X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;ds=sidebyside;f=RtAudio.cpp;h=e0713a0d58b7f3496f25884fa473c9e4b9bf2a5c;hb=241e666f546364dbc471d7ff2aa381f32d1d9885;hp=9c7f1546e56e4fafb1d9bc193159242caf1cd269;hpb=01cc43c819896862cb89ef34b6e0173c8a3f7b10;p=rtaudio.git

diff --git a/RtAudio.cpp b/RtAudio.cpp
index 9c7f154..e0713a0 100644
--- 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>
@@ -1905,7 +1906,10 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
  unlock:
   //MUTEX_UNLOCK( &stream_.mutex );
 
-  RtApi::tickStreamTime();
+  // Make sure to only tick duplex stream time once if using two devices
+  if ( stream_.mode != DUPLEX || (stream_.mode == DUPLEX && handle->id[0] != handle->id[1] && deviceId == handle->id[0] ) )
+    RtApi::tickStreamTime();
+  
   return SUCCESS;
 }
 
@@ -3841,8 +3845,9 @@ public:
       relOutIndex += bufferSize_;
     }
 
-    // "in" index can end on the "out" index but cannot begin at it
-    if ( inIndex_ < relOutIndex && inIndexEnd > relOutIndex ) {
+    // the "IN" index CAN BEGIN at the "OUT" index
+    // the "IN" index CANNOT END at the "OUT" index
+    if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) {
       return false; // not enough space between "in" index and "out" index
     }
 
@@ -3902,7 +3907,8 @@ public:
       relInIndex += bufferSize_;
     }
 
-    // "out" index can begin at and end on the "in" index
+    // the "OUT" index CANNOT BEGIN at the "IN" index
+    // the "OUT" index CAN END at the "IN" index
     if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) {
       return false; // not enough space between "out" index and "in" index
     }
@@ -4843,7 +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 );
@@ -4935,7 +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;
@@ -5239,6 +5245,9 @@ void RtApiWasapi::wasapiThread()
                                    captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
                                    stream_.callbackInfo.userData );
 
+        // tick stream time
+        RtApi::tickStreamTime();
+
         // Handle return value from callback
         if ( callbackResult == 1 ) {
           // instantiate a thread to stop this thread
@@ -5447,9 +5456,6 @@ void RtApiWasapi::wasapiThread()
       // unsetting the callbackPulled flag lets the stream know that
       // the audio device is ready for another callback output buffer.
       callbackPulled = false;
-
-      // tick stream time
-      RtApi::tickStreamTime();
     }
 
   }
@@ -8790,6 +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 );