clang-format on AudioDeviceBuffer
BUG=NONE R=magjed@webrtc.org Review URL: https://codereview.webrtc.org/2119093003 . Cr-Commit-Position: refs/heads/master@{#13377}
This commit is contained in:
henrika
parent
414dda1a10
commit
0fd6801c3c
@ -28,166 +28,160 @@ static const int kLogHighDelayIntervalFrames = 500; // 5 seconds.
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// ctor
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// ----------------------------------------------------------------------------
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AudioDeviceBuffer::AudioDeviceBuffer() :
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_id(-1),
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_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
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_critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
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_ptrCbAudioTransport(NULL),
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_recSampleRate(0),
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_playSampleRate(0),
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_recChannels(0),
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_playChannels(0),
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_recChannel(AudioDeviceModule::kChannelBoth),
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_recBytesPerSample(0),
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_playBytesPerSample(0),
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_recSamples(0),
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_recSize(0),
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_playSamples(0),
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_playSize(0),
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_recFile(*FileWrapper::Create()),
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_playFile(*FileWrapper::Create()),
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_currentMicLevel(0),
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_newMicLevel(0),
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_typingStatus(false),
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_playDelayMS(0),
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_recDelayMS(0),
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_clockDrift(0),
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// Set to the interval in order to log on the first occurrence.
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high_delay_counter_(kLogHighDelayIntervalFrames) {
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// valid ID will be set later by SetId, use -1 for now
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created", __FUNCTION__);
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memset(_recBuffer, 0, kMaxBufferSizeBytes);
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memset(_playBuffer, 0, kMaxBufferSizeBytes);
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AudioDeviceBuffer::AudioDeviceBuffer()
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: _id(-1),
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_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
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_critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
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_ptrCbAudioTransport(NULL),
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_recSampleRate(0),
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_playSampleRate(0),
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_recChannels(0),
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_playChannels(0),
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_recChannel(AudioDeviceModule::kChannelBoth),
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_recBytesPerSample(0),
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_playBytesPerSample(0),
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_recSamples(0),
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_recSize(0),
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_playSamples(0),
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_playSize(0),
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_recFile(*FileWrapper::Create()),
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_playFile(*FileWrapper::Create()),
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_currentMicLevel(0),
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_newMicLevel(0),
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_typingStatus(false),
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_playDelayMS(0),
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_recDelayMS(0),
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_clockDrift(0),
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// Set to the interval in order to log on the first occurrence.
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high_delay_counter_(kLogHighDelayIntervalFrames) {
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// valid ID will be set later by SetId, use -1 for now
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created",
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__FUNCTION__);
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memset(_recBuffer, 0, kMaxBufferSizeBytes);
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memset(_playBuffer, 0, kMaxBufferSizeBytes);
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}
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// ----------------------------------------------------------------------------
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// dtor
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// ----------------------------------------------------------------------------
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AudioDeviceBuffer::~AudioDeviceBuffer()
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
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{
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CriticalSectionScoped lock(&_critSect);
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AudioDeviceBuffer::~AudioDeviceBuffer() {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed",
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__FUNCTION__);
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{
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CriticalSectionScoped lock(&_critSect);
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_recFile.Flush();
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_recFile.CloseFile();
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delete &_recFile;
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_recFile.Flush();
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_recFile.CloseFile();
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delete &_recFile;
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_playFile.Flush();
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_playFile.CloseFile();
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delete &_playFile;
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}
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_playFile.Flush();
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_playFile.CloseFile();
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delete &_playFile;
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}
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delete &_critSect;
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delete &_critSectCb;
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delete &_critSect;
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delete &_critSectCb;
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}
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// ----------------------------------------------------------------------------
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// SetId
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// ----------------------------------------------------------------------------
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void AudioDeviceBuffer::SetId(uint32_t id)
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "AudioDeviceBuffer::SetId(id=%d)", id);
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_id = id;
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void AudioDeviceBuffer::SetId(uint32_t id) {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
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"AudioDeviceBuffer::SetId(id=%d)", id);
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_id = id;
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}
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// ----------------------------------------------------------------------------
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// RegisterAudioCallback
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::RegisterAudioCallback(AudioTransport* audioCallback)
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{
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CriticalSectionScoped lock(&_critSectCb);
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_ptrCbAudioTransport = audioCallback;
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int32_t AudioDeviceBuffer::RegisterAudioCallback(
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AudioTransport* audioCallback) {
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CriticalSectionScoped lock(&_critSectCb);
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_ptrCbAudioTransport = audioCallback;
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return 0;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// InitPlayout
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::InitPlayout()
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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return 0;
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int32_t AudioDeviceBuffer::InitPlayout() {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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return 0;
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}
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// ----------------------------------------------------------------------------
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// InitRecording
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::InitRecording()
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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return 0;
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int32_t AudioDeviceBuffer::InitRecording() {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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return 0;
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}
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// ----------------------------------------------------------------------------
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// SetRecordingSampleRate
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz)
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{
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CriticalSectionScoped lock(&_critSect);
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_recSampleRate = fsHz;
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return 0;
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int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
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CriticalSectionScoped lock(&_critSect);
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_recSampleRate = fsHz;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// SetPlayoutSampleRate
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz)
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{
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CriticalSectionScoped lock(&_critSect);
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_playSampleRate = fsHz;
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return 0;
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int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
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CriticalSectionScoped lock(&_critSect);
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_playSampleRate = fsHz;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// RecordingSampleRate
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::RecordingSampleRate() const
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{
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return _recSampleRate;
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int32_t AudioDeviceBuffer::RecordingSampleRate() const {
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return _recSampleRate;
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}
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// ----------------------------------------------------------------------------
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// PlayoutSampleRate
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::PlayoutSampleRate() const
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{
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return _playSampleRate;
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int32_t AudioDeviceBuffer::PlayoutSampleRate() const {
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return _playSampleRate;
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}
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// ----------------------------------------------------------------------------
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// SetRecordingChannels
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels)
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{
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CriticalSectionScoped lock(&_critSect);
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_recChannels = channels;
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_recBytesPerSample = 2*channels; // 16 bits per sample in mono, 32 bits in stereo
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return 0;
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int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
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CriticalSectionScoped lock(&_critSect);
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_recChannels = channels;
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_recBytesPerSample =
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2 * channels; // 16 bits per sample in mono, 32 bits in stereo
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return 0;
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}
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// ----------------------------------------------------------------------------
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// SetPlayoutChannels
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels)
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{
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CriticalSectionScoped lock(&_critSect);
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_playChannels = channels;
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// 16 bits per sample in mono, 32 bits in stereo
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_playBytesPerSample = 2*channels;
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return 0;
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int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
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CriticalSectionScoped lock(&_critSect);
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_playChannels = channels;
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// 16 bits per sample in mono, 32 bits in stereo
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_playBytesPerSample = 2 * channels;
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return 0;
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}
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// ----------------------------------------------------------------------------
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@ -201,88 +195,80 @@ int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels)
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// will be 2 instead of 4 four these cases.
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::SetRecordingChannel(const AudioDeviceModule::ChannelType channel)
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{
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CriticalSectionScoped lock(&_critSect);
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int32_t AudioDeviceBuffer::SetRecordingChannel(
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const AudioDeviceModule::ChannelType channel) {
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CriticalSectionScoped lock(&_critSect);
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if (_recChannels == 1)
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{
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return -1;
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}
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if (_recChannels == 1) {
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return -1;
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}
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if (channel == AudioDeviceModule::kChannelBoth)
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{
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// two bytes per channel
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_recBytesPerSample = 4;
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}
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else
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{
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// only utilize one out of two possible channels (left or right)
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_recBytesPerSample = 2;
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}
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_recChannel = channel;
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if (channel == AudioDeviceModule::kChannelBoth) {
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// two bytes per channel
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_recBytesPerSample = 4;
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} else {
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// only utilize one out of two possible channels (left or right)
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_recBytesPerSample = 2;
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}
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_recChannel = channel;
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return 0;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// RecordingChannel
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& channel) const
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{
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channel = _recChannel;
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return 0;
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int32_t AudioDeviceBuffer::RecordingChannel(
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AudioDeviceModule::ChannelType& channel) const {
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channel = _recChannel;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// RecordingChannels
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// ----------------------------------------------------------------------------
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size_t AudioDeviceBuffer::RecordingChannels() const
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{
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return _recChannels;
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size_t AudioDeviceBuffer::RecordingChannels() const {
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return _recChannels;
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}
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// ----------------------------------------------------------------------------
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// PlayoutChannels
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// ----------------------------------------------------------------------------
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size_t AudioDeviceBuffer::PlayoutChannels() const
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{
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return _playChannels;
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size_t AudioDeviceBuffer::PlayoutChannels() const {
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return _playChannels;
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}
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// ----------------------------------------------------------------------------
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// SetCurrentMicLevel
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level)
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{
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_currentMicLevel = level;
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return 0;
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int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level) {
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_currentMicLevel = level;
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return 0;
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}
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int32_t AudioDeviceBuffer::SetTypingStatus(bool typingStatus)
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{
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_typingStatus = typingStatus;
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return 0;
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int32_t AudioDeviceBuffer::SetTypingStatus(bool typingStatus) {
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_typingStatus = typingStatus;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// NewMicLevel
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// ----------------------------------------------------------------------------
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uint32_t AudioDeviceBuffer::NewMicLevel() const
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{
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return _newMicLevel;
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uint32_t AudioDeviceBuffer::NewMicLevel() const {
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return _newMicLevel;
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}
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// ----------------------------------------------------------------------------
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// SetVQEData
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// ----------------------------------------------------------------------------
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void AudioDeviceBuffer::SetVQEData(int playDelayMs, int recDelayMs,
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void AudioDeviceBuffer::SetVQEData(int playDelayMs,
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int recDelayMs,
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int clockDrift) {
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if (high_delay_counter_ < kLogHighDelayIntervalFrames) {
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++high_delay_counter_;
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@ -304,32 +290,30 @@ void AudioDeviceBuffer::SetVQEData(int playDelayMs, int recDelayMs,
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::StartInputFileRecording(
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const char fileName[kAdmMaxFileNameSize])
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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const char fileName[kAdmMaxFileNameSize]) {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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CriticalSectionScoped lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_recFile.Flush();
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_recFile.CloseFile();
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_recFile.Flush();
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_recFile.CloseFile();
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return _recFile.OpenFile(fileName, false) ? 0 : -1;
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return _recFile.OpenFile(fileName, false) ? 0 : -1;
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}
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// ----------------------------------------------------------------------------
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// StopInputFileRecording
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::StopInputFileRecording()
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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int32_t AudioDeviceBuffer::StopInputFileRecording() {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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CriticalSectionScoped lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_recFile.Flush();
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_recFile.CloseFile();
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_recFile.Flush();
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_recFile.CloseFile();
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return 0;
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return 0;
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}
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// ----------------------------------------------------------------------------
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@ -337,32 +321,30 @@ int32_t AudioDeviceBuffer::StopInputFileRecording()
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::StartOutputFileRecording(
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const char fileName[kAdmMaxFileNameSize])
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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const char fileName[kAdmMaxFileNameSize]) {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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CriticalSectionScoped lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_playFile.Flush();
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_playFile.CloseFile();
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_playFile.Flush();
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_playFile.CloseFile();
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return _playFile.OpenFile(fileName, false) ? 0 : -1;
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return _playFile.OpenFile(fileName, false) ? 0 : -1;
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}
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// ----------------------------------------------------------------------------
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// StopOutputFileRecording
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::StopOutputFileRecording()
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{
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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int32_t AudioDeviceBuffer::StopOutputFileRecording() {
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WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
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CriticalSectionScoped lock(&_critSect);
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CriticalSectionScoped lock(&_critSect);
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_playFile.Flush();
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_playFile.CloseFile();
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_playFile.Flush();
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_playFile.CloseFile();
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return 0;
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return 0;
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}
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// ----------------------------------------------------------------------------
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@ -381,202 +363,175 @@ int32_t AudioDeviceBuffer::StopOutputFileRecording()
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// ----------------------------------------------------------------------------
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int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
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size_t nSamples)
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{
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CriticalSectionScoped lock(&_critSect);
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size_t nSamples) {
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CriticalSectionScoped lock(&_critSect);
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if (_recBytesPerSample == 0)
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{
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assert(false);
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return -1;
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if (_recBytesPerSample == 0) {
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assert(false);
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return -1;
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}
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_recSamples = nSamples;
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_recSize = _recBytesPerSample * nSamples; // {2,4}*nSamples
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if (_recSize > kMaxBufferSizeBytes) {
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assert(false);
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return -1;
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}
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if (_recChannel == AudioDeviceModule::kChannelBoth) {
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// (default) copy the complete input buffer to the local buffer
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memcpy(&_recBuffer[0], audioBuffer, _recSize);
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} else {
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int16_t* ptr16In = (int16_t*)audioBuffer;
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int16_t* ptr16Out = (int16_t*)&_recBuffer[0];
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if (AudioDeviceModule::kChannelRight == _recChannel) {
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ptr16In++;
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}
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_recSamples = nSamples;
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_recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
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if (_recSize > kMaxBufferSizeBytes)
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{
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assert(false);
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return -1;
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// exctract left or right channel from input buffer to the local buffer
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for (size_t i = 0; i < _recSamples; i++) {
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*ptr16Out = *ptr16In;
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ptr16Out++;
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ptr16In++;
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ptr16In++;
|
||||
}
|
||||
}
|
||||
|
||||
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 (_recFile.is_open()) {
|
||||
// write to binary file in mono or stereo (interleaved)
|
||||
_recFile.Write(&_recBuffer[0], _recSize);
|
||||
}
|
||||
|
||||
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);
|
||||
}
|
||||
|
||||
return 0;
|
||||
return 0;
|
||||
}
|
||||
|
||||
// ----------------------------------------------------------------------------
|
||||
// DeliverRecordedData
|
||||
// ----------------------------------------------------------------------------
|
||||
|
||||
int32_t AudioDeviceBuffer::DeliverRecordedData()
|
||||
{
|
||||
CriticalSectionScoped lock(&_critSectCb);
|
||||
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;
|
||||
}
|
||||
// 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;
|
||||
size_t playChannels = 0;
|
||||
{
|
||||
CriticalSectionScoped lock(&_critSect);
|
||||
int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples) {
|
||||
uint32_t playSampleRate = 0;
|
||||
size_t playBytesPerSample = 0;
|
||||
size_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;
|
||||
// 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;
|
||||
}
|
||||
// Ensure that user has initialized all essential members
|
||||
if ((playBytesPerSample == 0) || (playChannels == 0) ||
|
||||
(playSampleRate == 0)) {
|
||||
assert(false);
|
||||
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;
|
||||
_playSamples = nSamples;
|
||||
_playSize = playBytesPerSample * nSamples; // {2,4}*nSamples
|
||||
if (_playSize > kMaxBufferSizeBytes) {
|
||||
assert(false);
|
||||
return -1;
|
||||
}
|
||||
|
||||
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");
|
||||
}
|
||||
if (nSamples != _playSamples) {
|
||||
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
|
||||
"invalid number of samples to be played out (%d)", nSamples);
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
|
||||
return static_cast<int32_t>(nSamplesOut);
|
||||
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);
|
||||
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;
|
||||
}
|
||||
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);
|
||||
memcpy(audioBuffer, &_playBuffer[0], _playSize);
|
||||
|
||||
if (_playFile.is_open()) {
|
||||
// write to binary file in mono or stereo (interleaved)
|
||||
_playFile.Write(&_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);
|
||||
return static_cast<int32_t>(_playSamples);
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
@ -19,103 +19,95 @@ namespace webrtc {
|
||||
class CriticalSectionWrapper;
|
||||
|
||||
const uint32_t kPulsePeriodMs = 1000;
|
||||
const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
|
||||
const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
|
||||
|
||||
class AudioDeviceObserver;
|
||||
|
||||
class AudioDeviceBuffer
|
||||
{
|
||||
public:
|
||||
AudioDeviceBuffer();
|
||||
virtual ~AudioDeviceBuffer();
|
||||
class AudioDeviceBuffer {
|
||||
public:
|
||||
AudioDeviceBuffer();
|
||||
virtual ~AudioDeviceBuffer();
|
||||
|
||||
void SetId(uint32_t id);
|
||||
int32_t RegisterAudioCallback(AudioTransport* audioCallback);
|
||||
void SetId(uint32_t id);
|
||||
int32_t RegisterAudioCallback(AudioTransport* audioCallback);
|
||||
|
||||
int32_t InitPlayout();
|
||||
int32_t InitRecording();
|
||||
int32_t InitPlayout();
|
||||
int32_t InitRecording();
|
||||
|
||||
virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
|
||||
virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
|
||||
int32_t RecordingSampleRate() const;
|
||||
int32_t PlayoutSampleRate() const;
|
||||
virtual int32_t SetRecordingSampleRate(uint32_t fsHz);
|
||||
virtual int32_t SetPlayoutSampleRate(uint32_t fsHz);
|
||||
int32_t RecordingSampleRate() const;
|
||||
int32_t PlayoutSampleRate() const;
|
||||
|
||||
virtual int32_t SetRecordingChannels(size_t channels);
|
||||
virtual int32_t SetPlayoutChannels(size_t channels);
|
||||
size_t RecordingChannels() const;
|
||||
size_t PlayoutChannels() const;
|
||||
int32_t SetRecordingChannel(
|
||||
const AudioDeviceModule::ChannelType channel);
|
||||
int32_t RecordingChannel(
|
||||
AudioDeviceModule::ChannelType& channel) const;
|
||||
virtual int32_t SetRecordingChannels(size_t channels);
|
||||
virtual int32_t SetPlayoutChannels(size_t channels);
|
||||
size_t RecordingChannels() const;
|
||||
size_t PlayoutChannels() const;
|
||||
int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
|
||||
int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
|
||||
|
||||
virtual int32_t SetRecordedBuffer(const void* audioBuffer,
|
||||
size_t nSamples);
|
||||
int32_t SetCurrentMicLevel(uint32_t level);
|
||||
virtual void SetVQEData(int playDelayMS,
|
||||
int recDelayMS,
|
||||
int clockDrift);
|
||||
virtual int32_t DeliverRecordedData();
|
||||
uint32_t NewMicLevel() const;
|
||||
virtual int32_t SetRecordedBuffer(const void* audioBuffer, size_t nSamples);
|
||||
int32_t SetCurrentMicLevel(uint32_t level);
|
||||
virtual void SetVQEData(int playDelayMS, int recDelayMS, int clockDrift);
|
||||
virtual int32_t DeliverRecordedData();
|
||||
uint32_t NewMicLevel() const;
|
||||
|
||||
virtual int32_t RequestPlayoutData(size_t nSamples);
|
||||
virtual int32_t GetPlayoutData(void* audioBuffer);
|
||||
virtual int32_t RequestPlayoutData(size_t nSamples);
|
||||
virtual int32_t GetPlayoutData(void* audioBuffer);
|
||||
|
||||
int32_t StartInputFileRecording(
|
||||
const char fileName[kAdmMaxFileNameSize]);
|
||||
int32_t StopInputFileRecording();
|
||||
int32_t StartOutputFileRecording(
|
||||
const char fileName[kAdmMaxFileNameSize]);
|
||||
int32_t StopOutputFileRecording();
|
||||
int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]);
|
||||
int32_t StopInputFileRecording();
|
||||
int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]);
|
||||
int32_t StopOutputFileRecording();
|
||||
|
||||
int32_t SetTypingStatus(bool typingStatus);
|
||||
int32_t SetTypingStatus(bool typingStatus);
|
||||
|
||||
private:
|
||||
int32_t _id;
|
||||
CriticalSectionWrapper& _critSect;
|
||||
CriticalSectionWrapper& _critSectCb;
|
||||
private:
|
||||
int32_t _id;
|
||||
CriticalSectionWrapper& _critSect;
|
||||
CriticalSectionWrapper& _critSectCb;
|
||||
|
||||
AudioTransport* _ptrCbAudioTransport;
|
||||
AudioTransport* _ptrCbAudioTransport;
|
||||
|
||||
uint32_t _recSampleRate;
|
||||
uint32_t _playSampleRate;
|
||||
uint32_t _recSampleRate;
|
||||
uint32_t _playSampleRate;
|
||||
|
||||
size_t _recChannels;
|
||||
size_t _playChannels;
|
||||
size_t _recChannels;
|
||||
size_t _playChannels;
|
||||
|
||||
// selected recording channel (left/right/both)
|
||||
AudioDeviceModule::ChannelType _recChannel;
|
||||
// selected recording channel (left/right/both)
|
||||
AudioDeviceModule::ChannelType _recChannel;
|
||||
|
||||
// 2 or 4 depending on mono or stereo
|
||||
size_t _recBytesPerSample;
|
||||
size_t _playBytesPerSample;
|
||||
// 2 or 4 depending on mono or stereo
|
||||
size_t _recBytesPerSample;
|
||||
size_t _playBytesPerSample;
|
||||
|
||||
// 10ms in stereo @ 96kHz
|
||||
int8_t _recBuffer[kMaxBufferSizeBytes];
|
||||
// 10ms in stereo @ 96kHz
|
||||
int8_t _recBuffer[kMaxBufferSizeBytes];
|
||||
|
||||
// one sample <=> 2 or 4 bytes
|
||||
size_t _recSamples;
|
||||
size_t _recSize; // in bytes
|
||||
// one sample <=> 2 or 4 bytes
|
||||
size_t _recSamples;
|
||||
size_t _recSize; // in bytes
|
||||
|
||||
// 10ms in stereo @ 96kHz
|
||||
int8_t _playBuffer[kMaxBufferSizeBytes];
|
||||
// 10ms in stereo @ 96kHz
|
||||
int8_t _playBuffer[kMaxBufferSizeBytes];
|
||||
|
||||
// one sample <=> 2 or 4 bytes
|
||||
size_t _playSamples;
|
||||
size_t _playSize; // in bytes
|
||||
// one sample <=> 2 or 4 bytes
|
||||
size_t _playSamples;
|
||||
size_t _playSize; // in bytes
|
||||
|
||||
FileWrapper& _recFile;
|
||||
FileWrapper& _playFile;
|
||||
FileWrapper& _recFile;
|
||||
FileWrapper& _playFile;
|
||||
|
||||
uint32_t _currentMicLevel;
|
||||
uint32_t _newMicLevel;
|
||||
uint32_t _currentMicLevel;
|
||||
uint32_t _newMicLevel;
|
||||
|
||||
bool _typingStatus;
|
||||
bool _typingStatus;
|
||||
|
||||
int _playDelayMS;
|
||||
int _recDelayMS;
|
||||
int _clockDrift;
|
||||
int high_delay_counter_;
|
||||
int _playDelayMS;
|
||||
int _recDelayMS;
|
||||
int _clockDrift;
|
||||
int high_delay_counter_;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user