\96Fix formatting
authorMarcus Tomlinson <themarcustomlinson@gmail.com>
Sat, 22 Apr 2017 17:22:42 +0000 (19:22 +0200)
committerMarcus Tomlinson <themarcustomlinson@gmail.com>
Sat, 22 Apr 2017 17:22:42 +0000 (19:22 +0200)
RtAudio.cpp

index 9ebe49dbb793809ddc89c53bbf509fb5eceaa0ea..65ed83072e31a65d28f20d841a6ba94a53839eea 100644 (file)
@@ -3860,50 +3860,50 @@ private:
 // 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
+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 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
+    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
         {\r
-            unsigned int inSample = (unsigned int)inSampleFraction;\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
+                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
+                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
+                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
+                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
+                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
+                memcpy( &(( double* ) outBuffer)[outSample * channelCount], &(( double* ) inBuffer)[inSample * channelCount], channelCount * sizeof( double ) );\r
                 break;\r
             }\r
 \r
@@ -3916,7 +3916,7 @@ void convertBufferWasapi(char* outBuffer,
         // 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
+            unsigned int inSample = ( unsigned int ) inSampleFraction;\r
 \r
             switch (format)\r
             {\r
@@ -3924,10 +3924,10 @@ void convertBufferWasapi(char* outBuffer,
             {\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
+                    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
@@ -3935,10 +3935,10 @@ void convertBufferWasapi(char* outBuffer,
             {\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
+                    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
@@ -3946,10 +3946,10 @@ void convertBufferWasapi(char* outBuffer,
             {\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
+                    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
@@ -3957,10 +3957,10 @@ void convertBufferWasapi(char* outBuffer,
             {\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
+                    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
@@ -3968,10 +3968,10 @@ void convertBufferWasapi(char* outBuffer,
             {\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
+                    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
@@ -3979,10 +3979,10 @@ void convertBufferWasapi(char* outBuffer,
             {\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
+                    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