| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/voice_engine/transmit_mixer.h" |
| |
| #include "webrtc/modules/utility/interface/audio_frame_operations.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/event_wrapper.h" |
| #include "webrtc/system_wrappers/interface/logging.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| #include "webrtc/voice_engine/channel.h" |
| #include "webrtc/voice_engine/channel_manager.h" |
| #include "webrtc/voice_engine/include/voe_external_media.h" |
| #include "webrtc/voice_engine/statistics.h" |
| #include "webrtc/voice_engine/utility.h" |
| #include "webrtc/voice_engine/voe_base_impl.h" |
| |
| #define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a)) |
| |
| namespace webrtc { |
| |
| namespace voe { |
| |
| // Used for downmixing before resampling. |
| // TODO(ajm): audio_device should advertise the maximum sample rate it can |
| // provide. |
| static const int kMaxMonoDeviceDataSizeSamples = 960; // 10 ms, 96 kHz, mono. |
| |
| // TODO(ajm): The thread safety of this is dubious... |
| void |
| TransmitMixer::OnPeriodicProcess() |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::OnPeriodicProcess()"); |
| |
| #if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION) |
| if (_typingNoiseWarning) |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (_voiceEngineObserverPtr) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::OnPeriodicProcess() => " |
| "CallbackOnError(VE_TYPING_NOISE_WARNING)"); |
| _voiceEngineObserverPtr->CallbackOnError(-1, |
| VE_TYPING_NOISE_WARNING); |
| } |
| _typingNoiseWarning = false; |
| } |
| #endif |
| |
| bool saturationWarning = false; |
| { |
| // Modify |_saturationWarning| under lock to avoid conflict with write op |
| // in ProcessAudio and also ensure that we don't hold the lock during the |
| // callback. |
| CriticalSectionScoped cs(&_critSect); |
| saturationWarning = _saturationWarning; |
| if (_saturationWarning) |
| _saturationWarning = false; |
| } |
| |
| if (saturationWarning) |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (_voiceEngineObserverPtr) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::OnPeriodicProcess() =>" |
| " CallbackOnError(VE_SATURATION_WARNING)"); |
| _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING); |
| } |
| } |
| } |
| |
| |
| void TransmitMixer::PlayNotification(int32_t id, |
| uint32_t durationMs) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::PlayNotification(id=%d, durationMs=%d)", |
| id, durationMs); |
| |
| // Not implement yet |
| } |
| |
| void TransmitMixer::RecordNotification(int32_t id, |
| uint32_t durationMs) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "TransmitMixer::RecordNotification(id=%d, durationMs=%d)", |
| id, durationMs); |
| |
| // Not implement yet |
| } |
| |
| void TransmitMixer::PlayFileEnded(int32_t id) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::PlayFileEnded(id=%d)", id); |
| |
| assert(id == _filePlayerId); |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| _filePlaying = false; |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::PlayFileEnded() =>" |
| "file player module is shutdown"); |
| } |
| |
| void |
| TransmitMixer::RecordFileEnded(int32_t id) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::RecordFileEnded(id=%d)", id); |
| |
| if (id == _fileRecorderId) |
| { |
| CriticalSectionScoped cs(&_critSect); |
| _fileRecording = false; |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::RecordFileEnded() => fileRecorder module" |
| "is shutdown"); |
| } else if (id == _fileCallRecorderId) |
| { |
| CriticalSectionScoped cs(&_critSect); |
| _fileCallRecording = false; |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::RecordFileEnded() => fileCallRecorder" |
| "module is shutdown"); |
| } |
| } |
| |
| int32_t |
| TransmitMixer::Create(TransmitMixer*& mixer, uint32_t instanceId) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1), |
| "TransmitMixer::Create(instanceId=%d)", instanceId); |
| mixer = new TransmitMixer(instanceId); |
| if (mixer == NULL) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1), |
| "TransmitMixer::Create() unable to allocate memory" |
| "for mixer"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| void |
| TransmitMixer::Destroy(TransmitMixer*& mixer) |
| { |
| if (mixer) |
| { |
| delete mixer; |
| mixer = NULL; |
| } |
| } |
| |
| TransmitMixer::TransmitMixer(uint32_t instanceId) : |
| _engineStatisticsPtr(NULL), |
| _channelManagerPtr(NULL), |
| audioproc_(NULL), |
| _voiceEngineObserverPtr(NULL), |
| _processThreadPtr(NULL), |
| _filePlayerPtr(NULL), |
| _fileRecorderPtr(NULL), |
| _fileCallRecorderPtr(NULL), |
| // Avoid conflict with other channels by adding 1024 - 1026, |
| // won't use as much as 1024 channels. |
| _filePlayerId(instanceId + 1024), |
| _fileRecorderId(instanceId + 1025), |
| _fileCallRecorderId(instanceId + 1026), |
| _filePlaying(false), |
| _fileRecording(false), |
| _fileCallRecording(false), |
| _audioLevel(), |
| _critSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| _timeActive(0), |
| _timeSinceLastTyping(0), |
| _penaltyCounter(0), |
| _typingNoiseWarning(false), |
| _timeWindow(10), // 10ms slots accepted to count as a hit |
| _costPerTyping(100), // Penalty added for a typing + activity coincide |
| _reportingThreshold(300), // Threshold for _penaltyCounter |
| _penaltyDecay(1), // how much we reduce _penaltyCounter every 10 ms. |
| _typeEventDelay(2), // how "old" event we check for |
| #endif |
| _saturationWarning(false), |
| _instanceId(instanceId), |
| _mixFileWithMicrophone(false), |
| _captureLevel(0), |
| external_postproc_ptr_(NULL), |
| external_preproc_ptr_(NULL), |
| _mute(false), |
| _remainingMuteMicTimeMs(0), |
| stereo_codec_(false), |
| swap_stereo_channels_(false) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::TransmitMixer() - ctor"); |
| } |
| |
| TransmitMixer::~TransmitMixer() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::~TransmitMixer() - dtor"); |
| _monitorModule.DeRegisterObserver(); |
| if (_processThreadPtr) |
| { |
| _processThreadPtr->DeRegisterModule(&_monitorModule); |
| } |
| DeRegisterExternalMediaProcessing(kRecordingAllChannelsMixed); |
| DeRegisterExternalMediaProcessing(kRecordingPreprocessing); |
| { |
| CriticalSectionScoped cs(&_critSect); |
| if (_fileRecorderPtr) |
| { |
| _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| _fileRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| _fileRecorderPtr = NULL; |
| } |
| if (_fileCallRecorderPtr) |
| { |
| _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| _fileCallRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| _fileCallRecorderPtr = NULL; |
| } |
| if (_filePlayerPtr) |
| { |
| _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| _filePlayerPtr->StopPlayingFile(); |
| FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| _filePlayerPtr = NULL; |
| } |
| } |
| delete &_critSect; |
| delete &_callbackCritSect; |
| } |
| |
| int32_t |
| TransmitMixer::SetEngineInformation(ProcessThread& processThread, |
| Statistics& engineStatistics, |
| ChannelManager& channelManager) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::SetEngineInformation()"); |
| |
| _processThreadPtr = &processThread; |
| _engineStatisticsPtr = &engineStatistics; |
| _channelManagerPtr = &channelManager; |
| |
| if (_processThreadPtr->RegisterModule(&_monitorModule) == -1) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::SetEngineInformation() failed to" |
| "register the monitor module"); |
| } else |
| { |
| _monitorModule.RegisterObserver(*this); |
| } |
| |
| return 0; |
| } |
| |
| int32_t |
| TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::RegisterVoiceEngineObserver()"); |
| CriticalSectionScoped cs(&_callbackCritSect); |
| |
| if (_voiceEngineObserverPtr) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_OPERATION, kTraceError, |
| "RegisterVoiceEngineObserver() observer already enabled"); |
| return -1; |
| } |
| _voiceEngineObserverPtr = &observer; |
| return 0; |
| } |
| |
| int32_t |
| TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::SetAudioProcessingModule(" |
| "audioProcessingModule=0x%x)", |
| audioProcessingModule); |
| audioproc_ = audioProcessingModule; |
| return 0; |
| } |
| |
| void TransmitMixer::GetSendCodecInfo(int* max_sample_rate, int* max_channels) { |
| ScopedChannel sc(*_channelManagerPtr); |
| void* iterator = NULL; |
| Channel* channel = sc.GetFirstChannel(iterator); |
| |
| *max_sample_rate = 8000; |
| *max_channels = 1; |
| while (channel != NULL) { |
| if (channel->Sending()) { |
| CodecInst codec; |
| channel->GetSendCodec(codec); |
| // TODO(tlegrand): Remove the 32 kHz restriction once we have full 48 kHz |
| // support in Audio Coding Module. |
| *max_sample_rate = std::min(32000, |
| std::max(*max_sample_rate, codec.plfreq)); |
| *max_channels = std::max(*max_channels, codec.channels); |
| } |
| channel = sc.GetNextChannel(iterator); |
| } |
| } |
| |
| int32_t |
| TransmitMixer::PrepareDemux(const void* audioSamples, |
| uint32_t nSamples, |
| uint8_t nChannels, |
| uint32_t samplesPerSec, |
| uint16_t totalDelayMS, |
| int32_t clockDrift, |
| uint16_t currentMicLevel, |
| bool keyPressed) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u," |
| "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%d," |
| "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec, |
| totalDelayMS, clockDrift, currentMicLevel); |
| |
| // --- Resample input audio and create/store the initial audio frame |
| if (GenerateAudioFrame(static_cast<const int16_t*>(audioSamples), |
| nSamples, |
| nChannels, |
| samplesPerSec) == -1) |
| { |
| return -1; |
| } |
| |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (external_preproc_ptr_) { |
| external_preproc_ptr_->Process(-1, kRecordingPreprocessing, |
| _audioFrame.data_, |
| _audioFrame.samples_per_channel_, |
| _audioFrame.sample_rate_hz_, |
| _audioFrame.num_channels_ == 2); |
| } |
| } |
| |
| // --- Near-end audio processing. |
| ProcessAudio(totalDelayMS, clockDrift, currentMicLevel); |
| |
| if (swap_stereo_channels_ && stereo_codec_) |
| // Only bother swapping if we're using a stereo codec. |
| AudioFrameOperations::SwapStereoChannels(&_audioFrame); |
| |
| // --- Annoying typing detection (utilizes the APM/VAD decision) |
| #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| TypingDetection(keyPressed); |
| #endif |
| |
| // --- Mute during DTMF tone if direct feedback is enabled |
| if (_remainingMuteMicTimeMs > 0) |
| { |
| AudioFrameOperations::Mute(_audioFrame); |
| _remainingMuteMicTimeMs -= 10; |
| if (_remainingMuteMicTimeMs < 0) |
| { |
| _remainingMuteMicTimeMs = 0; |
| } |
| } |
| |
| // --- Mute signal |
| if (_mute) |
| { |
| AudioFrameOperations::Mute(_audioFrame); |
| } |
| |
| // --- Mix with file (does not affect the mixing frequency) |
| if (_filePlaying) |
| { |
| MixOrReplaceAudioWithFile(_audioFrame.sample_rate_hz_); |
| } |
| |
| // --- Record to file |
| if (_fileRecording) |
| { |
| RecordAudioToFile(_audioFrame.sample_rate_hz_); |
| } |
| |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (external_postproc_ptr_) { |
| external_postproc_ptr_->Process(-1, kRecordingAllChannelsMixed, |
| _audioFrame.data_, |
| _audioFrame.samples_per_channel_, |
| _audioFrame.sample_rate_hz_, |
| _audioFrame.num_channels_ == 2); |
| } |
| } |
| |
| // --- Measure audio level of speech after all processing. |
| _audioLevel.ComputeLevel(_audioFrame); |
| return 0; |
| } |
| |
| int32_t |
| TransmitMixer::DemuxAndMix() |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::DemuxAndMix()"); |
| |
| ScopedChannel sc(*_channelManagerPtr); |
| void* iterator(NULL); |
| Channel* channelPtr = sc.GetFirstChannel(iterator); |
| while (channelPtr != NULL) |
| { |
| if (channelPtr->InputIsOnHold()) |
| { |
| channelPtr->UpdateLocalTimeStamp(); |
| } else if (channelPtr->Sending()) |
| { |
| // Demultiplex makes a copy of its input. |
| channelPtr->Demultiplex(_audioFrame); |
| channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_); |
| } |
| channelPtr = sc.GetNextChannel(iterator); |
| } |
| return 0; |
| } |
| |
| int32_t |
| TransmitMixer::EncodeAndSend() |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::EncodeAndSend()"); |
| |
| ScopedChannel sc(*_channelManagerPtr); |
| void* iterator(NULL); |
| Channel* channelPtr = sc.GetFirstChannel(iterator); |
| while (channelPtr != NULL) |
| { |
| if (channelPtr->Sending() && !channelPtr->InputIsOnHold()) |
| { |
| channelPtr->EncodeAndSend(); |
| } |
| channelPtr = sc.GetNextChannel(iterator); |
| } |
| return 0; |
| } |
| |
| uint32_t TransmitMixer::CaptureLevel() const |
| { |
| CriticalSectionScoped cs(&_critSect); |
| return _captureLevel; |
| } |
| |
| void |
| TransmitMixer::UpdateMuteMicrophoneTime(uint32_t lengthMs) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)", |
| lengthMs); |
| _remainingMuteMicTimeMs = lengthMs; |
| } |
| |
| int32_t |
| TransmitMixer::StopSend() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StopSend()"); |
| _audioLevel.Clear(); |
| return 0; |
| } |
| |
| int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName, |
| bool loop, |
| FileFormats format, |
| int startPosition, |
| float volumeScaling, |
| int stopPosition, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StartPlayingFileAsMicrophone(" |
| "fileNameUTF8[]=%s,loop=%d, format=%d, volumeScaling=%5.3f," |
| " startPosition=%d, stopPosition=%d)", fileName, loop, |
| format, volumeScaling, startPosition, stopPosition); |
| |
| if (_filePlaying) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_ALREADY_PLAYING, kTraceWarning, |
| "StartPlayingFileAsMicrophone() is already playing"); |
| return 0; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| // Destroy the old instance |
| if (_filePlayerPtr) |
| { |
| _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| _filePlayerPtr = NULL; |
| } |
| |
| // Dynamically create the instance |
| _filePlayerPtr |
| = FilePlayer::CreateFilePlayer(_filePlayerId, |
| (const FileFormats) format); |
| |
| if (_filePlayerPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceError, |
| "StartPlayingFileAsMicrophone() filePlayer format isnot correct"); |
| return -1; |
| } |
| |
| const uint32_t notificationTime(0); |
| |
| if (_filePlayerPtr->StartPlayingFile( |
| fileName, |
| loop, |
| startPosition, |
| volumeScaling, |
| notificationTime, |
| stopPosition, |
| (const CodecInst*) codecInst) != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_FILE, kTraceError, |
| "StartPlayingFile() failed to start file playout"); |
| _filePlayerPtr->StopPlayingFile(); |
| FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| _filePlayerPtr = NULL; |
| return -1; |
| } |
| |
| _filePlayerPtr->RegisterModuleFileCallback(this); |
| _filePlaying = true; |
| |
| return 0; |
| } |
| |
| int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream, |
| FileFormats format, |
| int startPosition, |
| float volumeScaling, |
| int stopPosition, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "TransmitMixer::StartPlayingFileAsMicrophone(format=%d," |
| " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)", |
| format, volumeScaling, startPosition, stopPosition); |
| |
| if (stream == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_FILE, kTraceError, |
| "StartPlayingFileAsMicrophone() NULL as input stream"); |
| return -1; |
| } |
| |
| if (_filePlaying) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_ALREADY_PLAYING, kTraceWarning, |
| "StartPlayingFileAsMicrophone() is already playing"); |
| return 0; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| // Destroy the old instance |
| if (_filePlayerPtr) |
| { |
| _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| _filePlayerPtr = NULL; |
| } |
| |
| // Dynamically create the instance |
| _filePlayerPtr |
| = FilePlayer::CreateFilePlayer(_filePlayerId, |
| (const FileFormats) format); |
| |
| if (_filePlayerPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceWarning, |
| "StartPlayingFileAsMicrophone() filePlayer format isnot correct"); |
| return -1; |
| } |
| |
| const uint32_t notificationTime(0); |
| |
| if (_filePlayerPtr->StartPlayingFile( |
| (InStream&) *stream, |
| startPosition, |
| volumeScaling, |
| notificationTime, |
| stopPosition, |
| (const CodecInst*) codecInst) != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_FILE, kTraceError, |
| "StartPlayingFile() failed to start file playout"); |
| _filePlayerPtr->StopPlayingFile(); |
| FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| _filePlayerPtr = NULL; |
| return -1; |
| } |
| _filePlayerPtr->RegisterModuleFileCallback(this); |
| _filePlaying = true; |
| |
| return 0; |
| } |
| |
| int TransmitMixer::StopPlayingFileAsMicrophone() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "TransmitMixer::StopPlayingFileAsMicrophone()"); |
| |
| if (!_filePlaying) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_OPERATION, kTraceWarning, |
| "StopPlayingFileAsMicrophone() isnot playing"); |
| return 0; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| if (_filePlayerPtr->StopPlayingFile() != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_CANNOT_STOP_PLAYOUT, kTraceError, |
| "StopPlayingFile() couldnot stop playing file"); |
| return -1; |
| } |
| |
| _filePlayerPtr->RegisterModuleFileCallback(NULL); |
| FilePlayer::DestroyFilePlayer(_filePlayerPtr); |
| _filePlayerPtr = NULL; |
| _filePlaying = false; |
| |
| return 0; |
| } |
| |
| int TransmitMixer::IsPlayingFileAsMicrophone() const |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::IsPlayingFileAsMicrophone()"); |
| return _filePlaying; |
| } |
| |
| int TransmitMixer::ScaleFileAsMicrophonePlayout(float scale) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)", |
| scale); |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| if (!_filePlaying) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_OPERATION, kTraceError, |
| "ScaleFileAsMicrophonePlayout() isnot playing file"); |
| return -1; |
| } |
| |
| if ((_filePlayerPtr == NULL) || |
| (_filePlayerPtr->SetAudioScaling(scale) != 0)) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_ARGUMENT, kTraceError, |
| "SetAudioScaling() failed to scale playout"); |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| int TransmitMixer::StartRecordingMicrophone(const char* fileName, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StartRecordingMicrophone(fileName=%s)", |
| fileName); |
| |
| if (_fileRecording) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "StartRecordingMicrophone() is already recording"); |
| return 0; |
| } |
| |
| FileFormats format; |
| const uint32_t notificationTime(0); // Not supported in VoE |
| CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| |
| if (codecInst != NULL && |
| (codecInst->channels < 0 || codecInst->channels > 2)) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_ARGUMENT, kTraceError, |
| "StartRecordingMicrophone() invalid compression"); |
| return (-1); |
| } |
| if (codecInst == NULL) |
| { |
| format = kFileFormatPcm16kHzFile; |
| codecInst = &dummyCodec; |
| } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| { |
| format = kFileFormatWavFile; |
| } else |
| { |
| format = kFileFormatCompressedFile; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| // Destroy the old instance |
| if (_fileRecorderPtr) |
| { |
| _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| _fileRecorderPtr = NULL; |
| } |
| |
| _fileRecorderPtr = |
| FileRecorder::CreateFileRecorder(_fileRecorderId, |
| (const FileFormats) format); |
| if (_fileRecorderPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceError, |
| "StartRecordingMicrophone() fileRecorder format isnot correct"); |
| return -1; |
| } |
| |
| if (_fileRecorderPtr->StartRecordingAudioFile( |
| fileName, |
| (const CodecInst&) *codecInst, |
| notificationTime) != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_FILE, kTraceError, |
| "StartRecordingAudioFile() failed to start file recording"); |
| _fileRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| _fileRecorderPtr = NULL; |
| return -1; |
| } |
| _fileRecorderPtr->RegisterModuleFileCallback(this); |
| _fileRecording = true; |
| |
| return 0; |
| } |
| |
| int TransmitMixer::StartRecordingMicrophone(OutStream* stream, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StartRecordingMicrophone()"); |
| |
| if (_fileRecording) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "StartRecordingMicrophone() is already recording"); |
| return 0; |
| } |
| |
| FileFormats format; |
| const uint32_t notificationTime(0); // Not supported in VoE |
| CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| |
| if (codecInst != NULL && codecInst->channels != 1) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_ARGUMENT, kTraceError, |
| "StartRecordingMicrophone() invalid compression"); |
| return (-1); |
| } |
| if (codecInst == NULL) |
| { |
| format = kFileFormatPcm16kHzFile; |
| codecInst = &dummyCodec; |
| } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| { |
| format = kFileFormatWavFile; |
| } else |
| { |
| format = kFileFormatCompressedFile; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| // Destroy the old instance |
| if (_fileRecorderPtr) |
| { |
| _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| _fileRecorderPtr = NULL; |
| } |
| |
| _fileRecorderPtr = |
| FileRecorder::CreateFileRecorder(_fileRecorderId, |
| (const FileFormats) format); |
| if (_fileRecorderPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceError, |
| "StartRecordingMicrophone() fileRecorder format isnot correct"); |
| return -1; |
| } |
| |
| if (_fileRecorderPtr->StartRecordingAudioFile(*stream, |
| *codecInst, |
| notificationTime) != 0) |
| { |
| _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| "StartRecordingAudioFile() failed to start file recording"); |
| _fileRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| _fileRecorderPtr = NULL; |
| return -1; |
| } |
| |
| _fileRecorderPtr->RegisterModuleFileCallback(this); |
| _fileRecording = true; |
| |
| return 0; |
| } |
| |
| |
| int TransmitMixer::StopRecordingMicrophone() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StopRecordingMicrophone()"); |
| |
| if (!_fileRecording) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "StopRecordingMicrophone() isnot recording"); |
| return 0; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| if (_fileRecorderPtr->StopRecording() != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_STOP_RECORDING_FAILED, kTraceError, |
| "StopRecording(), could not stop recording"); |
| return -1; |
| } |
| _fileRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_fileRecorderPtr); |
| _fileRecorderPtr = NULL; |
| _fileRecording = false; |
| |
| return 0; |
| } |
| |
| int TransmitMixer::StartRecordingCall(const char* fileName, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StartRecordingCall(fileName=%s)", fileName); |
| |
| if (_fileCallRecording) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "StartRecordingCall() is already recording"); |
| return 0; |
| } |
| |
| FileFormats format; |
| const uint32_t notificationTime(0); // Not supported in VoE |
| CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| |
| if (codecInst != NULL && codecInst->channels != 1) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_ARGUMENT, kTraceError, |
| "StartRecordingCall() invalid compression"); |
| return (-1); |
| } |
| if (codecInst == NULL) |
| { |
| format = kFileFormatPcm16kHzFile; |
| codecInst = &dummyCodec; |
| } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| { |
| format = kFileFormatWavFile; |
| } else |
| { |
| format = kFileFormatCompressedFile; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| // Destroy the old instance |
| if (_fileCallRecorderPtr) |
| { |
| _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| _fileCallRecorderPtr = NULL; |
| } |
| |
| _fileCallRecorderPtr |
| = FileRecorder::CreateFileRecorder(_fileCallRecorderId, |
| (const FileFormats) format); |
| if (_fileCallRecorderPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceError, |
| "StartRecordingCall() fileRecorder format isnot correct"); |
| return -1; |
| } |
| |
| if (_fileCallRecorderPtr->StartRecordingAudioFile( |
| fileName, |
| (const CodecInst&) *codecInst, |
| notificationTime) != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_FILE, kTraceError, |
| "StartRecordingAudioFile() failed to start file recording"); |
| _fileCallRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| _fileCallRecorderPtr = NULL; |
| return -1; |
| } |
| _fileCallRecorderPtr->RegisterModuleFileCallback(this); |
| _fileCallRecording = true; |
| |
| return 0; |
| } |
| |
| int TransmitMixer::StartRecordingCall(OutStream* stream, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StartRecordingCall()"); |
| |
| if (_fileCallRecording) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "StartRecordingCall() is already recording"); |
| return 0; |
| } |
| |
| FileFormats format; |
| const uint32_t notificationTime(0); // Not supported in VoE |
| CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 }; |
| |
| if (codecInst != NULL && codecInst->channels != 1) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_ARGUMENT, kTraceError, |
| "StartRecordingCall() invalid compression"); |
| return (-1); |
| } |
| if (codecInst == NULL) |
| { |
| format = kFileFormatPcm16kHzFile; |
| codecInst = &dummyCodec; |
| } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) || |
| (STR_CASE_CMP(codecInst->plname,"PCMA") == 0)) |
| { |
| format = kFileFormatWavFile; |
| } else |
| { |
| format = kFileFormatCompressedFile; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| // Destroy the old instance |
| if (_fileCallRecorderPtr) |
| { |
| _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| _fileCallRecorderPtr = NULL; |
| } |
| |
| _fileCallRecorderPtr = |
| FileRecorder::CreateFileRecorder(_fileCallRecorderId, |
| (const FileFormats) format); |
| if (_fileCallRecorderPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceError, |
| "StartRecordingCall() fileRecorder format isnot correct"); |
| return -1; |
| } |
| |
| if (_fileCallRecorderPtr->StartRecordingAudioFile(*stream, |
| *codecInst, |
| notificationTime) != 0) |
| { |
| _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| "StartRecordingAudioFile() failed to start file recording"); |
| _fileCallRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| _fileCallRecorderPtr = NULL; |
| return -1; |
| } |
| |
| _fileCallRecorderPtr->RegisterModuleFileCallback(this); |
| _fileCallRecording = true; |
| |
| return 0; |
| } |
| |
| int TransmitMixer::StopRecordingCall() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::StopRecordingCall()"); |
| |
| if (!_fileCallRecording) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| "StopRecordingCall() file isnot recording"); |
| return -1; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| if (_fileCallRecorderPtr->StopRecording() != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_STOP_RECORDING_FAILED, kTraceError, |
| "StopRecording(), could not stop recording"); |
| return -1; |
| } |
| |
| _fileCallRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr); |
| _fileCallRecorderPtr = NULL; |
| _fileCallRecording = false; |
| |
| return 0; |
| } |
| |
| void |
| TransmitMixer::SetMixWithMicStatus(bool mix) |
| { |
| _mixFileWithMicrophone = mix; |
| } |
| |
| int TransmitMixer::RegisterExternalMediaProcessing( |
| VoEMediaProcess* object, |
| ProcessingTypes type) { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::RegisterExternalMediaProcessing()"); |
| |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (!object) { |
| return -1; |
| } |
| |
| // Store the callback object according to the processing type. |
| if (type == kRecordingAllChannelsMixed) { |
| external_postproc_ptr_ = object; |
| } else if (type == kRecordingPreprocessing) { |
| external_preproc_ptr_ = object; |
| } else { |
| return -1; |
| } |
| return 0; |
| } |
| |
| int TransmitMixer::DeRegisterExternalMediaProcessing(ProcessingTypes type) { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::DeRegisterExternalMediaProcessing()"); |
| |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (type == kRecordingAllChannelsMixed) { |
| external_postproc_ptr_ = NULL; |
| } else if (type == kRecordingPreprocessing) { |
| external_preproc_ptr_ = NULL; |
| } else { |
| return -1; |
| } |
| return 0; |
| } |
| |
| int |
| TransmitMixer::SetMute(bool enable) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::SetMute(enable=%d)", enable); |
| _mute = enable; |
| return 0; |
| } |
| |
| bool |
| TransmitMixer::Mute() const |
| { |
| return _mute; |
| } |
| |
| int8_t TransmitMixer::AudioLevel() const |
| { |
| // Speech + file level [0,9] |
| return _audioLevel.Level(); |
| } |
| |
| int16_t TransmitMixer::AudioLevelFullRange() const |
| { |
| // Speech + file level [0,32767] |
| return _audioLevel.LevelFullRange(); |
| } |
| |
| bool TransmitMixer::IsRecordingCall() |
| { |
| return _fileCallRecording; |
| } |
| |
| bool TransmitMixer::IsRecordingMic() |
| { |
| |
| return _fileRecording; |
| } |
| |
| // TODO(andrew): use RemixAndResample for this. |
| int TransmitMixer::GenerateAudioFrame(const int16_t audio[], |
| int samples_per_channel, |
| int num_channels, |
| int sample_rate_hz) { |
| int destination_rate; |
| int num_codec_channels; |
| GetSendCodecInfo(&destination_rate, &num_codec_channels); |
| |
| // Never upsample the capture signal here. This should be done at the |
| // end of the send chain. |
| destination_rate = std::min(destination_rate, sample_rate_hz); |
| stereo_codec_ = num_codec_channels == 2; |
| |
| const int16_t* audio_ptr = audio; |
| int16_t mono_audio[kMaxMonoDeviceDataSizeSamples]; |
| assert(samples_per_channel <= kMaxMonoDeviceDataSizeSamples); |
| // If no stereo codecs are in use, we downmix a stereo stream from the |
| // device early in the chain, before resampling. |
| if (num_channels == 2 && !stereo_codec_) { |
| AudioFrameOperations::StereoToMono(audio, samples_per_channel, |
| mono_audio); |
| audio_ptr = mono_audio; |
| num_channels = 1; |
| } |
| |
| if (resampler_.InitializeIfNeeded(sample_rate_hz, |
| destination_rate, |
| num_channels) != 0) { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::GenerateAudioFrame() unable to resample"); |
| return -1; |
| } |
| |
| int out_length = resampler_.Resample(audio_ptr, |
| samples_per_channel * num_channels, |
| _audioFrame.data_, |
| AudioFrame::kMaxDataSizeSamples); |
| if (out_length == -1) { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::GenerateAudioFrame() resampling failed"); |
| return -1; |
| } |
| |
| _audioFrame.samples_per_channel_ = out_length / num_channels; |
| _audioFrame.id_ = _instanceId; |
| _audioFrame.timestamp_ = -1; |
| _audioFrame.sample_rate_hz_ = destination_rate; |
| _audioFrame.speech_type_ = AudioFrame::kNormalSpeech; |
| _audioFrame.vad_activity_ = AudioFrame::kVadUnknown; |
| _audioFrame.num_channels_ = num_channels; |
| |
| return 0; |
| } |
| |
| int32_t TransmitMixer::RecordAudioToFile( |
| uint32_t mixingFrequency) |
| { |
| CriticalSectionScoped cs(&_critSect); |
| if (_fileRecorderPtr == NULL) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::RecordAudioToFile() filerecorder doesnot" |
| "exist"); |
| return -1; |
| } |
| |
| if (_fileRecorderPtr->RecordAudioToFile(_audioFrame) != 0) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::RecordAudioToFile() file recording" |
| "failed"); |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| int32_t TransmitMixer::MixOrReplaceAudioWithFile( |
| int mixingFrequency) |
| { |
| scoped_array<int16_t> fileBuffer(new int16_t[640]); |
| |
| int fileSamples(0); |
| { |
| CriticalSectionScoped cs(&_critSect); |
| if (_filePlayerPtr == NULL) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| VoEId(_instanceId, -1), |
| "TransmitMixer::MixOrReplaceAudioWithFile()" |
| "fileplayer doesnot exist"); |
| return -1; |
| } |
| |
| if (_filePlayerPtr->Get10msAudioFromFile(fileBuffer.get(), |
| fileSamples, |
| mixingFrequency) == -1) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "TransmitMixer::MixOrReplaceAudioWithFile() file" |
| " mixing failed"); |
| return -1; |
| } |
| } |
| |
| assert(_audioFrame.samples_per_channel_ == fileSamples); |
| |
| if (_mixFileWithMicrophone) |
| { |
| // Currently file stream is always mono. |
| // TODO(xians): Change the code when FilePlayer supports real stereo. |
| Utility::MixWithSat(_audioFrame.data_, |
| _audioFrame.num_channels_, |
| fileBuffer.get(), |
| 1, |
| fileSamples); |
| } else |
| { |
| // Replace ACM audio with file. |
| // Currently file stream is always mono. |
| // TODO(xians): Change the code when FilePlayer supports real stereo. |
| _audioFrame.UpdateFrame(-1, |
| -1, |
| fileBuffer.get(), |
| fileSamples, |
| mixingFrequency, |
| AudioFrame::kNormalSpeech, |
| AudioFrame::kVadUnknown, |
| 1); |
| } |
| return 0; |
| } |
| |
| void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift, |
| int current_mic_level) { |
| if (audioproc_->set_num_channels(_audioFrame.num_channels_, |
| _audioFrame.num_channels_) != 0) { |
| LOG_FERR2(LS_ERROR, set_num_channels, _audioFrame.num_channels_, |
| _audioFrame.num_channels_); |
| } |
| |
| if (audioproc_->set_sample_rate_hz(_audioFrame.sample_rate_hz_) != 0) { |
| LOG_FERR1(LS_ERROR, set_sample_rate_hz, _audioFrame.sample_rate_hz_); |
| } |
| |
| if (audioproc_->set_stream_delay_ms(delay_ms) != 0) { |
| // Report as a warning; we can occasionally run into very large delays. |
| LOG_FERR1(LS_WARNING, set_stream_delay_ms, delay_ms); |
| } |
| |
| GainControl* agc = audioproc_->gain_control(); |
| if (agc->set_stream_analog_level(current_mic_level) != 0) { |
| LOG_FERR1(LS_ERROR, set_stream_analog_level, current_mic_level); |
| } |
| |
| EchoCancellation* aec = audioproc_->echo_cancellation(); |
| if (aec->is_drift_compensation_enabled()) { |
| aec->set_stream_drift_samples(clock_drift); |
| } |
| |
| int err = audioproc_->ProcessStream(&_audioFrame); |
| if (err != 0) { |
| LOG(LS_ERROR) << "ProcessStream() error: " << err; |
| } |
| |
| CriticalSectionScoped cs(&_critSect); |
| |
| // Store new capture level. Only updated when analog AGC is enabled. |
| _captureLevel = agc->stream_analog_level(); |
| |
| // Triggers a callback in OnPeriodicProcess(). |
| _saturationWarning |= agc->stream_is_saturated(); |
| } |
| |
| #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| int TransmitMixer::TypingDetection(bool keyPressed) |
| { |
| |
| // We let the VAD determine if we're using this feature or not. |
| if (_audioFrame.vad_activity_ == AudioFrame::kVadUnknown) |
| { |
| return (0); |
| } |
| |
| if (_audioFrame.vad_activity_ == AudioFrame::kVadActive) |
| _timeActive++; |
| else |
| _timeActive = 0; |
| |
| // Keep track if time since last typing event |
| if (keyPressed) |
| { |
| _timeSinceLastTyping = 0; |
| } |
| else |
| { |
| ++_timeSinceLastTyping; |
| } |
| |
| if ((_timeSinceLastTyping < _typeEventDelay) |
| && (_audioFrame.vad_activity_ == AudioFrame::kVadActive) |
| && (_timeActive < _timeWindow)) |
| { |
| _penaltyCounter += _costPerTyping; |
| if (_penaltyCounter > _reportingThreshold) |
| { |
| // Triggers a callback in OnPeriodicProcess(). |
| _typingNoiseWarning = true; |
| } |
| } |
| |
| if (_penaltyCounter > 0) |
| _penaltyCounter-=_penaltyDecay; |
| |
| return (0); |
| } |
| #endif |
| |
| int TransmitMixer::GetMixingFrequency() |
| { |
| assert(_audioFrame.sample_rate_hz_ != 0); |
| return _audioFrame.sample_rate_hz_; |
| } |
| |
| #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| int TransmitMixer::TimeSinceLastTyping(int &seconds) |
| { |
| // We check in VoEAudioProcessingImpl that this is only called when |
| // typing detection is active. |
| |
| // Round to whole seconds |
| seconds = (_timeSinceLastTyping + 50) / 100; |
| return(0); |
| } |
| #endif |
| |
| #ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
| int TransmitMixer::SetTypingDetectionParameters(int timeWindow, |
| int costPerTyping, |
| int reportingThreshold, |
| int penaltyDecay, |
| int typeEventDelay) |
| { |
| if(timeWindow != 0) |
| _timeWindow = timeWindow; |
| if(costPerTyping != 0) |
| _costPerTyping = costPerTyping; |
| if(reportingThreshold != 0) |
| _reportingThreshold = reportingThreshold; |
| if(penaltyDecay != 0) |
| _penaltyDecay = penaltyDecay; |
| if(typeEventDelay != 0) |
| _typeEventDelay = typeEventDelay; |
| |
| |
| return(0); |
| } |
| #endif |
| |
| void TransmitMixer::EnableStereoChannelSwapping(bool enable) { |
| swap_stereo_channels_ = enable; |
| } |
| |
| bool TransmitMixer::IsStereoChannelSwappingEnabled() { |
| return swap_stereo_channels_; |
| } |
| |
| } // namespace voe |
| |
| } // namespace webrtc |