| /* |
| * 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/modules/audio_device/audio_device_buffer.h" |
| |
| #include <assert.h> |
| #include <string.h> |
| |
| #include "webrtc/base/format_macros.h" |
| #include "webrtc/modules/audio_device/audio_device_config.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/logging.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| |
| namespace webrtc { |
| |
| static const int kHighDelayThresholdMs = 300; |
| static const int kLogHighDelayIntervalFrames = 500; // 5 seconds. |
| |
| // ---------------------------------------------------------------------------- |
| // 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), |
| _typingStatus(false), |
| _playDelayMS(0), |
| _recDelayMS(0), |
| _clockDrift(0), |
| // Set to the interval in order to log on the first occurrence. |
| high_delay_counter_(kLogHighDelayIntervalFrames) { |
| // 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; |
| } |
| |
| delete &_critSect; |
| delete &_critSectCb; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetId |
| // ---------------------------------------------------------------------------- |
| |
| void AudioDeviceBuffer::SetId(uint32_t id) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "AudioDeviceBuffer::SetId(id=%d)", id); |
| _id = id; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RegisterAudioCallback |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::RegisterAudioCallback(AudioTransport* audioCallback) |
| { |
| CriticalSectionScoped lock(&_critSectCb); |
| _ptrCbAudioTransport = audioCallback; |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // InitPlayout |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::InitPlayout() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // InitRecording |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::InitRecording() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetRecordingSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) |
| { |
| CriticalSectionScoped lock(&_critSect); |
| _recSampleRate = fsHz; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetPlayoutSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) |
| { |
| CriticalSectionScoped lock(&_critSect); |
| _playSampleRate = fsHz; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RecordingSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::RecordingSampleRate() const |
| { |
| return _recSampleRate; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // PlayoutSampleRate |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::PlayoutSampleRate() const |
| { |
| return _playSampleRate; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetRecordingChannels |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels) |
| { |
| CriticalSectionScoped lock(&_critSect); |
| _recChannels = channels; |
| _recBytesPerSample = 2*channels; // 16 bits per sample in mono, 32 bits in stereo |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetPlayoutChannels |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::SetPlayoutChannels(uint8_t 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. |
| // ---------------------------------------------------------------------------- |
| |
| int32_t 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 |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& channel) const |
| { |
| channel = _recChannel; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RecordingChannels |
| // ---------------------------------------------------------------------------- |
| |
| uint8_t AudioDeviceBuffer::RecordingChannels() const |
| { |
| return _recChannels; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // PlayoutChannels |
| // ---------------------------------------------------------------------------- |
| |
| uint8_t AudioDeviceBuffer::PlayoutChannels() const |
| { |
| return _playChannels; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetCurrentMicLevel |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level) |
| { |
| _currentMicLevel = level; |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::SetTypingStatus(bool typingStatus) |
| { |
| _typingStatus = typingStatus; |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // NewMicLevel |
| // ---------------------------------------------------------------------------- |
| |
| uint32_t AudioDeviceBuffer::NewMicLevel() const |
| { |
| return _newMicLevel; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // SetVQEData |
| // ---------------------------------------------------------------------------- |
| |
| void AudioDeviceBuffer::SetVQEData(int playDelayMs, int recDelayMs, |
| int clockDrift) { |
| if (high_delay_counter_ < kLogHighDelayIntervalFrames) { |
| ++high_delay_counter_; |
| } else { |
| if (playDelayMs + recDelayMs > kHighDelayThresholdMs) { |
| high_delay_counter_ = 0; |
| LOG(LS_WARNING) << "High audio device delay reported (render=" |
| << playDelayMs << " ms, capture=" << recDelayMs << " ms)"; |
| } |
| } |
| |
| _playDelayMS = playDelayMs; |
| _recDelayMS = recDelayMs; |
| _clockDrift = clockDrift; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StartInputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::StartInputFileRecording( |
| const char fileName[kAdmMaxFileNameSize]) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(&_critSect); |
| |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| |
| return (_recFile.OpenFile(fileName, false, false, false)); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StopInputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::StopInputFileRecording() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(&_critSect); |
| |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StartOutputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::StartOutputFileRecording( |
| const char fileName[kAdmMaxFileNameSize]) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__); |
| |
| CriticalSectionScoped lock(&_critSect); |
| |
| _playFile.Flush(); |
| _playFile.CloseFile(); |
| |
| return (_playFile.OpenFile(fileName, false, false, false)); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // StopOutputFileRecording |
| // ---------------------------------------------------------------------------- |
| |
| int32_t 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 |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer, |
| size_t 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 (_recChannel == AudioDeviceModule::kChannelBoth) |
| { |
| // (default) copy the complete input buffer to the local buffer |
| memcpy(&_recBuffer[0], audioBuffer, _recSize); |
| } |
| else |
| { |
| int16_t* ptr16In = (int16_t*)audioBuffer; |
| int16_t* ptr16Out = (int16_t*)&_recBuffer[0]; |
| |
| if (AudioDeviceModule::kChannelRight == _recChannel) |
| { |
| ptr16In++; |
| } |
| |
| // exctract left or right channel from input buffer to the local buffer |
| for (size_t 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 |
| // ---------------------------------------------------------------------------- |
| |
| int32_t 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; |
| } |
| |
| int32_t res(0); |
| uint32_t newMicLevel(0); |
| uint32_t totalDelayMS = _playDelayMS +_recDelayMS; |
| |
| res = _ptrCbAudioTransport->RecordedDataIsAvailable(&_recBuffer[0], |
| _recSamples, |
| _recBytesPerSample, |
| _recChannels, |
| _recSampleRate, |
| totalDelayMS, |
| _clockDrift, |
| _currentMicLevel, |
| _typingStatus, |
| newMicLevel); |
| if (res != -1) |
| { |
| _newMicLevel = newMicLevel; |
| } |
| |
| return 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // RequestPlayoutData |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) |
| { |
| uint32_t playSampleRate = 0; |
| size_t playBytesPerSample = 0; |
| uint8_t playChannels = 0; |
| { |
| CriticalSectionScoped lock(&_critSect); |
| |
| // Store copies under lock and use copies hereafter to avoid race with |
| // setter methods. |
| playSampleRate = _playSampleRate; |
| playBytesPerSample = _playBytesPerSample; |
| playChannels = _playChannels; |
| |
| // 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; |
| } |
| } |
| |
| size_t 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) |
| { |
| uint32_t res(0); |
| int64_t elapsed_time_ms = -1; |
| int64_t ntp_time_ms = -1; |
| res = _ptrCbAudioTransport->NeedMorePlayData(_playSamples, |
| playBytesPerSample, |
| playChannels, |
| playSampleRate, |
| &_playBuffer[0], |
| nSamplesOut, |
| &elapsed_time_ms, |
| &ntp_time_ms); |
| if (res != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "NeedMorePlayData() failed"); |
| } |
| } |
| |
| return static_cast<int32_t>(nSamplesOut); |
| } |
| |
| // ---------------------------------------------------------------------------- |
| // GetPlayoutData |
| // ---------------------------------------------------------------------------- |
| |
| int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) |
| { |
| CriticalSectionScoped lock(&_critSect); |
| |
| if (_playSize > kMaxBufferSizeBytes) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceUtility, _id, |
| "_playSize %" PRIuS " 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 static_cast<int32_t>(_playSamples); |
| } |
| |
| } // namespace webrtc |
