Merge pull request #88 from MarcusTomlinson/wasapi_sr_fix

Added interpolation to WASAPI's sample rate converter

1 files changed, 116 insertions, 28 deletions

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 882fa0e..f4bff00 100644
--- a/RtAudio.cpp
+++ b/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 +3871,129 @@ 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 )

+    // frame-by-frame, copy each relative input sample into it's corresponding output sample

+    for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )

     {

-      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;

+      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;

 

-    // jump to next in sample

-    inSampleFraction += sampleStep;

+      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;

+    }

   }

 }