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pulse: introduce pulseaudio input
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
192
libavdevice/pulse.c
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192
libavdevice/pulse.c
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/*
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* Pulseaudio input
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* Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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/**
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* @file
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* Pulseaudio input
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* @author Luca Barbato <lu_zero@gentoo.org>
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*
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* This avdevice decoder allows to capture audio from a Pulseaudio device using
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* the simple api.
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*
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*/
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#include <pulse/simple.h>
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#include <pulse/rtclock.h>
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#include <pulse/error.h>
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#include "libavformat/avformat.h"
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#include "libavutil/opt.h"
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#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
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typedef struct PulseData {
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AVClass *class;
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char *server;
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char *name;
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char *dev;
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char *stream_name;
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int sample_rate;
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int channels;
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int frame_size;
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pa_simple *s;
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int64_t pts;
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} PulseData;
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static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
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switch(codec_id) {
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case CODEC_ID_PCM_U8: return PA_SAMPLE_U8;
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case CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW;
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case CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
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case CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
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case CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
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case CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
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case CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
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case CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
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case CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
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case CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
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case CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
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default: return PA_SAMPLE_INVALID;
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}
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}
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static av_cold int pulse_read_header(AVFormatContext *s,
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AVFormatParameters *ap)
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{
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PulseData *pd = s->priv_data;
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AVStream *st;
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int ret;
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enum CodecID codec_id =
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s->audio_codec_id == CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
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const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
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pd->sample_rate,
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pd->channels };
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pa_buffer_attr attr = { -1 };
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st = avformat_new_stream(s, NULL);
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if (!st) {
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av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
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return AVERROR(ENOMEM);
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}
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attr.fragsize = pd->frame_size * 4;
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pd->s = pa_simple_new(pd->server, pd->name,
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PA_STREAM_RECORD,
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pd->dev, pd->stream_name, &ss,
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NULL, &attr, &ret);
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if (!pd->s) {
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av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
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pa_strerror(ret));
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return AVERROR(EIO);
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}
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/* take real parameters */
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = codec_id;
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st->codec->sample_rate = pd->sample_rate;
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st->codec->channels = pd->channels;
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av_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 int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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PulseData *pd = s->priv_data;
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int res;
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pa_usec_t latency, cur;
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uint64_t frame_duration =
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(pd->frame_size*1000000LL)/(pd->sample_rate * pd->channels);
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if (av_new_packet(pkt, pd->frame_size) < 0) {
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return AVERROR(ENOMEM);
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}
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cur = pa_rtclock_now();
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if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
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av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
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pa_strerror(res));
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av_free_packet(pkt);
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return AVERROR(EIO);
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}
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if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
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av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
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pa_strerror(res));
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return AVERROR(EIO);
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}
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if (!pd->pts) {
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pd->pts -= latency;
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}
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pd->pts += frame_duration;
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av_log(s, AV_LOG_DEBUG, "%"PRId64" time %"PRId64","
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" latency %"PRId64", %"PRId64"\n",
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av_gettime(), cur, latency, pd->pts);
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pkt->pts = pd->pts;
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return 0;
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}
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static av_cold int pulse_close(AVFormatContext *s)
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{
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PulseData *pd = s->priv_data;
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pa_simple_free(pd->s);
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return 0;
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}
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#define OFFSET(a) offsetof(PulseData, a)
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#define D AV_OPT_FLAG_DECODING_PARAM
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static const AVOption options[] = {
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{ "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
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{ "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = "libav"}, 0, 0, D },
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{ "dev", "device to use", OFFSET(dev), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
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{ "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
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{ "sample_rate", "", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D },
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{ "channels", "", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
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{ "frame_size", "", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D },
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{ NULL },
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};
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static const AVClass pulse_demuxer_class = {
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.class_name = "Pulse demuxer",
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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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};
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AVInputFormat ff_pulse_demuxer = {
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.name = "pulse",
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.long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
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.priv_data_size = sizeof(PulseData),
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.read_header = pulse_read_header,
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.read_packet = pulse_read_packet,
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.read_close = pulse_close,
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.flags = AVFMT_NOFILE,
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.priv_class = &pulse_demuxer_class,
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};
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