380 lines
12 KiB
C
380 lines
12 KiB
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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* Copyright 2004-2006 Lennart Poettering
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* Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at>
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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/rtclock.h>
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#include <pulse/error.h>
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#include "libavutil/internal.h"
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#include "libavutil/opt.h"
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#include "libavutil/time.h"
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#include "libavformat/avformat.h"
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#include "libavformat/internal.h"
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#include "pulse_audio_common.h"
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#include "timefilter.h"
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#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_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 *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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int fragment_size;
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pa_threaded_mainloop *mainloop;
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pa_context *context;
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pa_stream *stream;
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TimeFilter *timefilter;
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int last_period;
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int wallclock;
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} PulseData;
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#define CHECK_SUCCESS_GOTO(rerror, expression, label) \
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do { \
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if (!(expression)) { \
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rerror = AVERROR_EXTERNAL; \
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goto label; \
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} \
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} while (0)
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#define CHECK_DEAD_GOTO(p, rerror, label) \
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do { \
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if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \
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!(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \
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rerror = AVERROR_EXTERNAL; \
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goto label; \
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} \
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} while (0)
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static void context_state_cb(pa_context *c, void *userdata) {
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PulseData *p = userdata;
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switch (pa_context_get_state(c)) {
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case PA_CONTEXT_READY:
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case PA_CONTEXT_TERMINATED:
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case PA_CONTEXT_FAILED:
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pa_threaded_mainloop_signal(p->mainloop, 0);
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break;
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}
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}
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static void stream_state_cb(pa_stream *s, void * userdata) {
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PulseData *p = userdata;
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switch (pa_stream_get_state(s)) {
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case PA_STREAM_READY:
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case PA_STREAM_FAILED:
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case PA_STREAM_TERMINATED:
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pa_threaded_mainloop_signal(p->mainloop, 0);
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break;
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}
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}
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static void stream_request_cb(pa_stream *s, size_t length, void *userdata) {
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PulseData *p = userdata;
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pa_threaded_mainloop_signal(p->mainloop, 0);
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}
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static void stream_latency_update_cb(pa_stream *s, void *userdata) {
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PulseData *p = userdata;
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pa_threaded_mainloop_signal(p->mainloop, 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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if (pd->mainloop)
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pa_threaded_mainloop_stop(pd->mainloop);
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if (pd->stream)
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pa_stream_unref(pd->stream);
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pd->stream = NULL;
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if (pd->context) {
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pa_context_disconnect(pd->context);
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pa_context_unref(pd->context);
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}
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pd->context = NULL;
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if (pd->mainloop)
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pa_threaded_mainloop_free(pd->mainloop);
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pd->mainloop = NULL;
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ff_timefilter_destroy(pd->timefilter);
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pd->timefilter = NULL;
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return 0;
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}
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static av_cold int pulse_read_header(AVFormatContext *s)
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{
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PulseData *pd = s->priv_data;
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AVStream *st;
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char *device = NULL;
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int ret;
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enum AVCodecID codec_id =
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s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
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const pa_sample_spec ss = { ff_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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pa_channel_map cmap;
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pa_channel_map_init_extend(&cmap, pd->channels, PA_CHANNEL_MAP_WAVEEX);
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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->fragment_size;
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if (s->url[0] != '\0' && strcmp(s->url, "default"))
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device = s->url;
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if (!(pd->mainloop = pa_threaded_mainloop_new())) {
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pulse_close(s);
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return AVERROR_EXTERNAL;
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}
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if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) {
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pulse_close(s);
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return AVERROR_EXTERNAL;
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}
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pa_context_set_state_callback(pd->context, context_state_cb, pd);
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if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) {
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pulse_close(s);
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return AVERROR(pa_context_errno(pd->context));
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}
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pa_threaded_mainloop_lock(pd->mainloop);
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if (pa_threaded_mainloop_start(pd->mainloop) < 0) {
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ret = -1;
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goto unlock_and_fail;
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}
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for (;;) {
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pa_context_state_t state;
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state = pa_context_get_state(pd->context);
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if (state == PA_CONTEXT_READY)
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break;
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if (!PA_CONTEXT_IS_GOOD(state)) {
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ret = AVERROR(pa_context_errno(pd->context));
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goto unlock_and_fail;
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}
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/* Wait until the context is ready */
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pa_threaded_mainloop_wait(pd->mainloop);
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}
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if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, &cmap))) {
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ret = AVERROR(pa_context_errno(pd->context));
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goto unlock_and_fail;
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}
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pa_stream_set_state_callback(pd->stream, stream_state_cb, pd);
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pa_stream_set_read_callback(pd->stream, stream_request_cb, pd);
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pa_stream_set_write_callback(pd->stream, stream_request_cb, pd);
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pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd);
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ret = pa_stream_connect_record(pd->stream, device, &attr,
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PA_STREAM_INTERPOLATE_TIMING
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|PA_STREAM_ADJUST_LATENCY
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|PA_STREAM_AUTO_TIMING_UPDATE);
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if (ret < 0) {
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ret = AVERROR(pa_context_errno(pd->context));
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goto unlock_and_fail;
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}
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for (;;) {
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pa_stream_state_t state;
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state = pa_stream_get_state(pd->stream);
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if (state == PA_STREAM_READY)
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break;
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if (!PA_STREAM_IS_GOOD(state)) {
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ret = AVERROR(pa_context_errno(pd->context));
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goto unlock_and_fail;
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}
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/* Wait until the stream is ready */
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pa_threaded_mainloop_wait(pd->mainloop);
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}
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pa_threaded_mainloop_unlock(pd->mainloop);
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/* take real parameters */
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = codec_id;
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st->codecpar->sample_rate = pd->sample_rate;
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st->codecpar->channels = pd->channels;
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
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1000, 1.5E-6);
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if (!pd->timefilter) {
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pulse_close(s);
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return AVERROR(ENOMEM);
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}
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return 0;
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unlock_and_fail:
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pa_threaded_mainloop_unlock(pd->mainloop);
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pulse_close(s);
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return ret;
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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 ret;
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size_t read_length;
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const void *read_data = NULL;
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int64_t dts;
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pa_usec_t latency;
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int negative;
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pa_threaded_mainloop_lock(pd->mainloop);
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CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
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while (!read_data) {
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int r;
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r = pa_stream_peek(pd->stream, &read_data, &read_length);
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CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
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if (read_length <= 0) {
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pa_threaded_mainloop_wait(pd->mainloop);
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CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
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} else if (!read_data) {
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/* There's a hole in the stream, skip it. We could generate
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* silence, but that wouldn't work for compressed streams. */
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r = pa_stream_drop(pd->stream);
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CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
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}
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}
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if (av_new_packet(pkt, read_length) < 0) {
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ret = AVERROR(ENOMEM);
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goto unlock_and_fail;
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}
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dts = av_gettime();
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pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
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if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
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enum AVCodecID codec_id =
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s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
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int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
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int frame_duration = read_length / frame_size;
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if (negative) {
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dts += latency;
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} else
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dts -= latency;
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if (pd->wallclock)
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pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
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pd->last_period = frame_duration;
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} else {
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av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
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}
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memcpy(pkt->data, read_data, read_length);
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pa_stream_drop(pd->stream);
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pa_threaded_mainloop_unlock(pd->mainloop);
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return 0;
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unlock_and_fail:
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pa_threaded_mainloop_unlock(pd->mainloop);
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return ret;
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}
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static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
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{
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PulseData *s = h->priv_data;
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return ff_pulse_audio_get_devices(device_list, s->server, 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", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
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{ "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
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{ "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
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{ "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
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{ "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
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{ "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
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{ "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
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{ "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, 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 indev",
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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_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,
|
||
|
|
};
|