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

src/hostapi/dsound/pa_win_ds.c

@@ -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,