From: Marcus Tomlinson Date: Sat, 22 Apr 2017 17:24:22 +0000 (+0200) Subject: –Fix formatting X-Git-Url: https://git.carlh.net/gitweb/?a=commitdiff_plain;h=47585b1a05f3fd37cc491fcc432b68582417ffad;p=rtaudio-cdist.git –Fix formatting --- diff --git a/RtAudio.cpp b/RtAudio.cpp index 65ed830..95cf910 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -3869,129 +3869,129 @@ void convertBufferWasapi( char* outBuffer, unsigned int& outSampleCount, const RtAudioFormat& format ) { - // 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; + // 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 ); + 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 inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate + 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++) { - // 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; - } + unsigned int inSample = ( unsigned int ) inSampleFraction; - // jump to next in sample - inSampleFraction += sampleStep; - } + 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 + } + 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++) { - // 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; - } - } + unsigned int inSample = ( unsigned int ) inSampleFraction; - // jump to next in sample - inSampleFraction += sampleStep; + 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; } + } } //-----------------------------------------------------------------------------