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