–Added interpolation to WASAPI's sample rate converter

RtAudio.cpp

@@ -1,4 +1,4 @@
-/************************************************************************/
+/************************************************************************/
 /*! \class RtAudio
     \brief Realtime audio i/o C++ classes.
 
@@ -3859,8 +3859,7 @@ private:
 // 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 favors speed over quality, and works best with conversions between
-// one rate and its multiple.
+// 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,
@@ -3872,40 +3871,126 @@ void convertBufferWasapi( char* outBuffer,
 {
   // 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;
 
   outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );
 
-  // frame-by-frame, copy each relative input sample into it's corresponding output sample
-  for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
+  // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate
+  if (floor(sampleRatio) == sampleRatio || floor(sampleRatioInv) == sampleRatioInv)
   {
-    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;
+	  // 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;
+
+		  switch (format)
+		  {
+		  case RTAUDIO_SINT8:
+		  {
+			  for (unsigned int channel = 0; channel < channelCount; channel++)
+			  {
+				  char fromSample = ((char*)inBuffer)[(inSample * channelCount) + channel];
+				  char toSample = ((char*)inBuffer)[((inSample + 1) * channelCount) + channel];
+				  float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));
+				  ((char*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (char)sampleDiff;
+			  }
+			  break;
+		  }
+		  case RTAUDIO_SINT16:
+		  {
+			  for (unsigned int channel = 0; channel < channelCount; channel++)
+			  {
+				  short fromSample = ((short*)inBuffer)[(inSample * channelCount) + channel];
+				  short toSample = ((short*)inBuffer)[((inSample + 1) * channelCount) + channel];
+				  float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));
+				  ((short*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (short)sampleDiff;
+			  }
+			  break;
+		  }
+		  case RTAUDIO_SINT24:
+		  {
+			  for (unsigned int channel = 0; channel < channelCount; channel++)
+			  {
+				  int fromSample = ((S24*)inBuffer)[(inSample * channelCount) + channel].asInt();
+				  int toSample = ((S24*)inBuffer)[((inSample + 1) * channelCount) + channel].asInt();
+				  float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));
+				  ((S24*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (int)sampleDiff;
+			  }
+			  break;
+		  }
+		  case RTAUDIO_SINT32:
+		  {
+			  for (unsigned int channel = 0; channel < channelCount; channel++)
+			  {
+				  int fromSample = ((int*)inBuffer)[(inSample * channelCount) + channel];
+				  int toSample = ((int*)inBuffer)[((inSample + 1) * channelCount) + channel];
+				  float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));
+				  ((int*)outBuffer)[(outSample * channelCount) + channel] = fromSample + (int)sampleDiff;
+			  }
+			  break;
+		  }
+		  case RTAUDIO_FLOAT32:
+		  {
+			  for (unsigned int channel = 0; channel < channelCount; channel++)
+			  {
+				  float fromSample = ((float*)inBuffer)[(inSample * channelCount) + channel];
+				  float toSample = ((float*)inBuffer)[((inSample + 1) * channelCount) + channel];
+				  float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));
+				  ((float*)outBuffer)[(outSample * channelCount) + channel] = fromSample + sampleDiff;
+			  }
+			  break;
+		  }
+		  case RTAUDIO_FLOAT64:
+		  {
+			  for (unsigned int channel = 0; channel < channelCount; channel++)
+			  {
+				  double fromSample = ((double*)inBuffer)[(inSample * channelCount) + channel];
+				  double toSample = ((double*)inBuffer)[((inSample + 1) * channelCount) + channel];
+				  double sampleDiff = (toSample - fromSample) * (inSampleFraction - floor(inSampleFraction));
+				  ((double*)outBuffer)[(outSample * channelCount) + channel] = fromSample + sampleDiff;
+			  }
+			  break;
+		  }
+		  }
+
+		  // jump to next in sample
+		  inSampleFraction += sampleStep;
+	  }
   }
 }