| /* |
| * 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 "trace.h" |
| #include "critical_section_wrapper.h" |
| #include "audio_device_buffer.h" |
| #include "audio_device_utility.h" |
| #include "audio_device_config.h" |
| |
| #include <stdlib.h> |
| #include <string.h> |
| #include <cassert> |
| |
| #include "signal_processing_library.h" |
| |
| namespace webrtc { |
| |
| // ---------------------------------------------------------------------------- |
| // ctor |
| // ---------------------------------------------------------------------------- |
| |
| AudioDeviceBuffer::AudioDeviceBuffer() : |
| _id(-1), |
| _critSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()), |
| _ptrCbAudioTransport(NULL), |
| _recSampleRate(0), |
| _playSampleRate(0), |
| _recChannels(0), |
| _playChannels(0), |
| _recChannel(AudioDeviceModule::kChannelBoth), |
| _recBytesPerSample(0), |
| _playBytesPerSample(0), |
| _recSamples(0), |
| _recSize(0), |
| _playSamples(0), |
| _playSize(0), |
| _recFile(*FileWrapper::Create()), |
| _playFile(*FileWrapper::Create()), |
| _currentMicLevel(0), |
| _newMicLevel(0), |
| _playDelayMS(0), |
| _recDelayMS(0), |
| _clockDrift(0), |
| _measureDelay(false), // should always be 'false' (EXPERIMENTAL) |
| _pulseList(), |
| _lastPulseTime(AudioDeviceUtility::GetTimeInMS()) |
| { |
| // valid ID will be set later by SetId, use -1 for now |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created", __FUNCTION__); |
| memset(_recBuffer, 0, kMaxBufferSizeBytes); |
| memset(_playBuffer, 0, kMaxBufferSizeBytes); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // dtor |
| // ---------------------------------------------------------------------------- |
| |
| AudioDeviceBuffer::~AudioDeviceBuffer() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__); |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| delete &_recFile; |
| |
| _playFile.Flush(); |
| _playFile.CloseFile(); |
| delete &_playFile; |
| |
| _EmptyList(); |
| } |
| |
| delete &_critSect; |
| delete &_critSectCb; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetId |
| // ---------------------------------------------------------------------------- |
| |
| void AudioDeviceBuffer::SetId(WebRtc_UWord32 id) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "AudioDeviceBuffer::SetId(id=%d)", id); |
| _id = id; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RegisterAudioCallback |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::RegisterAudioCallback(AudioTransport* audioCallback) |
| { |
| |
| CriticalSectionScoped lock(_critSectCb); |
| _ptrCbAudioTransport = audioCallback; |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // InitPlayout |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::InitPlayout() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| if (_measureDelay) |
| { |
| _EmptyList(); |
| _lastPulseTime = AudioDeviceUtility::GetTimeInMS(); |
| } |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // InitRecording |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::InitRecording() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| if (_measureDelay) |
| { |
| _EmptyList(); |
| _lastPulseTime = AudioDeviceUtility::GetTimeInMS(); |
| } |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetRecordingSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetRecordingSampleRate(WebRtc_UWord32 fsHz) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingSampleRate(fsHz=%u)", fsHz); |
| |
| CriticalSectionScoped lock(_critSect); |
| _recSampleRate = fsHz; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetPlayoutSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetPlayoutSampleRate(WebRtc_UWord32 fsHz) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutSampleRate(fsHz=%u)", fsHz); |
| |
| CriticalSectionScoped lock(_critSect); |
| _playSampleRate = fsHz; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RecordingSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::RecordingSampleRate() const |
| { |
| return _recSampleRate; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // PlayoutSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::PlayoutSampleRate() const |
| { |
| return _playSampleRate; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetRecordingChannels |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetRecordingChannels(WebRtc_UWord8 channels) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingChannels(channels=%u)", channels); |
| |
| CriticalSectionScoped lock(_critSect); |
| _recChannels = channels; |
| _recBytesPerSample = 2*channels; // 16 bits per sample in mono, 32 bits in stereo |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetPlayoutChannels |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetPlayoutChannels(WebRtc_UWord8 channels) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutChannels(channels=%u)", channels); |
| |
| CriticalSectionScoped lock(_critSect); |
| _playChannels = channels; |
| // 16 bits per sample in mono, 32 bits in stereo |
| _playBytesPerSample = 2*channels; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetRecordingChannel |
| // |
| // Select which channel to use while recording. |
| // This API requires that stereo is enabled. |
| // |
| // Note that, the nChannel parameter in RecordedDataIsAvailable will be |
| // set to 2 even for kChannelLeft and kChannelRight. However, nBytesPerSample |
| // will be 2 instead of 4 four these cases. |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetRecordingChannel(const AudioDeviceModule::ChannelType channel) |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| if (_recChannels == 1) |
| { |
| return -1; |
| } |
| |
| if (channel == AudioDeviceModule::kChannelBoth) |
| { |
| // two bytes per channel |
| _recBytesPerSample = 4; |
| } |
| else |
| { |
| // only utilize one out of two possible channels (left or right) |
| _recBytesPerSample = 2; |
| } |
| _recChannel = channel; |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RecordingChannel |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& channel) const |
| { |
| channel = _recChannel; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RecordingChannels |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_UWord8 AudioDeviceBuffer::RecordingChannels() const |
| { |
| return _recChannels; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // PlayoutChannels |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_UWord8 AudioDeviceBuffer::PlayoutChannels() const |
| { |
| return _playChannels; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetCurrentMicLevel |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetCurrentMicLevel(WebRtc_UWord32 level) |
| { |
| _currentMicLevel = level; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // NewMicLevel |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_UWord32 AudioDeviceBuffer::NewMicLevel() const |
| { |
| return _newMicLevel; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetVQEData |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetVQEData(WebRtc_UWord32 playDelayMS, WebRtc_UWord32 recDelayMS, WebRtc_Word32 clockDrift) |
| { |
| if ((playDelayMS + recDelayMS) > 300) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "too long delay (play:%i rec:%i)", playDelayMS, recDelayMS, clockDrift); |
| } |
| |
| _playDelayMS = playDelayMS; |
| _recDelayMS = recDelayMS; |
| _clockDrift = clockDrift; |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StartInputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::StartInputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| |
| return (_recFile.OpenFile(fileName, false, false, false)); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StopInputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::StopInputFileRecording() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StartOutputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::StartOutputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| _playFile.Flush(); |
| _playFile.CloseFile(); |
| |
| return (_playFile.OpenFile(fileName, false, false, false)); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StopOutputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::StopOutputFileRecording() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(_critSect); |
| |
| _playFile.Flush(); |
| _playFile.CloseFile(); |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetRecordedBuffer |
| // |
| // Store recorded audio buffer in local memory ready for the actual |
| // "delivery" using a callback. |
| // |
| // This method can also parse out left or right channel from a stereo |
| // input signal, i.e., emulate mono. |
| // |
| // Examples: |
| // |
| // 16-bit,48kHz mono, 10ms => nSamples=480 => _recSize=2*480=960 bytes |
| // 16-bit,48kHz stereo,10ms => nSamples=480 => _recSize=4*480=1920 bytes |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::SetRecordedBuffer(const WebRtc_Word8* audioBuffer, WebRtc_UWord32 nSamples) |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| if (_recBytesPerSample == 0) |
| { |
| assert(false); |
| return -1; |
| } |
| |
| _recSamples = nSamples; |
| _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples |
| if (_recSize > kMaxBufferSizeBytes) |
| { |
| assert(false); |
| return -1; |
| } |
| |
| if (nSamples != _recSamples) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of recorded samples (%d)", nSamples); |
| return -1; |
| } |
| |
| if (_recChannel == AudioDeviceModule::kChannelBoth) |
| { |
| // (default) copy the complete input buffer to the local buffer |
| memcpy(&_recBuffer[0], audioBuffer, _recSize); |
| } |
| else |
| { |
| WebRtc_Word16* ptr16In = (WebRtc_Word16*)audioBuffer; |
| WebRtc_Word16* ptr16Out = (WebRtc_Word16*)&_recBuffer[0]; |
| |
| if (AudioDeviceModule::kChannelRight == _recChannel) |
| { |
| ptr16In++; |
| } |
| |
| // exctract left or right channel from input buffer to the local buffer |
| for (WebRtc_UWord32 i = 0; i < _recSamples; i++) |
| { |
| *ptr16Out = *ptr16In; |
| ptr16Out++; |
| ptr16In++; |
| ptr16In++; |
| } |
| } |
| |
| if (_recFile.Open()) |
| { |
| // write to binary file in mono or stereo (interleaved) |
| _recFile.Write(&_recBuffer[0], _recSize); |
| } |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // DeliverRecordedData |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::DeliverRecordedData() |
| { |
| CriticalSectionScoped lock(_critSectCb); |
| |
| // Ensure that user has initialized all essential members |
| if ((_recSampleRate == 0) || |
| (_recSamples == 0) || |
| (_recBytesPerSample == 0) || |
| (_recChannels == 0)) |
| { |
| assert(false); |
| return -1; |
| } |
| |
| if (_ptrCbAudioTransport == NULL) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to deliver recorded data (AudioTransport does not exist)"); |
| return 0; |
| } |
| |
| WebRtc_Word32 res(0); |
| WebRtc_UWord32 newMicLevel(0); |
| WebRtc_UWord32 totalDelayMS = _playDelayMS +_recDelayMS; |
| |
| if (_measureDelay) |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| memset(&_recBuffer[0], 0, _recSize); |
| WebRtc_UWord32 time = AudioDeviceUtility::GetTimeInMS(); |
| if (time - _lastPulseTime > 500) |
| { |
| _pulseList.PushBack(time); |
| _lastPulseTime = time; |
| |
| WebRtc_Word16* ptr16 = (WebRtc_Word16*)&_recBuffer[0]; |
| *ptr16 = 30000; |
| } |
| } |
| |
| res = _ptrCbAudioTransport->RecordedDataIsAvailable(&_recBuffer[0], |
| _recSamples, |
| _recBytesPerSample, |
| _recChannels, |
| _recSampleRate, |
| totalDelayMS, |
| _clockDrift, |
| _currentMicLevel, |
| newMicLevel); |
| if (res != -1) |
| { |
| _newMicLevel = newMicLevel; |
| } |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RequestPlayoutData |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::RequestPlayoutData(WebRtc_UWord32 nSamples) |
| { |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| // Ensure that user has initialized all essential members |
| if ((_playBytesPerSample == 0) || |
| (_playChannels == 0) || |
| (_playSampleRate == 0)) |
| { |
| assert(false); |
| return -1; |
| } |
| |
| _playSamples = nSamples; |
| _playSize = _playBytesPerSample * nSamples; // {2,4}*nSamples |
| if (_playSize > kMaxBufferSizeBytes) |
| { |
| assert(false); |
| return -1; |
| } |
| |
| if (nSamples != _playSamples) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of samples to be played out (%d)", nSamples); |
| return -1; |
| } |
| } |
| |
| WebRtc_UWord32 nSamplesOut(0); |
| |
| CriticalSectionScoped lock(_critSectCb); |
| |
| if (_ptrCbAudioTransport == NULL) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to feed data to playout (AudioTransport does not exist)"); |
| return 0; |
| } |
| |
| if (_ptrCbAudioTransport) |
| { |
| WebRtc_UWord32 res(0); |
| |
| res = _ptrCbAudioTransport->NeedMorePlayData(_playSamples, |
| _playBytesPerSample, |
| _playChannels, |
| _playSampleRate, |
| &_playBuffer[0], |
| nSamplesOut); |
| if (res != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "NeedMorePlayData() failed"); |
| } |
| |
| // --- Experimental delay-measurement implementation |
| // *** not be used in released code *** |
| |
| if (_measureDelay) |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| WebRtc_Word16 maxAbs = WebRtcSpl_MaxAbsValueW16((const WebRtc_Word16*)&_playBuffer[0], (WebRtc_Word16)nSamplesOut*_playChannels); |
| if (maxAbs > 1000) |
| { |
| WebRtc_UWord32 nowTime = AudioDeviceUtility::GetTimeInMS(); |
| |
| if (!_pulseList.Empty()) |
| { |
| ListItem* item = _pulseList.First(); |
| if (item) |
| { |
| WebRtc_Word16 maxIndex = WebRtcSpl_MaxAbsIndexW16((const WebRtc_Word16*)&_playBuffer[0], (WebRtc_Word16)nSamplesOut*_playChannels); |
| WebRtc_UWord32 pulseTime = item->GetUnsignedItem(); |
| WebRtc_UWord32 diff = nowTime - pulseTime + (10*maxIndex)/(nSamplesOut*_playChannels); |
| WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "diff time in playout delay (%d)", diff); |
| } |
| _pulseList.PopFront(); |
| } |
| } |
| } |
| } |
| |
| return nSamplesOut; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // GetPlayoutData |
| // ---------------------------------------------------------------------------- |
| |
| WebRtc_Word32 AudioDeviceBuffer::GetPlayoutData(WebRtc_Word8* audioBuffer) |
| { |
| CriticalSectionScoped lock(_critSect); |
| |
| if (_playSize > kMaxBufferSizeBytes) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "_playSize %i exceeds " |
| "kMaxBufferSizeBytes in AudioDeviceBuffer::GetPlayoutData", _playSize); |
| assert(false); |
| return -1; |
| } |
| |
| memcpy(audioBuffer, &_playBuffer[0], _playSize); |
| |
| if (_playFile.Open()) |
| { |
| // write to binary file in mono or stereo (interleaved) |
| _playFile.Write(&_playBuffer[0], _playSize); |
| } |
| |
| return _playSamples; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // _EmptyList |
| // ---------------------------------------------------------------------------- |
| |
| void AudioDeviceBuffer::_EmptyList() |
| { |
| while (!_pulseList.Empty()) |
| { |
| ListItem* item = _pulseList.First(); |
| if (item) |
| { |
| // WebRtc_UWord32 ts = item->GetUnsignedItem(); |
| } |
| _pulseList.PopFront(); |
| } |
| } |
| |
| } // namespace webrtc |
