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authorMarcus Tomlinson <themarcustomlinson@gmail.com>2017-04-22 19:26:44 +0200
committerMarcus Tomlinson <themarcustomlinson@gmail.com>2017-04-22 19:26:44 +0200
commitef482c9dd846a3eab87fa42735b85ed3e04a7519 (patch)
treed661c56262f408b9a0a2339f15286e3ee36be344 /RtAudio.cpp
parent47585b1a05f3fd37cc491fcc432b68582417ffad (diff)
–Fix formatting
Diffstat (limited to 'RtAudio.cpp')
-rw-r--r--RtAudio.cpp82
1 files changed, 41 insertions, 41 deletions
diff --git a/RtAudio.cpp b/RtAudio.cpp
index 95cf910..a5439a2 100644
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -3878,32 +3878,32 @@ void convertBufferWasapi( char* outBuffer,
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)
+ 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++)
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
{
unsigned int inSample = ( unsigned int ) inSampleFraction;
- switch (format)
+ switch ( format )
{
case RTAUDIO_SINT8:
- memcpy( &(( char* ) outBuffer)[outSample * channelCount], &(( char* ) inBuffer)[inSample * channelCount], channelCount * sizeof( char ) );
+ 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 ) );
+ 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 ) );
+ 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 ) );
+ 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 ) );
+ 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 ) );
+ memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
break;
}
@@ -3914,75 +3914,75 @@ void convertBufferWasapi( char* outBuffer,
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++)
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
{
unsigned int inSample = ( unsigned int ) inSampleFraction;
- switch (format)
+ switch ( format )
{
case RTAUDIO_SINT8:
{
- for (unsigned int channel = 0; channel < channelCount; channel++)
+ 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;
+ 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++)
+ 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;
+ 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++)
+ 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;
+ 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++)
+ 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;
+ 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++)
+ 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;
+ 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++)
+ 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;
+ 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;
}