From ef482c9dd846a3eab87fa42735b85ed3e04a7519 Mon Sep 17 00:00:00 2001 From: Marcus Tomlinson Date: Sat, 22 Apr 2017 19:26:44 +0200 Subject: [PATCH] =?utf8?q?=C2=96Fix=20formatting?= MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit --- RtAudio.cpp | 82 ++++++++++++++++++++++++++--------------------------- 1 file 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; } -- 2.30.2