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Fix excessive audio device logging on Windows

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Reverted the logic introduced in https://codereview.webrtc.org/2933953003
This is because the audio device buffer behavior changed with https://codereview.webrtc.org/3009193002, and the RequestPlayoutData method now returns number of samples in each channel, which creates mismatch the reverted CL.

Bug: webrtc:8548
Change-Id: Id4711ca48437ddd3483327c2a4c7827d09e5b770
Reviewed-on: https://webrtc-review.googlesource.com/24122
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20737}
This commit is contained in:
lliuu 2017-11-17 10:26:24 -08:00 committed by Commit Bot
parent b154c27e72
commit dc1e3e8414
2 changed files with 17 additions and 22 deletions
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36
modules/audio_device/win/audio_device_core_win.cc
View File

@ -397,8 +397,7 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
_builtInAecEnabled(false),
_playAudioFrameSize(0),
_playSampleRate(0),
_playBlockSizeInFrames(0),
_playBlockSizeInSamples(0),
_playBlockSize(0),
_playChannels(2),
_sndCardPlayDelay(0),
_sndCardRecDelay(0),
@ -1969,11 +1968,8 @@ int32_t AudioDeviceWindowsCore::InitPlayout() {
// Same in InitRecording().
if (hr == S_OK) {
_playAudioFrameSize = Wfx.nBlockAlign;
// 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;
// Block size is the number of samples each channel in 10ms.
_playBlockSize = Wfx.nSamplesPerSec / 100;
_playSampleRate = Wfx.nSamplesPerSec;
_devicePlaySampleRate =
Wfx.nSamplesPerSec; // The device itself continues to run at 44.1 kHz.
@ -1992,8 +1988,8 @@ int32_t AudioDeviceWindowsCore::InitPlayout() {
RTC_LOG(LS_VERBOSE) << "cbSize : " << Wfx.cbSize;
RTC_LOG(LS_VERBOSE) << "Additional settings:";
RTC_LOG(LS_VERBOSE) << "_playAudioFrameSize: " << _playAudioFrameSize;
RTC_LOG(LS_VERBOSE) << "_playBlockSizeInFrames : "
<< _playBlockSizeInFrames;
RTC_LOG(LS_VERBOSE) << "_playBlockSize : "
<< _playBlockSize;
RTC_LOG(LS_VERBOSE) << "_playChannels : " << _playChannels;
}
@ -2885,7 +2881,7 @@ DWORD AudioDeviceWindowsCore::DoRenderThread() {
// Derive initial rendering delay.
// Example: 10*(960/480) + 15 = 20 + 15 = 35ms
//
int playout_delay = 10 * (bufferLength / _playBlockSizeInFrames) +
int playout_delay = 10 * (bufferLength / _playBlockSize) +
(int)((latency + devPeriod) / 10000);
_sndCardPlayDelay = playout_delay;
_writtenSamples = 0;
@ -2966,26 +2962,26 @@ DWORD AudioDeviceWindowsCore::DoRenderThread() {
uint32_t framesAvailable = bufferLength - padding;
// Do we have 10 ms available in the render buffer?
if (framesAvailable < _playBlockSizeInFrames) {
if (framesAvailable < _playBlockSize) {
// 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 / _playBlockSizeInFrames);
const uint32_t n10msBuffers = (framesAvailable / _playBlockSize);
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(_playBlockSizeInFrames, &pData);
hr = _ptrRenderClient->GetBuffer(_playBlockSize, &pData);
EXIT_ON_ERROR(hr);
if (_ptrAudioBuffer) {
// Request data to be played out (#bytes =
// _playBlockSizeInFrames*_audioFrameSize)
// _playBlockSize*_audioFrameSize)
_UnLock();
int32_t nSamples =
_ptrAudioBuffer->RequestPlayoutData(_playBlockSizeInFrames);
_ptrAudioBuffer->RequestPlayoutData(_playBlockSize);
_Lock();
if (nSamples == -1) {
@ -3003,10 +2999,10 @@ DWORD AudioDeviceWindowsCore::DoRenderThread() {
<< " period";
goto Exit;
}
if (nSamples != static_cast<int32_t>(_playBlockSizeInSamples)) {
if (nSamples != static_cast<int32_t>(_playBlockSize)) {
RTC_LOG(LS_WARNING)
<< "nSamples(" << nSamples << ") != _playBlockSizeInSamples("
<< _playBlockSizeInSamples << ")";
<< "nSamples(" << nSamples << ") != _playBlockSize"
<< _playBlockSize << ")";
}
// Get the actual (stored) data
@ -3014,12 +3010,12 @@ DWORD AudioDeviceWindowsCore::DoRenderThread() {
}
DWORD dwFlags(0);
hr = _ptrRenderClient->ReleaseBuffer(_playBlockSizeInFrames, dwFlags);
hr = _ptrRenderClient->ReleaseBuffer(_playBlockSize, 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 += _playBlockSizeInFrames;
_writtenSamples += _playBlockSize;
}
// Check the current delay on the playout side.

3
modules/audio_device/win/audio_device_core_win.h
View File

@ -283,8 +283,7 @@ private: // thread functions
UINT _playAudioFrameSize;
uint32_t _playSampleRate;
uint32_t _devicePlaySampleRate;
uint32_t _playBlockSizeInFrames;
uint32_t _playBlockSizeInSamples;
uint32_t _playBlockSize;
uint32_t _devicePlayBlockSize;
uint32_t _playChannels;
uint32_t _sndCardPlayDelay;