3 * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
4 * Copyright 2004-2006 Lennart Poettering
5 * Copyright (c) 2014 Michael Niedermayer <michaelni@gmx.at>
7 * This file is part of FFmpeg.
9 * FFmpeg is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
14 * FFmpeg is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Lesser General Public License for more details.
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with FFmpeg; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 #include <pulse/rtclock.h>
25 #include <pulse/error.h>
27 #include "libavutil/internal.h"
28 #include "libavutil/opt.h"
29 #include "libavutil/time.h"
31 #include "libavformat/avformat.h"
32 #include "libavformat/internal.h"
33 #include "pulse_audio_common.h"
34 #include "timefilter.h"
36 #define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
38 typedef struct PulseData {
48 pa_threaded_mainloop *mainloop;
52 TimeFilter *timefilter;
58 #define CHECK_SUCCESS_GOTO(rerror, expression, label) \
60 if (!(expression)) { \
61 rerror = AVERROR_EXTERNAL; \
66 #define CHECK_DEAD_GOTO(p, rerror, label) \
68 if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \
69 !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \
70 rerror = AVERROR_EXTERNAL; \
75 static void context_state_cb(pa_context *c, void *userdata) {
76 PulseData *p = userdata;
78 switch (pa_context_get_state(c)) {
79 case PA_CONTEXT_READY:
80 case PA_CONTEXT_TERMINATED:
81 case PA_CONTEXT_FAILED:
82 pa_threaded_mainloop_signal(p->mainloop, 0);
87 static void stream_state_cb(pa_stream *s, void * userdata) {
88 PulseData *p = userdata;
90 switch (pa_stream_get_state(s)) {
92 case PA_STREAM_FAILED:
93 case PA_STREAM_TERMINATED:
94 pa_threaded_mainloop_signal(p->mainloop, 0);
99 static void stream_request_cb(pa_stream *s, size_t length, void *userdata) {
100 PulseData *p = userdata;
102 pa_threaded_mainloop_signal(p->mainloop, 0);
105 static void stream_latency_update_cb(pa_stream *s, void *userdata) {
106 PulseData *p = userdata;
108 pa_threaded_mainloop_signal(p->mainloop, 0);
111 static av_cold int pulse_close(AVFormatContext *s)
113 PulseData *pd = s->priv_data;
116 pa_threaded_mainloop_stop(pd->mainloop);
119 pa_stream_unref(pd->stream);
123 pa_context_disconnect(pd->context);
124 pa_context_unref(pd->context);
129 pa_threaded_mainloop_free(pd->mainloop);
132 ff_timefilter_destroy(pd->timefilter);
133 pd->timefilter = NULL;
138 static av_cold int pulse_read_header(AVFormatContext *s)
140 PulseData *pd = s->priv_data;
144 enum AVCodecID codec_id =
145 s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
146 const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id),
150 pa_buffer_attr attr = { -1 };
153 pa_channel_map_init_extend(&cmap, pd->channels, PA_CHANNEL_MAP_WAVEEX);
155 st = avformat_new_stream(s, NULL);
158 av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
159 return AVERROR(ENOMEM);
162 attr.fragsize = pd->fragment_size;
164 if (s->url[0] != '\0' && strcmp(s->url, "default"))
167 if (!(pd->mainloop = pa_threaded_mainloop_new())) {
169 return AVERROR_EXTERNAL;
172 if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) {
174 return AVERROR_EXTERNAL;
177 pa_context_set_state_callback(pd->context, context_state_cb, pd);
179 if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) {
181 return AVERROR(pa_context_errno(pd->context));
184 pa_threaded_mainloop_lock(pd->mainloop);
186 if (pa_threaded_mainloop_start(pd->mainloop) < 0) {
188 goto unlock_and_fail;
192 pa_context_state_t state;
194 state = pa_context_get_state(pd->context);
196 if (state == PA_CONTEXT_READY)
199 if (!PA_CONTEXT_IS_GOOD(state)) {
200 ret = AVERROR(pa_context_errno(pd->context));
201 goto unlock_and_fail;
204 /* Wait until the context is ready */
205 pa_threaded_mainloop_wait(pd->mainloop);
208 if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, &cmap))) {
209 ret = AVERROR(pa_context_errno(pd->context));
210 goto unlock_and_fail;
213 pa_stream_set_state_callback(pd->stream, stream_state_cb, pd);
214 pa_stream_set_read_callback(pd->stream, stream_request_cb, pd);
215 pa_stream_set_write_callback(pd->stream, stream_request_cb, pd);
216 pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd);
218 ret = pa_stream_connect_record(pd->stream, device, &attr,
219 PA_STREAM_INTERPOLATE_TIMING
220 |PA_STREAM_ADJUST_LATENCY
221 |PA_STREAM_AUTO_TIMING_UPDATE);
224 ret = AVERROR(pa_context_errno(pd->context));
225 goto unlock_and_fail;
229 pa_stream_state_t state;
231 state = pa_stream_get_state(pd->stream);
233 if (state == PA_STREAM_READY)
236 if (!PA_STREAM_IS_GOOD(state)) {
237 ret = AVERROR(pa_context_errno(pd->context));
238 goto unlock_and_fail;
241 /* Wait until the stream is ready */
242 pa_threaded_mainloop_wait(pd->mainloop);
245 pa_threaded_mainloop_unlock(pd->mainloop);
247 /* take real parameters */
248 st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
249 st->codecpar->codec_id = codec_id;
250 st->codecpar->sample_rate = pd->sample_rate;
251 st->codecpar->channels = pd->channels;
252 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
254 pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
257 if (!pd->timefilter) {
259 return AVERROR(ENOMEM);
265 pa_threaded_mainloop_unlock(pd->mainloop);
271 static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
273 PulseData *pd = s->priv_data;
276 const void *read_data = NULL;
281 pa_threaded_mainloop_lock(pd->mainloop);
283 CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
288 r = pa_stream_peek(pd->stream, &read_data, &read_length);
289 CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
291 if (read_length <= 0) {
292 pa_threaded_mainloop_wait(pd->mainloop);
293 CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
294 } else if (!read_data) {
295 /* There's a hole in the stream, skip it. We could generate
296 * silence, but that wouldn't work for compressed streams. */
297 r = pa_stream_drop(pd->stream);
298 CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
302 if (av_new_packet(pkt, read_length) < 0) {
303 ret = AVERROR(ENOMEM);
304 goto unlock_and_fail;
308 pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
310 if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
311 enum AVCodecID codec_id =
312 s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
313 int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
314 int frame_duration = read_length / frame_size;
322 pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
324 pd->last_period = frame_duration;
326 av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
329 memcpy(pkt->data, read_data, read_length);
330 pa_stream_drop(pd->stream);
332 pa_threaded_mainloop_unlock(pd->mainloop);
336 pa_threaded_mainloop_unlock(pd->mainloop);
340 static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
342 PulseData *s = h->priv_data;
343 return ff_pulse_audio_get_devices(device_list, s->server, 0);
346 #define OFFSET(a) offsetof(PulseData, a)
347 #define D AV_OPT_FLAG_DECODING_PARAM
349 static const AVOption options[] = {
350 { "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
351 { "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
352 { "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
353 { "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
354 { "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
355 { "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
356 { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
357 { "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, D },
361 static const AVClass pulse_demuxer_class = {
362 .class_name = "Pulse demuxer",
363 .item_name = av_default_item_name,
365 .version = LIBAVUTIL_VERSION_INT,
366 .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
369 AVInputFormat ff_pulse_demuxer = {
371 .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
372 .priv_data_size = sizeof(PulseData),
373 .read_header = pulse_read_header,
374 .read_packet = pulse_read_packet,
375 .read_close = pulse_close,
376 .get_device_list = pulse_get_device_list,
377 .flags = AVFMT_NOFILE,
378 .priv_class = &pulse_demuxer_class,