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tweaked dsound callback time info calculations. implemented adc time. resolves #122

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This commit is contained in:
rossb 2011-11-10 05:47:50 +00:00
commit fc14177a05
1 changed files with 21 additions and 4 deletions
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25
src/hostapi/dsound/pa_win_ds.c
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@ -2348,7 +2348,8 @@ static int TimeSlice( PaWinDsStream *stream )
long bytesProcessed;
HRESULT hresult;
double outputLatency = 0;
PaStreamCallbackTimeInfo timeInfo = {0,0,0}; /** @todo implement inputBufferAdcTime */
double inputLatency = 0;
PaStreamCallbackTimeInfo timeInfo = {0,0,0};
/* Input */
LPBYTE lpInBuf1 = NULL;
@ -2377,11 +2378,12 @@ static int TimeSlice( PaWinDsStream *stream )
filled = readPos - stream->readOffset;
if( filled < 0 ) filled += stream->inputBufferSizeBytes; // unwrap offset
bytesFilled = filled;
inputLatency = ((double)bytesFilled) * stream->secondsPerHostByte;
}
// FIXME: what happens if IDirectSoundCaptureBuffer_GetCurrentPosition fails?
framesToXfer = numInFramesReady = bytesFilled / stream->inputFrameSizeBytes;
outputLatency = ((double)bytesFilled) * stream->secondsPerHostByte; // FIXME: this doesn't look right. we're calculating output latency in input branch. also secondsPerHostByte is only initialized for the output stream
/** @todo Check for overflow */
}
@ -2396,6 +2398,14 @@ static int TimeSlice( PaWinDsStream *stream )
/* Check for underflow */
if( stream->outputUnderflowCount != previousUnderflowCount )
stream->callbackFlags |= paOutputUnderflow;
/* We are about to compute audio into the first byte of empty space in the output buffer.
This audio will reach the DAC after all of the current (non-empty) audio
in the buffer has played. Therefore the output time is the current time
plus the time it takes to play the non-empty bytes in the buffer,
computed here:
*/
outputLatency = ((double)(stream->outputBufferSizeBytes - bytesEmpty)) * stream->secondsPerHostByte;
}
/* if it's a full duplex stream, set framesToXfer to the minimum of input and output frames ready */
@ -2411,8 +2421,6 @@ static int TimeSlice( PaWinDsStream *stream )
/* The outputBufferDacTime parameter should indicates the time at which
the first sample of the output buffer is heard at the DACs. */
timeInfo.currentTime = PaUtil_GetTime();
timeInfo.outputBufferDacTime = timeInfo.currentTime + outputLatency; // FIXME: QueryOutputSpace gets the playback position, we could use that (?)
PaUtil_BeginBufferProcessing( &stream->bufferProcessor, &timeInfo, stream->callbackFlags );
stream->callbackFlags = 0;
@ -2420,6 +2428,8 @@ static int TimeSlice( PaWinDsStream *stream )
/* Input */
if( stream->bufferProcessor.inputChannelCount > 0 )
{
timeInfo.inputBufferAdcTime = timeInfo.currentTime - inputLatency;
bytesToXfer = framesToXfer * stream->inputFrameSizeBytes;
hresult = IDirectSoundCaptureBuffer_Lock ( stream->pDirectSoundInputBuffer,
stream->readOffset, bytesToXfer,
@ -2449,6 +2459,13 @@ static int TimeSlice( PaWinDsStream *stream )
/* Output */
if( stream->bufferProcessor.outputChannelCount > 0 )
{
/*
We don't currently add outputLatency here because it appears to produce worse
results than non adding it. Need to do more testing to verify this.
*/
/* timeInfo.outputBufferDacTime = timeInfo.currentTime + outputLatency; */
timeInfo.outputBufferDacTime = timeInfo.currentTime;
bytesToXfer = framesToXfer * stream->outputFrameSizeBytes;
hresult = IDirectSoundBuffer_Lock ( stream->pDirectSoundOutputBuffer,
stream->outputBufferWriteOffsetBytes, bytesToXfer,