ echo "libopencore-amrnb support ${libopencore_amrnb-no}"
echo "libopencore-amrwb support ${libopencore_amrwb-no}"
echo "libopencv support ${libopencv-no}"
echo "libopenjpeg enabled ${libopenjpeg-no}"
echo "libpulse enabled ${libpulse-no}"
echo "librtmp enabled ${librtmp-no}"
echo "libschroedinger enabled ${libschroedinger-no}"
echo "libspeex enabled ${libspeex-no}"

1
libavdevice/Makefile
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@ -16,6 +16,7 @@ OBJS-$(CONFIG_FBDEV_INDEV) += fbdev.o
OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o
OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o
OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o
OBJS-$(CONFIG_PULSE_INDEV) += pulse.o
OBJS-$(CONFIG_SNDIO_INDEV) += sndio_common.o sndio_dec.o
OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_common.o sndio_enc.o
OBJS-$(CONFIG_V4L2_INDEV) += v4l2.o

1
libavdevice/alldevices.c
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@ -45,6 +45,7 @@ void avdevice_register_all(void)
REGISTER_INDEV (FBDEV, fbdev);
REGISTER_INDEV (JACK, jack);
REGISTER_INOUTDEV (OSS, oss);
REGISTER_INDEV (PULSE, pulse);
REGISTER_INOUTDEV (SNDIO, sndio);
REGISTER_INDEV (V4L2, v4l2);
#if FF_API_V4L

192
libavdevice/pulse.c Normal file
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@ -0,0 +1,192 @@
/*
* 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
* @author Luca Barbato <lu_zero@gentoo.org>
*
* This avdevice decoder allows to capture audio from a Pulseaudio device using
* the simple api.
*
*/
#include <pulse/simple.h>
#include <pulse/rtclock.h>
#include <pulse/error.h>
#include "libavformat/avformat.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 *dev;
char *stream_name;
int sample_rate;
int channels;
int frame_size;
pa_simple *s;
int64_t pts;
} 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;
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->frame_size * 4;
pd->s = pa_simple_new(pd->server, pd->name,
PA_STREAM_RECORD,
pd->dev, 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;
av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
{
PulseData *pd = s->priv_data;
int res;
pa_usec_t latency, cur;
uint64_t frame_duration =
(pd->frame_size*1000000LL)/(pd->sample_rate * pd->channels);
if (av_new_packet(pkt, pd->frame_size) < 0) {
return AVERROR(ENOMEM);
}
cur = pa_rtclock_now();
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) {
pd->pts -= latency;
}
pd->pts += frame_duration;
av_log(s, AV_LOG_DEBUG, "%"PRId64" time %"PRId64","
" latency %"PRId64", %"PRId64"\n",
av_gettime(), cur, latency, pd->pts);
pkt->pts = pd->pts;
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 = "libav"}, 0, 0, D },
{ "dev", "device to use", OFFSET(dev), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
{ "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
{ "sample_rate", "", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D },
{ "channels", "", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
{ "frame_size", "", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 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,
};