| OLD | NEW |
| (Empty) |
| 1 /* | |
| 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #include "webrtc/api/test/fakeaudiocapturemodule.h" | |
| 12 | |
| 13 #include "webrtc/base/checks.h" | |
| 14 #include "webrtc/base/common.h" | |
| 15 #include "webrtc/base/refcount.h" | |
| 16 #include "webrtc/base/thread.h" | |
| 17 #include "webrtc/base/timeutils.h" | |
| 18 | |
| 19 // Audio sample value that is high enough that it doesn't occur naturally when | |
| 20 // frames are being faked. E.g. NetEq will not generate this large sample value | |
| 21 // unless it has received an audio frame containing a sample of this value. | |
| 22 // Even simpler buffers would likely just contain audio sample values of 0. | |
| 23 static const int kHighSampleValue = 10000; | |
| 24 | |
| 25 // Same value as src/modules/audio_device/main/source/audio_device_config.h in | |
| 26 // https://code.google.com/p/webrtc/ | |
| 27 static const int kAdmMaxIdleTimeProcess = 1000; | |
| 28 | |
| 29 // Constants here are derived by running VoE using a real ADM. | |
| 30 // The constants correspond to 10ms of mono audio at 44kHz. | |
| 31 static const int kTimePerFrameMs = 10; | |
| 32 static const uint8_t kNumberOfChannels = 1; | |
| 33 static const int kSamplesPerSecond = 44000; | |
| 34 static const int kTotalDelayMs = 0; | |
| 35 static const int kClockDriftMs = 0; | |
| 36 static const uint32_t kMaxVolume = 14392; | |
| 37 | |
| 38 enum { | |
| 39 MSG_START_PROCESS, | |
| 40 MSG_RUN_PROCESS, | |
| 41 }; | |
| 42 | |
| 43 FakeAudioCaptureModule::FakeAudioCaptureModule() | |
| 44 : last_process_time_ms_(0), | |
| 45 audio_callback_(nullptr), | |
| 46 recording_(false), | |
| 47 playing_(false), | |
| 48 play_is_initialized_(false), | |
| 49 rec_is_initialized_(false), | |
| 50 current_mic_level_(kMaxVolume), | |
| 51 started_(false), | |
| 52 next_frame_time_(0), | |
| 53 frames_received_(0) { | |
| 54 } | |
| 55 | |
| 56 FakeAudioCaptureModule::~FakeAudioCaptureModule() { | |
| 57 if (process_thread_) { | |
| 58 process_thread_->Stop(); | |
| 59 } | |
| 60 } | |
| 61 | |
| 62 rtc::scoped_refptr<FakeAudioCaptureModule> FakeAudioCaptureModule::Create() { | |
| 63 rtc::scoped_refptr<FakeAudioCaptureModule> capture_module( | |
| 64 new rtc::RefCountedObject<FakeAudioCaptureModule>()); | |
| 65 if (!capture_module->Initialize()) { | |
| 66 return nullptr; | |
| 67 } | |
| 68 return capture_module; | |
| 69 } | |
| 70 | |
| 71 int FakeAudioCaptureModule::frames_received() const { | |
| 72 rtc::CritScope cs(&crit_); | |
| 73 return frames_received_; | |
| 74 } | |
| 75 | |
| 76 int64_t FakeAudioCaptureModule::TimeUntilNextProcess() { | |
| 77 const int64_t current_time = rtc::TimeMillis(); | |
| 78 if (current_time < last_process_time_ms_) { | |
| 79 // TODO: wraparound could be handled more gracefully. | |
| 80 return 0; | |
| 81 } | |
| 82 const int64_t elapsed_time = current_time - last_process_time_ms_; | |
| 83 if (kAdmMaxIdleTimeProcess < elapsed_time) { | |
| 84 return 0; | |
| 85 } | |
| 86 return kAdmMaxIdleTimeProcess - elapsed_time; | |
| 87 } | |
| 88 | |
| 89 void FakeAudioCaptureModule::Process() { | |
| 90 last_process_time_ms_ = rtc::TimeMillis(); | |
| 91 } | |
| 92 | |
| 93 int32_t FakeAudioCaptureModule::ActiveAudioLayer( | |
| 94 AudioLayer* /*audio_layer*/) const { | |
| 95 RTC_NOTREACHED(); | |
| 96 return 0; | |
| 97 } | |
| 98 | |
| 99 webrtc::AudioDeviceModule::ErrorCode FakeAudioCaptureModule::LastError() const { | |
| 100 RTC_NOTREACHED(); | |
| 101 return webrtc::AudioDeviceModule::kAdmErrNone; | |
| 102 } | |
| 103 | |
| 104 int32_t FakeAudioCaptureModule::RegisterEventObserver( | |
| 105 webrtc::AudioDeviceObserver* /*event_callback*/) { | |
| 106 // Only used to report warnings and errors. This fake implementation won't | |
| 107 // generate any so discard this callback. | |
| 108 return 0; | |
| 109 } | |
| 110 | |
| 111 int32_t FakeAudioCaptureModule::RegisterAudioCallback( | |
| 112 webrtc::AudioTransport* audio_callback) { | |
| 113 rtc::CritScope cs(&crit_callback_); | |
| 114 audio_callback_ = audio_callback; | |
| 115 return 0; | |
| 116 } | |
| 117 | |
| 118 int32_t FakeAudioCaptureModule::Init() { | |
| 119 // Initialize is called by the factory method. Safe to ignore this Init call. | |
| 120 return 0; | |
| 121 } | |
| 122 | |
| 123 int32_t FakeAudioCaptureModule::Terminate() { | |
| 124 // Clean up in the destructor. No action here, just success. | |
| 125 return 0; | |
| 126 } | |
| 127 | |
| 128 bool FakeAudioCaptureModule::Initialized() const { | |
| 129 RTC_NOTREACHED(); | |
| 130 return 0; | |
| 131 } | |
| 132 | |
| 133 int16_t FakeAudioCaptureModule::PlayoutDevices() { | |
| 134 RTC_NOTREACHED(); | |
| 135 return 0; | |
| 136 } | |
| 137 | |
| 138 int16_t FakeAudioCaptureModule::RecordingDevices() { | |
| 139 RTC_NOTREACHED(); | |
| 140 return 0; | |
| 141 } | |
| 142 | |
| 143 int32_t FakeAudioCaptureModule::PlayoutDeviceName( | |
| 144 uint16_t /*index*/, | |
| 145 char /*name*/[webrtc::kAdmMaxDeviceNameSize], | |
| 146 char /*guid*/[webrtc::kAdmMaxGuidSize]) { | |
| 147 RTC_NOTREACHED(); | |
| 148 return 0; | |
| 149 } | |
| 150 | |
| 151 int32_t FakeAudioCaptureModule::RecordingDeviceName( | |
| 152 uint16_t /*index*/, | |
| 153 char /*name*/[webrtc::kAdmMaxDeviceNameSize], | |
| 154 char /*guid*/[webrtc::kAdmMaxGuidSize]) { | |
| 155 RTC_NOTREACHED(); | |
| 156 return 0; | |
| 157 } | |
| 158 | |
| 159 int32_t FakeAudioCaptureModule::SetPlayoutDevice(uint16_t /*index*/) { | |
| 160 // No playout device, just playing from file. Return success. | |
| 161 return 0; | |
| 162 } | |
| 163 | |
| 164 int32_t FakeAudioCaptureModule::SetPlayoutDevice(WindowsDeviceType /*device*/) { | |
| 165 if (play_is_initialized_) { | |
| 166 return -1; | |
| 167 } | |
| 168 return 0; | |
| 169 } | |
| 170 | |
| 171 int32_t FakeAudioCaptureModule::SetRecordingDevice(uint16_t /*index*/) { | |
| 172 // No recording device, just dropping audio. Return success. | |
| 173 return 0; | |
| 174 } | |
| 175 | |
| 176 int32_t FakeAudioCaptureModule::SetRecordingDevice( | |
| 177 WindowsDeviceType /*device*/) { | |
| 178 if (rec_is_initialized_) { | |
| 179 return -1; | |
| 180 } | |
| 181 return 0; | |
| 182 } | |
| 183 | |
| 184 int32_t FakeAudioCaptureModule::PlayoutIsAvailable(bool* /*available*/) { | |
| 185 RTC_NOTREACHED(); | |
| 186 return 0; | |
| 187 } | |
| 188 | |
| 189 int32_t FakeAudioCaptureModule::InitPlayout() { | |
| 190 play_is_initialized_ = true; | |
| 191 return 0; | |
| 192 } | |
| 193 | |
| 194 bool FakeAudioCaptureModule::PlayoutIsInitialized() const { | |
| 195 return play_is_initialized_; | |
| 196 } | |
| 197 | |
| 198 int32_t FakeAudioCaptureModule::RecordingIsAvailable(bool* /*available*/) { | |
| 199 RTC_NOTREACHED(); | |
| 200 return 0; | |
| 201 } | |
| 202 | |
| 203 int32_t FakeAudioCaptureModule::InitRecording() { | |
| 204 rec_is_initialized_ = true; | |
| 205 return 0; | |
| 206 } | |
| 207 | |
| 208 bool FakeAudioCaptureModule::RecordingIsInitialized() const { | |
| 209 return rec_is_initialized_; | |
| 210 } | |
| 211 | |
| 212 int32_t FakeAudioCaptureModule::StartPlayout() { | |
| 213 if (!play_is_initialized_) { | |
| 214 return -1; | |
| 215 } | |
| 216 { | |
| 217 rtc::CritScope cs(&crit_); | |
| 218 playing_ = true; | |
| 219 } | |
| 220 bool start = true; | |
| 221 UpdateProcessing(start); | |
| 222 return 0; | |
| 223 } | |
| 224 | |
| 225 int32_t FakeAudioCaptureModule::StopPlayout() { | |
| 226 bool start = false; | |
| 227 { | |
| 228 rtc::CritScope cs(&crit_); | |
| 229 playing_ = false; | |
| 230 start = ShouldStartProcessing(); | |
| 231 } | |
| 232 UpdateProcessing(start); | |
| 233 return 0; | |
| 234 } | |
| 235 | |
| 236 bool FakeAudioCaptureModule::Playing() const { | |
| 237 rtc::CritScope cs(&crit_); | |
| 238 return playing_; | |
| 239 } | |
| 240 | |
| 241 int32_t FakeAudioCaptureModule::StartRecording() { | |
| 242 if (!rec_is_initialized_) { | |
| 243 return -1; | |
| 244 } | |
| 245 { | |
| 246 rtc::CritScope cs(&crit_); | |
| 247 recording_ = true; | |
| 248 } | |
| 249 bool start = true; | |
| 250 UpdateProcessing(start); | |
| 251 return 0; | |
| 252 } | |
| 253 | |
| 254 int32_t FakeAudioCaptureModule::StopRecording() { | |
| 255 bool start = false; | |
| 256 { | |
| 257 rtc::CritScope cs(&crit_); | |
| 258 recording_ = false; | |
| 259 start = ShouldStartProcessing(); | |
| 260 } | |
| 261 UpdateProcessing(start); | |
| 262 return 0; | |
| 263 } | |
| 264 | |
| 265 bool FakeAudioCaptureModule::Recording() const { | |
| 266 rtc::CritScope cs(&crit_); | |
| 267 return recording_; | |
| 268 } | |
| 269 | |
| 270 int32_t FakeAudioCaptureModule::SetAGC(bool /*enable*/) { | |
| 271 // No AGC but not needed since audio is pregenerated. Return success. | |
| 272 return 0; | |
| 273 } | |
| 274 | |
| 275 bool FakeAudioCaptureModule::AGC() const { | |
| 276 RTC_NOTREACHED(); | |
| 277 return 0; | |
| 278 } | |
| 279 | |
| 280 int32_t FakeAudioCaptureModule::SetWaveOutVolume(uint16_t /*volume_left*/, | |
| 281 uint16_t /*volume_right*/) { | |
| 282 RTC_NOTREACHED(); | |
| 283 return 0; | |
| 284 } | |
| 285 | |
| 286 int32_t FakeAudioCaptureModule::WaveOutVolume( | |
| 287 uint16_t* /*volume_left*/, | |
| 288 uint16_t* /*volume_right*/) const { | |
| 289 RTC_NOTREACHED(); | |
| 290 return 0; | |
| 291 } | |
| 292 | |
| 293 int32_t FakeAudioCaptureModule::InitSpeaker() { | |
| 294 // No speaker, just playing from file. Return success. | |
| 295 return 0; | |
| 296 } | |
| 297 | |
| 298 bool FakeAudioCaptureModule::SpeakerIsInitialized() const { | |
| 299 RTC_NOTREACHED(); | |
| 300 return 0; | |
| 301 } | |
| 302 | |
| 303 int32_t FakeAudioCaptureModule::InitMicrophone() { | |
| 304 // No microphone, just playing from file. Return success. | |
| 305 return 0; | |
| 306 } | |
| 307 | |
| 308 bool FakeAudioCaptureModule::MicrophoneIsInitialized() const { | |
| 309 RTC_NOTREACHED(); | |
| 310 return 0; | |
| 311 } | |
| 312 | |
| 313 int32_t FakeAudioCaptureModule::SpeakerVolumeIsAvailable(bool* /*available*/) { | |
| 314 RTC_NOTREACHED(); | |
| 315 return 0; | |
| 316 } | |
| 317 | |
| 318 int32_t FakeAudioCaptureModule::SetSpeakerVolume(uint32_t /*volume*/) { | |
| 319 RTC_NOTREACHED(); | |
| 320 return 0; | |
| 321 } | |
| 322 | |
| 323 int32_t FakeAudioCaptureModule::SpeakerVolume(uint32_t* /*volume*/) const { | |
| 324 RTC_NOTREACHED(); | |
| 325 return 0; | |
| 326 } | |
| 327 | |
| 328 int32_t FakeAudioCaptureModule::MaxSpeakerVolume( | |
| 329 uint32_t* /*max_volume*/) const { | |
| 330 RTC_NOTREACHED(); | |
| 331 return 0; | |
| 332 } | |
| 333 | |
| 334 int32_t FakeAudioCaptureModule::MinSpeakerVolume( | |
| 335 uint32_t* /*min_volume*/) const { | |
| 336 RTC_NOTREACHED(); | |
| 337 return 0; | |
| 338 } | |
| 339 | |
| 340 int32_t FakeAudioCaptureModule::SpeakerVolumeStepSize( | |
| 341 uint16_t* /*step_size*/) const { | |
| 342 RTC_NOTREACHED(); | |
| 343 return 0; | |
| 344 } | |
| 345 | |
| 346 int32_t FakeAudioCaptureModule::MicrophoneVolumeIsAvailable( | |
| 347 bool* /*available*/) { | |
| 348 RTC_NOTREACHED(); | |
| 349 return 0; | |
| 350 } | |
| 351 | |
| 352 int32_t FakeAudioCaptureModule::SetMicrophoneVolume(uint32_t volume) { | |
| 353 rtc::CritScope cs(&crit_); | |
| 354 current_mic_level_ = volume; | |
| 355 return 0; | |
| 356 } | |
| 357 | |
| 358 int32_t FakeAudioCaptureModule::MicrophoneVolume(uint32_t* volume) const { | |
| 359 rtc::CritScope cs(&crit_); | |
| 360 *volume = current_mic_level_; | |
| 361 return 0; | |
| 362 } | |
| 363 | |
| 364 int32_t FakeAudioCaptureModule::MaxMicrophoneVolume( | |
| 365 uint32_t* max_volume) const { | |
| 366 *max_volume = kMaxVolume; | |
| 367 return 0; | |
| 368 } | |
| 369 | |
| 370 int32_t FakeAudioCaptureModule::MinMicrophoneVolume( | |
| 371 uint32_t* /*min_volume*/) const { | |
| 372 RTC_NOTREACHED(); | |
| 373 return 0; | |
| 374 } | |
| 375 | |
| 376 int32_t FakeAudioCaptureModule::MicrophoneVolumeStepSize( | |
| 377 uint16_t* /*step_size*/) const { | |
| 378 RTC_NOTREACHED(); | |
| 379 return 0; | |
| 380 } | |
| 381 | |
| 382 int32_t FakeAudioCaptureModule::SpeakerMuteIsAvailable(bool* /*available*/) { | |
| 383 RTC_NOTREACHED(); | |
| 384 return 0; | |
| 385 } | |
| 386 | |
| 387 int32_t FakeAudioCaptureModule::SetSpeakerMute(bool /*enable*/) { | |
| 388 RTC_NOTREACHED(); | |
| 389 return 0; | |
| 390 } | |
| 391 | |
| 392 int32_t FakeAudioCaptureModule::SpeakerMute(bool* /*enabled*/) const { | |
| 393 RTC_NOTREACHED(); | |
| 394 return 0; | |
| 395 } | |
| 396 | |
| 397 int32_t FakeAudioCaptureModule::MicrophoneMuteIsAvailable(bool* /*available*/) { | |
| 398 RTC_NOTREACHED(); | |
| 399 return 0; | |
| 400 } | |
| 401 | |
| 402 int32_t FakeAudioCaptureModule::SetMicrophoneMute(bool /*enable*/) { | |
| 403 RTC_NOTREACHED(); | |
| 404 return 0; | |
| 405 } | |
| 406 | |
| 407 int32_t FakeAudioCaptureModule::MicrophoneMute(bool* /*enabled*/) const { | |
| 408 RTC_NOTREACHED(); | |
| 409 return 0; | |
| 410 } | |
| 411 | |
| 412 int32_t FakeAudioCaptureModule::MicrophoneBoostIsAvailable( | |
| 413 bool* /*available*/) { | |
| 414 RTC_NOTREACHED(); | |
| 415 return 0; | |
| 416 } | |
| 417 | |
| 418 int32_t FakeAudioCaptureModule::SetMicrophoneBoost(bool /*enable*/) { | |
| 419 RTC_NOTREACHED(); | |
| 420 return 0; | |
| 421 } | |
| 422 | |
| 423 int32_t FakeAudioCaptureModule::MicrophoneBoost(bool* /*enabled*/) const { | |
| 424 RTC_NOTREACHED(); | |
| 425 return 0; | |
| 426 } | |
| 427 | |
| 428 int32_t FakeAudioCaptureModule::StereoPlayoutIsAvailable( | |
| 429 bool* available) const { | |
| 430 // No recording device, just dropping audio. Stereo can be dropped just | |
| 431 // as easily as mono. | |
| 432 *available = true; | |
| 433 return 0; | |
| 434 } | |
| 435 | |
| 436 int32_t FakeAudioCaptureModule::SetStereoPlayout(bool /*enable*/) { | |
| 437 // No recording device, just dropping audio. Stereo can be dropped just | |
| 438 // as easily as mono. | |
| 439 return 0; | |
| 440 } | |
| 441 | |
| 442 int32_t FakeAudioCaptureModule::StereoPlayout(bool* /*enabled*/) const { | |
| 443 RTC_NOTREACHED(); | |
| 444 return 0; | |
| 445 } | |
| 446 | |
| 447 int32_t FakeAudioCaptureModule::StereoRecordingIsAvailable( | |
| 448 bool* available) const { | |
| 449 // Keep thing simple. No stereo recording. | |
| 450 *available = false; | |
| 451 return 0; | |
| 452 } | |
| 453 | |
| 454 int32_t FakeAudioCaptureModule::SetStereoRecording(bool enable) { | |
| 455 if (!enable) { | |
| 456 return 0; | |
| 457 } | |
| 458 return -1; | |
| 459 } | |
| 460 | |
| 461 int32_t FakeAudioCaptureModule::StereoRecording(bool* /*enabled*/) const { | |
| 462 RTC_NOTREACHED(); | |
| 463 return 0; | |
| 464 } | |
| 465 | |
| 466 int32_t FakeAudioCaptureModule::SetRecordingChannel( | |
| 467 const ChannelType channel) { | |
| 468 if (channel != AudioDeviceModule::kChannelBoth) { | |
| 469 // There is no right or left in mono. I.e. kChannelBoth should be used for | |
| 470 // mono. | |
| 471 RTC_NOTREACHED(); | |
| 472 return -1; | |
| 473 } | |
| 474 return 0; | |
| 475 } | |
| 476 | |
| 477 int32_t FakeAudioCaptureModule::RecordingChannel(ChannelType* channel) const { | |
| 478 // Stereo recording not supported. However, WebRTC ADM returns kChannelBoth | |
| 479 // in that case. Do the same here. | |
| 480 *channel = AudioDeviceModule::kChannelBoth; | |
| 481 return 0; | |
| 482 } | |
| 483 | |
| 484 int32_t FakeAudioCaptureModule::SetPlayoutBuffer(const BufferType /*type*/, | |
| 485 uint16_t /*size_ms*/) { | |
| 486 RTC_NOTREACHED(); | |
| 487 return 0; | |
| 488 } | |
| 489 | |
| 490 int32_t FakeAudioCaptureModule::PlayoutBuffer(BufferType* /*type*/, | |
| 491 uint16_t* /*size_ms*/) const { | |
| 492 RTC_NOTREACHED(); | |
| 493 return 0; | |
| 494 } | |
| 495 | |
| 496 int32_t FakeAudioCaptureModule::PlayoutDelay(uint16_t* delay_ms) const { | |
| 497 // No delay since audio frames are dropped. | |
| 498 *delay_ms = 0; | |
| 499 return 0; | |
| 500 } | |
| 501 | |
| 502 int32_t FakeAudioCaptureModule::RecordingDelay(uint16_t* /*delay_ms*/) const { | |
| 503 RTC_NOTREACHED(); | |
| 504 return 0; | |
| 505 } | |
| 506 | |
| 507 int32_t FakeAudioCaptureModule::CPULoad(uint16_t* /*load*/) const { | |
| 508 RTC_NOTREACHED(); | |
| 509 return 0; | |
| 510 } | |
| 511 | |
| 512 int32_t FakeAudioCaptureModule::StartRawOutputFileRecording( | |
| 513 const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) { | |
| 514 RTC_NOTREACHED(); | |
| 515 return 0; | |
| 516 } | |
| 517 | |
| 518 int32_t FakeAudioCaptureModule::StopRawOutputFileRecording() { | |
| 519 RTC_NOTREACHED(); | |
| 520 return 0; | |
| 521 } | |
| 522 | |
| 523 int32_t FakeAudioCaptureModule::StartRawInputFileRecording( | |
| 524 const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) { | |
| 525 RTC_NOTREACHED(); | |
| 526 return 0; | |
| 527 } | |
| 528 | |
| 529 int32_t FakeAudioCaptureModule::StopRawInputFileRecording() { | |
| 530 RTC_NOTREACHED(); | |
| 531 return 0; | |
| 532 } | |
| 533 | |
| 534 int32_t FakeAudioCaptureModule::SetRecordingSampleRate( | |
| 535 const uint32_t /*samples_per_sec*/) { | |
| 536 RTC_NOTREACHED(); | |
| 537 return 0; | |
| 538 } | |
| 539 | |
| 540 int32_t FakeAudioCaptureModule::RecordingSampleRate( | |
| 541 uint32_t* /*samples_per_sec*/) const { | |
| 542 RTC_NOTREACHED(); | |
| 543 return 0; | |
| 544 } | |
| 545 | |
| 546 int32_t FakeAudioCaptureModule::SetPlayoutSampleRate( | |
| 547 const uint32_t /*samples_per_sec*/) { | |
| 548 RTC_NOTREACHED(); | |
| 549 return 0; | |
| 550 } | |
| 551 | |
| 552 int32_t FakeAudioCaptureModule::PlayoutSampleRate( | |
| 553 uint32_t* /*samples_per_sec*/) const { | |
| 554 RTC_NOTREACHED(); | |
| 555 return 0; | |
| 556 } | |
| 557 | |
| 558 int32_t FakeAudioCaptureModule::ResetAudioDevice() { | |
| 559 RTC_NOTREACHED(); | |
| 560 return 0; | |
| 561 } | |
| 562 | |
| 563 int32_t FakeAudioCaptureModule::SetLoudspeakerStatus(bool /*enable*/) { | |
| 564 RTC_NOTREACHED(); | |
| 565 return 0; | |
| 566 } | |
| 567 | |
| 568 int32_t FakeAudioCaptureModule::GetLoudspeakerStatus(bool* /*enabled*/) const { | |
| 569 RTC_NOTREACHED(); | |
| 570 return 0; | |
| 571 } | |
| 572 | |
| 573 void FakeAudioCaptureModule::OnMessage(rtc::Message* msg) { | |
| 574 switch (msg->message_id) { | |
| 575 case MSG_START_PROCESS: | |
| 576 StartProcessP(); | |
| 577 break; | |
| 578 case MSG_RUN_PROCESS: | |
| 579 ProcessFrameP(); | |
| 580 break; | |
| 581 default: | |
| 582 // All existing messages should be caught. Getting here should never | |
| 583 // happen. | |
| 584 RTC_NOTREACHED(); | |
| 585 } | |
| 586 } | |
| 587 | |
| 588 bool FakeAudioCaptureModule::Initialize() { | |
| 589 // Set the send buffer samples high enough that it would not occur on the | |
| 590 // remote side unless a packet containing a sample of that magnitude has been | |
| 591 // sent to it. Note that the audio processing pipeline will likely distort the | |
| 592 // original signal. | |
| 593 SetSendBuffer(kHighSampleValue); | |
| 594 last_process_time_ms_ = rtc::TimeMillis(); | |
| 595 return true; | |
| 596 } | |
| 597 | |
| 598 void FakeAudioCaptureModule::SetSendBuffer(int value) { | |
| 599 Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_); | |
| 600 const size_t buffer_size_in_samples = | |
| 601 sizeof(send_buffer_) / kNumberBytesPerSample; | |
| 602 for (size_t i = 0; i < buffer_size_in_samples; ++i) { | |
| 603 buffer_ptr[i] = value; | |
| 604 } | |
| 605 } | |
| 606 | |
| 607 void FakeAudioCaptureModule::ResetRecBuffer() { | |
| 608 memset(rec_buffer_, 0, sizeof(rec_buffer_)); | |
| 609 } | |
| 610 | |
| 611 bool FakeAudioCaptureModule::CheckRecBuffer(int value) { | |
| 612 const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_); | |
| 613 const size_t buffer_size_in_samples = | |
| 614 sizeof(rec_buffer_) / kNumberBytesPerSample; | |
| 615 for (size_t i = 0; i < buffer_size_in_samples; ++i) { | |
| 616 if (buffer_ptr[i] >= value) return true; | |
| 617 } | |
| 618 return false; | |
| 619 } | |
| 620 | |
| 621 bool FakeAudioCaptureModule::ShouldStartProcessing() { | |
| 622 return recording_ || playing_; | |
| 623 } | |
| 624 | |
| 625 void FakeAudioCaptureModule::UpdateProcessing(bool start) { | |
| 626 if (start) { | |
| 627 if (!process_thread_) { | |
| 628 process_thread_.reset(new rtc::Thread()); | |
| 629 process_thread_->Start(); | |
| 630 } | |
| 631 process_thread_->Post(RTC_FROM_HERE, this, MSG_START_PROCESS); | |
| 632 } else { | |
| 633 if (process_thread_) { | |
| 634 process_thread_->Stop(); | |
| 635 process_thread_.reset(nullptr); | |
| 636 } | |
| 637 started_ = false; | |
| 638 } | |
| 639 } | |
| 640 | |
| 641 void FakeAudioCaptureModule::StartProcessP() { | |
| 642 RTC_CHECK(process_thread_->IsCurrent()); | |
| 643 if (started_) { | |
| 644 // Already started. | |
| 645 return; | |
| 646 } | |
| 647 ProcessFrameP(); | |
| 648 } | |
| 649 | |
| 650 void FakeAudioCaptureModule::ProcessFrameP() { | |
| 651 RTC_CHECK(process_thread_->IsCurrent()); | |
| 652 if (!started_) { | |
| 653 next_frame_time_ = rtc::TimeMillis(); | |
| 654 started_ = true; | |
| 655 } | |
| 656 | |
| 657 { | |
| 658 rtc::CritScope cs(&crit_); | |
| 659 // Receive and send frames every kTimePerFrameMs. | |
| 660 if (playing_) { | |
| 661 ReceiveFrameP(); | |
| 662 } | |
| 663 if (recording_) { | |
| 664 SendFrameP(); | |
| 665 } | |
| 666 } | |
| 667 | |
| 668 next_frame_time_ += kTimePerFrameMs; | |
| 669 const int64_t current_time = rtc::TimeMillis(); | |
| 670 const int64_t wait_time = | |
| 671 (next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0; | |
| 672 process_thread_->PostDelayed(RTC_FROM_HERE, wait_time, this, MSG_RUN_PROCESS); | |
| 673 } | |
| 674 | |
| 675 void FakeAudioCaptureModule::ReceiveFrameP() { | |
| 676 RTC_CHECK(process_thread_->IsCurrent()); | |
| 677 { | |
| 678 rtc::CritScope cs(&crit_callback_); | |
| 679 if (!audio_callback_) { | |
| 680 return; | |
| 681 } | |
| 682 ResetRecBuffer(); | |
| 683 size_t nSamplesOut = 0; | |
| 684 int64_t elapsed_time_ms = 0; | |
| 685 int64_t ntp_time_ms = 0; | |
| 686 if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample, | |
| 687 kNumberOfChannels, kSamplesPerSecond, | |
| 688 rec_buffer_, nSamplesOut, | |
| 689 &elapsed_time_ms, &ntp_time_ms) != 0) { | |
| 690 RTC_NOTREACHED(); | |
| 691 } | |
| 692 RTC_CHECK(nSamplesOut == kNumberSamples); | |
| 693 } | |
| 694 // The SetBuffer() function ensures that after decoding, the audio buffer | |
| 695 // should contain samples of similar magnitude (there is likely to be some | |
| 696 // distortion due to the audio pipeline). If one sample is detected to | |
| 697 // have the same or greater magnitude somewhere in the frame, an actual frame | |
| 698 // has been received from the remote side (i.e. faked frames are not being | |
| 699 // pulled). | |
| 700 if (CheckRecBuffer(kHighSampleValue)) { | |
| 701 rtc::CritScope cs(&crit_); | |
| 702 ++frames_received_; | |
| 703 } | |
| 704 } | |
| 705 | |
| 706 void FakeAudioCaptureModule::SendFrameP() { | |
| 707 RTC_CHECK(process_thread_->IsCurrent()); | |
| 708 rtc::CritScope cs(&crit_callback_); | |
| 709 if (!audio_callback_) { | |
| 710 return; | |
| 711 } | |
| 712 bool key_pressed = false; | |
| 713 uint32_t current_mic_level = 0; | |
| 714 MicrophoneVolume(¤t_mic_level); | |
| 715 if (audio_callback_->RecordedDataIsAvailable(send_buffer_, kNumberSamples, | |
| 716 kNumberBytesPerSample, | |
| 717 kNumberOfChannels, | |
| 718 kSamplesPerSecond, kTotalDelayMs, | |
| 719 kClockDriftMs, current_mic_level, | |
| 720 key_pressed, | |
| 721 current_mic_level) != 0) { | |
| 722 RTC_NOTREACHED(); | |
| 723 } | |
| 724 SetMicrophoneVolume(current_mic_level); | |
| 725 } | |
| OLD | NEW |