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- /*
- * Pulseaudio input
- * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
- *
- * 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
- */
- /**
- * @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/time.h"
- #include "libavutil/opt.h"
- #include "pulse_audio_common.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_simple *s;
- int64_t pts;
- int64_t frame_duration;
- int wallclock;
- } PulseData;
- static av_cold int pulse_read_header(AVFormatContext *s)
- {
- PulseData *pd = s->priv_data;
- AVStream *st;
- char *device = NULL;
- int ret, sample_bytes;
- 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 (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, pd->sample_rate); /* 64 bits pts in us */
- pd->pts = AV_NOPTS_VALUE;
- sample_bytes = (av_get_bits_per_sample(codec_id) >> 3) * pd->channels;
- if (pd->frame_size % sample_bytes) {
- av_log(s, AV_LOG_WARNING, "frame_size %i is not divisible by %i "
- "(channels * bytes_per_sample) \n", pd->frame_size, sample_bytes);
- }
- pd->frame_duration = pd->frame_size / sample_bytes;
- return 0;
- }
- static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
- {
- PulseData *pd = s->priv_data;
- int res;
- 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 (pd->pts == AV_NOPTS_VALUE) {
- pa_usec_t latency;
- 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);
- }
- pd->pts = -latency;
- if (pd->wallclock)
- pd->pts += av_gettime();
- }
- 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", "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,
- };
- 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,
- };
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