| 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 | #include "libavformat/avformat.h" |
| 27 | #include "libavformat/internal.h" |
| 28 | #include "libavutil/opt.h" |
| 29 | #include "libavutil/time.h" |
| 30 | #include "pulse_audio_common.h" |
| 31 | #include "timefilter.h" |
| 32 | |
| 33 | #define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE) |
| 34 | |
| 35 | typedef struct PulseData { |
| 36 | AVClass *class; |
| 37 | char *server; |
| 38 | char *name; |
| 39 | char *stream_name; |
| 40 | int sample_rate; |
| 41 | int channels; |
| 42 | int frame_size; |
| 43 | int fragment_size; |
| 44 | |
| 45 | pa_threaded_mainloop *mainloop; |
| 46 | pa_context *context; |
| 47 | pa_stream *stream; |
| 48 | |
| 49 | TimeFilter *timefilter; |
| 50 | int last_period; |
| 51 | int wallclock; |
| 52 | } PulseData; |
| 53 | |
| 54 | |
| 55 | #define CHECK_SUCCESS_GOTO(rerror, expression, label) \ |
| 56 | do { \ |
| 57 | if (!(expression)) { \ |
| 58 | rerror = AVERROR_EXTERNAL; \ |
| 59 | goto label; \ |
| 60 | } \ |
| 61 | } while(0); |
| 62 | |
| 63 | #define CHECK_DEAD_GOTO(p, rerror, label) \ |
| 64 | do { \ |
| 65 | if (!(p)->context || !PA_CONTEXT_IS_GOOD(pa_context_get_state((p)->context)) || \ |
| 66 | !(p)->stream || !PA_STREAM_IS_GOOD(pa_stream_get_state((p)->stream))) { \ |
| 67 | rerror = AVERROR_EXTERNAL; \ |
| 68 | goto label; \ |
| 69 | } \ |
| 70 | } while(0); |
| 71 | |
| 72 | static void context_state_cb(pa_context *c, void *userdata) { |
| 73 | PulseData *p = userdata; |
| 74 | |
| 75 | switch (pa_context_get_state(c)) { |
| 76 | case PA_CONTEXT_READY: |
| 77 | case PA_CONTEXT_TERMINATED: |
| 78 | case PA_CONTEXT_FAILED: |
| 79 | pa_threaded_mainloop_signal(p->mainloop, 0); |
| 80 | break; |
| 81 | } |
| 82 | } |
| 83 | |
| 84 | static void stream_state_cb(pa_stream *s, void * userdata) { |
| 85 | PulseData *p = userdata; |
| 86 | |
| 87 | switch (pa_stream_get_state(s)) { |
| 88 | case PA_STREAM_READY: |
| 89 | case PA_STREAM_FAILED: |
| 90 | case PA_STREAM_TERMINATED: |
| 91 | pa_threaded_mainloop_signal(p->mainloop, 0); |
| 92 | break; |
| 93 | } |
| 94 | } |
| 95 | |
| 96 | static void stream_request_cb(pa_stream *s, size_t length, void *userdata) { |
| 97 | PulseData *p = userdata; |
| 98 | |
| 99 | pa_threaded_mainloop_signal(p->mainloop, 0); |
| 100 | } |
| 101 | |
| 102 | static void stream_latency_update_cb(pa_stream *s, void *userdata) { |
| 103 | PulseData *p = userdata; |
| 104 | |
| 105 | pa_threaded_mainloop_signal(p->mainloop, 0); |
| 106 | } |
| 107 | |
| 108 | static av_cold int pulse_close(AVFormatContext *s) |
| 109 | { |
| 110 | PulseData *pd = s->priv_data; |
| 111 | |
| 112 | if (pd->mainloop) |
| 113 | pa_threaded_mainloop_stop(pd->mainloop); |
| 114 | |
| 115 | if (pd->stream) |
| 116 | pa_stream_unref(pd->stream); |
| 117 | pd->stream = NULL; |
| 118 | |
| 119 | if (pd->context) { |
| 120 | pa_context_disconnect(pd->context); |
| 121 | pa_context_unref(pd->context); |
| 122 | } |
| 123 | pd->context = NULL; |
| 124 | |
| 125 | if (pd->mainloop) |
| 126 | pa_threaded_mainloop_free(pd->mainloop); |
| 127 | pd->mainloop = NULL; |
| 128 | |
| 129 | ff_timefilter_destroy(pd->timefilter); |
| 130 | pd->timefilter = NULL; |
| 131 | |
| 132 | return 0; |
| 133 | } |
| 134 | |
| 135 | static av_cold int pulse_read_header(AVFormatContext *s) |
| 136 | { |
| 137 | PulseData *pd = s->priv_data; |
| 138 | AVStream *st; |
| 139 | char *device = NULL; |
| 140 | int ret; |
| 141 | enum AVCodecID codec_id = |
| 142 | s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; |
| 143 | const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id), |
| 144 | pd->sample_rate, |
| 145 | pd->channels }; |
| 146 | |
| 147 | pa_buffer_attr attr = { -1 }; |
| 148 | |
| 149 | st = avformat_new_stream(s, NULL); |
| 150 | |
| 151 | if (!st) { |
| 152 | av_log(s, AV_LOG_ERROR, "Cannot add stream\n"); |
| 153 | return AVERROR(ENOMEM); |
| 154 | } |
| 155 | |
| 156 | attr.fragsize = pd->fragment_size; |
| 157 | |
| 158 | if (s->filename[0] != '\0' && strcmp(s->filename, "default")) |
| 159 | device = s->filename; |
| 160 | |
| 161 | if (!(pd->mainloop = pa_threaded_mainloop_new())) { |
| 162 | pulse_close(s); |
| 163 | return AVERROR_EXTERNAL; |
| 164 | } |
| 165 | |
| 166 | if (!(pd->context = pa_context_new(pa_threaded_mainloop_get_api(pd->mainloop), pd->name))) { |
| 167 | pulse_close(s); |
| 168 | return AVERROR_EXTERNAL; |
| 169 | } |
| 170 | |
| 171 | pa_context_set_state_callback(pd->context, context_state_cb, pd); |
| 172 | |
| 173 | if (pa_context_connect(pd->context, pd->server, 0, NULL) < 0) { |
| 174 | pulse_close(s); |
| 175 | return AVERROR(pa_context_errno(pd->context)); |
| 176 | } |
| 177 | |
| 178 | pa_threaded_mainloop_lock(pd->mainloop); |
| 179 | |
| 180 | if (pa_threaded_mainloop_start(pd->mainloop) < 0) { |
| 181 | ret = -1; |
| 182 | goto unlock_and_fail; |
| 183 | } |
| 184 | |
| 185 | for (;;) { |
| 186 | pa_context_state_t state; |
| 187 | |
| 188 | state = pa_context_get_state(pd->context); |
| 189 | |
| 190 | if (state == PA_CONTEXT_READY) |
| 191 | break; |
| 192 | |
| 193 | if (!PA_CONTEXT_IS_GOOD(state)) { |
| 194 | ret = AVERROR(pa_context_errno(pd->context)); |
| 195 | goto unlock_and_fail; |
| 196 | } |
| 197 | |
| 198 | /* Wait until the context is ready */ |
| 199 | pa_threaded_mainloop_wait(pd->mainloop); |
| 200 | } |
| 201 | |
| 202 | if (!(pd->stream = pa_stream_new(pd->context, pd->stream_name, &ss, NULL))) { |
| 203 | ret = AVERROR(pa_context_errno(pd->context)); |
| 204 | goto unlock_and_fail; |
| 205 | } |
| 206 | |
| 207 | pa_stream_set_state_callback(pd->stream, stream_state_cb, pd); |
| 208 | pa_stream_set_read_callback(pd->stream, stream_request_cb, pd); |
| 209 | pa_stream_set_write_callback(pd->stream, stream_request_cb, pd); |
| 210 | pa_stream_set_latency_update_callback(pd->stream, stream_latency_update_cb, pd); |
| 211 | |
| 212 | ret = pa_stream_connect_record(pd->stream, device, &attr, |
| 213 | PA_STREAM_INTERPOLATE_TIMING |
| 214 | |PA_STREAM_ADJUST_LATENCY |
| 215 | |PA_STREAM_AUTO_TIMING_UPDATE); |
| 216 | |
| 217 | if (ret < 0) { |
| 218 | ret = AVERROR(pa_context_errno(pd->context)); |
| 219 | goto unlock_and_fail; |
| 220 | } |
| 221 | |
| 222 | for (;;) { |
| 223 | pa_stream_state_t state; |
| 224 | |
| 225 | state = pa_stream_get_state(pd->stream); |
| 226 | |
| 227 | if (state == PA_STREAM_READY) |
| 228 | break; |
| 229 | |
| 230 | if (!PA_STREAM_IS_GOOD(state)) { |
| 231 | ret = AVERROR(pa_context_errno(pd->context)); |
| 232 | goto unlock_and_fail; |
| 233 | } |
| 234 | |
| 235 | /* Wait until the stream is ready */ |
| 236 | pa_threaded_mainloop_wait(pd->mainloop); |
| 237 | } |
| 238 | |
| 239 | pa_threaded_mainloop_unlock(pd->mainloop); |
| 240 | |
| 241 | /* take real parameters */ |
| 242 | st->codec->codec_type = AVMEDIA_TYPE_AUDIO; |
| 243 | st->codec->codec_id = codec_id; |
| 244 | st->codec->sample_rate = pd->sample_rate; |
| 245 | st->codec->channels = pd->channels; |
| 246 | avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ |
| 247 | |
| 248 | pd->timefilter = ff_timefilter_new(1000000.0 / pd->sample_rate, |
| 249 | 1000, 1.5E-6); |
| 250 | |
| 251 | if (!pd->timefilter) { |
| 252 | pulse_close(s); |
| 253 | return AVERROR(ENOMEM); |
| 254 | } |
| 255 | |
| 256 | return 0; |
| 257 | |
| 258 | unlock_and_fail: |
| 259 | pa_threaded_mainloop_unlock(pd->mainloop); |
| 260 | |
| 261 | pulse_close(s); |
| 262 | return ret; |
| 263 | } |
| 264 | |
| 265 | static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt) |
| 266 | { |
| 267 | PulseData *pd = s->priv_data; |
| 268 | int ret; |
| 269 | size_t read_length; |
| 270 | const void *read_data = NULL; |
| 271 | int64_t dts; |
| 272 | pa_usec_t latency; |
| 273 | int negative; |
| 274 | |
| 275 | pa_threaded_mainloop_lock(pd->mainloop); |
| 276 | |
| 277 | CHECK_DEAD_GOTO(pd, ret, unlock_and_fail); |
| 278 | |
| 279 | while (!read_data) { |
| 280 | int r; |
| 281 | |
| 282 | r = pa_stream_peek(pd->stream, &read_data, &read_length); |
| 283 | CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail); |
| 284 | |
| 285 | if (read_length <= 0) { |
| 286 | pa_threaded_mainloop_wait(pd->mainloop); |
| 287 | CHECK_DEAD_GOTO(pd, ret, unlock_and_fail); |
| 288 | } else if (!read_data) { |
| 289 | /* There's a hole in the stream, skip it. We could generate |
| 290 | * silence, but that wouldn't work for compressed streams. */ |
| 291 | r = pa_stream_drop(pd->stream); |
| 292 | CHECK_SUCCESS_GOTO(ret, r == 0, unlock_and_fail); |
| 293 | } |
| 294 | } |
| 295 | |
| 296 | if (av_new_packet(pkt, read_length) < 0) { |
| 297 | ret = AVERROR(ENOMEM); |
| 298 | goto unlock_and_fail; |
| 299 | } |
| 300 | |
| 301 | dts = av_gettime(); |
| 302 | pa_operation_unref(pa_stream_update_timing_info(pd->stream, NULL, NULL)); |
| 303 | |
| 304 | if (pa_stream_get_latency(pd->stream, &latency, &negative) >= 0) { |
| 305 | enum AVCodecID codec_id = |
| 306 | s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id; |
| 307 | int frame_size = ((av_get_bits_per_sample(codec_id) >> 3) * pd->channels); |
| 308 | int frame_duration = read_length / frame_size; |
| 309 | |
| 310 | |
| 311 | if (negative) { |
| 312 | dts += latency; |
| 313 | } else |
| 314 | dts -= latency; |
| 315 | if (pd->wallclock) |
| 316 | pkt->pts = ff_timefilter_update(pd->timefilter, dts, pd->last_period); |
| 317 | |
| 318 | pd->last_period = frame_duration; |
| 319 | } else { |
| 320 | av_log(s, AV_LOG_WARNING, "pa_stream_get_latency() failed\n"); |
| 321 | } |
| 322 | |
| 323 | memcpy(pkt->data, read_data, read_length); |
| 324 | pa_stream_drop(pd->stream); |
| 325 | |
| 326 | pa_threaded_mainloop_unlock(pd->mainloop); |
| 327 | return 0; |
| 328 | |
| 329 | unlock_and_fail: |
| 330 | pa_threaded_mainloop_unlock(pd->mainloop); |
| 331 | return ret; |
| 332 | } |
| 333 | |
| 334 | static int pulse_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) |
| 335 | { |
| 336 | PulseData *s = h->priv_data; |
| 337 | return ff_pulse_audio_get_devices(device_list, s->server, 0); |
| 338 | } |
| 339 | |
| 340 | #define OFFSET(a) offsetof(PulseData, a) |
| 341 | #define D AV_OPT_FLAG_DECODING_PARAM |
| 342 | |
| 343 | static const AVOption options[] = { |
| 344 | { "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D }, |
| 345 | { "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D }, |
| 346 | { "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D }, |
| 347 | { "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D }, |
| 348 | { "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D }, |
| 349 | { "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D }, |
| 350 | { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D }, |
| 351 | { "wallclock", "set the initial pts using the current time", OFFSET(wallclock), AV_OPT_TYPE_INT, {.i64 = 1}, -1, 1, D }, |
| 352 | { NULL }, |
| 353 | }; |
| 354 | |
| 355 | static const AVClass pulse_demuxer_class = { |
| 356 | .class_name = "Pulse demuxer", |
| 357 | .item_name = av_default_item_name, |
| 358 | .option = options, |
| 359 | .version = LIBAVUTIL_VERSION_INT, |
| 360 | .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT, |
| 361 | }; |
| 362 | |
| 363 | AVInputFormat ff_pulse_demuxer = { |
| 364 | .name = "pulse", |
| 365 | .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"), |
| 366 | .priv_data_size = sizeof(PulseData), |
| 367 | .read_header = pulse_read_header, |
| 368 | .read_packet = pulse_read_packet, |
| 369 | .read_close = pulse_close, |
| 370 | .get_device_list = pulse_get_device_list, |
| 371 | .flags = AVFMT_NOFILE, |
| 372 | .priv_class = &pulse_demuxer_class, |
| 373 | }; |