Merge pull request #207 from mcwissink/capture-flags-wasapi

[rtaudio.git] / RtAudio.cpp

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 33ae87b08f63cb0d55a42ca871b209de770051ef..e0713a0d58b7f3496f25884fa473c9e4b9bf2a5c 100644 (file)
--- a/RtAudio.cpp
+++ b/RtAudio.cpp
@@ -7,10 +7,11 @@
     and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
     (DirectSound, ASIO and WASAPI) operating systems.
 
+    RtAudio GitHub site: https://github.com/thestk/rtaudio
     RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
 
     RtAudio: realtime audio i/o C++ classes
-    Copyright (c) 2001-2017 Gary P. Scavone
+    Copyright (c) 2001-2019 Gary P. Scavone
 
     Permission is hereby granted, free of charge, to any person
     obtaining a copy of this software and associated documentation files
@@ -38,7 +39,7 @@
 */
 /************************************************************************/
 
-// RtAudio: Version 5.0.0
+// RtAudio: Version 5.1.0
 
 #include "RtAudio.h"
 #include <iostream>
@@ -4848,7 +4849,7 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne
     stream_.doConvertBuffer[mode] = true;
 
   if ( stream_.doConvertBuffer[mode] )
-    setConvertInfo( mode, 0 );
+    setConvertInfo( mode, firstChannel );
 
   // Allocate necessary internal buffers
   bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
@@ -4940,7 +4941,7 @@ void RtApiWasapi::wasapiThread()
   // declare local stream variables
   RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
   BYTE* streamBuffer = NULL;
-  unsigned long captureFlags = 0;
+  DWORD captureFlags = 0;
   unsigned int bufferFrameCount = 0;
   unsigned int numFramesPadding = 0;
   unsigned int convBufferSize = 0;
@@ -8795,6 +8796,8 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
     stream_.doConvertBuffer[mode] = true;
   if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
     stream_.doConvertBuffer[mode] = true;
+  if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+    stream_.doConvertBuffer[mode] = true;
 
   // Allocate necessary internal buffers.
   bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );