| /* |
| * 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 "output_mixer.h" |
| |
| #include "audio_processing.h" |
| #include "audio_frame_operations.h" |
| #include "critical_section_wrapper.h" |
| #include "file_wrapper.h" |
| #include "output_mixer_internal.h" |
| #include "statistics.h" |
| #include "trace.h" |
| #include "voe_external_media.h" |
| |
| namespace webrtc { |
| |
| namespace voe { |
| |
| void |
| OutputMixer::NewMixedAudio(const WebRtc_Word32 id, |
| const AudioFrame& generalAudioFrame, |
| const AudioFrame** uniqueAudioFrames, |
| const WebRtc_UWord32 size) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::NewMixedAudio(id=%d, size=%u)", id, size); |
| |
| _audioFrame = generalAudioFrame; |
| _audioFrame.id_ = id; |
| } |
| |
| void OutputMixer::MixedParticipants( |
| const WebRtc_Word32 id, |
| const ParticipantStatistics* participantStatistics, |
| const WebRtc_UWord32 size) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::MixedParticipants(id=%d, size=%u)", id, size); |
| } |
| |
| void OutputMixer::VADPositiveParticipants( |
| const WebRtc_Word32 id, |
| const ParticipantStatistics* participantStatistics, |
| const WebRtc_UWord32 size) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::VADPositiveParticipants(id=%d, size=%u)", |
| id, size); |
| } |
| |
| void OutputMixer::MixedAudioLevel(const WebRtc_Word32 id, |
| const WebRtc_UWord32 level) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::MixedAudioLevel(id=%d, level=%u)", id, level); |
| } |
| |
| void OutputMixer::PlayNotification(const WebRtc_Word32 id, |
| const WebRtc_UWord32 durationMs) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::PlayNotification(id=%d, durationMs=%d)", |
| id, durationMs); |
| // Not implement yet |
| } |
| |
| void OutputMixer::RecordNotification(const WebRtc_Word32 id, |
| const WebRtc_UWord32 durationMs) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::RecordNotification(id=%d, durationMs=%d)", |
| id, durationMs); |
| |
| // Not implement yet |
| } |
| |
| void OutputMixer::PlayFileEnded(const WebRtc_Word32 id) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::PlayFileEnded(id=%d)", id); |
| |
| // not needed |
| } |
| |
| void OutputMixer::RecordFileEnded(const WebRtc_Word32 id) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::RecordFileEnded(id=%d)", id); |
| assert(id == _instanceId); |
| |
| CriticalSectionScoped cs(&_fileCritSect); |
| _outputFileRecording = false; |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::RecordFileEnded() =>" |
| "output file recorder module is shutdown"); |
| } |
| |
| WebRtc_Word32 |
| OutputMixer::Create(OutputMixer*& mixer, const WebRtc_UWord32 instanceId) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId, |
| "OutputMixer::Create(instanceId=%d)", instanceId); |
| mixer = new OutputMixer(instanceId); |
| if (mixer == NULL) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId, |
| "OutputMixer::Create() unable to allocate memory for" |
| "mixer"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| OutputMixer::OutputMixer(const WebRtc_UWord32 instanceId) : |
| _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| _mixerModule(*AudioConferenceMixer::Create(instanceId)), |
| _audioLevel(), |
| _dtmfGenerator(instanceId), |
| _instanceId(instanceId), |
| _externalMediaCallbackPtr(NULL), |
| _externalMedia(false), |
| _panLeft(1.0f), |
| _panRight(1.0f), |
| _mixingFrequencyHz(8000), |
| _outputFileRecorderPtr(NULL), |
| _outputFileRecording(false) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::OutputMixer() - ctor"); |
| |
| if ((_mixerModule.RegisterMixedStreamCallback(*this) == -1) || |
| (_mixerModule.RegisterMixerStatusCallback(*this, 100) == -1)) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::OutputMixer() failed to register mixer" |
| "callbacks"); |
| } |
| |
| _dtmfGenerator.Init(); |
| } |
| |
| void |
| OutputMixer::Destroy(OutputMixer*& mixer) |
| { |
| if (mixer) |
| { |
| delete mixer; |
| mixer = NULL; |
| } |
| } |
| |
| OutputMixer::~OutputMixer() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::~OutputMixer() - dtor"); |
| if (_externalMedia) |
| { |
| DeRegisterExternalMediaProcessing(); |
| } |
| { |
| CriticalSectionScoped cs(&_fileCritSect); |
| if (_outputFileRecorderPtr) |
| { |
| _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| _outputFileRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| _outputFileRecorderPtr = NULL; |
| } |
| } |
| _mixerModule.UnRegisterMixerStatusCallback(); |
| _mixerModule.UnRegisterMixedStreamCallback(); |
| delete &_mixerModule; |
| delete &_callbackCritSect; |
| delete &_fileCritSect; |
| } |
| |
| WebRtc_Word32 |
| OutputMixer::SetEngineInformation(voe::Statistics& engineStatistics) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::SetEngineInformation()"); |
| _engineStatisticsPtr = &engineStatistics; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| OutputMixer::SetAudioProcessingModule( |
| AudioProcessing* audioProcessingModule) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::SetAudioProcessingModule(" |
| "audioProcessingModule=0x%x)", audioProcessingModule); |
| _audioProcessingModulePtr = audioProcessingModule; |
| return 0; |
| } |
| |
| int OutputMixer::RegisterExternalMediaProcessing( |
| VoEMediaProcess& proccess_object) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::RegisterExternalMediaProcessing()"); |
| |
| CriticalSectionScoped cs(&_callbackCritSect); |
| _externalMediaCallbackPtr = &proccess_object; |
| _externalMedia = true; |
| |
| return 0; |
| } |
| |
| int OutputMixer::DeRegisterExternalMediaProcessing() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::DeRegisterExternalMediaProcessing()"); |
| |
| CriticalSectionScoped cs(&_callbackCritSect); |
| _externalMedia = false; |
| _externalMediaCallbackPtr = NULL; |
| |
| return 0; |
| } |
| |
| int OutputMixer::PlayDtmfTone(WebRtc_UWord8 eventCode, int lengthMs, |
| int attenuationDb) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "OutputMixer::PlayDtmfTone()"); |
| if (_dtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb) != 0) |
| { |
| _engineStatisticsPtr->SetLastError(VE_STILL_PLAYING_PREV_DTMF, |
| kTraceError, |
| "OutputMixer::PlayDtmfTone()"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| int OutputMixer::StartPlayingDtmfTone(WebRtc_UWord8 eventCode, |
| int attenuationDb) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "OutputMixer::StartPlayingDtmfTone()"); |
| if (_dtmfGenerator.StartTone(eventCode, attenuationDb) != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_STILL_PLAYING_PREV_DTMF, |
| kTraceError, |
| "OutputMixer::StartPlayingDtmfTone())"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| int OutputMixer::StopPlayingDtmfTone() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "OutputMixer::StopPlayingDtmfTone()"); |
| return (_dtmfGenerator.StopTone()); |
| } |
| |
| WebRtc_Word32 |
| OutputMixer::SetMixabilityStatus(MixerParticipant& participant, |
| const bool mixable) |
| { |
| return _mixerModule.SetMixabilityStatus(participant, mixable); |
| } |
| |
| WebRtc_Word32 |
| OutputMixer::SetAnonymousMixabilityStatus(MixerParticipant& participant, |
| const bool mixable) |
| { |
| return _mixerModule.SetAnonymousMixabilityStatus(participant,mixable); |
| } |
| |
| WebRtc_Word32 |
| OutputMixer::MixActiveChannels() |
| { |
| return _mixerModule.Process(); |
| } |
| |
| int |
| OutputMixer::GetSpeechOutputLevel(WebRtc_UWord32& level) |
| { |
| WebRtc_Word8 currentLevel = _audioLevel.Level(); |
| level = static_cast<WebRtc_UWord32> (currentLevel); |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "GetSpeechOutputLevel() => level=%u", level); |
| return 0; |
| } |
| |
| int |
| OutputMixer::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level) |
| { |
| WebRtc_Word16 currentLevel = _audioLevel.LevelFullRange(); |
| level = static_cast<WebRtc_UWord32> (currentLevel); |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "GetSpeechOutputLevelFullRange() => level=%u", level); |
| return 0; |
| } |
| |
| int |
| OutputMixer::SetOutputVolumePan(float left, float right) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::SetOutputVolumePan()"); |
| _panLeft = left; |
| _panRight = right; |
| return 0; |
| } |
| |
| int |
| OutputMixer::GetOutputVolumePan(float& left, float& right) |
| { |
| left = _panLeft; |
| right = _panRight; |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "GetOutputVolumePan() => left=%2.1f, right=%2.1f", |
| left, right); |
| return 0; |
| } |
| |
| int OutputMixer::StartRecordingPlayout(const char* fileName, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::StartRecordingPlayout(fileName=%s)", fileName); |
| |
| if (_outputFileRecording) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1), |
| "StartRecordingPlayout() is already recording"); |
| return 0; |
| } |
| |
| FileFormats format; |
| const WebRtc_UWord32 notificationTime(0); |
| CodecInst dummyCodec={100,"L16",16000,320,1,320000}; |
| |
| if ((codecInst != NULL) && |
| ((codecInst->channels < 1) || (codecInst->channels > 2))) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_ARGUMENT, kTraceError, |
| "StartRecordingPlayout() 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(&_fileCritSect); |
| |
| // Destroy the old instance |
| if (_outputFileRecorderPtr) |
| { |
| _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| _outputFileRecorderPtr = NULL; |
| } |
| |
| _outputFileRecorderPtr = FileRecorder::CreateFileRecorder( |
| _instanceId, |
| (const FileFormats)format); |
| if (_outputFileRecorderPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceError, |
| "StartRecordingPlayout() fileRecorder format isnot correct"); |
| return -1; |
| } |
| |
| if (_outputFileRecorderPtr->StartRecordingAudioFile( |
| fileName, |
| (const CodecInst&)*codecInst, |
| notificationTime) != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_FILE, kTraceError, |
| "StartRecordingAudioFile() failed to start file recording"); |
| _outputFileRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| _outputFileRecorderPtr = NULL; |
| return -1; |
| } |
| _outputFileRecorderPtr->RegisterModuleFileCallback(this); |
| _outputFileRecording = true; |
| |
| return 0; |
| } |
| |
| int OutputMixer::StartRecordingPlayout(OutStream* stream, |
| const CodecInst* codecInst) |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::StartRecordingPlayout()"); |
| |
| if (_outputFileRecording) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1), |
| "StartRecordingPlayout() is already recording"); |
| return 0; |
| } |
| |
| FileFormats format; |
| const WebRtc_UWord32 notificationTime(0); |
| CodecInst dummyCodec={100,"L16",16000,320,1,320000}; |
| |
| if (codecInst != NULL && codecInst->channels != 1) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_BAD_ARGUMENT, kTraceError, |
| "StartRecordingPlayout() 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(&_fileCritSect); |
| |
| // Destroy the old instance |
| if (_outputFileRecorderPtr) |
| { |
| _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| _outputFileRecorderPtr = NULL; |
| } |
| |
| _outputFileRecorderPtr = FileRecorder::CreateFileRecorder( |
| _instanceId, |
| (const FileFormats)format); |
| if (_outputFileRecorderPtr == NULL) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_INVALID_ARGUMENT, kTraceError, |
| "StartRecordingPlayout() fileRecorder format isnot correct"); |
| return -1; |
| } |
| |
| if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, |
| *codecInst, |
| notificationTime) != 0) |
| { |
| _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError, |
| "StartRecordingAudioFile() failed to start file recording"); |
| _outputFileRecorderPtr->StopRecording(); |
| FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| _outputFileRecorderPtr = NULL; |
| return -1; |
| } |
| |
| _outputFileRecorderPtr->RegisterModuleFileCallback(this); |
| _outputFileRecording = true; |
| |
| return 0; |
| } |
| |
| int OutputMixer::StopRecordingPlayout() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::StopRecordingPlayout()"); |
| |
| if (!_outputFileRecording) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1), |
| "StopRecordingPlayout() file isnot recording"); |
| return -1; |
| } |
| |
| CriticalSectionScoped cs(&_fileCritSect); |
| |
| if (_outputFileRecorderPtr->StopRecording() != 0) |
| { |
| _engineStatisticsPtr->SetLastError( |
| VE_STOP_RECORDING_FAILED, kTraceError, |
| "StopRecording(), could not stop recording"); |
| return -1; |
| } |
| _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); |
| FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr); |
| _outputFileRecorderPtr = NULL; |
| _outputFileRecording = false; |
| |
| return 0; |
| } |
| |
| int OutputMixer::GetMixedAudio(int sample_rate_hz, |
| int num_channels, |
| AudioFrame* frame) { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%d)", |
| sample_rate_hz, num_channels); |
| |
| // --- Record playout if enabled |
| { |
| CriticalSectionScoped cs(&_fileCritSect); |
| if (_outputFileRecording && _outputFileRecorderPtr) |
| _outputFileRecorderPtr->RecordAudioToFile(_audioFrame); |
| } |
| |
| frame->num_channels_ = num_channels; |
| frame->sample_rate_hz_ = sample_rate_hz; |
| // TODO(andrew): Ideally the downmixing would occur much earlier, in |
| // AudioCodingModule. |
| return RemixAndResample(_audioFrame, &_resampler, frame); |
| } |
| |
| WebRtc_Word32 |
| OutputMixer::DoOperationsOnCombinedSignal() |
| { |
| if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1), |
| "OutputMixer::DoOperationsOnCombinedSignal() => " |
| "mixing frequency = %d", _audioFrame.sample_rate_hz_); |
| _mixingFrequencyHz = _audioFrame.sample_rate_hz_; |
| } |
| |
| // --- Insert inband Dtmf tone |
| if (_dtmfGenerator.IsAddingTone()) |
| { |
| InsertInbandDtmfTone(); |
| } |
| |
| // Scale left and/or right channel(s) if balance is active |
| if (_panLeft != 1.0 || _panRight != 1.0) |
| { |
| if (_audioFrame.num_channels_ == 1) |
| { |
| AudioFrameOperations::MonoToStereo(&_audioFrame); |
| } |
| else |
| { |
| // Pure stereo mode (we are receiving a stereo signal). |
| } |
| |
| assert(_audioFrame.num_channels_ == 2); |
| AudioFrameOperations::Scale(_panLeft, _panRight, _audioFrame); |
| } |
| |
| // --- Far-end Voice Quality Enhancement (AudioProcessing Module) |
| |
| APMAnalyzeReverseStream(); |
| |
| // --- External media processing |
| |
| if (_externalMedia) |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| const bool isStereo = (_audioFrame.num_channels_ == 2); |
| if (_externalMediaCallbackPtr) |
| { |
| _externalMediaCallbackPtr->Process( |
| -1, |
| kPlaybackAllChannelsMixed, |
| (WebRtc_Word16*)_audioFrame.data_, |
| _audioFrame.samples_per_channel_, |
| _audioFrame.sample_rate_hz_, |
| isStereo); |
| } |
| } |
| |
| // --- Measure audio level (0-9) for the combined signal |
| _audioLevel.ComputeLevel(_audioFrame); |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // Private methods |
| // ---------------------------------------------------------------------------- |
| |
| void OutputMixer::APMAnalyzeReverseStream() { |
| // Convert from mixing to AudioProcessing sample rate, determined by the send |
| // side. Downmix to mono. |
| AudioFrame frame; |
| frame.num_channels_ = 1; |
| frame.sample_rate_hz_ = _audioProcessingModulePtr->sample_rate_hz(); |
| if (RemixAndResample(_audioFrame, &_apmResampler, &frame) == -1) |
| return; |
| |
| if (_audioProcessingModulePtr->AnalyzeReverseStream(&frame) == -1) { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1), |
| "AudioProcessingModule::AnalyzeReverseStream() => error"); |
| } |
| } |
| |
| int |
| OutputMixer::InsertInbandDtmfTone() |
| { |
| WebRtc_UWord16 sampleRate(0); |
| _dtmfGenerator.GetSampleRate(sampleRate); |
| if (sampleRate != _audioFrame.sample_rate_hz_) |
| { |
| // Update sample rate of Dtmf tone since the mixing frequency changed. |
| _dtmfGenerator.SetSampleRate( |
| (WebRtc_UWord16)(_audioFrame.sample_rate_hz_)); |
| // Reset the tone to be added taking the new sample rate into account. |
| _dtmfGenerator.ResetTone(); |
| } |
| |
| WebRtc_Word16 toneBuffer[320]; |
| WebRtc_UWord16 toneSamples(0); |
| if (_dtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "OutputMixer::InsertInbandDtmfTone() inserting Dtmf" |
| "tone failed"); |
| return -1; |
| } |
| |
| // replace mixed audio with Dtmf tone |
| if (_audioFrame.num_channels_ == 1) |
| { |
| // mono |
| memcpy(_audioFrame.data_, toneBuffer, sizeof(WebRtc_Word16) |
| * toneSamples); |
| } else |
| { |
| // stereo |
| for (int i = 0; i < _audioFrame.samples_per_channel_; i++) |
| { |
| _audioFrame.data_[2 * i] = toneBuffer[i]; |
| _audioFrame.data_[2 * i + 1] = 0; |
| } |
| } |
| assert(_audioFrame.samples_per_channel_ == toneSamples); |
| |
| return 0; |
| } |
| |
| } // namespace voe |
| |
| } // namespace webrtc |
