/* * Pulseaudio input * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org> * Copyright 2004-2006 Lennart Poettering * Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at> * * 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/rtclock.h> #include <pulse/error.h> #include "libavutil/internal.h" #include "libavutil/opt.h" #include "libavutil/time.h" #include "libavformat/avformat.h" #include "libavformat/internal.h" #include "pulse_audio_common.h" #include "timefilter.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_threaded_mainloop *mainloop; pa_context *context; pa_stream *stream; TimeFilter *timefilter; int last_period; int wallclock; } PulseData; #define CHECK_SUCCESS_GOTO(rerror, expression, label) \ do { \ if (!(expression)) { \ rerror = AVERROR_EXTERNAL; \ goto label; \ } \ } while (0) #define CHECK_DEAD_GOTO(p, rerror, label) \ do { \ if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \ !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \ rerror = AVERROR_EXTERNAL; \ goto label; \ } \ } while (0) static void context_state_cb(pa_context *c, void *userdata) { PulseData *p = userdata; switch (pa_context_get_state(c)) { case PA_CONTEXT_READY: case PA_CONTEXT_TERMINATED: case PA_CONTEXT_FAILED: pa_threaded_mainloop_signal(p->mainloop, 0); break; } } static void stream_state_cb(pa_stream *s, void * userdata) { PulseData *p = userdata; switch (pa_stream_get_state(s)) { case PA_STREAM_READY: case PA_STREAM_FAILED: case PA_STREAM_TERMINATED: pa_threaded_mainloop_signal(p->mainloop, 0); break; } } static void stream_request_cb(pa_stream *s, size_t length, void *userdata) { PulseData *p = userdata; pa_threaded_mainloop_signal(p->mainloop, 0); } static void stream_latency_update_cb(pa_stream *s, void *userdata) { PulseData *p = userdata; pa_threaded_mainloop_signal(p->mainloop, 0); } static av_cold int pulse_close(AVFormatContext *s) { PulseData *pd = s->priv_data; if (pd->mainloop) pa_threaded_mainloop_stop(pd->mainloop); if (pd->stream) pa_stream_unref(pd->stream); pd->stream = NULL; if (pd->context) { pa_context_disconnect(pd->context); pa_context_unref(pd->context); } pd->context = NULL; if (pd->mainloop) pa_threaded_mainloop_free(pd->mainloop); pd->mainloop = NULL; ff_timefilter_destroy(pd->timefilter); pd->timefilter = NULL; return 0; } static av_cold int pulse_read_header(AVFormatContext *s) { PulseData *pd = s->priv_data; AVStream *st; char *device = NULL; int ret; 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 }; pa_channel_map cmap; pa_channel_map_init_extend(&cmap, pd->channels, PA_CHANNEL_MAP_WAVEEX); 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 (s->url[0] != '\0' && strcmp(s->url, "default")) device = s->url; if (!(pd->mainloop = pa_threaded_mainloop_new())) { pulse_close(s); return AVERROR_EXTERNAL; } if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) { pulse_close(s); return AVERROR_EXTERNAL; } pa_context_set_state_callback(pd->context, context_state_cb, pd); if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) { pulse_close(s); return AVERROR(pa_context_errno(pd->context)); } pa_threaded_mainloop_lock(pd->mainloop); if (pa_threaded_mainloop_start(pd->mainloop) < 0) { ret = -1; goto unlock_and_fail; } for (;;) { pa_context_state_t state; state = pa_context_get_state(pd->context); if (state == PA_CONTEXT_READY) break; if (!PA_CONTEXT_IS_GOOD(state)) { ret = AVERROR(pa_context_errno(pd->context)); goto unlock_and_fail; } /* Wait until the context is ready */ pa_threaded_mainloop_wait(pd->mainloop); } if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, &cmap))) { ret = AVERROR(pa_context_errno(pd->context)); goto unlock_and_fail; } pa_stream_set_state_callback(pd->stream, stream_state_cb, pd); pa_stream_set_read_callback(pd->stream, stream_request_cb, pd); pa_stream_set_write_callback(pd->stream, stream_request_cb, pd); pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd); ret = pa_stream_connect_record(pd->stream, device, &attr, PA_STREAM_INTERPOLATE_TIMING |PA_STREAM_ADJUST_LATENCY |PA_STREAM_AUTO_TIMING_UPDATE); if (ret < 0) { ret = AVERROR(pa_context_errno(pd->context)); goto unlock_and_fail; } for (;;) { pa_stream_state_t state; state = pa_stream_get_state(pd->stream); if (state == PA_STREAM_READY) break; if (!PA_STREAM_IS_GOOD(state)) { ret = AVERROR(pa_context_errno(pd->context)); goto unlock_and_fail; } /* Wait until the stream is ready */ pa_threaded_mainloop_wait(pd->mainloop); } pa_threaded_mainloop_unlock(pd->mainloop); /* take real parameters */ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->codec_id = codec_id; st->codecpar->sample_rate = pd->sample_rate; st->codecpar->channels = pd->channels; avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate, 1000, 1.5E-6); if (!pd->timefilter) { pulse_close(s); return AVERROR(ENOMEM); } return 0; unlock_and_fail: pa_threaded_mainloop_unlock(pd->mainloop); pulse_close(s); return ret; } static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt) { PulseData *pd = s->priv_data; int ret; size_t read_length; const void *read_data = NULL; int64_t dts; pa_usec_t latency; int negative; pa_threaded_mainloop_lock(pd->mainloop); CHECK_DEAD_GOTO(pd, ret, unlock_and_fail); while (!read_data) { int r; r = pa_stream_peek(pd->stream, &read_data, &read_length); CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail); if (read_length <= 0) { pa_threaded_mainloop_wait(pd->mainloop); CHECK_DEAD_GOTO(pd, ret, unlock_and_fail); } else if (!read_data) { /* There's a hole in the stream, skip it. We could generate * silence, but that wouldn't work for compressed streams. */ r = pa_stream_drop(pd->stream); CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail); } } if (av_new_packet(pkt, read_length) < 0) { ret = AVERROR(ENOMEM); goto unlock_and_fail; } dts = av_gettime(); pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL)); if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) { enum AVCodecID codec_id = s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels); int frame_duration = read_length / frame_size; if (negative) { dts += latency; } else dts -= latency; if (pd->wallclock) pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period); pd->last_period = frame_duration; } else { av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n"); } memcpy(pkt->data, read_data, read_length); pa_stream_drop(pd->stream); pa_threaded_mainloop_unlock(pd->mainloop); return 0; unlock_and_fail: pa_threaded_mainloop_unlock(pd->mainloop); return ret; } static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) { PulseData *s = h->priv_data; return ff_pulse_audio_get_devices(device_list, s->server, 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 indev", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, .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, };