diff options
| author | Marcus Tomlinson <themarcustomlinson@gmail.com> | 2017-04-22 19:12:22 +0200 |
|---|---|---|
| committer | Marcus Tomlinson <themarcustomlinson@gmail.com> | 2017-04-22 19:12:22 +0200 |
| commit | dc20fccbbec601cca84adb248d0c9550471990fc (patch) | |
| tree | fd89cbfbac01fc07b2b6fd2f9fd3ace8229691a1 /RtAudio.cpp | |
| parent | a0e1549e6df433cec8129ec0ef98bf26dc075c21 (diff) | |
Added interpolation to WASAPI's sample rate converter
Diffstat (limited to 'RtAudio.cpp')
| -rw-r--r-- | RtAudio.cpp | 147 |
1 files changed, 116 insertions, 31 deletions
diff --git a/RtAudio.cpp b/RtAudio.cpp index 882fa0e..cdb98d7 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 +3871,126 @@ 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;
+ }
}
}
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