mirror of
https://github.com/FFmpeg/FFmpeg.git
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f607767d93
Signed-off-by: Lukasz Marek <lukasz.m.luki@gmail.com>
186 lines
6.5 KiB
C
186 lines
6.5 KiB
C
/*
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* Copyright (c) 2013 Lukasz Marek <lukasz.m.luki@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <pulse/simple.h>
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#include <pulse/error.h>
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "libavutil/log.h"
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#include "pulse_audio_common.h"
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typedef struct PulseData {
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AVClass *class;
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const char *server;
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const char *name;
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const char *stream_name;
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const char *device;
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pa_simple *pa;
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int64_t timestamp;
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int buffer_size;
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int buffer_duration;
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} PulseData;
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static av_cold int pulse_write_header(AVFormatContext *h)
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{
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PulseData *s = h->priv_data;
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AVStream *st = NULL;
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int ret;
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pa_sample_spec ss;
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pa_buffer_attr attr = { -1, -1, -1, -1, -1 };
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const char *stream_name = s->stream_name;
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if (h->nb_streams != 1 || h->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) {
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av_log(s, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
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return AVERROR(EINVAL);
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}
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st = h->streams[0];
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if (!stream_name) {
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if (h->filename[0])
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stream_name = h->filename;
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else
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stream_name = "Playback";
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}
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if (s->buffer_duration) {
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int64_t bytes = s->buffer_duration;
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bytes *= st->codec->channels * st->codec->sample_rate *
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av_get_bytes_per_sample(st->codec->sample_fmt);
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bytes /= 1000;
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attr.tlength = FFMAX(s->buffer_size, av_clip64(bytes, 0, UINT32_MAX - 1));
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av_log(s, AV_LOG_DEBUG,
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"Buffer duration: %ums recalculated into %"PRId64" bytes buffer.\n",
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s->buffer_duration, bytes);
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av_log(s, AV_LOG_DEBUG, "Real buffer length is %u bytes\n", attr.tlength);
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} else if (s->buffer_size)
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attr.tlength = s->buffer_size;
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ss.format = ff_codec_id_to_pulse_format(st->codec->codec_id);
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ss.rate = st->codec->sample_rate;
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ss.channels = st->codec->channels;
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s->pa = pa_simple_new(s->server, // Server
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s->name, // Application name
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PA_STREAM_PLAYBACK,
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s->device, // Device
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stream_name, // Description of a stream
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&ss, // Sample format
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NULL, // Use default channel map
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&attr, // Buffering attributes
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&ret); // Result
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if (!s->pa) {
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av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n", pa_strerror(ret));
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return AVERROR(EIO);
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}
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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return 0;
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}
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static av_cold int pulse_write_trailer(AVFormatContext *h)
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{
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PulseData *s = h->priv_data;
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pa_simple_flush(s->pa, NULL);
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pa_simple_free(s->pa);
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s->pa = NULL;
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return 0;
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}
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static int pulse_write_packet(AVFormatContext *h, AVPacket *pkt)
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{
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PulseData *s = h->priv_data;
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int error;
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if (!pkt) {
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if (pa_simple_flush(s->pa, &error) < 0) {
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av_log(s, AV_LOG_ERROR, "pa_simple_flush failed: %s\n", pa_strerror(error));
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return AVERROR(EIO);
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}
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return 1;
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}
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if (pkt->dts != AV_NOPTS_VALUE)
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s->timestamp = pkt->dts;
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if (pkt->duration) {
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s->timestamp += pkt->duration;
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} else {
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AVStream *st = h->streams[0];
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AVCodecContext *codec_ctx = st->codec;
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AVRational r = { 1, codec_ctx->sample_rate };
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int64_t samples = pkt->size / (av_get_bytes_per_sample(codec_ctx->sample_fmt) * codec_ctx->channels);
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s->timestamp += av_rescale_q(samples, r, st->time_base);
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}
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if (pa_simple_write(s->pa, pkt->data, pkt->size, &error) < 0) {
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av_log(s, AV_LOG_ERROR, "pa_simple_write failed: %s\n", pa_strerror(error));
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return AVERROR(EIO);
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}
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return 0;
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}
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static void pulse_get_output_timestamp(AVFormatContext *h, int stream, int64_t *dts, int64_t *wall)
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{
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PulseData *s = h->priv_data;
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pa_usec_t latency = pa_simple_get_latency(s->pa, NULL);
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*wall = av_gettime();
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*dts = s->timestamp - latency;
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}
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#define OFFSET(a) offsetof(PulseData, a)
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#define E AV_OPT_FLAG_ENCODING_PARAM
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static const AVOption options[] = {
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{ "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
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{ "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, E },
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{ "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
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{ "device", "set device name", OFFSET(device), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, E },
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{ "buffer_size", "set buffer size in bytes", OFFSET(buffer_size), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, E },
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{ "buffer_duration", "set buffer duration in millisecs", OFFSET(buffer_duration), AV_OPT_TYPE_INT, {.i64 = 0}, 0, INT_MAX, E },
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{ NULL }
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};
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static const AVClass pulse_muxer_class = {
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.class_name = "Pulse muxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
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};
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AVOutputFormat ff_pulse_muxer = {
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.name = "pulse",
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.long_name = NULL_IF_CONFIG_SMALL("Pulse audio output"),
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.priv_data_size = sizeof(PulseData),
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.audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
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.video_codec = AV_CODEC_ID_NONE,
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.write_header = pulse_write_header,
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.write_packet = pulse_write_packet,
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.write_trailer = pulse_write_trailer,
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.get_output_timestamp = pulse_get_output_timestamp,
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.flags = AVFMT_NOFILE | AVFMT_ALLOW_FLUSH,
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.priv_class = &pulse_muxer_class,
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};
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