From: Gary Scavone <gary@music.mcgill.ca>
Date: Wed, 2 Apr 2014 17:41:27 +0000 (-0400)
Subject: Updated WASAPI code from v0.1 to v0.2.
X-Git-Url: https://git.carlh.net/gitweb/?a=commitdiff_plain;h=0d0fb5433a2be8723a392b943111aea75b301333;p=rtaudio-cdist.git

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

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 ) {
diff --git a/tests/Windows/audioprobe.dsp b/tests/Windows/audioprobe.dsp
index cbdc11a..2eebcc3 100755
--- a/tests/Windows/audioprobe.dsp
+++ b/tests/Windows/audioprobe.dsp
@@ -42,7 +42,7 @@ RSC=rc.exe
 # PROP Ignore_Export_Lib 0
 # PROP Target_Dir ""
 # ADD BASE CPP /nologo /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_CONSOLE" /D "_MBCS" /YX /FD /c
-# ADD CPP /nologo /MT /W3 /GX /O2 /I "../../" /I "../../include" /D "NDEBUG" /D "__WINDOW_DS__" /D "__WINDOWS_DS__" /D "__WINDOWS_ASIO__" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /YX /FD /c
+# ADD CPP /nologo /MT /W3 /GX /O2 /I "../../" /I "../../include" /D "NDEBUG" /D "__WINDOWS_DS__" /D "__WINDOWS_ASIO__" /D "WIN32" /D "_CONSOLE" /D "_MBCS" /YX /FD /c
 # ADD BASE RSC /l 0x409 /d "NDEBUG"
 # ADD RSC /l 0x409 /d "NDEBUG"
 BSC32=bscmake.exe