andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 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/voice_engine/transmit_mixer.h" |
| 12 | |
| 13 | #include "webrtc/modules/utility/interface/audio_frame_operations.h" |
| 14 | #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| 15 | #include "webrtc/system_wrappers/interface/event_wrapper.h" |
| 16 | #include "webrtc/system_wrappers/interface/logging.h" |
| 17 | #include "webrtc/system_wrappers/interface/trace.h" |
| 18 | #include "webrtc/voice_engine/channel.h" |
| 19 | #include "webrtc/voice_engine/channel_manager.h" |
| 20 | #include "webrtc/voice_engine/include/voe_external_media.h" |
| 21 | #include "webrtc/voice_engine/statistics.h" |
| 22 | #include "webrtc/voice_engine/utility.h" |
| 23 | #include "webrtc/voice_engine/voe_base_impl.h" |
| 24 | |
| 25 | #define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a)) |
| 26 | |
| 27 | namespace webrtc { |
| 28 | namespace voe { |
| 29 | |
| 30 | // TODO(ajm): The thread safety of this is dubious... |
| 31 | void |
| 32 | TransmitMixer::OnPeriodicProcess() |
| 33 | { |
| 34 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| 35 | "TransmitMixer::OnPeriodicProcess()"); |
| 36 | |
| 37 | #if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION) |
| 38 | if (_typingNoiseWarningPending) |
| 39 | { |
| 40 | CriticalSectionScoped cs(&_callbackCritSect); |
| 41 | if (_voiceEngineObserverPtr) |
| 42 | { |
| 43 | if (_typingNoiseDetected) { |
| 44 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 45 | "TransmitMixer::OnPeriodicProcess() => " |
| 46 | "CallbackOnError(VE_TYPING_NOISE_WARNING)"); |
| 47 | _voiceEngineObserverPtr->CallbackOnError( |
| 48 | -1, |
| 49 | VE_TYPING_NOISE_WARNING); |
| 50 | } else { |
| 51 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 52 | "TransmitMixer::OnPeriodicProcess() => " |
| 53 | "CallbackOnError(VE_TYPING_NOISE_OFF_WARNING)"); |
| 54 | _voiceEngineObserverPtr->CallbackOnError( |
| 55 | -1, |
| 56 | VE_TYPING_NOISE_OFF_WARNING); |
| 57 | } |
| 58 | } |
| 59 | _typingNoiseWarningPending = false; |
| 60 | } |
| 61 | #endif |
| 62 | |
| 63 | bool saturationWarning = false; |
| 64 | { |
| 65 | // Modify |_saturationWarning| under lock to avoid conflict with write op |
| 66 | // in ProcessAudio and also ensure that we don't hold the lock during the |
| 67 | // callback. |
| 68 | CriticalSectionScoped cs(&_critSect); |
| 69 | saturationWarning = _saturationWarning; |
| 70 | if (_saturationWarning) |
| 71 | _saturationWarning = false; |
| 72 | } |
| 73 | |
| 74 | if (saturationWarning) |
| 75 | { |
| 76 | CriticalSectionScoped cs(&_callbackCritSect); |
| 77 | if (_voiceEngineObserverPtr) |
| 78 | { |
| 79 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 80 | "TransmitMixer::OnPeriodicProcess() =>" |
| 81 | " CallbackOnError(VE_SATURATION_WARNING)"); |
| 82 | _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING); |
| 83 | } |
| 84 | } |
| 85 | } |
| 86 | |
| 87 | |
| 88 | void TransmitMixer::PlayNotification(int32_t id, |
| 89 | uint32_t durationMs) |
| 90 | { |
| 91 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| 92 | "TransmitMixer::PlayNotification(id=%d, durationMs=%d)", |
| 93 | id, durationMs); |
| 94 | |
| 95 | // Not implement yet |
| 96 | } |
| 97 | |
| 98 | void TransmitMixer::RecordNotification(int32_t id, |
| 99 | uint32_t durationMs) |
| 100 | { |
| 101 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| 102 | "TransmitMixer::RecordNotification(id=%d, durationMs=%d)", |
| 103 | id, durationMs); |
| 104 | |
| 105 | // Not implement yet |
| 106 | } |
| 107 | |
| 108 | void TransmitMixer::PlayFileEnded(int32_t id) |
| 109 | { |
| 110 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| 111 | "TransmitMixer::PlayFileEnded(id=%d)", id); |
| 112 | |
| 113 | assert(id == _filePlayerId); |
| 114 | |
| 115 | CriticalSectionScoped cs(&_critSect); |
| 116 | |
| 117 | _filePlaying = false; |
| 118 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 119 | "TransmitMixer::PlayFileEnded() =>" |
| 120 | "file player module is shutdown"); |
| 121 | } |
| 122 | |
| 123 | void |
| 124 | TransmitMixer::RecordFileEnded(int32_t id) |
| 125 | { |
| 126 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| 127 | "TransmitMixer::RecordFileEnded(id=%d)", id); |
| 128 | |
| 129 | if (id == _fileRecorderId) |
| 130 | { |
| 131 | CriticalSectionScoped cs(&_critSect); |
| 132 | _fileRecording = false; |
| 133 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 134 | "TransmitMixer::RecordFileEnded() => fileRecorder module" |
| 135 | "is shutdown"); |
| 136 | } else if (id == _fileCallRecorderId) |
| 137 | { |
| 138 | CriticalSectionScoped cs(&_critSect); |
| 139 | _fileCallRecording = false; |
| 140 | WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 141 | "TransmitMixer::RecordFileEnded() => fileCallRecorder" |
| 142 | "module is shutdown"); |
| 143 | } |
| 144 | } |
| 145 | |
| 146 | int32_t |
| 147 | TransmitMixer::Create(TransmitMixer*& mixer, uint32_t instanceId) |
| 148 | { |
| 149 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1), |
| 150 | "TransmitMixer::Create(instanceId=%d)", instanceId); |
| 151 | mixer = new TransmitMixer(instanceId); |
| 152 | if (mixer == NULL) |
| 153 | { |
| 154 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1), |
| 155 | "TransmitMixer::Create() unable to allocate memory" |
| 156 | "for mixer"); |
| 157 | return -1; |
| 158 | } |
| 159 | return 0; |
| 160 | } |
| 161 | |
| 162 | void |
| 163 | TransmitMixer::Destroy(TransmitMixer*& mixer) |
| 164 | { |
| 165 | if (mixer) |
| 166 | { |
| 167 | delete mixer; |
| 168 | mixer = NULL; |
| 169 | } |
| 170 | } |
| 171 | |
| 172 | TransmitMixer::TransmitMixer(uint32_t instanceId) : |
| 173 | _engineStatisticsPtr(NULL), |
| 174 | _channelManagerPtr(NULL), |
| 175 | audioproc_(NULL), |
| 176 | _voiceEngineObserverPtr(NULL), |
| 177 | _processThreadPtr(NULL), |
| 178 | _filePlayerPtr(NULL), |
| 179 | _fileRecorderPtr(NULL), |
| 180 | _fileCallRecorderPtr(NULL), |
| 181 | // Avoid conflict with other channels by adding 1024 - 1026, |
| 182 | // won't use as much as 1024 channels. |
| 183 | _filePlayerId(instanceId + 1024), |
| 184 | _fileRecorderId(instanceId + 1025), |
| 185 | _fileCallRecorderId(instanceId + 1026), |
| 186 | _filePlaying(false), |
| 187 | _fileRecording(false), |
| 188 | _fileCallRecording(false), |
| 189 | _audioLevel(), |
| 190 | _critSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| 191 | _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| 192 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 193 | _typingNoiseWarningPending(false), |
| 194 | _typingNoiseDetected(false), |
| 195 | #endif |
| 196 | _saturationWarning(false), |
| 197 | _instanceId(instanceId), |
| 198 | _mixFileWithMicrophone(false), |
| 199 | _captureLevel(0), |
| 200 | external_postproc_ptr_(NULL), |
| 201 | external_preproc_ptr_(NULL), |
| 202 | _mute(false), |
| 203 | _remainingMuteMicTimeMs(0), |
| 204 | stereo_codec_(false), |
| 205 | swap_stereo_channels_(false) |
| 206 | { |
| 207 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), |
| 208 | "TransmitMixer::TransmitMixer() - ctor"); |
| 209 | } |
| 210 | |
| 211 | TransmitMixer::~TransmitMixer() |
| 212 | { |
| 213 | WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), |
| 214 | "TransmitMixer::~TransmitMixer() - dtor"); |
| 215 | _monitorModule.DeRegisterObserver(); |
| 216 | if (_processThreadPtr) |
| 217 | { |
| 218 | _processThreadPtr->DeRegisterModule(&_monitorModule); |
| 219 | } |
| 220 | DeRegisterExternalMediaProcessing(kRecordingAllChannelsMixed); |
| 221 | DeRegisterExternalMediaProcessing(kRecordingPreprocessing); |
| 222 | { |
| 223 | CriticalSectionScoped cs(&_critSect); |
| 224 | if (_fileRecorderPtr) |
| 225 | { |
| 226 | _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 227 | _fileRecorderPtr->StopRecording(); |
| 228 | FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| 229 | _fileRecorderPtr = NULL; |
| 230 | } |
| 231 | if (_fileCallRecorderPtr) |
| 232 | { |
| 233 | _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| 234 | _fileCallRecorderPtr->StopRecording(); |
| 235 | FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| 236 | _fileCallRecorderPtr = NULL; |
| 237 | } |
| 238 | if (_filePlayerPtr) |
| 239 | { |
| 240 | _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| 241 | _filePlayerPtr->StopPlayingFile(); |
| 242 | FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| 243 | _filePlayerPtr = NULL; |
| 244 | } |
| 245 | } |
| 246 | delete &_critSect; |
| 247 | delete &_callbackCritSect; |
| 248 | } |
| 249 | |
| 250 | int32_t |
| 251 | TransmitMixer::SetEngineInformation(ProcessThread& processThread, |
| 252 | Statistics& engineStatistics, |
| 253 | ChannelManager& channelManager) |
| 254 | { |
| 255 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 256 | "TransmitMixer::SetEngineInformation()"); |
| 257 | |
| 258 | _processThreadPtr = &processThread; |
| 259 | _engineStatisticsPtr = &engineStatistics; |
| 260 | _channelManagerPtr = &channelManager; |
| 261 | |
| 262 | if (_processThreadPtr->RegisterModule(&_monitorModule) == -1) |
| 263 | { |
| 264 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 265 | "TransmitMixer::SetEngineInformation() failed to" |
| 266 | "register the monitor module"); |
| 267 | } else |
| 268 | { |
| 269 | _monitorModule.RegisterObserver(*this); |
| 270 | } |
| 271 | |
| 272 | return 0; |
| 273 | } |
| 274 | |
| 275 | int32_t |
| 276 | TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) |
| 277 | { |
| 278 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 279 | "TransmitMixer::RegisterVoiceEngineObserver()"); |
| 280 | CriticalSectionScoped cs(&_callbackCritSect); |
| 281 | |
| 282 | if (_voiceEngineObserverPtr) |
| 283 | { |
| 284 | _engineStatisticsPtr->SetLastError( |
| 285 | VE_INVALID_OPERATION, kTraceError, |
| 286 | "RegisterVoiceEngineObserver() observer already enabled"); |
| 287 | return -1; |
| 288 | } |
| 289 | _voiceEngineObserverPtr = &observer; |
| 290 | return 0; |
| 291 | } |
| 292 | |
| 293 | int32_t |
| 294 | TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule) |
| 295 | { |
| 296 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 297 | "TransmitMixer::SetAudioProcessingModule(" |
| 298 | "audioProcessingModule=0x%x)", |
| 299 | audioProcessingModule); |
| 300 | audioproc_ = audioProcessingModule; |
| 301 | return 0; |
| 302 | } |
| 303 | |
| 304 | void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) { |
| 305 | *max_sample_rate = 8000; |
| 306 | *max_channels = 1; |
| 307 | for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid(); |
| 308 | it.Increment()) { |
| 309 | Channel* channel = it.GetChannel(); |
| 310 | if (channel->Sending()) { |
| 311 | CodecInst codec; |
| 312 | channel->GetSendCodec(codec); |
| 313 | *max_sample_rate = std::max(*max_sample_rate, codec.plfreq); |
| 314 | *max_channels = std::max(*max_channels, codec.channels); |
| 315 | } |
| 316 | } |
| 317 | } |
| 318 | |
| 319 | int32_t |
| 320 | TransmitMixer::PrepareDemux(const void* audioSamples, |
| 321 | uint32_t nSamples, |
| 322 | uint8_t nChannels, |
| 323 | uint32_t samplesPerSec, |
| 324 | uint16_t totalDelayMS, |
| 325 | int32_t clockDrift, |
| 326 | uint16_t currentMicLevel, |
| 327 | bool keyPressed) |
| 328 | { |
| 329 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| 330 | "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u," |
| 331 | "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%d," |
| 332 | "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec, |
| 333 | totalDelayMS, clockDrift, currentMicLevel); |
| 334 | |
| 335 | // --- Resample input audio and create/store the initial audio frame |
| 336 | GenerateAudioFrame(static_cast<const int16_t*>(audioSamples), |
| 337 | nSamples, |
| 338 | nChannels, |
| 339 | samplesPerSec); |
| 340 | |
| 341 | { |
| 342 | CriticalSectionScoped cs(&_callbackCritSect); |
| 343 | if (external_preproc_ptr_) { |
| 344 | external_preproc_ptr_->Process(-1, kRecordingPreprocessing, |
| 345 | _audioFrame.data_, |
| 346 | _audioFrame.samples_per_channel_, |
| 347 | _audioFrame.sample_rate_hz_, |
| 348 | _audioFrame.num_channels_ == 2); |
| 349 | } |
| 350 | } |
| 351 | |
| 352 | // --- Near-end audio processing. |
| 353 | ProcessAudio(totalDelayMS, clockDrift, currentMicLevel, keyPressed); |
| 354 | |
| 355 | if (swap_stereo_channels_ && stereo_codec_) |
| 356 | // Only bother swapping if we're using a stereo codec. |
| 357 | AudioFrameOperations::SwapStereoChannels(&_audioFrame); |
| 358 | |
| 359 | // --- Annoying typing detection (utilizes the APM/VAD decision) |
| 360 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 361 | TypingDetection(keyPressed); |
| 362 | #endif |
| 363 | |
| 364 | // --- Mute during DTMF tone if direct feedback is enabled |
| 365 | if (_remainingMuteMicTimeMs > 0) |
| 366 | { |
| 367 | AudioFrameOperations::Mute(_audioFrame); |
| 368 | _remainingMuteMicTimeMs -= 10; |
| 369 | if (_remainingMuteMicTimeMs < 0) |
| 370 | { |
| 371 | _remainingMuteMicTimeMs = 0; |
| 372 | } |
| 373 | } |
| 374 | |
| 375 | // --- Mute signal |
| 376 | if (_mute) |
| 377 | { |
| 378 | AudioFrameOperations::Mute(_audioFrame); |
| 379 | } |
| 380 | |
| 381 | // --- Mix with file (does not affect the mixing frequency) |
| 382 | if (_filePlaying) |
| 383 | { |
| 384 | MixOrReplaceAudioWithFile(_audioFrame.sample_rate_hz_); |
| 385 | } |
| 386 | |
| 387 | // --- Record to file |
| 388 | bool file_recording = false; |
| 389 | { |
| 390 | CriticalSectionScoped cs(&_critSect); |
| 391 | file_recording = _fileRecording; |
| 392 | } |
| 393 | if (file_recording) |
| 394 | { |
| 395 | RecordAudioToFile(_audioFrame.sample_rate_hz_); |
| 396 | } |
| 397 | |
| 398 | { |
| 399 | CriticalSectionScoped cs(&_callbackCritSect); |
| 400 | if (external_postproc_ptr_) { |
| 401 | external_postproc_ptr_->Process(-1, kRecordingAllChannelsMixed, |
| 402 | _audioFrame.data_, |
| 403 | _audioFrame.samples_per_channel_, |
| 404 | _audioFrame.sample_rate_hz_, |
| 405 | _audioFrame.num_channels_ == 2); |
| 406 | } |
| 407 | } |
| 408 | |
| 409 | // --- Measure audio level of speech after all processing. |
| 410 | _audioLevel.ComputeLevel(_audioFrame); |
| 411 | return 0; |
| 412 | } |
| 413 | |
| 414 | int32_t |
| 415 | TransmitMixer::DemuxAndMix() |
| 416 | { |
| 417 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| 418 | "TransmitMixer::DemuxAndMix()"); |
| 419 | |
| 420 | for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid(); |
| 421 | it.Increment()) |
| 422 | { |
| 423 | Channel* channelPtr = it.GetChannel(); |
henrika@webrtc.org | 692224a | 2014-04-17 10:45:01 +0000 | [diff] [blame] | 424 | if (channelPtr->Sending()) |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 425 | { |
| 426 | // Demultiplex makes a copy of its input. |
| 427 | channelPtr->Demultiplex(_audioFrame); |
| 428 | channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_); |
| 429 | } |
| 430 | } |
| 431 | return 0; |
| 432 | } |
| 433 | |
| 434 | void TransmitMixer::DemuxAndMix(const int voe_channels[], |
| 435 | int number_of_voe_channels) { |
| 436 | for (int i = 0; i < number_of_voe_channels; ++i) { |
| 437 | voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]); |
| 438 | voe::Channel* channel_ptr = ch.channel(); |
| 439 | if (channel_ptr) { |
henrika@webrtc.org | 692224a | 2014-04-17 10:45:01 +0000 | [diff] [blame] | 440 | if (channel_ptr->Sending()) { |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 441 | // Demultiplex makes a copy of its input. |
| 442 | channel_ptr->Demultiplex(_audioFrame); |
| 443 | channel_ptr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_); |
| 444 | } |
| 445 | } |
| 446 | } |
| 447 | } |
| 448 | |
| 449 | int32_t |
| 450 | TransmitMixer::EncodeAndSend() |
| 451 | { |
| 452 | WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| 453 | "TransmitMixer::EncodeAndSend()"); |
| 454 | |
| 455 | for (ChannelManager::Iterator it(_channelManagerPtr); it.IsValid(); |
| 456 | it.Increment()) |
| 457 | { |
| 458 | Channel* channelPtr = it.GetChannel(); |
henrika@webrtc.org | 692224a | 2014-04-17 10:45:01 +0000 | [diff] [blame] | 459 | if (channelPtr->Sending()) |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 460 | { |
| 461 | channelPtr->EncodeAndSend(); |
| 462 | } |
| 463 | } |
| 464 | return 0; |
| 465 | } |
| 466 | |
| 467 | void TransmitMixer::EncodeAndSend(const int voe_channels[], |
| 468 | int number_of_voe_channels) { |
| 469 | for (int i = 0; i < number_of_voe_channels; ++i) { |
| 470 | voe::ChannelOwner ch = _channelManagerPtr->GetChannel(voe_channels[i]); |
| 471 | voe::Channel* channel_ptr = ch.channel(); |
henrika@webrtc.org | 692224a | 2014-04-17 10:45:01 +0000 | [diff] [blame] | 472 | if (channel_ptr && channel_ptr->Sending()) |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 473 | channel_ptr->EncodeAndSend(); |
| 474 | } |
| 475 | } |
| 476 | |
| 477 | uint32_t TransmitMixer::CaptureLevel() const |
| 478 | { |
| 479 | return _captureLevel; |
| 480 | } |
| 481 | |
| 482 | void |
| 483 | TransmitMixer::UpdateMuteMicrophoneTime(uint32_t lengthMs) |
| 484 | { |
| 485 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 486 | "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)", |
| 487 | lengthMs); |
| 488 | _remainingMuteMicTimeMs = lengthMs; |
| 489 | } |
| 490 | |
| 491 | int32_t |
| 492 | TransmitMixer::StopSend() |
| 493 | { |
| 494 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 495 | "TransmitMixer::StopSend()"); |
| 496 | _audioLevel.Clear(); |
| 497 | return 0; |
| 498 | } |
| 499 | |
| 500 | int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName, |
| 501 | bool loop, |
| 502 | FileFormats format, |
| 503 | int startPosition, |
| 504 | float volumeScaling, |
| 505 | int stopPosition, |
| 506 | const CodecInst* codecInst) |
| 507 | { |
| 508 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 509 | "TransmitMixer::StartPlayingFileAsMicrophone(" |
| 510 | "fileNameUTF8[]=%s,loop=%d, format=%d, volumeScaling=%5.3f," |
| 511 | " startPosition=%d, stopPosition=%d)", fileName, loop, |
| 512 | format, volumeScaling, startPosition, stopPosition); |
| 513 | |
| 514 | if (_filePlaying) |
| 515 | { |
| 516 | _engineStatisticsPtr->SetLastError( |
| 517 | VE_ALREADY_PLAYING, kTraceWarning, |
| 518 | "StartPlayingFileAsMicrophone() is already playing"); |
| 519 | return 0; |
| 520 | } |
| 521 | |
| 522 | CriticalSectionScoped cs(&_critSect); |
| 523 | |
| 524 | // Destroy the old instance |
| 525 | if (_filePlayerPtr) |
| 526 | { |
| 527 | _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| 528 | FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| 529 | _filePlayerPtr = NULL; |
| 530 | } |
| 531 | |
| 532 | // Dynamically create the instance |
| 533 | _filePlayerPtr |
| 534 | = FilePlayer::CreateFilePlayer(_filePlayerId, |
| 535 | (const FileFormats) format); |
| 536 | |
| 537 | if (_filePlayerPtr == NULL) |
| 538 | { |
| 539 | _engineStatisticsPtr->SetLastError( |
| 540 | VE_INVALID_ARGUMENT, kTraceError, |
| 541 | "StartPlayingFileAsMicrophone() filePlayer format isnot correct"); |
| 542 | return -1; |
| 543 | } |
| 544 | |
| 545 | const uint32_t notificationTime(0); |
| 546 | |
| 547 | if (_filePlayerPtr->StartPlayingFile( |
| 548 | fileName, |
| 549 | loop, |
| 550 | startPosition, |
| 551 | volumeScaling, |
| 552 | notificationTime, |
| 553 | stopPosition, |
| 554 | (const CodecInst*) codecInst) != 0) |
| 555 | { |
| 556 | _engineStatisticsPtr->SetLastError( |
| 557 | VE_BAD_FILE, kTraceError, |
| 558 | "StartPlayingFile() failed to start file playout"); |
| 559 | _filePlayerPtr->StopPlayingFile(); |
| 560 | FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| 561 | _filePlayerPtr = NULL; |
| 562 | return -1; |
| 563 | } |
| 564 | |
| 565 | _filePlayerPtr->RegisterModuleFileCallback(this); |
| 566 | _filePlaying = true; |
| 567 | |
| 568 | return 0; |
| 569 | } |
| 570 | |
| 571 | int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream, |
| 572 | FileFormats format, |
| 573 | int startPosition, |
| 574 | float volumeScaling, |
| 575 | int stopPosition, |
| 576 | const CodecInst* codecInst) |
| 577 | { |
| 578 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 579 | "TransmitMixer::StartPlayingFileAsMicrophone(format=%d," |
| 580 | " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
| 581 | format, volumeScaling, startPosition, stopPosition); |
| 582 | |
| 583 | if (stream == NULL) |
| 584 | { |
| 585 | _engineStatisticsPtr->SetLastError( |
| 586 | VE_BAD_FILE, kTraceError, |
| 587 | "StartPlayingFileAsMicrophone() NULL as input stream"); |
| 588 | return -1; |
| 589 | } |
| 590 | |
| 591 | if (_filePlaying) |
| 592 | { |
| 593 | _engineStatisticsPtr->SetLastError( |
| 594 | VE_ALREADY_PLAYING, kTraceWarning, |
| 595 | "StartPlayingFileAsMicrophone() is already playing"); |
| 596 | return 0; |
| 597 | } |
| 598 | |
| 599 | CriticalSectionScoped cs(&_critSect); |
| 600 | |
| 601 | // Destroy the old instance |
| 602 | if (_filePlayerPtr) |
| 603 | { |
| 604 | _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| 605 | FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| 606 | _filePlayerPtr = NULL; |
| 607 | } |
| 608 | |
| 609 | // Dynamically create the instance |
| 610 | _filePlayerPtr |
| 611 | = FilePlayer::CreateFilePlayer(_filePlayerId, |
| 612 | (const FileFormats) format); |
| 613 | |
| 614 | if (_filePlayerPtr == NULL) |
| 615 | { |
| 616 | _engineStatisticsPtr->SetLastError( |
| 617 | VE_INVALID_ARGUMENT, kTraceWarning, |
| 618 | "StartPlayingFileAsMicrophone() filePlayer format isnot correct"); |
| 619 | return -1; |
| 620 | } |
| 621 | |
| 622 | const uint32_t notificationTime(0); |
| 623 | |
| 624 | if (_filePlayerPtr->StartPlayingFile( |
| 625 | (InStream&) *stream, |
| 626 | startPosition, |
| 627 | volumeScaling, |
| 628 | notificationTime, |
| 629 | stopPosition, |
| 630 | (const CodecInst*) codecInst) != 0) |
| 631 | { |
| 632 | _engineStatisticsPtr->SetLastError( |
| 633 | VE_BAD_FILE, kTraceError, |
| 634 | "StartPlayingFile() failed to start file playout"); |
| 635 | _filePlayerPtr->StopPlayingFile(); |
| 636 | FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| 637 | _filePlayerPtr = NULL; |
| 638 | return -1; |
| 639 | } |
| 640 | _filePlayerPtr->RegisterModuleFileCallback(this); |
| 641 | _filePlaying = true; |
| 642 | |
| 643 | return 0; |
| 644 | } |
| 645 | |
| 646 | int TransmitMixer::StopPlayingFileAsMicrophone() |
| 647 | { |
| 648 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| 649 | "TransmitMixer::StopPlayingFileAsMicrophone()"); |
| 650 | |
| 651 | if (!_filePlaying) |
| 652 | { |
| 653 | _engineStatisticsPtr->SetLastError( |
| 654 | VE_INVALID_OPERATION, kTraceWarning, |
| 655 | "StopPlayingFileAsMicrophone() isnot playing"); |
| 656 | return 0; |
| 657 | } |
| 658 | |
| 659 | CriticalSectionScoped cs(&_critSect); |
| 660 | |
| 661 | if (_filePlayerPtr->StopPlayingFile() != 0) |
| 662 | { |
| 663 | _engineStatisticsPtr->SetLastError( |
| 664 | VE_CANNOT_STOP_PLAYOUT, kTraceError, |
| 665 | "StopPlayingFile() couldnot stop playing file"); |
| 666 | return -1; |
| 667 | } |
| 668 | |
| 669 | _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| 670 | FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| 671 | _filePlayerPtr = NULL; |
| 672 | _filePlaying = false; |
| 673 | |
| 674 | return 0; |
| 675 | } |
| 676 | |
| 677 | int TransmitMixer::IsPlayingFileAsMicrophone() const |
| 678 | { |
| 679 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 680 | "TransmitMixer::IsPlayingFileAsMicrophone()"); |
| 681 | return _filePlaying; |
| 682 | } |
| 683 | |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 684 | int TransmitMixer::StartRecordingMicrophone(const char* fileName, |
| 685 | const CodecInst* codecInst) |
| 686 | { |
| 687 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 688 | "TransmitMixer::StartRecordingMicrophone(fileName=%s)", |
| 689 | fileName); |
| 690 | |
| 691 | CriticalSectionScoped cs(&_critSect); |
| 692 | |
| 693 | if (_fileRecording) |
| 694 | { |
| 695 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 696 | "StartRecordingMicrophone() is already recording"); |
| 697 | return 0; |
| 698 | } |
| 699 | |
| 700 | FileFormats format; |
| 701 | const uint32_t notificationTime(0); // Not supported in VoE |
| 702 | CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| 703 | |
| 704 | if (codecInst != NULL && |
| 705 | (codecInst->channels < 0 || codecInst->channels > 2)) |
| 706 | { |
| 707 | _engineStatisticsPtr->SetLastError( |
| 708 | VE_BAD_ARGUMENT, kTraceError, |
| 709 | "StartRecordingMicrophone() invalid compression"); |
| 710 | return (-1); |
| 711 | } |
| 712 | if (codecInst == NULL) |
| 713 | { |
| 714 | format = kFileFormatPcm16kHzFile; |
| 715 | codecInst = &dummyCodec; |
| 716 | } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 717 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 718 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 719 | { |
| 720 | format = kFileFormatWavFile; |
| 721 | } else |
| 722 | { |
| 723 | format = kFileFormatCompressedFile; |
| 724 | } |
| 725 | |
| 726 | // Destroy the old instance |
| 727 | if (_fileRecorderPtr) |
| 728 | { |
| 729 | _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 730 | FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| 731 | _fileRecorderPtr = NULL; |
| 732 | } |
| 733 | |
| 734 | _fileRecorderPtr = |
| 735 | FileRecorder::CreateFileRecorder(_fileRecorderId, |
| 736 | (const FileFormats) format); |
| 737 | if (_fileRecorderPtr == NULL) |
| 738 | { |
| 739 | _engineStatisticsPtr->SetLastError( |
| 740 | VE_INVALID_ARGUMENT, kTraceError, |
| 741 | "StartRecordingMicrophone() fileRecorder format isnot correct"); |
| 742 | return -1; |
| 743 | } |
| 744 | |
| 745 | if (_fileRecorderPtr->StartRecordingAudioFile( |
| 746 | fileName, |
| 747 | (const CodecInst&) *codecInst, |
| 748 | notificationTime) != 0) |
| 749 | { |
| 750 | _engineStatisticsPtr->SetLastError( |
| 751 | VE_BAD_FILE, kTraceError, |
| 752 | "StartRecordingAudioFile() failed to start file recording"); |
| 753 | _fileRecorderPtr->StopRecording(); |
| 754 | FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| 755 | _fileRecorderPtr = NULL; |
| 756 | return -1; |
| 757 | } |
| 758 | _fileRecorderPtr->RegisterModuleFileCallback(this); |
| 759 | _fileRecording = true; |
| 760 | |
| 761 | return 0; |
| 762 | } |
| 763 | |
| 764 | int TransmitMixer::StartRecordingMicrophone(OutStream* stream, |
| 765 | const CodecInst* codecInst) |
| 766 | { |
| 767 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 768 | "TransmitMixer::StartRecordingMicrophone()"); |
| 769 | |
| 770 | CriticalSectionScoped cs(&_critSect); |
| 771 | |
| 772 | if (_fileRecording) |
| 773 | { |
| 774 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 775 | "StartRecordingMicrophone() is already recording"); |
| 776 | return 0; |
| 777 | } |
| 778 | |
| 779 | FileFormats format; |
| 780 | const uint32_t notificationTime(0); // Not supported in VoE |
| 781 | CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| 782 | |
| 783 | if (codecInst != NULL && codecInst->channels != 1) |
| 784 | { |
| 785 | _engineStatisticsPtr->SetLastError( |
| 786 | VE_BAD_ARGUMENT, kTraceError, |
| 787 | "StartRecordingMicrophone() invalid compression"); |
| 788 | return (-1); |
| 789 | } |
| 790 | if (codecInst == NULL) |
| 791 | { |
| 792 | format = kFileFormatPcm16kHzFile; |
| 793 | codecInst = &dummyCodec; |
| 794 | } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 795 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 796 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 797 | { |
| 798 | format = kFileFormatWavFile; |
| 799 | } else |
| 800 | { |
| 801 | format = kFileFormatCompressedFile; |
| 802 | } |
| 803 | |
| 804 | // Destroy the old instance |
| 805 | if (_fileRecorderPtr) |
| 806 | { |
| 807 | _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 808 | FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| 809 | _fileRecorderPtr = NULL; |
| 810 | } |
| 811 | |
| 812 | _fileRecorderPtr = |
| 813 | FileRecorder::CreateFileRecorder(_fileRecorderId, |
| 814 | (const FileFormats) format); |
| 815 | if (_fileRecorderPtr == NULL) |
| 816 | { |
| 817 | _engineStatisticsPtr->SetLastError( |
| 818 | VE_INVALID_ARGUMENT, kTraceError, |
| 819 | "StartRecordingMicrophone() fileRecorder format isnot correct"); |
| 820 | return -1; |
| 821 | } |
| 822 | |
| 823 | if (_fileRecorderPtr->StartRecordingAudioFile(*stream, |
| 824 | *codecInst, |
| 825 | notificationTime) != 0) |
| 826 | { |
| 827 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 828 | "StartRecordingAudioFile() failed to start file recording"); |
| 829 | _fileRecorderPtr->StopRecording(); |
| 830 | FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| 831 | _fileRecorderPtr = NULL; |
| 832 | return -1; |
| 833 | } |
| 834 | |
| 835 | _fileRecorderPtr->RegisterModuleFileCallback(this); |
| 836 | _fileRecording = true; |
| 837 | |
| 838 | return 0; |
| 839 | } |
| 840 | |
| 841 | |
| 842 | int TransmitMixer::StopRecordingMicrophone() |
| 843 | { |
| 844 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 845 | "TransmitMixer::StopRecordingMicrophone()"); |
| 846 | |
| 847 | CriticalSectionScoped cs(&_critSect); |
| 848 | |
| 849 | if (!_fileRecording) |
| 850 | { |
| 851 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 852 | "StopRecordingMicrophone() isnot recording"); |
| 853 | return 0; |
| 854 | } |
| 855 | |
| 856 | if (_fileRecorderPtr->StopRecording() != 0) |
| 857 | { |
| 858 | _engineStatisticsPtr->SetLastError( |
| 859 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 860 | "StopRecording(), could not stop recording"); |
| 861 | return -1; |
| 862 | } |
| 863 | _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| 864 | FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| 865 | _fileRecorderPtr = NULL; |
| 866 | _fileRecording = false; |
| 867 | |
| 868 | return 0; |
| 869 | } |
| 870 | |
| 871 | int TransmitMixer::StartRecordingCall(const char* fileName, |
| 872 | const CodecInst* codecInst) |
| 873 | { |
| 874 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 875 | "TransmitMixer::StartRecordingCall(fileName=%s)", fileName); |
| 876 | |
| 877 | if (_fileCallRecording) |
| 878 | { |
| 879 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 880 | "StartRecordingCall() is already recording"); |
| 881 | return 0; |
| 882 | } |
| 883 | |
| 884 | FileFormats format; |
| 885 | const uint32_t notificationTime(0); // Not supported in VoE |
| 886 | CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| 887 | |
| 888 | if (codecInst != NULL && codecInst->channels != 1) |
| 889 | { |
| 890 | _engineStatisticsPtr->SetLastError( |
| 891 | VE_BAD_ARGUMENT, kTraceError, |
| 892 | "StartRecordingCall() invalid compression"); |
| 893 | return (-1); |
| 894 | } |
| 895 | if (codecInst == NULL) |
| 896 | { |
| 897 | format = kFileFormatPcm16kHzFile; |
| 898 | codecInst = &dummyCodec; |
| 899 | } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 900 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 901 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 902 | { |
| 903 | format = kFileFormatWavFile; |
| 904 | } else |
| 905 | { |
| 906 | format = kFileFormatCompressedFile; |
| 907 | } |
| 908 | |
| 909 | CriticalSectionScoped cs(&_critSect); |
| 910 | |
| 911 | // Destroy the old instance |
| 912 | if (_fileCallRecorderPtr) |
| 913 | { |
| 914 | _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| 915 | FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| 916 | _fileCallRecorderPtr = NULL; |
| 917 | } |
| 918 | |
| 919 | _fileCallRecorderPtr |
| 920 | = FileRecorder::CreateFileRecorder(_fileCallRecorderId, |
| 921 | (const FileFormats) format); |
| 922 | if (_fileCallRecorderPtr == NULL) |
| 923 | { |
| 924 | _engineStatisticsPtr->SetLastError( |
| 925 | VE_INVALID_ARGUMENT, kTraceError, |
| 926 | "StartRecordingCall() fileRecorder format isnot correct"); |
| 927 | return -1; |
| 928 | } |
| 929 | |
| 930 | if (_fileCallRecorderPtr->StartRecordingAudioFile( |
| 931 | fileName, |
| 932 | (const CodecInst&) *codecInst, |
| 933 | notificationTime) != 0) |
| 934 | { |
| 935 | _engineStatisticsPtr->SetLastError( |
| 936 | VE_BAD_FILE, kTraceError, |
| 937 | "StartRecordingAudioFile() failed to start file recording"); |
| 938 | _fileCallRecorderPtr->StopRecording(); |
| 939 | FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| 940 | _fileCallRecorderPtr = NULL; |
| 941 | return -1; |
| 942 | } |
| 943 | _fileCallRecorderPtr->RegisterModuleFileCallback(this); |
| 944 | _fileCallRecording = true; |
| 945 | |
| 946 | return 0; |
| 947 | } |
| 948 | |
| 949 | int TransmitMixer::StartRecordingCall(OutStream* stream, |
| 950 | const CodecInst* codecInst) |
| 951 | { |
| 952 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 953 | "TransmitMixer::StartRecordingCall()"); |
| 954 | |
| 955 | if (_fileCallRecording) |
| 956 | { |
| 957 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 958 | "StartRecordingCall() is already recording"); |
| 959 | return 0; |
| 960 | } |
| 961 | |
| 962 | FileFormats format; |
| 963 | const uint32_t notificationTime(0); // Not supported in VoE |
| 964 | CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| 965 | |
| 966 | if (codecInst != NULL && codecInst->channels != 1) |
| 967 | { |
| 968 | _engineStatisticsPtr->SetLastError( |
| 969 | VE_BAD_ARGUMENT, kTraceError, |
| 970 | "StartRecordingCall() invalid compression"); |
| 971 | return (-1); |
| 972 | } |
| 973 | if (codecInst == NULL) |
| 974 | { |
| 975 | format = kFileFormatPcm16kHzFile; |
| 976 | codecInst = &dummyCodec; |
| 977 | } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| 978 | (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| 979 | (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| 980 | { |
| 981 | format = kFileFormatWavFile; |
| 982 | } else |
| 983 | { |
| 984 | format = kFileFormatCompressedFile; |
| 985 | } |
| 986 | |
| 987 | CriticalSectionScoped cs(&_critSect); |
| 988 | |
| 989 | // Destroy the old instance |
| 990 | if (_fileCallRecorderPtr) |
| 991 | { |
| 992 | _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| 993 | FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| 994 | _fileCallRecorderPtr = NULL; |
| 995 | } |
| 996 | |
| 997 | _fileCallRecorderPtr = |
| 998 | FileRecorder::CreateFileRecorder(_fileCallRecorderId, |
| 999 | (const FileFormats) format); |
| 1000 | if (_fileCallRecorderPtr == NULL) |
| 1001 | { |
| 1002 | _engineStatisticsPtr->SetLastError( |
| 1003 | VE_INVALID_ARGUMENT, kTraceError, |
| 1004 | "StartRecordingCall() fileRecorder format isnot correct"); |
| 1005 | return -1; |
| 1006 | } |
| 1007 | |
| 1008 | if (_fileCallRecorderPtr->StartRecordingAudioFile(*stream, |
| 1009 | *codecInst, |
| 1010 | notificationTime) != 0) |
| 1011 | { |
| 1012 | _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| 1013 | "StartRecordingAudioFile() failed to start file recording"); |
| 1014 | _fileCallRecorderPtr->StopRecording(); |
| 1015 | FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| 1016 | _fileCallRecorderPtr = NULL; |
| 1017 | return -1; |
| 1018 | } |
| 1019 | |
| 1020 | _fileCallRecorderPtr->RegisterModuleFileCallback(this); |
| 1021 | _fileCallRecording = true; |
| 1022 | |
| 1023 | return 0; |
| 1024 | } |
| 1025 | |
| 1026 | int TransmitMixer::StopRecordingCall() |
| 1027 | { |
| 1028 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 1029 | "TransmitMixer::StopRecordingCall()"); |
| 1030 | |
| 1031 | if (!_fileCallRecording) |
| 1032 | { |
| 1033 | WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| 1034 | "StopRecordingCall() file isnot recording"); |
| 1035 | return -1; |
| 1036 | } |
| 1037 | |
| 1038 | CriticalSectionScoped cs(&_critSect); |
| 1039 | |
| 1040 | if (_fileCallRecorderPtr->StopRecording() != 0) |
| 1041 | { |
| 1042 | _engineStatisticsPtr->SetLastError( |
| 1043 | VE_STOP_RECORDING_FAILED, kTraceError, |
| 1044 | "StopRecording(), could not stop recording"); |
| 1045 | return -1; |
| 1046 | } |
| 1047 | |
| 1048 | _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| 1049 | FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| 1050 | _fileCallRecorderPtr = NULL; |
| 1051 | _fileCallRecording = false; |
| 1052 | |
| 1053 | return 0; |
| 1054 | } |
| 1055 | |
| 1056 | void |
| 1057 | TransmitMixer::SetMixWithMicStatus(bool mix) |
| 1058 | { |
| 1059 | _mixFileWithMicrophone = mix; |
| 1060 | } |
| 1061 | |
| 1062 | int TransmitMixer::RegisterExternalMediaProcessing( |
| 1063 | VoEMediaProcess* object, |
| 1064 | ProcessingTypes type) { |
| 1065 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 1066 | "TransmitMixer::RegisterExternalMediaProcessing()"); |
| 1067 | |
| 1068 | CriticalSectionScoped cs(&_callbackCritSect); |
| 1069 | if (!object) { |
| 1070 | return -1; |
| 1071 | } |
| 1072 | |
| 1073 | // Store the callback object according to the processing type. |
| 1074 | if (type == kRecordingAllChannelsMixed) { |
| 1075 | external_postproc_ptr_ = object; |
| 1076 | } else if (type == kRecordingPreprocessing) { |
| 1077 | external_preproc_ptr_ = object; |
| 1078 | } else { |
| 1079 | return -1; |
| 1080 | } |
| 1081 | return 0; |
| 1082 | } |
| 1083 | |
| 1084 | int TransmitMixer::DeRegisterExternalMediaProcessing(ProcessingTypes type) { |
| 1085 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 1086 | "TransmitMixer::DeRegisterExternalMediaProcessing()"); |
| 1087 | |
| 1088 | CriticalSectionScoped cs(&_callbackCritSect); |
| 1089 | if (type == kRecordingAllChannelsMixed) { |
| 1090 | external_postproc_ptr_ = NULL; |
| 1091 | } else if (type == kRecordingPreprocessing) { |
| 1092 | external_preproc_ptr_ = NULL; |
| 1093 | } else { |
| 1094 | return -1; |
| 1095 | } |
| 1096 | return 0; |
| 1097 | } |
| 1098 | |
| 1099 | int |
| 1100 | TransmitMixer::SetMute(bool enable) |
| 1101 | { |
| 1102 | WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| 1103 | "TransmitMixer::SetMute(enable=%d)", enable); |
| 1104 | _mute = enable; |
| 1105 | return 0; |
| 1106 | } |
| 1107 | |
| 1108 | bool |
| 1109 | TransmitMixer::Mute() const |
| 1110 | { |
| 1111 | return _mute; |
| 1112 | } |
| 1113 | |
| 1114 | int8_t TransmitMixer::AudioLevel() const |
| 1115 | { |
| 1116 | // Speech + file level [0,9] |
| 1117 | return _audioLevel.Level(); |
| 1118 | } |
| 1119 | |
| 1120 | int16_t TransmitMixer::AudioLevelFullRange() const |
| 1121 | { |
| 1122 | // Speech + file level [0,32767] |
| 1123 | return _audioLevel.LevelFullRange(); |
| 1124 | } |
| 1125 | |
| 1126 | bool TransmitMixer::IsRecordingCall() |
| 1127 | { |
| 1128 | return _fileCallRecording; |
| 1129 | } |
| 1130 | |
| 1131 | bool TransmitMixer::IsRecordingMic() |
| 1132 | { |
| 1133 | CriticalSectionScoped cs(&_critSect); |
| 1134 | return _fileRecording; |
| 1135 | } |
| 1136 | |
| 1137 | void TransmitMixer::GenerateAudioFrame(const int16_t* audio, |
| 1138 | int samples_per_channel, |
| 1139 | int num_channels, |
| 1140 | int sample_rate_hz) { |
| 1141 | int codec_rate; |
| 1142 | int num_codec_channels; |
| 1143 | GetSendCodecInfo(&codec_rate, &num_codec_channels); |
| 1144 | // TODO(ajm): This currently restricts the sample rate to 32 kHz. |
| 1145 | // See: https://code.google.com/p/webrtc/issues/detail?id=3146 |
| 1146 | // When 48 kHz is supported natively by AudioProcessing, this will have |
| 1147 | // to be changed to handle 44.1 kHz. |
andrew@webrtc.org | c1878ac | 2014-04-30 18:58:23 +0000 | [diff] [blame] | 1148 | int max_sample_rate_hz = kAudioProcMaxNativeSampleRateHz; |
| 1149 | if (audioproc_->echo_control_mobile()->is_enabled()) { |
| 1150 | // AECM only supports 8 and 16 kHz. |
| 1151 | max_sample_rate_hz = 16000; |
| 1152 | } |
| 1153 | codec_rate = std::min(codec_rate, max_sample_rate_hz); |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 1154 | stereo_codec_ = num_codec_channels == 2; |
| 1155 | |
| 1156 | if (!mono_buffer_.get()) { |
| 1157 | // Temporary space for DownConvertToCodecFormat. |
| 1158 | mono_buffer_.reset(new int16_t[kMaxMonoDataSizeSamples]); |
| 1159 | } |
| 1160 | DownConvertToCodecFormat(audio, |
| 1161 | samples_per_channel, |
| 1162 | num_channels, |
| 1163 | sample_rate_hz, |
| 1164 | num_codec_channels, |
| 1165 | codec_rate, |
| 1166 | mono_buffer_.get(), |
| 1167 | &resampler_, |
| 1168 | &_audioFrame); |
| 1169 | } |
| 1170 | |
| 1171 | int32_t TransmitMixer::RecordAudioToFile( |
| 1172 | uint32_t mixingFrequency) |
| 1173 | { |
| 1174 | CriticalSectionScoped cs(&_critSect); |
| 1175 | if (_fileRecorderPtr == NULL) |
| 1176 | { |
| 1177 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 1178 | "TransmitMixer::RecordAudioToFile() filerecorder doesnot" |
| 1179 | "exist"); |
| 1180 | return -1; |
| 1181 | } |
| 1182 | |
| 1183 | if (_fileRecorderPtr->RecordAudioToFile(_audioFrame) != 0) |
| 1184 | { |
| 1185 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 1186 | "TransmitMixer::RecordAudioToFile() file recording" |
| 1187 | "failed"); |
| 1188 | return -1; |
| 1189 | } |
| 1190 | |
| 1191 | return 0; |
| 1192 | } |
| 1193 | |
| 1194 | int32_t TransmitMixer::MixOrReplaceAudioWithFile( |
| 1195 | int mixingFrequency) |
| 1196 | { |
andrew@webrtc.org | ba47616 | 2014-04-25 23:10:28 +0000 | [diff] [blame] | 1197 | scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]); |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 1198 | |
| 1199 | int fileSamples(0); |
| 1200 | { |
| 1201 | CriticalSectionScoped cs(&_critSect); |
| 1202 | if (_filePlayerPtr == NULL) |
| 1203 | { |
| 1204 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| 1205 | VoEId(_instanceId, -1), |
| 1206 | "TransmitMixer::MixOrReplaceAudioWithFile()" |
| 1207 | "fileplayer doesnot exist"); |
| 1208 | return -1; |
| 1209 | } |
| 1210 | |
| 1211 | if (_filePlayerPtr->Get10msAudioFromFile(fileBuffer.get(), |
| 1212 | fileSamples, |
| 1213 | mixingFrequency) == -1) |
| 1214 | { |
| 1215 | WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| 1216 | "TransmitMixer::MixOrReplaceAudioWithFile() file" |
| 1217 | " mixing failed"); |
| 1218 | return -1; |
| 1219 | } |
| 1220 | } |
| 1221 | |
| 1222 | assert(_audioFrame.samples_per_channel_ == fileSamples); |
| 1223 | |
| 1224 | if (_mixFileWithMicrophone) |
| 1225 | { |
| 1226 | // Currently file stream is always mono. |
| 1227 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
| 1228 | MixWithSat(_audioFrame.data_, |
| 1229 | _audioFrame.num_channels_, |
| 1230 | fileBuffer.get(), |
| 1231 | 1, |
| 1232 | fileSamples); |
| 1233 | } else |
| 1234 | { |
| 1235 | // Replace ACM audio with file. |
| 1236 | // Currently file stream is always mono. |
| 1237 | // TODO(xians): Change the code when FilePlayer supports real stereo. |
| 1238 | _audioFrame.UpdateFrame(-1, |
tommi@webrtc.org | 9fbd3ec | 2014-07-11 19:09:59 +0000 | [diff] [blame^] | 1239 | 0xFFFFFFFF, |
andrew@webrtc.org | f7c73b5 | 2014-04-03 21:56:01 +0000 | [diff] [blame] | 1240 | fileBuffer.get(), |
| 1241 | fileSamples, |
| 1242 | mixingFrequency, |
| 1243 | AudioFrame::kNormalSpeech, |
| 1244 | AudioFrame::kVadUnknown, |
| 1245 | 1); |
| 1246 | } |
| 1247 | return 0; |
| 1248 | } |
| 1249 | |
| 1250 | void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift, |
| 1251 | int current_mic_level, bool key_pressed) { |
| 1252 | if (audioproc_->set_stream_delay_ms(delay_ms) != 0) { |
| 1253 | // A redundant warning is reported in AudioDevice, which we've throttled |
| 1254 | // to avoid flooding the logs. Relegate this one to LS_VERBOSE to avoid |
| 1255 | // repeating the problem here. |
| 1256 | LOG_FERR1(LS_VERBOSE, set_stream_delay_ms, delay_ms); |
| 1257 | } |
| 1258 | |
| 1259 | GainControl* agc = audioproc_->gain_control(); |
| 1260 | if (agc->set_stream_analog_level(current_mic_level) != 0) { |
| 1261 | LOG_FERR1(LS_ERROR, set_stream_analog_level, current_mic_level); |
| 1262 | assert(false); |
| 1263 | } |
| 1264 | |
| 1265 | EchoCancellation* aec = audioproc_->echo_cancellation(); |
| 1266 | if (aec->is_drift_compensation_enabled()) { |
| 1267 | aec->set_stream_drift_samples(clock_drift); |
| 1268 | } |
| 1269 | |
| 1270 | audioproc_->set_stream_key_pressed(key_pressed); |
| 1271 | |
| 1272 | int err = audioproc_->ProcessStream(&_audioFrame); |
| 1273 | if (err != 0) { |
| 1274 | LOG(LS_ERROR) << "ProcessStream() error: " << err; |
| 1275 | assert(false); |
| 1276 | } |
| 1277 | |
| 1278 | // Store new capture level. Only updated when analog AGC is enabled. |
| 1279 | _captureLevel = agc->stream_analog_level(); |
| 1280 | |
| 1281 | CriticalSectionScoped cs(&_critSect); |
| 1282 | // Triggers a callback in OnPeriodicProcess(). |
| 1283 | _saturationWarning |= agc->stream_is_saturated(); |
| 1284 | } |
| 1285 | |
| 1286 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 1287 | void TransmitMixer::TypingDetection(bool keyPressed) |
| 1288 | { |
| 1289 | // We let the VAD determine if we're using this feature or not. |
| 1290 | if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) { |
| 1291 | return; |
| 1292 | } |
| 1293 | |
| 1294 | bool vadActive = _audioFrame.vad_activity_ == AudioFrame::kVadActive; |
| 1295 | if (_typingDetection.Process(keyPressed, vadActive)) { |
| 1296 | _typingNoiseWarningPending = true; |
| 1297 | _typingNoiseDetected = true; |
| 1298 | } else { |
| 1299 | // If there is already a warning pending, do not change the state. |
| 1300 | // Otherwise set a warning pending if last callback was for noise detected. |
| 1301 | if (!_typingNoiseWarningPending && _typingNoiseDetected) { |
| 1302 | _typingNoiseWarningPending = true; |
| 1303 | _typingNoiseDetected = false; |
| 1304 | } |
| 1305 | } |
| 1306 | } |
| 1307 | #endif |
| 1308 | |
| 1309 | int TransmitMixer::GetMixingFrequency() |
| 1310 | { |
| 1311 | assert(_audioFrame.sample_rate_hz_ != 0); |
| 1312 | return _audioFrame.sample_rate_hz_; |
| 1313 | } |
| 1314 | |
| 1315 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 1316 | int TransmitMixer::TimeSinceLastTyping(int &seconds) |
| 1317 | { |
| 1318 | // We check in VoEAudioProcessingImpl that this is only called when |
| 1319 | // typing detection is active. |
| 1320 | seconds = _typingDetection.TimeSinceLastDetectionInSeconds(); |
| 1321 | return 0; |
| 1322 | } |
| 1323 | #endif |
| 1324 | |
| 1325 | #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| 1326 | int TransmitMixer::SetTypingDetectionParameters(int timeWindow, |
| 1327 | int costPerTyping, |
| 1328 | int reportingThreshold, |
| 1329 | int penaltyDecay, |
| 1330 | int typeEventDelay) |
| 1331 | { |
| 1332 | _typingDetection.SetParameters(timeWindow, |
| 1333 | costPerTyping, |
| 1334 | reportingThreshold, |
| 1335 | penaltyDecay, |
| 1336 | typeEventDelay, |
| 1337 | 0); |
| 1338 | return 0; |
| 1339 | } |
| 1340 | #endif |
| 1341 | |
| 1342 | void TransmitMixer::EnableStereoChannelSwapping(bool enable) { |
| 1343 | swap_stereo_channels_ = enable; |
| 1344 | } |
| 1345 | |
| 1346 | bool TransmitMixer::IsStereoChannelSwappingEnabled() { |
| 1347 | return swap_stereo_channels_; |
| 1348 | } |
| 1349 | |
| 1350 | } // namespace voe |
| 1351 | } // namespace webrtc |