outSampleCount = ( unsigned int ) roundf( inSampleCount * sampleRatio );\r
\r
// if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate\r
- if (floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv)\r
+ if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )\r
{\r
// frame-by-frame, copy each relative input sample into it's corresponding output sample\r
- for (unsigned int outSample = 0; outSample < outSampleCount; outSample++)\r
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )\r
{\r
unsigned int inSample = ( unsigned int ) inSampleFraction;\r
\r
- switch (format)\r
+ switch ( format )\r
{\r
- case RTAUDIO_SINT8:\r
- memcpy( &(( char* ) outBuffer)[outSample * channelCount], &(( char* ) inBuffer)[inSample * channelCount], channelCount * sizeof( char ) );\r
- break;\r
- case RTAUDIO_SINT16:\r
- memcpy( &(( short* ) outBuffer)[outSample * channelCount], &(( short* ) inBuffer)[inSample * channelCount], channelCount * sizeof( short ) );\r
- break;\r
- case RTAUDIO_SINT24:\r
- memcpy( &(( S24* ) outBuffer)[outSample * channelCount], &(( S24* ) inBuffer)[inSample * channelCount], channelCount * sizeof( S24 ) );\r
- break;\r
- case RTAUDIO_SINT32:\r
- memcpy( &(( int* ) outBuffer)[outSample * channelCount], &(( int* ) inBuffer)[inSample * channelCount], channelCount * sizeof( int ) );\r
- break;\r
- case RTAUDIO_FLOAT32:\r
- memcpy( &(( float* ) outBuffer)[outSample * channelCount], &(( float* ) inBuffer)[inSample * channelCount], channelCount * sizeof( float ) );\r
- break;\r
- case RTAUDIO_FLOAT64:\r
- memcpy( &(( double* ) outBuffer)[outSample * channelCount], &(( double* ) inBuffer)[inSample * channelCount], channelCount * sizeof( double ) );\r
- break;\r
+ case RTAUDIO_SINT8:\r
+ memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );\r
+ break;\r
+ case RTAUDIO_SINT16:\r
+ memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );\r
+ break;\r
+ case RTAUDIO_SINT24:\r
+ memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );\r
+ break;\r
+ case RTAUDIO_SINT32:\r
+ memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );\r
+ break;\r
+ case RTAUDIO_FLOAT32:\r
+ memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );\r
+ break;\r
+ case RTAUDIO_FLOAT64:\r
+ memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );\r
+ break;\r
}\r
\r
// jump to next in sample\r
else // else interpolate\r
{\r
// frame-by-frame, copy each relative input sample into it's corresponding output sample\r
- for (unsigned int outSample = 0; outSample < outSampleCount; outSample++)\r
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )\r
{\r
unsigned int inSample = ( unsigned int ) inSampleFraction;\r
+ float inSampleDec = inSampleFraction - inSample;\r
+ unsigned int frameInSample = inSample * channelCount;\r
+ unsigned int frameOutSample = outSample * channelCount;\r
\r
- switch (format)\r
+ switch ( format )\r
{\r
- case RTAUDIO_SINT8:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\r
+ case RTAUDIO_SINT8:\r
{\r
- char fromSample = (( char* ) inBuffer)[(inSample * channelCount) + channel];\r
- char toSample = (( char* ) inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor( inSampleFraction ));\r
- (( char* ) outBuffer)[(outSample * channelCount) + channel] = fromSample + ( char ) sampleDiff;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];\r
+ char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+ char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
}\r
- break;\r
- }\r
- case RTAUDIO_SINT16:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\r
+ case RTAUDIO_SINT16:\r
{\r
- short fromSample = (( short* ) inBuffer)[(inSample * channelCount) + channel];\r
- short toSample = (( short* ) inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor( inSampleFraction ));\r
- (( short* ) outBuffer)[(outSample * channelCount) + channel] = fromSample + ( short ) sampleDiff;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];\r
+ short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+ short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
}\r
- break;\r
- }\r
- case RTAUDIO_SINT24:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\r
+ case RTAUDIO_SINT24:\r
{\r
- int fromSample = (( S24* ) inBuffer)[(inSample * channelCount) + channel].asInt();\r
- int toSample = (( S24* ) inBuffer)[((inSample + 1) * channelCount) + channel].asInt();\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor( inSampleFraction ));\r
- (( S24* ) outBuffer)[(outSample * channelCount) + channel] = fromSample + ( int ) sampleDiff;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();\r
+ int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();\r
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
}\r
- break;\r
- }\r
- case RTAUDIO_SINT32:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\r
+ case RTAUDIO_SINT32:\r
{\r
- int fromSample = (( int* ) inBuffer)[(inSample * channelCount) + channel];\r
- int toSample = (( int* ) inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor( inSampleFraction ));\r
- (( int* ) outBuffer)[(outSample * channelCount) + channel] = fromSample + ( int ) sampleDiff;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];\r
+ int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
}\r
- break;\r
- }\r
- case RTAUDIO_FLOAT32:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\r
+ case RTAUDIO_FLOAT32:\r
{\r
- float fromSample = (( float* ) inBuffer)[(inSample * channelCount) + channel];\r
- float toSample = (( float* ) inBuffer)[((inSample + 1) * channelCount) + channel];\r
- float sampleDiff = (toSample - fromSample) * (inSampleFraction - floor( inSampleFraction ));\r
- (( float* ) outBuffer)[(outSample * channelCount) + channel] = fromSample + sampleDiff;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];\r
+ float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+ float sampleDiff = ( toSample - fromSample ) * inSampleDec;\r
+ ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
}\r
- break;\r
- }\r
- case RTAUDIO_FLOAT64:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\r
+ case RTAUDIO_FLOAT64:\r
{\r
- double fromSample = (( double* ) inBuffer)[(inSample * channelCount) + channel];\r
- double toSample = (( double* ) inBuffer)[((inSample + 1) * channelCount) + channel];\r
- double sampleDiff = (toSample - fromSample) * (inSampleFraction - floor( inSampleFraction ));\r
- (( double* ) outBuffer)[(outSample * channelCount) + channel] = fromSample + sampleDiff;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];\r
+ double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];\r
+ double sampleDiff = ( toSample - fromSample ) * inSampleDec;\r
+ ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
}\r
- break;\r
- }\r
}\r
\r
// jump to next in sample\r
info.nativeFormats |= RTAUDIO_SINT8;\r
if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )\r
info.nativeFormats |= RTAUDIO_SINT32;\r
+#ifdef AFMT_FLOAT
if ( mask & AFMT_FLOAT )\r
info.nativeFormats |= RTAUDIO_FLOAT32;\r
+#endif
if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )\r
info.nativeFormats |= RTAUDIO_SINT24;\r
\r
}\r
\r
// Verify the sample rate setup worked.\r
- if ( abs( srate - sampleRate ) > 100 ) {\r
+ if ( abs( srate - (int)sampleRate ) > 100 ) {
close( fd );\r
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";\r
errorText_ = errorStream_.str();\r
projects / rtaudio-cdist.git / blobdiff