mirror of
https://git.videolan.org/git/ffmpeg.git
synced 2024-08-20 00:05:06 +02:00
![Derek Buitenhuis](/assets/img/avatar_default.png)
* commit '9200514ad8717c63f82101dc394f4378854325bf': lavf: replace AVStream.codec with AVStream.codecpar This has been a HUGE effort from: - Derek Buitenhuis <derek.buitenhuis@gmail.com> - Hendrik Leppkes <h.leppkes@gmail.com> - wm4 <nfxjfg@googlemail.com> - Clément Bœsch <clement@stupeflix.com> - James Almer <jamrial@gmail.com> - Michael Niedermayer <michael@niedermayer.cc> - Rostislav Pehlivanov <atomnuker@gmail.com> Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
377 lines
12 KiB
C
377 lines
12 KiB
C
/*
|
|
* Pulseaudio input
|
|
* Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
|
|
* Copyright 2004-2006 Lennart Poettering
|
|
* Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at>
|
|
*
|
|
* 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
|
|
*/
|
|
|
|
#include <pulse/rtclock.h>
|
|
#include <pulse/error.h>
|
|
|
|
#include "libavutil/internal.h"
|
|
#include "libavutil/opt.h"
|
|
#include "libavutil/time.h"
|
|
|
|
#include "libavformat/avformat.h"
|
|
#include "libavformat/internal.h"
|
|
#include "pulse_audio_common.h"
|
|
#include "timefilter.h"
|
|
|
|
#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_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_threaded_mainloop *mainloop;
|
|
pa_context *context;
|
|
pa_stream *stream;
|
|
|
|
TimeFilter *timefilter;
|
|
int last_period;
|
|
int wallclock;
|
|
} PulseData;
|
|
|
|
|
|
#define CHECK_SUCCESS_GOTO(rerror, expression, label) \
|
|
do { \
|
|
if (!(expression)) { \
|
|
rerror = AVERROR_EXTERNAL; \
|
|
goto label; \
|
|
} \
|
|
} while (0)
|
|
|
|
#define CHECK_DEAD_GOTO(p, rerror, label) \
|
|
do { \
|
|
if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \
|
|
!(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \
|
|
rerror = AVERROR_EXTERNAL; \
|
|
goto label; \
|
|
} \
|
|
} while (0)
|
|
|
|
static void context_state_cb(pa_context *c, void *userdata) {
|
|
PulseData *p = userdata;
|
|
|
|
switch (pa_context_get_state(c)) {
|
|
case PA_CONTEXT_READY:
|
|
case PA_CONTEXT_TERMINATED:
|
|
case PA_CONTEXT_FAILED:
|
|
pa_threaded_mainloop_signal(p->mainloop, 0);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void stream_state_cb(pa_stream *s, void * userdata) {
|
|
PulseData *p = userdata;
|
|
|
|
switch (pa_stream_get_state(s)) {
|
|
case PA_STREAM_READY:
|
|
case PA_STREAM_FAILED:
|
|
case PA_STREAM_TERMINATED:
|
|
pa_threaded_mainloop_signal(p->mainloop, 0);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void stream_request_cb(pa_stream *s, size_t length, void *userdata) {
|
|
PulseData *p = userdata;
|
|
|
|
pa_threaded_mainloop_signal(p->mainloop, 0);
|
|
}
|
|
|
|
static void stream_latency_update_cb(pa_stream *s, void *userdata) {
|
|
PulseData *p = userdata;
|
|
|
|
pa_threaded_mainloop_signal(p->mainloop, 0);
|
|
}
|
|
|
|
static av_cold int pulse_close(AVFormatContext *s)
|
|
{
|
|
PulseData *pd = s->priv_data;
|
|
|
|
if (pd->mainloop)
|
|
pa_threaded_mainloop_stop(pd->mainloop);
|
|
|
|
if (pd->stream)
|
|
pa_stream_unref(pd->stream);
|
|
pd->stream = NULL;
|
|
|
|
if (pd->context) {
|
|
pa_context_disconnect(pd->context);
|
|
pa_context_unref(pd->context);
|
|
}
|
|
pd->context = NULL;
|
|
|
|
if (pd->mainloop)
|
|
pa_threaded_mainloop_free(pd->mainloop);
|
|
pd->mainloop = NULL;
|
|
|
|
ff_timefilter_destroy(pd->timefilter);
|
|
pd->timefilter = NULL;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int pulse_read_header(AVFormatContext *s)
|
|
{
|
|
PulseData *pd = s->priv_data;
|
|
AVStream *st;
|
|
char *device = NULL;
|
|
int ret;
|
|
enum AVCodecID codec_id =
|
|
s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
|
|
const pa_sample_spec ss = { ff_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 (s->filename[0] != '\0' && strcmp(s->filename, "default"))
|
|
device = s->filename;
|
|
|
|
if (!(pd->mainloop = pa_threaded_mainloop_new())) {
|
|
pulse_close(s);
|
|
return AVERROR_EXTERNAL;
|
|
}
|
|
|
|
if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) {
|
|
pulse_close(s);
|
|
return AVERROR_EXTERNAL;
|
|
}
|
|
|
|
pa_context_set_state_callback(pd->context, context_state_cb, pd);
|
|
|
|
if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) {
|
|
pulse_close(s);
|
|
return AVERROR(pa_context_errno(pd->context));
|
|
}
|
|
|
|
pa_threaded_mainloop_lock(pd->mainloop);
|
|
|
|
if (pa_threaded_mainloop_start(pd->mainloop) < 0) {
|
|
ret = -1;
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
for (;;) {
|
|
pa_context_state_t state;
|
|
|
|
state = pa_context_get_state(pd->context);
|
|
|
|
if (state == PA_CONTEXT_READY)
|
|
break;
|
|
|
|
if (!PA_CONTEXT_IS_GOOD(state)) {
|
|
ret = AVERROR(pa_context_errno(pd->context));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
/* Wait until the context is ready */
|
|
pa_threaded_mainloop_wait(pd->mainloop);
|
|
}
|
|
|
|
if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, NULL))) {
|
|
ret = AVERROR(pa_context_errno(pd->context));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
pa_stream_set_state_callback(pd->stream, stream_state_cb, pd);
|
|
pa_stream_set_read_callback(pd->stream, stream_request_cb, pd);
|
|
pa_stream_set_write_callback(pd->stream, stream_request_cb, pd);
|
|
pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd);
|
|
|
|
ret = pa_stream_connect_record(pd->stream, device, &attr,
|
|
PA_STREAM_INTERPOLATE_TIMING
|
|
|PA_STREAM_ADJUST_LATENCY
|
|
|PA_STREAM_AUTO_TIMING_UPDATE);
|
|
|
|
if (ret < 0) {
|
|
ret = AVERROR(pa_context_errno(pd->context));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
for (;;) {
|
|
pa_stream_state_t state;
|
|
|
|
state = pa_stream_get_state(pd->stream);
|
|
|
|
if (state == PA_STREAM_READY)
|
|
break;
|
|
|
|
if (!PA_STREAM_IS_GOOD(state)) {
|
|
ret = AVERROR(pa_context_errno(pd->context));
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
/* Wait until the stream is ready */
|
|
pa_threaded_mainloop_wait(pd->mainloop);
|
|
}
|
|
|
|
pa_threaded_mainloop_unlock(pd->mainloop);
|
|
|
|
/* take real parameters */
|
|
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
|
|
st->codecpar->codec_id = codec_id;
|
|
st->codecpar->sample_rate = pd->sample_rate;
|
|
st->codecpar->channels = pd->channels;
|
|
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
|
|
|
|
pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
|
|
1000, 1.5E-6);
|
|
|
|
if (!pd->timefilter) {
|
|
pulse_close(s);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
return 0;
|
|
|
|
unlock_and_fail:
|
|
pa_threaded_mainloop_unlock(pd->mainloop);
|
|
|
|
pulse_close(s);
|
|
return ret;
|
|
}
|
|
|
|
static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
|
|
{
|
|
PulseData *pd = s->priv_data;
|
|
int ret;
|
|
size_t read_length;
|
|
const void *read_data = NULL;
|
|
int64_t dts;
|
|
pa_usec_t latency;
|
|
int negative;
|
|
|
|
pa_threaded_mainloop_lock(pd->mainloop);
|
|
|
|
CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
|
|
|
|
while (!read_data) {
|
|
int r;
|
|
|
|
r = pa_stream_peek(pd->stream, &read_data, &read_length);
|
|
CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
|
|
|
|
if (read_length <= 0) {
|
|
pa_threaded_mainloop_wait(pd->mainloop);
|
|
CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
|
|
} else if (!read_data) {
|
|
/* There's a hole in the stream, skip it. We could generate
|
|
* silence, but that wouldn't work for compressed streams. */
|
|
r = pa_stream_drop(pd->stream);
|
|
CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
|
|
}
|
|
}
|
|
|
|
if (av_new_packet(pkt, read_length) < 0) {
|
|
ret = AVERROR(ENOMEM);
|
|
goto unlock_and_fail;
|
|
}
|
|
|
|
dts = av_gettime();
|
|
pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
|
|
|
|
if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
|
|
enum AVCodecID codec_id =
|
|
s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
|
|
int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
|
|
int frame_duration = read_length / frame_size;
|
|
|
|
|
|
if (negative) {
|
|
dts += latency;
|
|
} else
|
|
dts -= latency;
|
|
if (pd->wallclock)
|
|
pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
|
|
|
|
pd->last_period = frame_duration;
|
|
} else {
|
|
av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
|
|
}
|
|
|
|
memcpy(pkt->data, read_data, read_length);
|
|
pa_stream_drop(pd->stream);
|
|
|
|
pa_threaded_mainloop_unlock(pd->mainloop);
|
|
return 0;
|
|
|
|
unlock_and_fail:
|
|
pa_threaded_mainloop_unlock(pd->mainloop);
|
|
return ret;
|
|
}
|
|
|
|
static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
|
|
{
|
|
PulseData *s = h->priv_data;
|
|
return ff_pulse_audio_get_devices(device_list, s->server, 0);
|
|
}
|
|
|
|
#define OFFSET(a) offsetof(PulseData, a)
|
|
#define D AV_OPT_FLAG_DECODING_PARAM
|
|
|
|
static const AVOption options[] = {
|
|
{ "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
|
|
{ "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
|
|
{ "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
|
|
{ "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
|
|
{ "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
|
|
{ "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
|
|
{ "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
|
|
{ "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, D },
|
|
{ NULL },
|
|
};
|
|
|
|
static const AVClass pulse_demuxer_class = {
|
|
.class_name = "Pulse demuxer",
|
|
.item_name = av_default_item_name,
|
|
.option = options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
|
|
};
|
|
|
|
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,
|
|
.get_device_list = pulse_get_device_list,
|
|
.flags = AVFMT_NOFILE,
|
|
.priv_class = &pulse_demuxer_class,
|
|
};
|