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 };
152 const pa_buffer_attr *queried_attr;
154 pa_channel_map_init_extend(&cmap, pd->channels, PA_CHANNEL_MAP_WAVEEX);
156 st = avformat_new_stream(s, NULL);
159 av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
160 return AVERROR(ENOMEM);
163 attr.fragsize = pd->fragment_size;
165 if (s->url[0] != '\0' && strcmp(s->url, "default"))
168 if (!(pd->mainloop = pa_threaded_mainloop_new())) {
170 return AVERROR_EXTERNAL;
173 if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) {
175 return AVERROR_EXTERNAL;
178 pa_context_set_state_callback(pd->context, context_state_cb, pd);
180 if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) {
182 return AVERROR(pa_context_errno(pd->context));
185 pa_threaded_mainloop_lock(pd->mainloop);
187 if (pa_threaded_mainloop_start(pd->mainloop) < 0) {
189 goto unlock_and_fail;
193 pa_context_state_t state;
195 state = pa_context_get_state(pd->context);
197 if (state == PA_CONTEXT_READY)
200 if (!PA_CONTEXT_IS_GOOD(state)) {
201 ret = AVERROR(pa_context_errno(pd->context));
202 goto unlock_and_fail;
205 /* Wait until the context is ready */
206 pa_threaded_mainloop_wait(pd->mainloop);
209 if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, &cmap))) {
210 ret = AVERROR(pa_context_errno(pd->context));
211 goto unlock_and_fail;
214 pa_stream_set_state_callback(pd->stream, stream_state_cb, pd);
215 pa_stream_set_read_callback(pd->stream, stream_request_cb, pd);
216 pa_stream_set_write_callback(pd->stream, stream_request_cb, pd);
217 pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd);
219 ret = pa_stream_connect_record(pd->stream, device, &attr,
220 PA_STREAM_INTERPOLATE_TIMING
221 |PA_STREAM_ADJUST_LATENCY
222 |PA_STREAM_AUTO_TIMING_UPDATE);
225 ret = AVERROR(pa_context_errno(pd->context));
226 goto unlock_and_fail;
230 pa_stream_state_t state;
232 state = pa_stream_get_state(pd->stream);
234 if (state == PA_STREAM_READY)
237 if (!PA_STREAM_IS_GOOD(state)) {
238 ret = AVERROR(pa_context_errno(pd->context));
239 goto unlock_and_fail;
242 /* Wait until the stream is ready */
243 pa_threaded_mainloop_wait(pd->mainloop);
246 /* Query actual fragment size */
247 queried_attr = pa_stream_get_buffer_attr(pd->stream);
248 if (!queried_attr || queried_attr->fragsize > INT_MAX/100) {
249 ret = AVERROR_EXTERNAL;
250 goto unlock_and_fail;
252 pd->fragment_size = queried_attr->fragsize;
254 pa_threaded_mainloop_unlock(pd->mainloop);
256 /* take real parameters */
257 st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
258 st->codecpar->codec_id = codec_id;
259 st->codecpar->sample_rate = pd->sample_rate;
260 st->codecpar->channels = pd->channels;
261 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
263 pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
266 if (!pd->timefilter) {
268 return AVERROR(ENOMEM);
274 pa_threaded_mainloop_unlock(pd->mainloop);
280 static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
282 PulseData *pd = s->priv_data;
285 const void *read_data = NULL;
290 pa_threaded_mainloop_lock(pd->mainloop);
292 CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
297 r = pa_stream_peek(pd->stream, &read_data, &read_length);
298 CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
300 if (read_length <= 0) {
301 pa_threaded_mainloop_wait(pd->mainloop);
302 CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
303 } else if (!read_data) {
304 /* There's a hole in the stream, skip it. We could generate
305 * silence, but that wouldn't work for compressed streams. */
306 r = pa_stream_drop(pd->stream);
307 CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
311 if (av_new_packet(pkt, read_length) < 0) {
312 ret = AVERROR(ENOMEM);
313 goto unlock_and_fail;
317 pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
319 if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
320 enum AVCodecID codec_id =
321 s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
322 int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
323 int frame_duration = read_length / frame_size;
331 pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
333 pd->last_period = frame_duration;
335 av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
338 memcpy(pkt->data, read_data, read_length);
339 pa_stream_drop(pd->stream);
341 pa_threaded_mainloop_unlock(pd->mainloop);
345 pa_threaded_mainloop_unlock(pd->mainloop);
349 static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
351 PulseData *s = h->priv_data;
352 return ff_pulse_audio_get_devices(device_list, s->server, 0);
355 #define OFFSET(a) offsetof(PulseData, a)
356 #define D AV_OPT_FLAG_DECODING_PARAM
358 static const AVOption options[] = {
359 { "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
360 { "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
361 { "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
362 { "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
363 { "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
364 { "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
365 { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
366 { "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, D },
370 static const AVClass pulse_demuxer_class = {
371 .class_name = "Pulse indev",
372 .item_name = av_default_item_name,
374 .version = LIBAVUTIL_VERSION_INT,
375 .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
378 AVInputFormat ff_pulse_demuxer = {
380 .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
381 .priv_data_size = sizeof(PulseData),
382 .read_header = pulse_read_header,
383 .read_packet = pulse_read_packet,
384 .read_close = pulse_close,
385 .get_device_list = pulse_get_device_list,
386 .flags = AVFMT_NOFILE,
387 .priv_class = &pulse_demuxer_class,