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| 1 // Copyright 2016 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. |
| 4 |
| 5 #include "remoting/protocol/webrtc_audio_module.h" |
| 6 |
| 7 #include "base/bind.h" |
| 8 #include "base/stl_util.h" |
| 9 #include "base/threading/thread_task_runner_handle.h" |
| 10 |
| 11 namespace remoting { |
| 12 namespace protocol { |
| 13 |
| 14 namespace { |
| 15 |
| 16 const int kSamplingRate = 48000; |
| 17 |
| 18 // Webrtc uses 10ms frames. |
| 19 const int kFrameLengthMs = 10; |
| 20 const int kSamplesPerFrame = kSamplingRate * kFrameLengthMs / 1000; |
| 21 |
| 22 constexpr base::TimeDelta kPollInterval = |
| 23 base::TimeDelta::FromMilliseconds(5 * kFrameLengthMs); |
| 24 const int kChannels = 2; |
| 25 const int kBytesPerSample = 2; |
| 26 |
| 27 } // namespace |
| 28 |
| 29 // webrtc::AudioDeviceModule is a generic interface that aims to provide all |
| 30 // functionality normally supported audio input/output devices, but most of |
| 31 // the functions are never called in Webrtc. This class implements only |
| 32 // functions that are actually used. All unused functions are marked as |
| 33 // NOTREACHED(). |
| 34 |
| 35 WebrtcAudioModule::WebrtcAudioModule() {} |
| 36 WebrtcAudioModule::~WebrtcAudioModule() {} |
| 37 |
| 38 void WebrtcAudioModule::SetAudioTaskRunner( |
| 39 scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner) { |
| 40 DCHECK(!audio_task_runner_); |
| 41 DCHECK(audio_task_runner); |
| 42 audio_task_runner_ = audio_task_runner; |
| 43 } |
| 44 |
| 45 int64_t WebrtcAudioModule::TimeUntilNextProcess() { |
| 46 // We don't need to do anything in Process(), so return an arbitrary value |
| 47 // that's not too low, so that Process() doesn't get called too frequently. |
| 48 return 1000000; |
| 49 } |
| 50 |
| 51 void WebrtcAudioModule::Process() {} |
| 52 |
| 53 int32_t WebrtcAudioModule::ActiveAudioLayer(AudioLayer* audio_layer) const { |
| 54 NOTREACHED(); |
| 55 return -1; |
| 56 } |
| 57 |
| 58 WebrtcAudioModule::ErrorCode WebrtcAudioModule::LastError() const { |
| 59 return kAdmErrNone; |
| 60 } |
| 61 |
| 62 int32_t WebrtcAudioModule::RegisterEventObserver( |
| 63 webrtc::AudioDeviceObserver* event_callback) { |
| 64 return 0; |
| 65 } |
| 66 |
| 67 int32_t WebrtcAudioModule::RegisterAudioCallback( |
| 68 webrtc::AudioTransport* audio_transport) { |
| 69 base::AutoLock lock(lock_); |
| 70 audio_transport_ = audio_transport; |
| 71 return 0; |
| 72 } |
| 73 |
| 74 int32_t WebrtcAudioModule::Init() { |
| 75 base::AutoLock auto_lock(lock_); |
| 76 initialized_ = true; |
| 77 return 0; |
| 78 } |
| 79 |
| 80 int32_t WebrtcAudioModule::Terminate() { |
| 81 base::AutoLock auto_lock(lock_); |
| 82 initialized_ = false; |
| 83 return 0; |
| 84 } |
| 85 |
| 86 bool WebrtcAudioModule::Initialized() const { |
| 87 base::AutoLock auto_lock(lock_); |
| 88 return initialized_; |
| 89 } |
| 90 |
| 91 int16_t WebrtcAudioModule::PlayoutDevices() { |
| 92 return 0; |
| 93 } |
| 94 |
| 95 int16_t WebrtcAudioModule::RecordingDevices() { |
| 96 return 0; |
| 97 } |
| 98 |
| 99 int32_t WebrtcAudioModule::PlayoutDeviceName( |
| 100 uint16_t index, |
| 101 char name[webrtc::kAdmMaxDeviceNameSize], |
| 102 char guid[webrtc::kAdmMaxGuidSize]) { |
| 103 return 0; |
| 104 } |
| 105 |
| 106 int32_t WebrtcAudioModule::RecordingDeviceName( |
| 107 uint16_t index, |
| 108 char name[webrtc::kAdmMaxDeviceNameSize], |
| 109 char guid[webrtc::kAdmMaxGuidSize]) { |
| 110 return 0; |
| 111 } |
| 112 |
| 113 int32_t WebrtcAudioModule::SetPlayoutDevice(uint16_t index) { |
| 114 return 0; |
| 115 } |
| 116 |
| 117 int32_t WebrtcAudioModule::SetPlayoutDevice(WindowsDeviceType device) { |
| 118 return 0; |
| 119 } |
| 120 |
| 121 int32_t WebrtcAudioModule::SetRecordingDevice(uint16_t index) { |
| 122 return 0; |
| 123 } |
| 124 |
| 125 int32_t WebrtcAudioModule::SetRecordingDevice(WindowsDeviceType device) { |
| 126 return 0; |
| 127 } |
| 128 |
| 129 int32_t WebrtcAudioModule::PlayoutIsAvailable(bool* available) { |
| 130 NOTREACHED(); |
| 131 return -1; |
| 132 } |
| 133 |
| 134 int32_t WebrtcAudioModule::InitPlayout() { |
| 135 return 0; |
| 136 } |
| 137 |
| 138 bool WebrtcAudioModule::PlayoutIsInitialized() const { |
| 139 base::AutoLock auto_lock(lock_); |
| 140 return initialized_; |
| 141 } |
| 142 |
| 143 int32_t WebrtcAudioModule::RecordingIsAvailable(bool* available) { |
| 144 NOTREACHED(); |
| 145 return -1; |
| 146 } |
| 147 |
| 148 int32_t WebrtcAudioModule::InitRecording() { |
| 149 return 0; |
| 150 } |
| 151 |
| 152 bool WebrtcAudioModule::RecordingIsInitialized() const { |
| 153 return false; |
| 154 } |
| 155 |
| 156 int32_t WebrtcAudioModule::StartPlayout() { |
| 157 base::AutoLock auto_lock(lock_); |
| 158 if (!playing_ && audio_task_runner_) { |
| 159 audio_task_runner_->PostTask( |
| 160 FROM_HERE, |
| 161 base::Bind(&WebrtcAudioModule::StartPlayoutOnAudioThread, this)); |
| 162 playing_ = true; |
| 163 } |
| 164 return 0; |
| 165 } |
| 166 |
| 167 int32_t WebrtcAudioModule::StopPlayout() { |
| 168 base::AutoLock auto_lock(lock_); |
| 169 if (playing_) { |
| 170 audio_task_runner_->PostTask( |
| 171 FROM_HERE, |
| 172 base::Bind(&WebrtcAudioModule::StopPlayoutOnAudioThread, this)); |
| 173 playing_ = false; |
| 174 } |
| 175 return 0; |
| 176 } |
| 177 |
| 178 bool WebrtcAudioModule::Playing() const { |
| 179 base::AutoLock auto_lock(lock_); |
| 180 return playing_; |
| 181 } |
| 182 |
| 183 int32_t WebrtcAudioModule::StartRecording() { |
| 184 return 0; |
| 185 } |
| 186 |
| 187 int32_t WebrtcAudioModule::StopRecording() { |
| 188 return 0; |
| 189 } |
| 190 |
| 191 bool WebrtcAudioModule::Recording() const { |
| 192 return false; |
| 193 } |
| 194 |
| 195 int32_t WebrtcAudioModule::SetAGC(bool enable) { |
| 196 return 0; |
| 197 } |
| 198 |
| 199 bool WebrtcAudioModule::AGC() const { |
| 200 NOTREACHED(); |
| 201 return false; |
| 202 } |
| 203 |
| 204 int32_t WebrtcAudioModule::SetWaveOutVolume(uint16_t volume_left, |
| 205 uint16_t volume_right) { |
| 206 NOTREACHED(); |
| 207 return -1; |
| 208 } |
| 209 |
| 210 int32_t WebrtcAudioModule::WaveOutVolume(uint16_t* volume_left, |
| 211 uint16_t* volume_right) const { |
| 212 NOTREACHED(); |
| 213 return -1; |
| 214 } |
| 215 |
| 216 int32_t WebrtcAudioModule::InitSpeaker() { |
| 217 return 0; |
| 218 } |
| 219 |
| 220 bool WebrtcAudioModule::SpeakerIsInitialized() const { |
| 221 return false; |
| 222 } |
| 223 |
| 224 int32_t WebrtcAudioModule::InitMicrophone() { |
| 225 return 0; |
| 226 } |
| 227 |
| 228 bool WebrtcAudioModule::MicrophoneIsInitialized() const { |
| 229 return false; |
| 230 } |
| 231 |
| 232 int32_t WebrtcAudioModule::SpeakerVolumeIsAvailable(bool* available) { |
| 233 NOTREACHED(); |
| 234 return -1; |
| 235 } |
| 236 |
| 237 int32_t WebrtcAudioModule::SetSpeakerVolume(uint32_t volume) { |
| 238 NOTREACHED(); |
| 239 return -1; |
| 240 } |
| 241 |
| 242 int32_t WebrtcAudioModule::SpeakerVolume(uint32_t* volume) const { |
| 243 NOTREACHED(); |
| 244 return -1; |
| 245 } |
| 246 |
| 247 int32_t WebrtcAudioModule::MaxSpeakerVolume(uint32_t* max_volume) const { |
| 248 NOTREACHED(); |
| 249 return -1; |
| 250 } |
| 251 |
| 252 int32_t WebrtcAudioModule::MinSpeakerVolume(uint32_t* min_volume) const { |
| 253 NOTREACHED(); |
| 254 return -1; |
| 255 } |
| 256 |
| 257 int32_t WebrtcAudioModule::SpeakerVolumeStepSize(uint16_t* step_size) const { |
| 258 NOTREACHED(); |
| 259 return -1; |
| 260 } |
| 261 |
| 262 int32_t WebrtcAudioModule::MicrophoneVolumeIsAvailable(bool* available) { |
| 263 NOTREACHED(); |
| 264 return -1; |
| 265 } |
| 266 |
| 267 int32_t WebrtcAudioModule::SetMicrophoneVolume(uint32_t volume) { |
| 268 NOTREACHED(); |
| 269 return -1; |
| 270 } |
| 271 |
| 272 int32_t WebrtcAudioModule::MicrophoneVolume(uint32_t* volume) const { |
| 273 NOTREACHED(); |
| 274 return -1; |
| 275 } |
| 276 |
| 277 int32_t WebrtcAudioModule::MaxMicrophoneVolume(uint32_t* max_volume) const { |
| 278 NOTREACHED(); |
| 279 return -1; |
| 280 } |
| 281 |
| 282 int32_t WebrtcAudioModule::MinMicrophoneVolume(uint32_t* min_volume) const { |
| 283 NOTREACHED(); |
| 284 return -1; |
| 285 } |
| 286 |
| 287 int32_t WebrtcAudioModule::MicrophoneVolumeStepSize(uint16_t* step_size) const { |
| 288 NOTREACHED(); |
| 289 return -1; |
| 290 } |
| 291 |
| 292 int32_t WebrtcAudioModule::SpeakerMuteIsAvailable(bool* available) { |
| 293 NOTREACHED(); |
| 294 return -1; |
| 295 } |
| 296 |
| 297 int32_t WebrtcAudioModule::SetSpeakerMute(bool enable) { |
| 298 NOTREACHED(); |
| 299 return -1; |
| 300 } |
| 301 |
| 302 int32_t WebrtcAudioModule::SpeakerMute(bool* enabled) const { |
| 303 NOTREACHED(); |
| 304 return -1; |
| 305 } |
| 306 |
| 307 int32_t WebrtcAudioModule::MicrophoneMuteIsAvailable(bool* available) { |
| 308 NOTREACHED(); |
| 309 return -1; |
| 310 } |
| 311 |
| 312 int32_t WebrtcAudioModule::SetMicrophoneMute(bool enable) { |
| 313 NOTREACHED(); |
| 314 return -1; |
| 315 } |
| 316 |
| 317 int32_t WebrtcAudioModule::MicrophoneMute(bool* enabled) const { |
| 318 NOTREACHED(); |
| 319 return -1; |
| 320 } |
| 321 |
| 322 int32_t WebrtcAudioModule::MicrophoneBoostIsAvailable(bool* available) { |
| 323 NOTREACHED(); |
| 324 return -1; |
| 325 } |
| 326 |
| 327 int32_t WebrtcAudioModule::SetMicrophoneBoost(bool enable) { |
| 328 NOTREACHED(); |
| 329 return -1; |
| 330 } |
| 331 |
| 332 int32_t WebrtcAudioModule::MicrophoneBoost(bool* enabled) const { |
| 333 NOTREACHED(); |
| 334 return -1; |
| 335 } |
| 336 |
| 337 int32_t WebrtcAudioModule::StereoPlayoutIsAvailable(bool* available) const { |
| 338 *available = true; |
| 339 return 0; |
| 340 } |
| 341 |
| 342 int32_t WebrtcAudioModule::SetStereoPlayout(bool enable) { |
| 343 DCHECK(enable); |
| 344 return 0; |
| 345 } |
| 346 |
| 347 int32_t WebrtcAudioModule::StereoPlayout(bool* enabled) const { |
| 348 NOTREACHED(); |
| 349 return -1; |
| 350 } |
| 351 |
| 352 int32_t WebrtcAudioModule::StereoRecordingIsAvailable(bool* available) const { |
| 353 *available = false; |
| 354 return 0; |
| 355 } |
| 356 |
| 357 int32_t WebrtcAudioModule::SetStereoRecording(bool enable) { |
| 358 return 0; |
| 359 } |
| 360 |
| 361 int32_t WebrtcAudioModule::StereoRecording(bool* enabled) const { |
| 362 NOTREACHED(); |
| 363 return -1; |
| 364 } |
| 365 |
| 366 int32_t WebrtcAudioModule::SetRecordingChannel(const ChannelType channel) { |
| 367 return 0; |
| 368 } |
| 369 |
| 370 int32_t WebrtcAudioModule::RecordingChannel(ChannelType* channel) const { |
| 371 NOTREACHED(); |
| 372 return -1; |
| 373 } |
| 374 |
| 375 int32_t WebrtcAudioModule::SetPlayoutBuffer(const BufferType type, |
| 376 uint16_t size_ms) { |
| 377 NOTREACHED(); |
| 378 return -1; |
| 379 } |
| 380 |
| 381 int32_t WebrtcAudioModule::PlayoutBuffer(BufferType* type, |
| 382 uint16_t* size_ms) const { |
| 383 NOTREACHED(); |
| 384 return -1; |
| 385 } |
| 386 |
| 387 int32_t WebrtcAudioModule::PlayoutDelay(uint16_t* delay_ms) const { |
| 388 *delay_ms = 0; |
| 389 return 0; |
| 390 } |
| 391 |
| 392 int32_t WebrtcAudioModule::RecordingDelay(uint16_t* delay_ms) const { |
| 393 NOTREACHED(); |
| 394 return -1; |
| 395 } |
| 396 |
| 397 int32_t WebrtcAudioModule::CPULoad(uint16_t* load) const { |
| 398 NOTREACHED(); |
| 399 return -1; |
| 400 } |
| 401 |
| 402 int32_t WebrtcAudioModule::StartRawOutputFileRecording( |
| 403 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) { |
| 404 NOTREACHED(); |
| 405 return -1; |
| 406 } |
| 407 |
| 408 int32_t WebrtcAudioModule::StopRawOutputFileRecording() { |
| 409 NOTREACHED(); |
| 410 return -1; |
| 411 } |
| 412 |
| 413 int32_t WebrtcAudioModule::StartRawInputFileRecording( |
| 414 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) { |
| 415 NOTREACHED(); |
| 416 return -1; |
| 417 } |
| 418 |
| 419 int32_t WebrtcAudioModule::StopRawInputFileRecording() { |
| 420 NOTREACHED(); |
| 421 return -1; |
| 422 } |
| 423 |
| 424 int32_t WebrtcAudioModule::SetRecordingSampleRate( |
| 425 const uint32_t samples_per_sec) { |
| 426 NOTREACHED(); |
| 427 return -1; |
| 428 } |
| 429 |
| 430 int32_t WebrtcAudioModule::RecordingSampleRate( |
| 431 uint32_t* samples_per_sec) const { |
| 432 NOTREACHED(); |
| 433 return -1; |
| 434 } |
| 435 |
| 436 int32_t WebrtcAudioModule::SetPlayoutSampleRate( |
| 437 const uint32_t samples_per_sec) { |
| 438 NOTREACHED(); |
| 439 return -1; |
| 440 } |
| 441 |
| 442 int32_t WebrtcAudioModule::PlayoutSampleRate(uint32_t* samples_per_sec) const { |
| 443 NOTREACHED(); |
| 444 return -1; |
| 445 } |
| 446 |
| 447 int32_t WebrtcAudioModule::ResetAudioDevice() { |
| 448 NOTREACHED(); |
| 449 return -1; |
| 450 } |
| 451 |
| 452 int32_t WebrtcAudioModule::SetLoudspeakerStatus(bool enable) { |
| 453 NOTREACHED(); |
| 454 return -1; |
| 455 } |
| 456 |
| 457 int32_t WebrtcAudioModule::GetLoudspeakerStatus(bool* enabled) const { |
| 458 NOTREACHED(); |
| 459 return -1; |
| 460 } |
| 461 |
| 462 bool WebrtcAudioModule::BuiltInAECIsAvailable() const { |
| 463 return false; |
| 464 } |
| 465 |
| 466 bool WebrtcAudioModule::BuiltInAGCIsAvailable() const { |
| 467 return false; |
| 468 } |
| 469 |
| 470 bool WebrtcAudioModule::BuiltInNSIsAvailable() const { |
| 471 return false; |
| 472 } |
| 473 |
| 474 int32_t WebrtcAudioModule::EnableBuiltInAEC(bool enable) { |
| 475 NOTREACHED(); |
| 476 return -1; |
| 477 } |
| 478 |
| 479 int32_t WebrtcAudioModule::EnableBuiltInAGC(bool enable) { |
| 480 NOTREACHED(); |
| 481 return -1; |
| 482 } |
| 483 |
| 484 int32_t WebrtcAudioModule::EnableBuiltInNS(bool enable) { |
| 485 NOTREACHED(); |
| 486 return -1; |
| 487 } |
| 488 |
| 489 #if defined(WEBRTC_IOS) |
| 490 int WebrtcAudioModule::GetPlayoutAudioParameters( |
| 491 AudioParameters* params) const { |
| 492 NOTREACHED(); |
| 493 return -1; |
| 494 } |
| 495 |
| 496 int WebrtcAudioModule::GetRecordAudioParameters(AudioParameters* params) const { |
| 497 NOTREACHED(); |
| 498 return -1; |
| 499 } |
| 500 #endif // WEBRTC_IOS |
| 501 |
| 502 void WebrtcAudioModule::StartPlayoutOnAudioThread() { |
| 503 DCHECK(audio_task_runner_->BelongsToCurrentThread()); |
| 504 poll_timer_.Start( |
| 505 FROM_HERE, kPollInterval, |
| 506 base::Bind(&WebrtcAudioModule::PollFromSource, base::Unretained(this))); |
| 507 } |
| 508 |
| 509 void WebrtcAudioModule::StopPlayoutOnAudioThread() { |
| 510 DCHECK(audio_task_runner_->BelongsToCurrentThread()); |
| 511 poll_timer_.Stop(); |
| 512 } |
| 513 |
| 514 void WebrtcAudioModule::PollFromSource() { |
| 515 DCHECK(audio_task_runner_->BelongsToCurrentThread()); |
| 516 |
| 517 base::AutoLock lock(lock_); |
| 518 if (!audio_transport_) |
| 519 return; |
| 520 |
| 521 for (int i = 0; i < kPollInterval.InMilliseconds() / kFrameLengthMs; i++) { |
| 522 int64_t elapsed_time_ms = -1; |
| 523 int64_t ntp_time_ms = -1; |
| 524 char data[kBytesPerSample * kChannels * kSamplesPerFrame]; |
| 525 audio_transport_->PullRenderData(kBytesPerSample * 8, kSamplingRate, |
| 526 kChannels, kSamplesPerFrame, data, |
| 527 &elapsed_time_ms, &ntp_time_ms); |
| 528 } |
| 529 } |
| 530 |
| 531 } // namespace protocol |
| 532 } // namespace remoting |
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