+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
+// between HW and the user. The convertBufferWasapi function is used to perform this conversion
+// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
+// This sample rate converter works best with conversions between one rate and its multiple.
+void convertBufferWasapi( char* outBuffer,
+ const char* inBuffer,
+ const unsigned int& channelCount,
+ const unsigned int& inSampleRate,
+ const unsigned int& outSampleRate,
+ const unsigned int& inSampleCount,
+ unsigned int& outSampleCount,
+ const RtAudioFormat& format )
+{
+ // calculate the new outSampleCount and relative sampleStep
+ float sampleRatio = ( float ) outSampleRate / inSampleRate;
+ float sampleRatioInv = ( float ) 1 / sampleRatio;
+ float sampleStep = 1.0f / sampleRatio;
+ float inSampleFraction = 0.0f;
+
+ // for cmath functions
+ using namespace std;
+
+ outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
+
+ // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate
+ if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )
+ {
+ // frame-by-frame, copy each relative input sample into it's corresponding output sample
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
+ {
+ unsigned int inSample = ( unsigned int ) inSampleFraction;
+
+ switch ( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
+ break;
+ }
+
+ // jump to next in sample
+ inSampleFraction += sampleStep;
+ }
+ }
+ else // else interpolate
+ {
+ // frame-by-frame, copy each relative input sample into it's corresponding output sample
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
+ {
+ unsigned int inSample = ( unsigned int ) inSampleFraction;
+ float inSampleDec = inSampleFraction - inSample;
+ unsigned int frameInSample = inSample * channelCount;
+ unsigned int frameOutSample = outSample * channelCount;
+
+ switch ( format )
+ {
+ case RTAUDIO_SINT8:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];
+ char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];
+ char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_SINT16:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];
+ short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];
+ short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_SINT24:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();
+ int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_SINT32:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];
+ int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_FLOAT32:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];
+ float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];
+ float sampleDiff = ( toSample - fromSample ) * inSampleDec;
+ ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_FLOAT64:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];
+ double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];
+ double sampleDiff = ( toSample - fromSample ) * inSampleDec;
+ ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ }
+
+ // jump to next in sample
+ inSampleFraction += sampleStep;
+ }
+ }
+}
+
+//-----------------------------------------------------------------------------
+