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Landing 1399004, Minor clean up on the un-used _measureDelay code

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Those code is/will never used, removing it makes the code better.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@3961 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
xians@webrtc.org 2013-05-06 11:52:47 +00:00
parent 59aaebc3cd
commit 233c58de47
2 changed files with 1 additions and 92 deletions
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86
webrtc/modules/audio_device/audio_device_buffer.cc
View File

@ -48,11 +48,7 @@ AudioDeviceBuffer::AudioDeviceBuffer() :
_newMicLevel(0),
_playDelayMS(0),
_recDelayMS(0),
_clockDrift(0),
_measureDelay(false), // should always be 'false' (EXPERIMENTAL)
_pulseList(),
_lastPulseTime(AudioDeviceUtility::GetTimeInMS())
{
_clockDrift(0) {
// valid ID will be set later by SetId, use -1 for now
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created", __FUNCTION__);
memset(_recBuffer, 0, kMaxBufferSizeBytes);
@ -76,8 +72,6 @@ AudioDeviceBuffer::~AudioDeviceBuffer()
_playFile.Flush();
_playFile.CloseFile();
delete &_playFile;
_EmptyList();
}
delete &_critSect;
@ -113,15 +107,6 @@ int32_t AudioDeviceBuffer::RegisterAudioCallback(AudioTransport* audioCallback)
int32_t AudioDeviceBuffer::InitPlayout()
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
CriticalSectionScoped lock(&_critSect);
if (_measureDelay)
{
_EmptyList();
_lastPulseTime = AudioDeviceUtility::GetTimeInMS();
}
return 0;
}
@ -132,15 +117,6 @@ int32_t AudioDeviceBuffer::InitPlayout()
int32_t AudioDeviceBuffer::InitRecording()
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
CriticalSectionScoped lock(&_critSect);
if (_measureDelay)
{
_EmptyList();
_lastPulseTime = AudioDeviceUtility::GetTimeInMS();
}
return 0;
}
@ -485,22 +461,6 @@ int32_t AudioDeviceBuffer::DeliverRecordedData()
uint32_t newMicLevel(0);
uint32_t totalDelayMS = _playDelayMS +_recDelayMS;
if (_measureDelay)
{
CriticalSectionScoped lock(&_critSect);
memset(&_recBuffer[0], 0, _recSize);
uint32_t time = AudioDeviceUtility::GetTimeInMS();
if (time - _lastPulseTime > 500)
{
_pulseList.PushBack(time);
_lastPulseTime = time;
int16_t* ptr16 = (int16_t*)&_recBuffer[0];
*ptr16 = 30000;
}
}
res = _ptrCbAudioTransport->RecordedDataIsAvailable(&_recBuffer[0],
_recSamples,
_recBytesPerSample,
@ -584,33 +544,6 @@ int32_t AudioDeviceBuffer::RequestPlayoutData(uint32_t nSamples)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "NeedMorePlayData() failed");
}
// --- Experimental delay-measurement implementation
// *** not be used in released code ***
if (_measureDelay)
{
CriticalSectionScoped lock(&_critSect);
int16_t maxAbs = WebRtcSpl_MaxAbsValueW16((const int16_t*)&_playBuffer[0], (int16_t)nSamplesOut*_playChannels);
if (maxAbs > 1000)
{
uint32_t nowTime = AudioDeviceUtility::GetTimeInMS();
if (!_pulseList.Empty())
{
ListItem* item = _pulseList.First();
if (item)
{
int16_t maxIndex = WebRtcSpl_MaxAbsIndexW16((const int16_t*)&_playBuffer[0], (int16_t)nSamplesOut*_playChannels);
uint32_t pulseTime = item->GetUnsignedItem();
uint32_t diff = nowTime - pulseTime + (10*maxIndex)/(nSamplesOut*_playChannels);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "diff time in playout delay (%d)", diff);
}
_pulseList.PopFront();
}
}
}
}
return nSamplesOut;
@ -643,21 +576,4 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
return _playSamples;
}
// ----------------------------------------------------------------------------
// _EmptyList
// ----------------------------------------------------------------------------
void AudioDeviceBuffer::_EmptyList()
{
while (!_pulseList.Empty())
{
ListItem* item = _pulseList.First();
if (item)
{
// uint32_t ts = item->GetUnsignedItem();
}
_pulseList.PopFront();
}
}
} // namespace webrtc

7
webrtc/modules/audio_device/audio_device_buffer.h
View File

@ -70,9 +70,6 @@ public:
AudioDeviceBuffer();
~AudioDeviceBuffer();
private:
void _EmptyList();
private:
int32_t _id;
CriticalSectionWrapper& _critSect;
@ -117,10 +114,6 @@ private:
uint32_t _recDelayMS;
int32_t _clockDrift;
bool _measureDelay;
ListWrapper _pulseList;
uint32_t _lastPulseTime;
};
} // namespace webrtc