// between HW and the user. The convertBufferWasapi function is used to perform this conversion\r
// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.\r
// This sample rate converter works best with conversions between one rate and its multiple.\r
-void convertBufferWasapi(char* outBuffer,\r
- const char* inBuffer,\r
- const unsigned int& channelCount,\r
- const unsigned int& inSampleRate,\r
- const unsigned int& outSampleRate,\r
- const unsigned int& inSampleCount,\r
- unsigned int& outSampleCount,\r
- const RtAudioFormat& format)\r
-{\r
- // calculate the new outSampleCount and relative sampleStep\r
- float sampleRatio = (float)outSampleRate / inSampleRate;\r
- float sampleRatioInv = (float)1 / sampleRatio;\r
- float sampleStep = 1.0f / sampleRatio;\r
- float inSampleFraction = 0.0f;\r
-\r
- 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
+void convertBufferWasapi( char* outBuffer,\r
+ const char* inBuffer,\r
+ const unsigned int& channelCount,\r
+ const unsigned int& inSampleRate,\r
+ const unsigned int& outSampleRate,\r
+ const unsigned int& inSampleCount,\r
+ unsigned int& outSampleCount,\r
+ const RtAudioFormat& format )\r
+{\r
+ // calculate the new outSampleCount and relative sampleStep\r
+ float sampleRatio = ( float ) outSampleRate / inSampleRate;\r
+ float sampleRatioInv = ( float ) 1 / sampleRatio;\r
+ float sampleStep = 1.0f / sampleRatio;\r
+ float inSampleFraction = 0.0f;\r
+\r
+ 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
+ {\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
{\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
- {\r
- unsigned int inSample = (unsigned int)inSampleFraction;\r
-\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
- }\r
+ unsigned int inSample = ( unsigned int ) inSampleFraction;\r
\r
- // jump to next in sample\r
- inSampleFraction += sampleStep;\r
- }\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
+ }\r
+\r
+ // jump to next in sample\r
+ inSampleFraction += sampleStep;\r
}\r
- else // else interpolate\r
+ }\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
{\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
- {\r
- unsigned int inSample = (unsigned int)inSampleFraction;\r
-\r
- switch (format)\r
- {\r
- case RTAUDIO_SINT8:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\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
- }\r
- break;\r
- }\r
- case RTAUDIO_SINT16:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\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
- }\r
- break;\r
- }\r
- case RTAUDIO_SINT24:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\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
- }\r
- break;\r
- }\r
- case RTAUDIO_SINT32:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\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
- }\r
- break;\r
- }\r
- case RTAUDIO_FLOAT32:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\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
- }\r
- break;\r
- }\r
- case RTAUDIO_FLOAT64:\r
- {\r
- for (unsigned int channel = 0; channel < channelCount; channel++)\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
- }\r
- break;\r
- }\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
- // jump to next in sample\r
- inSampleFraction += sampleStep;\r
+ switch ( format )\r
+ {\r
+ case RTAUDIO_SINT8:\r
+ {\r
+ char* convInBuffer = ( char* ) inBuffer;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ char fromSample = convInBuffer[ frameInSample + channel ];\r
+ char toSample = convInBuffer[ frameInSample + channelCount + channel ];\r
+ char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
}\r
+ case RTAUDIO_SINT16:\r
+ {\r
+ short* convInBuffer = ( short* ) inBuffer;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ short fromSample = convInBuffer[ frameInSample + channel ];\r
+ short toSample = convInBuffer[ frameInSample + channelCount + channel ];\r
+ short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
+ }\r
+ case RTAUDIO_SINT24:\r
+ {\r
+ S24* convInBuffer = ( S24* ) inBuffer;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ int fromSample = convInBuffer[ frameInSample + channel ].asInt();\r
+ int toSample = convInBuffer[ frameInSample + channelCount + channel ].asInt();\r
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
+ }\r
+ case RTAUDIO_SINT32:\r
+ {\r
+ int* convInBuffer = ( int* ) inBuffer;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ int fromSample = convInBuffer[ frameInSample + channel ];\r
+ int toSample = convInBuffer[ frameInSample + channelCount + channel ];\r
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );\r
+ ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
+ }\r
+ case RTAUDIO_FLOAT32:\r
+ {\r
+ float* convInBuffer = ( float* ) inBuffer;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ float fromSample = convInBuffer[ frameInSample + channel ];\r
+ float toSample = convInBuffer[ frameInSample + channelCount + channel ];\r
+ float sampleDiff = ( toSample - fromSample ) * inSampleDec;\r
+ ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
+ }\r
+ case RTAUDIO_FLOAT64:\r
+ {\r
+ double* convInBuffer = ( double* ) inBuffer;\r
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )\r
+ {\r
+ double fromSample = convInBuffer[ frameInSample + channel ];\r
+ double toSample = convInBuffer[ frameInSample + channelCount + channel ];\r
+ double sampleDiff = ( toSample - fromSample ) * inSampleDec;\r
+ ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;\r
+ }\r
+ break;\r
+ }\r
+ }\r
+\r
+ // jump to next in sample\r
+ inSampleFraction += sampleStep;\r
}\r
+ }\r
}\r
\r
//-----------------------------------------------------------------------------\r
projects / rtaudio-cdist.git / blobdiff