summaryrefslogtreecommitdiff
path: root/libavdevice/pulse_audio_dec.c (plain)
blob: 95a1d6ecfa6204246cf4f150cb66d1fc59b01187
1/*
2 * Pulseaudio input
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>
6 *
7 * This file is part of FFmpeg.
8 *
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.
13 *
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.
18 *
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
22 */
23
24#include <pulse/rtclock.h>
25#include <pulse/error.h>
26
27#include "libavutil/internal.h"
28#include "libavutil/opt.h"
29#include "libavutil/time.h"
30
31#include "libavformat/avformat.h"
32#include "libavformat/internal.h"
33#include "pulse_audio_common.h"
34#include "timefilter.h"
35
36#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
37
38typedef struct PulseData {
39 AVClass *class;
40 char *server;
41 char *name;
42 char *stream_name;
43 int sample_rate;
44 int channels;
45 int frame_size;
46 int fragment_size;
47
48 pa_threaded_mainloop *mainloop;
49 pa_context *context;
50 pa_stream *stream;
51
52 TimeFilter *timefilter;
53 int last_period;
54 int wallclock;
55} PulseData;
56
57
58#define CHECK_SUCCESS_GOTO(rerror, expression, label) \
59 do { \
60 if (!(expression)) { \
61 rerror = AVERROR_EXTERNAL; \
62 goto label; \
63 } \
64 } while (0)
65
66#define CHECK_DEAD_GOTO(p, rerror, label) \
67 do { \
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; \
71 goto label; \
72 } \
73 } while (0)
74
75static void context_state_cb(pa_context *c, void *userdata) {
76 PulseData *p = userdata;
77
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);
83 break;
84 }
85}
86
87static void stream_state_cb(pa_stream *s, void * userdata) {
88 PulseData *p = userdata;
89
90 switch (pa_stream_get_state(s)) {
91 case PA_STREAM_READY:
92 case PA_STREAM_FAILED:
93 case PA_STREAM_TERMINATED:
94 pa_threaded_mainloop_signal(p->mainloop, 0);
95 break;
96 }
97}
98
99static void stream_request_cb(pa_stream *s, size_t length, void *userdata) {
100 PulseData *p = userdata;
101
102 pa_threaded_mainloop_signal(p->mainloop, 0);
103}
104
105static void stream_latency_update_cb(pa_stream *s, void *userdata) {
106 PulseData *p = userdata;
107
108 pa_threaded_mainloop_signal(p->mainloop, 0);
109}
110
111static av_cold int pulse_close(AVFormatContext *s)
112{
113 PulseData *pd = s->priv_data;
114
115 if (pd->mainloop)
116 pa_threaded_mainloop_stop(pd->mainloop);
117
118 if (pd->stream)
119 pa_stream_unref(pd->stream);
120 pd->stream = NULL;
121
122 if (pd->context) {
123 pa_context_disconnect(pd->context);
124 pa_context_unref(pd->context);
125 }
126 pd->context = NULL;
127
128 if (pd->mainloop)
129 pa_threaded_mainloop_free(pd->mainloop);
130 pd->mainloop = NULL;
131
132 ff_timefilter_destroy(pd->timefilter);
133 pd->timefilter = NULL;
134
135 return 0;
136}
137
138static av_cold int pulse_read_header(AVFormatContext *s)
139{
140 PulseData *pd = s->priv_data;
141 AVStream *st;
142 char *device = NULL;
143 int ret;
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),
147 pd->sample_rate,
148 pd->channels };
149
150 pa_buffer_attr attr = { -1 };
151
152 st = avformat_new_stream(s, NULL);
153
154 if (!st) {
155 av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
156 return AVERROR(ENOMEM);
157 }
158
159 attr.fragsize = pd->fragment_size;
160
161 if (s->filename[0] != '\0' && strcmp(s->filename, "default"))
162 device = s->filename;
163
164 if (!(pd->mainloop = pa_threaded_mainloop_new())) {
165 pulse_close(s);
166 return AVERROR_EXTERNAL;
167 }
168
169 if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) {
170 pulse_close(s);
171 return AVERROR_EXTERNAL;
172 }
173
174 pa_context_set_state_callback(pd->context, context_state_cb, pd);
175
176 if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) {
177 pulse_close(s);
178 return AVERROR(pa_context_errno(pd->context));
179 }
180
181 pa_threaded_mainloop_lock(pd->mainloop);
182
183 if (pa_threaded_mainloop_start(pd->mainloop) < 0) {
184 ret = -1;
185 goto unlock_and_fail;
186 }
187
188 for (;;) {
189 pa_context_state_t state;
190
191 state = pa_context_get_state(pd->context);
192
193 if (state == PA_CONTEXT_READY)
194 break;
195
196 if (!PA_CONTEXT_IS_GOOD(state)) {
197 ret = AVERROR(pa_context_errno(pd->context));
198 goto unlock_and_fail;
199 }
200
201 /* Wait until the context is ready */
202 pa_threaded_mainloop_wait(pd->mainloop);
203 }
204
205 if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, NULL))) {
206 ret = AVERROR(pa_context_errno(pd->context));
207 goto unlock_and_fail;
208 }
209
210 pa_stream_set_state_callback(pd->stream, stream_state_cb, pd);
211 pa_stream_set_read_callback(pd->stream, stream_request_cb, pd);
212 pa_stream_set_write_callback(pd->stream, stream_request_cb, pd);
213 pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd);
214
215 ret = pa_stream_connect_record(pd->stream, device, &attr,
216 PA_STREAM_INTERPOLATE_TIMING
217 |PA_STREAM_ADJUST_LATENCY
218 |PA_STREAM_AUTO_TIMING_UPDATE);
219
220 if (ret < 0) {
221 ret = AVERROR(pa_context_errno(pd->context));
222 goto unlock_and_fail;
223 }
224
225 for (;;) {
226 pa_stream_state_t state;
227
228 state = pa_stream_get_state(pd->stream);
229
230 if (state == PA_STREAM_READY)
231 break;
232
233 if (!PA_STREAM_IS_GOOD(state)) {
234 ret = AVERROR(pa_context_errno(pd->context));
235 goto unlock_and_fail;
236 }
237
238 /* Wait until the stream is ready */
239 pa_threaded_mainloop_wait(pd->mainloop);
240 }
241
242 pa_threaded_mainloop_unlock(pd->mainloop);
243
244 /* take real parameters */
245 st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
246 st->codecpar->codec_id = codec_id;
247 st->codecpar->sample_rate = pd->sample_rate;
248 st->codecpar->channels = pd->channels;
249 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
250
251 pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate,
252 1000, 1.5E-6);
253
254 if (!pd->timefilter) {
255 pulse_close(s);
256 return AVERROR(ENOMEM);
257 }
258
259 return 0;
260
261unlock_and_fail:
262 pa_threaded_mainloop_unlock(pd->mainloop);
263
264 pulse_close(s);
265 return ret;
266}
267
268static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
269{
270 PulseData *pd = s->priv_data;
271 int ret;
272 size_t read_length;
273 const void *read_data = NULL;
274 int64_t dts;
275 pa_usec_t latency;
276 int negative;
277
278 pa_threaded_mainloop_lock(pd->mainloop);
279
280 CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
281
282 while (!read_data) {
283 int r;
284
285 r = pa_stream_peek(pd->stream, &read_data, &read_length);
286 CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
287
288 if (read_length <= 0) {
289 pa_threaded_mainloop_wait(pd->mainloop);
290 CHECK_DEAD_GOTO(pd, ret, unlock_and_fail);
291 } else if (!read_data) {
292 /* There's a hole in the stream, skip it. We could generate
293 * silence, but that wouldn't work for compressed streams. */
294 r = pa_stream_drop(pd->stream);
295 CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail);
296 }
297 }
298
299 if (av_new_packet(pkt, read_length) < 0) {
300 ret = AVERROR(ENOMEM);
301 goto unlock_and_fail;
302 }
303
304 dts = av_gettime();
305 pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL));
306
307 if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) {
308 enum AVCodecID codec_id =
309 s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
310 int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels);
311 int frame_duration = read_length / frame_size;
312
313
314 if (negative) {
315 dts += latency;
316 } else
317 dts -= latency;
318 if (pd->wallclock)
319 pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period);
320
321 pd->last_period = frame_duration;
322 } else {
323 av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n");
324 }
325
326 memcpy(pkt->data, read_data, read_length);
327 pa_stream_drop(pd->stream);
328
329 pa_threaded_mainloop_unlock(pd->mainloop);
330 return 0;
331
332unlock_and_fail:
333 pa_threaded_mainloop_unlock(pd->mainloop);
334 return ret;
335}
336
337static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
338{
339 PulseData *s = h->priv_data;
340 return ff_pulse_audio_get_devices(device_list, s->server, 0);
341}
342
343#define OFFSET(a) offsetof(PulseData, a)
344#define D AV_OPT_FLAG_DECODING_PARAM
345
346static const AVOption options[] = {
347 { "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
348 { "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
349 { "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
350 { "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
351 { "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
352 { "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
353 { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
354 { "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, D },
355 { NULL },
356};
357
358static const AVClass pulse_demuxer_class = {
359 .class_name = "Pulse demuxer",
360 .item_name = av_default_item_name,
361 .option = options,
362 .version = LIBAVUTIL_VERSION_INT,
363 .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
364};
365
366AVInputFormat ff_pulse_demuxer = {
367 .name = "pulse",
368 .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
369 .priv_data_size = sizeof(PulseData),
370 .read_header = pulse_read_header,
371 .read_packet = pulse_read_packet,
372 .read_close = pulse_close,
373 .get_device_list = pulse_get_device_list,
374 .flags = AVFMT_NOFILE,
375 .priv_class = &pulse_demuxer_class,
376};
377
generated by cgit at 2024-05-17 16:02:02 (GMT)