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Fix play block size mismatch in Win audio device.

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All of the buffer size returned by Windows Core Audio APIs are in unit
of audio frames (which is sample times number of channels), while
WebRTC's AudioDeviceBuffer RequestPlayoutData method takes in samples
per channel (equivalent to frames per channel) but returns number of
audio samples in all the channels. This CL makes sure that we compare
playout block size in frames with frames and size in samples with
samples, which should fix the excessive logging issues and audio quality
problems due to the mismatch when comparing.

BUG=webrtc:7797

Review-Url: https://codereview.webrtc.org/2933953003
Cr-Commit-Position: refs/heads/master@{#18546}
This commit is contained in:
lliuu 2017-06-12 16:54:07 -07:00 committed by Commit Bot
parent 84da736e92
commit c35c7dedc0
2 changed files with 187 additions and 177 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

360
webrtc/modules/audio_device/win/audio_device_core_win.cc
View File

@ -404,165 +404,167 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported()
// AudioDeviceWindowsCore() - ctor
// ----------------------------------------------------------------------------
AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id) :
_comInit(ScopedCOMInitializer::kMTA),
_id(id),
_ptrAudioBuffer(NULL),
_ptrEnumerator(NULL),
_ptrRenderCollection(NULL),
_ptrCaptureCollection(NULL),
_ptrDeviceOut(NULL),
_ptrDeviceIn(NULL),
_ptrClientOut(NULL),
_ptrClientIn(NULL),
_ptrRenderClient(NULL),
_ptrCaptureClient(NULL),
_ptrCaptureVolume(NULL),
_ptrRenderSimpleVolume(NULL),
_dmo(NULL),
_mediaBuffer(NULL),
_builtInAecEnabled(false),
_playAudioFrameSize(0),
_playSampleRate(0),
_playBlockSizePerChannel(0),
_playBlockSize(0),
_playChannels(2),
_sndCardPlayDelay(0),
_sndCardRecDelay(0),
_writtenSamples(0),
_readSamples(0),
_playAcc(0),
_recAudioFrameSize(0),
_recSampleRate(0),
_recBlockSize(0),
_recChannels(2),
_avrtLibrary(NULL),
_winSupportAvrt(false),
_hRenderSamplesReadyEvent(NULL),
_hPlayThread(NULL),
_hCaptureSamplesReadyEvent(NULL),
_hRecThread(NULL),
_hShutdownRenderEvent(NULL),
_hShutdownCaptureEvent(NULL),
_hRenderStartedEvent(NULL),
_hCaptureStartedEvent(NULL),
_hGetCaptureVolumeThread(NULL),
_hSetCaptureVolumeThread(NULL),
_hSetCaptureVolumeEvent(NULL),
_hMmTask(NULL),
_initialized(false),
_recording(false),
_playing(false),
_recIsInitialized(false),
_playIsInitialized(false),
_speakerIsInitialized(false),
_microphoneIsInitialized(false),
_AGC(false),
_playWarning(0),
_playError(0),
_recWarning(0),
_recError(0),
_playBufType(AudioDeviceModule::kAdaptiveBufferSize),
_playBufDelay(80),
_playBufDelayFixed(80),
_usingInputDeviceIndex(false),
_usingOutputDeviceIndex(false),
_inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_outputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_inputDeviceIndex(0),
_outputDeviceIndex(0),
_newMicLevel(0)
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
assert(_comInit.succeeded());
AudioDeviceWindowsCore::AudioDeviceWindowsCore(const int32_t id)
: _comInit(ScopedCOMInitializer::kMTA),
_id(id),
_ptrAudioBuffer(NULL),
_ptrEnumerator(NULL),
_ptrRenderCollection(NULL),
_ptrCaptureCollection(NULL),
_ptrDeviceOut(NULL),
_ptrDeviceIn(NULL),
_ptrClientOut(NULL),
_ptrClientIn(NULL),
_ptrRenderClient(NULL),
_ptrCaptureClient(NULL),
_ptrCaptureVolume(NULL),
_ptrRenderSimpleVolume(NULL),
_dmo(NULL),
_mediaBuffer(NULL),
_builtInAecEnabled(false),
_playAudioFrameSize(0),
_playSampleRate(0),
_playBlockSizeInFrames(0),
_playBlockSizeInSamples(0),
_playChannels(2),
_sndCardPlayDelay(0),
_sndCardRecDelay(0),
_writtenSamples(0),
_readSamples(0),
_playAcc(0),
_recAudioFrameSize(0),
_recSampleRate(0),
_recBlockSize(0),
_recChannels(2),
_avrtLibrary(NULL),
_winSupportAvrt(false),
_hRenderSamplesReadyEvent(NULL),
_hPlayThread(NULL),
_hCaptureSamplesReadyEvent(NULL),
_hRecThread(NULL),
_hShutdownRenderEvent(NULL),
_hShutdownCaptureEvent(NULL),
_hRenderStartedEvent(NULL),
_hCaptureStartedEvent(NULL),
_hGetCaptureVolumeThread(NULL),
_hSetCaptureVolumeThread(NULL),
_hSetCaptureVolumeEvent(NULL),
_hMmTask(NULL),
_initialized(false),
_recording(false),
_playing(false),
_recIsInitialized(false),
_playIsInitialized(false),
_speakerIsInitialized(false),
_microphoneIsInitialized(false),
_AGC(false),
_playWarning(0),
_playError(0),
_recWarning(0),
_recError(0),
_playBufType(AudioDeviceModule::kAdaptiveBufferSize),
_playBufDelay(80),
_playBufDelayFixed(80),
_usingInputDeviceIndex(false),
_usingOutputDeviceIndex(false),
_inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_outputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
_inputDeviceIndex(0),
_outputDeviceIndex(0),
_newMicLevel(0) {
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
assert(_comInit.succeeded());
// Try to load the Avrt DLL
if (!_avrtLibrary)
{
// Get handle to the Avrt DLL module.
_avrtLibrary = LoadLibrary(TEXT("Avrt.dll"));
if (_avrtLibrary)
{
// Handle is valid (should only happen if OS larger than vista & win7).
// Try to get the function addresses.
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() The Avrt DLL module is now loaded");
// Try to load the Avrt DLL
if (!_avrtLibrary) {
// Get handle to the Avrt DLL module.
_avrtLibrary = LoadLibrary(TEXT("Avrt.dll"));
if (_avrtLibrary) {
// Handle is valid (should only happen if OS larger than vista & win7).
// Try to get the function addresses.
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"AudioDeviceWindowsCore::AudioDeviceWindowsCore() The Avrt "
"DLL module is now loaded");
_PAvRevertMmThreadCharacteristics = (PAvRevertMmThreadCharacteristics)GetProcAddress(_avrtLibrary, "AvRevertMmThreadCharacteristics");
_PAvSetMmThreadCharacteristicsA = (PAvSetMmThreadCharacteristicsA)GetProcAddress(_avrtLibrary, "AvSetMmThreadCharacteristicsA");
_PAvSetMmThreadPriority = (PAvSetMmThreadPriority)GetProcAddress(_avrtLibrary, "AvSetMmThreadPriority");
_PAvRevertMmThreadCharacteristics =
(PAvRevertMmThreadCharacteristics)GetProcAddress(
_avrtLibrary, "AvRevertMmThreadCharacteristics");
_PAvSetMmThreadCharacteristicsA =
(PAvSetMmThreadCharacteristicsA)GetProcAddress(
_avrtLibrary, "AvSetMmThreadCharacteristicsA");
_PAvSetMmThreadPriority = (PAvSetMmThreadPriority)GetProcAddress(
_avrtLibrary, "AvSetMmThreadPriority");
if ( _PAvRevertMmThreadCharacteristics &&
_PAvSetMmThreadCharacteristicsA &&
_PAvSetMmThreadPriority)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvRevertMmThreadCharacteristics() is OK");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvSetMmThreadCharacteristicsA() is OK");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvSetMmThreadPriority() is OK");
_winSupportAvrt = true;
}
}
if (_PAvRevertMmThreadCharacteristics &&
_PAvSetMmThreadCharacteristicsA && _PAvSetMmThreadPriority) {
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"AudioDeviceWindowsCore::AudioDeviceWindowsCore() "
"AvRevertMmThreadCharacteristics() is OK");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"AudioDeviceWindowsCore::AudioDeviceWindowsCore() "
"AvSetMmThreadCharacteristicsA() is OK");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"AudioDeviceWindowsCore::AudioDeviceWindowsCore() "
"AvSetMmThreadPriority() is OK");
_winSupportAvrt = true;
}
}
}
// Create our samples ready events - we want auto reset events that start in the not-signaled state.
// The state of an auto-reset event object remains signaled until a single waiting thread is released,
// at which time the system automatically sets the state to nonsignaled. If no threads are waiting,
// the event object's state remains signaled.
// (Except for _hShutdownCaptureEvent, which is used to shutdown multiple threads).
_hRenderSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hCaptureSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hShutdownRenderEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hShutdownCaptureEvent = CreateEvent(NULL, TRUE, FALSE, NULL);
_hRenderStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hCaptureStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
// Create our samples ready events - we want auto reset events that start in
// the not-signaled state. The state of an auto-reset event object remains
// signaled until a single waiting thread is released, at which time the
// system automatically sets the state to nonsignaled. If no threads are
// waiting, the event object's state remains signaled. (Except for
// _hShutdownCaptureEvent, which is used to shutdown multiple threads).
_hRenderSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hCaptureSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hShutdownRenderEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hShutdownCaptureEvent = CreateEvent(NULL, TRUE, FALSE, NULL);
_hRenderStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hCaptureStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
_perfCounterFreq.QuadPart = 1;
_perfCounterFactor = 0.0;
_avgCPULoad = 0.0;
_perfCounterFreq.QuadPart = 1;
_perfCounterFactor = 0.0;
_avgCPULoad = 0.0;
// list of number of channels to use on recording side
_recChannelsPrioList[0] = 2; // stereo is prio 1
_recChannelsPrioList[1] = 1; // mono is prio 2
_recChannelsPrioList[2] = 4; // quad is prio 3
// list of number of channels to use on recording side
_recChannelsPrioList[0] = 2; // stereo is prio 1
_recChannelsPrioList[1] = 1; // mono is prio 2
_recChannelsPrioList[2] = 4; // quad is prio 3
// list of number of channels to use on playout side
_playChannelsPrioList[0] = 2; // stereo is prio 1
_playChannelsPrioList[1] = 1; // mono is prio 2
// list of number of channels to use on playout side
_playChannelsPrioList[0] = 2; // stereo is prio 1
_playChannelsPrioList[1] = 1; // mono is prio 2
HRESULT hr;
HRESULT hr;
// We know that this API will work since it has already been verified in
// CoreAudioIsSupported, hence no need to check for errors here as well.
// We know that this API will work since it has already been verified in
// CoreAudioIsSupported, hence no need to check for errors here as well.
// Retrive the IMMDeviceEnumerator API (should load the MMDevAPI.dll)
// TODO(henrika): we should probably move this allocation to Init() instead
// and deallocate in Terminate() to make the implementation more symmetric.
CoCreateInstance(
__uuidof(MMDeviceEnumerator),
NULL,
CLSCTX_ALL,
__uuidof(IMMDeviceEnumerator),
reinterpret_cast<void**>(&_ptrEnumerator));
assert(NULL != _ptrEnumerator);
// Retrive the IMMDeviceEnumerator API (should load the MMDevAPI.dll)
// TODO(henrika): we should probably move this allocation to Init() instead
// and deallocate in Terminate() to make the implementation more symmetric.
CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL,
__uuidof(IMMDeviceEnumerator),
reinterpret_cast<void**>(&_ptrEnumerator));
assert(NULL != _ptrEnumerator);
// DMO initialization for built-in WASAPI AEC.
{
IMediaObject* ptrDMO = NULL;
hr = CoCreateInstance(CLSID_CWMAudioAEC,
NULL,
CLSCTX_INPROC_SERVER,
IID_IMediaObject,
reinterpret_cast<void**>(&ptrDMO));
if (FAILED(hr) || ptrDMO == NULL)
{
// Since we check that _dmo is non-NULL in EnableBuiltInAEC(), the
// feature is prevented from being enabled.
_builtInAecEnabled = false;
_TraceCOMError(hr);
}
_dmo = ptrDMO;
SAFE_RELEASE(ptrDMO);
// DMO initialization for built-in WASAPI AEC.
{
IMediaObject* ptrDMO = NULL;
hr = CoCreateInstance(CLSID_CWMAudioAEC, NULL, CLSCTX_INPROC_SERVER,
IID_IMediaObject, reinterpret_cast<void**>(&ptrDMO));
if (FAILED(hr) || ptrDMO == NULL) {
// Since we check that _dmo is non-NULL in EnableBuiltInAEC(), the
// feature is prevented from being enabled.
_builtInAecEnabled = false;
_TraceCOMError(hr);
}
_dmo = ptrDMO;
SAFE_RELEASE(ptrDMO);
}
}
// ----------------------------------------------------------------------------
@ -2283,8 +2285,11 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
if (hr == S_OK)
{
_playAudioFrameSize = Wfx.nBlockAlign;
_playBlockSizePerChannel = Wfx.nSamplesPerSec/100;
_playBlockSize = _playBlockSizePerChannel*Wfx.nChannels;
// Block size in frames is the number of samples each channel in 10ms.
_playBlockSizeInFrames = Wfx.nSamplesPerSec / 100;
// Block size in samples is block size in frames times number of
// channels.
_playBlockSizeInSamples = _playBlockSizeInFrames * Wfx.nChannels;
_playSampleRate = Wfx.nSamplesPerSec;
_devicePlaySampleRate = Wfx.nSamplesPerSec; // The device itself continues to run at 44.1 kHz.
_devicePlayBlockSize = Wfx.nSamplesPerSec/100;
@ -2300,7 +2305,8 @@ int32_t AudioDeviceWindowsCore::InitPlayout()
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", Wfx.cbSize);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Additional settings:");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playAudioFrameSize: %d", _playAudioFrameSize);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playBlockSize : %d", _playBlockSize);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"_playBlockSizeInFrames : %d", _playBlockSizeInFrames);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playChannels : %d", _playChannels);
}
@ -3463,8 +3469,8 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
// Derive initial rendering delay.
// Example: 10*(960/480) + 15 = 20 + 15 = 35ms
//
int playout_delay = 10 * (bufferLength / _playBlockSize) +
(int)((latency + devPeriod) / 10000);
int playout_delay = 10 * (bufferLength / _playBlockSizeInFrames) +
(int)((latency + devPeriod) / 10000);
_sndCardPlayDelay = playout_delay;
_writtenSamples = 0;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
@ -3545,38 +3551,38 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
// WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "#avaliable audio frames = %u", framesAvailable);
// Do we have 10 ms available in the render buffer?
if (framesAvailable < _playBlockSize)
{
// Not enough space in render buffer to store next render packet.
_UnLock();
break;
if (framesAvailable < _playBlockSizeInFrames) {
// Not enough space in render buffer to store next render packet.
_UnLock();
break;
}
// Write n*10ms buffers to the render buffer
const uint32_t n10msBuffers = (framesAvailable / _playBlockSize);
const uint32_t n10msBuffers =
(framesAvailable / _playBlockSizeInFrames);
for (uint32_t n = 0; n < n10msBuffers; n++)
{
// Get pointer (i.e., grab the buffer) to next space in the shared render buffer.
hr = _ptrRenderClient->GetBuffer(_playBlockSize, &pData);
hr =
_ptrRenderClient->GetBuffer(_playBlockSizeInFrames, &pData);
EXIT_ON_ERROR(hr);
QueryPerformanceCounter(&t1); // measure time: START
if (_ptrAudioBuffer)
{
// Request data to be played out (#bytes = _playBlockSize*_audioFrameSize)
_UnLock();
int32_t nSamples =
_ptrAudioBuffer->RequestPlayoutData(
_playBlockSizePerChannel);
_Lock();
// Request data to be played out (#bytes =
// _playBlockSizeInFrames*_audioFrameSize)
_UnLock();
int32_t nSamples = _ptrAudioBuffer->RequestPlayoutData(
_playBlockSizeInFrames);
_Lock();
if (nSamples == -1)
{
_UnLock();
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"failed to read data from render client");
goto Exit;
if (nSamples == -1) {
_UnLock();
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
"failed to read data from render client");
goto Exit;
}
// Sanity check to ensure that essential states are not modified during the unlocked period
@ -3586,9 +3592,12 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
goto Exit;
}
if (nSamples != static_cast<int32_t>(_playBlockSize))
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "nSamples(%d) != _playBlockSize(%d)", nSamples, _playBlockSize);
if (nSamples !=
static_cast<int32_t>(_playBlockSizeInSamples)) {
WEBRTC_TRACE(
kTraceWarning, kTraceAudioDevice, _id,
"nSamples(%d) != _playBlockSizeInSamples(%d)",
nSamples, _playBlockSizeInSamples);
}
// Get the actual (stored) data
@ -3600,12 +3609,13 @@ DWORD AudioDeviceWindowsCore::DoRenderThread()
_playAcc += time;
DWORD dwFlags(0);
hr = _ptrRenderClient->ReleaseBuffer(_playBlockSize, dwFlags);
hr = _ptrRenderClient->ReleaseBuffer(_playBlockSizeInFrames,
dwFlags);
// See http://msdn.microsoft.com/en-us/library/dd316605(VS.85).aspx
// for more details regarding AUDCLNT_E_DEVICE_INVALIDATED.
EXIT_ON_ERROR(hr);
_writtenSamples += _playBlockSize;
_writtenSamples += _playBlockSizeInFrames;
}
// Check the current delay on the playout side.

4
webrtc/modules/audio_device/win/audio_device_core_win.h
View File

@ -313,8 +313,8 @@ private: // thread functions
UINT _playAudioFrameSize;
uint32_t _playSampleRate;
uint32_t _devicePlaySampleRate;
uint32_t _playBlockSizePerChannel;
uint32_t _playBlockSize;
uint32_t _playBlockSizeInFrames;
uint32_t _playBlockSizeInSamples;
uint32_t _devicePlayBlockSize;
uint32_t _playChannels;
uint32_t _sndCardPlayDelay;