\96Fix formatting

diff --git a/RtAudio.cpp b/RtAudio.cpp
index a5439a2394b7bd243d92230884ecc8204bcf616b..41b6b6abe1f8a68ccbe18bf908242ca9cfbf794e 100644 (file)
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -3887,24 +3887,24 @@ void convertBufferWasapi( char* outBuffer,
 \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
@@ -3920,72 +3920,72 @@ void convertBufferWasapi( char* outBuffer,
 \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 )[ ( 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
-        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 )[ ( 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
-        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 )[ ( 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
-        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 )[ ( 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
-        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 )[ ( 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
-        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 )[ ( 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
-        break;\r
-      }\r
       }\r
 \r
       // jump to next in sample\r