diff options
Diffstat (limited to 'RtAudio.cpp')
| -rw-r--r-- | RtAudio.cpp | 144 |
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;
+ }
}
}
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