// Processing
DWORD CTimeStretchFilter::ThreadProc()
{
Log("CTimeStretchFilter::timestretch thread - starting up - thread ID: %d", m_ThreadId);
SetThreadName(0, "TimeStretchFilter");
AudioSinkCommand command;
CComPtr<IMediaSample> sample;
while (true)
{
m_csResources.Unlock();
HRESULT hr = GetNextSampleOrCommand(&command, &sample.p, INFINITE, &m_hSampleEvents, &m_dwSampleWaitObjects);
m_csResources.Lock();
if (hr == MPAR_S_THREAD_STOPPING)
{
Log("CTimeStretchFilter::timestretch thread - closing down - thread ID: %d", m_ThreadId);
SetEvent(m_hCurrentSampleReleased);
CloseThread();
m_csResources.Unlock();
return 0;
}
else
{
if (command == ASC_Flush)
{
Log("CTimeStretchFilter::timestretch thread - flushing");
m_rtInSampleTime = m_rtNextIncomingSampleTime = 0;
if (m_pNextOutSample)
m_pNextOutSample.Release();
flush();
sample.Release();
SetEvent(m_hCurrentSampleReleased);
}
else if (command == ASC_Pause || command == ASC_Resume)
continue;
else if (sample)
{
BYTE *pMediaBuffer = NULL;
long size = sample->GetActualDataLength();
if (sample->IsDiscontinuity() == S_OK)
{
sample->SetDiscontinuity(false);
m_bDiscontinuity = true;
}
if (CheckSample(sample) == S_FALSE)
{
DeleteMediaType(m_pMediaType);
sample->GetMediaType(&m_pMediaType);
}
CheckStreamContinuity(sample);
m_nSampleNum++;
hr = sample->GetPointer(&pMediaBuffer);
if ((hr == S_OK) && m_pMemAllocator)
{
uint unprocessedSamplesBefore = numUnprocessedSamples();
uint unprocessedSamplesAfter = 0;
UINT32 nFrames = size / m_pOutputFormat->Format.nBlockAlign;
REFERENCE_TIME estimatedSampleDuration = nFrames * UNITS / m_pOutputFormat->Format.nSamplesPerSec;
double bias = m_pClock->GetBias();
double adjustment = m_pClock->Adjustment();
double AVMult = m_pClock->SuggestedAudioMultiplier(estimatedSampleDuration, bias, adjustment);
setTempoInternal(AVMult, 1.0); // this should be the same as previous line, but in future we want to get rid of the 2nd parameter
// Process the sample
putSamplesInternal((const short*)pMediaBuffer, size / m_pOutputFormat->Format.nBlockAlign);
unprocessedSamplesAfter = numUnprocessedSamples();
UINT32 nInFrames = (size / m_pOutputFormat->Format.nBlockAlign) - unprocessedSamplesAfter + unprocessedSamplesBefore;
UINT32 nOutFrames = numSamples();
CreateOutput(nInFrames, nOutFrames, bias, adjustment, AVMult, false);
}
}
}
}
}
// Processing
DWORD CTimeStretchFilter::ThreadProc()
{
Log("CTimeStretchFilter::timestretch thread - starting up - thread ID: %d", m_ThreadId);
SetThreadName(0, "TimeStretchFilter");
AudioSinkCommand command;
CComPtr<IMediaSample> sample;
while (true)
{
m_csResources.Unlock();
HRESULT hr = GetNextSampleOrCommand(&command, &sample.p, INFINITE, &m_hSampleEvents, &m_dwSampleWaitObjects);
m_csResources.Lock();
if (hr == MPAR_S_THREAD_STOPPING)
{
Log("CTimeStretchFilter::timestretch thread - closing down - thread ID: %d", m_ThreadId);
SetEvent(m_hCurrentSampleReleased);
CloseThread();
m_csResources.Unlock();
return 0;
}
else
{
if (command == ASC_Flush)
{
Log("CTimeStretchFilter::timestretch thread - flushing");
m_rtInSampleTime = m_rtNextIncomingSampleTime = 0;
m_rtLastOuputStart = m_rtLastOuputEnd = -1;
if (m_pNextOutSample)
m_pNextOutSample.Release();
flush();
m_pClock->Flush();
sample.Release();
SetEvent(m_hCurrentSampleReleased);
}
else if (command == ASC_Pause || command == ASC_Resume)
continue;
else if (sample)
{
BYTE *pMediaBuffer = NULL;
long size = sample->GetActualDataLength();
if (sample->IsDiscontinuity() == S_OK)
{
sample->SetDiscontinuity(false);
m_bDiscontinuity = true;
}
REFERENCE_TIME rtDrained = 0;
if (CheckSample(sample, &rtDrained) == S_FALSE)
{
DeleteMediaType(m_pMediaType);
sample->GetMediaType(&m_pMediaType);
}
CheckStreamContinuity(sample, rtDrained);
m_nSampleNum++;
hr = sample->GetPointer(&pMediaBuffer);
if ((hr == S_OK) && m_pMemAllocator)
{
REFERENCE_TIME rtStart = 0;
REFERENCE_TIME rtAdjustedStart = 0;
REFERENCE_TIME rtEnd = 0;
REFERENCE_TIME rtAdjustedEnd = 0;
REFERENCE_TIME rtAHwTime = 0;
REFERENCE_TIME rtRCTime = 0;
m_pClock->GetHWTime(&rtRCTime, &rtAHwTime);
sample->GetTime(&rtStart, &rtEnd);
REFERENCE_TIME sampleDuration = rtEnd - rtStart;
uint unprocessedSamplesBefore = numUnprocessedSamples();
uint unprocessedSamplesAfter = 0;
UINT32 nFrames = size / m_pOutputFormat->Format.nBlockAlign;
double bias = m_pClock->GetBias();
double adjustment = m_pClock->Adjustment();
double AVMult = m_pClock->SuggestedAudioMultiplier(rtAHwTime, rtRCTime, bias, adjustment);
setTempoInternal(AVMult, 1.0);
if (m_rtLastOuputEnd == -1)
m_rtLastOuputEnd = rtStart / AVMult - 1;
m_rtLastOuputStart = m_rtLastOuputEnd + 1;
// Process the sample
putSamplesInternal((const short*)pMediaBuffer, size / m_pOutputFormat->Format.nBlockAlign);
unprocessedSamplesAfter = numUnprocessedSamples();
UINT32 nInFrames = (size / m_pOutputFormat->Format.nBlockAlign) - unprocessedSamplesAfter + unprocessedSamplesBefore;
UINT32 nOutFrames = numSamples();
// TODO: Soundtouch can provide less samples than asked (but never more) so a cummulative error is possible. This will not happen over the course of a long TV stint, but could be solved for correctness
// m_rtLastOuputEnd += (nOutFrames + unprocessedSamplesAfter - unprocessedSamplesBefore) * UNITS / m_pOutputFormat->Format.nSamplesPerSec;
//rtStart = m_rtInSampleTime;
rtEnd = rtStart + sampleDuration;
rtAdjustedStart = m_rtLastOuputEnd +1;
rtAdjustedEnd = rtAdjustedStart + sampleDuration / AVMult;
m_rtLastOuputEnd += sampleDuration / AVMult;
CreateOutput(nInFrames, nOutFrames, bias, adjustment, AVMult, false);
m_pClock->AddSample(rtStart, rtAdjustedStart, rtEnd, rtAdjustedEnd);
}
}
}
}
}
