Updated WASAPI code from v0.1 to v0.2.

1 files changed, 7 insertions, 16 deletions

diff --git a/RtAudio.cpp b/RtAudio.cpp
index c5a2a87..202de26 100644
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -4560,11 +4560,7 @@ void RtApiWasapi::wasapiThread()
     EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to get capture buffer size" );

 

     // scale outBufferSize according to stream->user sample rate ratio

-    // (outBufferSize must be a multiple of the input channel count)

-    unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio );

-    if ( outBufferSize % stream_.nDeviceChannels[INPUT] )

-      outBufferSize += stream_.nDeviceChannels[INPUT] - ( outBufferSize % stream_.nDeviceChannels[INPUT] );

-

+    unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];

     inBufferSize *= stream_.nDeviceChannels[INPUT];

 

     // set captureBuffer size

@@ -4620,12 +4616,7 @@ void RtApiWasapi::wasapiThread()
     EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to get render buffer size" );

 

     // scale inBufferSize according to user->stream sample rate ratio

-    // (inBufferSize must be a multiple of the output channel count)

-    unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * renderSrRatio );

-    if ( inBufferSize % stream_.nDeviceChannels[OUTPUT] ) {

-      inBufferSize += stream_.nDeviceChannels[OUTPUT] - ( inBufferSize % stream_.nDeviceChannels[OUTPUT] );

-    }

-

+    unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];

     outBufferSize *= stream_.nDeviceChannels[OUTPUT];

 

     // set renderBuffer size

@@ -4660,14 +4651,14 @@ void RtApiWasapi::wasapiThread()
   char* convBuffer = NULL;

 

   if ( stream_.mode == INPUT ) {

-    convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio * formatBytes( stream_.deviceFormat[INPUT] ) ) );

+    convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ) );

   }

   else if ( stream_.mode == OUTPUT ) {

-    convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * renderSrRatio * formatBytes( stream_.deviceFormat[OUTPUT] ) ) );

+    convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );

   }

   else if ( stream_.mode == DUPLEX ) {

-    convBuffer = ( char* ) malloc( max( ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio * formatBytes( stream_.deviceFormat[INPUT] ) ),

-                                        ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * renderSrRatio * formatBytes( stream_.deviceFormat[OUTPUT] ) ) ) );

+    convBuffer = ( char* ) malloc( max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),

+                                        ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ) );

   }

 

   // stream process loop

@@ -4682,7 +4673,7 @@ void RtApiWasapi::wasapiThread()
       if ( captureAudioClient ) {

         // Pull callback buffer from inputBuffer

         callbackPulled = captureBuffer.pullBuffer( convBuffer,

-                                                   ( unsigned int ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio ),

+                                                   ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],

                                                    stream_.deviceFormat[INPUT] );

 

         if ( callbackPulled ) {