| /* |
| * 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 "webrtc/base/bind.h" |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/format_macros.h" |
| #include "webrtc/base/timeutils.h" |
| #include "webrtc/modules/audio_device/audio_device_config.h" |
| |
| namespace webrtc { |
| |
| static const int kHighDelayThresholdMs = 300; |
| static const int kLogHighDelayIntervalFrames = 500; // 5 seconds. |
| |
| static const char kTimerQueueName[] = "AudioDeviceBufferTimer"; |
| |
| // Time between two sucessive calls to LogStats(). |
| static const size_t kTimerIntervalInSeconds = 10; |
| static const size_t kTimerIntervalInMilliseconds = |
| kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec; |
| |
| AudioDeviceBuffer::AudioDeviceBuffer() |
| : _ptrCbAudioTransport(nullptr), |
| task_queue_(kTimerQueueName), |
| timer_has_started_(false), |
| _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), |
| num_stat_reports_(0), |
| rec_callbacks_(0), |
| last_rec_callbacks_(0), |
| play_callbacks_(0), |
| last_play_callbacks_(0), |
| rec_samples_(0), |
| last_rec_samples_(0), |
| play_samples_(0), |
| last_play_samples_(0), |
| last_log_stat_time_(0) { |
| LOG(INFO) << "AudioDeviceBuffer::ctor"; |
| memset(_recBuffer, 0, kMaxBufferSizeBytes); |
| memset(_playBuffer, 0, kMaxBufferSizeBytes); |
| } |
| |
| AudioDeviceBuffer::~AudioDeviceBuffer() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| LOG(INFO) << "AudioDeviceBuffer::~dtor"; |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| delete &_recFile; |
| |
| _playFile.Flush(); |
| _playFile.CloseFile(); |
| delete &_playFile; |
| } |
| |
| int32_t AudioDeviceBuffer::RegisterAudioCallback( |
| AudioTransport* audioCallback) { |
| LOG(INFO) << __FUNCTION__; |
| rtc::CritScope lock(&_critSectCb); |
| _ptrCbAudioTransport = audioCallback; |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::InitPlayout() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| LOG(INFO) << __FUNCTION__; |
| if (!timer_has_started_) { |
| StartTimer(); |
| timer_has_started_ = true; |
| } |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::InitRecording() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| LOG(INFO) << __FUNCTION__; |
| if (!timer_has_started_) { |
| StartTimer(); |
| timer_has_started_ = true; |
| } |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) { |
| LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")"; |
| rtc::CritScope lock(&_critSect); |
| _recSampleRate = fsHz; |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) { |
| LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")"; |
| rtc::CritScope lock(&_critSect); |
| _playSampleRate = fsHz; |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::RecordingSampleRate() const { |
| return _recSampleRate; |
| } |
| |
| int32_t AudioDeviceBuffer::PlayoutSampleRate() const { |
| return _playSampleRate; |
| } |
| |
| int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) { |
| rtc::CritScope lock(&_critSect); |
| _recChannels = channels; |
| _recBytesPerSample = |
| 2 * channels; // 16 bits per sample in mono, 32 bits in stereo |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) { |
| rtc::CritScope lock(&_critSect); |
| _playChannels = channels; |
| // 16 bits per sample in mono, 32 bits in stereo |
| _playBytesPerSample = 2 * channels; |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::SetRecordingChannel( |
| const AudioDeviceModule::ChannelType channel) { |
| rtc::CritScope 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; |
| } |
| |
| int32_t AudioDeviceBuffer::RecordingChannel( |
| AudioDeviceModule::ChannelType& channel) const { |
| channel = _recChannel; |
| return 0; |
| } |
| |
| size_t AudioDeviceBuffer::RecordingChannels() const { |
| return _recChannels; |
| } |
| |
| size_t AudioDeviceBuffer::PlayoutChannels() const { |
| return _playChannels; |
| } |
| |
| int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level) { |
| _currentMicLevel = level; |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::SetTypingStatus(bool typingStatus) { |
| _typingStatus = typingStatus; |
| return 0; |
| } |
| |
| uint32_t AudioDeviceBuffer::NewMicLevel() const { |
| return _newMicLevel; |
| } |
| |
| 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; |
| } |
| |
| int32_t AudioDeviceBuffer::StartInputFileRecording( |
| const char fileName[kAdmMaxFileNameSize]) { |
| rtc::CritScope lock(&_critSect); |
| |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| |
| return _recFile.OpenFile(fileName, false) ? 0 : -1; |
| } |
| |
| int32_t AudioDeviceBuffer::StopInputFileRecording() { |
| rtc::CritScope lock(&_critSect); |
| |
| _recFile.Flush(); |
| _recFile.CloseFile(); |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::StartOutputFileRecording( |
| const char fileName[kAdmMaxFileNameSize]) { |
| rtc::CritScope lock(&_critSect); |
| |
| _playFile.Flush(); |
| _playFile.CloseFile(); |
| |
| return _playFile.OpenFile(fileName, false) ? 0 : -1; |
| } |
| |
| int32_t AudioDeviceBuffer::StopOutputFileRecording() { |
| rtc::CritScope lock(&_critSect); |
| |
| _playFile.Flush(); |
| _playFile.CloseFile(); |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer, |
| size_t nSamples) { |
| rtc::CritScope 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.is_open()) { |
| // write to binary file in mono or stereo (interleaved) |
| _recFile.Write(&_recBuffer[0], _recSize); |
| } |
| |
| // Update some stats but do it on the task queue to ensure that the members |
| // are modified and read on the same thread. |
| task_queue_.PostTask( |
| rtc::Bind(&AudioDeviceBuffer::UpdateRecStats, this, nSamples)); |
| |
| return 0; |
| } |
| |
| int32_t AudioDeviceBuffer::DeliverRecordedData() { |
| rtc::CritScope lock(&_critSectCb); |
| // Ensure that user has initialized all essential members |
| if ((_recSampleRate == 0) || (_recSamples == 0) || |
| (_recBytesPerSample == 0) || (_recChannels == 0)) { |
| RTC_NOTREACHED(); |
| return -1; |
| } |
| |
| if (!_ptrCbAudioTransport) { |
| LOG(LS_WARNING) << "Invalid audio transport"; |
| 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; |
| } |
| |
| int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) { |
| uint32_t playSampleRate = 0; |
| size_t playBytesPerSample = 0; |
| size_t playChannels = 0; |
| |
| // TOOD(henrika): improve bad locking model and make it more clear that only |
| // 10ms buffer sizes is supported in WebRTC. |
| { |
| rtc::CritScope 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)) { |
| RTC_NOTREACHED(); |
| return -1; |
| } |
| |
| _playSamples = nSamples; |
| _playSize = playBytesPerSample * nSamples; // {2,4}*nSamples |
| RTC_CHECK_LE(_playSize, kMaxBufferSizeBytes); |
| RTC_CHECK_EQ(nSamples, _playSamples); |
| } |
| |
| size_t nSamplesOut(0); |
| |
| rtc::CritScope lock(&_critSectCb); |
| |
| // It is currently supported to start playout without a valid audio |
| // transport object. Leads to warning and silence. |
| if (!_ptrCbAudioTransport) { |
| LOG(LS_WARNING) << "Invalid audio transport"; |
| return 0; |
| } |
| |
| 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) { |
| LOG(LS_ERROR) << "NeedMorePlayData() failed"; |
| } |
| |
| // Update some stats but do it on the task queue to ensure that access of |
| // members is serialized hence avoiding usage of locks. |
| task_queue_.PostTask( |
| rtc::Bind(&AudioDeviceBuffer::UpdatePlayStats, this, nSamplesOut)); |
| |
| return static_cast<int32_t>(nSamplesOut); |
| } |
| |
| int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) { |
| rtc::CritScope lock(&_critSect); |
| RTC_CHECK_LE(_playSize, kMaxBufferSizeBytes); |
| |
| memcpy(audioBuffer, &_playBuffer[0], _playSize); |
| |
| if (_playFile.is_open()) { |
| // write to binary file in mono or stereo (interleaved) |
| _playFile.Write(&_playBuffer[0], _playSize); |
| } |
| |
| return static_cast<int32_t>(_playSamples); |
| } |
| |
| void AudioDeviceBuffer::StartTimer() { |
| last_log_stat_time_ = rtc::TimeMillis(); |
| task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this), |
| kTimerIntervalInMilliseconds); |
| } |
| |
| void AudioDeviceBuffer::LogStats() { |
| RTC_DCHECK(task_queue_.IsCurrent()); |
| |
| int64_t now_time = rtc::TimeMillis(); |
| int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds; |
| int64_t time_since_last = rtc::TimeDiff(now_time, last_log_stat_time_); |
| last_log_stat_time_ = now_time; |
| |
| // Log the latest statistics but skip the first 10 seconds since we are not |
| // sure of the exact starting point. I.e., the first log printout will be |
| // after ~20 seconds. |
| if (++num_stat_reports_ > 1) { |
| uint32_t diff_samples = rec_samples_ - last_rec_samples_; |
| uint32_t rate = diff_samples / kTimerIntervalInSeconds; |
| LOG(INFO) << "[REC : " << time_since_last << "msec, " |
| << _recSampleRate / 1000 |
| << "kHz] callbacks: " << rec_callbacks_ - last_rec_callbacks_ |
| << ", " |
| << "samples: " << diff_samples << ", " |
| << "rate: " << rate; |
| |
| diff_samples = play_samples_ - last_play_samples_; |
| rate = diff_samples / kTimerIntervalInSeconds; |
| LOG(INFO) << "[PLAY: " << time_since_last << "msec, " |
| << _playSampleRate / 1000 |
| << "kHz] callbacks: " << play_callbacks_ - last_play_callbacks_ |
| << ", " |
| << "samples: " << diff_samples << ", " |
| << "rate: " << rate; |
| } |
| |
| last_rec_callbacks_ = rec_callbacks_; |
| last_play_callbacks_ = play_callbacks_; |
| last_rec_samples_ = rec_samples_; |
| last_play_samples_ = play_samples_; |
| |
| int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis(); |
| RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval"; |
| |
| // Update some stats but do it on the task queue to ensure that access of |
| // members is serialized hence avoiding usage of locks. |
| task_queue_.PostDelayedTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this), |
| time_to_wait_ms); |
| } |
| |
| void AudioDeviceBuffer::UpdateRecStats(size_t num_samples) { |
| RTC_DCHECK(task_queue_.IsCurrent()); |
| ++rec_callbacks_; |
| rec_samples_ += num_samples; |
| } |
| |
| void AudioDeviceBuffer::UpdatePlayStats(size_t num_samples) { |
| RTC_DCHECK(task_queue_.IsCurrent()); |
| ++play_callbacks_; |
| play_samples_ += num_samples; |
| } |
| |
| } // namespace webrtc |