/*
* Copyright (c) 2013 Lukasz Marek <lukasz.m.luki@gmail.com>
*
* 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/simple.h>
#include <pulse/error.h>
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "libavutil/opt.h"
#include "libavutil/time.h"
#include "libavutil/log.h"
#include "pulse_audio_common.h"
typedef struct PulseData {
AVClass *class;
const char *server;
const char *name;
const char *stream_name;
const char *device;
pa_simple *pa;
int64_t timestamp;
int buffer_size;
int buffer_duration;
} PulseData;
static av_cold int pulse_write_header(AVFormatContext *h)
{
PulseData *s = h->priv_data;
AVStream *st = NULL;
int ret;
pa_sample_spec ss;
pa_buffer_attr attr = { -1, -1, -1, -1, -1 };
const char *stream_name = s->stream_name;
if (h->nb_streams != 1 || h->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
av_log(s, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
return AVERROR(EINVAL);
}
st = h->streams[0];
if (!stream_name) {
if (h->filename[0])
stream_name = h->filename;
else
stream_name = "Playback";
}
if (s->buffer_duration) {
int64_t bytes = s->buffer_duration;
bytes *= st->codec->channels * st->codec->sample_rate *
av_get_bytes_per_sample(st->codec->sample_fmt);
bytes /= 1000;
attr.tlength = FFMAX(s->buffer_size, av_clip64(bytes, 0, UINT32_MAX - 1));
av_log(s, AV_LOG_DEBUG,
"Buffer duration: %ums recalculated into %"PRId64" bytes buffer.\n",
s->buffer_duration, bytes);
av_log(s, AV_LOG_DEBUG, "Real buffer length is %u bytes\n", attr.tlength);
} else if (s->buffer_size)
attr.tlength = s->buffer_size;
ss.format = ff_codec_id_to_pulse_format(st->codec->codec_id);
ss.rate = st->codec->sample_rate;
ss.channels = st->codec->channels;
s->pa = pa_simple_new(s->server, // Server
s->name, // Application name
PA_STREAM_PLAYBACK,
s->device, // Device
stream_name, // Description of a stream
&ss, // Sample format
NULL, // Use default channel map
&attr, // Buffering attributes
&ret); // Result
if (!s->pa) {
av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n", pa_strerror(ret));
return AVERROR(EIO);
}
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
return 0;
}
static av_cold int pulse_write_trailer(AVFormatContext *h)
{
PulseData *s = h->priv_data;
pa_simple_flush(s->pa, NULL);
pa_simple_free(s->pa);
s->pa = NULL;
return 0;
}
static int pulse_write_packet(AVFormatContext *h, AVPacket *pkt)
{
PulseData *s = h->priv_data;
int error;
if (!pkt) {
if (pa_simple_flush(s->pa, &error) < 0) {
av_log(s, AV_LOG_ERROR, "pa_simple_flush failed: %s\n", pa_strerror(error));
return AVERROR(EIO);
}
return 1;
}
if (pkt->dts != AV_NOPTS_VALUE)
s->timestamp = pkt->dts;
if (pkt->duration) {
s->timestamp += pkt->duration;
} else {
AVStream *st = h->streams[0];
AVCodecContext *codec_ctx = st->codec;
AVRational r = { 1, codec_ctx->sample_rate };
int64_t samples = pkt->size / (av_get_bytes_per_sample(codec_ctx->sample_fmt) * codec_ctx->channels);
s->timestamp += av_rescale_q(samples, r, st->time_base);
}
if (pa_simple_write(s->pa, pkt->data, pkt->size, &error) < 0) {
av_log(s, AV_LOG_ERROR, "pa_simple_write failed: %s\n", pa_strerror(error));
return AVERROR(EIO);
}
return 0;
}
static void pulse_get_output_timestamp(AVFormatContext *h, int stream, int64_t *dts, int64_t *wall)
{
PulseData *s = h->priv_data;
pa_usec_t latency = pa_simple_get_latency(s->pa, NULL);
*wall = av_gettime();
*dts = s->timestamp - latency;
}
#define OFFSET(a) offsetof(PulseData, a)
#define E AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
{ "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
{ "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, E },
{ "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
{ "device", "set device name", OFFSET(device), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
{ "buffer_size", "set buffer size in bytes", OFFSET(buffer_size), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, E },
{ "buffer_duration", "set buffer duration in millisecs", OFFSET(buffer_duration), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, E },
{ NULL }
};
static const AVClass pulse_muxer_class = {
.class_name = "Pulse muxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVOutputFormat ff_pulse_muxer = {
.name = "pulse",
.long_name = NULL_IF_CONFIG_SMALL("Pulse audio output"),
.priv_data_size = sizeof(PulseData),
.audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
.video_codec = AV_CODEC_ID_NONE,
.write_header = pulse_write_header,
.write_packet = pulse_write_packet,
.write_trailer = pulse_write_trailer,
.get_output_timestamp = pulse_get_output_timestamp,
.flags = AVFMT_NOFILE | AVFMT_ALLOW_FLUSH,
.priv_class = &pulse_muxer_class,
};