diff -Nru mixxx-1.6.0/COPYING mixxx-1.6.1/COPYING
--- mixxx-1.6.0/COPYING	2007-02-25 18:50:59.000000000 -0600
+++ mixxx-1.6.1/COPYING	2008-09-20 18:05:46.000000000 -0500
@@ -1,3 +1,3 @@
-Mixxx is Copyright (C) 2000-2007 by its respective authors. This version
+Mixxx is Copyright (C) 2000-2008 by its respective authors. This version
 of the program is distributed under the General Public Licence version 2, 
 as described in the file LICENSE distributed with the program.
diff -Nru mixxx-1.6.0/debian/changelog mixxx-1.6.1/debian/changelog
--- mixxx-1.6.0/debian/changelog	2008-11-15 15:24:51.000000000 -0600
+++ mixxx-1.6.1/debian/changelog	2008-11-15 15:24:55.000000000 -0600
@@ -1,3 +1,16 @@
+mixxx (1.6.1-1ubuntu1) jaunty; urgency=low
+
+  * Merge from debian unstable, remaining changes (LP: #298514):
+    - Update desktop file.
+
+ -- Nathan Handler <nhandler@ubuntu.com>  Sat, 15 Nov 2008 14:13:30 -0600
+
+mixxx (1.6.1-1) unstable; urgency=low
+
+  * New Upstream Version
+
+ -- Free Ekanayaka <freee@debian.org>  Sat, 15 Nov 2008 00:00:37 +0100
+
 mixxx (1.6.0-1ubuntu1) jaunty; urgency=low
 
   * Merge from debian unstable, remaining changes (LP: #262971):
diff -Nru mixxx-1.6.0/lib/soundtouch/3dnow_win.cpp mixxx-1.6.1/lib/soundtouch/3dnow_win.cpp
--- mixxx-1.6.0/lib/soundtouch/3dnow_win.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/3dnow_win.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -35,7 +35,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: 3dnow_win.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/AAFilter.cpp mixxx-1.6.1/lib/soundtouch/AAFilter.cpp
--- mixxx-1.6.0/lib/soundtouch/AAFilter.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/AAFilter.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -12,7 +12,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: AAFilter.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/AAFilter.h mixxx-1.6.1/lib/soundtouch/AAFilter.h
--- mixxx-1.6.0/lib/soundtouch/AAFilter.h	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/AAFilter.h	2005-02-10 07:11:55.000000000 -0600
@@ -13,7 +13,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: AAFilter.h 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/cpu_detect.h mixxx-1.6.1/lib/soundtouch/cpu_detect.h
--- mixxx-1.6.0/lib/soundtouch/cpu_detect.h	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/cpu_detect.h	2005-02-10 07:11:55.000000000 -0600
@@ -12,7 +12,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: cpu_detect.h 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/cpu_detect_x86_gcc.cpp mixxx-1.6.1/lib/soundtouch/cpu_detect_x86_gcc.cpp
--- mixxx-1.6.0/lib/soundtouch/cpu_detect_x86_gcc.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/cpu_detect_x86_gcc.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -12,7 +12,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: cpu_detect_x86_gcc.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/cpu_detect_x86_win.cpp mixxx-1.6.1/lib/soundtouch/cpu_detect_x86_win.cpp
--- mixxx-1.6.0/lib/soundtouch/cpu_detect_x86_win.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/cpu_detect_x86_win.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -12,7 +12,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: cpu_detect_x86_win.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/FIFOSampleBuffer.cpp mixxx-1.6.1/lib/soundtouch/FIFOSampleBuffer.cpp
--- mixxx-1.6.0/lib/soundtouch/FIFOSampleBuffer.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/FIFOSampleBuffer.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -15,7 +15,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: FIFOSampleBuffer.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/FIFOSampleBuffer.h mixxx-1.6.1/lib/soundtouch/FIFOSampleBuffer.h
--- mixxx-1.6.0/lib/soundtouch/FIFOSampleBuffer.h	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/FIFOSampleBuffer.h	2005-02-10 07:11:55.000000000 -0600
@@ -15,7 +15,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: FIFOSampleBuffer.h 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/FIFOSamplePipe.h mixxx-1.6.1/lib/soundtouch/FIFOSamplePipe.h
--- mixxx-1.6.0/lib/soundtouch/FIFOSamplePipe.h	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/FIFOSamplePipe.h	2005-02-10 07:11:55.000000000 -0600
@@ -17,7 +17,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: FIFOSamplePipe.h 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/FIRFilter.cpp mixxx-1.6.1/lib/soundtouch/FIRFilter.cpp
--- mixxx-1.6.0/lib/soundtouch/FIRFilter.cpp	2006-08-10 07:08:06.000000000 -0500
+++ mixxx-1.6.1/lib/soundtouch/FIRFilter.cpp	2006-08-10 07:08:06.000000000 -0500
@@ -11,7 +11,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2006-08-10 08:08:06 -0400 (Thu, 10 Aug 2006) $
+// Last changed  : $Date: 2006-08-10 05:08:06 -0700 (Thu, 10 Aug 2006) $
 // File revision : $Revision: 967 $
 //
 // $Id: FIRFilter.cpp 967 2006-08-10 12:08:06Z adam_d $
diff -Nru mixxx-1.6.0/lib/soundtouch/FIRFilter.h mixxx-1.6.1/lib/soundtouch/FIRFilter.h
--- mixxx-1.6.0/lib/soundtouch/FIRFilter.h	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/FIRFilter.h	2005-02-10 07:11:55.000000000 -0600
@@ -11,7 +11,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: FIRFilter.h 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/mmx_gcc.cpp mixxx-1.6.1/lib/soundtouch/mmx_gcc.cpp
--- mixxx-1.6.0/lib/soundtouch/mmx_gcc.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/mmx_gcc.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -15,7 +15,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: mmx_gcc.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/mmx_win.cpp mixxx-1.6.1/lib/soundtouch/mmx_win.cpp
--- mixxx-1.6.0/lib/soundtouch/mmx_win.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/mmx_win.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -15,7 +15,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: mmx_win.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/RateTransposer.cpp mixxx-1.6.1/lib/soundtouch/RateTransposer.cpp
--- mixxx-1.6.0/lib/soundtouch/RateTransposer.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/RateTransposer.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -10,7 +10,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: RateTransposer.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/RateTransposer.h mixxx-1.6.1/lib/soundtouch/RateTransposer.h
--- mixxx-1.6.0/lib/soundtouch/RateTransposer.h	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/RateTransposer.h	2005-02-10 07:11:55.000000000 -0600
@@ -14,7 +14,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: RateTransposer.h 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/SoundTouch.cpp mixxx-1.6.1/lib/soundtouch/SoundTouch.cpp
--- mixxx-1.6.0/lib/soundtouch/SoundTouch.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/SoundTouch.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -41,7 +41,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: SoundTouch.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/SoundTouch.h mixxx-1.6.1/lib/soundtouch/SoundTouch.h
--- mixxx-1.6.0/lib/soundtouch/SoundTouch.h	2006-08-10 14:32:06.000000000 -0500
+++ mixxx-1.6.1/lib/soundtouch/SoundTouch.h	2006-08-10 14:32:06.000000000 -0500
@@ -41,7 +41,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2006-08-10 15:32:06 -0400 (Thu, 10 Aug 2006) $
+// Last changed  : $Date: 2006-08-10 12:32:06 -0700 (Thu, 10 Aug 2006) $
 // File revision : $Revision: 975 $
 //
 // $Id: SoundTouch.h 975 2006-08-10 19:32:06Z adam_d $
diff -Nru mixxx-1.6.0/lib/soundtouch/sse_win.cpp mixxx-1.6.1/lib/soundtouch/sse_win.cpp
--- mixxx-1.6.0/lib/soundtouch/sse_win.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/sse_win.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -23,7 +23,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: sse_win.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/STTypes.h mixxx-1.6.1/lib/soundtouch/STTypes.h
--- mixxx-1.6.0/lib/soundtouch/STTypes.h	2008-05-11 10:28:27.000000000 -0500
+++ mixxx-1.6.1/lib/soundtouch/STTypes.h	2008-05-11 10:28:27.000000000 -0500
@@ -8,7 +8,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2008-05-11 11:28:27 -0400 (Sun, 11 May 2008) $
+// Last changed  : $Date: 2008-05-11 08:28:27 -0700 (Sun, 11 May 2008) $
 // File revision : $Revision: 1981 $
 //
 // $Id: STTypes.h 1981 2008-05-11 15:28:27Z gamegod $
diff -Nru mixxx-1.6.0/lib/soundtouch/TDStretch.cpp mixxx-1.6.1/lib/soundtouch/TDStretch.cpp
--- mixxx-1.6.0/lib/soundtouch/TDStretch.cpp	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/TDStretch.cpp	2005-02-10 07:11:55.000000000 -0600
@@ -13,7 +13,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: TDStretch.cpp 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/lib/soundtouch/TDStretch.h mixxx-1.6.1/lib/soundtouch/TDStretch.h
--- mixxx-1.6.0/lib/soundtouch/TDStretch.h	2005-02-10 07:11:55.000000000 -0600
+++ mixxx-1.6.1/lib/soundtouch/TDStretch.h	2005-02-10 07:11:55.000000000 -0600
@@ -13,7 +13,7 @@
 ///
 ////////////////////////////////////////////////////////////////////////////////
 //
-// Last changed  : $Date: 2005-02-10 08:11:55 -0500 (Thu, 10 Feb 2005) $
+// Last changed  : $Date: 2005-02-10 05:11:55 -0800 (Thu, 10 Feb 2005) $
 // File revision : $Revision: 857 $
 //
 // $Id: TDStretch.h 857 2005-02-10 13:11:55Z tuehaste $
diff -Nru mixxx-1.6.0/LICENSE mixxx-1.6.1/LICENSE
--- mixxx-1.6.0/LICENSE	2007-11-11 11:51:46.000000000 -0600
+++ mixxx-1.6.1/LICENSE	2008-09-20 18:05:46.000000000 -0500
@@ -1,5 +1,5 @@
-Mixxx version 1.6.0, Digital DJ'ing software.
-Copyright (C) 2001-2007 Haste Andersen and Mixxx developers
+Mixxx version 1.6.1, Digital DJ'ing software.
+Copyright (C) 2001-2008 Mixxx development team 
 
 Depending on your platform, you may receive a copy of PortAudio 
 (http://www.portaudio.com/) which is distributed under the 
diff -Nru mixxx-1.6.0/.mixxx_flags.svn mixxx-1.6.1/.mixxx_flags.svn
--- mixxx-1.6.0/.mixxx_flags.svn	2008-08-05 15:43:17.000000000 -0500
+++ mixxx-1.6.1/.mixxx_flags.svn	1969-12-31 18:00:00.000000000 -0600
@@ -1 +0,0 @@
-#define BUILD_FLAGS "hifieq vinylcontrol optimize=1 "
diff -Nru mixxx-1.6.0/Mixxx.nsi mixxx-1.6.1/Mixxx.nsi
--- mixxx-1.6.0/Mixxx.nsi	2008-07-29 10:08:58.000000000 -0500
+++ mixxx-1.6.1/Mixxx.nsi	2008-09-11 23:51:33.000000000 -0500
@@ -15,7 +15,7 @@
 BrandingText " "
 
 ; The file to write
-OutFile "mixxx-1.6.0-beta4-win.exe"
+OutFile "mixxx-1.6.0-win.exe"
 
 ; The default installation directory
 InstallDir $PROGRAMFILES\Mixxx
@@ -66,8 +66,9 @@
   SetOutPath $INSTDIR\midi
   File "dist\midi\*.xml"
 
-  SetOutPath $INSTDIR\promo
-  File "dist\promo\*"
+  ;Disabled for initial 1.6.0 release
+  ;SetOutPath $INSTDIR\promo
+  ;File "dist\promo\*"
 
   SetOutPath $INSTDIR\keyboard
   File "src\keyboard\Standard.kbd.cfg"
@@ -76,9 +77,6 @@
   SetOutPath "$INSTDIR\skins"
   File "src\skins\cross.png"
 
-  SetOutPath "$INSTDIR\skins\Natt"
-  File "src\skins\Natt\*"
-
   SetOutPath "$INSTDIR\skins\Collusion (1280)"
   File "src\skins\Collusion (1280)\*"
 
@@ -1279,7 +1277,6 @@
   Delete $INSTDIR\skins\outlineClose\*.*
   Delete $INSTDIR\skins\outlineSmall\*.*
   Delete $INSTDIR\skins\outlineMini\*.*
-  Delete "$INSTDIR\skins\Natt\*.*"
   Delete "$INSTDIR\skins\Collusion (1280)\*.*"
   Delete "$INSTDIR\skins\Collusion (1280-WS)\*.*"
   Delete $INSTDIR\skins\hercules\*.*
@@ -1288,12 +1285,11 @@
   Delete $INSTDIR\skins\*.*
   Delete $INSTDIR\keyboard\*.*
   Delete $INSTDIR\midi\*.*
-  Delete $INSTDIR\promo\*.*
+  ;Delete $INSTDIR\promo\*.*
   RMDir "$INSTDIR\skins\outline"
   RMDir "$INSTDIR\skins\outlineClose"
   RMDir "$INSTDIR\skins\outlineSmall"
   RMDir "$INSTDIR\skins\outlineMini"
-  RMDir "$INSTDIR\skins\Natt"
   RMDir "$INSTDIR\skins\Collusion (1280)"
   RMDir "$INSTDIR\skins\Collusion (1280-WS)"
   RMDir "$INSTDIR\skins\hercules"
@@ -1302,7 +1298,7 @@
   RMDir "$INSTDIR\skins"
   RMDir "$INSTDIR\midi"
   RMDir "$INSTDIR\keyboard"
-  RMDir "$INSTDIR\promo"
+  ;RMDir "$INSTDIR\promo"
 
 
   ; Remove shortcuts, if any
diff -Nru mixxx-1.6.0/.mixxx_version.svn mixxx-1.6.1/.mixxx_version.svn
--- mixxx-1.6.0/.mixxx_version.svn	2008-08-05 15:43:17.000000000 -0500
+++ mixxx-1.6.1/.mixxx_version.svn	1969-12-31 18:00:00.000000000 -0600
@@ -1 +0,0 @@
-#define BUILD_REV ""
diff -Nru mixxx-1.6.0/README mixxx-1.6.1/README
--- mixxx-1.6.0/README	2008-07-06 12:21:19.000000000 -0500
+++ mixxx-1.6.1/README	2008-09-20 18:05:46.000000000 -0500
@@ -1,4 +1,4 @@
-Mixxx 1.6.0
+Mixxx 1.6.1
 
 * HOMEPAGE *************************************************
 http://www.mixxx.org
@@ -17,46 +17,8 @@
 
 * NEWS *****************************************************
 
-Since Mixxx 1.5.0 was released (March 4th, 2007), the following changes have been made:
-
-* New MIDI mappings for Tascam US-428, M-Audio X-Session Pro, Evolution X-Session, FaderFox DJ2, and the M-Audio Torq Xponent
-* ALSA Sequencer MIDI support courtesy of Cedric Gestes
-* A couple of MIDI bug fixes (knobs now center properly, thanks to Sacha Berger)
-* Added support for 14-bit MIDI pitch wheel controllers (thanks to Adam Sugerman)
-* Hercules support on Linux improved (jog wheels work again)
-* New nCut skin from Frank Willascheck
-* Big stability improvements (3 bug fixes)
-* Multiple soundcards can now be used for output (master/headphones), in case you don't have a soundcard with 4 outputs on it.
-* Adam's wicked colour scheme support for skins
-* Can now change skins without restarting Mixxx (more hard work from Adam)
-* Channel VU meters are now pre-fader
-* VU meters are now much more smooth
-* Added clipping indicators (courtesy of John Sully)
-* Higher quality EQs and other sound quality improvements (also from John Sully)
-* New MIDI mapping format now in XML, supports controlling LEDs
-* Better Hercules support on Windows and Linux
-* Initial support for recording output
-* New BPM detection algorithm (Micah Lee/GSoC)
-* New media library (Nathan Prado/GSoC)
-* LADSPA effects support (Pawel Bartkiewicz/GSoC)
-* BPM Tap tempo
-* Library search function
-* Ported to QT4 (!)
-* Moved build system to SCONS
-* Redesigned preferences dialogs
-* Rewritten audio core (Albert)
-* Vinyl control support for Serato, Traktor Scratch, and FinalScratch (FS needs work, but the others are good)
-* Software preamp for vinyl control (can use turntables without a preamp)
-* Track info editor (double-click in library)
-* New library browse mode (CTAF)
-* Starts in fullscreen mode if launched with the -f flag.
-* Several MP3 decoder performance and stability improvements (John Sully)
-* Support for merengue
-* Reorganized "File" menu
-* NEXT mode now works as expected (plays the next track in the table)
-* Partial play queue
-* Lots of little OS X improvements
-* Improved consistency of fullscreen mode 
+Check the Mixxx blog to find out what's new in this version:
+http://mixxxblog.blogspot.com
 
 * UPGRADING ************************************************
 
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/.sconsign.tmp and /tmp/MHYQbGTmm3/mixxx-1.6.1/.sconsign.tmp differ
diff -Nru mixxx-1.6.0/src/bpmdetect.cpp mixxx-1.6.1/src/bpmdetect.cpp
--- mixxx-1.6.0/src/bpmdetect.cpp	2008-05-11 10:27:38.000000000 -0500
+++ mixxx-1.6.1/src/bpmdetect.cpp	2008-08-14 00:23:43.000000000 -0500
@@ -81,6 +81,8 @@
 #include "cmetrics.h"
 #endif
 
+#include <QtDebug>
+
 using namespace soundtouch;
 
 #define INPUT_BLOCK_SAMPLES       2048
@@ -165,6 +167,7 @@
     outcount = 0;
     for (count = 0; count < numsamples; count++)
     {
+//        qDebug() << "======"<< "count:"<< count << "numsamples:"<<numsamples << "outcount:"<< outcount << "decimateSum:"<< decimateSum << "decimateCount:"<< decimateCount;
         decimateSum += src[count];
 
         decimateCount++;
diff -Nru mixxx-1.6.0/src/bpmdetector.cpp mixxx-1.6.1/src/bpmdetector.cpp
--- mixxx-1.6.0/src/bpmdetector.cpp	2008-06-13 21:36:34.000000000 -0500
+++ mixxx-1.6.1/src/bpmdetector.cpp	2008-09-07 04:36:04.000000000 -0500
@@ -65,23 +65,27 @@
     delete m_pEngineSpectralFwd;
 }
 
+int BpmDetector::queueCount() {
+    return m_qQueue.count();
+}
+
 void BpmDetector::enqueue(TrackInfoObject * pTrackInfoObject, BpmReceiver * pBpmReceiver)
 {
     int minBpm, maxBpm;
     bool entire;
-    
+
     m_qMutex.lock();
     BpmDetectionPackage * package = new BpmDetectionPackage();
     package->_TrackInfoObject = pTrackInfoObject;
     package->_BpmReceiver = pBpmReceiver;
-    
+
     // Read
     minBpm = m_Config->getValueString(ConfigKey("[BPM]","BPMRangeStart")).toInt();
     maxBpm = m_Config->getValueString(ConfigKey("[BPM]","BPMRangeEnd")).toInt();
     entire = (bool)m_Config->getValueString(ConfigKey("[BPM]","AnalyzeEntireSong")).toInt();
-    
+
     package->_Scheme = new BpmScheme("Default", minBpm, maxBpm, entire);
-  
+
     m_qQueue.enqueue(package);
     m_qMutex.unlock();
     m_qWait.wakeAll();
@@ -112,6 +116,8 @@
     int channels = 2;
     float frequency = 44100;
 
+    qDebug() << "BPM detection starting for" << pTrackInfoObject->getFilename();
+
     if(pTrackInfoObject->getSampleRate())
     {
         frequency = pTrackInfoObject->getSampleRate();
@@ -122,7 +128,7 @@
     }
 
     length = pSoundSource->length();
-
+    if (length <= 0) return;
     if(!pScheme->getAnalyzeEntireSong())
     {
         length = length / 2;
@@ -191,14 +197,14 @@
 #ifdef __C_METRICS__
                 cm_writemsg_ascii(1, "BPM detection failed, setting to 0.");
 #endif
-                
+
                 if(pBpmReceiver){
                     pBpmReceiver->setComplete(pTrackInfoObject, true, BPM);
-                }    
-                
+                }
+
                 delete pSoundSource;
                 return;
-            }  
+            }
 
 }
 
@@ -212,7 +218,7 @@
 
 //The fallback is broken and is causing crashes.  This will break us out and set the BPM to 0
 //Remove these lines if you get the old BPM detection working
-    
+
 //qDebug() << "BPM detection failed the first time. Trying old version.";
 
 
@@ -281,7 +287,7 @@
     pPeaks->update(0, iBeatBlockLength);
 
     // Initialize beat probability vector
-    ProbabilityVector * bpv = new ProbabilityVector(60.f/(float)pScheme->getMaxBpm(), 
+    ProbabilityVector * bpv = new ProbabilityVector(60.f/(float)pScheme->getMaxBpm(),
                                                     60.f/(float)pScheme->getMinBpm(), kiBeatBins);
 
     // Calculate BPM
@@ -328,7 +334,6 @@
 
     qDebug() << "BPM detection successful for" << pTrackInfoObject->getFilename();
 
-    
 }
 
 void BpmDetector::run()
diff -Nru mixxx-1.6.0/src/bpmdetector.h mixxx-1.6.1/src/bpmdetector.h
--- mixxx-1.6.0/src/bpmdetector.h	2008-06-13 21:35:45.000000000 -0500
+++ mixxx-1.6.1/src/bpmdetector.h	2008-08-14 00:23:43.000000000 -0500
@@ -66,6 +66,7 @@
     /** Puts an TrackInfoObject into the queue of BPM detection. Thread safe, blocking. */
     void enqueue(TrackInfoObject *pTrackInfoObject, BpmReceiver *pBpmReceiver=NULL);
     void enqueue(TrackInfoObject *pTrackInfoObject, BpmScheme *scheme, BpmReceiver *pBpmReceiver=NULL);
+    int queueCount();
 
 protected:
     /** Main thread loop */
diff -Nru mixxx-1.6.0/src/defs_audiofiles.h mixxx-1.6.1/src/defs_audiofiles.h
--- mixxx-1.6.0/src/defs_audiofiles.h	2008-03-02 00:35:46.000000000 -0600
+++ mixxx-1.6.1/src/defs_audiofiles.h	2008-08-24 01:38:08.000000000 -0500
@@ -8,9 +8,9 @@
 #define __DEFS_AUDIOFILES_H__
 
 /** The types of audio files we support */
-#define MIXXX_SUPPORTED_AUDIO_FILETYPES "*.wav *.mp3 *.ogg *.aiff *.aif *.flac"
+#define MIXXX_SUPPORTED_AUDIO_FILETYPES "*.wav *.mp3 *.m4a *.ogg *.aiff *.aif *.flac"
 
 /** A regex for the types of audio files we support */
-#define MIXXX_SUPPORTED_AUDIO_FILETYPES_REGEX "\\.(mp3|ogg|aiff|aif|wav|flac)"
+#define MIXXX_SUPPORTED_AUDIO_FILETYPES_REGEX "\\.(mp3|m4a|ogg|aiff|aif|wav|flac)"
 
 #endif
diff -Nru mixxx-1.6.0/src/defs.h mixxx-1.6.1/src/defs.h
--- mixxx-1.6.0/src/defs.h	2008-07-29 18:51:22.000000000 -0500
+++ mixxx-1.6.1/src/defs.h	2008-09-20 18:05:46.000000000 -0500
@@ -17,7 +17,7 @@
 #ifndef DEFS_H
 #define DEFS_H
 
-#define VERSION "1.6.0"
+#define VERSION "1.6.1"
 #define MIXXX_PROMO_DIR "promo"
 
 #include <math.h>
diff -Nru mixxx-1.6.0/src/dlgpreferences.h mixxx-1.6.1/src/dlgpreferences.h
--- mixxx-1.6.0/src/dlgpreferences.h	2008-01-30 17:44:00.000000000 -0600
+++ mixxx-1.6.1/src/dlgpreferences.h	2008-08-11 03:57:49.000000000 -0500
@@ -31,7 +31,6 @@
 
 class MixxxApp;
 class MixxxView;
-class PlayerProxy;
 class SoundManager;
 class Track;
 class DlgPrefSound;
diff -Nru mixxx-1.6.0/src/dlgprefsound.cpp mixxx-1.6.1/src/dlgprefsound.cpp
--- mixxx-1.6.0/src/dlgprefsound.cpp	2008-07-06 13:11:20.000000000 -0500
+++ mixxx-1.6.1/src/dlgprefsound.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -16,7 +16,6 @@
 ***************************************************************************/
 
 #include "dlgprefsound.h"
-//#include "playerproxy.h"
 #include <qcombobox.h>
 #include <QButtonGroup>
 #include <QtDebug>
@@ -41,7 +40,6 @@
                            ConfigObject<ConfigValue> * _config) : QWidget(parent), Ui::DlgPrefSoundDlg()
 {
     m_bLatencySliderDrag = false;
-    //player = _player;
     m_pSoundManager = _soundman;
     config = _config;
 	m_parent = parent;
@@ -125,7 +123,6 @@
     // API's
     ComboBoxSoundApi->clear();
     ComboBoxSoundApi->insertItem(0, "None");
-    //QStringList api = player->getSoundApiList();
     QList<QString> apis = m_pSoundManager->getHostAPIList();
     QListIterator<QString> api_it(apis);
     QString api;
@@ -192,7 +189,6 @@
 
     // Sample rate
     ComboBoxSamplerates->clear();
-    //QStringList srates = player->getSampleRates();
     QList<QString> srates = m_pSoundManager->getSamplerateList();
     QListIterator<QString> srate_it(srates);
     QString srate;
@@ -293,7 +289,6 @@
     qDebug() << "request msec " << getSliderLatencyMsec(SliderLatency->value());
 
     // Close devices, and open using config data
-    //player->close();
     m_pSoundManager->closeDevices();
 
     // Not much to do if the API is None...
diff -Nru mixxx-1.6.0/src/dlgprefsound.h mixxx-1.6.1/src/dlgprefsound.h
--- mixxx-1.6.0/src/dlgprefsound.h	2007-12-07 21:04:36.000000000 -0600
+++ mixxx-1.6.1/src/dlgprefsound.h	2008-08-11 03:57:49.000000000 -0500
@@ -60,8 +60,7 @@
     int getSliderLatencyMsec(int);
     /** Transform latency value in msec to slider value */
     int getSliderLatencyVal(int);
-    /** Pointer to player device */
-    //PlayerProxy *player;
+    /** Pointer to the sound manager */
     SoundManager* m_pSoundManager;
     /** Pointer to config object */
     ConfigObject<ConfigValue> *config;
diff -Nru mixxx-1.6.0/src/dlgprefvinyl.cpp mixxx-1.6.1/src/dlgprefvinyl.cpp
--- mixxx-1.6.0/src/dlgprefvinyl.cpp	2008-04-30 11:12:24.000000000 -0500
+++ mixxx-1.6.1/src/dlgprefvinyl.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -23,7 +23,6 @@
 #include <QList>
 #include <QButtonGroup>
 #include "dlgprefvinyl.h"
-#include "playerproxy.h"
 #include <qcombobox.h>
 #include <qcheckbox.h>
 #include <qpushbutton.h>
@@ -42,7 +41,6 @@
 DlgPrefVinyl::DlgPrefVinyl(QWidget * parent, SoundManager * soundman,
                            ConfigObject<ConfigValue> * _config) : QWidget(parent), Ui::DlgPrefVinylDlg()
 {
-    //player = _player;
     m_pSoundManager = soundman;
     config = _config;
 
@@ -247,7 +245,6 @@
 
     // Apply Soundcard options
 
-    //player->close();
     //m_pSoundManager->closeDevices();
 
     //NOTE: Soundcard options (input device selection) is applied by DlgPrefSound...
diff -Nru mixxx-1.6.0/src/enginebuffer.cpp mixxx-1.6.1/src/enginebuffer.cpp
--- mixxx-1.6.0/src/enginebuffer.cpp	2008-07-29 19:01:33.000000000 -0500
+++ mixxx-1.6.1/src/enginebuffer.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -33,7 +33,6 @@
 #include "enginebufferscalesrc.h"
 #include "enginebufferscaledummy.h"
 #include "mathstuff.h"
-#include "player.h"
 #include "enginebuffercue.h"
 
 // Static default values for rate buttons
diff -Nru mixxx-1.6.0/src/enginebuffercue.cpp mixxx-1.6.1/src/enginebuffercue.cpp
--- mixxx-1.6.0/src/enginebuffercue.cpp	2008-07-03 22:58:51.000000000 -0500
+++ mixxx-1.6.1/src/enginebuffercue.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -18,7 +18,6 @@
 #include "controlobject.h"
 #include "enginebuffercue.h"
 #include "enginebuffer.h"
-#include "player.h"
 #include "mathstuff.h"
 
 EngineBufferCue::EngineBufferCue(const char * group, EngineBuffer * pEngineBuffer)
diff -Nru mixxx-1.6.0/src/libraryscanner.cpp mixxx-1.6.1/src/libraryscanner.cpp
--- mixxx-1.6.0/src/libraryscanner.cpp	2008-01-26 00:34:52.000000000 -0600
+++ mixxx-1.6.1/src/libraryscanner.cpp	2008-08-14 00:24:33.000000000 -0500
@@ -31,7 +31,7 @@
 {
     m_qLibraryPlaylist = library_playlist;
     m_qLibraryPath = libraryPath;
-    
+
     qDebug() << "Constructed LibraryScanner!!!";
 }
 
@@ -41,15 +41,15 @@
 }
 
 void LibraryScanner::run()
-{ 
+{
     //m_pProgress->slotStartTiming();
-    
+
     //Start scanning the library.
     m_qLibraryPlaylist->addPath(m_qLibraryPath);
-    
+
     qDebug() << "Scan finished cleanly";
     //m_pProgress->slotStopTiming();
-    
+
     emit(scanFinished());
 }
 
@@ -63,13 +63,13 @@
     //we need the signals to get processed immediately, we have to use BlockingQueuedConnection. (DirectConnection isn't an
     //option for sending signals across threads.)
     connect(m_qLibraryPlaylist, SIGNAL(startedLoading()), m_pProgress, SLOT(slotStartTiming()), Qt::BlockingQueuedConnection);
-    connect(m_qLibraryPlaylist, SIGNAL(finishedLoading()), m_pProgress, SLOT(slotStopTiming()), Qt::BlockingQueuedConnection);  
+    connect(m_qLibraryPlaylist, SIGNAL(finishedLoading()), m_pProgress, SLOT(slotStopTiming()), Qt::BlockingQueuedConnection);
     connect(m_qLibraryPlaylist, SIGNAL(progressLoading(QString)), m_pProgress, SLOT(slotCheckTiming(QString)), Qt::BlockingQueuedConnection);
-    
+
     //connect(m_pProgress, SIGNAL(scanCancelled()), this, SLOT(terminate()));  //This causes a deadlock, don't use it.
     connect(m_pProgress, SIGNAL(scanCancelled()), m_qLibraryPlaylist, SLOT(slotCancelLibraryScan()));
-    
-    //connect(m_qPlaylists->at(0), SIGNAL(finishedLoading()), this, SIGNAL(scanFinished()), Qt::BlockingQueuedConnection);  
+
+    //connect(m_qPlaylists->at(0), SIGNAL(finishedLoading()), this, SIGNAL(scanFinished()), Qt::BlockingQueuedConnection);
 
     scan();
 }
diff -Nru mixxx-1.6.0/src/main.cpp mixxx-1.6.1/src/main.cpp
--- mixxx-1.6.0/src/main.cpp	2008-07-13 22:21:19.000000000 -0500
+++ mixxx-1.6.1/src/main.cpp	2008-08-18 23:31:42.000000000 -0500
@@ -37,7 +37,8 @@
 #include <ladspaloader.h>
 #endif
 
-#ifdef A_THING_THAT_IS_NOT_DEFFED //Q_WS_WIN
+#ifdef Q_WS_WIN
+#ifdef DEBUGCONSOLE
 #include <io.h> // Debug Console
 #include <windows.h>
 
@@ -62,7 +63,8 @@
         setvbuf( stderr, NULL, _IONBF, 0 );
     }
 }
-#endif
+#endif // DEBUGCONSOLE
+#endif // Q_WS_WIN
 
 QApplication * a;
 
@@ -131,11 +133,11 @@
 #ifdef Q_WS_WIN
     // For windows write all debug messages to a logfile:
     Logfile.setName( "mixxx.log" );
-#ifndef QT3_SUPPORT
+  #ifndef QT3_SUPPORT
     Logfile.open(IO_WriteOnly | IO_Translate);
-#else
+  #else
     Logfile.open(QIODevice::WriteOnly | QIODevice::Text);
-#endif
+  #endif
   #ifdef DEBUGCONSOLE
     InitDebugConsole();
     qInstallMsgHandler( MessageOutput );
diff -Nru mixxx-1.6.0/src/midi/Hercules DJ Console Rmx.midi.xml mixxx-1.6.1/src/midi/Hercules DJ Console Rmx.midi.xml
--- mixxx-1.6.0/src/midi/Hercules DJ Console Rmx.midi.xml	2008-04-05 23:52:11.000000000 -0500
+++ mixxx-1.6.1/src/midi/Hercules DJ Console Rmx.midi.xml	2008-09-18 23:42:45.000000000 -0500
@@ -71,7 +71,7 @@
 			<miditype>Ctrl</miditype>
 			<midino>0x14</midino>
 			<options>
-				<switch/>
+				<button/>
 			</options>
 		</control>
 		<control>
@@ -79,12 +79,18 @@
 			<key>PrevTrack</key>
 			<miditype>Ctrl</miditype>
 			<midino>0x09</midino>
+			<options>
+				<button/>
+			</options>
 		</control>
 		<control>
 			<group>[Channel1]</group>
 			<key>NextTrack</key>
 			<miditype>Ctrl</miditype>
 			<midino>0x0A</midino>
+			<options>
+				<button/>
+			</options>
 		</control>
 		
 		<control>
@@ -150,7 +156,7 @@
 			<miditype>Ctrl</miditype>
 			<midino>0x18</midino>
 			<options>
-				<switch/>
+				<button/>
 			</options>
 		</control>
 		<control>
@@ -158,12 +164,18 @@
 			<key>PrevTrack</key>
 			<miditype>Ctrl</miditype>
 			<midino>0x21</midino>
+			<options>
+				<button/>
+			</options>
 		</control>
 		<control>
 			<group>[Channel2]</group>
 			<key>NextTrack</key>
 			<miditype>Ctrl</miditype>
 			<midino>0x22</midino>
+			<options>
+				<button/>
+			</options>
 		</control>
 
 <!-- New for RMX -->
@@ -171,93 +183,104 @@
     		<control>
    			<group>[Channel1]</group>
 			<key>filterHighKill</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x0E</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
     		<control>
    			<group>[Channel2]</group>
 			<key>filterHighKill</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x26</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
     		<control>
    			<group>[Channel1]</group>
 			<key>filterMidKill</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x0F</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
     		<control>
    			<group>[Channel2]</group>
 			<key>filterMidKill</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x27</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
     		<control>
    			<group>[Channel1]</group>
 			<key>filterLowKill</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x10</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
     		<control>
    			<group>[Channel2]</group>
 			<key>filterLowKill</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x28</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
 
 <!-- Sync Buttons -->
 		<control>
 			<group>[Channel1]</group>
 			<key>beatsync</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x07</midino>
-			<midichan>1</midichan>
 		</control>
 		<control>
 			<group>[Channel2]</group>
 			<key>beatsync</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x1B</midino>
-			<midichan>1</midichan>
 		</control>
 
-
 <!-- Up/Down -->
 		<control>
 			<group>[Playlist]</group>
-			<key>SelectNextTrack</key>
-			<miditype>Key</miditype>
+			<key>SelectPrevTrack</key>
+			<miditype>Ctrl</miditype>
 			<midino>0x2A</midino>
-			<midichan>1</midichan>
 		</control>
 		<control>
 			<group>[Playlist]</group>
-			<key>SelectPrevTrack</key>
-			<miditype>Key</miditype>
+			<key>SelectNextTrack</key>
+			<miditype>Ctrl</miditype>
 			<midino>0x2B</midino>
-			<midichan>1</midichan>
 		</control>
 
 <!-- Load Deck A/B -->
 		<control>
 			<group>[Channel1]</group>
 			<key>LoadSelectedTrack</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x12</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
 		<control>
 			<group>[Channel2]</group>
 			<key>LoadSelectedTrack</key>
-			<miditype>Key</miditype>
+			<miditype>Ctrl</miditype>
 			<midino>0x16</midino>
-			<midichan>1</midichan>
+			<options>
+				<button/>
+			</options>
 		</control>
 
 <!-- A/B Gain  -->
@@ -275,6 +298,7 @@
 		</control>
 
 <!-- Scratch -->
+<!-- There is no Master scratch control in Mixxx. -Albert 09/15/2008
 		<control>
 			<group>[Master]</group>
 			<key>scratch</key>
@@ -282,7 +306,7 @@
 			<midino>0x29</midino>
 			<midichan>1</midichan>
 		</control>
-
+-->
 <!-- Balance -->
 		<control>
 			<group>[Master]</group>
@@ -329,6 +353,7 @@
 		</light>
 
 <!-- Scratch -->
+<!--
 		<light>
 			<group>[Master]</group>
 			<key>scratch</key>
@@ -336,7 +361,7 @@
 			<midino>0x29</midino>
 			<threshold>0.5</threshold>
 		</light>
-
+-->
 <!-- Headphone Cue -->
 		<light>
 			<group>[Channel1]</group>
diff -Nru mixxx-1.6.0/src/midiledhandler.cpp mixxx-1.6.1/src/midiledhandler.cpp
--- mixxx-1.6.0/src/midiledhandler.cpp	2008-08-05 10:59:27.000000000 -0500
+++ mixxx-1.6.1/src/midiledhandler.cpp	2008-09-15 22:45:48.000000000 -0500
@@ -10,15 +10,16 @@
                                double max, unsigned char status, unsigned char byte1)
     : m_min(min), m_max(max), m_midi(midi), m_status(status), m_byte1(byte1) {
 
+    //OMGWTFBBQ: Massive hack to temporarily fix LP #254564 for the 1.6.0 release.
+    //           Something's funky with our <lights> blocks handling? -- Albert 08/05/2008
+    if (group.isEmpty() || name.isEmpty()) return;
+
     m_cobj = ControlObject::getControl(ConfigKey(group, name));
-    
-
-    //OMGWTFBBQ: Massive hack to temporarily fix LP #254564 for the 1.6.0 release. 
-    //           Something's funky with our <lights> blocks handling? -- Albert 08/05/2008
-    if (group.isEmpty() || name.isEmpty()) return;
-
+
     //m_cobj should never be null, so Q_ASSERT here to make sure that we hear about it if it is null.
-    Q_ASSERT(m_cobj);
+    // Q_ASSERT(m_cobj);
+    Q_ASSERT_X(m_cobj, "MidiLedHandler", "Invalid config group: '" + group + "' name:'" + name + "'");
+
     connect(m_cobj, SIGNAL(valueChangedFromEngine(double)), this, SLOT(controlChanged(double)));
     connect(m_cobj, SIGNAL(valueChanged(double)), this, SLOT(controlChanged(double)));
 }
@@ -37,12 +38,12 @@
 
     if (!node.isNull() && node.isElement()) {
         QDomNode light = node.firstChild();
-
-        while (!light.isNull()) {
+
+        while (!light.isNull()) {
             if(light.nodeName() == "light") {
                 QString group = WWidget::selectNodeQString(light, "group");
                 QString key = WWidget::selectNodeQString(light, "key");
-    
+
                 unsigned char status = (unsigned char)WWidget::selectNodeInt(light, "status");
                 unsigned char midino = (unsigned char)WWidget::selectNodeInt(light, "midino");
 				float min = 0.0f;
@@ -56,9 +57,9 @@
 				if (!light.firstChildElement("maximum").isNull()) {
 					max = WWidget::selectNodeFloat(light, "maximum");
 				}
-    
-                allhandlers.append(new MidiLedHandler(group, key, midi, min, max, status, midino));
-            }
+
+                allhandlers.append(new MidiLedHandler(group, key, midi, min, max, status, midino));
+            }
             light = light.nextSibling();
         }
     }
diff -Nru mixxx-1.6.0/src/midiobjectwin.cpp mixxx-1.6.1/src/midiobjectwin.cpp
--- mixxx-1.6.0/src/midiobjectwin.cpp	2008-04-06 19:24:52.000000000 -0500
+++ mixxx-1.6.1/src/midiobjectwin.cpp	2008-09-18 00:26:57.000000000 -0500
@@ -18,6 +18,7 @@
 #include "midiobjectwin.h"
 #include "midiledhandler.h"
 #include <atlconv.h>
+#include <QtCore>
 #include <QtDebug>
 
 #ifdef __WIN__
@@ -124,8 +125,11 @@
 
 void MidiObjectWin::handleMidi(char channel, char midicontrol, char midivalue)
 {
-    qDebug() << QString("midi miditype: ") << QString(channel& 240) << QString(" ch: ") << QString(channel&15) << QString(", ctrl: ") << QString(midicontrol) << QString(", val: ") << QString(midivalue);
-    send((MidiCategory)(channel & 240), channel&15, midicontrol, midivalue);
+    qDebug() << QString("midi miditype: %1 ch: %2, ctrl: %3, val: %4").arg(QString::number(channel& 240, 16).toUpper())
+       .arg(QString::number(channel&15, 16).toUpper())
+       .arg(QString::number(midicontrol, 16).toUpper())
+       .arg(QString::number(midivalue, 16).toUpper());
+    receive((MidiCategory)(channel & 240), channel&15, midicontrol, midivalue, device); // void receive(MidiCategory category, char channel, char control, char value, QString device);
 }
 
 // C/C++ wrapper function
diff -Nru mixxx-1.6.0/src/mixxx.cpp mixxx-1.6.1/src/mixxx.cpp
--- mixxx-1.6.0/src/mixxx.cpp	2008-07-15 22:46:38.000000000 -0500
+++ mixxx-1.6.1/src/mixxx.cpp	2008-09-23 01:52:28.000000000 -0500
@@ -41,7 +41,6 @@
 #include "log.h"
 #include "dlgabout.h"
 
-#include "playerproxy.h"
 #include "soundmanager.h"
 #include "defs_urls.h"
 #include "defs_audiofiles.h"
@@ -93,7 +92,6 @@
 #endif
 
     //Reset pointer to players
-    //player = 0;
     soundmanager = 0;
     m_pTrack = 0;
     prefDlg = 0;
@@ -204,8 +202,6 @@
     master = new EngineMaster(config, buffer1, buffer2, channel1, channel2, "[Master]");
 
     // Initialize player device
-    //Player::setMaster(master);
-    //player = new PlayerProxy(config);
 
     soundmanager = new SoundManager(config, master);
     soundmanager->queryDevices();
@@ -242,7 +238,7 @@
 
     view=new MixxxView(frame, kbdconfig, qSkinPath, config);
 
-    // TODO : Move this to WaveformViewerFactory or something.
+    // TODO rryan : Move this to WaveformViewerFactory or something.
     /*
     if (bVisualsWaveform && !view->activeWaveform())
     {
@@ -255,20 +251,6 @@
     }
     */
 
-    // Tell EngineBuffer to notify the visuals if they are WVisualWaveform
-    if (view->activeWaveform())
-    {
-        // Dynamic zoom on visuals
-        if (view->m_bZoom)
-        {
-            // TODO rryan : USE THESE CONTROLOBJECTS IN THE NEWVISUAL CODE
-            //ControlObject::connectControls(ConfigKey("[Channel1]", "rate"), ConfigKey("[Channel1]", "VisualLengthScale-marks"));
-            //ControlObject::connectControls(ConfigKey("[Channel1]", "rate"), ConfigKey("[Channel1]", "VisualLengthScale-signal"));
-            //ControlObject::connectControls(ConfigKey("[Channel2]", "rate"), ConfigKey("[Channel2]", "VisualLengthScale-marks"));
-            //ControlObject::connectControls(ConfigKey("[Channel2]", "rate"), ConfigKey("[Channel2]", "VisualLengthScale-signal"));
-        }
-    }
-
     // Verify path for xml track file.
     QFile trackfile(config->getValueString(ConfigKey("[Playlist]","Listfile")));
     if ((config->getValueString(ConfigKey("[Playlist]","Listfile")).length()<1) || (!trackfile.exists()))
@@ -334,9 +316,6 @@
 #endif
 
     // Try open player device If that fails, the preference panel is opened.
-    //if (!player->open())
-    //    prefDlg->setHidden(false);
-
     if (soundmanager->setupDevices() != 0)
     {
 
@@ -377,15 +356,7 @@
     // Initialize visualization of temporal effects
     channel1->setEngineBuffer(buffer1);
     channel2->setEngineBuffer(buffer2);
-
-    // Dynamic scaling of temporal effect curves
-    if (view->m_bZoom)
-    {
-        // TODO rryan, use these!
-        //ControlObject::connectControls(ConfigKey("[Channel1]", "rate"), ConfigKey("[Channel1]", "VisualLengthScale-temporal"));
-        //ControlObject::connectControls(ConfigKey("[Channel2]", "rate"), ConfigKey("[Channel2]", "VisualLengthScale-temporal"));
-    }
-
+    
 #ifdef __SCRIPT__
     scriptEng = new ScriptEngine(this, m_pTrack);
 #endif
@@ -437,10 +408,6 @@
     qDebug() << "Write track xml, " << qTime.elapsed();
     m_pTrack->writeXML(config->getValueString(ConfigKey("[Playlist]","Listfile")));
 
-    //qDebug() << "close player, " << qTime.elapsed();
-    //player->close();
-    //qDebug() << "player->close() done";
-
     qDebug() << "close soundmanager" << qTime.elapsed();
     soundmanager->closeDevices();
     qDebug() << "soundmanager->close() done";
@@ -449,8 +416,6 @@
     config->set(ConfigKey("[Controls]","TrackEndModeCh1"), ConfigValue((int)ControlObject::getControl(ConfigKey("[Channel1]","TrackEndMode"))->get()));
     config->set(ConfigKey("[Controls]","TrackEndModeCh2"), ConfigValue((int)ControlObject::getControl(ConfigKey("[Channel2]","TrackEndMode"))->get()));
 
-    //qDebug() << "delete player, " << qTime.elapsed();
-    //delete player;
     qDebug() << "delete soundmanager, " << qTime.elapsed();
     delete soundmanager;
     qDebug() << "delete master, " << qTime.elapsed();
@@ -521,6 +486,10 @@
     playlistsImport->setShortcut(tr("Ctrl+I"));
     playlistsImport->setShortcutContext(Qt::ApplicationShortcut);
 
+    batchBpmDetect = new QAction(tr("Batch BPM Detect (up to 100 songs per run)"), this);
+    connect(batchBpmDetect, SIGNAL(activated()), m_pTrack, SLOT(slotBatchBPMDetection()));
+
+
 #ifdef __IPOD__
     iPodToggle = new QAction(tr("iPod &Active"), this);
     iPodToggle->setShortcut(tr("Ctrl+A"));
@@ -676,6 +645,7 @@
     libraryMenu->addSeparator();
     libraryMenu->addAction(playlistsNew);
     libraryMenu->addAction(playlistsImport);
+    //libraryMenu->addAction(batchBpmDetect);
 
 #ifdef __IPOD__
     libraryMenu->addSeparator();
@@ -888,7 +858,8 @@
 
 void MixxxApp::slotOptionsFullScreen(bool toggle)
 {
-    // Making a fullscreen window on linux and windows is harder than you could possibly imagine...
+
+// Making a fullscreen window on linux and windows is harder than you could possibly imagine...
     if (toggle)
     {
 #ifdef __LINUX__
@@ -1076,6 +1047,8 @@
 "Ulrich Heske<br>"
 "James Hagerman<br>"
 "quil0m80<br>"
+"Michael Pujos<br>"
+"Mark Glines<br>"
 "</p>"
 "<p align=\"center\"><b>And special thanks to:</b></p>"
 "<p align=\"center\">"
diff -Nru mixxx-1.6.0/src/mixxx.h mixxx-1.6.1/src/mixxx.h
--- mixxx-1.6.0/src/mixxx.h	2008-07-13 22:21:19.000000000 -0500
+++ mixxx-1.6.1/src/mixxx.h	2008-08-14 00:23:43.000000000 -0500
@@ -159,7 +159,6 @@
     
     EngineChannel *channel1, *channel2;
     EngineMaster *master;
-    //PlayerProxy *player;
     SoundManager *soundmanager;
     MidiObject *midi;
     ControlObject *control;
@@ -212,6 +211,8 @@
 
     QAction *iPodToggle;
 
+    QAction *batchBpmDetect;
+
     QAction *libraryRescan;
 
     QAction *optionsBeatMark;
diff -Nru mixxx-1.6.0/src/mixxx.vcproj mixxx-1.6.1/src/mixxx.vcproj
--- mixxx-1.6.0/src/mixxx.vcproj	2007-06-27 16:37:06.000000000 -0500
+++ mixxx-1.6.1/src/mixxx.vcproj	2008-08-11 05:02:03.000000000 -0500
@@ -530,18 +530,6 @@
 				>
 			</File>
 			<File
-				RelativePath="player.cpp"
-				>
-			</File>
-			<File
-				RelativePath="playerportaudio.cpp"
-				>
-			</File>
-			<File
-				RelativePath="playerproxy.cpp"
-				>
-			</File>
-			<File
 				RelativePath="powermate.cpp"
 				>
 			</File>
@@ -1835,18 +1823,6 @@
 				>
 			</File>
 			<File
-				RelativePath="player.h"
-				>
-			</File>
-			<File
-				RelativePath="playerportaudio.h"
-				>
-			</File>
-			<File
-				RelativePath="playerproxy.h"
-				>
-			</File>
-			<File
 				RelativePath="..\..\mixxx-winlib\portaudio.h"
 				>
 			</File>
diff -Nru mixxx-1.6.0/src/mixxxview.cpp mixxx-1.6.1/src/mixxxview.cpp
--- mixxx-1.6.0/src/mixxxview.cpp	2008-07-29 10:16:10.000000000 -0500
+++ mixxx-1.6.1/src/mixxxview.cpp	2008-09-16 21:12:45.000000000 -0500
@@ -99,7 +99,6 @@
     m_pNumberPosCh2 = 0;
     m_pSliderRateCh1 = 0;
     m_pSliderRateCh2 = 0;
-    m_bZoom = false;
     m_bVisualWaveform = false;
     m_pOverviewCh1 = 0;
     m_pOverviewCh2 = 0;
@@ -590,8 +589,6 @@
 		    ((QWidget*)m_pVisualCh2)->repaint();
 		}
 		
-                if (!WWidget::selectNode(node, "Zoom").isNull() && WWidget::selectNodeQString(node, "Zoom")=="true")
-                    m_bZoom = true;
             }
 
 
diff -Nru mixxx-1.6.0/src/mixxxview.h mixxx-1.6.1/src/mixxxview.h
--- mixxx-1.6.0/src/mixxxview.h	2008-07-13 22:21:19.000000000 -0500
+++ mixxx-1.6.1/src/mixxxview.h	2008-09-16 21:12:45.000000000 -0500
@@ -74,8 +74,6 @@
     WNumberPos *m_pNumberPosCh1, *m_pNumberPosCh2;
     /** Pointer to rate slider widgets */
     WSliderComposed *m_pSliderRateCh1, *m_pSliderRateCh2;
-    /** Allow dynamic zoom on visuals */
-    bool m_bZoom;
     /** Pointer to ComboBox*/
     QComboBox *m_pComboBox;
     //WComboBox *m_pComboBox;
diff -Nru mixxx-1.6.0/src/old/playeralsa.cpp mixxx-1.6.1/src/old/playeralsa.cpp
--- mixxx-1.6.0/src/old/playeralsa.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playeralsa.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,757 @@
+/***************************************************************************
+                          playeralsa.cpp  -  description
+                             -------------------
+    begin                : Wed Feb 20 2002
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+    email                :
+***************************************************************************/
+
+/***************************************************************************
+*                                                                         *
+*   This program is free software; you can redistribute it and/or modify  *
+*   it under the terms of the GNU General Public License as published by  *
+*   the Free Software Foundation; either version 2 of the License, or     *
+*   (at your option) any later version.                                   *
+*                                                                         *
+***************************************************************************/
+
+#include "playeralsa.h"
+#include "controlobject.h"
+
+#ifndef PLAYERTEST
+#include "rtthread.h"
+#endif
+
+// Tutorials /home/peter/pad-alsa-audio.html /home/peter/alsa090_howto.html
+// Docs      /usr/share/doc/alsa-lib-1.0.4/doxygen/html/index.html
+// Example   /usr/share/doc/alsa-lib-1.0.4/doxygen/html/_2test_2pcm_8c-example.html
+// QT        /usr/share/doc/qt-devel-3.1.1/html/index.html
+
+/** Maximum frame size used with ALSA. Used to determine no of buffers
+ *   * when setting latency */
+const int kiMaxFrameSize = 512;
+
+
+#ifndef PLAYERTEST
+PlayerALSA::PlayerALSA(ConfigObject<ConfigValue> * config) : Player(config)
+#else
+PlayerALSA::PlayerALSA()
+#endif
+{
+    int err;
+
+    handle = 0;
+    err = snd_pcm_hw_params_malloc(&hwparams);
+    if (err < 0)
+        qCritical("Couldn't allocate memory for hw params: %s\n", snd_strerror(err));
+
+    err = snd_pcm_sw_params_malloc(&swparams);
+    if (err < 0)
+        qCritical("Couldn't allocate memory for sw params: %s\n", snd_strerror(err));
+
+    err = snd_pcm_sw_params_malloc(&swparams);
+    if (err < 0)
+        qCritical("Couldn't allocate memory for sw params: %s\n", snd_strerror(err));
+
+    isopen = false;
+    masterleft = masterright = -1;
+    headleft = headright = -1;
+    twrite = false;
+    m_iChannels = alsa_channels;
+    qDebug() << "Alsa constructed";
+}
+
+PlayerALSA::~PlayerALSA()
+{
+    if (isopen) close();
+
+    qDebug() << "Alsa destroying...";
+    if (hwparams)
+    {
+        snd_pcm_hw_params_free(hwparams);
+    }
+    if (swparams)
+    {
+        snd_pcm_sw_params_free(swparams);
+    }
+    qDebug() << "Alsa destroyed";
+}
+
+bool PlayerALSA::initialize()
+{
+    qDebug() << "Alsa initialized";
+
+    return true;
+}
+
+bool PlayerALSA::open()
+{
+#ifndef PLAYERTEST
+    Player::open();
+#endif
+    QString name;
+    QString devname, devtmp;
+    int err;
+
+#ifndef PLAYERTEST
+    QRegExp rx("(\\S+) \\(ch (\\d+)\\)");
+
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceMasterLeft"));
+    if (name != "None")
+    {
+        if (rx.search(name) < 0)
+            qWarning() << "can't find device name or channel number in" << name;
+
+        devname = rx.cap(1);
+    }
+    else
+    {
+        setDefaults(); // initialise defaults
+        return open();
+    }
+#else
+    devname = QString("surround40:0");
+#endif
+
+    // If no device was selected return false
+    if (!devname)
+        return false;
+
+    qDebug() << "Alsa opening pcm_open:" << devname;
+
+    if ((err = snd_pcm_open(&handle, devname.ascii(),
+                            SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+        // AD: Try for default device instead
+        if ((err = snd_pcm_open(&handle, "default",
+                                SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+            qWarning("Playback open error: %s\n", snd_strerror(err));
+            return false;
+        } else {
+            qWarning("Using default ALSA device, you may want to set up a ~/.asoundrc to enable all the features of your sound card");
+        }
+    }
+
+    qDebug() << "Alsa setting hw";
+    if ((err = set_hwparams()) < 0)
+    {
+        qWarning("Setting of hwparams failed: %s\n", snd_strerror(err));
+        snd_pcm_close(handle);
+        return false;
+    }
+
+    qDebug() << "Alsa setting sw";
+    if ((err = set_swparams()) < 0)
+    {
+        qWarning("Setting of swparams failed: %s\n", snd_strerror(err));
+        snd_pcm_close(handle);
+        return false;
+    }
+
+    masterleft = masterright = -1;
+    headleft = headright = -1;
+
+#ifndef PLAYERTEST
+    int temp;
+
+    // master left
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceMasterLeft"));
+    qDebug() << "Alsa opening..." << name;
+    if (name != "None")
+    {
+        if (rx.search(name) < 0)
+        {
+            qWarning() << "can't find device name or channel number in" << name;
+        }
+//      devname = rx.cap(1);
+        temp = rx.cap(2).toInt();
+        if (temp > 0 && temp <= m_iChannels)
+        {
+            masterleft = temp - 1;
+        }
+    }
+    qDebug() << "Alsa opening...ML " << masterleft;
+
+    // master right
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceMasterRight"));
+    if (name != "None")
+    {
+        if (rx.search(name) < 0)
+        {
+            qWarning() << "can't find device name or channel number in" << name;
+        }
+        devtmp = rx.cap(1);
+        if (devname != devtmp)
+        {
+            qWarning("device name mismatch, only one device supported");
+        }
+        temp = rx.cap(2).toInt();
+        if (temp > 0 && temp <= m_iChannels)
+        {
+            masterright = temp - 1;
+        }
+    }
+    qDebug() << "Alsa opening...MR " << masterright;
+
+    // head left
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceHeadLeft"));
+    if (name != "None")
+    {
+        if (rx.search(name) < 0)
+        {
+            qWarning() << "can't find device name or channel number in" << name;
+        }
+        devtmp = rx.cap(1);
+        if (devname != devtmp)
+        {
+            qWarning("device name mismatch, only one device supported");
+        }
+        temp = rx.cap(2).toInt();
+        if (temp > 0 && temp <= m_iChannels)
+        {
+            headleft = temp - 1;
+        }
+    }
+    qDebug() << "Alsa opening...HL " << headleft;
+
+    // head right
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceHeadRight"));
+    if (name != "None")
+    {
+        if (rx.search(name) < 0)
+        {
+            qWarning() << "can't find device name or channel number in" << name;
+        }
+        devtmp = rx.cap(1);
+        if (devname != devtmp)
+        {
+            qWarning("device name mismatch, only one device supported");
+        }
+        temp = rx.cap(2).toInt();
+        if (temp > 0 && temp <= m_iChannels)
+        {
+            headright = temp - 1;
+        }
+    }
+
+    qDebug() << "Alsa opening...HR " << headright;
+    // Check if any of the devices in the config database needs to be set to "None"
+    if (masterleft < 0)
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterLeft"), ConfigValue("None"));
+    if (masterright < 0)
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterRight"), ConfigValue("None"));
+    if (headleft < 0)
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadLeft"), ConfigValue("None"));
+    if (headright < 0)
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadRight"), ConfigValue("None"));
+
+#else
+    masterleft = 0;
+    masterright = 1;
+    headleft = 2;
+    headright = 3;
+#endif
+
+    twrite = true;
+    isopen = true;
+
+    QThread::start();
+
+    return true;
+}
+
+/*
+ *   Underrun and suspend recovery
+ */
+int PlayerALSA::xrun_recovery(int err)
+{
+    if (err == -EPIPE) {        /* under-run */
+        err = snd_pcm_prepare(handle);
+        if (err < 0)
+            qWarning("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err));
+        return 0;
+    }
+    else if (err == -ESTRPIPE)
+    {
+        while ((err = snd_pcm_resume(handle)) == -EAGAIN)
+            sleep(1);/* wait until the suspend flag is released */
+        if (err < 0)
+        {
+            err = snd_pcm_prepare(handle);
+            if (err < 0)
+                qWarning("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err));
+        }
+        return 0;
+    }
+    return err;
+}
+
+/**
+ * thread for writing samples out
+ */
+void PlayerALSA::run()
+{
+    CSAMPLE * sptr;
+    int err, ctr;
+
+#ifndef PLAYERTEST
+    rtThread();
+#endif
+
+    if (isformatfloat) {
+        float * optr, * output;
+
+        output = new float[buffer_size * m_iChannels];
+        qDebug() << "Alsa allocating output buffer (FLOAT) " << output;
+
+        while (twrite)
+        {
+            sptr = prepareBuffer((int) period_size);
+            optr = output;
+            for (int i = 0; i < (int) period_size; i++)
+            {
+                for (int j = 0; j < m_iChannels; j++) optr[j] = 0;
+                if (masterleft >= 0) optr[masterleft]  = *sptr / 32768.;
+                sptr++;
+                if (masterright >= 0) optr[masterright] = *sptr / 32768.;
+                sptr++;
+                if (headleft >= 0) optr[headleft]  = *sptr / 32768.;
+                sptr++;
+                if (headright >= 0) optr[headright] = *sptr / 32768.;
+                sptr++;
+                optr += m_iChannels;
+            }
+            ctr = period_size;
+            optr = output;
+            while (ctr > 0)
+            {
+                err = snd_pcm_writei(handle, optr, ctr);
+                if (err == -EAGAIN) continue;
+                if (err < 0)
+                {
+                    if (xrun_recovery(err) < 0)
+                        qCritical("Write error: %s\n", snd_strerror(err));
+
+                    break;
+                }
+                optr += err*m_iChannels;
+                ctr -= err;
+            }
+        }
+        if (output)
+            delete output;
+    }
+    else
+    {
+        short int * optr, * output;
+
+        output = new short int[buffer_size * m_iChannels];
+        qDebug() << "Alsa allocating output buffer (S16) " << output;
+
+        while (twrite)
+        {
+            sptr = prepareBuffer((int) period_size);
+            optr = output;
+            for (int i = 0; i < (int) period_size; i++)
+            {
+                for (int j = 0; j < m_iChannels; j++) optr[j] = 0;
+                if (masterleft >= 0) optr[masterleft]  = (short int) *sptr;
+                sptr++;
+                if (masterright >= 0) optr[masterright] = (short int) *sptr;
+                sptr++;
+                if (headleft >= 0) optr[headleft]  = (short int) *sptr;
+                sptr++;
+                if (headright >= 0) optr[headright] = (short int) *sptr;
+                sptr++;
+                optr += m_iChannels;
+            }
+            ctr = period_size;
+            optr = output;
+            while (ctr > 0)
+            {
+                err = snd_pcm_writei(handle, optr, ctr);
+                if (err == -EAGAIN) continue;
+                if (err < 0)
+                {
+                    if (xrun_recovery(err) < 0)
+                        qCritical("Write error: %s\n", snd_strerror(err));
+                    break;
+                }
+                optr += err*m_iChannels;
+                ctr -= err;
+            }
+        }
+        if (output) delete output;
+    }
+    qDebug() << "Alsa leaving thread";
+}
+
+#ifdef PLAYERTEST
+/**
+ * generate a middle A slowly shifting across the channels
+ */
+CSAMPLE * PlayerALSA::prepareBuffer(int count)
+{
+    const double freq = 440.;
+    const unsigned int rate = 44100;
+    static CSAMPLE * out = 0;
+
+    static double phase = 0;
+    double max_phase = 1.0 / freq;
+    double step = 1.0 / (double) rate;
+    double res;
+    CSAMPLE * sptr;
+    int chn;
+    static int channel = 0;
+    static int ccount = 0;
+
+    if (out == 0)
+        out = new CSAMPLE[80000];
+    sptr = out;
+    while (count-- > 0)
+    {
+        res = 32768. * sin((phase * 2 * M_PI) / max_phase - M_PI);
+        for (chn = 0; chn < alsa_channels; chn++)
+        {
+            if (chn == channel)
+                *sptr = res;
+            else
+                *sptr = 0;
+            sptr++;
+        }
+        phase += step;
+        if (phase >= max_phase)
+        {
+            phase -= max_phase;
+            ccount++;
+            if (ccount == (3 * 440))
+            {
+                ccount = 0;
+                channel++;
+                if (channel == alsa_channels) channel = 0;
+            }
+        }
+    }
+    return out;
+}
+#endif
+
+void PlayerALSA::close()
+{
+    if (!isopen)
+        return;
+
+    qDebug() << "Alsa closing";
+    twrite = false;
+    qDebug() << "Alsa waiting for thread for close";
+    QThread::wait();
+
+    snd_pcm_drop(handle);
+    snd_pcm_close(handle);
+
+    isopen = false;
+}
+
+void PlayerALSA::setDefaults()
+{
+    qDebug() << "Alsa setdefs";
+#ifndef PLAYERTEST
+    // Get list of interfaces
+    QStringList interfaces = getInterfaces();
+
+    // Set first interfaces to master left
+    QStringList::iterator it = interfaces.begin();
+    if (*it)
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterLeft"), ConfigValue((*it)));
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterLeft"), ConfigValue("None"));
+
+    // Set second interface to master right
+    ++it;
+    if (*it)
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterRight"), ConfigValue((*it)));
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterRight"), ConfigValue("None"));
+
+    // Set head left and right to none
+    m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadLeft"), ConfigValue("None"));
+    m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadRight"), ConfigValue("None"));
+
+    // Set default sample rate
+    QStringList srates = getSampleRates();
+    it = srates.begin();
+    while (*it)
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]", "Samplerate"), ConfigValue((*it)));
+        if ((*it).toInt() >= 44100)
+            break;
+        ++it;
+    }
+
+    // Set currently used latency in config database
+    m_pConfig->set(ConfigKey("[Soundcard]", "Latency"), ConfigValue(default_latency));
+#endif
+}
+
+QStringList PlayerALSA::getInterfaces()
+{
+    qDebug() << "Alsa getinter";
+    QStringList result;
+
+    result.append("mixxx (ch 1)");
+    result.append("mixxx (ch 2)");
+    result.append("mixxx (ch 3)");
+    result.append("mixxx (ch 4)");
+
+    return result;
+}
+
+QStringList PlayerALSA::getSampleRates()
+{
+    qDebug() << "Alsa getsample";
+    QStringList result;
+    result.append("11025");
+    result.append("22050");
+    result.append("44100");
+    result.append("48000");
+    return result;
+}
+
+QString PlayerALSA::getSoundApi()
+{
+    qDebug() << "Alsa getapi";
+    return QString("Alsa");
+}
+
+int PlayerALSA::set_hwparams()
+{
+    unsigned int rate, rrate;
+    unsigned int buffer_time = 500000;  /* ring buffer length in us */
+    unsigned int period_time = 100000;  /* period time in us */
+
+    int err, dir;
+
+    /* choose all parameters */
+    err = snd_pcm_hw_params_any(handle, hwparams);
+    if (err < 0)
+    {
+        qWarning("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* set the interleaved read/write format */
+    err = snd_pcm_hw_params_set_access(handle, hwparams, alsa_access);
+    if (err < 0)
+    {
+        qWarning("Access type not available for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* set the sample format */
+    err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_FLOAT);
+    if (err < 0)
+    {
+        qWarning("Sample format (FLOAT) not available for playback: %s\n", snd_strerror(err));
+        qWarning("Falling back to S16");
+        err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16);
+        if (err < 0)
+        {
+            qWarning("Sample format (S16) not available for playback: %s\n", snd_strerror(err));
+            return err;
+        }
+        isformatfloat = false;
+    } else {
+        isformatfloat = true;
+    }
+
+    /* find min and max number of channels */
+    err = snd_pcm_hw_params_get_channels_min(hwparams, &rate);
+    if (err < 0)
+    {
+        qWarning("Channels count (%i) not available for playbacks: %s\n", rate, snd_strerror(err));
+    }
+    qDebug() << "Channels min " << rate;
+    err = snd_pcm_hw_params_get_channels_max(hwparams, &rate);
+    if (err < 0)
+    {
+        qWarning("Channels count (%i) not available for playbacks: %s\n", rate, snd_strerror(err));
+    }
+    qDebug() << "Channels max " << rate;
+    m_iChannels = math_min(rate, alsa_channels);
+    qDebug() << "Set channels = " << m_iChannels;
+
+    /* set the count of channels */
+    err = snd_pcm_hw_params_set_channels(handle, hwparams, m_iChannels);
+    if (err < 0)
+    {
+        qWarning("Channels count (%i) not available for playbacks: %s\n", m_iChannels, snd_strerror(err));
+        return err;
+    }
+    /* set the stream rate */
+#ifndef PLAYERTEST
+    rate = m_pConfig->getValueString(ConfigKey("[Soundcard]", "Samplerate")).toUInt();
+#else
+    rate = 44100;
+#endif
+    rrate = rate;
+    err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rrate, 0);
+    if (err < 0)
+    {
+        qWarning("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err));
+        return err;
+    }
+    if (rrate != rate)
+    {
+        qWarning("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err);
+        return -EINVAL;
+    }
+
+#ifndef PLAYERTEST
+//     setPlaySrate(rrate);
+//     m_pControlObjectSampleRate->queueFromThread((double)rrate);
+
+    // Update SRATE and Latency ControlObjects
+    m_pControlObjectSampleRate->queueFromThread((double)rrate);
+//     m_pControlObjectLatency->queueFromThread((double)iLatencyMSec);
+//
+#endif
+
+    /* set the buffer time */
+#ifndef PLAYERTEST
+    buffer_time = m_pConfig->getValueString(ConfigKey("[Soundcard]", "Latency")).toUInt() * 1000;
+#else
+    buffer_time = 50000;
+#endif
+    /* figure out size and make it even */
+    buffer_size = (int) rrate*(1e-6*buffer_time) + 1;
+
+    snd_pcm_uframes_t max = 0;
+    snd_pcm_uframes_t min = 0;
+
+    // FIXME: Should check return values to be on the safe side
+    snd_pcm_hw_params_get_buffer_size_max(hwparams, &max);
+    snd_pcm_hw_params_get_buffer_size_min(hwparams, &min);
+    qDebug() << "Allowed buffer size range: " << min << "i -> " << max << "i";
+
+    // Make sure buffer size we're aiming for is really allowed
+    if (buffer_size < min) { buffer_size = (int)min; }
+    if (buffer_size > max) { buffer_size = (int)max; }
+
+    buffer_size -= (buffer_size & 1);
+
+    if (buffer_size < min) { buffer_size += 2; }
+
+    /* test buffer size first */
+    err = snd_pcm_hw_params_test_buffer_size(handle, hwparams, buffer_size);
+    if (err < 0) {
+        qWarning("Unable to test buffer size %d", buffer_size);
+        return err;
+    }
+    err = snd_pcm_hw_params_set_buffer_size(handle, hwparams, buffer_size);
+    if (err < 0)
+    {
+        qWarning("Unable to set buffer size for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    qWarning("Buffer %dus [actual %dus] (size: %d)", (int) buffer_time, (int) (1e6*buffer_size/rrate), buffer_size);
+
+    /*
+     * work out maximum number of periods so that only kiMaxFrameSize is
+     * written out at a time, then use this as a starting point to find
+     * period that works
+     */
+    int period_no_start = 2;
+
+    if (buffer_size/kiMaxFrameSize>2)
+        period_no_start = buffer_size/kiMaxFrameSize;
+
+    for (period_no = period_no_start; period_no > 1; period_no--)
+    {
+        unsigned int period_time = buffer_time / period_no;
+        qWarning("Period %dus (no: %d)", (int) period_time, period_no);
+
+        int dir = 0;
+        err = snd_pcm_hw_params_test_periods(handle, hwparams, period_no, dir);
+        if (err < 0)
+        {
+            qWarning("Unable to test periods %i (dir %d) for playback: %s\n", period_no, dir,
+                     snd_strerror(err));
+        }
+        else
+        {
+            err = snd_pcm_hw_params_set_periods(handle, hwparams, period_no, dir);
+            if (err < 0)
+            {
+                qWarning("Unable to set periods %i for playback: %s\n", period_no, snd_strerror(err));
+                return err;
+            }
+            err = snd_pcm_hw_params_get_period_size(hwparams, &period_size, &dir);
+            if (err < 0)
+            {
+                qWarning("Unable to get period size for playback: %s\n", snd_strerror(err));
+                return err;
+            }
+            qWarning("Period %df, buffer %df", (int) period_size, (int) buffer_size);
+            break;
+        }
+    }
+
+    if (period_no == 1)
+        return -1;
+
+    /* write the parameters to device */
+    err = snd_pcm_hw_params(handle, hwparams);
+    if (err < 0)
+    {
+        qWarning("Unable to set hw params for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    qWarning("Rate: %d, buf %df (%dus), per %df (%dus)", rrate,
+             (int) buffer_size, (int) (buffer_size / (rrate * 1e-6)),
+             (int) period_size, (int) (period_size / (rrate * 1e-6)));
+
+    // Update SRATE and Latency ControlObjects
+    m_pControlObjectSampleRate->queueFromThread((double)rrate);
+    m_pControlObjectLatency->queueFromThread((double)buffer_time/1000.);
+
+    return 0;
+}
+
+int PlayerALSA::set_swparams()
+{
+    int err;
+
+    /* get the current swparams */
+    err = snd_pcm_sw_params_current(handle, swparams);
+    if (err < 0)
+    {
+        qWarning("Unable to determine current swparams for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* start the transfer when the buffer is full */
+    err = snd_pcm_sw_params_set_start_threshold(handle, swparams, (buffer_size/period_size)*period_size);
+    if (err < 0)
+    {
+        qWarning("Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* allow the transfer when at least period_size samples can be processed */
+    err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size);
+    if (err < 0)
+    {
+        qWarning("Unable to set avail min for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* align all transfers to 1 sample */
+    err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
+    if (err < 0)
+    {
+        qWarning("Unable to set transfer align for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    /* write the parameters to the playback device */
+    err = snd_pcm_sw_params(handle, swparams);
+    if (err < 0)
+    {
+        qWarning("Unable to set sw params for playback: %s\n", snd_strerror(err));
+        return err;
+    }
+    return 0;
+}
diff -Nru mixxx-1.6.0/src/old/playeralsa.h mixxx-1.6.1/src/old/playeralsa.h
--- mixxx-1.6.0/src/old/playeralsa.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playeralsa.h	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,107 @@
+/***************************************************************************
+                          playeralsa.h  -  description
+                             -------------------
+    begin                : Wed Feb 20 2002
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+    email                : 
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#ifndef PLAYERALSA_H
+#define PLAYERALSA_H
+
+/**
+  *@author Tue and Ken Haste Andersen, Peter Chang
+  */
+
+#ifndef PLAYERTEST
+#include "player.h"
+#else
+// some definitions to avoid using player.h in test mode
+
+#include <math.h>
+
+#include <qstring.h>
+#include <qstringlist.h>
+
+typedef float CSAMPLE; // defines the CSAMPLE type used for intermediate calculations
+#endif
+
+#include <qregexp.h>
+#include <qthread.h>
+#include <alsa/asoundlib.h>
+
+/**
+ * there are two independent defines S16_OUTPUT and DIRECT_OUTPUT
+ *  1) S16_OUTPUT makes the class output short integers to ALSA
+ *  2) DIRECT_OUTPUT uses mmapped output
+ * 
+ * NB by default, these are not defined and so the class uses floats via the
+ * plug plugin. Defining S16_OUTPUT allows the bare hw to be used (via surround40).
+ * XXX: Mmapped output doesn't seem to work with the plug plugin...
+ */
+
+#ifndef PLAYERTEST
+class PlayerALSA : public Player, public QThread 
+{
+public:
+    PlayerALSA(ConfigObject<ConfigValue> *config);
+#else
+class PlayerALSA : public QThread {
+public:
+    PlayerALSA();
+#endif
+    ~PlayerALSA();
+    bool initialize();
+    bool open();
+    void close();
+    void setDefaults();
+    QStringList getInterfaces();
+    QStringList getSampleRates();
+    static QString getSoundApi();
+    QString getSoundApiName() { return getSoundApi(); };
+    /** Satisfy virtual declaration in EngineObject */
+    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
+
+    /** Main loop of player. Executed in a separate thread by QT */
+    void run();
+
+protected:
+/** ALSA parameters */
+    snd_pcm_t *handle;
+    snd_pcm_hw_params_t *hwparams;
+    snd_pcm_sw_params_t *swparams;
+
+    snd_pcm_uframes_t buffer_size;
+    snd_pcm_uframes_t period_size;
+    unsigned int period_no;
+    bool isformatfloat;
+
+    /** True if devices are open */
+    bool isopen;
+    int masterleft, masterright, headleft, headright;
+
+#ifdef PLAYERTEST
+    CSAMPLE *prepareBuffer(int nframes);
+#endif
+    int setPeriodSize(bool setMinimum);
+    int set_hwparams();
+    int set_swparams();
+    int xrun_recovery(int err);
+
+    bool twrite; // flag for thread
+
+private:
+    static const snd_pcm_access_t alsa_access = SND_PCM_ACCESS_RW_INTERLEAVED;
+    static const int alsa_channels = 4;
+    static const int default_latency = 200;
+};
+#endif
diff -Nru mixxx-1.6.0/src/old/playerasio.cpp mixxx-1.6.1/src/old/playerasio.cpp
--- mixxx-1.6.0/src/old/playerasio.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerasio.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,670 @@
+/***************************************************************************
+                          PlayerAsio.cpp  -  description
+                             -------------------
+    begin                : Thu June 02 2005
+    copyright            : (C) 2005 by Ingo Kossyk
+    email                :
+***************************************************************************/
+
+/***************************************************************************
+*                                                                         *
+*   This program is free software; you can redistribute it and/or modify  *
+*   it under the terms of the GNU General Public License as published by  *
+*   the Free Software Foundation; either version 2 of the License, or     *
+*   (at your option) any later version.                                   *
+*                                                                         *
+***************************************************************************/
+
+
+// PlayerAsio.cpp: implementation of the PlayerAsio class.
+//
+//////////////////////////////////////////////////////////////////////
+
+//////////////////////////////////////////////////////////////////////
+// ToDo:
+// - Implement all conversion routines for the different drivers
+// - Testing on different soundcards
+// - Correctly (???) setting the latency in the preferences and locking it
+// - Samplerate Changed needs to be implemented if changed in the Preferences
+//////////////////////////////////////////////////////////////////////
+
+#include "qstringlist.h"
+#include "PlayerAsio.h"
+#include "playerproxy.h"
+#include "controlobject.h"
+//////////////////////////////////////////////////////////////////////
+// External Declarations
+//////////////////////////////////////////////////////////////////////
+
+__declspec(dllimport) AsioDrivers* asioDrivers;
+ASIOCallbacks asioCallbacks;
+DriverInfo asioDriverInfo;
+
+//////////////////////////////////////////////////////////////////////
+// Construction/Destruction
+//////////////////////////////////////////////////////////////////////
+
+const int MAX_DRIVER_NAME_LENGTH = 32;
+
+PlayerAsio::PlayerAsio(ConfigObject<ConfigValue> * config) : Player(config)
+
+{
+    m_pTempBuffer = 0;
+    m_reenter = 0;
+    bufferLength = 0;
+    driverIsLoaded = false;
+
+    for (int i = 0 ; i < 32 ; ++i) {
+        driverList[i] =static_cast<char *>(malloc(32));
+
+        /* check memory */
+        if(!driverList[i]) qDebug() << "ERROR: Could not allocate Memory!";
+    }
+
+
+    loadAsioDriver("dummy");
+
+    drvNum = asioDrivers->getDriverNames(driverList,32);
+
+}
+
+PlayerAsio::~PlayerAsio()
+{
+
+    ASIOStop();
+    ASIOExit();
+
+}
+//////////////////////////////////////////////////////////////////////
+// Class functions
+//////////////////////////////////////////////////////////////////////
+
+/*******
+ *
+ * Conversion of different Ouputtypes
+ *
+ ********/
+const double fScaler24 = (double)0xffff;
+
+void PlayerAsio::float32toInt24inPlace(float * buffer, long frames, bool swap)
+{
+    double sc = fScaler24 + .5;
+    long a;
+    char * b = (char *)(buffer);
+    char * aa = (char *)(&a);
+
+    while(--frames >= 0)
+    {
+        a = static_cast<long>(static_cast<double>(*buffer++) *sc);
+
+
+        if (swap)
+        {
+            *b++ = aa[3];
+            *b++ = aa[2];
+            *b++ = aa[1];
+        }
+        else
+        {
+            *b++ = aa[1];
+            *b++ = aa[2];
+            *b++ = aa[3];
+        }
+    }
+}
+
+
+void PlayerAsio::Output_Float32_Int24(int channelNumber, float * inputBuffer, float * outputBuffer, bool swap)
+{
+    for(int i=0; i < bufferLength; i++)
+    {
+        m_pTempBuffer[i] = inputBuffer[i*sizeof(float) + channelNumber];
+    }
+
+    float32toInt24inPlace(m_pTempBuffer,bufferLength,swap);
+    memcpy(outputBuffer,m_pTempBuffer,bufferLength*3);
+
+}
+
+void PlayerAsio::Output_Float32_Int32(int channelNumber, float * inputBuffer, float * outputBuffer, bool swap)
+{
+    for(int i=0; i < bufferLength; i++)
+    {
+        m_pTempBuffer[i] = inputBuffer[i*sizeof(float) + channelNumber];
+    }
+
+    float32toInt32inPlace(static_cast<float *>(m_pTempBuffer),bufferLength,swap);
+    memcpy(outputBuffer,m_pTempBuffer,bufferLength*4);
+
+}
+
+const double fScaler32 = (double)0xffff;
+
+void PlayerAsio::float32toInt32inPlace(float * buffer, long frames,bool swap)
+{
+    double sc = fScaler32 + .49999;
+    long a;
+    char * b = (char *)(buffer);
+    char * aa = (char *)(&a);
+
+    while(--frames >= 0)
+    {
+        a = static_cast<long>(static_cast<double>(*buffer++) * sc);
+
+
+        if(swap){
+            *b++ = aa[3];
+            *b++ = aa[2];
+            *b++ = aa[1];
+            *b++ = aa[0];
+        }
+        else
+        {
+            *b++ = aa[0];
+            *b++ = aa[1];
+            *b++ = aa[2];
+            *b++ = aa[3];
+        }
+    }
+}
+
+void PlayerAsio::Output_Float32_Int16(int channelNumber, float * inputBuffer, short * outputBuffer, bool swap)
+{
+    short temp;
+    for (int i=0; i < bufferLength; i++)
+    {
+        temp = static_cast<short>(inputBuffer[i*sizeof(float) + channelNumber]);
+        outputBuffer[i] = static_cast<short>(temp);
+        if (swap) SwapBuffer(&outputBuffer[i]);
+    }
+}
+
+void PlayerAsio::SwapBuffer(short * buffer)
+{
+    short a;
+
+    char * b = (char *)(buffer);
+    char * aa = (char *)(&a);
+
+    a = *buffer;
+
+    *b++ = aa[1];
+    *b++ = aa[0];
+
+}
+/*******
+ *
+ * Init
+ *
+ ********/
+
+bool PlayerAsio::initialize() {
+    return true;
+}
+
+bool PlayerAsio::initDriver()
+{
+
+
+    if (ASIOInit (&asioDriverInfo.driverInfo) == ASE_OK ){
+
+
+        return true;
+    }
+    return false;
+
+
+}
+
+
+int PlayerAsio::getBufferSize(void)
+{
+    return asioDriverInfo.preferredSize;
+}
+
+
+int PlayerAsio::getChannelCount(void)
+{
+    return asioDriverInfo.outputChannels;
+}
+
+
+int PlayerAsio::getSampleRate(void)
+{
+    return asioDriverInfo.sampleRate;
+}
+
+
+QStringList PlayerAsio::getSampleRates() {
+    QStringList result;
+    result.append("44100");
+    return result;
+}
+
+
+QStringList PlayerAsio::getInterfaces() {
+    QStringList result;
+    LPASIODRVSTRUCT lpdrv = asioDrivers->lpdrvlist;
+    while(lpdrv) {
+        result.append(lpdrv->drvname);
+        lpdrv = lpdrv->next;
+    }
+    return result;
+}
+
+long PlayerAsio::init_asio_static_data (DriverInfo * asioDriverInfo)
+{
+
+    if(ASIOGetChannels(&asioDriverInfo->inputChannels, &asioDriverInfo->outputChannels) == ASE_OK)
+    {
+
+
+        // get the usable buffer sizes
+        if(ASIOGetBufferSize(&asioDriverInfo->minSize, &asioDriverInfo->maxSize, &asioDriverInfo->preferredSize, &asioDriverInfo->granularity) == ASE_OK)
+        {
+
+            // get the currently selected sample rate
+            if(ASIOGetSampleRate(&asioDriverInfo->sampleRate) == ASE_OK)
+            {
+                //Get SampleRate From Preferences
+                int sampleRate = m_pConfig->getValueString(ConfigKey("[Soundcard]", "Samplerate")).toInt();
+                //Calculate Latency
+                int iLatencyMSec = (1000*asioDriverInfo->preferredSize*4)/(sampleRate);
+
+                if (asioDriverInfo->sampleRate <= 0.0 || asioDriverInfo->sampleRate > 96000.0)
+                {
+
+                    if(ASIOSetSampleRate(sampleRate) == ASE_OK)
+                    {
+                        if(ASIOGetSampleRate(&asioDriverInfo->sampleRate) == ASE_OK)
+                            qDebug() << "Asio State : SAMPLERATE CHANGED";
+                        else
+                            return -6;
+                    }
+                    else
+                        return -5;
+                } else if(asioDriverInfo->sampleRate != m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt())
+                {
+
+                    if(ASIOSetSampleRate(sampleRate) == ASE_OK)
+                    {
+                        if(ASIOGetSampleRate(&asioDriverInfo->sampleRate) == ASE_OK)
+                            qDebug() << "Asio State : SAMPLERATE CHANGED";
+                        else
+                            return -6;
+                    }
+                    else
+                        return -5;
+                }
+
+
+                // Driver does not store it's internal sample rate, so set it to a know one.
+                // Usually you should check beforehand, that the selected sample rate is valid
+                // with ASIOCanSampleRate().
+                // Update SRATE and Latency ControlObjects
+                m_pControlObjectSampleRate->queueFromThread(static_cast<double>(sampleRate));
+                m_pControlObjectLatency->queueFromThread(static_cast<double>(iLatencyMSec));
+                m_pConfig->set(ConfigKey("[Soundcard]","Latency"), iLatencyMSec);
+                qDebug() << "Asio State: SAMPLERATE SET";
+
+                // check wether the driver requires the ASIOOutputReady() optimization
+                // (can be used by the driver to reduce output latency by one block)
+                if(ASIOOutputReady() == ASE_OK)
+                    asioDriverInfo->postOutput = true;
+                else
+                    asioDriverInfo->postOutput = false;
+
+
+                return 0;
+            }
+            return -3;
+        }
+        return -2;
+    }
+    return -1;
+}
+
+ASIOError PlayerAsio::create_asio_buffers (DriverInfo * asioDriverInfo)
+{
+
+    long i=0;
+    ASIOError result;
+
+    bufferLength = asioDriverInfo->preferredSize;
+    m_pTempBuffer = static_cast<float *>(malloc(bufferLength*4));
+
+    // fill the bufferInfos from the start without a gap
+    ASIOBufferInfo * info = asioDriverInfo->bufferInfos;
+
+    asioDriverInfo->inputBuffers = 0;
+
+    // prepare inputs (Though this is not necessaily required, no opened inputs will work, too
+    if (asioDriverInfo->inputChannels > kMaxInputChannels)
+        asioDriverInfo->inputBuffers = kMaxInputChannels;
+    else
+        asioDriverInfo->inputBuffers = asioDriverInfo->inputChannels;
+
+    for(i = 0; i < asioDriverInfo->inputBuffers; i++, info++)
+    {
+        info->isInput = ASIOTrue;
+        info->channelNum = i;
+        info->buffers[0] = info->buffers[1] = 0;
+    }
+    // prepare outputs
+    if (asioDriverInfo->outputChannels > kMaxOutputChannels)
+        asioDriverInfo->outputBuffers = kMaxOutputChannels;
+    else
+        asioDriverInfo->outputBuffers = asioDriverInfo->outputChannels;
+
+    qDebug(QString::number(asioDriverInfo->outputChannels));
+
+    // prepare outputs
+    if (asioDriverInfo->outputChannels > kMaxOutputChannels)
+        asioDriverInfo->outputBuffers = kMaxOutputChannels;
+    else
+        asioDriverInfo->outputBuffers = asioDriverInfo->outputChannels;
+    for(i = 0; i < asioDriverInfo->outputBuffers; i++, info++)
+    {
+        info->isInput = ASIOFalse;
+        info->channelNum = i;
+        info->buffers[0] = info->buffers[1] = 0;
+    }
+
+    // create and activate buffers
+    result = ASIOCreateBuffers(asioDriverInfo->bufferInfos,
+                               asioDriverInfo->outputBuffers+asioDriverInfo->inputBuffers,
+                               asioDriverInfo->preferredSize, &asioCallbacks);
+
+    if (result == ASE_OK)
+    {
+        // now get all the buffer details, sample word length, name, word clock group and activation
+        for (i = 0; i < asioDriverInfo->inputBuffers + asioDriverInfo->outputBuffers; i++)
+        {
+            asioDriverInfo->channelInfos[i].channel = asioDriverInfo->bufferInfos[i].channelNum;
+            asioDriverInfo->channelInfos[i].isInput = asioDriverInfo->bufferInfos[i].isInput;
+
+            result = ASIOGetChannelInfo(&asioDriverInfo->channelInfos[i]);
+
+            if (result != ASE_OK){
+                qDebug() << "ERROR: Buffer init failed !";
+                break;
+            }
+        }
+
+        if (result == ASE_OK)
+        {
+            // get the input and output latencies
+            // Latencies often are only valid after ASIOCreateBuffers()
+            // (input latency is the age of the first sample in the currently returned audio block)
+            // (output latency is the time the first sample in the currently returned audio block requires to get to the output)
+            result = ASIOGetLatencies(&asioDriverInfo->inputLatency, &asioDriverInfo->outputLatency);
+            //if (result == ASE_OK)
+            //emit(initBuffers(asioDriverInfo->preferredSize));
+        }
+    }
+    return result;
+}
+
+/*******
+ *
+ * Open / Close Calls
+ *
+ ********/
+bool PlayerAsio::open()
+{
+    if (driverIsLoaded)
+        close();
+
+    QString name;
+
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
+
+    char driverName[MAX_DRIVER_NAME_LENGTH];
+    strcpy(driverName, name.latin1());
+
+    if (loadAsioDriver(driverName))
+    {
+        qDebug() << "Asio State: DRIVER LOADED";
+        if(initDriver()){
+            qDebug() << "Asio State: DRIVER INITIALIZED";
+            init_asio_static_data (&asioDriverInfo);
+            asioCallbacks.bufferSwitch = bufferSwitch;
+            asioCallbacks.sampleRateDidChange = sampleRateChanged;
+            asioCallbacks.asioMessage = asioMessages;
+            asioCallbacks.bufferSwitchTimeInfo = bufferSwitchTimeInfo;
+            create_asio_buffers(&asioDriverInfo);
+            startAsio();
+            return true;
+        }
+    }
+    return false;
+}
+
+void PlayerAsio::startAsio()
+{
+    if (ASIOStart() == ASE_OK)
+    {
+        driverIsLoaded = true;
+        qDebug() << "Asio streaming started";
+    }
+}
+
+void PlayerAsio::close()
+{
+    if(ASIOStop() == ASE_OK)
+    {
+        driverIsLoaded = false;
+        qDebug() << "Asio streaming stopped";
+    }
+
+    if(ASIOExit() == ASE_OK)
+    {
+        qDebug() << "Asio exited";
+    }
+}
+
+void PlayerAsio::setDefaults()
+{
+    // Get list of interfaces
+    QStringList interfaces = getInterfaces();
+
+    // Set first interfaces to master left
+    QStringList::iterator it = interfaces.begin();
+    if (it!=interfaces.end())
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((*it)));
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((*it)));
+    }
+    else
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+    }
+
+    // Set head left and right to none
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+
+}
+
+/*******
+ *
+ * Asio and mixxx Callbacks
+ *
+ ********/
+
+//----------------------------------------------------------------------------------
+void bufferSwitch(long index, ASIOBool processNow)
+{   // the actual processing callback.
+
+    ASIOTime timeInfo;
+    memset (&timeInfo, 0, sizeof (timeInfo));
+
+
+    bufferSwitchTimeInfo (&timeInfo, index, processNow);
+
+}
+
+
+//----------------------------------------------------------------------------------
+void sampleRateChanged(ASIOSampleRate sRate)
+{
+    // do whatever you need to do if the sample rate changed
+    // usually this only happens during external sync.
+    // Audio processing is not stopped by the driver, actual sample rate
+    // might not have even changed, maybe only the sample rate status of an
+    // AES/EBU or S/PDIF digital input at the audio device.
+    // You might have to update time/sample related conversion routines, etc.
+}
+
+//----------------------------------------------------------------------------------
+long asioMessages(long selector, long value, void * message, double * opt)
+{
+    // currently the parameters "value", "message" and "opt" are not used.
+    long ret = 0;
+    switch(selector)
+    {
+    case kAsioSelectorSupported:
+        if(value == kAsioResetRequest
+           || value == kAsioEngineVersion
+           || value == kAsioResyncRequest
+           || value == kAsioLatenciesChanged
+           // the following three were added for ASIO 2.0, you don't necessarily have to support them
+           || value == kAsioSupportsTimeInfo
+           || value == kAsioSupportsTimeCode
+           || value == kAsioSupportsInputMonitor)
+            ret = 1L;
+        break;
+    case kAsioResetRequest:
+        // defer the task and perform the reset of the driver during the next "safe" situation
+        // You cannot reset the driver right now, as this code is called from the driver.
+        // Reset the driver is done by completely destruct is. I.e. ASIOStop(), ASIODisposeBuffers(), Destruction
+        // Afterwards you initialize the driver again.
+        //asioDriverInfo.stopped;  // In this sample the processing will just stop
+        ret = 1L;
+        break;
+    case kAsioResyncRequest:
+        // This informs the application, that the driver encountered some non fatal data loss.
+        // It is used for synchronization purposes of different media.
+        // Added mainly to work around the Win16Mutex problems in Windows 95/98 with the
+        // Windows Multimedia system, which could loose data because the Mutex was hold too long
+        // by another thread.
+        // However a driver can issue it in other situations, too.
+        ret = 1L;
+        break;
+    case kAsioLatenciesChanged:
+        // This will inform the host application that the drivers were latencies changed.
+        // Beware, it this does not mean that the buffer sizes have changed!
+        // You might need to update internal delay data.
+        ret = 1L;
+        break;
+    case kAsioEngineVersion:
+        // return the supported ASIO version of the host application
+        // If a host applications does not implement this selector, ASIO 1.0 is assumed
+        // by the driver
+        ret = 2L;
+        break;
+    case kAsioSupportsTimeInfo:
+        // informs the driver wether the asioCallbacks.bufferSwitchTimeInfo() callback
+        // is supported.
+        // For compatibility with ASIO 1.0 drivers the host application should always support
+        // the "old" bufferSwitch method, too.
+        ret = 1;
+        break;
+    case kAsioSupportsTimeCode:
+        // informs the driver wether application is interested in time code info.
+        // If an application does not need to know about time code, the driver has less work
+        // to do.
+        ret = 0;
+        break;
+    }
+    return ret;
+}
+
+void PlayerAsio::processCallback(long bufferIndex) {
+
+    float * inputBuffer = prepareBuffer(bufferLength);
+    // perform the processing
+    for (int i = 0; i < asioDriverInfo.inputBuffers + asioDriverInfo.outputBuffers; i++)
+    {
+        if (asioDriverInfo.bufferInfos[i].isInput == false)
+        {
+            ASIOBufferInfo * bufferInfo = &asioDriverInfo.bufferInfos[i];
+
+            m_reenter++;
+            if (m_reenter > 1) {
+                qDebug() << "ASIO : reentered callback ???!!!";
+                return;
+            }
+
+            if (!driverIsLoaded) {
+                qDebug() << "ASIO : callback and not started ???!!!";
+            }
+
+
+            bool swap = true;
+
+
+            void * outputBuffer = bufferInfo->buffers[bufferIndex];
+            ASIOSampleType bufferType = asioDriverInfo.channelInfos[i].type;
+            switch (bufferType) {
+            case ASIOSTInt16LSB:
+                Output_Float32_Int16(i, inputBuffer, static_cast<short *>(outputBuffer), !swap);
+                break;
+            case ASIOSTInt16MSB:
+                Output_Float32_Int16(i, inputBuffer, static_cast<short *>(outputBuffer), swap);
+                break;
+            case ASIOSTInt32LSB:
+                Output_Float32_Int32(i, inputBuffer, static_cast<float *>(outputBuffer), !swap);
+                break;
+            case ASIOSTInt32MSB:
+                Output_Float32_Int32(i, inputBuffer, static_cast<float *>(outputBuffer), swap);
+                break;
+            case ASIOSTFloat32LSB:
+                //Output_Float32_Float32(i, inputBuffer, static_cast<float*>(outputBuffer), swap);
+                break;
+            case ASIOSTFloat32MSB:
+                //Output_Float32_Float32(i, inputBuffer, static_cast<float*>(outputBuffer), !swap);
+                memset(static_cast<float *>(outputBuffer),0,bufferLength*4);
+                break;
+            case ASIOSTInt24LSB:        // used for 20 bits as well
+                Output_Float32_Int24(i, inputBuffer,static_cast<float *>(outputBuffer), !swap);
+                //memset(static_cast<float*>(outputBuffer),0,bufferLength*4);
+                break;
+            case ASIOSTInt24MSB:        // used for 20 bits as well
+            case ASIOSTFloat64LSB:      // IEEE 754 64 bit double float, as found on Intel x86 architecture
+            case ASIOSTFloat64MSB:      // IEEE 754 64 bit double float, as found on Intel x86 architecture
+
+                // these are used for 32 bit data buffer, with different alignment of the data inside
+                // 32 bit PCI bus systems can more easily used with these
+
+            case ASIOSTInt32LSB16:      // 32 bit data with 16 bit alignment
+            case ASIOSTInt32LSB18:      // 32 bit data with 18 bit alignment
+            case ASIOSTInt32LSB20:      // 32 bit data with 20 bit alignment
+            case ASIOSTInt32LSB24:      // 32 bit data with 24 bit alignment
+
+
+            case ASIOSTInt32MSB16:      // 32 bit data with 16 bit alignment
+            case ASIOSTInt32MSB18:      // 32 bit data with 18 bit alignment
+            case ASIOSTInt32MSB20:      // 32 bit data with 20 bit alignment
+            case ASIOSTInt32MSB24:      // 32 bit data with 24 bit alignment
+                // not implemented !!
+
+                break;
+
+
+            }
+
+            m_reenter--;
+        }
+    }
+    if (asioDriverInfo.postOutput)
+        ASIOOutputReady();
+}
+
+ASIOTime * bufferSwitchTimeInfo(ASIOTime * timeInfo, long index, ASIOBool processNow)
+{
+    ((PlayerAsio *)(PlayerProxy::getPlayer()))->processCallback(index);
+
+    return 0L;
+}
diff -Nru mixxx-1.6.0/src/old/playerasio.h mixxx-1.6.1/src/old/playerasio.h
--- mixxx-1.6.0/src/old/playerasio.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerasio.h	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,133 @@
+// PlayerAsio.h: interface for the PlayerAsio class.
+//
+//////////////////////////////////////////////////////////////////////
+
+#include <qobject.h>
+#include <qmutex.h>
+#include <qsignal.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include "player.h"
+#include "asiosys.h"
+#include "asio.h"
+#include "asiodrivers.h"
+
+
+#if !defined(PlayerAsio_H)
+#define PlayerAsio_H
+
+void __cdecl bufferSwitch(long index, ASIOBool processNow);
+long __cdecl asioMessages(long selector, long value, void* message, double* opt);
+void __cdecl sampleRateChanged(ASIOSampleRate sRate);
+ASIOTime* __cdecl bufferSwitchTimeInfo(ASIOTime *timeInfo, long index, ASIOBool processNow);
+
+enum {
+    // number of input and outputs supported by the host application
+    // you can change these to higher or lower values
+    kMaxInputChannels = 0,
+    kMaxOutputChannels = 4
+};
+
+
+// internal data storage
+typedef struct DriverInfo
+{
+    
+    ASIODriverInfo driverInfo;
+    
+    
+    long           inputChannels;
+    long           outputChannels;
+    
+    
+    long           minSize;
+    long           maxSize;
+    long           preferredSize;
+    long           granularity;
+    
+    
+    ASIOSampleRate sampleRate;
+    
+    
+    bool           postOutput;
+    
+    
+    long           inputLatency;
+    long           outputLatency;
+    
+   
+    long inputBuffers;  // becomes number of actual created input buffers
+    long outputBuffers; // becomes number of actual created output buffers
+    ASIOBufferInfo bufferInfos[kMaxInputChannels + kMaxOutputChannels]; // buffer info's
+    
+    // ASIOGetChannelInfo()
+    ASIOChannelInfo channelInfos[kMaxInputChannels + kMaxOutputChannels]; // channel info's
+    
+    
+    double         nanoSeconds;
+    double         samples;
+    double         tcSamples;   // time code samples
+    
+    
+    ASIOTime       tInfo;           // time info state
+    unsigned long  sysRefTime;      // system reference time, when bufferSwitch() was called
+    
+  
+} DriverInfo;
+
+
+class PlayerAsio : public Player 
+{
+    
+public:
+    
+    
+    PlayerAsio(ConfigObject<ConfigValue> *config);
+    virtual ~PlayerAsio();
+    
+    bool initialize();
+    bool open();
+    void close();
+    void startAsio();
+    
+    int getBufferSize(void);
+    int getSampleRate(void);
+    int getChannelCount(void);
+    
+    QStringList getInterfaces();
+    QStringList getSampleRates();
+    
+    void setDefaults();
+    
+    static QString getSoundApi() { return "ASIO"; }
+    QString getSoundApiName() { return getSoundApi(); };
+    
+    void process(const CSAMPLE *, const CSAMPLE *, const int) { };
+    void processCallback(long bufferIndex);
+    
+    
+private:
+    bool initDriver(void);
+
+    void Output_Float32_Int16(int channelNumber, float* inputBuffer, short* outputBuffer, bool swap); 
+    void Output_Float32_Int32(int channelNumber, float* inputBuffer, float* outputBuffer, bool swap); 
+    void Output_Float32_Int24(int channelNumber, float* inputBuffer, float* outputBuffer, bool swap); 
+    void float32toInt24inPlace(float* buffer, long frames, bool swap);
+    void float32toInt32inPlace(float* buffer, long frames, bool swap);
+    void SwapBuffer(short* buffer);
+    
+    ASIOError create_asio_buffers (DriverInfo *asioDriverInfo);
+    long init_asio_static_data (DriverInfo *asioDriverInfo);
+    
+    AsioDrivers* driver;
+    int m_reenter;
+    bool driverIsLoaded;
+    int drvNum;
+    int bufferLength;
+    char* driverList[32];
+    /** Used in sample convertion */
+    float *m_pTempBuffer;
+     
+};
+
+#endif 
diff -Nru mixxx-1.6.0/src/old/player.cpp mixxx-1.6.1/src/old/player.cpp
--- mixxx-1.6.0/src/old/player.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/player.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,151 @@
+/***************************************************************************
+                          player.cpp  -  description
+                             -------------------
+    begin                : Wed Feb 20 2002
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+    email                :
+***************************************************************************/
+
+/***************************************************************************
+*                                                                         *
+*   This program is free software; you can redistribute it and/or modify  *
+*   it under the terms of the GNU General Public License as published by  *
+*   the Free Software Foundation; either version 2 of the License, or     *
+*   (at your option) any later version.                                   *
+*                                                                         *
+***************************************************************************/
+
+#include "player.h"
+#include "enginemaster.h"
+#include "controlobject.h"
+
+#ifdef RECORD_OUTPUT
+// HACK TO RECORD OUTPUT. WRITTEN TO FILE AT PROGRAM EXIT
+  #include <audiofile.h>
+  #include <q3ptrlist.h>
+typedef struct {
+    short int * pBuffer;
+    int size;
+} recordObject;
+Q3PtrList<recordObject> m_qRecordList;
+#endif
+
+//Static variable memory allocation
+short int Player::m_iBufferSize = 0;
+short int Player::m_iChannels[MAX_AUDIODEVICES];
+EngineMaster * Player::m_pMaster = 0;
+
+
+
+/* -------- ------------------------------------------------------
+   Purpose: Initializes the audio hardware.
+   Input:   Size of the output buffer in samples
+   Output:  Pointer to internal synthesis data structure.
+   -------- ------------------------------------------------------ */
+Player::Player(ConfigObject<ConfigValue> * pConfig)
+{
+    m_pConfig = pConfig;
+    m_pBuffer = new CSAMPLE[MAX_BUFFER_LEN];
+    m_pControlObjectSampleRate = ControlObject::getControl(ConfigKey("[Master]","samplerate"));
+    m_pControlObjectLatency = new ControlObject(ConfigKey("[Master]","latency"));;
+
+    for (int i = 0; i < MAX_AUDIODEVICES; i++)
+    {
+        m_iChannels[i] = 0;
+    }
+}
+
+/* -------- ------------------------------------------------------
+   Purpose: Terminate and deallocate the synthesis system
+   Input:   -
+   Output:  -
+   -------- ------------------------------------------------------ */
+Player::~Player()
+{
+    delete [] m_pBuffer;
+
+#ifdef RECORD_OUTPUT
+    //
+    // HACK: Write recorded sound
+    //
+
+    qDebug() << "Writing output to file... please wait!";
+
+    // Setup file format
+    AFfilesetup outputSetup = afNewFileSetup();
+    afInitFileFormat(outputSetup, AF_FILE_WAVE);
+    afInitRate(outputSetup, AF_DEFAULT_TRACK, m_pControlObjectSampleRate->get());
+    afInitChannels(outputSetup, AF_DEFAULT_TRACK, 2);
+    afInitSampleFormat (outputSetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+    //afInitByteOrder (outputSetup, AF_DEFAULT_TRACK, AF_BYTEORDER_BIGENDIAN);
+
+    // Open file handle
+    AFfilehandle fh = afOpenFile("output.wav","w",outputSetup);
+    if (!fh)
+        qCritical("Could not write wave output file");
+    recordObject * r;
+    for (r = m_qRecordList.first(); r; r=m_qRecordList.next())
+    {
+        afWriteFrames(fh,AF_DEFAULT_TRACK,r->pBuffer,r->size/2);
+    }
+    afCloseFile(fh);
+#endif
+}
+
+/*
+   short int Player::getBufferSize()
+   {
+   //     if (m_iChannels>0)
+   //         return m_iBufferSize/m_iChannels;
+   //     else
+        return m_iBufferSize;
+   }
+ */
+
+void Player::setMaster(EngineMaster * pMaster)
+{
+    m_pMaster = pMaster;
+}
+
+bool Player::open()
+{
+    if (m_pConfig->getValueString(ConfigKey("[Soundcard]","PitchIndpTimeStretch")).toInt())
+        qDebug() << "pitch true";
+
+    // Set sound scale method
+    if (m_pMaster)
+        m_pMaster->setPitchIndpTimeStretch(m_pConfig->getValueString(ConfigKey("[Soundcard]","PitchIndpTimeStretch")).toInt());
+
+    return true;
+}
+
+/* -------- ------------------------------------------------------
+   Purpose: Internal callback function used for preparing samples
+            for playback. This is where the synthesis is done.
+   Input:   Number of samples for each channel (4 channels in all)
+   Output:  pointer to resulting buffer of samples
+   -------- ------------------------------------------------------ */
+CSAMPLE * Player::prepareBuffer(int iBufferSize)
+{
+    // First, sync control parameters with changes from GUI thread
+    ControlObject::sync();
+
+    // Process a block of samples for output. iBufferSize is the
+    // number of samples for one channel, but the EngineObject
+    // architecture expects number of samples for two channels
+    // as input so...
+    m_pMaster->process(0, m_pBuffer, iBufferSize*2);
+
+#ifdef RECORD_OUTPUT
+    // HACK: RECORD SOUND
+    recordObject * r = new recordObject;
+    m_qRecordList.append(r);
+    r->size = iBufferSize*2;
+    r->pBuffer = new short int[r->size];
+    for (int j=0; j<r->size; ++j)
+        r->pBuffer[j] = (short int)(m_pBuffer[j]);
+#endif
+
+    return m_pBuffer;
+}
+
diff -Nru mixxx-1.6.0/src/old/player.h mixxx-1.6.1/src/old/player.h
--- mixxx-1.6.0/src/old/player.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/player.h	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,79 @@
+/***************************************************************************
+                          player.h  -  description
+                             -------------------
+    begin                : Wed Feb 20 2002
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+    email                : 
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#ifndef PLAYER_H
+#define PLAYER_H
+
+#include "configobject.h"
+#include <qstring.h>
+#include "engineobject.h"
+#include <qstringlist.h>
+
+#define MAX_AUDIODEVICES 2
+
+class EngineMaster;
+class ControlObject;
+
+class Player : public EngineObject
+{
+public:
+    Player(ConfigObject<ConfigValue> *pConfig);
+    virtual ~Player();
+    /** Initialize the API. Returns true on success. No other methods in a Player
+      * class can be called before this method returns true. Called by the proxy
+      * class. */
+    virtual bool initialize() = 0;
+    /** Returns the current buffer size (total latency) used by the playback system */
+    //static short int getBufferSize();
+    /** Set EngineMaster object */
+    static void setMaster(EngineMaster *pMaster);
+    /** Open devices according to config database, and start audio stream.
+      * Returns true on success. */
+    virtual bool open();
+    /** Close devices, and stop audio stream */
+    virtual void close() = 0;
+    /** Store default configuration in config database */
+    virtual void setDefaults() = 0;
+    /** Return list of interfaces */
+    virtual QStringList getInterfaces() = 0;
+    /** Return list of input interfaces */
+    virtual QStringList getInputInterfaces() = 0;    
+    /** Return list of sample rates */
+    virtual QStringList getSampleRates() = 0;
+    /** Return name of sound api */
+    virtual QString getSoundApiName() = 0;
+
+protected:
+    /** Prepares one buffer of sound by calling the engine */
+    CSAMPLE *prepareBuffer(int iBufferSize);
+
+    /** Pointer to config database */
+    ConfigObject<ConfigValue> *m_pConfig;
+    /** Pointer to EngineMaster object */
+    static EngineMaster *m_pMaster;
+    CSAMPLE *m_pBuffer;
+    /** Current buffer size in use. */
+    static short int m_iBufferSize;
+    /** Current number of channels in use */
+    static short int m_iChannels[MAX_AUDIODEVICES];
+    /** Pointer to object holding sample rate */
+    ControlObject *m_pControlObjectSampleRate;
+    /** Pointer to object holding latency */
+    ControlObject *m_pControlObjectLatency;
+};
+
+#endif
diff -Nru mixxx-1.6.0/src/old/playerjack.cpp mixxx-1.6.1/src/old/playerjack.cpp
--- mixxx-1.6.0/src/old/playerjack.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerjack.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,370 @@
+/***************************************************************************
+                          playerjack.cpp  -  description
+                             -------------------
+    begin                : Sat Nov 15 2003
+    copyright            : (C) 2003 by Tue Haste Andersen
+    email                :
+***************************************************************************/
+
+/***************************************************************************
+*                                                                         *
+*   This program is free software; you can redistribute it and/or modify  *
+*   it under the terms of the GNU General Public License as published by  *
+*   the Free Software Foundation; either version 2 of the License, or     *
+*   (at your option) any later version.                                   *
+*                                                                         *
+***************************************************************************/
+
+#include "playerjack.h"
+#include "controlobject.h"
+#include <qmessagebox.h>
+#include <QtDebug>
+#include <qapplication.h>
+#include "controlobject.h"
+
+PlayerJack::PlayerJack(ConfigObject<ConfigValue> * config) : Player(config) //, mLibJack("libjack.so")
+{
+    ports = 0;
+    m_bOpen = false;
+
+/*
+    //mLibJack = new QLibrary("jack");
+    if (!mLibJack.load())
+        qDebug() << "JACK lib path" << mLibJack.library();
+
+    //if (mLibJack.isLoaded())
+    {
+        jack_set_error_function = (jack_set_error_function_t) mLibJack.resolve("jack_set_error_function");
+        jack_port_unregister = (jack_port_unregister_t) mLibJack.resolve("jack_port_unregister");
+        jack_client_close = (jack_client_close_t) mLibJack.resolve("jack_client_close");
+        jack_client_new = (jack_client_new_t) mLibJack.resolve("jack_client_new");
+        jack_set_process_callback = (jack_set_process_callback_t) mLibJack.resolve("jack_set_process_callback");
+        jack_set_sample_rate_callback = (jack_set_sample_rate_callback_t) mLibJack.resolve("jack_set_sample_rate_callback");
+        jack_on_shutdown =(jack_on_shutdown_t) mLibJack.resolve("jack_on_shutdown");
+        jack_port_register =(jack_port_register_t) mLibJack.resolve("jack_port_register");
+        jack_activate =(jack_activate_t) mLibJack.resolve("jack_activate");
+        jack_connect =(jack_connect_t) mLibJack.resolve("jack_connect");
+        jack_port_name =(jack_port_name_t) mLibJack.resolve("jack_port_name");
+        jack_deactivate =(jack_deactivate_t) mLibJack.resolve("jack_deactivate");
+        jack_get_ports =(jack_get_ports_t) mLibJack.resolve("jack_get_ports");
+        jack_get_sample_rate =(jack_get_sample_rate_t) mLibJack.resolve("jack_get_sample_rate");
+        jack_port_get_buffer =(jack_port_get_buffer_t) mLibJack.resolve("jack_port_get_buffer");
+    }
+ */
+}
+
+PlayerJack::~PlayerJack()
+{
+//    if (!mLibJack.isLoaded())
+//        return;
+
+    close();
+
+    if (client)
+    {
+        jack_port_unregister(client, output_master_left);
+        jack_port_unregister(client, output_master_right);
+        jack_port_unregister(client, output_head_left);
+        jack_port_unregister(client, output_head_right);
+
+        jack_client_close(client);
+    }
+
+    if (ports)
+        free(ports);
+}
+
+bool PlayerJack::initialize()
+{
+    // Verify that library is loaded, and all function pointers has been retreived
+    /*
+       if (!mLibJack.isLoaded())
+       {
+        qDebug() << "lib not loaded";
+        return false;
+       }
+       else
+       if (!jack_set_error_function |
+            !jack_port_unregister |
+            !jack_client_close |
+            !jack_client_new |
+            !jack_set_process_callback |
+            !jack_set_sample_rate_callback |
+            !jack_on_shutdown |
+            !jack_port_register |
+            !jack_activate |
+            !jack_connect |
+            !jack_port_name |
+            !jack_deactivate |
+            !jack_get_ports |
+            !jack_get_sample_rate |
+            !jack_port_get_buffer)
+       {
+        qDebug() << "API function pointer error";
+        return false;
+       }
+     */
+
+    jack_set_error_function(jackError);
+
+    if ((client = jack_client_new("Mixxx")) == 0)
+        return false;
+
+    jack_set_process_callback(client, jackProcess, this);
+    jack_set_sample_rate_callback(client, jackSrate, this);
+    jack_on_shutdown(client, jackShutdown, this);
+
+    output_master_left  = jack_port_register(client, "Master left", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    output_master_right = jack_port_register(client, "Master right", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    output_head_left    = jack_port_register(client, "Head left", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    output_head_right   = jack_port_register(client, "Head right", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+
+    return true;
+}
+
+bool PlayerJack::open()
+{
+    Player::open();
+
+    /* tell the JACK server that we are ready to roll */
+    if (jack_activate(client))
+    {
+        qDebug() << "Jack: Cannot activate client";
+        return false;
+    }
+
+    // Connect to the ports
+    QString name;
+    m_iChannels = 0;
+
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
+    if (name != "None")
+    {
+        qDebug() << "connecting" << name << "to master left";
+        if (jack_connect(client, jack_port_name(output_master_left), name.latin1()))
+            m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+        else
+        {
+            m_iBufferSize = jack_port_get_total_latency(client, output_master_left);
+            m_iChannels++;
+        }
+    }
+
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterRight"));
+    if (name != "None")
+    {
+        qDebug() << "connecting" << name << "to master right";
+        if (jack_connect(client, jack_port_name(output_master_right), name.latin1()))
+            m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+        else
+        {
+            m_iBufferSize = jack_port_get_total_latency(client, output_master_right);
+            m_iChannels++;
+        }
+    }
+
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadLeft"));
+    if (name != "None")
+    {
+        qDebug() << "connecting" << name << "to head left";
+        if (jack_connect(client, jack_port_name(output_head_left), name.latin1()))
+            m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+        else
+        {
+            m_iBufferSize = jack_port_get_total_latency(client, output_head_left);
+            m_iChannels++;
+        }
+    }
+
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadRight"));
+    if (name != "None")
+    {
+        qDebug() << "connecting" << name << "to head right";
+        if (jack_connect(client, jack_port_name(output_head_right), name.latin1()))
+            m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+        else
+        {
+            m_iBufferSize = jack_port_get_total_latency(client, output_head_right);
+            m_iChannels++;
+        }
+    }
+
+    qDebug() << "channels " << m_iChannels << ", latency " << m_iBufferSize;
+
+
+    // Update the config database with the used sample rate
+    QStringList srates = getSampleRates();
+    QStringList::iterator it = srates.begin();
+    if (*it)
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue((*it)));
+
+        // Set currently used latency in config database
+        int msec = (int)(1000.*(float)m_iBufferSize/((float)(*it).toInt()));
+        m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
+
+        // Update SRATE and Latency ControlObjects
+        m_pControlObjectSampleRate->queueFromThread((double)((*it).toInt()));
+        m_pControlObjectLatency->queueFromThread((double)msec);
+
+    }
+
+    m_bOpen = true;
+    return true;
+}
+
+void PlayerJack::close()
+{
+    // Deactivate jack and disconnect all ports
+    if (m_bOpen && client)
+        jack_deactivate(client);
+    m_bOpen = false;
+}
+
+void PlayerJack::setDefaults()
+{
+    // Get list of interfaces
+    QStringList interfaces = getInterfaces();
+
+    // Set first interfaces to master left
+    QStringList::iterator it = interfaces.begin();
+    if (*it)
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((*it)));
+    }
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+
+    // Set second interface to master right
+    if (*it) ++it;
+    if (*it)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((*it)));
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+
+    // Set head left and right to none
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+
+    // Set default sample rate
+    QStringList srates = getSampleRates();
+    it = srates.begin();
+    if (*it)
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue((*it)));
+
+        // Set currently used latency in config database
+        int msec = 1000*m_iBufferSize/(4*(*it).toInt());
+        m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
+    }
+}
+
+QStringList PlayerJack::getInterfaces()
+{
+    QStringList result;
+
+    if (!client && !initialize())
+    {
+        qDebug() << "Jack: Failed to reinitialize the connection to Jack";
+        return result;
+    }
+
+    if ((ports = jack_get_ports (client, 0, 0, JackPortIsPhysical|JackPortIsInput)) == 0)
+        qDebug() << "Jack: Cannot find any physical playback ports";
+    else
+    {
+        int i=0;
+        while (ports[i]!=0)
+        {
+            result.append(ports[i]);
+            ++i;
+        }
+    }
+
+    return result;
+}
+
+QStringList PlayerJack::getSampleRates()
+{
+    QStringList result;
+
+    if (client)
+        result.append(QString("%1").arg((int)jack_get_sample_rate(client)));
+
+    return result;
+}
+
+QString PlayerJack::getSoundApi()
+{
+    return QString("Jack");
+}
+
+int PlayerJack::callbackProcess(int iBufferSize)
+{
+//     qDebug() << "buffer size " << iBufferSize;
+    jack_default_audio_sample_t * out_ml = (jack_default_audio_sample_t *)jack_port_get_buffer(output_master_left, iBufferSize);
+    jack_default_audio_sample_t * out_mr = (jack_default_audio_sample_t *)jack_port_get_buffer(output_master_right, iBufferSize);
+    jack_default_audio_sample_t * out_hl = (jack_default_audio_sample_t *)jack_port_get_buffer(output_head_left, iBufferSize);
+    jack_default_audio_sample_t * out_hr = (jack_default_audio_sample_t *)jack_port_get_buffer(output_head_right, iBufferSize);
+
+    if (m_bOpen)
+    {
+        CSAMPLE * buffer = prepareBuffer(iBufferSize);
+
+        for (int i=0; i<iBufferSize; ++i)
+        {
+            *out_ml++=buffer[(i*4)  ]/32768.;
+            *out_mr++=buffer[(i*4)+1]/32768.;
+            *out_hl++=buffer[(i*4)+2]/32768.;
+            *out_hr++=buffer[(i*4)+3]/32768.;
+        }
+    }
+
+    return 0;
+}
+
+void PlayerJack::callbackSetSrate(int srate)
+{
+//    setPlaySrate(srate);
+    m_pControlObjectSampleRate->queueFromThread((double)srate);
+}
+
+void PlayerJack::callbackSetBufferSize(int iBufferSize)
+{
+    m_iBufferSize = iBufferSize;
+}
+
+void PlayerJack::callbackShutdown()
+{
+    m_pConfig->set(ConfigKey("[Soundcard]","SoundApi"), ConfigValue("None"));
+
+    qWarning("Jack connection was killed.\n\nThis could be due to a high CPU load. Try increasing the latency\nwhen starting the Jack sound server.");
+}
+
+void jackError(const char * desc)
+{
+    qDebug() << "Jack experienced an error:" << desc;
+}
+
+int jackProcess(jack_nframes_t nframes, void * arg)
+{
+    ((PlayerJack *)arg)->callbackProcess(nframes);
+    return 0;
+}
+
+int jackSrate(jack_nframes_t nframes, void * arg)
+{
+    // Update SRATE in EngineObject
+    ((PlayerJack *)arg)->callbackSetSrate(nframes);
+    return 0;
+}
+
+void jackShutdown(void * arg)
+{
+    ((PlayerJack *)arg)->callbackShutdown();
+}
+
+void jackBufferSize(jack_nframes_t nframes, void * arg)
+{
+    ((PlayerJack *)arg)->callbackSetBufferSize(nframes);
+}
diff -Nru mixxx-1.6.0/src/old/playerjack.h mixxx-1.6.1/src/old/playerjack.h
--- mixxx-1.6.0/src/old/playerjack.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerjack.h	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,120 @@
+/***************************************************************************
+                          playerjack.h  -  description
+                             -------------------
+    begin                : Sat Nov 15 2003
+    copyright            : (C) 2003 by Tue Haste Andersen
+    email                : 
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#ifndef PLAYERJACK_H
+#define PLAYERJACK_H
+
+#include "player.h"
+#include <jack/jack.h>
+#include <qlibrary.h>
+
+/**
+  *@author Tue and Ken Haste Andersen
+  */
+
+class PlayerJack : public Player  
+{
+public:
+    PlayerJack(ConfigObject<ConfigValue> *config);
+    ~PlayerJack();
+    bool initialize();
+    bool open();
+    void close();
+    void setDefaults();
+    QStringList getInterfaces();
+    QStringList getSampleRates();
+    static QString getSoundApi();
+    QString getSoundApiName() { return getSoundApi(); };
+    /** Satisfy virtual declaration in EngineObject */
+    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
+    /** Process samples. Called from jack callback */
+    int callbackProcess(int iBufferSize);
+    /** Used to set sample rate from Jack callback */
+    void callbackSetSrate(int iSrate);
+    /** Used to set current buffer size */
+    void callbackSetBufferSize(int iBufferSize);
+    /** Used to reinitialize when shut down by Jack server */
+    void callbackShutdown();
+
+protected:
+    /** Pointer to QLibrary object */
+    //QLibrary mLibJack;
+    /** Pointer to client info */
+    jack_client_t *client;
+    /** Null terminated array of port names */
+    const char **ports;
+    /** Playback ports */
+    jack_port_t *output_master_left, *output_master_right, *output_head_left, *output_head_right;
+    /** True if devices are open */
+    bool m_bOpen;
+
+/*
+    typedef void (*jack_set_error_function_t)(void (*func)(const char *));
+    jack_set_error_function_t jack_set_error_function;
+
+    typedef int (*jack_port_unregister_t)(jack_client_t *, jack_port_t *);
+    jack_port_unregister_t jack_port_unregister;
+
+    typedef int (*jack_client_close_t)(jack_client_t *client);
+    jack_client_close_t jack_client_close;
+
+    typedef jack_client_t* (*jack_client_new_t)(const char *client_name);
+    jack_client_new_t jack_client_new;
+
+    typedef int (*jack_set_process_callback_t)(jack_client_t *client,JackProcessCallback process_callback,void *arg);
+    jack_set_process_callback_t jack_set_process_callback;
+
+    typedef int (*jack_set_sample_rate_callback_t)(jack_client_t *client,JackSampleRateCallback srate_callback,void *arg);
+    jack_set_sample_rate_callback_t jack_set_sample_rate_callback;
+
+    typedef void (*jack_on_shutdown_t)(jack_client_t *client, void (*function)(void *arg), void *arg);
+    jack_on_shutdown_t jack_on_shutdown;
+
+    typedef jack_port_t* (*jack_port_register_t)(jack_client_t *,const char *port_name,const char *port_type,unsigned long flags,unsigned long buffer_size);
+    jack_port_register_t jack_port_register;
+
+    typedef int (*jack_activate_t)(jack_client_t *client);
+    jack_activate_t jack_activate;
+
+    typedef int (*jack_connect_t)(jack_client_t *,const char *source_port,const char *destination_port);
+    jack_connect_t jack_connect;
+
+    typedef const char* (*jack_port_name_t)(const jack_port_t *port);
+    jack_port_name_t jack_port_name;
+
+    typedef int (*jack_deactivate_t)(jack_client_t *client);
+    jack_deactivate_t jack_deactivate;
+
+    typedef const char** (*jack_get_ports_t)(jack_client_t *,const char *port_name_pattern,const char *type_name_pattern,unsigned long flags);
+    jack_get_ports_t jack_get_ports;
+
+    typedef jack_nframes_t (*jack_get_sample_rate_t)(jack_client_t *);
+    jack_get_sample_rate_t jack_get_sample_rate;
+
+    typedef void* (*jack_port_get_buffer_t)(jack_port_t *, jack_nframes_t);
+    jack_port_get_buffer_t jack_port_get_buffer;
+*/
+};
+
+// Jack callbacks:
+void jackError(const char *desc);
+int jackProcess(jack_nframes_t nframes, void *arg);
+int jackSrate(jack_nframes_t nframes, void *arg);
+void jackShutdown(void *arg);
+void jackBufferSize(jack_nframes_t nframes, void *arg);
+
+#endif
diff -Nru mixxx-1.6.0/src/old/playerportaudio.cpp mixxx-1.6.1/src/old/playerportaudio.cpp
--- mixxx-1.6.0/src/old/playerportaudio.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerportaudio.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,938 @@
+/***************************************************************************
+                          playerportaudiov19.cpp  -  description
+                             -------------------
+    begin                : Wed Feb 20 2002
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+                                           (C) 2006/2007 by Albert Santoni
+                                           (C) 2006 by Adam Davison
+    email                :
+***************************************************************************/
+
+/***************************************************************************
+*                                                                         *
+*   This program is free software; you can redistribute it and/or modify  *
+*   it under the terms of the GNU General Public License as published by  *
+*   the Free Software Foundation; either version 2 of the License, or     *
+*   (at your option) any later version.                                   *
+*                                                                         *
+***************************************************************************/
+
+#include "playerportaudio.h"
+#include "controlobject.h"
+
+// Mathstuff required for Win32
+#include "mathstuff.h"
+//Added by qt3to4:
+#include <Q3ValueList>
+#include <QtDebug>
+#include <QWaitCondition>
+
+bool PlayerPortAudio::m_painited = false;
+
+PlayerPortAudio::PlayerPortAudio(ConfigObject<ConfigValue> * config, QString api_name) : Player(config)
+{
+    for (int i = 0; i < MAX_AUDIODEVICES; i++)
+    {
+        m_devId[i]      = -1;
+        m_inputDevId[i] = -1;
+        m_pStream[i]    =  0;
+        m_iChannels[i]  = -1;
+    }
+
+    m_iNumberOfBuffers = 2;
+    m_iMasterLeftCh = -1;
+    m_iMasterRigthCh = -1;
+    m_iHeadLeftCh = -1;
+    m_iHeadRightCh = -1;
+    m_iNumActiveDevices = 0;
+    m_iPreviousDevIndex = -1;
+#ifdef __VINYLCONTROL__
+    m_VinylControl[0] = NULL;
+    m_VinylControl[1] = NULL;
+#endif
+
+    m_HostAPI = api_name;
+    m_bInit = false;
+}
+
+PlayerPortAudio::~PlayerPortAudio()
+{
+    //Close all the audio devices.
+    close();
+
+    if (m_bInit) {
+        Pa_Terminate();
+        m_painited = false;
+    }
+}
+
+bool PlayerPortAudio::initialize()
+{
+    PaError err = Pa_Initialize();
+    if (err!=paNoError)
+    {
+        qDebug() << "PortAudio error: " << Pa_GetErrorText(err);
+        m_bInit = false;
+    }
+    else
+        m_bInit = true;
+
+    return m_bInit;
+}
+
+bool PlayerPortAudio::open()
+{
+    Player::open();
+
+#ifdef __VINYLCONTROL__
+    //Create a new VinylControl object so that it updates it's settings.
+    //TODO: FIX THIS SAMPLERATE HACKAGE
+    int iTempSrate = m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt();
+    if (iTempSrate <= 0)
+        iTempSrate = 44100;
+    m_VinylControl[0] = new VinylControlProxy(m_pConfig, "[Channel1]", iTempSrate);
+    m_VinylControl[1] = new VinylControlProxy(m_pConfig, "[Channel2]", 44100);
+#endif
+
+    // Find out which device to open. Select the first one listed as either Master Left,
+    // Master Right, Head Left, Head Right. If other devices are requested for opening
+    // than the one selected here, set them to "None" in the config database
+
+    // For the record, with MAX_AUDIODEVICES currently set to 2, we're referring to
+    // Master Left/Right as device 1 and Headphones as device 2 (unless Master and
+    // Headphones are on the same device, then we only use device 1.)
+    PaDeviceIndex id[MAX_AUDIODEVICES];
+    PaDeviceIndex id_input[MAX_AUDIODEVICES];
+    int iChannelMax[MAX_AUDIODEVICES]; /** Maximum number of channels needed */
+    PaDeviceIndex temp_id = -1;
+    QString name;
+
+    for (int i = 0; i < MAX_AUDIODEVICES; i++)
+    {
+        iChannelMax[i] = -1;
+        id[i] = -1;
+        id_input[i] = -1;
+    }
+
+    m_iMasterLeftCh = -1;
+    m_iMasterRigthCh = -1;
+    m_iHeadLeftCh = -1;
+    m_iHeadRightCh = -1;
+
+    waitForNextOutput.wakeAll();
+
+    // Master left
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
+    temp_id = getDeviceID(name);
+    if (temp_id >= 0)
+    {
+        if (getChannelNo(name)>=0)
+        {
+            id[0] = temp_id;
+            iChannelMax[0] = getChannelNo(name);
+            m_iMasterLeftCh = getChannelNo(name)-1;
+        }
+    }
+
+    // Master right
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterRight"));
+    temp_id = getDeviceID(name);
+    if (getChannelNo(name)>=0                      //Make sure we got a valid number of channels
+        && ((id[0]==-1 && temp_id>=0)              //Make sure the left channel and this (right) channel IDs are valid
+            || (temp_id!=-1 && id[0]==temp_id))) //No idea...
+    {
+        id[0] = math_max(temp_id, id[0]);
+        iChannelMax[0] = math_max(iChannelMax[0], getChannelNo(name));
+        m_iMasterRigthCh = getChannelNo(name)-1;
+    }
+
+    // Head left
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadLeft"));
+    temp_id = getDeviceID(name);
+    if (getChannelNo(name)>=0 && ((id[1]==-1 && temp_id>=0) || (temp_id!=-1 && id[1]==temp_id)))
+    {
+        id[1] = temp_id;
+        iChannelMax[1] = math_max(iChannelMax[1], getChannelNo(name));
+        m_iHeadLeftCh = getChannelNo(name)-1;
+    }
+
+    // Head right
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadRight"));
+    temp_id = getDeviceID(name);
+    if (getChannelNo(name)>=0 && ((id[1]==-1 && temp_id>=0) || (temp_id!=-1 && id[1]==temp_id)))
+    {
+        id[1] = temp_id;
+        iChannelMax[1] = math_max(iChannelMax[1], getChannelNo(name));
+        m_iHeadRightCh = getChannelNo(name)-1;
+    }
+
+    // Inputs
+    name = m_pConfig->getValueString(ConfigKey("[VinylControl]","DeviceInputDeck1"));
+    temp_id = getInputDeviceID(name);
+    if (temp_id != -1)
+        id_input[0] = temp_id;
+
+    name = m_pConfig->getValueString(ConfigKey("[VinylControl]","DeviceInputDeck2"));
+    temp_id = getInputDeviceID(name);
+    if (temp_id != -1)
+        id_input[1] = temp_id;
+
+    // Check if any of the devices in the config database needs to be set to "None"
+    if (m_iMasterLeftCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+    if (m_iMasterRigthCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+    if (m_iHeadLeftCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+    if (m_iHeadRightCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+
+    // Number of channels to open
+    int iChannels[MAX_AUDIODEVICES];
+    if (id[0] == id[1]) //Check if we're supposed to use the same soundcard for both Master and Headphone output
+    {
+        iChannels[0] = math_max(iChannelMax[0], iChannelMax[1]);
+        iChannels[1] = -1;
+    }
+    else
+    {
+        iChannels[0] = iChannelMax[0];
+        iChannels[1] = iChannelMax[1];
+    }
+
+    // Sample rate
+    int iSrate = m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt();
+    if (iSrate <= 0)
+        iSrate = 44100;
+
+    // Get latency in msec
+    int iLatencyMSec = m_pConfig->getValueString(ConfigKey("[Soundcard]","Latency")).toInt();
+
+    if (iLatencyMSec <= 0)     //Make sure we don't get a crazy latency value.
+        iLatencyMSec = 100;
+
+    // Latency in samples
+    //int iLatencySamples = (int)((float)(iSrate*iChannels)/1000.f*(float)iLatencyMSec); //Pre-multisoundcard output
+    int iLatencySamples = (int)((float)(iSrate*math_max(iChannels[0], iChannels[1]))/1000.f*(float)iLatencyMSec);
+
+    // Round to the nearest multiple of 4.
+    iLatencySamples -= (iLatencySamples % 4);
+    iLatencySamples += 4;
+
+    qDebug() << "iLatencySamples: " << iLatencySamples;
+
+    // Apply simple rule to determine number of buffers
+    if (iLatencySamples/kiMaxFrameSize<2)
+        m_iNumberOfBuffers = 2;
+    else
+        m_iNumberOfBuffers = iLatencySamples/kiMaxFrameSize;
+
+    // Frame size...
+    int iFramesPerBuffer = iLatencySamples/m_iNumberOfBuffers;
+
+    // Ensure the chosen configuration is valid
+    //TODO: Fix this in our PortAudio-v19 implementation. PA19 ditches the Pa_GetMinNumBuffers() function, as it's
+    //		not needed anymore.
+    /*
+       if (m_iNumberOfBuffers<Pa_GetMinNumBuffers(iFramesPerBuffer,iSrate))
+       {
+        m_iNumberOfBuffers = Pa_GetMinNumBuffers(iFramesPerBuffer,iSrate);
+
+        iLatencyMSec = (1000*iFramesPerBuffer*m_iNumberOfBuffers)/(iSrate*iChannels);
+        m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(iLatencyMSec));
+       }
+     */
+
+    // Callback function to use
+    PaStreamCallback * callback = paV19Callback;
+    qDebug() << "PortAudio: kiMaxFrameSize: " << kiMaxFrameSize << ", iLatencyMSec: " << iLatencyMSec;
+    qDebug() << "PortAudio: id[0] " << id[0] << ", sr " << iSrate << ", ch[0] " << iChannels[0] << ", bufsize " << iFramesPerBuffer << ", bufno " << m_iNumberOfBuffers << ", req. latency " << iLatencyMSec << " msec";
+    qDebug() << "PortAudio: id[1] " << id[1] << ", sr " << iSrate << ", ch[1] " << iChannels[1] << ", bufsize " << iFramesPerBuffer << ", bufno " << m_iNumberOfBuffers << ", req. latency " << iLatencyMSec << " msec";
+
+    qDebug() << "Device 1 indices (id[0]: " << id[0] << ", id_input[0]: " << id_input[0] << ")";
+    qDebug() << "Device 2 indices (id[1]: " << id[1] << ", id_input[1]: " << id_input[1] << ")";
+
+
+    //If all devices are set to "None", then just return.
+    if ((id[0] < 0) && (id[1] < 0))
+        return false;
+
+    PaStreamParameters outputParams[MAX_AUDIODEVICES];
+    PaStreamParameters inputParams[MAX_AUDIODEVICES];
+    bool bDeviceAlreadyOpened = false;
+
+    //Set up and open each soundcard that we need!
+    for (int i = 0; i < MAX_AUDIODEVICES; i++)
+    {
+        bDeviceAlreadyOpened = false;
+
+        for (int j = 0; j < i; j++)         //Check to see if we've already opened this device (eg. Maybe we're already using it for Master output)
+        {
+            if (j != i)
+            {
+                if (id[j] == id[i] && id[j] != -1) {
+                    bDeviceAlreadyOpened = true;
+                }
+            }
+        }
+        if (((id[i] != -1) && !bDeviceAlreadyOpened) || (id_input[i] != -1))         //Make sure we're supposed to open this device...
+        {
+            qDebug() << "PortAudio: Trying to open device id " << id[i] << ", with channels " << iChannels[i] << " at samplerate " << iSrate;
+            PaStreamParameters * p_outputParams;
+            outputParams[i].device = id[i];
+            outputParams[i].channelCount = iChannels[i];
+            outputParams[i].sampleFormat = paFloat32;
+            outputParams[i].suggestedLatency = ((float)iLatencyMSec) / 1000.0f;             //Latency in seconds.
+            outputParams[i].hostApiSpecificStreamInfo = NULL;
+
+            PaStreamParameters * p_inputParams;            //TODO: Does this need to be made an array like inputParams?
+            inputParams[i].device = id_input[i];             //TODO: Isn't the "input" device always the same as the output device? (if there is an output device?)
+            //	     Maybe that should be restricted in the preferences GUI...
+            // NO, it's not always the same as the output device! Consider using 4 outputs on your
+            // master soundcard, and a different soundcard for input. The master's id is the same as the first
+            // for the second stream, but the input id for the second stream is something different.
+            inputParams[i].channelCount = 2;
+            inputParams[i].sampleFormat = paInt16;             //As needed by scratchlib
+            inputParams[i].suggestedLatency = ((float)iLatencyMSec) / 1000.0f;             //Latency in seconds.
+            inputParams[i].hostApiSpecificStreamInfo = NULL;
+
+            if (id[i] < 0 || (i > 0 && (id[1] == id[0])))             //TODO: comment this the same as the below input one.... kinda
+                p_outputParams = NULL;
+            else
+                p_outputParams = &(outputParams[i]);
+
+            if (id_input[i] < 0)             //If we're not supposed to open an input device, set the pointer to be null.
+                p_inputParams = NULL;                 //NULL input params in Pa_OpenStream means don't open the device for input.
+            else
+                p_inputParams = &(inputParams[i]);
+
+            PaError err = paNoError;
+
+            // Set up a struct containing all the data we want to pass back to the callback
+            callbackStuff[i].player = this;
+            callbackStuff[i].devIndex = i;         //The audio device's index (NOT THE PORTAUDIO ID!!!)
+
+            // Try open device using iChannelMax
+            err = Pa_OpenStream(&m_pStream[i],
+                                p_inputParams,                                  // Input parameters
+                                p_outputParams,                                 // Output parameters
+                                iSrate,                                                 // Sample rate
+                                iFramesPerBuffer,                               // Frames per buffer
+                                paClipOff,                                              // Stream flags
+                                callback,                                               // Stream callback
+                                &(callbackStuff[i]));                                      // Pointer passed to the callback function
+
+            if (err == paNoError)
+                m_iChannels[i] = iChannels[i];
+            else
+            {
+                // Try open device using maximum supported channels by soundcard
+                iChannels[i] = Pa_GetDeviceInfo(id[i])->maxOutputChannels;
+                outputParams[i].channelCount = iChannels[i];
+                err = Pa_OpenStream(&m_pStream[i],
+                                    p_inputParams,                              // Input parameters
+                                    p_outputParams,                             // Output parameters
+                                    iSrate,                                                     // Sample rate
+                                    iFramesPerBuffer,                                   // Frames per buffer
+                                    paClipOff,                                                  // Stream flags
+                                    callback,                                                   // Stream callback
+                                    &(callbackStuff[i]));                                          // Pointer passed to the callback function
+                if (err==paNoError)
+                    m_iChannels[i] = iChannels[i];
+            }
+
+            if( err != paNoError )         //*** TODO/WARNING: I don't think this block makes any sense whatsoever... - Albert April 21/07
+            {
+                // Try open device using only two channels
+                outputParams[i].channelCount = 2;
+                err = Pa_OpenStream(&m_pStream[i],
+                                    p_inputParams,                              // Input parameters
+                                    p_outputParams,                             // Output parameters
+                                    iSrate,                                                     // Sample rate
+                                    iFramesPerBuffer,                                   // Frames per buffer
+                                    paClipOff,                                                  // Stream flags
+                                    callback,                                                   // Stream callback
+                                    &(callbackStuff[i]));                                          // Pointer passed to the callback function
+                if (err==paNoError)
+                {
+                    m_iChannels[i] = 2;
+
+                    // Update channel variables and config database
+                    if (m_iMasterLeftCh>1)
+                    {
+                        m_iMasterLeftCh = -1;
+                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+                    }
+                    if (m_iMasterRigthCh>1)
+                    {
+                        m_iMasterRigthCh = -1;
+                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+                    }
+                    if (m_iHeadLeftCh>1)
+                    {
+                        m_iHeadLeftCh = -1;
+                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+                    }
+                    if (m_iHeadRightCh>1)
+                    {
+                        m_iHeadRightCh = -1;
+                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+                    }
+                }
+            }
+
+            if( err != paNoError )         //Make sure we opened the soundcard successfully.
+            {
+                qDebug() << "PortAudio: Open stream error: " << Pa_GetErrorText(err);
+                qDebug() << "PortAudio: More error info: " << Pa_GetLastHostErrorInfo()->errorText;
+
+                m_devId[i] = -1;
+                m_inputDevId[i] = -1;
+                m_iChannels[i] = -1;
+
+                return false;
+            }
+
+            m_devId[i] = id[i];
+            m_inputDevId[i] = id_input[i];
+
+            // Update SRATE and Latency ControlObjects
+            m_pControlObjectSampleRate->queueFromThread((double)iSrate);
+            m_pControlObjectLatency->queueFromThread((double)iLatencyMSec);
+
+            // Start stream
+            err = Pa_StartStream(m_pStream[i]);
+            if (err != paNoError)
+            {
+                qDebug() << "PortAudio: Start stream " << i << " error: " << Pa_GetErrorText(err);
+                m_pStream[i] = 0;
+            }
+            else
+                qDebug() << "PortAudio: Started stream " << i << " successfully";
+
+            //If the Master and Headphone devices are the same, or the Headphone device is set to "None",
+            //then we can just break out of this loop:
+            /*if ((id[0] == id[1]) || (id[1] == -1))
+               {
+                    calculateNumActiveDevices();
+                    return true;
+               }*/
+        }
+    }
+
+    /*
+       //Print out all the information about the soundcards and their channels.
+       qDebug() << "PortAudio: ==Soundcard Summary==";
+       qDebug() << "Device 1 index: " << m_devId[0];
+       qDebug() << "Device 2 index: " << m_devId[1];
+       qDebug() << "Channels for device 1: " << m_iChannels[0];
+       qDebug() << "Channels for device 2: " << m_iChannels[1];
+       qDebug() << "m_iMasterLeftCh: " << m_iMasterLeftCh;
+       qDebug() << "m_iMasterRightCh: " << m_iMasterRigthCh;
+       qDebug() << "m_iHeadLeftCh: " << m_iHeadLeftCh;
+       qDebug() << "m_iHeadRightCh: " << m_iHeadRightCh;
+     */
+
+    calculateNumActiveDevices();
+    return true;
+}
+
+//Find out how many active/open soundcards there are.
+void PlayerPortAudio::calculateNumActiveDevices()
+{
+    int count = 0;
+
+    for (int i = 0; i < MAX_AUDIODEVICES; i++)
+    {
+        if (m_devId[i] != -1)
+            count++;
+    }
+    m_iNumActiveDevices = count;
+}
+
+void PlayerPortAudio::close()
+{
+    //waitForNextOutput.wakeAll();
+
+    for (int i = 0; i < MAX_AUDIODEVICES; i++)
+    {
+        m_devId[i] = -1;
+        m_iChannels[i] = 0;
+
+        // Stop streams
+        if (m_pStream[i])
+        {
+            PaError err = Pa_StopStream(m_pStream[i]);
+            if( err != paNoError )
+                qDebug() << "PortAudio: Stop stream " << i << " error: " << Pa_GetErrorText(err) << ",";
+
+            // Close streams
+            err = Pa_CloseStream(m_pStream[i]);
+            if( err != paNoError )
+                qDebug() << "PortAudio: Close stream " << i << " error: " << Pa_GetErrorText(err);
+        }
+
+        m_pStream[i] = 0;
+        waitForNextOutput.wakeAll();
+        calculateNumActiveDevices();
+    }
+
+    //waitForNextOutput.wakeAll();
+
+    m_iMasterLeftCh = -1;
+    m_iMasterRigthCh = -1;
+    m_iHeadLeftCh = -1;
+    m_iHeadRightCh = -1;
+}
+
+void PlayerPortAudio::setDefaults()
+{
+    // Get list of interfaces
+    QStringList interfaces = getInterfaces();
+
+    // Set first interfaces to master left
+    QStringListIterator it(interfaces);
+
+    if (it.hasNext())
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((it.next())));
+    }
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+
+    // Set second interface to master right
+    if (it.hasNext())
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((it.next())));
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+
+    // Set head left and right to none
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+
+    // Set default sample rate
+    QStringList srates = getSampleRates();
+    QStringListIterator itSRates(srates);
+    while (it.hasNext())
+    {
+        QString s;
+        s = itSRates.next();
+        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue(s));
+        if ((s).toInt()>=44100)
+            break;
+    }
+
+    // Set currently used latency in config database
+    //int msec = (int)(1000.*(2.*1024.)/(2.*(float)(*it).toInt()));
+    int msec = 100;     // More conservative defaults
+
+    m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
+}
+
+QStringList PlayerPortAudio::getInterfaces()
+{
+    //qDebug() << "PortAudio: getInterfaces()";
+
+    QStringList result;
+    const PaHostApiInfo * apiInfo = NULL;
+    const PaDeviceInfo * devInfo = NULL;
+    QString api;
+
+    if (m_HostAPI == "None")
+        return result;
+
+    PaDeviceIndex numDevices = Pa_GetDeviceCount();
+
+    for (int i = 0; i < numDevices; i++)
+    {
+        devInfo = Pa_GetDeviceInfo(i);
+
+        // Add the device if it is an output device:
+        //if (devInfo != NULL && devInfo->maxOutputChannels > 0)
+        if (devInfo != NULL)
+        {
+            apiInfo = Pa_GetHostApiInfo(devInfo->hostApi);
+            //api = apiInfo->name;
+            //qDebug() << "Api name: " << apiInfo->name;
+            qDebug() << devInfo->name;
+            //qDebug() << "m_HostAPI: " << m_HostAPI << "devInfo->hostApi: " << new QString(devInfo->hostApi);
+
+            //... and make sure the interface matches the API we've selected.
+            if (m_HostAPI == apiInfo->name)
+            {
+                //qDebug() << "name " << devInfo->name << ", API " << devInfo->hostApi << ", maxOutputChannels: " << devInfo->maxOutputChannels;
+
+                for (int j=1; j <= devInfo->maxOutputChannels; ++j)
+                    result.append(QString("%1 (channel %2)").arg(devInfo->name).arg(j));
+            }
+        }
+    }
+    //qDebug() << "PortAudio: getInterfaces() end";
+    return result;
+}
+
+/**
+ * Get the audio input/capture interfaces from PortAudio
+ */
+QStringList PlayerPortAudio::getInputInterfaces()
+{
+    qDebug() << "PortAudio: getInputInterfaces()";
+
+    QStringList result;
+    const PaHostApiInfo * apiInfo = NULL;
+    const PaDeviceInfo * devInfo = NULL;
+    QString api;
+
+    if (m_HostAPI == "None")
+        return result;
+
+    PaDeviceIndex numDevices = Pa_GetDeviceCount();
+
+    for (int i = 0; i < numDevices; i++)
+    {
+        devInfo = Pa_GetDeviceInfo(i);
+
+        // Add the device if it is an input device:
+        //if (devInfo != NULL && devInfo->maxOutputChannels > 0)
+        if (devInfo != NULL && (devInfo->maxInputChannels > 0))
+        {
+            apiInfo = Pa_GetHostApiInfo(devInfo->hostApi);
+            //api = apiInfo->name;
+            qDebug() << "Api name: " << apiInfo->name;
+            qDebug(devInfo->name);
+            //qDebug() << "m_HostAPI: " << m_HostAPI << "devInfo->hostApi: " << new QString(devInfo->hostApi);
+
+            //... and make sure the interface matches the API we've selected.
+            if (m_HostAPI == apiInfo->name )
+            {
+                qDebug() << "name " << devInfo->name << ", API " << devInfo->hostApi << ", maxInputChannels: " << devInfo->maxInputChannels;
+
+                result.append(QString("%1").arg(devInfo->name));
+
+                //for (int j=1; j <= devInfo->maxInputChannels; ++j)
+                // 	result.append(QString("%1 (channel %2)").arg(devInfo->name).arg(j));
+            }
+        }
+    }
+    qDebug() << "PortAudio: getInputInterfaces() end";
+    return result;
+}
+
+
+QStringList PlayerPortAudio::getSampleRates()
+{
+    // Returns a sorted list of supported sample rates of the currently opened device.
+    // If no device is open, return the list of sample rates supported by the
+    // default device
+    qDebug() << "PortAudio: getSampleRates()";
+
+    PaError err;
+    PaDeviceIndex id = m_devId[0];              //TODO: This is a hack to pick the samplerates from the first soundcard....
+    //Figure out something smarter...
+    //Something smarter = poll the samplerates from both soundcards, and
+    //only display the samlerates that are supported by both. - Albert April 30, 2007
+    //The problem with doing this right now is that Pa_IsFormatSupported when using
+    //ALSA takes ages for some reason, and polling both soundcards here would
+    //double this 5 second freeze that Mixxx experiences because of Pa_IsFormatSupported
+    //(buggily) blocking.
+    //qDebug() << "m_devId[0]: " << m_devId[0];
+    if (id<0)
+        id = Pa_GetDefaultOutputDevice();
+
+    //const PaDeviceInfo *devInfo = Pa_GetDeviceInfo(id);
+
+    PaStreamParameters outputParams;
+    outputParams.device = id;
+    outputParams.channelCount = 2;
+    outputParams.sampleFormat = paFloat32;
+    outputParams.suggestedLatency = .150;     //Latency in seconds.
+    outputParams.hostApiSpecificStreamInfo = NULL;
+
+    Q3ValueList<double> desiredSampleRates; //A list of all the sample rates we're going to suggest.
+    Q3ValueList<double> validSampleRates; //A list containing all the supported sample rates.
+
+    //Here's all the sample rates we're going to suggest to PortAudio. We check which ones
+    //are actually supported below.
+    desiredSampleRates.append(11025.0);
+    desiredSampleRates.append(22050.0);
+    desiredSampleRates.append(44100.0);
+    desiredSampleRates.append(48000.0);
+    desiredSampleRates.append(96000.0);
+
+    // Sample rates
+    //if (devInfo)
+    {
+        for (unsigned int j=0; j < desiredSampleRates.count(); j++)
+        {
+            //Check if each sample rate is supported, if so, add them to the list of supported sample rates.
+            qDebug() << "PortAudio: checking if sample rate is supported...";
+            err = Pa_IsFormatSupported(NULL, &outputParams, desiredSampleRates[j]);
+            if (err == paFormatIsSupported)     //The format IS supported.
+            {
+                validSampleRates.append(desiredSampleRates[j]);
+                qDebug() << "Supported...";
+            }
+            else
+            {
+                qDebug() << "PortAudio error: " << Pa_GetErrorText(err) << ", id was: " << id;
+            }
+        }
+    }
+
+    //If for some reason our enumeration of the sample rates failed, throw
+    //in some default values. (PortAudio might be being sketchy...)
+    if (validSampleRates.count() == 0)
+    {
+        validSampleRates.append(44100.0);
+        validSampleRates.append(48000.0);
+        validSampleRates.append(96000.0);
+    }
+
+    // Sort list
+#ifndef QT3_SUPPORT
+    qSort(validSampleRates);
+#endif
+
+    // Convert srlist to stringlist
+    QStringList result;
+    for (unsigned int i = 0; i < validSampleRates.count(); ++i)
+        result.append(QString("%1").arg((*validSampleRates.at(i))));
+
+    qDebug() << "PortAudio: getSampleRates() end";
+
+    return result;
+}
+
+QStringList PlayerPortAudio::getSoundApiList()
+{
+    //Note: Need to put the active API at the top of the list.
+    QStringList apiList;
+    const PaHostApiInfo * apiInfo = NULL;
+
+    //We need to initialize PortAudio before we find out what APIs are present.
+    //Even if this gets called while PortAudio is already initialized, the docs
+    //say this is OK (I think...).
+
+    PaError err = paNoError;
+
+    // So this little hackfest saves buggy drivers from being really buggy - AD
+    if (!m_painited) {
+        err = Pa_Initialize();
+        m_painited = true;
+    }
+
+    if (err == paNoError)
+    {
+        for (int i = 0; i < Pa_GetHostApiCount(); i++)
+        {
+            apiInfo = Pa_GetHostApiInfo(i);
+            //qDebug() << "Api name: " << apiInfo->name;
+            apiList.append(apiInfo->name);
+        }
+//		Pa_Terminate();
+    }
+    else
+        qDebug() << "PortAudio error: " << Pa_GetErrorText(err);
+
+    return apiList;
+
+    /*
+     #ifdef __LINUX__
+       return QStringList("OSS (PA)");
+     #endif
+     #ifdef __MACX__
+       return QStringList("CoreAudio (PA)");
+     #endif
+     #ifdef __WIN__
+       return QStringList("WMME (PA)");
+     #endif
+     */
+
+}
+
+
+PaDeviceIndex PlayerPortAudio::getDeviceID(QString name)
+{
+    //qDebug() << "PortAudio: getDeviceID(" + name + ")";
+    PaDeviceIndex no = Pa_GetDeviceCount();
+    for (int i=0; i<no; i++)
+    {
+        const PaDeviceInfo * devInfo = Pa_GetDeviceInfo(i);
+
+        // Add the device if it is an output device:
+        if (devInfo != 0 && devInfo->maxOutputChannels > 0)
+        {
+            for (int j = 1; j <= devInfo->maxOutputChannels; ++j)
+                if (QString("%1 (channel %2)").arg(devInfo->name).arg(j) == name)
+                {
+                    //qDebug() << "PortAudio: getDeviceID(" + name + "), returning device id " << i;
+                    return i;
+                }
+        }
+    }
+    return -1;
+}
+
+
+PaDeviceIndex PlayerPortAudio::getInputDeviceID(QString name)
+{
+    qDebug() << "PortAudio: getInputDeviceID(" + name + ")";
+    PaDeviceIndex no = Pa_GetDeviceCount();
+    for (int i=0; i<no; i++)
+    {
+        const PaDeviceInfo * devInfo = Pa_GetDeviceInfo(i);
+
+        // Add the device if it is an input device:
+        if (devInfo != 0 && devInfo->maxOutputChannels > 0)
+        {
+            for (int j = 1; j <= devInfo->maxInputChannels; ++j)
+            {
+                //qDebug() << "Comparing: " << QString("%1").arg(devInfo->name);
+                //qDebug() << "and: " << name;
+                if (QString("%1").arg(devInfo->name) == name)
+                {
+                    qDebug() << "PortAudio: getInputDeviceID(" + name + "), returning device id " << i;
+                    return i;
+                }
+            }
+        }
+    }
+    qDebug() << "PortAudio: getInputDeviceID(" + name + "), returning device id -1";
+    return -1;
+}
+
+PaDeviceIndex PlayerPortAudio::getChannelNo(QString name)
+{
+    PaDeviceIndex no = Pa_GetDeviceCount();
+    for (int i=0; i<no; i++)
+    {
+        const PaDeviceInfo * devInfo = Pa_GetDeviceInfo(i);
+
+        // Add the device if it is an output device:
+        if (devInfo!=0 && devInfo->maxOutputChannels > 0)
+        {
+            for (int j=1; j<=devInfo->maxOutputChannels; ++j)
+                if (QString("%1 (channel %2)").arg(devInfo->name).arg(j) == name)
+                    return j;
+        }
+    }
+    return -1;
+}
+
+/** -------- ------------------------------------------------------
+        Purpose: This callback function gets called everytime a soundcard runs out of samples to play.
+                         This is where the soundcard/PortAudio ask Mixxx for more samples, and get them in
+                         an orderly fashion (*thread safety). This is also where the "routing" of audio samples
+                         take place.
+        -------- ------------------------------------------------------
+ */
+int PlayerPortAudio::callbackProcess(unsigned long framesPerBuffer, float * out, short * in, int devIndex)
+{
+    //if (m_iBufferSize==0)
+    //    m_iBufferSize = iBufferSize*m_iNumberOfBuffers;
+
+    static float * tmp; //Leave this as static - it's shared between two threads.
+    float * output = out;
+    int i;
+
+    /*
+       if (!out)
+       {
+        qDebug() << "output is NULL!";
+       prevDevice.lock();
+       m_iPreviousDevIndex = devIndex; //Save this devIndex as the previous one
+        prevDevice.unlock();
+        waitForNextOutput.wakeAll(); //Allow the other thread to give at 'er
+        return 0;
+       }*/
+
+    prevDevice.lock();
+    int iPreviousDevIndex = m_iPreviousDevIndex;
+    prevDevice.unlock();
+
+    if (iPreviousDevIndex == devIndex && m_iNumActiveDevices > 1)
+    {
+        lockSamples.lock();
+        waitForNextOutput.wait(&lockSamples);
+        lockSamples.unlock();
+    }
+
+    //Only fill the buffer with sound data from Mixxx once.
+    lockSamples.lock();
+    if (devIndex == 0)
+    {
+        tmp = prepareBuffer(iBufferSize);
+    }
+    lockSamples.unlock();
+
+    // Reset sample for each open channel
+    for (i=0; i<framesPerBuffer*m_iChannels[devIndex]; i++)
+        output[i] = 0.;
+
+    // Copy to output buffer
+    for (i=0; i<iBufferSize; i++)
+    {
+        if (devIndex == 0) //For the first sound device
+        {
+            if (m_iMasterLeftCh>=0  && m_iChannels[devIndex]>=1) { output[m_iMasterLeftCh]  += tmp[(i*4)  ]/32768.;}
+            if (m_iMasterRigthCh>=0 && m_iChannels[devIndex]>=2) output[m_iMasterRigthCh] += tmp[(i*4)+1]/32768.;
+            if (m_iHeadLeftCh>=0    && m_iChannels[devIndex]>=3) output[m_iHeadLeftCh]    += tmp[(i*4)+2]/32768.;
+            if (m_iHeadRightCh>=0   && m_iChannels[devIndex]>=4) output[m_iHeadRightCh]   += tmp[(i*4)+3]/32768.;
+        }
+        else if (devIndex == 1)         //If there's a second sound device, route the headphones to it.
+        {
+            //if (m_iMasterLeftCh>=0  && m_iChannels[devIndex]>=1) { output[m_iMasterLeftCh]  += tmp[(i*4)  ]/32768.;}
+            //if (m_iMasterRigthCh>=0 && m_iChannels[devIndex]>=2) output[m_iMasterRigthCh] += tmp[(i*4)+1]/32768.;
+            //qDebug() << "m_iHeadLeftCh:" << m_iHeadLeftCh << "devIndex: " << devIndex << "tmp[blah]" << tmp[(i*4)+2]/32768.;
+            if (m_iHeadLeftCh>=0    && m_iChannels[devIndex]>=1) output[m_iHeadLeftCh]    += tmp[(i*4)+2]/32768.;
+            if (m_iHeadRightCh>=0   && m_iChannels[devIndex]>=2) output[m_iHeadRightCh]   += tmp[(i*4)+3]/32768.;
+            //qDebug() << "headphones!";
+        }
+        for (int j=0; j < m_iChannels[devIndex]; ++j)
+            *output++;
+    }
+
+    if (in)
+    {
+#ifdef __VINYLCONTROL__
+        m_VinylControl[devIndex]->AnalyseSamples(in, iBufferSize);
+#endif
+    }
+
+    prevDevice.lock();
+    m_iPreviousDevIndex = devIndex; //Save this devIndex as the previous one
+    prevDevice.unlock();
+    waitForNextOutput.wakeAll();     //Allow the other thread to give at 'er
+
+    return 0;
+}
+
+
+/* -------- ------------------------------------------------------
+   Purpose: Wrapper function to call processing loop function,
+            implemented as a method in a class. Used in PortAudio,
+            which knows nothing about C++.
+   Input:   .
+   Output:  -
+   -------- ------------------------------------------------------ */
+int paV19Callback(const void * inputBuffer, void * outputBuffer,
+                  unsigned long framesPerBuffer,
+                  const PaStreamCallbackTimeInfo * timeInfo,
+                  PaStreamCallbackFlags statusFlags,
+                  void * _callbackStuff)
+{
+    /*
+       //Variables that are used in the human-readable form of function call from hell (below).
+       static PlayerPortAudio* _player;
+       static int devIndex;
+       _player = ((PAPlayerCallbackStuff*)_callbackStuff)->player;
+       devIndex = ((PAPlayerCallbackStuff*)_callbackStuff)->devIndex;
+     */
+
+    //Human-readable form of the function call from hell:
+    //return _player->callbackProcess(framesPerBuffer, (float *)outputBuffer, devIndex);
+
+    //Function call from hell (might provide a little bit of cheapo thread safety to do it this way):
+    return ((PAPlayerCallbackStuff *)_callbackStuff)->player->callbackProcess(framesPerBuffer, (float *)outputBuffer, (short *)inputBuffer, ((PAPlayerCallbackStuff *)_callbackStuff)->devIndex);
+}
+
diff -Nru mixxx-1.6.0/src/old/playerportaudio.h mixxx-1.6.1/src/old/playerportaudio.h
--- mixxx-1.6.0/src/old/playerportaudio.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerportaudio.h	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,109 @@
+/***************************************************************************
+                          playerportaudiov19.h  -  description
+                             -------------------
+    begin                : Wed Feb 20 2002
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+    email                : 
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#ifndef PLAYERPORTAUDIO_H
+#define PLAYERPORTAUDIO_H
+
+#include <q3semaphore.h>
+#include <qthread.h>
+#include <QMutex>
+#include <QWaitCondition>
+#include "player.h"
+#include "portaudio.h"
+#ifdef __VINYLCONTROL__
+#include "vinylcontrolproxy.h"
+#endif
+
+/**
+  *@author Tue and Ken Haste Andersen
+  */
+
+/** Maximum frame size used with PortAudio. Used to determine no of buffers
+  * when setting latency */
+const int kiMaxFrameSize = 1024;
+
+class PlayerPortAudio; //Forward declaration
+
+/** A struct to some stuff we need to pass along to the callback through PortAudio **/
+struct PAPlayerCallbackStuff
+{
+	PlayerPortAudio* player;
+	int devIndex;
+};
+
+class PlayerPortAudio : public Player  {
+public:
+    PlayerPortAudio(ConfigObject<ConfigValue> *config, QString api_name);
+    ~PlayerPortAudio();
+    bool initialize();
+    bool open();
+    void close();
+    void setDefaults();
+    QStringList getInterfaces();
+    QStringList getInputInterfaces();
+    QStringList getSampleRates();
+    static QStringList getSoundApiList();
+    QString getSoundApiName() { return getSoundApiList().front(); };
+    void calculateNumActiveDevices();
+    /** Satisfy virtual declaration in EngineObject */
+    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
+    /** Process samples. Called from PortAudio callback */
+    int callbackProcess(int iBufferSize, float *out, short *in, int devIndex);
+
+    static bool m_painited;
+protected:
+    /** Get id of device with a given name. Returns -1 if device is not found */
+    PaDeviceIndex getDeviceID(QString name);
+    PaDeviceIndex getInputDeviceID(QString name);
+    /** Get channel number of device with a given name. Returns -1 if device is not found */
+    int getChannelNo(QString name);
+
+    /** PortAudio stream */
+    PaStream *m_pStream[MAX_AUDIODEVICES];
+    /** Id of currently open device. -1 if no device is open */
+    PaDeviceIndex m_devId[MAX_AUDIODEVICES];
+    PaDeviceIndex m_inputDevId[MAX_AUDIODEVICES]; //For input devices
+    /** Channels used for each output from Mixxx. Set to -1 when not in use */
+    int m_iMasterLeftCh, m_iMasterRigthCh, m_iHeadLeftCh, m_iHeadRightCh;
+    /** True if PortAudio was sucessfully initialized */
+    bool m_bInit;
+    /** A struct to hold some information/pointers we need to pass to our callback function */
+    PAPlayerCallbackStuff callbackStuff[MAX_AUDIODEVICES];
+    /** Number of buffers */
+    int m_iNumberOfBuffers;
+    /** Number of active/open soundcards **/
+    int m_iNumActiveDevices;
+    /** Name of the current audio API inside PortAudio **/
+    QString m_HostAPI;
+    /** Mutex so that two threads don't try to prepare a new buffer full of samples from Mixxx at the same time */
+    QMutex lockSamples; 
+    /** Wait condition that forces multiple PortAudio callbacks in separate threads to play nicely */
+    QWaitCondition waitForNextOutput;
+    QMutex waitMutex;
+    int m_iPreviousDevIndex;
+    QMutex prevDevice; 
+#ifdef __VINYLCONTROL__
+    VinylControlProxy *m_VinylControl[2];
+#endif
+};
+
+int paV19Callback(const void *inputBuffer, void *outputBuffer,
+                      unsigned long framesPerBuffer,
+                      const PaStreamCallbackTimeInfo* timeInfo,
+                      PaStreamCallbackFlags statusFlags,
+                      void *_player);
+#endif
diff -Nru mixxx-1.6.0/src/old/playerproxy.cpp mixxx-1.6.1/src/old/playerproxy.cpp
--- mixxx-1.6.0/src/old/playerproxy.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerproxy.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,189 @@
+/***************************************************************************
+                          playerproxy.cpp  -  description
+                             -------------------
+    begin                : Fri Nov 21 2003
+    copyright            : (C) 2003 by Tue and Ken Haste Andersen
+    email                :
+***************************************************************************/
+
+/***************************************************************************
+*                                                                         *
+*   This program is free software; you can redistribute it and/or modify  *
+*   it under the terms of the GNU General Public License as published by  *
+*   the Free Software Foundation; either version 2 of the License, or     *
+*   (at your option) any later version.                                   *
+*                                                                         *
+***************************************************************************/
+
+#include "playerproxy.h"
+
+#ifdef __PORTAUDIO__
+    #include "playerportaudio.h"
+#endif
+#ifdef __JACK__
+    #include "playerjack.h"
+#endif
+#ifdef __RTAUDIO__
+    #include "playerrtaudio.h"
+#endif
+#ifdef __ASIO__
+    #include "playerasio.h"
+#endif
+#ifdef __ALSA__
+    #include "playeralsa.h"
+#endif
+
+Player * PlayerProxy::m_pPlayer = 0;
+
+PlayerProxy::PlayerProxy(ConfigObject<ConfigValue> * pConfig) : Player(pConfig)
+{
+    // Set API based on info stored in config database
+    setSoundApi(m_pConfig->getValueString(ConfigKey("[Soundcard]","SoundApi")));
+
+    if (!m_pPlayer)
+    {
+        // No valid API is stored in the database, so select the default
+        QStringList api = getSoundApiList();
+        QStringList::iterator it = api.begin();
+        while (!m_pPlayer && it!=api.end())
+        {
+            if (setSoundApi((*it)))
+            {
+                m_pConfig->set(ConfigKey("[Soundcard]","SoundApi"), ConfigValue((*it)));
+                m_pPlayer->setDefaults();
+            }
+            *it++;
+        }
+    }
+}
+
+PlayerProxy::~PlayerProxy()
+{
+    if (m_pPlayer)
+        delete m_pPlayer;
+}
+
+bool PlayerProxy::open()
+{
+    if (m_pPlayer)
+        return m_pPlayer->open();
+    else
+        return false;
+}
+
+void PlayerProxy::close()
+{
+    if (m_pPlayer)
+        m_pPlayer->close();
+}
+
+void PlayerProxy::setDefaults()
+{
+    if (m_pPlayer)
+        m_pPlayer->setDefaults();
+}
+
+QStringList PlayerProxy::getInterfaces()
+{
+    if (m_pPlayer)
+        return m_pPlayer->getInterfaces();
+    else
+        return QStringList();
+}
+
+QStringList PlayerProxy::getInputInterfaces()
+{
+    if (m_pPlayer)
+        return m_pPlayer->getInputInterfaces();
+    else
+        return QStringList();
+}
+
+QStringList PlayerProxy::getSampleRates()
+{
+    if (m_pPlayer)
+        return m_pPlayer->getSampleRates();
+    else
+        return QStringList();
+}
+
+QString PlayerProxy::getSoundApiName()
+{
+    if (m_pPlayer)
+        return m_pPlayer->getSoundApiName();
+    else
+        return QString();
+}
+
+
+QStringList PlayerProxy::getSoundApiList()
+{
+    QStringList result;
+#ifdef __PORTAUDIO__
+    result += PlayerPortAudio::getSoundApiList();
+#endif
+#ifdef __JACK__
+    result.append(PlayerJack::getSoundApi());
+#endif
+#ifdef __RTAUDIO__
+    result.append(PlayerRtAudio::getSoundApi());
+#endif
+#ifdef __ASIO__
+    result.append(PlayerAsio::getSoundApi());
+#endif
+#ifdef __ALSA__
+    result.append(PlayerALSA::getSoundApi());
+#endif
+
+    return result;
+}
+
+bool PlayerProxy::setSoundApi(QString name)
+{
+    if (m_pPlayer)
+        delete m_pPlayer;
+    m_pPlayer = 0;
+
+#ifdef __PORTAUDIO__
+    if (PlayerPortAudio::getSoundApiList().contains(name))
+        m_pPlayer = new PlayerPortAudio(m_pConfig, name);
+#endif
+
+#ifdef __JACK__
+    if (name == PlayerJack::getSoundApi())
+        m_pPlayer = new PlayerJack(m_pConfig);
+#endif
+
+#ifdef __RTAUDIO__
+    if (name == PlayerRtAudio::getSoundApi())
+        m_pPlayer = new PlayerRtAudio(m_pConfig);
+#endif
+
+#ifdef __ASIO__
+    if (name == PlayerAsio::getSoundApi())
+        m_pPlayer = new PlayerAsio(m_pConfig);
+#endif
+
+#ifdef __ALSA__
+    if (name == PlayerALSA::getSoundApi())
+        m_pPlayer = new PlayerALSA(m_pConfig);
+#endif
+
+    // Try initializing the selected API
+    bool init = false;
+    if (m_pPlayer)
+        init = m_pPlayer->initialize();
+
+    if (!init)
+    {
+        if (m_pPlayer)
+            delete m_pPlayer;
+        m_pPlayer = 0;
+    }
+
+    if (m_pPlayer)
+        return true;
+    else
+        return false;
+}
+
diff -Nru mixxx-1.6.0/src/old/playerproxy.h mixxx-1.6.1/src/old/playerproxy.h
--- mixxx-1.6.0/src/old/playerproxy.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerproxy.h	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,54 @@
+/***************************************************************************
+                          playerproxy.h  -  description
+                             -------------------
+    begin                : Fri Nov 21 2003
+    copyright            : (C) 2003 by Tue Haste Andersen
+    email                : 
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#ifndef PLAYERPROXY_H
+#define PLAYERPROXY_H
+
+#include "player.h"
+
+class PlayerProxy : public Player
+{
+public:
+    PlayerProxy(ConfigObject<ConfigValue> *pConfig);
+    ~PlayerProxy();
+    bool initialize() { return false; };
+    bool open();
+    void close();
+    void setDefaults();
+    QStringList getInterfaces();
+    QStringList getInputInterfaces();
+    QStringList getSampleRates();
+    QString getSoundApiName();
+    /** Return list of supported sound api's */
+    static QStringList getSoundApiList();
+    /** Select active sound api */
+    bool setSoundApi(QString name);
+    /** Satisfy virtual declaration in EngineObject */
+    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
+    /** Static function to return active player class. Needed for ASIO implementation */
+    static Player *getPlayer() { return m_pPlayer; }
+
+protected:
+    /** Pointer to active Player class */
+    static Player *m_pPlayer;
+};
+
+#endif
+
+
+
+
diff -Nru mixxx-1.6.0/src/old/playerrtaudio.cpp mixxx-1.6.1/src/old/playerrtaudio.cpp
--- mixxx-1.6.0/src/old/playerrtaudio.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerrtaudio.cpp	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,442 @@
+/***************************************************************************
+                          playerrtaudio.cpp  -  description
+                             -------------------
+    begin                : Thu May 20 2004
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+    email                :
+***************************************************************************/
+
+/***************************************************************************
+*                                                                         *
+*   This program is free software; you can redistribute it and/or modify  *
+*   it under the terms of the GNU General Public License as published by  *
+*   the Free Software Foundation; either version 2 of the License, or     *
+*   (at your option) any later version.                                   *
+*                                                                         *
+***************************************************************************/
+
+#include "playerrtaudio.h"
+#include "controlobject.h"
+
+/** Maximum frame size used with RtAudio. Used to determine no of buffers
+ * when setting latency */
+const int kiMaxFrameSize = 64;
+
+
+PlayerRtAudio::PlayerRtAudio(ConfigObject<ConfigValue> * config) : Player(config)
+{
+    m_pRtAudio = 0;
+    m_bInit = false;
+
+    m_devId = -1;
+    m_iNumberOfBuffers = 2;
+    m_iChannels = -1;
+    m_iMasterLeftCh = -1;
+    m_iMasterRigthCh = -1;
+    m_iHeadLeftCh = -1;
+    m_iHeadRightCh = -1;
+}
+
+PlayerRtAudio::~PlayerRtAudio()
+{
+    if (m_devId>=1)
+    {
+        try {
+            m_pRtAudio->stopStream();
+            m_pRtAudio->closeStream();
+        }
+        catch (RtError &error)
+        {
+            error.printMessage();
+        }
+    }
+
+    if (m_bInit)
+        delete m_pRtAudio;
+}
+
+bool PlayerRtAudio::initialize()
+{
+    try {
+        m_pRtAudio = new RtAudio();
+    }
+    catch (RtError &error)
+    {
+        error.printMessage();
+        m_pRtAudio = 0;
+        m_bInit = false;
+    }
+    m_bInit = true;
+
+    return m_bInit;
+}
+
+bool PlayerRtAudio::open()
+{
+    Player::open();
+
+    // Find out which device to open. Select the first one listed as either Master Left,
+    // Master Right, Head Left, Head Right. If other devices are requested for opening
+    // than the one selected here, set them to "None" in the config database
+    int id = -1;
+    int temp = -1;
+    QString name;
+    /** Maximum number of channels needed */
+    int iChannelMax = -1;
+
+    m_iMasterLeftCh = -1;
+    m_iMasterRigthCh = -1;
+    m_iHeadLeftCh = -1;
+    m_iHeadRightCh = -1;
+
+    // Master left
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
+    temp = getDeviceID(name);
+    if (temp>=0)
+    {
+        if (getChannelNo(name)>=0)
+        {
+            id = temp;
+            iChannelMax = getChannelNo(name);
+            m_iMasterLeftCh = getChannelNo(name)-1;
+        }
+    }
+
+    // Master right
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterRight"));
+    temp = getDeviceID(name);
+    if (getChannelNo(name)>=0 && ((id==-1 && temp>=0) || (temp!=-1 && id==temp)))
+    {
+        id = temp;
+        iChannelMax = math_max(iChannelMax, getChannelNo(name));
+        m_iMasterRigthCh = getChannelNo(name)-1;
+    }
+
+    // Head left
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadLeft"));
+    temp = getDeviceID(name);
+    if (getChannelNo(name)>=0 && ((id==-1 && temp>=0) || (temp!=-1 && id==temp)))
+    {
+        id = temp;
+        iChannelMax = math_max(iChannelMax, getChannelNo(name));
+        m_iHeadLeftCh = getChannelNo(name)-1;
+    }
+
+    // Head right
+    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadRight"));
+    temp = getDeviceID(name);
+    if (getChannelNo(name)>=0 && ((id==-1 && temp>=0) || (temp!=-1 && id==temp)))
+    {
+        id = temp;
+        iChannelMax = math_max(iChannelMax, getChannelNo(name));
+        m_iHeadRightCh = getChannelNo(name)-1;
+    }
+
+    // Check if any of the devices in the config database needs to be set to "None"
+    if (m_iMasterLeftCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+    if (m_iMasterRigthCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+    if (m_iHeadLeftCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+    if (m_iHeadRightCh<0)
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+
+    // Number of channels to open
+    int iChannels = iChannelMax;
+
+    // Sample rate
+    int iSrate = m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt();
+
+    // Get latency in msec
+    int iLatencyMSec = m_pConfig->getValueString(ConfigKey("[Soundcard]","Latency")).toInt();
+
+    // Latency in samples
+    int iLatencySamples = (int)((float)(iSrate*iChannels)/1000.f*(float)iLatencyMSec);
+
+    // Apply simple rule to determine number of buffers
+    if (iLatencySamples/kiMaxFrameSize<2)
+        m_iNumberOfBuffers = 2;
+    else
+        m_iNumberOfBuffers = iLatencySamples/kiMaxFrameSize;
+
+    // Frame size...
+    int iFramesPerBuffer = iLatencySamples/m_iNumberOfBuffers;
+
+    qDebug() << "RtAudio: id " << id << ", sr " << iSrate << ", ch " << iChannels << ", bufsize " << iFramesPerBuffer << ", bufno " << m_iNumberOfBuffers;
+
+    if (id<1)
+    {
+        m_devId = -1;
+        return false;
+    }
+
+    // Start playback
+    try
+    {
+        // Update SRATE and Latency ControlObjects
+        m_pControlObjectSampleRate->queueFromThread((double)iSrate);
+        m_pControlObjectLatency->queueFromThread((double)iLatencyMSec);
+
+        // Open stream
+        m_pRtAudio->openStream(id, iChannels, 0, 0, RTAUDIO_FLOAT32, iSrate, &iFramesPerBuffer, m_iNumberOfBuffers);
+
+        // Set callback function
+        m_pRtAudio->setStreamCallback(&rtCallback, (void *)this);
+
+        m_iChannels = iChannels;
+        m_devId = id;
+
+        // Start playback
+        m_pRtAudio->startStream();
+    }
+    catch (RtError &error)
+    {
+        error.printMessage();
+
+        m_devId = -1;
+        m_iChannels = -1;
+
+        return false;
+    }
+
+    return true;
+}
+
+void PlayerRtAudio::close()
+{
+    m_iChannels = 0;
+    m_iMasterLeftCh = -1;
+    m_iMasterRigthCh = -1;
+    m_iHeadLeftCh = -1;
+    m_iHeadRightCh = -1;
+
+    if (m_devId>0)
+    {
+        qDebug() << "close";
+        try
+        {
+            m_pRtAudio->stopStream();
+            m_pRtAudio->closeStream();
+        }
+        catch (RtError &error)
+        {
+            error.printMessage();
+        }
+    }
+    m_devId = -1;
+}
+
+void PlayerRtAudio::setDefaults()
+{
+    // Get list of interfaces
+    QStringList interfaces = getInterfaces();
+
+    // Set first interfaces to master left
+    QStringList::iterator it = interfaces.begin();
+    if (it!=interfaces.end())
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((*it)));
+    }
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
+
+    // Set second interface to master right
+    ++it;
+    if (it!=interfaces.end())
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((*it)));
+    else
+        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
+
+    // Set head left and right to none
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
+    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
+
+    // Set default sample rate
+    QStringList srates = getSampleRates();
+    it = srates.begin();
+    while (it!=srates.end())
+    {
+        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue((*it)));
+
+        if ((*it).toInt()>=44100)
+            break;
+
+        ++it;
+    }
+
+    // Set currently used latency in config database
+    int msec = (int)(1000.*(2.*1024.)/(2.*(float)(*it).toInt()));
+    m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
+}
+
+QStringList PlayerRtAudio::getInterfaces()
+{
+    QStringList result;
+
+    int no = m_pRtAudio->getDeviceCount();
+    RtAudioDeviceInfo info;
+    for (int i=1; i<=no; i++)
+    {
+        bool bGotInfo = false;
+        try {
+            info = m_pRtAudio->getDeviceInfo(i);
+            bGotInfo = true;
+        }
+        catch (RtError &error)
+        {
+            error.printMessage();
+        }
+
+        // Add the device if it is an output device:
+        if (bGotInfo && info.outputChannels > 0)
+        {
+            qDebug() << "name " << info.name.c_str();
+            for (int j=1; j<=info.outputChannels; ++j)
+                result.append(QString("%1 (channel %2)").arg(info.name.c_str()).arg(j));
+        }
+    }
+
+    return result;
+}
+
+QStringList PlayerRtAudio::getSampleRates()
+{
+    // Returns the list of supported sample rates of the currently opened device.
+    // If no device is open, return the list of sample rates supported by the
+    // default device
+    int id = m_devId;
+    if (id<1)
+        id = 1;
+
+    RtAudioDeviceInfo info;
+    try {
+        info = m_pRtAudio->getDeviceInfo(id);
+    }
+    catch (RtError &error)
+    {
+        error.printMessage();
+        return QStringList();
+    }
+
+    // Sample rates
+    QValueList<int> srlist;
+
+    if (info.sampleRates.size() > 0)
+    {
+        for (unsigned int j=0; j<info.sampleRates.size(); ++j)
+            srlist.append(info.sampleRates[j]);
+    }
+
+    // Sort list
+#ifndef QT3_SUPPORT
+    qHeapSort(srlist);
+#endif
+
+    // Convert srlist to stringlist
+    QStringList result;
+    for (int i=0; i<srlist.count(); ++i)
+        result.append(QString("%1").arg((*srlist.at(i))));
+
+    return result;
+}
+
+QString PlayerRtAudio::getSoundApi()
+{
+#ifdef __WIN__
+    return QString("DirectSound");
+#endif
+#ifdef __LINUX__
+    return QString("ALSA");
+#endif
+}
+
+int PlayerRtAudio::getDeviceID(QString name)
+{
+    int no = m_pRtAudio->getDeviceCount();
+    for (int i=1; i<=no; i++)
+    {
+        RtAudioDeviceInfo info;
+        try {
+            info = m_pRtAudio->getDeviceInfo(i);
+        }
+        catch (RtError &error)
+        {
+            error.printMessage();
+            return -1;
+        }
+
+        // Add the device if it is an output device:
+        if (info.outputChannels > 0)
+        {
+            for (int j=1; j<=info.outputChannels; ++j)
+                if (QString("%1 (channel %2)").arg(info.name.c_str()).arg(j) == name)
+                    return i;
+        }
+    }
+    return -1;
+}
+
+int PlayerRtAudio::getChannelNo(QString name)
+{
+    int no = m_pRtAudio->getDeviceCount();
+    for (int i=1; i<=no; i++)
+    {
+        RtAudioDeviceInfo info;
+        try {
+            info = m_pRtAudio->getDeviceInfo(i);
+        }
+        catch (RtError &error)
+        {
+            error.printMessage();
+            return -1;
+        }
+
+        // Add the device if it is an output device:
+        if (info.outputChannels > 0)
+        {
+            for (int j=1; j<=info.outputChannels; ++j)
+                if (QString("%1 (channel %2)").arg(info.name.c_str()).arg(j) == name)
+                    return j;
+        }
+    }
+    return -1;
+}
+
+int PlayerRtAudio::callbackProcess(int iBufferSize, float * out)
+{
+    //m_iBufferSize = iBufferSize*m_iNumberOfBuffers;
+
+    float * tmp = prepareBuffer(iBufferSize);
+    float * output = out;
+    int i;
+
+    // Reset sample for each open channel
+    for (i=0; i<iBufferSize*m_iChannels; i++)
+        output[i] = 0.;
+
+    // Copy to output buffer
+    for (i=0; i<iBufferSize; i++)
+    {
+        if (m_iMasterLeftCh>=0) output[m_iMasterLeftCh]  += tmp[(i*4)  ]/32768.;
+        if (m_iMasterRigthCh>=0) output[m_iMasterRigthCh] += tmp[(i*4)+1]/32768.;
+        if (m_iHeadLeftCh>=0) output[m_iHeadLeftCh]    += tmp[(i*4)+2]/32768.;
+        if (m_iHeadRightCh>=0) output[m_iHeadRightCh]   += tmp[(i*4)+3]/32768.;
+
+        for (int j=0; j<m_iChannels; ++j)
+            *output++;
+    }
+
+    return 0;
+}
+
+/* -------- ------------------------------------------------------
+   Purpose: Wrapper function to call processing loop function,
+            implemented as a method in a class. Used in RtAudio.
+   Input:   .
+   Output:  -
+   -------- ------------------------------------------------------ */
+int rtCallback(char * outputBuffer, int framesPerBuffer, void * pPlayer)
+{
+    return ((PlayerRtAudio *)pPlayer)->callbackProcess(framesPerBuffer, (float *)outputBuffer);
+}
diff -Nru mixxx-1.6.0/src/old/playerrtaudio.h mixxx-1.6.1/src/old/playerrtaudio.h
--- mixxx-1.6.0/src/old/playerrtaudio.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/old/playerrtaudio.h	2008-08-11 05:02:03.000000000 -0500
@@ -0,0 +1,65 @@
+/***************************************************************************
+                          playerrtaudio.h  -  description
+                             -------------------
+    begin                : Thu May 20 2004
+    copyright            : (C) 2002 by Tue and Ken Haste Andersen
+    email                :
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#ifndef PLAYERRTAUDIO_H
+#define PLAYERRTAUDIO_H
+
+#include "player.h"
+#include <RtAudio.h>
+
+/**
+  *@author Tue and Ken Haste Andersen
+  */
+
+
+class PlayerRtAudio : public Player  {
+public:
+    PlayerRtAudio(ConfigObject<ConfigValue> *config);
+    ~PlayerRtAudio();
+    bool initialize();
+    bool open();
+    void close();
+    void setDefaults();
+    QStringList getInterfaces();
+    QStringList getSampleRates();
+    static QString getSoundApi();
+    QString getSoundApiName() { return getSoundApi(); };
+    /** Satisfy virtual declaration in EngineObject */
+    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
+    /** Process samples. Called from RtAudio callback */
+    int callbackProcess(int iBufferSize, float *out);
+
+protected:
+    /** Pointer to RtAudio object */
+    RtAudio *m_pRtAudio;
+    /** Get id of device with a given name. Returns -1 if device is not found */
+    int getDeviceID(QString name);
+    /** Get channel number of device with a given name. Returns -1 if device is no found */
+    int getChannelNo(QString name);
+    /** Id of currently open device. -1 if no device is open */
+    int m_devId;
+    /** Channels used for each output from Mixxx. Set to -1 when not in use */
+    int m_iMasterLeftCh, m_iMasterRigthCh, m_iHeadLeftCh, m_iHeadRightCh;
+    /** True if RtAudio was sucessfully initialized */
+    bool m_bInit;
+    /** Number of buffers */
+    int m_iNumberOfBuffers;
+};
+
+int rtCallback(char *outputBuffer, int framesPerBuffer, void *_player);
+
+#endif
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/osx/VolumeIcon.icns and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/osx/VolumeIcon.icns differ
diff -Nru mixxx-1.6.0/src/playeralsa.cpp mixxx-1.6.1/src/playeralsa.cpp
--- mixxx-1.6.0/src/playeralsa.cpp	2008-03-27 00:04:43.000000000 -0500
+++ mixxx-1.6.1/src/playeralsa.cpp	1969-12-31 18:00:00.000000000 -0600
@@ -1,757 +0,0 @@
-/***************************************************************************
-                          playeralsa.cpp  -  description
-                             -------------------
-    begin                : Wed Feb 20 2002
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-    email                :
-***************************************************************************/
-
-/***************************************************************************
-*                                                                         *
-*   This program is free software; you can redistribute it and/or modify  *
-*   it under the terms of the GNU General Public License as published by  *
-*   the Free Software Foundation; either version 2 of the License, or     *
-*   (at your option) any later version.                                   *
-*                                                                         *
-***************************************************************************/
-
-#include "playeralsa.h"
-#include "controlobject.h"
-
-#ifndef PLAYERTEST
-#include "rtthread.h"
-#endif
-
-// Tutorials /home/peter/pad-alsa-audio.html /home/peter/alsa090_howto.html
-// Docs      /usr/share/doc/alsa-lib-1.0.4/doxygen/html/index.html
-// Example   /usr/share/doc/alsa-lib-1.0.4/doxygen/html/_2test_2pcm_8c-example.html
-// QT        /usr/share/doc/qt-devel-3.1.1/html/index.html
-
-/** Maximum frame size used with ALSA. Used to determine no of buffers
- *   * when setting latency */
-const int kiMaxFrameSize = 512;
-
-
-#ifndef PLAYERTEST
-PlayerALSA::PlayerALSA(ConfigObject<ConfigValue> * config) : Player(config)
-#else
-PlayerALSA::PlayerALSA()
-#endif
-{
-    int err;
-
-    handle = 0;
-    err = snd_pcm_hw_params_malloc(&hwparams);
-    if (err < 0)
-        qCritical("Couldn't allocate memory for hw params: %s\n", snd_strerror(err));
-
-    err = snd_pcm_sw_params_malloc(&swparams);
-    if (err < 0)
-        qCritical("Couldn't allocate memory for sw params: %s\n", snd_strerror(err));
-
-    err = snd_pcm_sw_params_malloc(&swparams);
-    if (err < 0)
-        qCritical("Couldn't allocate memory for sw params: %s\n", snd_strerror(err));
-
-    isopen = false;
-    masterleft = masterright = -1;
-    headleft = headright = -1;
-    twrite = false;
-    m_iChannels = alsa_channels;
-    qDebug() << "Alsa constructed";
-}
-
-PlayerALSA::~PlayerALSA()
-{
-    if (isopen) close();
-
-    qDebug() << "Alsa destroying...";
-    if (hwparams)
-    {
-        snd_pcm_hw_params_free(hwparams);
-    }
-    if (swparams)
-    {
-        snd_pcm_sw_params_free(swparams);
-    }
-    qDebug() << "Alsa destroyed";
-}
-
-bool PlayerALSA::initialize()
-{
-    qDebug() << "Alsa initialized";
-
-    return true;
-}
-
-bool PlayerALSA::open()
-{
-#ifndef PLAYERTEST
-    Player::open();
-#endif
-    QString name;
-    QString devname, devtmp;
-    int err;
-
-#ifndef PLAYERTEST
-    QRegExp rx("(\\S+) \\(ch (\\d+)\\)");
-
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceMasterLeft"));
-    if (name != "None")
-    {
-        if (rx.search(name) < 0)
-            qWarning() << "can't find device name or channel number in" << name;
-
-        devname = rx.cap(1);
-    }
-    else
-    {
-        setDefaults(); // initialise defaults
-        return open();
-    }
-#else
-    devname = QString("surround40:0");
-#endif
-
-    // If no device was selected return false
-    if (!devname)
-        return false;
-
-    qDebug() << "Alsa opening pcm_open:" << devname;
-
-    if ((err = snd_pcm_open(&handle, devname.ascii(),
-                            SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
-        // AD: Try for default device instead
-        if ((err = snd_pcm_open(&handle, "default",
-                                SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
-            qWarning("Playback open error: %s\n", snd_strerror(err));
-            return false;
-        } else {
-            qWarning("Using default ALSA device, you may want to set up a ~/.asoundrc to enable all the features of your sound card");
-        }
-    }
-
-    qDebug() << "Alsa setting hw";
-    if ((err = set_hwparams()) < 0)
-    {
-        qWarning("Setting of hwparams failed: %s\n", snd_strerror(err));
-        snd_pcm_close(handle);
-        return false;
-    }
-
-    qDebug() << "Alsa setting sw";
-    if ((err = set_swparams()) < 0)
-    {
-        qWarning("Setting of swparams failed: %s\n", snd_strerror(err));
-        snd_pcm_close(handle);
-        return false;
-    }
-
-    masterleft = masterright = -1;
-    headleft = headright = -1;
-
-#ifndef PLAYERTEST
-    int temp;
-
-    // master left
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceMasterLeft"));
-    qDebug() << "Alsa opening..." << name;
-    if (name != "None")
-    {
-        if (rx.search(name) < 0)
-        {
-            qWarning() << "can't find device name or channel number in" << name;
-        }
-//      devname = rx.cap(1);
-        temp = rx.cap(2).toInt();
-        if (temp > 0 && temp <= m_iChannels)
-        {
-            masterleft = temp - 1;
-        }
-    }
-    qDebug() << "Alsa opening...ML " << masterleft;
-
-    // master right
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceMasterRight"));
-    if (name != "None")
-    {
-        if (rx.search(name) < 0)
-        {
-            qWarning() << "can't find device name or channel number in" << name;
-        }
-        devtmp = rx.cap(1);
-        if (devname != devtmp)
-        {
-            qWarning("device name mismatch, only one device supported");
-        }
-        temp = rx.cap(2).toInt();
-        if (temp > 0 && temp <= m_iChannels)
-        {
-            masterright = temp - 1;
-        }
-    }
-    qDebug() << "Alsa opening...MR " << masterright;
-
-    // head left
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceHeadLeft"));
-    if (name != "None")
-    {
-        if (rx.search(name) < 0)
-        {
-            qWarning() << "can't find device name or channel number in" << name;
-        }
-        devtmp = rx.cap(1);
-        if (devname != devtmp)
-        {
-            qWarning("device name mismatch, only one device supported");
-        }
-        temp = rx.cap(2).toInt();
-        if (temp > 0 && temp <= m_iChannels)
-        {
-            headleft = temp - 1;
-        }
-    }
-    qDebug() << "Alsa opening...HL " << headleft;
-
-    // head right
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]", "DeviceHeadRight"));
-    if (name != "None")
-    {
-        if (rx.search(name) < 0)
-        {
-            qWarning() << "can't find device name or channel number in" << name;
-        }
-        devtmp = rx.cap(1);
-        if (devname != devtmp)
-        {
-            qWarning("device name mismatch, only one device supported");
-        }
-        temp = rx.cap(2).toInt();
-        if (temp > 0 && temp <= m_iChannels)
-        {
-            headright = temp - 1;
-        }
-    }
-
-    qDebug() << "Alsa opening...HR " << headright;
-    // Check if any of the devices in the config database needs to be set to "None"
-    if (masterleft < 0)
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterLeft"), ConfigValue("None"));
-    if (masterright < 0)
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterRight"), ConfigValue("None"));
-    if (headleft < 0)
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadLeft"), ConfigValue("None"));
-    if (headright < 0)
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadRight"), ConfigValue("None"));
-
-#else
-    masterleft = 0;
-    masterright = 1;
-    headleft = 2;
-    headright = 3;
-#endif
-
-    twrite = true;
-    isopen = true;
-
-    QThread::start();
-
-    return true;
-}
-
-/*
- *   Underrun and suspend recovery
- */
-int PlayerALSA::xrun_recovery(int err)
-{
-    if (err == -EPIPE) {        /* under-run */
-        err = snd_pcm_prepare(handle);
-        if (err < 0)
-            qWarning("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err));
-        return 0;
-    }
-    else if (err == -ESTRPIPE)
-    {
-        while ((err = snd_pcm_resume(handle)) == -EAGAIN)
-            sleep(1);/* wait until the suspend flag is released */
-        if (err < 0)
-        {
-            err = snd_pcm_prepare(handle);
-            if (err < 0)
-                qWarning("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err));
-        }
-        return 0;
-    }
-    return err;
-}
-
-/**
- * thread for writing samples out
- */
-void PlayerALSA::run()
-{
-    CSAMPLE * sptr;
-    int err, ctr;
-
-#ifndef PLAYERTEST
-    rtThread();
-#endif
-
-    if (isformatfloat) {
-        float * optr, * output;
-
-        output = new float[buffer_size * m_iChannels];
-        qDebug() << "Alsa allocating output buffer (FLOAT) " << output;
-
-        while (twrite)
-        {
-            sptr = prepareBuffer((int) period_size);
-            optr = output;
-            for (int i = 0; i < (int) period_size; i++)
-            {
-                for (int j = 0; j < m_iChannels; j++) optr[j] = 0;
-                if (masterleft >= 0) optr[masterleft]  = *sptr / 32768.;
-                sptr++;
-                if (masterright >= 0) optr[masterright] = *sptr / 32768.;
-                sptr++;
-                if (headleft >= 0) optr[headleft]  = *sptr / 32768.;
-                sptr++;
-                if (headright >= 0) optr[headright] = *sptr / 32768.;
-                sptr++;
-                optr += m_iChannels;
-            }
-            ctr = period_size;
-            optr = output;
-            while (ctr > 0)
-            {
-                err = snd_pcm_writei(handle, optr, ctr);
-                if (err == -EAGAIN) continue;
-                if (err < 0)
-                {
-                    if (xrun_recovery(err) < 0)
-                        qCritical("Write error: %s\n", snd_strerror(err));
-
-                    break;
-                }
-                optr += err*m_iChannels;
-                ctr -= err;
-            }
-        }
-        if (output)
-            delete output;
-    }
-    else
-    {
-        short int * optr, * output;
-
-        output = new short int[buffer_size * m_iChannels];
-        qDebug() << "Alsa allocating output buffer (S16) " << output;
-
-        while (twrite)
-        {
-            sptr = prepareBuffer((int) period_size);
-            optr = output;
-            for (int i = 0; i < (int) period_size; i++)
-            {
-                for (int j = 0; j < m_iChannels; j++) optr[j] = 0;
-                if (masterleft >= 0) optr[masterleft]  = (short int) *sptr;
-                sptr++;
-                if (masterright >= 0) optr[masterright] = (short int) *sptr;
-                sptr++;
-                if (headleft >= 0) optr[headleft]  = (short int) *sptr;
-                sptr++;
-                if (headright >= 0) optr[headright] = (short int) *sptr;
-                sptr++;
-                optr += m_iChannels;
-            }
-            ctr = period_size;
-            optr = output;
-            while (ctr > 0)
-            {
-                err = snd_pcm_writei(handle, optr, ctr);
-                if (err == -EAGAIN) continue;
-                if (err < 0)
-                {
-                    if (xrun_recovery(err) < 0)
-                        qCritical("Write error: %s\n", snd_strerror(err));
-                    break;
-                }
-                optr += err*m_iChannels;
-                ctr -= err;
-            }
-        }
-        if (output) delete output;
-    }
-    qDebug() << "Alsa leaving thread";
-}
-
-#ifdef PLAYERTEST
-/**
- * generate a middle A slowly shifting across the channels
- */
-CSAMPLE * PlayerALSA::prepareBuffer(int count)
-{
-    const double freq = 440.;
-    const unsigned int rate = 44100;
-    static CSAMPLE * out = 0;
-
-    static double phase = 0;
-    double max_phase = 1.0 / freq;
-    double step = 1.0 / (double) rate;
-    double res;
-    CSAMPLE * sptr;
-    int chn;
-    static int channel = 0;
-    static int ccount = 0;
-
-    if (out == 0)
-        out = new CSAMPLE[80000];
-    sptr = out;
-    while (count-- > 0)
-    {
-        res = 32768. * sin((phase * 2 * M_PI) / max_phase - M_PI);
-        for (chn = 0; chn < alsa_channels; chn++)
-        {
-            if (chn == channel)
-                *sptr = res;
-            else
-                *sptr = 0;
-            sptr++;
-        }
-        phase += step;
-        if (phase >= max_phase)
-        {
-            phase -= max_phase;
-            ccount++;
-            if (ccount == (3 * 440))
-            {
-                ccount = 0;
-                channel++;
-                if (channel == alsa_channels) channel = 0;
-            }
-        }
-    }
-    return out;
-}
-#endif
-
-void PlayerALSA::close()
-{
-    if (!isopen)
-        return;
-
-    qDebug() << "Alsa closing";
-    twrite = false;
-    qDebug() << "Alsa waiting for thread for close";
-    QThread::wait();
-
-    snd_pcm_drop(handle);
-    snd_pcm_close(handle);
-
-    isopen = false;
-}
-
-void PlayerALSA::setDefaults()
-{
-    qDebug() << "Alsa setdefs";
-#ifndef PLAYERTEST
-    // Get list of interfaces
-    QStringList interfaces = getInterfaces();
-
-    // Set first interfaces to master left
-    QStringList::iterator it = interfaces.begin();
-    if (*it)
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterLeft"), ConfigValue((*it)));
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterLeft"), ConfigValue("None"));
-
-    // Set second interface to master right
-    ++it;
-    if (*it)
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterRight"), ConfigValue((*it)));
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]", "DeviceMasterRight"), ConfigValue("None"));
-
-    // Set head left and right to none
-    m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadLeft"), ConfigValue("None"));
-    m_pConfig->set(ConfigKey("[Soundcard]", "DeviceHeadRight"), ConfigValue("None"));
-
-    // Set default sample rate
-    QStringList srates = getSampleRates();
-    it = srates.begin();
-    while (*it)
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]", "Samplerate"), ConfigValue((*it)));
-        if ((*it).toInt() >= 44100)
-            break;
-        ++it;
-    }
-
-    // Set currently used latency in config database
-    m_pConfig->set(ConfigKey("[Soundcard]", "Latency"), ConfigValue(default_latency));
-#endif
-}
-
-QStringList PlayerALSA::getInterfaces()
-{
-    qDebug() << "Alsa getinter";
-    QStringList result;
-
-    result.append("mixxx (ch 1)");
-    result.append("mixxx (ch 2)");
-    result.append("mixxx (ch 3)");
-    result.append("mixxx (ch 4)");
-
-    return result;
-}
-
-QStringList PlayerALSA::getSampleRates()
-{
-    qDebug() << "Alsa getsample";
-    QStringList result;
-    result.append("11025");
-    result.append("22050");
-    result.append("44100");
-    result.append("48000");
-    return result;
-}
-
-QString PlayerALSA::getSoundApi()
-{
-    qDebug() << "Alsa getapi";
-    return QString("Alsa");
-}
-
-int PlayerALSA::set_hwparams()
-{
-    unsigned int rate, rrate;
-    unsigned int buffer_time = 500000;  /* ring buffer length in us */
-    unsigned int period_time = 100000;  /* period time in us */
-
-    int err, dir;
-
-    /* choose all parameters */
-    err = snd_pcm_hw_params_any(handle, hwparams);
-    if (err < 0)
-    {
-        qWarning("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
-        return err;
-    }
-    /* set the interleaved read/write format */
-    err = snd_pcm_hw_params_set_access(handle, hwparams, alsa_access);
-    if (err < 0)
-    {
-        qWarning("Access type not available for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    /* set the sample format */
-    err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_FLOAT);
-    if (err < 0)
-    {
-        qWarning("Sample format (FLOAT) not available for playback: %s\n", snd_strerror(err));
-        qWarning("Falling back to S16");
-        err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16);
-        if (err < 0)
-        {
-            qWarning("Sample format (S16) not available for playback: %s\n", snd_strerror(err));
-            return err;
-        }
-        isformatfloat = false;
-    } else {
-        isformatfloat = true;
-    }
-
-    /* find min and max number of channels */
-    err = snd_pcm_hw_params_get_channels_min(hwparams, &rate);
-    if (err < 0)
-    {
-        qWarning("Channels count (%i) not available for playbacks: %s\n", rate, snd_strerror(err));
-    }
-    qDebug() << "Channels min " << rate;
-    err = snd_pcm_hw_params_get_channels_max(hwparams, &rate);
-    if (err < 0)
-    {
-        qWarning("Channels count (%i) not available for playbacks: %s\n", rate, snd_strerror(err));
-    }
-    qDebug() << "Channels max " << rate;
-    m_iChannels = math_min(rate, alsa_channels);
-    qDebug() << "Set channels = " << m_iChannels;
-
-    /* set the count of channels */
-    err = snd_pcm_hw_params_set_channels(handle, hwparams, m_iChannels);
-    if (err < 0)
-    {
-        qWarning("Channels count (%i) not available for playbacks: %s\n", m_iChannels, snd_strerror(err));
-        return err;
-    }
-    /* set the stream rate */
-#ifndef PLAYERTEST
-    rate = m_pConfig->getValueString(ConfigKey("[Soundcard]", "Samplerate")).toUInt();
-#else
-    rate = 44100;
-#endif
-    rrate = rate;
-    err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rrate, 0);
-    if (err < 0)
-    {
-        qWarning("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err));
-        return err;
-    }
-    if (rrate != rate)
-    {
-        qWarning("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err);
-        return -EINVAL;
-    }
-
-#ifndef PLAYERTEST
-//     setPlaySrate(rrate);
-//     m_pControlObjectSampleRate->queueFromThread((double)rrate);
-
-    // Update SRATE and Latency ControlObjects
-    m_pControlObjectSampleRate->queueFromThread((double)rrate);
-//     m_pControlObjectLatency->queueFromThread((double)iLatencyMSec);
-//
-#endif
-
-    /* set the buffer time */
-#ifndef PLAYERTEST
-    buffer_time = m_pConfig->getValueString(ConfigKey("[Soundcard]", "Latency")).toUInt() * 1000;
-#else
-    buffer_time = 50000;
-#endif
-    /* figure out size and make it even */
-    buffer_size = (int) rrate*(1e-6*buffer_time) + 1;
-
-    snd_pcm_uframes_t max = 0;
-    snd_pcm_uframes_t min = 0;
-
-    // FIXME: Should check return values to be on the safe side
-    snd_pcm_hw_params_get_buffer_size_max(hwparams, &max);
-    snd_pcm_hw_params_get_buffer_size_min(hwparams, &min);
-    qDebug() << "Allowed buffer size range: " << min << "i -> " << max << "i";
-
-    // Make sure buffer size we're aiming for is really allowed
-    if (buffer_size < min) { buffer_size = (int)min; }
-    if (buffer_size > max) { buffer_size = (int)max; }
-
-    buffer_size -= (buffer_size & 1);
-
-    if (buffer_size < min) { buffer_size += 2; }
-
-    /* test buffer size first */
-    err = snd_pcm_hw_params_test_buffer_size(handle, hwparams, buffer_size);
-    if (err < 0) {
-        qWarning("Unable to test buffer size %d", buffer_size);
-        return err;
-    }
-    err = snd_pcm_hw_params_set_buffer_size(handle, hwparams, buffer_size);
-    if (err < 0)
-    {
-        qWarning("Unable to set buffer size for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    qWarning("Buffer %dus [actual %dus] (size: %d)", (int) buffer_time, (int) (1e6*buffer_size/rrate), buffer_size);
-
-    /*
-     * work out maximum number of periods so that only kiMaxFrameSize is
-     * written out at a time, then use this as a starting point to find
-     * period that works
-     */
-    int period_no_start = 2;
-
-    if (buffer_size/kiMaxFrameSize>2)
-        period_no_start = buffer_size/kiMaxFrameSize;
-
-    for (period_no = period_no_start; period_no > 1; period_no--)
-    {
-        unsigned int period_time = buffer_time / period_no;
-        qWarning("Period %dus (no: %d)", (int) period_time, period_no);
-
-        int dir = 0;
-        err = snd_pcm_hw_params_test_periods(handle, hwparams, period_no, dir);
-        if (err < 0)
-        {
-            qWarning("Unable to test periods %i (dir %d) for playback: %s\n", period_no, dir,
-                     snd_strerror(err));
-        }
-        else
-        {
-            err = snd_pcm_hw_params_set_periods(handle, hwparams, period_no, dir);
-            if (err < 0)
-            {
-                qWarning("Unable to set periods %i for playback: %s\n", period_no, snd_strerror(err));
-                return err;
-            }
-            err = snd_pcm_hw_params_get_period_size(hwparams, &period_size, &dir);
-            if (err < 0)
-            {
-                qWarning("Unable to get period size for playback: %s\n", snd_strerror(err));
-                return err;
-            }
-            qWarning("Period %df, buffer %df", (int) period_size, (int) buffer_size);
-            break;
-        }
-    }
-
-    if (period_no == 1)
-        return -1;
-
-    /* write the parameters to device */
-    err = snd_pcm_hw_params(handle, hwparams);
-    if (err < 0)
-    {
-        qWarning("Unable to set hw params for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    qWarning("Rate: %d, buf %df (%dus), per %df (%dus)", rrate,
-             (int) buffer_size, (int) (buffer_size / (rrate * 1e-6)),
-             (int) period_size, (int) (period_size / (rrate * 1e-6)));
-
-    // Update SRATE and Latency ControlObjects
-    m_pControlObjectSampleRate->queueFromThread((double)rrate);
-    m_pControlObjectLatency->queueFromThread((double)buffer_time/1000.);
-
-    return 0;
-}
-
-int PlayerALSA::set_swparams()
-{
-    int err;
-
-    /* get the current swparams */
-    err = snd_pcm_sw_params_current(handle, swparams);
-    if (err < 0)
-    {
-        qWarning("Unable to determine current swparams for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    /* start the transfer when the buffer is full */
-    err = snd_pcm_sw_params_set_start_threshold(handle, swparams, (buffer_size/period_size)*period_size);
-    if (err < 0)
-    {
-        qWarning("Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    /* allow the transfer when at least period_size samples can be processed */
-    err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_size);
-    if (err < 0)
-    {
-        qWarning("Unable to set avail min for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    /* align all transfers to 1 sample */
-    err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
-    if (err < 0)
-    {
-        qWarning("Unable to set transfer align for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    /* write the parameters to the playback device */
-    err = snd_pcm_sw_params(handle, swparams);
-    if (err < 0)
-    {
-        qWarning("Unable to set sw params for playback: %s\n", snd_strerror(err));
-        return err;
-    }
-    return 0;
-}
diff -Nru mixxx-1.6.0/src/playeralsa.h mixxx-1.6.1/src/playeralsa.h
--- mixxx-1.6.0/src/playeralsa.h	2005-03-18 17:01:48.000000000 -0600
+++ mixxx-1.6.1/src/playeralsa.h	1969-12-31 18:00:00.000000000 -0600
@@ -1,107 +0,0 @@
-/***************************************************************************
-                          playeralsa.h  -  description
-                             -------------------
-    begin                : Wed Feb 20 2002
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-    email                : 
- ***************************************************************************/
-
-/***************************************************************************
- *                                                                         *
- *   This program is free software; you can redistribute it and/or modify  *
- *   it under the terms of the GNU General Public License as published by  *
- *   the Free Software Foundation; either version 2 of the License, or     *
- *   (at your option) any later version.                                   *
- *                                                                         *
- ***************************************************************************/
-
-#ifndef PLAYERALSA_H
-#define PLAYERALSA_H
-
-/**
-  *@author Tue and Ken Haste Andersen, Peter Chang
-  */
-
-#ifndef PLAYERTEST
-#include "player.h"
-#else
-// some definitions to avoid using player.h in test mode
-
-#include <math.h>
-
-#include <qstring.h>
-#include <qstringlist.h>
-
-typedef float CSAMPLE; // defines the CSAMPLE type used for intermediate calculations
-#endif
-
-#include <qregexp.h>
-#include <qthread.h>
-#include <alsa/asoundlib.h>
-
-/**
- * there are two independent defines S16_OUTPUT and DIRECT_OUTPUT
- *  1) S16_OUTPUT makes the class output short integers to ALSA
- *  2) DIRECT_OUTPUT uses mmapped output
- * 
- * NB by default, these are not defined and so the class uses floats via the
- * plug plugin. Defining S16_OUTPUT allows the bare hw to be used (via surround40).
- * XXX: Mmapped output doesn't seem to work with the plug plugin...
- */
-
-#ifndef PLAYERTEST
-class PlayerALSA : public Player, public QThread 
-{
-public:
-    PlayerALSA(ConfigObject<ConfigValue> *config);
-#else
-class PlayerALSA : public QThread {
-public:
-    PlayerALSA();
-#endif
-    ~PlayerALSA();
-    bool initialize();
-    bool open();
-    void close();
-    void setDefaults();
-    QStringList getInterfaces();
-    QStringList getSampleRates();
-    static QString getSoundApi();
-    QString getSoundApiName() { return getSoundApi(); };
-    /** Satisfy virtual declaration in EngineObject */
-    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
-
-    /** Main loop of player. Executed in a separate thread by QT */
-    void run();
-
-protected:
-/** ALSA parameters */
-    snd_pcm_t *handle;
-    snd_pcm_hw_params_t *hwparams;
-    snd_pcm_sw_params_t *swparams;
-
-    snd_pcm_uframes_t buffer_size;
-    snd_pcm_uframes_t period_size;
-    unsigned int period_no;
-    bool isformatfloat;
-
-    /** True if devices are open */
-    bool isopen;
-    int masterleft, masterright, headleft, headright;
-
-#ifdef PLAYERTEST
-    CSAMPLE *prepareBuffer(int nframes);
-#endif
-    int setPeriodSize(bool setMinimum);
-    int set_hwparams();
-    int set_swparams();
-    int xrun_recovery(int err);
-
-    bool twrite; // flag for thread
-
-private:
-    static const snd_pcm_access_t alsa_access = SND_PCM_ACCESS_RW_INTERLEAVED;
-    static const int alsa_channels = 4;
-    static const int default_latency = 200;
-};
-#endif
diff -Nru mixxx-1.6.0/src/playerasio.cpp mixxx-1.6.1/src/playerasio.cpp
--- mixxx-1.6.0/src/playerasio.cpp	2008-03-27 00:04:43.000000000 -0500
+++ mixxx-1.6.1/src/playerasio.cpp	1969-12-31 18:00:00.000000000 -0600
@@ -1,670 +0,0 @@
-/***************************************************************************
-                          PlayerAsio.cpp  -  description
-                             -------------------
-    begin                : Thu June 02 2005
-    copyright            : (C) 2005 by Ingo Kossyk
-    email                :
-***************************************************************************/
-
-/***************************************************************************
-*                                                                         *
-*   This program is free software; you can redistribute it and/or modify  *
-*   it under the terms of the GNU General Public License as published by  *
-*   the Free Software Foundation; either version 2 of the License, or     *
-*   (at your option) any later version.                                   *
-*                                                                         *
-***************************************************************************/
-
-
-// PlayerAsio.cpp: implementation of the PlayerAsio class.
-//
-//////////////////////////////////////////////////////////////////////
-
-//////////////////////////////////////////////////////////////////////
-// ToDo:
-// - Implement all conversion routines for the different drivers
-// - Testing on different soundcards
-// - Correctly (???) setting the latency in the preferences and locking it
-// - Samplerate Changed needs to be implemented if changed in the Preferences
-//////////////////////////////////////////////////////////////////////
-
-#include "qstringlist.h"
-#include "PlayerAsio.h"
-#include "playerproxy.h"
-#include "controlobject.h"
-//////////////////////////////////////////////////////////////////////
-// External Declarations
-//////////////////////////////////////////////////////////////////////
-
-__declspec(dllimport) AsioDrivers* asioDrivers;
-ASIOCallbacks asioCallbacks;
-DriverInfo asioDriverInfo;
-
-//////////////////////////////////////////////////////////////////////
-// Construction/Destruction
-//////////////////////////////////////////////////////////////////////
-
-const int MAX_DRIVER_NAME_LENGTH = 32;
-
-PlayerAsio::PlayerAsio(ConfigObject<ConfigValue> * config) : Player(config)
-
-{
-    m_pTempBuffer = 0;
-    m_reenter = 0;
-    bufferLength = 0;
-    driverIsLoaded = false;
-
-    for (int i = 0 ; i < 32 ; ++i) {
-        driverList[i] =static_cast<char *>(malloc(32));
-
-        /* check memory */
-        if(!driverList[i]) qDebug() << "ERROR: Could not allocate Memory!";
-    }
-
-
-    loadAsioDriver("dummy");
-
-    drvNum = asioDrivers->getDriverNames(driverList,32);
-
-}
-
-PlayerAsio::~PlayerAsio()
-{
-
-    ASIOStop();
-    ASIOExit();
-
-}
-//////////////////////////////////////////////////////////////////////
-// Class functions
-//////////////////////////////////////////////////////////////////////
-
-/*******
- *
- * Conversion of different Ouputtypes
- *
- ********/
-const double fScaler24 = (double)0xffff;
-
-void PlayerAsio::float32toInt24inPlace(float * buffer, long frames, bool swap)
-{
-    double sc = fScaler24 + .5;
-    long a;
-    char * b = (char *)(buffer);
-    char * aa = (char *)(&a);
-
-    while(--frames >= 0)
-    {
-        a = static_cast<long>(static_cast<double>(*buffer++) *sc);
-
-
-        if (swap)
-        {
-            *b++ = aa[3];
-            *b++ = aa[2];
-            *b++ = aa[1];
-        }
-        else
-        {
-            *b++ = aa[1];
-            *b++ = aa[2];
-            *b++ = aa[3];
-        }
-    }
-}
-
-
-void PlayerAsio::Output_Float32_Int24(int channelNumber, float * inputBuffer, float * outputBuffer, bool swap)
-{
-    for(int i=0; i < bufferLength; i++)
-    {
-        m_pTempBuffer[i] = inputBuffer[i*sizeof(float) + channelNumber];
-    }
-
-    float32toInt24inPlace(m_pTempBuffer,bufferLength,swap);
-    memcpy(outputBuffer,m_pTempBuffer,bufferLength*3);
-
-}
-
-void PlayerAsio::Output_Float32_Int32(int channelNumber, float * inputBuffer, float * outputBuffer, bool swap)
-{
-    for(int i=0; i < bufferLength; i++)
-    {
-        m_pTempBuffer[i] = inputBuffer[i*sizeof(float) + channelNumber];
-    }
-
-    float32toInt32inPlace(static_cast<float *>(m_pTempBuffer),bufferLength,swap);
-    memcpy(outputBuffer,m_pTempBuffer,bufferLength*4);
-
-}
-
-const double fScaler32 = (double)0xffff;
-
-void PlayerAsio::float32toInt32inPlace(float * buffer, long frames,bool swap)
-{
-    double sc = fScaler32 + .49999;
-    long a;
-    char * b = (char *)(buffer);
-    char * aa = (char *)(&a);
-
-    while(--frames >= 0)
-    {
-        a = static_cast<long>(static_cast<double>(*buffer++) * sc);
-
-
-        if(swap){
-            *b++ = aa[3];
-            *b++ = aa[2];
-            *b++ = aa[1];
-            *b++ = aa[0];
-        }
-        else
-        {
-            *b++ = aa[0];
-            *b++ = aa[1];
-            *b++ = aa[2];
-            *b++ = aa[3];
-        }
-    }
-}
-
-void PlayerAsio::Output_Float32_Int16(int channelNumber, float * inputBuffer, short * outputBuffer, bool swap)
-{
-    short temp;
-    for (int i=0; i < bufferLength; i++)
-    {
-        temp = static_cast<short>(inputBuffer[i*sizeof(float) + channelNumber]);
-        outputBuffer[i] = static_cast<short>(temp);
-        if (swap) SwapBuffer(&outputBuffer[i]);
-    }
-}
-
-void PlayerAsio::SwapBuffer(short * buffer)
-{
-    short a;
-
-    char * b = (char *)(buffer);
-    char * aa = (char *)(&a);
-
-    a = *buffer;
-
-    *b++ = aa[1];
-    *b++ = aa[0];
-
-}
-/*******
- *
- * Init
- *
- ********/
-
-bool PlayerAsio::initialize() {
-    return true;
-}
-
-bool PlayerAsio::initDriver()
-{
-
-
-    if (ASIOInit (&asioDriverInfo.driverInfo) == ASE_OK ){
-
-
-        return true;
-    }
-    return false;
-
-
-}
-
-
-int PlayerAsio::getBufferSize(void)
-{
-    return asioDriverInfo.preferredSize;
-}
-
-
-int PlayerAsio::getChannelCount(void)
-{
-    return asioDriverInfo.outputChannels;
-}
-
-
-int PlayerAsio::getSampleRate(void)
-{
-    return asioDriverInfo.sampleRate;
-}
-
-
-QStringList PlayerAsio::getSampleRates() {
-    QStringList result;
-    result.append("44100");
-    return result;
-}
-
-
-QStringList PlayerAsio::getInterfaces() {
-    QStringList result;
-    LPASIODRVSTRUCT lpdrv = asioDrivers->lpdrvlist;
-    while(lpdrv) {
-        result.append(lpdrv->drvname);
-        lpdrv = lpdrv->next;
-    }
-    return result;
-}
-
-long PlayerAsio::init_asio_static_data (DriverInfo * asioDriverInfo)
-{
-
-    if(ASIOGetChannels(&asioDriverInfo->inputChannels, &asioDriverInfo->outputChannels) == ASE_OK)
-    {
-
-
-        // get the usable buffer sizes
-        if(ASIOGetBufferSize(&asioDriverInfo->minSize, &asioDriverInfo->maxSize, &asioDriverInfo->preferredSize, &asioDriverInfo->granularity) == ASE_OK)
-        {
-
-            // get the currently selected sample rate
-            if(ASIOGetSampleRate(&asioDriverInfo->sampleRate) == ASE_OK)
-            {
-                //Get SampleRate From Preferences
-                int sampleRate = m_pConfig->getValueString(ConfigKey("[Soundcard]", "Samplerate")).toInt();
-                //Calculate Latency
-                int iLatencyMSec = (1000*asioDriverInfo->preferredSize*4)/(sampleRate);
-
-                if (asioDriverInfo->sampleRate <= 0.0 || asioDriverInfo->sampleRate > 96000.0)
-                {
-
-                    if(ASIOSetSampleRate(sampleRate) == ASE_OK)
-                    {
-                        if(ASIOGetSampleRate(&asioDriverInfo->sampleRate) == ASE_OK)
-                            qDebug() << "Asio State : SAMPLERATE CHANGED";
-                        else
-                            return -6;
-                    }
-                    else
-                        return -5;
-                } else if(asioDriverInfo->sampleRate != m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt())
-                {
-
-                    if(ASIOSetSampleRate(sampleRate) == ASE_OK)
-                    {
-                        if(ASIOGetSampleRate(&asioDriverInfo->sampleRate) == ASE_OK)
-                            qDebug() << "Asio State : SAMPLERATE CHANGED";
-                        else
-                            return -6;
-                    }
-                    else
-                        return -5;
-                }
-
-
-                // Driver does not store it's internal sample rate, so set it to a know one.
-                // Usually you should check beforehand, that the selected sample rate is valid
-                // with ASIOCanSampleRate().
-                // Update SRATE and Latency ControlObjects
-                m_pControlObjectSampleRate->queueFromThread(static_cast<double>(sampleRate));
-                m_pControlObjectLatency->queueFromThread(static_cast<double>(iLatencyMSec));
-                m_pConfig->set(ConfigKey("[Soundcard]","Latency"), iLatencyMSec);
-                qDebug() << "Asio State: SAMPLERATE SET";
-
-                // check wether the driver requires the ASIOOutputReady() optimization
-                // (can be used by the driver to reduce output latency by one block)
-                if(ASIOOutputReady() == ASE_OK)
-                    asioDriverInfo->postOutput = true;
-                else
-                    asioDriverInfo->postOutput = false;
-
-
-                return 0;
-            }
-            return -3;
-        }
-        return -2;
-    }
-    return -1;
-}
-
-ASIOError PlayerAsio::create_asio_buffers (DriverInfo * asioDriverInfo)
-{
-
-    long i=0;
-    ASIOError result;
-
-    bufferLength = asioDriverInfo->preferredSize;
-    m_pTempBuffer = static_cast<float *>(malloc(bufferLength*4));
-
-    // fill the bufferInfos from the start without a gap
-    ASIOBufferInfo * info = asioDriverInfo->bufferInfos;
-
-    asioDriverInfo->inputBuffers = 0;
-
-    // prepare inputs (Though this is not necessaily required, no opened inputs will work, too
-    if (asioDriverInfo->inputChannels > kMaxInputChannels)
-        asioDriverInfo->inputBuffers = kMaxInputChannels;
-    else
-        asioDriverInfo->inputBuffers = asioDriverInfo->inputChannels;
-
-    for(i = 0; i < asioDriverInfo->inputBuffers; i++, info++)
-    {
-        info->isInput = ASIOTrue;
-        info->channelNum = i;
-        info->buffers[0] = info->buffers[1] = 0;
-    }
-    // prepare outputs
-    if (asioDriverInfo->outputChannels > kMaxOutputChannels)
-        asioDriverInfo->outputBuffers = kMaxOutputChannels;
-    else
-        asioDriverInfo->outputBuffers = asioDriverInfo->outputChannels;
-
-    qDebug(QString::number(asioDriverInfo->outputChannels));
-
-    // prepare outputs
-    if (asioDriverInfo->outputChannels > kMaxOutputChannels)
-        asioDriverInfo->outputBuffers = kMaxOutputChannels;
-    else
-        asioDriverInfo->outputBuffers = asioDriverInfo->outputChannels;
-    for(i = 0; i < asioDriverInfo->outputBuffers; i++, info++)
-    {
-        info->isInput = ASIOFalse;
-        info->channelNum = i;
-        info->buffers[0] = info->buffers[1] = 0;
-    }
-
-    // create and activate buffers
-    result = ASIOCreateBuffers(asioDriverInfo->bufferInfos,
-                               asioDriverInfo->outputBuffers+asioDriverInfo->inputBuffers,
-                               asioDriverInfo->preferredSize, &asioCallbacks);
-
-    if (result == ASE_OK)
-    {
-        // now get all the buffer details, sample word length, name, word clock group and activation
-        for (i = 0; i < asioDriverInfo->inputBuffers + asioDriverInfo->outputBuffers; i++)
-        {
-            asioDriverInfo->channelInfos[i].channel = asioDriverInfo->bufferInfos[i].channelNum;
-            asioDriverInfo->channelInfos[i].isInput = asioDriverInfo->bufferInfos[i].isInput;
-
-            result = ASIOGetChannelInfo(&asioDriverInfo->channelInfos[i]);
-
-            if (result != ASE_OK){
-                qDebug() << "ERROR: Buffer init failed !";
-                break;
-            }
-        }
-
-        if (result == ASE_OK)
-        {
-            // get the input and output latencies
-            // Latencies often are only valid after ASIOCreateBuffers()
-            // (input latency is the age of the first sample in the currently returned audio block)
-            // (output latency is the time the first sample in the currently returned audio block requires to get to the output)
-            result = ASIOGetLatencies(&asioDriverInfo->inputLatency, &asioDriverInfo->outputLatency);
-            //if (result == ASE_OK)
-            //emit(initBuffers(asioDriverInfo->preferredSize));
-        }
-    }
-    return result;
-}
-
-/*******
- *
- * Open / Close Calls
- *
- ********/
-bool PlayerAsio::open()
-{
-    if (driverIsLoaded)
-        close();
-
-    QString name;
-
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
-
-    char driverName[MAX_DRIVER_NAME_LENGTH];
-    strcpy(driverName, name.latin1());
-
-    if (loadAsioDriver(driverName))
-    {
-        qDebug() << "Asio State: DRIVER LOADED";
-        if(initDriver()){
-            qDebug() << "Asio State: DRIVER INITIALIZED";
-            init_asio_static_data (&asioDriverInfo);
-            asioCallbacks.bufferSwitch = bufferSwitch;
-            asioCallbacks.sampleRateDidChange = sampleRateChanged;
-            asioCallbacks.asioMessage = asioMessages;
-            asioCallbacks.bufferSwitchTimeInfo = bufferSwitchTimeInfo;
-            create_asio_buffers(&asioDriverInfo);
-            startAsio();
-            return true;
-        }
-    }
-    return false;
-}
-
-void PlayerAsio::startAsio()
-{
-    if (ASIOStart() == ASE_OK)
-    {
-        driverIsLoaded = true;
-        qDebug() << "Asio streaming started";
-    }
-}
-
-void PlayerAsio::close()
-{
-    if(ASIOStop() == ASE_OK)
-    {
-        driverIsLoaded = false;
-        qDebug() << "Asio streaming stopped";
-    }
-
-    if(ASIOExit() == ASE_OK)
-    {
-        qDebug() << "Asio exited";
-    }
-}
-
-void PlayerAsio::setDefaults()
-{
-    // Get list of interfaces
-    QStringList interfaces = getInterfaces();
-
-    // Set first interfaces to master left
-    QStringList::iterator it = interfaces.begin();
-    if (it!=interfaces.end())
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((*it)));
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((*it)));
-    }
-    else
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-    }
-
-    // Set head left and right to none
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-
-}
-
-/*******
- *
- * Asio and mixxx Callbacks
- *
- ********/
-
-//----------------------------------------------------------------------------------
-void bufferSwitch(long index, ASIOBool processNow)
-{   // the actual processing callback.
-
-    ASIOTime timeInfo;
-    memset (&timeInfo, 0, sizeof (timeInfo));
-
-
-    bufferSwitchTimeInfo (&timeInfo, index, processNow);
-
-}
-
-
-//----------------------------------------------------------------------------------
-void sampleRateChanged(ASIOSampleRate sRate)
-{
-    // do whatever you need to do if the sample rate changed
-    // usually this only happens during external sync.
-    // Audio processing is not stopped by the driver, actual sample rate
-    // might not have even changed, maybe only the sample rate status of an
-    // AES/EBU or S/PDIF digital input at the audio device.
-    // You might have to update time/sample related conversion routines, etc.
-}
-
-//----------------------------------------------------------------------------------
-long asioMessages(long selector, long value, void * message, double * opt)
-{
-    // currently the parameters "value", "message" and "opt" are not used.
-    long ret = 0;
-    switch(selector)
-    {
-    case kAsioSelectorSupported:
-        if(value == kAsioResetRequest
-           || value == kAsioEngineVersion
-           || value == kAsioResyncRequest
-           || value == kAsioLatenciesChanged
-           // the following three were added for ASIO 2.0, you don't necessarily have to support them
-           || value == kAsioSupportsTimeInfo
-           || value == kAsioSupportsTimeCode
-           || value == kAsioSupportsInputMonitor)
-            ret = 1L;
-        break;
-    case kAsioResetRequest:
-        // defer the task and perform the reset of the driver during the next "safe" situation
-        // You cannot reset the driver right now, as this code is called from the driver.
-        // Reset the driver is done by completely destruct is. I.e. ASIOStop(), ASIODisposeBuffers(), Destruction
-        // Afterwards you initialize the driver again.
-        //asioDriverInfo.stopped;  // In this sample the processing will just stop
-        ret = 1L;
-        break;
-    case kAsioResyncRequest:
-        // This informs the application, that the driver encountered some non fatal data loss.
-        // It is used for synchronization purposes of different media.
-        // Added mainly to work around the Win16Mutex problems in Windows 95/98 with the
-        // Windows Multimedia system, which could loose data because the Mutex was hold too long
-        // by another thread.
-        // However a driver can issue it in other situations, too.
-        ret = 1L;
-        break;
-    case kAsioLatenciesChanged:
-        // This will inform the host application that the drivers were latencies changed.
-        // Beware, it this does not mean that the buffer sizes have changed!
-        // You might need to update internal delay data.
-        ret = 1L;
-        break;
-    case kAsioEngineVersion:
-        // return the supported ASIO version of the host application
-        // If a host applications does not implement this selector, ASIO 1.0 is assumed
-        // by the driver
-        ret = 2L;
-        break;
-    case kAsioSupportsTimeInfo:
-        // informs the driver wether the asioCallbacks.bufferSwitchTimeInfo() callback
-        // is supported.
-        // For compatibility with ASIO 1.0 drivers the host application should always support
-        // the "old" bufferSwitch method, too.
-        ret = 1;
-        break;
-    case kAsioSupportsTimeCode:
-        // informs the driver wether application is interested in time code info.
-        // If an application does not need to know about time code, the driver has less work
-        // to do.
-        ret = 0;
-        break;
-    }
-    return ret;
-}
-
-void PlayerAsio::processCallback(long bufferIndex) {
-
-    float * inputBuffer = prepareBuffer(bufferLength);
-    // perform the processing
-    for (int i = 0; i < asioDriverInfo.inputBuffers + asioDriverInfo.outputBuffers; i++)
-    {
-        if (asioDriverInfo.bufferInfos[i].isInput == false)
-        {
-            ASIOBufferInfo * bufferInfo = &asioDriverInfo.bufferInfos[i];
-
-            m_reenter++;
-            if (m_reenter > 1) {
-                qDebug() << "ASIO : reentered callback ???!!!";
-                return;
-            }
-
-            if (!driverIsLoaded) {
-                qDebug() << "ASIO : callback and not started ???!!!";
-            }
-
-
-            bool swap = true;
-
-
-            void * outputBuffer = bufferInfo->buffers[bufferIndex];
-            ASIOSampleType bufferType = asioDriverInfo.channelInfos[i].type;
-            switch (bufferType) {
-            case ASIOSTInt16LSB:
-                Output_Float32_Int16(i, inputBuffer, static_cast<short *>(outputBuffer), !swap);
-                break;
-            case ASIOSTInt16MSB:
-                Output_Float32_Int16(i, inputBuffer, static_cast<short *>(outputBuffer), swap);
-                break;
-            case ASIOSTInt32LSB:
-                Output_Float32_Int32(i, inputBuffer, static_cast<float *>(outputBuffer), !swap);
-                break;
-            case ASIOSTInt32MSB:
-                Output_Float32_Int32(i, inputBuffer, static_cast<float *>(outputBuffer), swap);
-                break;
-            case ASIOSTFloat32LSB:
-                //Output_Float32_Float32(i, inputBuffer, static_cast<float*>(outputBuffer), swap);
-                break;
-            case ASIOSTFloat32MSB:
-                //Output_Float32_Float32(i, inputBuffer, static_cast<float*>(outputBuffer), !swap);
-                memset(static_cast<float *>(outputBuffer),0,bufferLength*4);
-                break;
-            case ASIOSTInt24LSB:        // used for 20 bits as well
-                Output_Float32_Int24(i, inputBuffer,static_cast<float *>(outputBuffer), !swap);
-                //memset(static_cast<float*>(outputBuffer),0,bufferLength*4);
-                break;
-            case ASIOSTInt24MSB:        // used for 20 bits as well
-            case ASIOSTFloat64LSB:      // IEEE 754 64 bit double float, as found on Intel x86 architecture
-            case ASIOSTFloat64MSB:      // IEEE 754 64 bit double float, as found on Intel x86 architecture
-
-                // these are used for 32 bit data buffer, with different alignment of the data inside
-                // 32 bit PCI bus systems can more easily used with these
-
-            case ASIOSTInt32LSB16:      // 32 bit data with 16 bit alignment
-            case ASIOSTInt32LSB18:      // 32 bit data with 18 bit alignment
-            case ASIOSTInt32LSB20:      // 32 bit data with 20 bit alignment
-            case ASIOSTInt32LSB24:      // 32 bit data with 24 bit alignment
-
-
-            case ASIOSTInt32MSB16:      // 32 bit data with 16 bit alignment
-            case ASIOSTInt32MSB18:      // 32 bit data with 18 bit alignment
-            case ASIOSTInt32MSB20:      // 32 bit data with 20 bit alignment
-            case ASIOSTInt32MSB24:      // 32 bit data with 24 bit alignment
-                // not implemented !!
-
-                break;
-
-
-            }
-
-            m_reenter--;
-        }
-    }
-    if (asioDriverInfo.postOutput)
-        ASIOOutputReady();
-}
-
-ASIOTime * bufferSwitchTimeInfo(ASIOTime * timeInfo, long index, ASIOBool processNow)
-{
-    ((PlayerAsio *)(PlayerProxy::getPlayer()))->processCallback(index);
-
-    return 0L;
-}
diff -Nru mixxx-1.6.0/src/playerasio.h mixxx-1.6.1/src/playerasio.h
--- mixxx-1.6.0/src/playerasio.h	2007-02-03 03:06:06.000000000 -0600
+++ mixxx-1.6.1/src/playerasio.h	1969-12-31 18:00:00.000000000 -0600
@@ -1,133 +0,0 @@
-// PlayerAsio.h: interface for the PlayerAsio class.
-//
-//////////////////////////////////////////////////////////////////////
-
-#include <qobject.h>
-#include <qmutex.h>
-#include <qsignal.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include "player.h"
-#include "asiosys.h"
-#include "asio.h"
-#include "asiodrivers.h"
-
-
-#if !defined(PlayerAsio_H)
-#define PlayerAsio_H
-
-void __cdecl bufferSwitch(long index, ASIOBool processNow);
-long __cdecl asioMessages(long selector, long value, void* message, double* opt);
-void __cdecl sampleRateChanged(ASIOSampleRate sRate);
-ASIOTime* __cdecl bufferSwitchTimeInfo(ASIOTime *timeInfo, long index, ASIOBool processNow);
-
-enum {
-    // number of input and outputs supported by the host application
-    // you can change these to higher or lower values
-    kMaxInputChannels = 0,
-    kMaxOutputChannels = 4
-};
-
-
-// internal data storage
-typedef struct DriverInfo
-{
-    
-    ASIODriverInfo driverInfo;
-    
-    
-    long           inputChannels;
-    long           outputChannels;
-    
-    
-    long           minSize;
-    long           maxSize;
-    long           preferredSize;
-    long           granularity;
-    
-    
-    ASIOSampleRate sampleRate;
-    
-    
-    bool           postOutput;
-    
-    
-    long           inputLatency;
-    long           outputLatency;
-    
-   
-    long inputBuffers;  // becomes number of actual created input buffers
-    long outputBuffers; // becomes number of actual created output buffers
-    ASIOBufferInfo bufferInfos[kMaxInputChannels + kMaxOutputChannels]; // buffer info's
-    
-    // ASIOGetChannelInfo()
-    ASIOChannelInfo channelInfos[kMaxInputChannels + kMaxOutputChannels]; // channel info's
-    
-    
-    double         nanoSeconds;
-    double         samples;
-    double         tcSamples;   // time code samples
-    
-    
-    ASIOTime       tInfo;           // time info state
-    unsigned long  sysRefTime;      // system reference time, when bufferSwitch() was called
-    
-  
-} DriverInfo;
-
-
-class PlayerAsio : public Player 
-{
-    
-public:
-    
-    
-    PlayerAsio(ConfigObject<ConfigValue> *config);
-    virtual ~PlayerAsio();
-    
-    bool initialize();
-    bool open();
-    void close();
-    void startAsio();
-    
-    int getBufferSize(void);
-    int getSampleRate(void);
-    int getChannelCount(void);
-    
-    QStringList getInterfaces();
-    QStringList getSampleRates();
-    
-    void setDefaults();
-    
-    static QString getSoundApi() { return "ASIO"; }
-    QString getSoundApiName() { return getSoundApi(); };
-    
-    void process(const CSAMPLE *, const CSAMPLE *, const int) { };
-    void processCallback(long bufferIndex);
-    
-    
-private:
-    bool initDriver(void);
-
-    void Output_Float32_Int16(int channelNumber, float* inputBuffer, short* outputBuffer, bool swap); 
-    void Output_Float32_Int32(int channelNumber, float* inputBuffer, float* outputBuffer, bool swap); 
-    void Output_Float32_Int24(int channelNumber, float* inputBuffer, float* outputBuffer, bool swap); 
-    void float32toInt24inPlace(float* buffer, long frames, bool swap);
-    void float32toInt32inPlace(float* buffer, long frames, bool swap);
-    void SwapBuffer(short* buffer);
-    
-    ASIOError create_asio_buffers (DriverInfo *asioDriverInfo);
-    long init_asio_static_data (DriverInfo *asioDriverInfo);
-    
-    AsioDrivers* driver;
-    int m_reenter;
-    bool driverIsLoaded;
-    int drvNum;
-    int bufferLength;
-    char* driverList[32];
-    /** Used in sample convertion */
-    float *m_pTempBuffer;
-     
-};
-
-#endif 
diff -Nru mixxx-1.6.0/src/player.cpp mixxx-1.6.1/src/player.cpp
--- mixxx-1.6.0/src/player.cpp	2008-03-27 00:04:43.000000000 -0500
+++ mixxx-1.6.1/src/player.cpp	1969-12-31 18:00:00.000000000 -0600
@@ -1,151 +0,0 @@
-/***************************************************************************
-                          player.cpp  -  description
-                             -------------------
-    begin                : Wed Feb 20 2002
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-    email                :
-***************************************************************************/
-
-/***************************************************************************
-*                                                                         *
-*   This program is free software; you can redistribute it and/or modify  *
-*   it under the terms of the GNU General Public License as published by  *
-*   the Free Software Foundation; either version 2 of the License, or     *
-*   (at your option) any later version.                                   *
-*                                                                         *
-***************************************************************************/
-
-#include "player.h"
-#include "enginemaster.h"
-#include "controlobject.h"
-
-#ifdef RECORD_OUTPUT
-// HACK TO RECORD OUTPUT. WRITTEN TO FILE AT PROGRAM EXIT
-  #include <audiofile.h>
-  #include <q3ptrlist.h>
-typedef struct {
-    short int * pBuffer;
-    int size;
-} recordObject;
-Q3PtrList<recordObject> m_qRecordList;
-#endif
-
-//Static variable memory allocation
-short int Player::m_iBufferSize = 0;
-short int Player::m_iChannels[MAX_AUDIODEVICES];
-EngineMaster * Player::m_pMaster = 0;
-
-
-
-/* -------- ------------------------------------------------------
-   Purpose: Initializes the audio hardware.
-   Input:   Size of the output buffer in samples
-   Output:  Pointer to internal synthesis data structure.
-   -------- ------------------------------------------------------ */
-Player::Player(ConfigObject<ConfigValue> * pConfig)
-{
-    m_pConfig = pConfig;
-    m_pBuffer = new CSAMPLE[MAX_BUFFER_LEN];
-    m_pControlObjectSampleRate = ControlObject::getControl(ConfigKey("[Master]","samplerate"));
-    m_pControlObjectLatency = new ControlObject(ConfigKey("[Master]","latency"));;
-
-    for (int i = 0; i < MAX_AUDIODEVICES; i++)
-    {
-        m_iChannels[i] = 0;
-    }
-}
-
-/* -------- ------------------------------------------------------
-   Purpose: Terminate and deallocate the synthesis system
-   Input:   -
-   Output:  -
-   -------- ------------------------------------------------------ */
-Player::~Player()
-{
-    delete [] m_pBuffer;
-
-#ifdef RECORD_OUTPUT
-    //
-    // HACK: Write recorded sound
-    //
-
-    qDebug() << "Writing output to file... please wait!";
-
-    // Setup file format
-    AFfilesetup outputSetup = afNewFileSetup();
-    afInitFileFormat(outputSetup, AF_FILE_WAVE);
-    afInitRate(outputSetup, AF_DEFAULT_TRACK, m_pControlObjectSampleRate->get());
-    afInitChannels(outputSetup, AF_DEFAULT_TRACK, 2);
-    afInitSampleFormat (outputSetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
-    //afInitByteOrder (outputSetup, AF_DEFAULT_TRACK, AF_BYTEORDER_BIGENDIAN);
-
-    // Open file handle
-    AFfilehandle fh = afOpenFile("output.wav","w",outputSetup);
-    if (!fh)
-        qCritical("Could not write wave output file");
-    recordObject * r;
-    for (r = m_qRecordList.first(); r; r=m_qRecordList.next())
-    {
-        afWriteFrames(fh,AF_DEFAULT_TRACK,r->pBuffer,r->size/2);
-    }
-    afCloseFile(fh);
-#endif
-}
-
-/*
-   short int Player::getBufferSize()
-   {
-   //     if (m_iChannels>0)
-   //         return m_iBufferSize/m_iChannels;
-   //     else
-        return m_iBufferSize;
-   }
- */
-
-void Player::setMaster(EngineMaster * pMaster)
-{
-    m_pMaster = pMaster;
-}
-
-bool Player::open()
-{
-    if (m_pConfig->getValueString(ConfigKey("[Soundcard]","PitchIndpTimeStretch")).toInt())
-        qDebug() << "pitch true";
-
-    // Set sound scale method
-    if (m_pMaster)
-        m_pMaster->setPitchIndpTimeStretch(m_pConfig->getValueString(ConfigKey("[Soundcard]","PitchIndpTimeStretch")).toInt());
-
-    return true;
-}
-
-/* -------- ------------------------------------------------------
-   Purpose: Internal callback function used for preparing samples
-            for playback. This is where the synthesis is done.
-   Input:   Number of samples for each channel (4 channels in all)
-   Output:  pointer to resulting buffer of samples
-   -------- ------------------------------------------------------ */
-CSAMPLE * Player::prepareBuffer(int iBufferSize)
-{
-    // First, sync control parameters with changes from GUI thread
-    ControlObject::sync();
-
-    // Process a block of samples for output. iBufferSize is the
-    // number of samples for one channel, but the EngineObject
-    // architecture expects number of samples for two channels
-    // as input so...
-    m_pMaster->process(0, m_pBuffer, iBufferSize*2);
-
-#ifdef RECORD_OUTPUT
-    // HACK: RECORD SOUND
-    recordObject * r = new recordObject;
-    m_qRecordList.append(r);
-    r->size = iBufferSize*2;
-    r->pBuffer = new short int[r->size];
-    for (int j=0; j<r->size; ++j)
-        r->pBuffer[j] = (short int)(m_pBuffer[j]);
-#endif
-
-    return m_pBuffer;
-}
-
diff -Nru mixxx-1.6.0/src/player.h mixxx-1.6.1/src/player.h
--- mixxx-1.6.0/src/player.h	2007-08-21 23:16:40.000000000 -0500
+++ mixxx-1.6.1/src/player.h	1969-12-31 18:00:00.000000000 -0600
@@ -1,79 +0,0 @@
-/***************************************************************************
-                          player.h  -  description
-                             -------------------
-    begin                : Wed Feb 20 2002
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-    email                : 
- ***************************************************************************/
-
-/***************************************************************************
- *                                                                         *
- *   This program is free software; you can redistribute it and/or modify  *
- *   it under the terms of the GNU General Public License as published by  *
- *   the Free Software Foundation; either version 2 of the License, or     *
- *   (at your option) any later version.                                   *
- *                                                                         *
- ***************************************************************************/
-
-#ifndef PLAYER_H
-#define PLAYER_H
-
-#include "configobject.h"
-#include <qstring.h>
-#include "engineobject.h"
-#include <qstringlist.h>
-
-#define MAX_AUDIODEVICES 2
-
-class EngineMaster;
-class ControlObject;
-
-class Player : public EngineObject
-{
-public:
-    Player(ConfigObject<ConfigValue> *pConfig);
-    virtual ~Player();
-    /** Initialize the API. Returns true on success. No other methods in a Player
-      * class can be called before this method returns true. Called by the proxy
-      * class. */
-    virtual bool initialize() = 0;
-    /** Returns the current buffer size (total latency) used by the playback system */
-    //static short int getBufferSize();
-    /** Set EngineMaster object */
-    static void setMaster(EngineMaster *pMaster);
-    /** Open devices according to config database, and start audio stream.
-      * Returns true on success. */
-    virtual bool open();
-    /** Close devices, and stop audio stream */
-    virtual void close() = 0;
-    /** Store default configuration in config database */
-    virtual void setDefaults() = 0;
-    /** Return list of interfaces */
-    virtual QStringList getInterfaces() = 0;
-    /** Return list of input interfaces */
-    virtual QStringList getInputInterfaces() = 0;    
-    /** Return list of sample rates */
-    virtual QStringList getSampleRates() = 0;
-    /** Return name of sound api */
-    virtual QString getSoundApiName() = 0;
-
-protected:
-    /** Prepares one buffer of sound by calling the engine */
-    CSAMPLE *prepareBuffer(int iBufferSize);
-
-    /** Pointer to config database */
-    ConfigObject<ConfigValue> *m_pConfig;
-    /** Pointer to EngineMaster object */
-    static EngineMaster *m_pMaster;
-    CSAMPLE *m_pBuffer;
-    /** Current buffer size in use. */
-    static short int m_iBufferSize;
-    /** Current number of channels in use */
-    static short int m_iChannels[MAX_AUDIODEVICES];
-    /** Pointer to object holding sample rate */
-    ControlObject *m_pControlObjectSampleRate;
-    /** Pointer to object holding latency */
-    ControlObject *m_pControlObjectLatency;
-};
-
-#endif
diff -Nru mixxx-1.6.0/src/playerjack.cpp mixxx-1.6.1/src/playerjack.cpp
--- mixxx-1.6.0/src/playerjack.cpp	2008-03-27 00:04:43.000000000 -0500
+++ mixxx-1.6.1/src/playerjack.cpp	1969-12-31 18:00:00.000000000 -0600
@@ -1,370 +0,0 @@
-/***************************************************************************
-                          playerjack.cpp  -  description
-                             -------------------
-    begin                : Sat Nov 15 2003
-    copyright            : (C) 2003 by Tue Haste Andersen
-    email                :
-***************************************************************************/
-
-/***************************************************************************
-*                                                                         *
-*   This program is free software; you can redistribute it and/or modify  *
-*   it under the terms of the GNU General Public License as published by  *
-*   the Free Software Foundation; either version 2 of the License, or     *
-*   (at your option) any later version.                                   *
-*                                                                         *
-***************************************************************************/
-
-#include "playerjack.h"
-#include "controlobject.h"
-#include <qmessagebox.h>
-#include <QtDebug>
-#include <qapplication.h>
-#include "controlobject.h"
-
-PlayerJack::PlayerJack(ConfigObject<ConfigValue> * config) : Player(config) //, mLibJack("libjack.so")
-{
-    ports = 0;
-    m_bOpen = false;
-
-/*
-    //mLibJack = new QLibrary("jack");
-    if (!mLibJack.load())
-        qDebug() << "JACK lib path" << mLibJack.library();
-
-    //if (mLibJack.isLoaded())
-    {
-        jack_set_error_function = (jack_set_error_function_t) mLibJack.resolve("jack_set_error_function");
-        jack_port_unregister = (jack_port_unregister_t) mLibJack.resolve("jack_port_unregister");
-        jack_client_close = (jack_client_close_t) mLibJack.resolve("jack_client_close");
-        jack_client_new = (jack_client_new_t) mLibJack.resolve("jack_client_new");
-        jack_set_process_callback = (jack_set_process_callback_t) mLibJack.resolve("jack_set_process_callback");
-        jack_set_sample_rate_callback = (jack_set_sample_rate_callback_t) mLibJack.resolve("jack_set_sample_rate_callback");
-        jack_on_shutdown =(jack_on_shutdown_t) mLibJack.resolve("jack_on_shutdown");
-        jack_port_register =(jack_port_register_t) mLibJack.resolve("jack_port_register");
-        jack_activate =(jack_activate_t) mLibJack.resolve("jack_activate");
-        jack_connect =(jack_connect_t) mLibJack.resolve("jack_connect");
-        jack_port_name =(jack_port_name_t) mLibJack.resolve("jack_port_name");
-        jack_deactivate =(jack_deactivate_t) mLibJack.resolve("jack_deactivate");
-        jack_get_ports =(jack_get_ports_t) mLibJack.resolve("jack_get_ports");
-        jack_get_sample_rate =(jack_get_sample_rate_t) mLibJack.resolve("jack_get_sample_rate");
-        jack_port_get_buffer =(jack_port_get_buffer_t) mLibJack.resolve("jack_port_get_buffer");
-    }
- */
-}
-
-PlayerJack::~PlayerJack()
-{
-//    if (!mLibJack.isLoaded())
-//        return;
-
-    close();
-
-    if (client)
-    {
-        jack_port_unregister(client, output_master_left);
-        jack_port_unregister(client, output_master_right);
-        jack_port_unregister(client, output_head_left);
-        jack_port_unregister(client, output_head_right);
-
-        jack_client_close(client);
-    }
-
-    if (ports)
-        free(ports);
-}
-
-bool PlayerJack::initialize()
-{
-    // Verify that library is loaded, and all function pointers has been retreived
-    /*
-       if (!mLibJack.isLoaded())
-       {
-        qDebug() << "lib not loaded";
-        return false;
-       }
-       else
-       if (!jack_set_error_function |
-            !jack_port_unregister |
-            !jack_client_close |
-            !jack_client_new |
-            !jack_set_process_callback |
-            !jack_set_sample_rate_callback |
-            !jack_on_shutdown |
-            !jack_port_register |
-            !jack_activate |
-            !jack_connect |
-            !jack_port_name |
-            !jack_deactivate |
-            !jack_get_ports |
-            !jack_get_sample_rate |
-            !jack_port_get_buffer)
-       {
-        qDebug() << "API function pointer error";
-        return false;
-       }
-     */
-
-    jack_set_error_function(jackError);
-
-    if ((client = jack_client_new("Mixxx")) == 0)
-        return false;
-
-    jack_set_process_callback(client, jackProcess, this);
-    jack_set_sample_rate_callback(client, jackSrate, this);
-    jack_on_shutdown(client, jackShutdown, this);
-
-    output_master_left  = jack_port_register(client, "Master left", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
-    output_master_right = jack_port_register(client, "Master right", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
-    output_head_left    = jack_port_register(client, "Head left", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
-    output_head_right   = jack_port_register(client, "Head right", JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
-
-    return true;
-}
-
-bool PlayerJack::open()
-{
-    Player::open();
-
-    /* tell the JACK server that we are ready to roll */
-    if (jack_activate(client))
-    {
-        qDebug() << "Jack: Cannot activate client";
-        return false;
-    }
-
-    // Connect to the ports
-    QString name;
-    m_iChannels = 0;
-
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
-    if (name != "None")
-    {
-        qDebug() << "connecting" << name << "to master left";
-        if (jack_connect(client, jack_port_name(output_master_left), name.latin1()))
-            m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-        else
-        {
-            m_iBufferSize = jack_port_get_total_latency(client, output_master_left);
-            m_iChannels++;
-        }
-    }
-
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterRight"));
-    if (name != "None")
-    {
-        qDebug() << "connecting" << name << "to master right";
-        if (jack_connect(client, jack_port_name(output_master_right), name.latin1()))
-            m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-        else
-        {
-            m_iBufferSize = jack_port_get_total_latency(client, output_master_right);
-            m_iChannels++;
-        }
-    }
-
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadLeft"));
-    if (name != "None")
-    {
-        qDebug() << "connecting" << name << "to head left";
-        if (jack_connect(client, jack_port_name(output_head_left), name.latin1()))
-            m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-        else
-        {
-            m_iBufferSize = jack_port_get_total_latency(client, output_head_left);
-            m_iChannels++;
-        }
-    }
-
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadRight"));
-    if (name != "None")
-    {
-        qDebug() << "connecting" << name << "to head right";
-        if (jack_connect(client, jack_port_name(output_head_right), name.latin1()))
-            m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-        else
-        {
-            m_iBufferSize = jack_port_get_total_latency(client, output_head_right);
-            m_iChannels++;
-        }
-    }
-
-    qDebug() << "channels " << m_iChannels << ", latency " << m_iBufferSize;
-
-
-    // Update the config database with the used sample rate
-    QStringList srates = getSampleRates();
-    QStringList::iterator it = srates.begin();
-    if (*it)
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue((*it)));
-
-        // Set currently used latency in config database
-        int msec = (int)(1000.*(float)m_iBufferSize/((float)(*it).toInt()));
-        m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
-
-        // Update SRATE and Latency ControlObjects
-        m_pControlObjectSampleRate->queueFromThread((double)((*it).toInt()));
-        m_pControlObjectLatency->queueFromThread((double)msec);
-
-    }
-
-    m_bOpen = true;
-    return true;
-}
-
-void PlayerJack::close()
-{
-    // Deactivate jack and disconnect all ports
-    if (m_bOpen && client)
-        jack_deactivate(client);
-    m_bOpen = false;
-}
-
-void PlayerJack::setDefaults()
-{
-    // Get list of interfaces
-    QStringList interfaces = getInterfaces();
-
-    // Set first interfaces to master left
-    QStringList::iterator it = interfaces.begin();
-    if (*it)
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((*it)));
-    }
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-
-    // Set second interface to master right
-    if (*it) ++it;
-    if (*it)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((*it)));
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-
-    // Set head left and right to none
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-
-    // Set default sample rate
-    QStringList srates = getSampleRates();
-    it = srates.begin();
-    if (*it)
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue((*it)));
-
-        // Set currently used latency in config database
-        int msec = 1000*m_iBufferSize/(4*(*it).toInt());
-        m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
-    }
-}
-
-QStringList PlayerJack::getInterfaces()
-{
-    QStringList result;
-
-    if (!client && !initialize())
-    {
-        qDebug() << "Jack: Failed to reinitialize the connection to Jack";
-        return result;
-    }
-
-    if ((ports = jack_get_ports (client, 0, 0, JackPortIsPhysical|JackPortIsInput)) == 0)
-        qDebug() << "Jack: Cannot find any physical playback ports";
-    else
-    {
-        int i=0;
-        while (ports[i]!=0)
-        {
-            result.append(ports[i]);
-            ++i;
-        }
-    }
-
-    return result;
-}
-
-QStringList PlayerJack::getSampleRates()
-{
-    QStringList result;
-
-    if (client)
-        result.append(QString("%1").arg((int)jack_get_sample_rate(client)));
-
-    return result;
-}
-
-QString PlayerJack::getSoundApi()
-{
-    return QString("Jack");
-}
-
-int PlayerJack::callbackProcess(int iBufferSize)
-{
-//     qDebug() << "buffer size " << iBufferSize;
-    jack_default_audio_sample_t * out_ml = (jack_default_audio_sample_t *)jack_port_get_buffer(output_master_left, iBufferSize);
-    jack_default_audio_sample_t * out_mr = (jack_default_audio_sample_t *)jack_port_get_buffer(output_master_right, iBufferSize);
-    jack_default_audio_sample_t * out_hl = (jack_default_audio_sample_t *)jack_port_get_buffer(output_head_left, iBufferSize);
-    jack_default_audio_sample_t * out_hr = (jack_default_audio_sample_t *)jack_port_get_buffer(output_head_right, iBufferSize);
-
-    if (m_bOpen)
-    {
-        CSAMPLE * buffer = prepareBuffer(iBufferSize);
-
-        for (int i=0; i<iBufferSize; ++i)
-        {
-            *out_ml++=buffer[(i*4)  ]/32768.;
-            *out_mr++=buffer[(i*4)+1]/32768.;
-            *out_hl++=buffer[(i*4)+2]/32768.;
-            *out_hr++=buffer[(i*4)+3]/32768.;
-        }
-    }
-
-    return 0;
-}
-
-void PlayerJack::callbackSetSrate(int srate)
-{
-//    setPlaySrate(srate);
-    m_pControlObjectSampleRate->queueFromThread((double)srate);
-}
-
-void PlayerJack::callbackSetBufferSize(int iBufferSize)
-{
-    m_iBufferSize = iBufferSize;
-}
-
-void PlayerJack::callbackShutdown()
-{
-    m_pConfig->set(ConfigKey("[Soundcard]","SoundApi"), ConfigValue("None"));
-
-    qWarning("Jack connection was killed.\n\nThis could be due to a high CPU load. Try increasing the latency\nwhen starting the Jack sound server.");
-}
-
-void jackError(const char * desc)
-{
-    qDebug() << "Jack experienced an error:" << desc;
-}
-
-int jackProcess(jack_nframes_t nframes, void * arg)
-{
-    ((PlayerJack *)arg)->callbackProcess(nframes);
-    return 0;
-}
-
-int jackSrate(jack_nframes_t nframes, void * arg)
-{
-    // Update SRATE in EngineObject
-    ((PlayerJack *)arg)->callbackSetSrate(nframes);
-    return 0;
-}
-
-void jackShutdown(void * arg)
-{
-    ((PlayerJack *)arg)->callbackShutdown();
-}
-
-void jackBufferSize(jack_nframes_t nframes, void * arg)
-{
-    ((PlayerJack *)arg)->callbackSetBufferSize(nframes);
-}
diff -Nru mixxx-1.6.0/src/playerjack.h mixxx-1.6.1/src/playerjack.h
--- mixxx-1.6.0/src/playerjack.h	2005-03-18 17:01:48.000000000 -0600
+++ mixxx-1.6.1/src/playerjack.h	1969-12-31 18:00:00.000000000 -0600
@@ -1,120 +0,0 @@
-/***************************************************************************
-                          playerjack.h  -  description
-                             -------------------
-    begin                : Sat Nov 15 2003
-    copyright            : (C) 2003 by Tue Haste Andersen
-    email                : 
- ***************************************************************************/
-
-/***************************************************************************
- *                                                                         *
- *   This program is free software; you can redistribute it and/or modify  *
- *   it under the terms of the GNU General Public License as published by  *
- *   the Free Software Foundation; either version 2 of the License, or     *
- *   (at your option) any later version.                                   *
- *                                                                         *
- ***************************************************************************/
-
-#ifndef PLAYERJACK_H
-#define PLAYERJACK_H
-
-#include "player.h"
-#include <jack/jack.h>
-#include <qlibrary.h>
-
-/**
-  *@author Tue and Ken Haste Andersen
-  */
-
-class PlayerJack : public Player  
-{
-public:
-    PlayerJack(ConfigObject<ConfigValue> *config);
-    ~PlayerJack();
-    bool initialize();
-    bool open();
-    void close();
-    void setDefaults();
-    QStringList getInterfaces();
-    QStringList getSampleRates();
-    static QString getSoundApi();
-    QString getSoundApiName() { return getSoundApi(); };
-    /** Satisfy virtual declaration in EngineObject */
-    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
-    /** Process samples. Called from jack callback */
-    int callbackProcess(int iBufferSize);
-    /** Used to set sample rate from Jack callback */
-    void callbackSetSrate(int iSrate);
-    /** Used to set current buffer size */
-    void callbackSetBufferSize(int iBufferSize);
-    /** Used to reinitialize when shut down by Jack server */
-    void callbackShutdown();
-
-protected:
-    /** Pointer to QLibrary object */
-    //QLibrary mLibJack;
-    /** Pointer to client info */
-    jack_client_t *client;
-    /** Null terminated array of port names */
-    const char **ports;
-    /** Playback ports */
-    jack_port_t *output_master_left, *output_master_right, *output_head_left, *output_head_right;
-    /** True if devices are open */
-    bool m_bOpen;
-
-/*
-    typedef void (*jack_set_error_function_t)(void (*func)(const char *));
-    jack_set_error_function_t jack_set_error_function;
-
-    typedef int (*jack_port_unregister_t)(jack_client_t *, jack_port_t *);
-    jack_port_unregister_t jack_port_unregister;
-
-    typedef int (*jack_client_close_t)(jack_client_t *client);
-    jack_client_close_t jack_client_close;
-
-    typedef jack_client_t* (*jack_client_new_t)(const char *client_name);
-    jack_client_new_t jack_client_new;
-
-    typedef int (*jack_set_process_callback_t)(jack_client_t *client,JackProcessCallback process_callback,void *arg);
-    jack_set_process_callback_t jack_set_process_callback;
-
-    typedef int (*jack_set_sample_rate_callback_t)(jack_client_t *client,JackSampleRateCallback srate_callback,void *arg);
-    jack_set_sample_rate_callback_t jack_set_sample_rate_callback;
-
-    typedef void (*jack_on_shutdown_t)(jack_client_t *client, void (*function)(void *arg), void *arg);
-    jack_on_shutdown_t jack_on_shutdown;
-
-    typedef jack_port_t* (*jack_port_register_t)(jack_client_t *,const char *port_name,const char *port_type,unsigned long flags,unsigned long buffer_size);
-    jack_port_register_t jack_port_register;
-
-    typedef int (*jack_activate_t)(jack_client_t *client);
-    jack_activate_t jack_activate;
-
-    typedef int (*jack_connect_t)(jack_client_t *,const char *source_port,const char *destination_port);
-    jack_connect_t jack_connect;
-
-    typedef const char* (*jack_port_name_t)(const jack_port_t *port);
-    jack_port_name_t jack_port_name;
-
-    typedef int (*jack_deactivate_t)(jack_client_t *client);
-    jack_deactivate_t jack_deactivate;
-
-    typedef const char** (*jack_get_ports_t)(jack_client_t *,const char *port_name_pattern,const char *type_name_pattern,unsigned long flags);
-    jack_get_ports_t jack_get_ports;
-
-    typedef jack_nframes_t (*jack_get_sample_rate_t)(jack_client_t *);
-    jack_get_sample_rate_t jack_get_sample_rate;
-
-    typedef void* (*jack_port_get_buffer_t)(jack_port_t *, jack_nframes_t);
-    jack_port_get_buffer_t jack_port_get_buffer;
-*/
-};
-
-// Jack callbacks:
-void jackError(const char *desc);
-int jackProcess(jack_nframes_t nframes, void *arg);
-int jackSrate(jack_nframes_t nframes, void *arg);
-void jackShutdown(void *arg);
-void jackBufferSize(jack_nframes_t nframes, void *arg);
-
-#endif
diff -Nru mixxx-1.6.0/src/playerportaudio.cpp mixxx-1.6.1/src/playerportaudio.cpp
--- mixxx-1.6.0/src/playerportaudio.cpp	2008-03-27 00:04:43.000000000 -0500
+++ mixxx-1.6.1/src/playerportaudio.cpp	1969-12-31 18:00:00.000000000 -0600
@@ -1,938 +0,0 @@
-/***************************************************************************
-                          playerportaudiov19.cpp  -  description
-                             -------------------
-    begin                : Wed Feb 20 2002
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-                                           (C) 2006/2007 by Albert Santoni
-                                           (C) 2006 by Adam Davison
-    email                :
-***************************************************************************/
-
-/***************************************************************************
-*                                                                         *
-*   This program is free software; you can redistribute it and/or modify  *
-*   it under the terms of the GNU General Public License as published by  *
-*   the Free Software Foundation; either version 2 of the License, or     *
-*   (at your option) any later version.                                   *
-*                                                                         *
-***************************************************************************/
-
-#include "playerportaudio.h"
-#include "controlobject.h"
-
-// Mathstuff required for Win32
-#include "mathstuff.h"
-//Added by qt3to4:
-#include <Q3ValueList>
-#include <QtDebug>
-#include <QWaitCondition>
-
-bool PlayerPortAudio::m_painited = false;
-
-PlayerPortAudio::PlayerPortAudio(ConfigObject<ConfigValue> * config, QString api_name) : Player(config)
-{
-    for (int i = 0; i < MAX_AUDIODEVICES; i++)
-    {
-        m_devId[i]      = -1;
-        m_inputDevId[i] = -1;
-        m_pStream[i]    =  0;
-        m_iChannels[i]  = -1;
-    }
-
-    m_iNumberOfBuffers = 2;
-    m_iMasterLeftCh = -1;
-    m_iMasterRigthCh = -1;
-    m_iHeadLeftCh = -1;
-    m_iHeadRightCh = -1;
-    m_iNumActiveDevices = 0;
-    m_iPreviousDevIndex = -1;
-#ifdef __VINYLCONTROL__
-    m_VinylControl[0] = NULL;
-    m_VinylControl[1] = NULL;
-#endif
-
-    m_HostAPI = api_name;
-    m_bInit = false;
-}
-
-PlayerPortAudio::~PlayerPortAudio()
-{
-    //Close all the audio devices.
-    close();
-
-    if (m_bInit) {
-        Pa_Terminate();
-        m_painited = false;
-    }
-}
-
-bool PlayerPortAudio::initialize()
-{
-    PaError err = Pa_Initialize();
-    if (err!=paNoError)
-    {
-        qDebug() << "PortAudio error: " << Pa_GetErrorText(err);
-        m_bInit = false;
-    }
-    else
-        m_bInit = true;
-
-    return m_bInit;
-}
-
-bool PlayerPortAudio::open()
-{
-    Player::open();
-
-#ifdef __VINYLCONTROL__
-    //Create a new VinylControl object so that it updates it's settings.
-    //TODO: FIX THIS SAMPLERATE HACKAGE
-    int iTempSrate = m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt();
-    if (iTempSrate <= 0)
-        iTempSrate = 44100;
-    m_VinylControl[0] = new VinylControlProxy(m_pConfig, "[Channel1]", iTempSrate);
-    m_VinylControl[1] = new VinylControlProxy(m_pConfig, "[Channel2]", 44100);
-#endif
-
-    // Find out which device to open. Select the first one listed as either Master Left,
-    // Master Right, Head Left, Head Right. If other devices are requested for opening
-    // than the one selected here, set them to "None" in the config database
-
-    // For the record, with MAX_AUDIODEVICES currently set to 2, we're referring to
-    // Master Left/Right as device 1 and Headphones as device 2 (unless Master and
-    // Headphones are on the same device, then we only use device 1.)
-    PaDeviceIndex id[MAX_AUDIODEVICES];
-    PaDeviceIndex id_input[MAX_AUDIODEVICES];
-    int iChannelMax[MAX_AUDIODEVICES]; /** Maximum number of channels needed */
-    PaDeviceIndex temp_id = -1;
-    QString name;
-
-    for (int i = 0; i < MAX_AUDIODEVICES; i++)
-    {
-        iChannelMax[i] = -1;
-        id[i] = -1;
-        id_input[i] = -1;
-    }
-
-    m_iMasterLeftCh = -1;
-    m_iMasterRigthCh = -1;
-    m_iHeadLeftCh = -1;
-    m_iHeadRightCh = -1;
-
-    waitForNextOutput.wakeAll();
-
-    // Master left
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
-    temp_id = getDeviceID(name);
-    if (temp_id >= 0)
-    {
-        if (getChannelNo(name)>=0)
-        {
-            id[0] = temp_id;
-            iChannelMax[0] = getChannelNo(name);
-            m_iMasterLeftCh = getChannelNo(name)-1;
-        }
-    }
-
-    // Master right
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterRight"));
-    temp_id = getDeviceID(name);
-    if (getChannelNo(name)>=0                      //Make sure we got a valid number of channels
-        && ((id[0]==-1 && temp_id>=0)              //Make sure the left channel and this (right) channel IDs are valid
-            || (temp_id!=-1 && id[0]==temp_id))) //No idea...
-    {
-        id[0] = math_max(temp_id, id[0]);
-        iChannelMax[0] = math_max(iChannelMax[0], getChannelNo(name));
-        m_iMasterRigthCh = getChannelNo(name)-1;
-    }
-
-    // Head left
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadLeft"));
-    temp_id = getDeviceID(name);
-    if (getChannelNo(name)>=0 && ((id[1]==-1 && temp_id>=0) || (temp_id!=-1 && id[1]==temp_id)))
-    {
-        id[1] = temp_id;
-        iChannelMax[1] = math_max(iChannelMax[1], getChannelNo(name));
-        m_iHeadLeftCh = getChannelNo(name)-1;
-    }
-
-    // Head right
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadRight"));
-    temp_id = getDeviceID(name);
-    if (getChannelNo(name)>=0 && ((id[1]==-1 && temp_id>=0) || (temp_id!=-1 && id[1]==temp_id)))
-    {
-        id[1] = temp_id;
-        iChannelMax[1] = math_max(iChannelMax[1], getChannelNo(name));
-        m_iHeadRightCh = getChannelNo(name)-1;
-    }
-
-    // Inputs
-    name = m_pConfig->getValueString(ConfigKey("[VinylControl]","DeviceInputDeck1"));
-    temp_id = getInputDeviceID(name);
-    if (temp_id != -1)
-        id_input[0] = temp_id;
-
-    name = m_pConfig->getValueString(ConfigKey("[VinylControl]","DeviceInputDeck2"));
-    temp_id = getInputDeviceID(name);
-    if (temp_id != -1)
-        id_input[1] = temp_id;
-
-    // Check if any of the devices in the config database needs to be set to "None"
-    if (m_iMasterLeftCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-    if (m_iMasterRigthCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-    if (m_iHeadLeftCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-    if (m_iHeadRightCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-
-    // Number of channels to open
-    int iChannels[MAX_AUDIODEVICES];
-    if (id[0] == id[1]) //Check if we're supposed to use the same soundcard for both Master and Headphone output
-    {
-        iChannels[0] = math_max(iChannelMax[0], iChannelMax[1]);
-        iChannels[1] = -1;
-    }
-    else
-    {
-        iChannels[0] = iChannelMax[0];
-        iChannels[1] = iChannelMax[1];
-    }
-
-    // Sample rate
-    int iSrate = m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt();
-    if (iSrate <= 0)
-        iSrate = 44100;
-
-    // Get latency in msec
-    int iLatencyMSec = m_pConfig->getValueString(ConfigKey("[Soundcard]","Latency")).toInt();
-
-    if (iLatencyMSec <= 0)     //Make sure we don't get a crazy latency value.
-        iLatencyMSec = 100;
-
-    // Latency in samples
-    //int iLatencySamples = (int)((float)(iSrate*iChannels)/1000.f*(float)iLatencyMSec); //Pre-multisoundcard output
-    int iLatencySamples = (int)((float)(iSrate*math_max(iChannels[0], iChannels[1]))/1000.f*(float)iLatencyMSec);
-
-    // Round to the nearest multiple of 4.
-    iLatencySamples -= (iLatencySamples % 4);
-    iLatencySamples += 4;
-
-    qDebug() << "iLatencySamples: " << iLatencySamples;
-
-    // Apply simple rule to determine number of buffers
-    if (iLatencySamples/kiMaxFrameSize<2)
-        m_iNumberOfBuffers = 2;
-    else
-        m_iNumberOfBuffers = iLatencySamples/kiMaxFrameSize;
-
-    // Frame size...
-    int iFramesPerBuffer = iLatencySamples/m_iNumberOfBuffers;
-
-    // Ensure the chosen configuration is valid
-    //TODO: Fix this in our PortAudio-v19 implementation. PA19 ditches the Pa_GetMinNumBuffers() function, as it's
-    //		not needed anymore.
-    /*
-       if (m_iNumberOfBuffers<Pa_GetMinNumBuffers(iFramesPerBuffer,iSrate))
-       {
-        m_iNumberOfBuffers = Pa_GetMinNumBuffers(iFramesPerBuffer,iSrate);
-
-        iLatencyMSec = (1000*iFramesPerBuffer*m_iNumberOfBuffers)/(iSrate*iChannels);
-        m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(iLatencyMSec));
-       }
-     */
-
-    // Callback function to use
-    PaStreamCallback * callback = paV19Callback;
-    qDebug() << "PortAudio: kiMaxFrameSize: " << kiMaxFrameSize << ", iLatencyMSec: " << iLatencyMSec;
-    qDebug() << "PortAudio: id[0] " << id[0] << ", sr " << iSrate << ", ch[0] " << iChannels[0] << ", bufsize " << iFramesPerBuffer << ", bufno " << m_iNumberOfBuffers << ", req. latency " << iLatencyMSec << " msec";
-    qDebug() << "PortAudio: id[1] " << id[1] << ", sr " << iSrate << ", ch[1] " << iChannels[1] << ", bufsize " << iFramesPerBuffer << ", bufno " << m_iNumberOfBuffers << ", req. latency " << iLatencyMSec << " msec";
-
-    qDebug() << "Device 1 indices (id[0]: " << id[0] << ", id_input[0]: " << id_input[0] << ")";
-    qDebug() << "Device 2 indices (id[1]: " << id[1] << ", id_input[1]: " << id_input[1] << ")";
-
-
-    //If all devices are set to "None", then just return.
-    if ((id[0] < 0) && (id[1] < 0))
-        return false;
-
-    PaStreamParameters outputParams[MAX_AUDIODEVICES];
-    PaStreamParameters inputParams[MAX_AUDIODEVICES];
-    bool bDeviceAlreadyOpened = false;
-
-    //Set up and open each soundcard that we need!
-    for (int i = 0; i < MAX_AUDIODEVICES; i++)
-    {
-        bDeviceAlreadyOpened = false;
-
-        for (int j = 0; j < i; j++)         //Check to see if we've already opened this device (eg. Maybe we're already using it for Master output)
-        {
-            if (j != i)
-            {
-                if (id[j] == id[i] && id[j] != -1) {
-                    bDeviceAlreadyOpened = true;
-                }
-            }
-        }
-        if (((id[i] != -1) && !bDeviceAlreadyOpened) || (id_input[i] != -1))         //Make sure we're supposed to open this device...
-        {
-            qDebug() << "PortAudio: Trying to open device id " << id[i] << ", with channels " << iChannels[i] << " at samplerate " << iSrate;
-            PaStreamParameters * p_outputParams;
-            outputParams[i].device = id[i];
-            outputParams[i].channelCount = iChannels[i];
-            outputParams[i].sampleFormat = paFloat32;
-            outputParams[i].suggestedLatency = ((float)iLatencyMSec) / 1000.0f;             //Latency in seconds.
-            outputParams[i].hostApiSpecificStreamInfo = NULL;
-
-            PaStreamParameters * p_inputParams;            //TODO: Does this need to be made an array like inputParams?
-            inputParams[i].device = id_input[i];             //TODO: Isn't the "input" device always the same as the output device? (if there is an output device?)
-            //	     Maybe that should be restricted in the preferences GUI...
-            // NO, it's not always the same as the output device! Consider using 4 outputs on your
-            // master soundcard, and a different soundcard for input. The master's id is the same as the first
-            // for the second stream, but the input id for the second stream is something different.
-            inputParams[i].channelCount = 2;
-            inputParams[i].sampleFormat = paInt16;             //As needed by scratchlib
-            inputParams[i].suggestedLatency = ((float)iLatencyMSec) / 1000.0f;             //Latency in seconds.
-            inputParams[i].hostApiSpecificStreamInfo = NULL;
-
-            if (id[i] < 0 || (i > 0 && (id[1] == id[0])))             //TODO: comment this the same as the below input one.... kinda
-                p_outputParams = NULL;
-            else
-                p_outputParams = &(outputParams[i]);
-
-            if (id_input[i] < 0)             //If we're not supposed to open an input device, set the pointer to be null.
-                p_inputParams = NULL;                 //NULL input params in Pa_OpenStream means don't open the device for input.
-            else
-                p_inputParams = &(inputParams[i]);
-
-            PaError err = paNoError;
-
-            // Set up a struct containing all the data we want to pass back to the callback
-            callbackStuff[i].player = this;
-            callbackStuff[i].devIndex = i;         //The audio device's index (NOT THE PORTAUDIO ID!!!)
-
-            // Try open device using iChannelMax
-            err = Pa_OpenStream(&m_pStream[i],
-                                p_inputParams,                                  // Input parameters
-                                p_outputParams,                                 // Output parameters
-                                iSrate,                                                 // Sample rate
-                                iFramesPerBuffer,                               // Frames per buffer
-                                paClipOff,                                              // Stream flags
-                                callback,                                               // Stream callback
-                                &(callbackStuff[i]));                                      // Pointer passed to the callback function
-
-            if (err == paNoError)
-                m_iChannels[i] = iChannels[i];
-            else
-            {
-                // Try open device using maximum supported channels by soundcard
-                iChannels[i] = Pa_GetDeviceInfo(id[i])->maxOutputChannels;
-                outputParams[i].channelCount = iChannels[i];
-                err = Pa_OpenStream(&m_pStream[i],
-                                    p_inputParams,                              // Input parameters
-                                    p_outputParams,                             // Output parameters
-                                    iSrate,                                                     // Sample rate
-                                    iFramesPerBuffer,                                   // Frames per buffer
-                                    paClipOff,                                                  // Stream flags
-                                    callback,                                                   // Stream callback
-                                    &(callbackStuff[i]));                                          // Pointer passed to the callback function
-                if (err==paNoError)
-                    m_iChannels[i] = iChannels[i];
-            }
-
-            if( err != paNoError )         //*** TODO/WARNING: I don't think this block makes any sense whatsoever... - Albert April 21/07
-            {
-                // Try open device using only two channels
-                outputParams[i].channelCount = 2;
-                err = Pa_OpenStream(&m_pStream[i],
-                                    p_inputParams,                              // Input parameters
-                                    p_outputParams,                             // Output parameters
-                                    iSrate,                                                     // Sample rate
-                                    iFramesPerBuffer,                                   // Frames per buffer
-                                    paClipOff,                                                  // Stream flags
-                                    callback,                                                   // Stream callback
-                                    &(callbackStuff[i]));                                          // Pointer passed to the callback function
-                if (err==paNoError)
-                {
-                    m_iChannels[i] = 2;
-
-                    // Update channel variables and config database
-                    if (m_iMasterLeftCh>1)
-                    {
-                        m_iMasterLeftCh = -1;
-                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-                    }
-                    if (m_iMasterRigthCh>1)
-                    {
-                        m_iMasterRigthCh = -1;
-                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-                    }
-                    if (m_iHeadLeftCh>1)
-                    {
-                        m_iHeadLeftCh = -1;
-                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-                    }
-                    if (m_iHeadRightCh>1)
-                    {
-                        m_iHeadRightCh = -1;
-                        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-                    }
-                }
-            }
-
-            if( err != paNoError )         //Make sure we opened the soundcard successfully.
-            {
-                qDebug() << "PortAudio: Open stream error: " << Pa_GetErrorText(err);
-                qDebug() << "PortAudio: More error info: " << Pa_GetLastHostErrorInfo()->errorText;
-
-                m_devId[i] = -1;
-                m_inputDevId[i] = -1;
-                m_iChannels[i] = -1;
-
-                return false;
-            }
-
-            m_devId[i] = id[i];
-            m_inputDevId[i] = id_input[i];
-
-            // Update SRATE and Latency ControlObjects
-            m_pControlObjectSampleRate->queueFromThread((double)iSrate);
-            m_pControlObjectLatency->queueFromThread((double)iLatencyMSec);
-
-            // Start stream
-            err = Pa_StartStream(m_pStream[i]);
-            if (err != paNoError)
-            {
-                qDebug() << "PortAudio: Start stream " << i << " error: " << Pa_GetErrorText(err);
-                m_pStream[i] = 0;
-            }
-            else
-                qDebug() << "PortAudio: Started stream " << i << " successfully";
-
-            //If the Master and Headphone devices are the same, or the Headphone device is set to "None",
-            //then we can just break out of this loop:
-            /*if ((id[0] == id[1]) || (id[1] == -1))
-               {
-                    calculateNumActiveDevices();
-                    return true;
-               }*/
-        }
-    }
-
-    /*
-       //Print out all the information about the soundcards and their channels.
-       qDebug() << "PortAudio: ==Soundcard Summary==";
-       qDebug() << "Device 1 index: " << m_devId[0];
-       qDebug() << "Device 2 index: " << m_devId[1];
-       qDebug() << "Channels for device 1: " << m_iChannels[0];
-       qDebug() << "Channels for device 2: " << m_iChannels[1];
-       qDebug() << "m_iMasterLeftCh: " << m_iMasterLeftCh;
-       qDebug() << "m_iMasterRightCh: " << m_iMasterRigthCh;
-       qDebug() << "m_iHeadLeftCh: " << m_iHeadLeftCh;
-       qDebug() << "m_iHeadRightCh: " << m_iHeadRightCh;
-     */
-
-    calculateNumActiveDevices();
-    return true;
-}
-
-//Find out how many active/open soundcards there are.
-void PlayerPortAudio::calculateNumActiveDevices()
-{
-    int count = 0;
-
-    for (int i = 0; i < MAX_AUDIODEVICES; i++)
-    {
-        if (m_devId[i] != -1)
-            count++;
-    }
-    m_iNumActiveDevices = count;
-}
-
-void PlayerPortAudio::close()
-{
-    //waitForNextOutput.wakeAll();
-
-    for (int i = 0; i < MAX_AUDIODEVICES; i++)
-    {
-        m_devId[i] = -1;
-        m_iChannels[i] = 0;
-
-        // Stop streams
-        if (m_pStream[i])
-        {
-            PaError err = Pa_StopStream(m_pStream[i]);
-            if( err != paNoError )
-                qDebug() << "PortAudio: Stop stream " << i << " error: " << Pa_GetErrorText(err) << ",";
-
-            // Close streams
-            err = Pa_CloseStream(m_pStream[i]);
-            if( err != paNoError )
-                qDebug() << "PortAudio: Close stream " << i << " error: " << Pa_GetErrorText(err);
-        }
-
-        m_pStream[i] = 0;
-        waitForNextOutput.wakeAll();
-        calculateNumActiveDevices();
-    }
-
-    //waitForNextOutput.wakeAll();
-
-    m_iMasterLeftCh = -1;
-    m_iMasterRigthCh = -1;
-    m_iHeadLeftCh = -1;
-    m_iHeadRightCh = -1;
-}
-
-void PlayerPortAudio::setDefaults()
-{
-    // Get list of interfaces
-    QStringList interfaces = getInterfaces();
-
-    // Set first interfaces to master left
-    QStringListIterator it(interfaces);
-
-    if (it.hasNext())
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((it.next())));
-    }
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-
-    // Set second interface to master right
-    if (it.hasNext())
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((it.next())));
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-
-    // Set head left and right to none
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-
-    // Set default sample rate
-    QStringList srates = getSampleRates();
-    QStringListIterator itSRates(srates);
-    while (it.hasNext())
-    {
-        QString s;
-        s = itSRates.next();
-        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue(s));
-        if ((s).toInt()>=44100)
-            break;
-    }
-
-    // Set currently used latency in config database
-    //int msec = (int)(1000.*(2.*1024.)/(2.*(float)(*it).toInt()));
-    int msec = 100;     // More conservative defaults
-
-    m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
-}
-
-QStringList PlayerPortAudio::getInterfaces()
-{
-    //qDebug() << "PortAudio: getInterfaces()";
-
-    QStringList result;
-    const PaHostApiInfo * apiInfo = NULL;
-    const PaDeviceInfo * devInfo = NULL;
-    QString api;
-
-    if (m_HostAPI == "None")
-        return result;
-
-    PaDeviceIndex numDevices = Pa_GetDeviceCount();
-
-    for (int i = 0; i < numDevices; i++)
-    {
-        devInfo = Pa_GetDeviceInfo(i);
-
-        // Add the device if it is an output device:
-        //if (devInfo != NULL && devInfo->maxOutputChannels > 0)
-        if (devInfo != NULL)
-        {
-            apiInfo = Pa_GetHostApiInfo(devInfo->hostApi);
-            //api = apiInfo->name;
-            //qDebug() << "Api name: " << apiInfo->name;
-            qDebug() << devInfo->name;
-            //qDebug() << "m_HostAPI: " << m_HostAPI << "devInfo->hostApi: " << new QString(devInfo->hostApi);
-
-            //... and make sure the interface matches the API we've selected.
-            if (m_HostAPI == apiInfo->name)
-            {
-                //qDebug() << "name " << devInfo->name << ", API " << devInfo->hostApi << ", maxOutputChannels: " << devInfo->maxOutputChannels;
-
-                for (int j=1; j <= devInfo->maxOutputChannels; ++j)
-                    result.append(QString("%1 (channel %2)").arg(devInfo->name).arg(j));
-            }
-        }
-    }
-    //qDebug() << "PortAudio: getInterfaces() end";
-    return result;
-}
-
-/**
- * Get the audio input/capture interfaces from PortAudio
- */
-QStringList PlayerPortAudio::getInputInterfaces()
-{
-    qDebug() << "PortAudio: getInputInterfaces()";
-
-    QStringList result;
-    const PaHostApiInfo * apiInfo = NULL;
-    const PaDeviceInfo * devInfo = NULL;
-    QString api;
-
-    if (m_HostAPI == "None")
-        return result;
-
-    PaDeviceIndex numDevices = Pa_GetDeviceCount();
-
-    for (int i = 0; i < numDevices; i++)
-    {
-        devInfo = Pa_GetDeviceInfo(i);
-
-        // Add the device if it is an input device:
-        //if (devInfo != NULL && devInfo->maxOutputChannels > 0)
-        if (devInfo != NULL && (devInfo->maxInputChannels > 0))
-        {
-            apiInfo = Pa_GetHostApiInfo(devInfo->hostApi);
-            //api = apiInfo->name;
-            qDebug() << "Api name: " << apiInfo->name;
-            qDebug(devInfo->name);
-            //qDebug() << "m_HostAPI: " << m_HostAPI << "devInfo->hostApi: " << new QString(devInfo->hostApi);
-
-            //... and make sure the interface matches the API we've selected.
-            if (m_HostAPI == apiInfo->name )
-            {
-                qDebug() << "name " << devInfo->name << ", API " << devInfo->hostApi << ", maxInputChannels: " << devInfo->maxInputChannels;
-
-                result.append(QString("%1").arg(devInfo->name));
-
-                //for (int j=1; j <= devInfo->maxInputChannels; ++j)
-                // 	result.append(QString("%1 (channel %2)").arg(devInfo->name).arg(j));
-            }
-        }
-    }
-    qDebug() << "PortAudio: getInputInterfaces() end";
-    return result;
-}
-
-
-QStringList PlayerPortAudio::getSampleRates()
-{
-    // Returns a sorted list of supported sample rates of the currently opened device.
-    // If no device is open, return the list of sample rates supported by the
-    // default device
-    qDebug() << "PortAudio: getSampleRates()";
-
-    PaError err;
-    PaDeviceIndex id = m_devId[0];              //TODO: This is a hack to pick the samplerates from the first soundcard....
-    //Figure out something smarter...
-    //Something smarter = poll the samplerates from both soundcards, and
-    //only display the samlerates that are supported by both. - Albert April 30, 2007
-    //The problem with doing this right now is that Pa_IsFormatSupported when using
-    //ALSA takes ages for some reason, and polling both soundcards here would
-    //double this 5 second freeze that Mixxx experiences because of Pa_IsFormatSupported
-    //(buggily) blocking.
-    //qDebug() << "m_devId[0]: " << m_devId[0];
-    if (id<0)
-        id = Pa_GetDefaultOutputDevice();
-
-    //const PaDeviceInfo *devInfo = Pa_GetDeviceInfo(id);
-
-    PaStreamParameters outputParams;
-    outputParams.device = id;
-    outputParams.channelCount = 2;
-    outputParams.sampleFormat = paFloat32;
-    outputParams.suggestedLatency = .150;     //Latency in seconds.
-    outputParams.hostApiSpecificStreamInfo = NULL;
-
-    Q3ValueList<double> desiredSampleRates; //A list of all the sample rates we're going to suggest.
-    Q3ValueList<double> validSampleRates; //A list containing all the supported sample rates.
-
-    //Here's all the sample rates we're going to suggest to PortAudio. We check which ones
-    //are actually supported below.
-    desiredSampleRates.append(11025.0);
-    desiredSampleRates.append(22050.0);
-    desiredSampleRates.append(44100.0);
-    desiredSampleRates.append(48000.0);
-    desiredSampleRates.append(96000.0);
-
-    // Sample rates
-    //if (devInfo)
-    {
-        for (unsigned int j=0; j < desiredSampleRates.count(); j++)
-        {
-            //Check if each sample rate is supported, if so, add them to the list of supported sample rates.
-            qDebug() << "PortAudio: checking if sample rate is supported...";
-            err = Pa_IsFormatSupported(NULL, &outputParams, desiredSampleRates[j]);
-            if (err == paFormatIsSupported)     //The format IS supported.
-            {
-                validSampleRates.append(desiredSampleRates[j]);
-                qDebug() << "Supported...";
-            }
-            else
-            {
-                qDebug() << "PortAudio error: " << Pa_GetErrorText(err) << ", id was: " << id;
-            }
-        }
-    }
-
-    //If for some reason our enumeration of the sample rates failed, throw
-    //in some default values. (PortAudio might be being sketchy...)
-    if (validSampleRates.count() == 0)
-    {
-        validSampleRates.append(44100.0);
-        validSampleRates.append(48000.0);
-        validSampleRates.append(96000.0);
-    }
-
-    // Sort list
-#ifndef QT3_SUPPORT
-    qSort(validSampleRates);
-#endif
-
-    // Convert srlist to stringlist
-    QStringList result;
-    for (unsigned int i = 0; i < validSampleRates.count(); ++i)
-        result.append(QString("%1").arg((*validSampleRates.at(i))));
-
-    qDebug() << "PortAudio: getSampleRates() end";
-
-    return result;
-}
-
-QStringList PlayerPortAudio::getSoundApiList()
-{
-    //Note: Need to put the active API at the top of the list.
-    QStringList apiList;
-    const PaHostApiInfo * apiInfo = NULL;
-
-    //We need to initialize PortAudio before we find out what APIs are present.
-    //Even if this gets called while PortAudio is already initialized, the docs
-    //say this is OK (I think...).
-
-    PaError err = paNoError;
-
-    // So this little hackfest saves buggy drivers from being really buggy - AD
-    if (!m_painited) {
-        err = Pa_Initialize();
-        m_painited = true;
-    }
-
-    if (err == paNoError)
-    {
-        for (int i = 0; i < Pa_GetHostApiCount(); i++)
-        {
-            apiInfo = Pa_GetHostApiInfo(i);
-            //qDebug() << "Api name: " << apiInfo->name;
-            apiList.append(apiInfo->name);
-        }
-//		Pa_Terminate();
-    }
-    else
-        qDebug() << "PortAudio error: " << Pa_GetErrorText(err);
-
-    return apiList;
-
-    /*
-     #ifdef __LINUX__
-       return QStringList("OSS (PA)");
-     #endif
-     #ifdef __MACX__
-       return QStringList("CoreAudio (PA)");
-     #endif
-     #ifdef __WIN__
-       return QStringList("WMME (PA)");
-     #endif
-     */
-
-}
-
-
-PaDeviceIndex PlayerPortAudio::getDeviceID(QString name)
-{
-    //qDebug() << "PortAudio: getDeviceID(" + name + ")";
-    PaDeviceIndex no = Pa_GetDeviceCount();
-    for (int i=0; i<no; i++)
-    {
-        const PaDeviceInfo * devInfo = Pa_GetDeviceInfo(i);
-
-        // Add the device if it is an output device:
-        if (devInfo != 0 && devInfo->maxOutputChannels > 0)
-        {
-            for (int j = 1; j <= devInfo->maxOutputChannels; ++j)
-                if (QString("%1 (channel %2)").arg(devInfo->name).arg(j) == name)
-                {
-                    //qDebug() << "PortAudio: getDeviceID(" + name + "), returning device id " << i;
-                    return i;
-                }
-        }
-    }
-    return -1;
-}
-
-
-PaDeviceIndex PlayerPortAudio::getInputDeviceID(QString name)
-{
-    qDebug() << "PortAudio: getInputDeviceID(" + name + ")";
-    PaDeviceIndex no = Pa_GetDeviceCount();
-    for (int i=0; i<no; i++)
-    {
-        const PaDeviceInfo * devInfo = Pa_GetDeviceInfo(i);
-
-        // Add the device if it is an input device:
-        if (devInfo != 0 && devInfo->maxOutputChannels > 0)
-        {
-            for (int j = 1; j <= devInfo->maxInputChannels; ++j)
-            {
-                //qDebug() << "Comparing: " << QString("%1").arg(devInfo->name);
-                //qDebug() << "and: " << name;
-                if (QString("%1").arg(devInfo->name) == name)
-                {
-                    qDebug() << "PortAudio: getInputDeviceID(" + name + "), returning device id " << i;
-                    return i;
-                }
-            }
-        }
-    }
-    qDebug() << "PortAudio: getInputDeviceID(" + name + "), returning device id -1";
-    return -1;
-}
-
-PaDeviceIndex PlayerPortAudio::getChannelNo(QString name)
-{
-    PaDeviceIndex no = Pa_GetDeviceCount();
-    for (int i=0; i<no; i++)
-    {
-        const PaDeviceInfo * devInfo = Pa_GetDeviceInfo(i);
-
-        // Add the device if it is an output device:
-        if (devInfo!=0 && devInfo->maxOutputChannels > 0)
-        {
-            for (int j=1; j<=devInfo->maxOutputChannels; ++j)
-                if (QString("%1 (channel %2)").arg(devInfo->name).arg(j) == name)
-                    return j;
-        }
-    }
-    return -1;
-}
-
-/** -------- ------------------------------------------------------
-        Purpose: This callback function gets called everytime a soundcard runs out of samples to play.
-                         This is where the soundcard/PortAudio ask Mixxx for more samples, and get them in
-                         an orderly fashion (*thread safety). This is also where the "routing" of audio samples
-                         take place.
-        -------- ------------------------------------------------------
- */
-int PlayerPortAudio::callbackProcess(unsigned long framesPerBuffer, float * out, short * in, int devIndex)
-{
-    //if (m_iBufferSize==0)
-    //    m_iBufferSize = iBufferSize*m_iNumberOfBuffers;
-
-    static float * tmp; //Leave this as static - it's shared between two threads.
-    float * output = out;
-    int i;
-
-    /*
-       if (!out)
-       {
-        qDebug() << "output is NULL!";
-       prevDevice.lock();
-       m_iPreviousDevIndex = devIndex; //Save this devIndex as the previous one
-        prevDevice.unlock();
-        waitForNextOutput.wakeAll(); //Allow the other thread to give at 'er
-        return 0;
-       }*/
-
-    prevDevice.lock();
-    int iPreviousDevIndex = m_iPreviousDevIndex;
-    prevDevice.unlock();
-
-    if (iPreviousDevIndex == devIndex && m_iNumActiveDevices > 1)
-    {
-        lockSamples.lock();
-        waitForNextOutput.wait(&lockSamples);
-        lockSamples.unlock();
-    }
-
-    //Only fill the buffer with sound data from Mixxx once.
-    lockSamples.lock();
-    if (devIndex == 0)
-    {
-        tmp = prepareBuffer(iBufferSize);
-    }
-    lockSamples.unlock();
-
-    // Reset sample for each open channel
-    for (i=0; i<framesPerBuffer*m_iChannels[devIndex]; i++)
-        output[i] = 0.;
-
-    // Copy to output buffer
-    for (i=0; i<iBufferSize; i++)
-    {
-        if (devIndex == 0) //For the first sound device
-        {
-            if (m_iMasterLeftCh>=0  && m_iChannels[devIndex]>=1) { output[m_iMasterLeftCh]  += tmp[(i*4)  ]/32768.;}
-            if (m_iMasterRigthCh>=0 && m_iChannels[devIndex]>=2) output[m_iMasterRigthCh] += tmp[(i*4)+1]/32768.;
-            if (m_iHeadLeftCh>=0    && m_iChannels[devIndex]>=3) output[m_iHeadLeftCh]    += tmp[(i*4)+2]/32768.;
-            if (m_iHeadRightCh>=0   && m_iChannels[devIndex]>=4) output[m_iHeadRightCh]   += tmp[(i*4)+3]/32768.;
-        }
-        else if (devIndex == 1)         //If there's a second sound device, route the headphones to it.
-        {
-            //if (m_iMasterLeftCh>=0  && m_iChannels[devIndex]>=1) { output[m_iMasterLeftCh]  += tmp[(i*4)  ]/32768.;}
-            //if (m_iMasterRigthCh>=0 && m_iChannels[devIndex]>=2) output[m_iMasterRigthCh] += tmp[(i*4)+1]/32768.;
-            //qDebug() << "m_iHeadLeftCh:" << m_iHeadLeftCh << "devIndex: " << devIndex << "tmp[blah]" << tmp[(i*4)+2]/32768.;
-            if (m_iHeadLeftCh>=0    && m_iChannels[devIndex]>=1) output[m_iHeadLeftCh]    += tmp[(i*4)+2]/32768.;
-            if (m_iHeadRightCh>=0   && m_iChannels[devIndex]>=2) output[m_iHeadRightCh]   += tmp[(i*4)+3]/32768.;
-            //qDebug() << "headphones!";
-        }
-        for (int j=0; j < m_iChannels[devIndex]; ++j)
-            *output++;
-    }
-
-    if (in)
-    {
-#ifdef __VINYLCONTROL__
-        m_VinylControl[devIndex]->AnalyseSamples(in, iBufferSize);
-#endif
-    }
-
-    prevDevice.lock();
-    m_iPreviousDevIndex = devIndex; //Save this devIndex as the previous one
-    prevDevice.unlock();
-    waitForNextOutput.wakeAll();     //Allow the other thread to give at 'er
-
-    return 0;
-}
-
-
-/* -------- ------------------------------------------------------
-   Purpose: Wrapper function to call processing loop function,
-            implemented as a method in a class. Used in PortAudio,
-            which knows nothing about C++.
-   Input:   .
-   Output:  -
-   -------- ------------------------------------------------------ */
-int paV19Callback(const void * inputBuffer, void * outputBuffer,
-                  unsigned long framesPerBuffer,
-                  const PaStreamCallbackTimeInfo * timeInfo,
-                  PaStreamCallbackFlags statusFlags,
-                  void * _callbackStuff)
-{
-    /*
-       //Variables that are used in the human-readable form of function call from hell (below).
-       static PlayerPortAudio* _player;
-       static int devIndex;
-       _player = ((PAPlayerCallbackStuff*)_callbackStuff)->player;
-       devIndex = ((PAPlayerCallbackStuff*)_callbackStuff)->devIndex;
-     */
-
-    //Human-readable form of the function call from hell:
-    //return _player->callbackProcess(framesPerBuffer, (float *)outputBuffer, devIndex);
-
-    //Function call from hell (might provide a little bit of cheapo thread safety to do it this way):
-    return ((PAPlayerCallbackStuff *)_callbackStuff)->player->callbackProcess(framesPerBuffer, (float *)outputBuffer, (short *)inputBuffer, ((PAPlayerCallbackStuff *)_callbackStuff)->devIndex);
-}
-
diff -Nru mixxx-1.6.0/src/playerportaudio.h mixxx-1.6.1/src/playerportaudio.h
--- mixxx-1.6.0/src/playerportaudio.h	2007-08-21 23:16:40.000000000 -0500
+++ mixxx-1.6.1/src/playerportaudio.h	1969-12-31 18:00:00.000000000 -0600
@@ -1,109 +0,0 @@
-/***************************************************************************
-                          playerportaudiov19.h  -  description
-                             -------------------
-    begin                : Wed Feb 20 2002
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-    email                : 
- ***************************************************************************/
-
-/***************************************************************************
- *                                                                         *
- *   This program is free software; you can redistribute it and/or modify  *
- *   it under the terms of the GNU General Public License as published by  *
- *   the Free Software Foundation; either version 2 of the License, or     *
- *   (at your option) any later version.                                   *
- *                                                                         *
- ***************************************************************************/
-
-#ifndef PLAYERPORTAUDIO_H
-#define PLAYERPORTAUDIO_H
-
-#include <q3semaphore.h>
-#include <qthread.h>
-#include <QMutex>
-#include <QWaitCondition>
-#include "player.h"
-#include "portaudio.h"
-#ifdef __VINYLCONTROL__
-#include "vinylcontrolproxy.h"
-#endif
-
-/**
-  *@author Tue and Ken Haste Andersen
-  */
-
-/** Maximum frame size used with PortAudio. Used to determine no of buffers
-  * when setting latency */
-const int kiMaxFrameSize = 1024;
-
-class PlayerPortAudio; //Forward declaration
-
-/** A struct to some stuff we need to pass along to the callback through PortAudio **/
-struct PAPlayerCallbackStuff
-{
-	PlayerPortAudio* player;
-	int devIndex;
-};
-
-class PlayerPortAudio : public Player  {
-public:
-    PlayerPortAudio(ConfigObject<ConfigValue> *config, QString api_name);
-    ~PlayerPortAudio();
-    bool initialize();
-    bool open();
-    void close();
-    void setDefaults();
-    QStringList getInterfaces();
-    QStringList getInputInterfaces();
-    QStringList getSampleRates();
-    static QStringList getSoundApiList();
-    QString getSoundApiName() { return getSoundApiList().front(); };
-    void calculateNumActiveDevices();
-    /** Satisfy virtual declaration in EngineObject */
-    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
-    /** Process samples. Called from PortAudio callback */
-    int callbackProcess(int iBufferSize, float *out, short *in, int devIndex);
-
-    static bool m_painited;
-protected:
-    /** Get id of device with a given name. Returns -1 if device is not found */
-    PaDeviceIndex getDeviceID(QString name);
-    PaDeviceIndex getInputDeviceID(QString name);
-    /** Get channel number of device with a given name. Returns -1 if device is not found */
-    int getChannelNo(QString name);
-
-    /** PortAudio stream */
-    PaStream *m_pStream[MAX_AUDIODEVICES];
-    /** Id of currently open device. -1 if no device is open */
-    PaDeviceIndex m_devId[MAX_AUDIODEVICES];
-    PaDeviceIndex m_inputDevId[MAX_AUDIODEVICES]; //For input devices
-    /** Channels used for each output from Mixxx. Set to -1 when not in use */
-    int m_iMasterLeftCh, m_iMasterRigthCh, m_iHeadLeftCh, m_iHeadRightCh;
-    /** True if PortAudio was sucessfully initialized */
-    bool m_bInit;
-    /** A struct to hold some information/pointers we need to pass to our callback function */
-    PAPlayerCallbackStuff callbackStuff[MAX_AUDIODEVICES];
-    /** Number of buffers */
-    int m_iNumberOfBuffers;
-    /** Number of active/open soundcards **/
-    int m_iNumActiveDevices;
-    /** Name of the current audio API inside PortAudio **/
-    QString m_HostAPI;
-    /** Mutex so that two threads don't try to prepare a new buffer full of samples from Mixxx at the same time */
-    QMutex lockSamples; 
-    /** Wait condition that forces multiple PortAudio callbacks in separate threads to play nicely */
-    QWaitCondition waitForNextOutput;
-    QMutex waitMutex;
-    int m_iPreviousDevIndex;
-    QMutex prevDevice; 
-#ifdef __VINYLCONTROL__
-    VinylControlProxy *m_VinylControl[2];
-#endif
-};
-
-int paV19Callback(const void *inputBuffer, void *outputBuffer,
-                      unsigned long framesPerBuffer,
-                      const PaStreamCallbackTimeInfo* timeInfo,
-                      PaStreamCallbackFlags statusFlags,
-                      void *_player);
-#endif
diff -Nru mixxx-1.6.0/src/playerproxy.cpp mixxx-1.6.1/src/playerproxy.cpp
--- mixxx-1.6.0/src/playerproxy.cpp	2007-09-09 17:52:24.000000000 -0500
+++ mixxx-1.6.1/src/playerproxy.cpp	1969-12-31 18:00:00.000000000 -0600
@@ -1,189 +0,0 @@
-/***************************************************************************
-                          playerproxy.cpp  -  description
-                             -------------------
-    begin                : Fri Nov 21 2003
-    copyright            : (C) 2003 by Tue and Ken Haste Andersen
-    email                :
-***************************************************************************/
-
-/***************************************************************************
-*                                                                         *
-*   This program is free software; you can redistribute it and/or modify  *
-*   it under the terms of the GNU General Public License as published by  *
-*   the Free Software Foundation; either version 2 of the License, or     *
-*   (at your option) any later version.                                   *
-*                                                                         *
-***************************************************************************/
-
-#include "playerproxy.h"
-
-#ifdef __PORTAUDIO__
-    #include "playerportaudio.h"
-#endif
-#ifdef __JACK__
-    #include "playerjack.h"
-#endif
-#ifdef __RTAUDIO__
-    #include "playerrtaudio.h"
-#endif
-#ifdef __ASIO__
-    #include "playerasio.h"
-#endif
-#ifdef __ALSA__
-    #include "playeralsa.h"
-#endif
-
-Player * PlayerProxy::m_pPlayer = 0;
-
-PlayerProxy::PlayerProxy(ConfigObject<ConfigValue> * pConfig) : Player(pConfig)
-{
-    // Set API based on info stored in config database
-    setSoundApi(m_pConfig->getValueString(ConfigKey("[Soundcard]","SoundApi")));
-
-    if (!m_pPlayer)
-    {
-        // No valid API is stored in the database, so select the default
-        QStringList api = getSoundApiList();
-        QStringList::iterator it = api.begin();
-        while (!m_pPlayer && it!=api.end())
-        {
-            if (setSoundApi((*it)))
-            {
-                m_pConfig->set(ConfigKey("[Soundcard]","SoundApi"), ConfigValue((*it)));
-                m_pPlayer->setDefaults();
-            }
-            *it++;
-        }
-    }
-}
-
-PlayerProxy::~PlayerProxy()
-{
-    if (m_pPlayer)
-        delete m_pPlayer;
-}
-
-bool PlayerProxy::open()
-{
-    if (m_pPlayer)
-        return m_pPlayer->open();
-    else
-        return false;
-}
-
-void PlayerProxy::close()
-{
-    if (m_pPlayer)
-        m_pPlayer->close();
-}
-
-void PlayerProxy::setDefaults()
-{
-    if (m_pPlayer)
-        m_pPlayer->setDefaults();
-}
-
-QStringList PlayerProxy::getInterfaces()
-{
-    if (m_pPlayer)
-        return m_pPlayer->getInterfaces();
-    else
-        return QStringList();
-}
-
-QStringList PlayerProxy::getInputInterfaces()
-{
-    if (m_pPlayer)
-        return m_pPlayer->getInputInterfaces();
-    else
-        return QStringList();
-}
-
-QStringList PlayerProxy::getSampleRates()
-{
-    if (m_pPlayer)
-        return m_pPlayer->getSampleRates();
-    else
-        return QStringList();
-}
-
-QString PlayerProxy::getSoundApiName()
-{
-    if (m_pPlayer)
-        return m_pPlayer->getSoundApiName();
-    else
-        return QString();
-}
-
-
-QStringList PlayerProxy::getSoundApiList()
-{
-    QStringList result;
-#ifdef __PORTAUDIO__
-    result += PlayerPortAudio::getSoundApiList();
-#endif
-#ifdef __JACK__
-    result.append(PlayerJack::getSoundApi());
-#endif
-#ifdef __RTAUDIO__
-    result.append(PlayerRtAudio::getSoundApi());
-#endif
-#ifdef __ASIO__
-    result.append(PlayerAsio::getSoundApi());
-#endif
-#ifdef __ALSA__
-    result.append(PlayerALSA::getSoundApi());
-#endif
-
-    return result;
-}
-
-bool PlayerProxy::setSoundApi(QString name)
-{
-    if (m_pPlayer)
-        delete m_pPlayer;
-    m_pPlayer = 0;
-
-#ifdef __PORTAUDIO__
-    if (PlayerPortAudio::getSoundApiList().contains(name))
-        m_pPlayer = new PlayerPortAudio(m_pConfig, name);
-#endif
-
-#ifdef __JACK__
-    if (name == PlayerJack::getSoundApi())
-        m_pPlayer = new PlayerJack(m_pConfig);
-#endif
-
-#ifdef __RTAUDIO__
-    if (name == PlayerRtAudio::getSoundApi())
-        m_pPlayer = new PlayerRtAudio(m_pConfig);
-#endif
-
-#ifdef __ASIO__
-    if (name == PlayerAsio::getSoundApi())
-        m_pPlayer = new PlayerAsio(m_pConfig);
-#endif
-
-#ifdef __ALSA__
-    if (name == PlayerALSA::getSoundApi())
-        m_pPlayer = new PlayerALSA(m_pConfig);
-#endif
-
-    // Try initializing the selected API
-    bool init = false;
-    if (m_pPlayer)
-        init = m_pPlayer->initialize();
-
-    if (!init)
-    {
-        if (m_pPlayer)
-            delete m_pPlayer;
-        m_pPlayer = 0;
-    }
-
-    if (m_pPlayer)
-        return true;
-    else
-        return false;
-}
-
diff -Nru mixxx-1.6.0/src/playerproxy.h mixxx-1.6.1/src/playerproxy.h
--- mixxx-1.6.0/src/playerproxy.h	2007-08-21 23:16:40.000000000 -0500
+++ mixxx-1.6.1/src/playerproxy.h	1969-12-31 18:00:00.000000000 -0600
@@ -1,54 +0,0 @@
-/***************************************************************************
-                          playerproxy.h  -  description
-                             -------------------
-    begin                : Fri Nov 21 2003
-    copyright            : (C) 2003 by Tue Haste Andersen
-    email                : 
- ***************************************************************************/
-
-/***************************************************************************
- *                                                                         *
- *   This program is free software; you can redistribute it and/or modify  *
- *   it under the terms of the GNU General Public License as published by  *
- *   the Free Software Foundation; either version 2 of the License, or     *
- *   (at your option) any later version.                                   *
- *                                                                         *
- ***************************************************************************/
-
-#ifndef PLAYERPROXY_H
-#define PLAYERPROXY_H
-
-#include "player.h"
-
-class PlayerProxy : public Player
-{
-public:
-    PlayerProxy(ConfigObject<ConfigValue> *pConfig);
-    ~PlayerProxy();
-    bool initialize() { return false; };
-    bool open();
-    void close();
-    void setDefaults();
-    QStringList getInterfaces();
-    QStringList getInputInterfaces();
-    QStringList getSampleRates();
-    QString getSoundApiName();
-    /** Return list of supported sound api's */
-    static QStringList getSoundApiList();
-    /** Select active sound api */
-    bool setSoundApi(QString name);
-    /** Satisfy virtual declaration in EngineObject */
-    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
-    /** Static function to return active player class. Needed for ASIO implementation */
-    static Player *getPlayer() { return m_pPlayer; }
-
-protected:
-    /** Pointer to active Player class */
-    static Player *m_pPlayer;
-};
-
-#endif
-
-
-
-
diff -Nru mixxx-1.6.0/src/playerrtaudio.cpp mixxx-1.6.1/src/playerrtaudio.cpp
--- mixxx-1.6.0/src/playerrtaudio.cpp	2008-03-27 00:04:43.000000000 -0500
+++ mixxx-1.6.1/src/playerrtaudio.cpp	1969-12-31 18:00:00.000000000 -0600
@@ -1,442 +0,0 @@
-/***************************************************************************
-                          playerrtaudio.cpp  -  description
-                             -------------------
-    begin                : Thu May 20 2004
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-    email                :
-***************************************************************************/
-
-/***************************************************************************
-*                                                                         *
-*   This program is free software; you can redistribute it and/or modify  *
-*   it under the terms of the GNU General Public License as published by  *
-*   the Free Software Foundation; either version 2 of the License, or     *
-*   (at your option) any later version.                                   *
-*                                                                         *
-***************************************************************************/
-
-#include "playerrtaudio.h"
-#include "controlobject.h"
-
-/** Maximum frame size used with RtAudio. Used to determine no of buffers
- * when setting latency */
-const int kiMaxFrameSize = 64;
-
-
-PlayerRtAudio::PlayerRtAudio(ConfigObject<ConfigValue> * config) : Player(config)
-{
-    m_pRtAudio = 0;
-    m_bInit = false;
-
-    m_devId = -1;
-    m_iNumberOfBuffers = 2;
-    m_iChannels = -1;
-    m_iMasterLeftCh = -1;
-    m_iMasterRigthCh = -1;
-    m_iHeadLeftCh = -1;
-    m_iHeadRightCh = -1;
-}
-
-PlayerRtAudio::~PlayerRtAudio()
-{
-    if (m_devId>=1)
-    {
-        try {
-            m_pRtAudio->stopStream();
-            m_pRtAudio->closeStream();
-        }
-        catch (RtError &error)
-        {
-            error.printMessage();
-        }
-    }
-
-    if (m_bInit)
-        delete m_pRtAudio;
-}
-
-bool PlayerRtAudio::initialize()
-{
-    try {
-        m_pRtAudio = new RtAudio();
-    }
-    catch (RtError &error)
-    {
-        error.printMessage();
-        m_pRtAudio = 0;
-        m_bInit = false;
-    }
-    m_bInit = true;
-
-    return m_bInit;
-}
-
-bool PlayerRtAudio::open()
-{
-    Player::open();
-
-    // Find out which device to open. Select the first one listed as either Master Left,
-    // Master Right, Head Left, Head Right. If other devices are requested for opening
-    // than the one selected here, set them to "None" in the config database
-    int id = -1;
-    int temp = -1;
-    QString name;
-    /** Maximum number of channels needed */
-    int iChannelMax = -1;
-
-    m_iMasterLeftCh = -1;
-    m_iMasterRigthCh = -1;
-    m_iHeadLeftCh = -1;
-    m_iHeadRightCh = -1;
-
-    // Master left
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterLeft"));
-    temp = getDeviceID(name);
-    if (temp>=0)
-    {
-        if (getChannelNo(name)>=0)
-        {
-            id = temp;
-            iChannelMax = getChannelNo(name);
-            m_iMasterLeftCh = getChannelNo(name)-1;
-        }
-    }
-
-    // Master right
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceMasterRight"));
-    temp = getDeviceID(name);
-    if (getChannelNo(name)>=0 && ((id==-1 && temp>=0) || (temp!=-1 && id==temp)))
-    {
-        id = temp;
-        iChannelMax = math_max(iChannelMax, getChannelNo(name));
-        m_iMasterRigthCh = getChannelNo(name)-1;
-    }
-
-    // Head left
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadLeft"));
-    temp = getDeviceID(name);
-    if (getChannelNo(name)>=0 && ((id==-1 && temp>=0) || (temp!=-1 && id==temp)))
-    {
-        id = temp;
-        iChannelMax = math_max(iChannelMax, getChannelNo(name));
-        m_iHeadLeftCh = getChannelNo(name)-1;
-    }
-
-    // Head right
-    name = m_pConfig->getValueString(ConfigKey("[Soundcard]","DeviceHeadRight"));
-    temp = getDeviceID(name);
-    if (getChannelNo(name)>=0 && ((id==-1 && temp>=0) || (temp!=-1 && id==temp)))
-    {
-        id = temp;
-        iChannelMax = math_max(iChannelMax, getChannelNo(name));
-        m_iHeadRightCh = getChannelNo(name)-1;
-    }
-
-    // Check if any of the devices in the config database needs to be set to "None"
-    if (m_iMasterLeftCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-    if (m_iMasterRigthCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-    if (m_iHeadLeftCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-    if (m_iHeadRightCh<0)
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-
-    // Number of channels to open
-    int iChannels = iChannelMax;
-
-    // Sample rate
-    int iSrate = m_pConfig->getValueString(ConfigKey("[Soundcard]","Samplerate")).toInt();
-
-    // Get latency in msec
-    int iLatencyMSec = m_pConfig->getValueString(ConfigKey("[Soundcard]","Latency")).toInt();
-
-    // Latency in samples
-    int iLatencySamples = (int)((float)(iSrate*iChannels)/1000.f*(float)iLatencyMSec);
-
-    // Apply simple rule to determine number of buffers
-    if (iLatencySamples/kiMaxFrameSize<2)
-        m_iNumberOfBuffers = 2;
-    else
-        m_iNumberOfBuffers = iLatencySamples/kiMaxFrameSize;
-
-    // Frame size...
-    int iFramesPerBuffer = iLatencySamples/m_iNumberOfBuffers;
-
-    qDebug() << "RtAudio: id " << id << ", sr " << iSrate << ", ch " << iChannels << ", bufsize " << iFramesPerBuffer << ", bufno " << m_iNumberOfBuffers;
-
-    if (id<1)
-    {
-        m_devId = -1;
-        return false;
-    }
-
-    // Start playback
-    try
-    {
-        // Update SRATE and Latency ControlObjects
-        m_pControlObjectSampleRate->queueFromThread((double)iSrate);
-        m_pControlObjectLatency->queueFromThread((double)iLatencyMSec);
-
-        // Open stream
-        m_pRtAudio->openStream(id, iChannels, 0, 0, RTAUDIO_FLOAT32, iSrate, &iFramesPerBuffer, m_iNumberOfBuffers);
-
-        // Set callback function
-        m_pRtAudio->setStreamCallback(&rtCallback, (void *)this);
-
-        m_iChannels = iChannels;
-        m_devId = id;
-
-        // Start playback
-        m_pRtAudio->startStream();
-    }
-    catch (RtError &error)
-    {
-        error.printMessage();
-
-        m_devId = -1;
-        m_iChannels = -1;
-
-        return false;
-    }
-
-    return true;
-}
-
-void PlayerRtAudio::close()
-{
-    m_iChannels = 0;
-    m_iMasterLeftCh = -1;
-    m_iMasterRigthCh = -1;
-    m_iHeadLeftCh = -1;
-    m_iHeadRightCh = -1;
-
-    if (m_devId>0)
-    {
-        qDebug() << "close";
-        try
-        {
-            m_pRtAudio->stopStream();
-            m_pRtAudio->closeStream();
-        }
-        catch (RtError &error)
-        {
-            error.printMessage();
-        }
-    }
-    m_devId = -1;
-}
-
-void PlayerRtAudio::setDefaults()
-{
-    // Get list of interfaces
-    QStringList interfaces = getInterfaces();
-
-    // Set first interfaces to master left
-    QStringList::iterator it = interfaces.begin();
-    if (it!=interfaces.end())
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue((*it)));
-    }
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterLeft"),ConfigValue("None"));
-
-    // Set second interface to master right
-    ++it;
-    if (it!=interfaces.end())
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue((*it)));
-    else
-        m_pConfig->set(ConfigKey("[Soundcard]","DeviceMasterRight"),ConfigValue("None"));
-
-    // Set head left and right to none
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadLeft"),ConfigValue("None"));
-    m_pConfig->set(ConfigKey("[Soundcard]","DeviceHeadRight"),ConfigValue("None"));
-
-    // Set default sample rate
-    QStringList srates = getSampleRates();
-    it = srates.begin();
-    while (it!=srates.end())
-    {
-        m_pConfig->set(ConfigKey("[Soundcard]","Samplerate"),ConfigValue((*it)));
-
-        if ((*it).toInt()>=44100)
-            break;
-
-        ++it;
-    }
-
-    // Set currently used latency in config database
-    int msec = (int)(1000.*(2.*1024.)/(2.*(float)(*it).toInt()));
-    m_pConfig->set(ConfigKey("[Soundcard]","Latency"), ConfigValue(msec));
-}
-
-QStringList PlayerRtAudio::getInterfaces()
-{
-    QStringList result;
-
-    int no = m_pRtAudio->getDeviceCount();
-    RtAudioDeviceInfo info;
-    for (int i=1; i<=no; i++)
-    {
-        bool bGotInfo = false;
-        try {
-            info = m_pRtAudio->getDeviceInfo(i);
-            bGotInfo = true;
-        }
-        catch (RtError &error)
-        {
-            error.printMessage();
-        }
-
-        // Add the device if it is an output device:
-        if (bGotInfo && info.outputChannels > 0)
-        {
-            qDebug() << "name " << info.name.c_str();
-            for (int j=1; j<=info.outputChannels; ++j)
-                result.append(QString("%1 (channel %2)").arg(info.name.c_str()).arg(j));
-        }
-    }
-
-    return result;
-}
-
-QStringList PlayerRtAudio::getSampleRates()
-{
-    // Returns the list of supported sample rates of the currently opened device.
-    // If no device is open, return the list of sample rates supported by the
-    // default device
-    int id = m_devId;
-    if (id<1)
-        id = 1;
-
-    RtAudioDeviceInfo info;
-    try {
-        info = m_pRtAudio->getDeviceInfo(id);
-    }
-    catch (RtError &error)
-    {
-        error.printMessage();
-        return QStringList();
-    }
-
-    // Sample rates
-    QValueList<int> srlist;
-
-    if (info.sampleRates.size() > 0)
-    {
-        for (unsigned int j=0; j<info.sampleRates.size(); ++j)
-            srlist.append(info.sampleRates[j]);
-    }
-
-    // Sort list
-#ifndef QT3_SUPPORT
-    qHeapSort(srlist);
-#endif
-
-    // Convert srlist to stringlist
-    QStringList result;
-    for (int i=0; i<srlist.count(); ++i)
-        result.append(QString("%1").arg((*srlist.at(i))));
-
-    return result;
-}
-
-QString PlayerRtAudio::getSoundApi()
-{
-#ifdef __WIN__
-    return QString("DirectSound");
-#endif
-#ifdef __LINUX__
-    return QString("ALSA");
-#endif
-}
-
-int PlayerRtAudio::getDeviceID(QString name)
-{
-    int no = m_pRtAudio->getDeviceCount();
-    for (int i=1; i<=no; i++)
-    {
-        RtAudioDeviceInfo info;
-        try {
-            info = m_pRtAudio->getDeviceInfo(i);
-        }
-        catch (RtError &error)
-        {
-            error.printMessage();
-            return -1;
-        }
-
-        // Add the device if it is an output device:
-        if (info.outputChannels > 0)
-        {
-            for (int j=1; j<=info.outputChannels; ++j)
-                if (QString("%1 (channel %2)").arg(info.name.c_str()).arg(j) == name)
-                    return i;
-        }
-    }
-    return -1;
-}
-
-int PlayerRtAudio::getChannelNo(QString name)
-{
-    int no = m_pRtAudio->getDeviceCount();
-    for (int i=1; i<=no; i++)
-    {
-        RtAudioDeviceInfo info;
-        try {
-            info = m_pRtAudio->getDeviceInfo(i);
-        }
-        catch (RtError &error)
-        {
-            error.printMessage();
-            return -1;
-        }
-
-        // Add the device if it is an output device:
-        if (info.outputChannels > 0)
-        {
-            for (int j=1; j<=info.outputChannels; ++j)
-                if (QString("%1 (channel %2)").arg(info.name.c_str()).arg(j) == name)
-                    return j;
-        }
-    }
-    return -1;
-}
-
-int PlayerRtAudio::callbackProcess(int iBufferSize, float * out)
-{
-    //m_iBufferSize = iBufferSize*m_iNumberOfBuffers;
-
-    float * tmp = prepareBuffer(iBufferSize);
-    float * output = out;
-    int i;
-
-    // Reset sample for each open channel
-    for (i=0; i<iBufferSize*m_iChannels; i++)
-        output[i] = 0.;
-
-    // Copy to output buffer
-    for (i=0; i<iBufferSize; i++)
-    {
-        if (m_iMasterLeftCh>=0) output[m_iMasterLeftCh]  += tmp[(i*4)  ]/32768.;
-        if (m_iMasterRigthCh>=0) output[m_iMasterRigthCh] += tmp[(i*4)+1]/32768.;
-        if (m_iHeadLeftCh>=0) output[m_iHeadLeftCh]    += tmp[(i*4)+2]/32768.;
-        if (m_iHeadRightCh>=0) output[m_iHeadRightCh]   += tmp[(i*4)+3]/32768.;
-
-        for (int j=0; j<m_iChannels; ++j)
-            *output++;
-    }
-
-    return 0;
-}
-
-/* -------- ------------------------------------------------------
-   Purpose: Wrapper function to call processing loop function,
-            implemented as a method in a class. Used in RtAudio.
-   Input:   .
-   Output:  -
-   -------- ------------------------------------------------------ */
-int rtCallback(char * outputBuffer, int framesPerBuffer, void * pPlayer)
-{
-    return ((PlayerRtAudio *)pPlayer)->callbackProcess(framesPerBuffer, (float *)outputBuffer);
-}
diff -Nru mixxx-1.6.0/src/playerrtaudio.h mixxx-1.6.1/src/playerrtaudio.h
--- mixxx-1.6.0/src/playerrtaudio.h	2005-03-03 06:01:40.000000000 -0600
+++ mixxx-1.6.1/src/playerrtaudio.h	1969-12-31 18:00:00.000000000 -0600
@@ -1,65 +0,0 @@
-/***************************************************************************
-                          playerrtaudio.h  -  description
-                             -------------------
-    begin                : Thu May 20 2004
-    copyright            : (C) 2002 by Tue and Ken Haste Andersen
-    email                :
- ***************************************************************************/
-
-/***************************************************************************
- *                                                                         *
- *   This program is free software; you can redistribute it and/or modify  *
- *   it under the terms of the GNU General Public License as published by  *
- *   the Free Software Foundation; either version 2 of the License, or     *
- *   (at your option) any later version.                                   *
- *                                                                         *
- ***************************************************************************/
-
-#ifndef PLAYERRTAUDIO_H
-#define PLAYERRTAUDIO_H
-
-#include "player.h"
-#include <RtAudio.h>
-
-/**
-  *@author Tue and Ken Haste Andersen
-  */
-
-
-class PlayerRtAudio : public Player  {
-public:
-    PlayerRtAudio(ConfigObject<ConfigValue> *config);
-    ~PlayerRtAudio();
-    bool initialize();
-    bool open();
-    void close();
-    void setDefaults();
-    QStringList getInterfaces();
-    QStringList getSampleRates();
-    static QString getSoundApi();
-    QString getSoundApiName() { return getSoundApi(); };
-    /** Satisfy virtual declaration in EngineObject */
-    void process(const CSAMPLE *, const CSAMPLE *, const int) {};
-    /** Process samples. Called from RtAudio callback */
-    int callbackProcess(int iBufferSize, float *out);
-
-protected:
-    /** Pointer to RtAudio object */
-    RtAudio *m_pRtAudio;
-    /** Get id of device with a given name. Returns -1 if device is not found */
-    int getDeviceID(QString name);
-    /** Get channel number of device with a given name. Returns -1 if device is no found */
-    int getChannelNo(QString name);
-    /** Id of currently open device. -1 if no device is open */
-    int m_devId;
-    /** Channels used for each output from Mixxx. Set to -1 when not in use */
-    int m_iMasterLeftCh, m_iMasterRigthCh, m_iHeadLeftCh, m_iHeadRightCh;
-    /** True if RtAudio was sucessfully initialized */
-    bool m_bInit;
-    /** Number of buffers */
-    int m_iNumberOfBuffers;
-};
-
-int rtCallback(char *outputBuffer, int framesPerBuffer, void *_player);
-
-#endif
diff -Nru mixxx-1.6.0/src/SConscript mixxx-1.6.1/src/SConscript
--- mixxx-1.6.0/src/SConscript	2008-07-29 10:43:09.000000000 -0500
+++ mixxx-1.6.1/src/SConscript	2008-09-23 01:51:13.000000000 -0500
@@ -325,12 +325,19 @@
 
 if not env.GetOption('clean') and not SCons.Util.containsAny(os.sys.argv, ['-h', '--help']):
 	conf = Configure(env, custom_tests = { 'CheckForPKGConfig' : CheckForPKGConfig, 'CheckForPKG' : CheckForPKG })
-	
+
+#	if platform == 'linux' and conf.CheckCHeader('mp4.h'):
+#		print "Enabling M4A support!"
+#		env.Append(CXXFLAGS = '-D__M4A__')
+#		sources += Split("""soundsourcem4a.cpp """);  # MP4/M4A Support
+#		env.Append(LIBS = 'libmp4v2')
+#		env.Append(LIBS = 'libfaad')
+
 #TODO: Add all of the other configure checks as custom_tests properly.
 	
 # On Posix default SCons.LIBPREFIX = 'lib', on Windows default SCons.LIBPREFIX = ''
 
-	if not conf.CheckLibWithHeader('portaudio', 'portaudio.h', 'C'):
+	if not conf.CheckLibWithHeader('portaudio', 'portaudio.h', 'C') and not (platform == 'win32' and conf.CheckLib(['portaudio'])):
 		print 'Did not find libportaudio.a, portaudio.lib, or the PortAudio-v19 development header files - exiting!'
 		Exit(1)
 
@@ -630,10 +637,23 @@
 flags_ipod = getFlags(env, 'ipod', 0)
 if int(flags_ipod):
 	env.Append(CXXFLAGS = '-D__IPOD__')
-	# env.Append(LIBS = 'libgpod-1.0')
-	# env.Append(LIBS = 'glib-2.0')
-	env.ParseConfig('pkg-config libgpod-1.0 --silence-errors --cflags --libs')
-	env.ParseConfig('pkg-config glib-2.0 --silence-errors --cflags --libs')
+
+	if platform == 'win32':
+		env.Append(LIBS = 'gpod');
+		# You must check v-this-v directory out from http://publicsvn.songbirdnest.com/vendor-binaries/trunk/windows-i686-msvc8/libgpod/
+		env.Append(LIBPATH='../../../windows-i686-msvc8/libgpod/release/lib')
+		# Following building the following must be added to the dist folder in order for mixxx to run with ipod support on Win32
+		# \windows-i686-msvc8\libgpod\release\lib\libgpod.dll
+		# \windows-i686-msvc8\glib\release\bin\libgobject-2.0-0.dll
+		# \windows-i686-msvc8\glib\release\bin\libglib-2.0-0.dll
+		# \windows-i686-msvc8\libiconv\release\bin\iconv.dll
+		# \windows-i686-msvc8\gettext\release\binintl.dll
+	if platform == 'linux' or platform == 'osx':
+		# env.Append(LIBS = 'libgpod-1.0')
+		# env.Append(LIBS = 'glib-2.0')
+		env.ParseConfig('pkg-config libgpod-1.0 --silence-errors --cflags --libs')
+		env.ParseConfig('pkg-config glib-2.0 --silence-errors --cflags --libs')
+		
 	sources += Split("""wipodtracksmodel.cpp """) #IPOD
 	print "iPod support... enabled"
 	build_flags += 'ipod '
@@ -681,14 +701,17 @@
 else:
 	print "Vinyl Control... disabled"
 
-flags_msvcdebug = getFlags(env, 'msvcdebug', 1)
-if int(flags_msvcdebug) and platform == 'win32':
-	env.Append(LINKFLAGS = '/DEBUG')
-	env.Append(CXXFLAGS = '/ZI')
-	print "MSVC Debugging... enabled"
-	build_flags += 'msvcdebug '
-else:
-	print "MSVC Debugging... disabled"
+flags_msvcdebug = getFlags(env, 'msvcdebug', 0)
+if platform == 'win32':
+	if int(flags_msvcdebug):
+		env.Append(CCFLAGS = '/MDd') # required for sndfile w/ flac support on windows
+		env.Append(LINKFLAGS = '/DEBUG')
+		env.Append(CXXFLAGS = '/ZI')
+		print "MSVC Debugging... enabled"
+		build_flags += 'msvcdebug '
+	else:
+		env.Append(CCFLAGS = '/MD') # required for sndfile w/ flac support on windows
+		print "MSVC Debugging... disabled"
 
 flags_script = getFlags(env, 'script', 0)
 if int(flags_script):
@@ -1010,8 +1033,14 @@
 	#env.Alias('mixxx', icon)
 	env.Alias('mixxx', binary)
 
-
-
+	def cleanSVNDirsFromDist():
+		if os.path.exists("dist"):
+			print "Cleaning .svn directories from dist... ",
+			os.system('cmd.exe /c @FOR /F "tokens=*" %D IN (\'dir /b /ad /s dist\*.svn*\') do @(rd /s /q "%D") 2> NUL')
+			print "Done."
+	import atexit
+	atexit.register(cleanSVNDirsFromDist)
+	
 def BuildRelease(target, source, env):
 	print
 	print "==== Mixxx Post-Build Checks ===="
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/back-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/back-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/back-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/back-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/back-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/back-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/back-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/back-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/crosshandle-on-.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/crosshandle-on-.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/crossslider-off-.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/crossslider-off-.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/cue-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/cue-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/cue-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/cue-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/cue-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/cue-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/cue-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/cue-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/endoftrack-loop-ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/endoftrack-loop-ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/endoftrack-loop-ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/endoftrack-loop-ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/endoftrack-next-ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/endoftrack-next-ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/endoftrack-next-ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/endoftrack-next-ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/endoftrack-stop-ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/endoftrack-stop-ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/endoftrack-stop-ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/endoftrack-stop-ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/flanger-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/flanger-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/flanger-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/flanger-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/flanger-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/flanger-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/flanger-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/flanger-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/fwd-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/fwd-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/fwd-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/fwd-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/fwd-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/fwd-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/fwd-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/fwd-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/head-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/head-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/head-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/head-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/head-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/head-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/head-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/head-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg0.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg0.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg10.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg10.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg11.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg11.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg12.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg12.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg13.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg13.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg14.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg14.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg15.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg15.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg16.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg16.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg17.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg17.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg18.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg18.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg19.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg19.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg20.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg20.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg21.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg21.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg22.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg22.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg23.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg23.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg24.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg24.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg25.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg25.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg26.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg26.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg27.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg27.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg28.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg28.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg29.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg29.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg30.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg30.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg31.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg31.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg32.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg32.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg33.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg33.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg34.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg34.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg35.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg35.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg36.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg36.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg37.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg37.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg38.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg38.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg39.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg39.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg3.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg3.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg40.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg40.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg41.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg41.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg42.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg42.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg43.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg43.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg44.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg44.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg4.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg4.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg5.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg5.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg6.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg6.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg7.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg7.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg8.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg8.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobg9.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobg9.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm0.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm0.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm10.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm10.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm11.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm11.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm12.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm12.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm13.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm13.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm14.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm14.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm15.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm15.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm16.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm16.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm17.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm17.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm18.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm18.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm19.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm19.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm20.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm20.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm21.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm21.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm22.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm22.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm23.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm23.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm24.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm24.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm25.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm25.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm26.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm26.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm27.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm27.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm28.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm28.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm29.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm29.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm30.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm30.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm31.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm31.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm32.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm32.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm33.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm33.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm34.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm34.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm35.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm35.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm36.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm36.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm37.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm37.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm38.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm38.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm39.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm39.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm3.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm3.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm40.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm40.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm41.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm41.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm42.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm42.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm43.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm43.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm44.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm44.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm4.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm4.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm5.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm5.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm6.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm6.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm7.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm7.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm8.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm8.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/knobm9.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/knobm9.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/master_vol_meter_off.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/master_vol_meter_off.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/master_vol_meter_on.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/master_vol_meter_on.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/play-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/play-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/play-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/play-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/play-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/play-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/play-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/play-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/playpos-marker1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/playpos-marker1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/playpos-marker2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/playpos-marker2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/playpos-slider.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/playpos-slider.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/ratedown-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/ratedown-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/ratedown-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/ratedown-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/ratedown-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/ratedown-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/ratedown-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/ratedown-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/ratehandle-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/ratehandle-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/ratehandle-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/ratehandle-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rateslider-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rateslider-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rateslider-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rateslider-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rateup-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rateup-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rateup-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rateup-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rateup-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rateup-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rateup-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rateup-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rev-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rev-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rev-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rev-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rev-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rev-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/rev-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/rev-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/scene.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/scene.png differ
diff -Nru mixxx-1.6.0/src/skins/Natt/skin.xml mixxx-1.6.1/src/skins/Natt/skin.xml
--- mixxx-1.6.0/src/skins/Natt/skin.xml	2008-07-11 02:25:36.000000000 -0500
+++ mixxx-1.6.1/src/skins/Natt/skin.xml	1969-12-31 18:00:00.000000000 -0600
@@ -1,1087 +0,0 @@
-<!DOCTYPE skin>
-<skin>
-
- <Background>
-  <Path>scene.png</Path>
- </Background>
-
- <SliderComposed>
-  <Tooltip>Crossfader</Tooltip>
-  <Handle>crosshandle-on-.png</Handle>
-  <Slider>crossslider-off-.png</Slider>
-  <Pos>310,237</Pos>
-  <Horizontal>true</Horizontal>
-  <Connection>
-   <ConfigKey>[Master],crossfader</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-  </Connection>
- </SliderComposed>
-
- <SliderComposed>
-  <Tooltip>Playback rate</Tooltip>
-  <Handle>ratehandle-on-Ch1.png</Handle>
-  <Slider>rateslider-off-Ch1.png</Slider>
-  <Pos>10,170</Pos>
-  <Horizontal>false</Horizontal>
-  <Connection>
-   <ConfigKey>[Channel1],rate</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-  </Connection>
- </SliderComposed>
-
- <SliderComposed>
-  <Tooltip>Playback rate</Tooltip>
-  <Handle>ratehandle-on-Ch2.png</Handle>
-  <Slider>rateslider-off-Ch2.png</Slider>
-  <Pos>706,170</Pos>
-  <Horizontal>false</Horizontal>
-  <Connection>
-   <ConfigKey>[Channel2],rate</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-  </Connection>
- </SliderComposed>
-
- <PushButton>
-  <Tooltip>Tempo syncronization</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>sync-on-Ch1.png</Pressed>
-   <Unpressed>sync-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>45,170</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],beatsync</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Tempo syncronization</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>sync-on-Ch2.png</Pressed>
-   <Unpressed>sync-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>741,170</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],beatsync</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>rateup-on-Ch1.png</Pressed>
-   <Unpressed>rateup-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>45,326</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],rate_perm_up</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_perm_up_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>rateup-on-Ch2.png</Pressed>
-   <Unpressed>rateup-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>741,326</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],rate_perm_up</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_perm_up_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>ratedown-on-Ch1.png</Pressed>
-   <Unpressed>ratedown-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>45,306</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],rate_perm_down</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_perm_down_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>ratedown-on-Ch2.png</Pressed>
-   <Unpressed>ratedown-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>741,306</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],rate_perm_down</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_perm_down_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Temporary change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>rateup-on-Ch1.png</Pressed>
-   <Unpressed>rateup-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>72,326</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_up</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_up</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_up_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_up_small</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Temporary change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>ratedown-on-Ch1.png</Pressed>
-   <Unpressed>ratedown-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>72,306</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_down</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_down</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_down_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],rate_temp_down_small</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Temporary change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>rateup-on-Ch2.png</Pressed>
-   <Unpressed>rateup-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>768,326</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_up</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_up</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_up_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_up_small</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Temporary change playback rate (left/right click: large/small effect)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>ratedown-on-Ch2.png</Pressed>
-   <Unpressed>ratedown-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>768,306</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_down</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_down</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_down_small</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],rate_temp_down_small</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <SliderComposed>
-  <Tooltip>Channel volume</Tooltip>
-  <Handle>volhandle-on-Ch1.png</Handle>
-  <Slider>volslider-off-Ch1.png</Slider>
-  <Pos>8,413</Pos>
-  <Horizontal>false</Horizontal>
-  <Connection>
-   <ConfigKey>[Channel1],volume</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-  </Connection>
- </SliderComposed>
-
- <SliderComposed>
-  <Tooltip>Channel volume</Tooltip>
-  <Handle>volhandle-on-Ch2.png</Handle>
-  <Slider>volslider-off-Ch2.png</Slider>
-  <Pos>703,413</Pos>
-  <Horizontal>false</Horizontal>
-  <Connection>
-   <ConfigKey>[Channel2],volume</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-  </Connection>
- </SliderComposed>
-
- <VuMeter>
-  <Tooltip>Master VuMeter</Tooltip>
-  <PathVu>master_vol_meter_on.png</PathVu>
-  <PathBack>master_vol_meter_off.png</PathBack>
-  <Pos>388,170</Pos>
-  <PeakHoldSize>5</PeakHoldSize>
-  <PeakHoldTime>400</PeakHoldTime>
-  <Horizontal>false</Horizontal>
-  <Connection>
-   <ConfigKey>[Master],VuMeter</ConfigKey>
-  </Connection>
- </VuMeter>
-
- <VuMeter>
-  <Tooltip>Channel VuMeter</Tooltip>
-  <PathVu>vumeter-on-Ch1.png</PathVu>
-  <PathBack>vumeter-off-Ch1.png</PathBack>
-  <Pos>82,413</Pos>
-  <PeakHoldSize>5</PeakHoldSize>
-  <PeakHoldTime>400</PeakHoldTime>
-  <Connection>
-   <ConfigKey>[Channel1],VuMeter</ConfigKey>
-  </Connection>
- </VuMeter>
-
- <VuMeter>
-  <Tooltip>Channel VuMeter</Tooltip>
-  <PathVu>vumeter-on-Ch2.png</PathVu>
-  <PathBack>vumeter-off-Ch2.png</PathBack>
-  <Pos>777,413</Pos>
-  <PeakHoldSize>5</PeakHoldSize>
-  <PeakHoldTime>400</PeakHoldTime>
-  <Connection>
-   <ConfigKey>[Channel2],VuMeter</ConfigKey>
-  </Connection>
- </VuMeter>
-
- <PushButton>
-  <Tooltip>End of track mode</Tooltip>
-  <NumberStates>3</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>endoftrack-stop-ch1.png</Pressed>
-   <Unpressed>endoftrack-stop-ch1.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>endoftrack-next-ch1.png</Pressed>
-   <Unpressed>endoftrack-next-ch1.png</Unpressed>
-  </State>
-  <State>
-   <Number>2</Number>
-   <Pressed>endoftrack-loop-ch1.png</Pressed>
-   <Unpressed>endoftrack-loop-ch1.png</Unpressed>
-  </State>
-  <Pos>334,170</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],TrackEndMode</ConfigKey>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>End of track mode</Tooltip>
-  <NumberStates>3</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>endoftrack-stop-ch2.png</Pressed>
-   <Unpressed>endoftrack-stop-ch2.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>endoftrack-next-ch2.png</Pressed>
-   <Unpressed>endoftrack-next-ch2.png</Unpressed>
-  </State>
-  <State>
-   <Number>2</Number>
-   <Pressed>endoftrack-loop-ch2.png</Pressed>
-   <Unpressed>endoftrack-loop-ch2.png</Unpressed>
-  </State>
-  <Pos>417,170</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],TrackEndMode</ConfigKey>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Left click: Pause/play, Right click: Set cue point</Tooltip>
-  <NumberStates>2</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>play-off-Ch1.png</Pressed>
-   <Unpressed>play-off-Ch1.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>play-on-Ch1.png</Pressed>
-   <Unpressed>play-on-Ch1.png</Unpressed>
-  </State>
-  <Pos>10,60</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],play</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],cue_set</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Left click: Pause/play, Right click: Set cue point</Tooltip>
-  <NumberStates>2</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>play-off-Ch2.png</Pressed>
-   <Unpressed>play-off-Ch2.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>play-on-Ch2.png</Pressed>
-   <Unpressed>play-on-Ch2.png</Unpressed>
-  </State>
-  <Pos>706,60</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],play</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],cue_set</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Go to and play (while playing), Set cue point (while stopped), Go to and stop (right-click)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>cue-on-Ch1.png</Pressed>
-   <Unpressed>cue-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>54,126</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],cue_default</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],cue_gotoandstop</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Go to and play (while playing), Set cue point (while stopped), Go to and stop (right-click)</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>cue-on-Ch2.png</Pressed>
-   <Unpressed>cue-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>750,126</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],cue_default</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],cue_gotoandstop</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Reverse playback</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>rev-on-Ch1.png</Pressed>
-   <Unpressed>rev-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>10,126</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],reverse</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],reverse</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Reverse playback</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>rev-on-Ch2.png</Pressed>
-   <Unpressed>rev-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>706,126</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],reverse</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],reverse</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Left click: Fast forward, Right click: End of track</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>fwd-on-Ch1.png</Pressed>
-   <Unpressed>fwd-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>54,97</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],fwd</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],fwd</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],end</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Left click: Fast backward, Right click: Start of track</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>back-on-Ch1.png</Pressed>
-   <Unpressed>back-off-Ch1.png</Unpressed>
-  </State>
-  <Pos>10,97</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],back</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],back</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],start</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Left click: Fast forward, Right click: End of track</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>fwd-on-Ch2.png</Pressed>
-   <Unpressed>fwd-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>750,97</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],fwd</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],fwd</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],end</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Left click: Fast backward, Right click: Start of track</Tooltip>
-  <NumberStates>1</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>back-on-Ch2.png</Pressed>
-   <Unpressed>back-off-Ch2.png</Unpressed>
-  </State>
-  <Pos>706,97</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],back</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],back</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-   <ButtonState>LeftButton</ButtonState>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],start</ConfigKey>
-   <EmitOnDownPress>true</EmitOnDownPress>
-   <ButtonState>RightButton</ButtonState>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Flanger on/off</Tooltip>
-  <NumberStates>2</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>flanger-off-Ch1.png</Pressed>
-   <Unpressed>flanger-off-Ch1.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>flanger-on-Ch1.png</Pressed>
-   <Unpressed>flanger-on-Ch1.png</Unpressed>
-  </State>
-  <Pos>46,364</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],flanger</ConfigKey>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Flanger on/off</Tooltip>
-  <NumberStates>2</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>flanger-off-Ch2.png</Pressed>
-   <Unpressed>flanger-off-Ch2.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>flanger-on-Ch2.png</Pressed>
-   <Unpressed>flanger-on-Ch2.png</Unpressed>
-  </State>
-  <Pos>742,364</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],flanger</ConfigKey>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Headphone cue</Tooltip>
-  <NumberStates>2</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>head-off-Ch1.png</Pressed>
-   <Unpressed>head-off-Ch1.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>head-on-Ch1.png</Pressed>
-   <Unpressed>head-on-Ch1.png</Unpressed>
-  </State>
-  <Pos>2,364</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],pfl</ConfigKey>
-  </Connection>
- </PushButton>
-
- <PushButton>
-  <Tooltip>Headphone cue</Tooltip>
-  <NumberStates>2</NumberStates>
-  <State>
-   <Number>0</Number>
-   <Pressed>head-off-Ch2.png</Pressed>
-   <Unpressed>head-off-Ch2.png</Unpressed>
-  </State>
-  <State>
-   <Number>1</Number>
-   <Pressed>head-on-Ch2.png</Pressed>
-   <Unpressed>head-on-Ch2.png</Unpressed>
-  </State>
-  <Pos>698,364</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],pfl</ConfigKey>
-  </Connection>
- </PushButton>
-
- <Knob>
-  <Tooltip>Master volume</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>547,243</Pos>
-  <Connection>
-   <ConfigKey>[Master],volume</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Master balance</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>600,243</Pos>
-  <Connection>
-   <ConfigKey>[Master],balance</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Headphone volume</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>504,243</Pos>
-  <Connection>
-   <ConfigKey>[Master],headVolume</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Mix between headphone and main</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>128,243</Pos>
-  <Connection>
-   <ConfigKey>[Master],headMix</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Flanger depth</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>224,243</Pos>
-  <Connection>
-   <ConfigKey>[Flanger],lfoDepth</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Flanger delay</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>267,243</Pos>
-  <Connection>
-   <ConfigKey>[Flanger],lfoDelay</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Flanger LFO period</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>171,243</Pos>
-  <Connection>
-   <ConfigKey>[Flanger],lfoPeriod</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Channel pre gain</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>46,424</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],pregain</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Channel highpass filter</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <DisabledPath>knobg%1.png</DisabledPath>
-  <Pos>46,470</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],filterHigh</ConfigKey>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],filterHighKill</ConfigKey>
-   <OnOff>true</OnOff>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Channel bandpass filter</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <DisabledPath>knobg%1.png</DisabledPath>
-  <Pos>46,517</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],filterMid</ConfigKey>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],filterMidKill</ConfigKey>
-   <OnOff>true</OnOff>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Channel lowpass filter</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <DisabledPath>knobg%1.png</DisabledPath>
-  <Pos>46,565</Pos>
-  <Connection>
-   <ConfigKey>[Channel1],filterLow</ConfigKey>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel1],filterLowKill</ConfigKey>
-   <OnOff>true</OnOff>
-  </Connection>
- </Knob>
-
-<Knob>
-  <Tooltip>Channel pre gain</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <Pos>741,424</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],pregain</ConfigKey>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Channel highpass filter</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <DisabledPath>knobg%1.png</DisabledPath>
-  <Pos>741,470</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],filterHigh</ConfigKey>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],filterHighKill</ConfigKey>
-   <OnOff>true</OnOff>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Channel bandpass filter</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <DisabledPath>knobg%1.png</DisabledPath>
-  <Pos>741,517</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],filterMid</ConfigKey>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],filterMidKill</ConfigKey>
-   <OnOff>true</OnOff>
-  </Connection>
- </Knob>
-
- <Knob>
-  <Tooltip>Channel lowpass filter</Tooltip>
-  <NumberStates>41</NumberStates>
-  <Path>knobm%1.png</Path>
-  <DisabledPath>knobg%1.png</DisabledPath>
-  <Pos>741,565</Pos>
-  <Connection>
-   <ConfigKey>[Channel2],filterLow</ConfigKey>
-  </Connection>
-  <Connection>
-   <ConfigKey>[Channel2],filterLowKill</ConfigKey>
-   <OnOff>true</OnOff>
-  </Connection>
- </Knob>
-
- <Visual>
-  <Tooltip>Drag with mouse to control sound playback. Drop tracks from external file manager here</Tooltip>
-  <Channel>1</Channel>
-  <Pos>116,47</Pos>
-  <Size>570,48</Size>
-  <BgColor>#161D2E</BgColor>
-  <SignalColor>#49BADA</SignalColor>
-  <BeatColor>#D8E1EA</BeatColor>
-  <MarkerColor>#30C0E9</MarkerColor>
-  <FisheyeColor>#ffffff</FisheyeColor>
-  <Zoom>true</Zoom>
- </Visual>
-
- <Visual>
-  <Tooltip>Drag with mouse to control sound playback. Drop tracks from external file manager here</Tooltip>
-  <Channel>2</Channel>
-  <Pos>116,102</Pos>
-  <Size>570,48</Size>
-  <BgColor>#161D2E</BgColor>
-  <SignalColor>#E08F47</SignalColor>
-  <BeatColor>#D8E1EA</BeatColor>
-  <MarkerColor>#30C0E9</MarkerColor>
-  <FisheyeColor>#ffffff</FisheyeColor>
-  <Zoom>true</Zoom>
- </Visual>
-
- <Text>
-  <Channel>1</Channel>
-  <Pos>136,6</Pos>
-  <Size>250,25</Size>
-  <FgColor>#ffffff</FgColor>
-  <BgColor>#0E1526</BgColor>
- </Text>
-
- <Text>
-  <Channel>2</Channel>
-  <Pos>418,6</Pos>
-  <Size>250,25</Size>
-  <FgColor>#ffffff</FgColor>
-  <BgColor>#0E1526</BgColor>
- </Text>
-
- <NumberBpm>
-  <Tooltip>Tempo display (Beat Per Minute)</Tooltip>
-  <Channel>1</Channel>
-  <Pos>118,170</Pos>
-  <Size>78,16</Size>
-  <NumberOfDigits>6</NumberOfDigits>
-  <BgColor>#0E1526</BgColor>
-  <FgColor>#ffffff</FgColor>
-  <Connection>
-   <ConfigKey>[Channel1],bpm</ConfigKey>
-  </Connection>
- </NumberBpm>
-
- <NumberBpm>
-  <Tooltip>Tempo display (Beat Per Minute)</Tooltip>
-  <Channel>2</Channel>
-  <Pos>484,170</Pos>
-  <Size>78,16</Size>
-  <NumberOfDigits>6</NumberOfDigits>
-  <BgColor>#0E1526</BgColor>
-  <FgColor>#ffffff</FgColor>
-  <Connection>
-   <ConfigKey>[Channel2],bpm</ConfigKey>
-  </Connection>
- </NumberBpm>
-
- <NumberPos>
-  <Tooltip>Play position/remaining time</Tooltip>
-  <Channel>1</Channel>
-  <Pos>220,170</Pos>
-  <Size>94,16</Size>
-  <Alignment>right</Alignment>
-  <NumberOfDigits>6</NumberOfDigits>
-  <BgColor>#0E1526</BgColor>
-  <FgColor>#ffffff</FgColor>
-  <Connection>
-   <ConfigKey>[Channel1],playposition</ConfigKey>
-  </Connection>
- </NumberPos>
-
- <NumberPos>
-  <Tooltip>Play position/remaining time</Tooltip>
-  <Channel>2</Channel>
-  <Pos>591,170</Pos>
-  <Size>94,16</Size>
-  <Alignment>right</Alignment>
-  <NumberOfDigits>6</NumberOfDigits>
-  <BgColor>#0E1526</BgColor>
-  <FgColor>#ffffff</FgColor>
-  <Connection>
-   <ConfigKey>[Channel2],playposition</ConfigKey>
-  </Connection>
- </NumberPos>
-
- <NumberRate>
-  <Tooltip>Playback rate factor</Tooltip>
-  <Channel>1</Channel>
-  <Pos>45,250</Pos>
-  <Size>47,16</Size>
-  <NumberOfDigits>3</NumberOfDigits>
-  <BgColor>#0E1526</BgColor>
-  <FgColor>#ffffff</FgColor>
-  <Connection>
-   <ConfigKey>[Channel1],rate</ConfigKey>
-  </Connection>
- </NumberRate>
-
- <NumberRate>
-  <Tooltip>Playback rate factor</Tooltip>
-  <Channel>2</Channel>
-  <Pos>740,250</Pos>
-  <Size>47,16</Size>
-  <NumberOfDigits>3</NumberOfDigits>
-  <BgColor>#0E1526</BgColor>
-  <FgColor>#ffffff</FgColor>
-  <Connection>
-   <ConfigKey>[Channel2],rate</ConfigKey>
-  </Connection>
- </NumberRate>
-
- <XSliderComposed>
-  <Tooltip>Play position - Use mouse to seek</Tooltip>
-  <Handle>playpos-marker1.png</Handle>
-  <Slider>playpos-slider.png</Slider>
-  <Pos>112,177</Pos>
-  <Horizontal>true</Horizontal>
-  <EventWhileDrag>no</EventWhileDrag>
-  <Connection>
-   <ConfigKey>[Channel1],playposition</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-  </Connection>
- </XSliderComposed>
-
- <XSliderComposed>
-  <Tooltip>Play position - Use mouse to seek</Tooltip>
-  <Handle>playpos-marker2.png</Handle>
-  <Slider>playpos-slider.png</Slider>
-  <Pos>487,177</Pos>
-  <Horizontal>true</Horizontal>
-  <EventWhileDrag>no</EventWhileDrag>
-  <Connection>
-   <ConfigKey>[Channel2],playposition</ConfigKey>
-   <EmitOnDownPress>false</EmitOnDownPress>
-  </Connection>
- </XSliderComposed>
-
- <Overview>
-	 <Tooltip>Overview display of waveform</Tooltip>
-	 <Channel>1</Channel>
-	 <Pos>118,193</Pos>
-	 <Size>264,36</Size>
-	 <BgColor>#161D2E</BgColor>
-	 <SignalColor>#5D9BA8</SignalColor>
-	 <MarkerColor>#ffffff</MarkerColor>
-	 <Connection>
-		 <ConfigKey>[Channel1],playposition</ConfigKey>
-		 <EmitOnDownPress>false</EmitOnDownPress>
-	 </Connection>
- </Overview>
-
- <Overview>
-	 <Tooltip>Overview display of waveform</Tooltip>
-	 <Channel>2</Channel>
-	 <Pos>420,193</Pos>
-	 <Size>264,36</Size>
-	 <BgColor>#161D2E</BgColor>
-	 <SignalColor>#5D9BA8</SignalColor>
-	 <MarkerColor>#ffffff</MarkerColor>
-	 <Connection>
-		 <ConfigKey>[Channel2],playposition</ConfigKey>
-		 <EmitOnDownPress>false</EmitOnDownPress>
-	 </Connection>
- </Overview>
-
- <ComboBox>
-  <Pos>109,284</Pos>
-  <Size>164,25</Size>
-  <BgColor>#161D2E</BgColor>
- </ComboBox>
-
- <Search>
-  <Pos>285,284</Pos>
-  <Size>325,25</Size>
-  <BgColor>#161D2E</BgColor>
- </Search>
-
- <TableView>
-  <Pos>109,309</Pos>
-  <Size>582,275</Size>
-  <FgColor>#9C9F5D</FgColor>
-  <BgColor>#161D2E</BgColor>
-  <BgColorRowEven>#161D2E</BgColorRowEven>
-  <BgColorRowUneven>#161D2E</BgColorRowUneven>
-  <BgColorBpmNoConfirm>#161D2E</BgColorBpmNoConfirm>
-  <BgColorBpmConfirm>#161D2E</BgColorBpmConfirm>
- </TableView>
-</skin>
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/sync-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/sync-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/sync-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/sync-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/sync-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/sync-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/sync-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/sync-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/volhandle-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/volhandle-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/volhandle-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/volhandle-on-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/volslider-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/volslider-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/volslider-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/volslider-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/vumeter-off-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/vumeter-off-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/vumeter-off-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/vumeter-off-Ch2.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/vumeter-on-Ch1.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/vumeter-on-Ch1.png differ
Binary files /tmp/OngMa6ps8D/mixxx-1.6.0/src/skins/Natt/vumeter-on-Ch2.png and /tmp/MHYQbGTmm3/mixxx-1.6.1/src/skins/Natt/vumeter-on-Ch2.png differ
diff -Nru mixxx-1.6.0/src/soundsourcem4a.cpp mixxx-1.6.1/src/soundsourcem4a.cpp
--- mixxx-1.6.0/src/soundsourcem4a.cpp	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/soundsourcem4a.cpp	2008-08-24 01:38:08.000000000 -0500
@@ -0,0 +1,138 @@
+/***************************************************************************
+                          soundsourcem4a.cpp  -  mp4/m4a decoder
+                             -------------------
+    copyright            : (C) 2008 by Garth Dahlstrom
+    email                : ironstorm@users.sf.net
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#include "trackinfoobject.h"
+#include "soundsourcem4a.h"
+#include <QtDebug>
+
+#ifdef __WIN__
+#include <io.h>
+#include <fcntl.h>
+#endif
+
+SoundSourceM4A::SoundSourceM4A(QString qFileName) : SoundSource(qFileName)
+{
+   QString mp4FileName;
+   mp4FileName = qFileName;
+   trackId = MP4_INVALID_TRACK_ID;
+   sampleId = 0;
+   mp4file = MP4Read(mp4FileName);
+   if (mp4file == MP4_INVALID_FILE_HANDLE) {
+     qDebug() << "mp4: " << mp4FileName << "could not be opened using the MP4 decoder.";
+     filelength = 0;
+     return;
+   }
+   trackId = MP4FindTrackId(mp4file, 0); // We are only interested in first track for the initial dev iteration
+   if (trackId == MP4_INVALID_TRACK_ID) {
+	   qDebug() << "trackId is invalid.";
+	   return;
+   }
+   channels = 2; //FIXME: hard coded M4A to 2 channels
+
+   filelength = MP4GetTrackNumberOfSamples(mp4file, trackId);
+   SRATE = (filelength * 1024 * 1000) / MP4ConvertFromTrackDuration(mp4file, trackId, MP4GetTrackDuration(mp4file, trackId), MP4_MSECS_TIME_SCALE);
+   SRATE = 44100; // FIXME: Hard-coded SRATE, above formula for SRATE overflows an unsigned long... :(
+   qDebug() << "SRATE:"<< SRATE;
+   qDebug() << "filelength:" << filelength;
+}
+
+SoundSourceM4A::~SoundSourceM4A(){
+   if (mp4file != MP4_INVALID_FILE_HANDLE) {
+      MP4Close(mp4file);
+   }
+}
+
+long SoundSourceM4A::seek(long filepos){
+	Q_ASSERT(filepos % 2 == 0);
+	sampleId = 1 + (2 * filepos / READCHUNKSIZE); // sampleId is 1 indexed
+    qDebug() << "seek sampleId:"<< sampleId << "filepos:"<< filepos;
+}
+
+unsigned SoundSourceM4A::read(volatile unsigned long size, const SAMPLE* destination){
+	Q_ASSERT(size % 2 == 0);
+
+    MP4Timestamp sampleTime;
+    MP4Duration sampleDuration, sampleRenderingOffset;
+    bool isSyncSample;
+
+//    uint32_t sample_size = MP4GetTrackMaxSampleSize(mp4file, trackId);
+//    uint8_t *sample = (uint8_t *)malloc(sample_size);
+    unsigned int this_size = MP4GetTrackMaxSampleSize(mp4file, trackId); // size;
+
+    uint8_t *sample = (uint8_t*) destination;
+
+	bool ret = MP4ReadSample(mp4file,
+			trackId,
+			sampleId,
+			&sample,
+			&this_size,
+			&sampleTime,
+			&sampleDuration,
+			&sampleRenderingOffset,
+			&isSyncSample);
+    qDebug() << "read Track:"<< trackId << "Sample:" << sampleId << "Length:" << this_size << "success:"<<ret;
+	if (ret == false) {
+		qDebug() << "read: Sample read error\n";
+		return -1;
+	}
+    return this_size;
+}
+
+inline long unsigned SoundSourceM4A::length(){
+   return filelength;
+}
+
+int SoundSourceM4A::ParseHeader( TrackInfoObject * Track){
+    QString mp4FileName = Track->getLocation();
+
+    char *value;
+    MP4FileHandle mp4file = MP4Read(mp4FileName);
+    if (mp4file == MP4_INVALID_FILE_HANDLE) {
+     qDebug() << "mp4: " << mp4FileName << "could not be opened using the MP4 decoder.";
+     return ERR;
+   }
+
+   if (MP4GetMetadataName(mp4file, &value) && value != NULL) {
+     Track->setTitle(value);
+     free(value);
+   }
+   if (MP4GetMetadataArtist(mp4file, &value) && value != NULL) {
+     Track->setArtist(value);
+     free(value);
+   }
+   if (MP4GetMetadataComment(mp4file, &value) && value != NULL) {
+     Track->setComment(value);
+     free(value);
+   }
+   u_int16_t bpm = 0;
+   if (MP4GetMetadataTempo(mp4file, &bpm)) {
+       if(bpm > 0) {
+           Track->setBpm(bpm);
+           Track->setBpmConfirm(true);
+       }
+   }
+   Track->setHeaderParsed(true);
+   Track->setType("m4a");
+
+   int trackId = MP4FindTrackId(mp4file, 0); // We are only interested in first track for the initial dev iteration
+   Track->setDuration(MP4ConvertFromTrackDuration(mp4file, trackId, MP4GetTrackDuration(mp4file, trackId), MP4_SECS_TIME_SCALE));
+   Track->setBitrate(MP4GetTrackBitRate(mp4file, trackId)/1000);
+
+   Track->setChannels(2); // FIXME: hard-coded to 2 channels
+
+   MP4Close(mp4file);
+   return OK;
+}
diff -Nru mixxx-1.6.0/src/soundsourcem4a.h mixxx-1.6.1/src/soundsourcem4a.h
--- mixxx-1.6.0/src/soundsourcem4a.h	1969-12-31 18:00:00.000000000 -0600
+++ mixxx-1.6.1/src/soundsourcem4a.h	2008-08-24 01:38:08.000000000 -0500
@@ -0,0 +1,43 @@
+/***************************************************************************
+                          soundsourcemp4.h  -  mp4/m4a decoder
+                             -------------------
+    copyright            : (C) 2008 by Garth Dahlstrom
+    email                : ironstorm@users.sf.net
+ ***************************************************************************/
+
+/***************************************************************************
+ *                                                                         *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.                                   *
+ *                                                                         *
+ ***************************************************************************/
+
+#ifndef SOUNDSOURCEM4A_H
+#define SOUNDSOURCEM4A_H
+
+#include <qstring.h>
+#include "soundsource.h"
+#include "mp4.h"
+
+class TrackInfoObject;
+
+
+class SoundSourceM4A : public SoundSource {
+ public:
+  SoundSourceM4A(QString qFileName);
+  ~SoundSourceM4A();
+  long seek(long);
+  unsigned read(unsigned long size, const SAMPLE*);
+  inline long unsigned length();
+  static int ParseHeader( TrackInfoObject * );
+ private:
+  int channels;
+  int trackId;
+  long sampleId;
+  unsigned long filelength;
+  MP4FileHandle mp4file;
+};
+
+#endif
diff -Nru mixxx-1.6.0/src/soundsourcemp3.cpp mixxx-1.6.1/src/soundsourcemp3.cpp
--- mixxx-1.6.0/src/soundsourcemp3.cpp	2008-08-05 09:35:21.000000000 -0500
+++ mixxx-1.6.1/src/soundsourcemp3.cpp	2008-08-19 17:47:01.000000000 -0500
@@ -95,12 +95,12 @@
         m_iAvgFrameSize = 0;
 
     mad_header_finish (&Header); // This is a macro for nothing.
-    
+
     if (currentframe==0)
         bitrate = 0;
     else
         bitrate = bitrate/currentframe;
-    
+
     framecount = currentframe;
     currentframe = 0;
 /*
@@ -393,14 +393,14 @@
                 *(destination++) = madScale(Synth.pcm.samples[0][i]);
 
             // This is safe because we have Q_ASSERTed that samples_wanted is even.
-            Total_samples_decoded += 2; 
+            Total_samples_decoded += 2;
 
         }
 
         if(Total_samples_decoded >= samples_wanted) {
             if(i < Synth.pcm.length)
                 rest = i;
-            else 
+            else
                 rest = -1;
             return Total_samples_decoded;
         }
@@ -506,12 +506,10 @@
             // http://www.id3.org/id3v2.4.0-frames.txt
             QString s;
             getField(tag,"TIT2",&s); // TIT2 : Title
-            if (s.length()>2)
-                Track->setTitle(s);
+            Track->setTitle(s);
             s="";
             getField(tag,"TPE1",&s); // TPE1 : Artist
-            if (s.length()>2)
-                Track->setArtist(s);
+            Track->setArtist(s);
             s="";
             getField(tag,"TBPM",&s); // TBPM: the bpm
             float bpm = 0;
@@ -533,7 +531,7 @@
         // this closes 'tag' for us
         id3_file_close(fh);
     }
-    
+
     // Get file length. This has to be done by one of these options:
     // 1) looking for the tag named TLEN (above),
     //   -- too buggy to rely on
@@ -547,13 +545,13 @@
     //   -- We do not do this, 7/2008
 
     // Open file, initialize MAD and read beginnning of file
-    
+
     QFile file(location);
     if (!file.open(QIODevice::ReadOnly)) {
         qDebug() << "SSMP3::ParseHeader Open failed:" << location;
         return ERR;
     }
-    
+
     mad_stream Stream;
     mad_header Header;
 
@@ -567,7 +565,7 @@
     unsigned long bytesperframe = 0;
     bool constantbitrate = true;
     int frames = 0;
-    
+
     // Number of bytes to read at a time to determine duration
     const unsigned int READLENGTH = 5000;
     char *inputbuf = new char[READLENGTH];
@@ -592,7 +590,7 @@
         // This preserves skiplen, so if we had a buffer error earlier
         // the skip will occur when we give it more data.
         mad_stream_buffer(&Stream, (unsigned char *) inputbuf, readbytes);
-        
+
         while((Stream.bufend - Stream.this_frame) > 0) {
             if(mad_header_decode(&Header,&Stream) == -1) {
                 if(!MAD_RECOVERABLE(Stream.error)) {
diff -Nru mixxx-1.6.0/src/soundsourceproxy.cpp mixxx-1.6.1/src/soundsourceproxy.cpp
--- mixxx-1.6.0/src/soundsourceproxy.cpp	2008-05-11 10:27:38.000000000 -0500
+++ mixxx-1.6.1/src/soundsourceproxy.cpp	2008-08-24 01:38:08.000000000 -0500
@@ -19,6 +19,9 @@
 #include "trackinfoobject.h"
 #include "soundsourceproxy.h"
 #include "soundsourcemp3.h"
+#ifdef __M4A__
+	#include "soundsourcem4a.h"
+#endif
 #include "soundsourceoggvorbis.h"
 #ifdef __SNDFILE__
 #include "soundsourcesndfile.h"
@@ -42,6 +45,10 @@
 #endif
     if (qFilename.toLower().endsWith(".mp3"))
         m_pSoundSource = new SoundSourceMp3(qFilename);
+#ifdef __M4A__
+    else if (qFilename.toLower().endsWith(".m4a"))
+        m_pSoundSource = new SoundSourceM4A(qFilename);
+#endif
     else if (qFilename.toLower().endsWith(".ogg"))
         m_pSoundSource = new SoundSourceOggVorbis(qFilename);
     else
@@ -64,6 +71,10 @@
 
     if (qFilename.toLower().endsWith(".mp3"))
 	m_pSoundSource = new SoundSourceMp3(qFilename);
+#ifdef __M4A__
+    else if (qFilename.toLower().endsWith(".m4a"))
+	m_pSoundSource = new SoundSourceM4A(qFilename);
+#endif
     else if (qFilename.toLower().endsWith(".ogg"))
 	m_pSoundSource = new SoundSourceOggVorbis(qFilename);
     else
@@ -108,6 +119,10 @@
 #endif
     if (qFilename.toLower().endsWith(".mp3"))
 	return SoundSourceMp3::ParseHeader(p);
+#ifdef __M4A__
+    else if (qFilename.toLower().endsWith(".m4a"))
+	return SoundSourceM4A::ParseHeader(p);
+#endif
     else if (qFilename.toLower().endsWith(".ogg"))
 	return SoundSourceOggVorbis::ParseHeader(p);
     else if (qFilename.toLower().endsWith(".wav") || qFilename.toLower().endsWith(".aif") ||
diff -Nru mixxx-1.6.0/src/soundsourcesndfile.cpp mixxx-1.6.1/src/soundsourcesndfile.cpp
--- mixxx-1.6.0/src/soundsourcesndfile.cpp	2008-03-27 00:04:43.000000000 -0500
+++ mixxx-1.6.1/src/soundsourcesndfile.cpp	2008-08-15 22:09:05.000000000 -0500
@@ -132,6 +132,16 @@
     Track->setSampleRate(info.samplerate);
     Track->setChannels(info.channels);
 
+    const char *string;
+    string = sf_get_string(fh, SF_STR_ARTIST);
+//    qDebug() << location << "SF_STR_ARTIST" << string;
+    if(string && strlen(string))
+        Track->setArtist(string);
+    string = sf_get_string(fh, SF_STR_TITLE);
+    if(string && strlen(string))
+        Track->setTitle(string);
+//    qDebug() << location << "SF_STR_TITLE" << string;
+
     sf_close( fh );
     return OK;
 }
diff -Nru mixxx-1.6.0/src/track.cpp mixxx-1.6.1/src/track.cpp
--- mixxx-1.6.0/src/track.cpp	2008-08-04 14:00:57.000000000 -0500
+++ mixxx-1.6.1/src/track.cpp	2008-08-18 00:08:49.000000000 -0500
@@ -10,6 +10,10 @@
 //
 //
 
+#ifdef __WIN__
+#include <windows.h> // for Sleep()  on Windows
+#endif
+
 #include "track.h"
 #include "trackinfoobject.h"
 #include "trackcollection.h"
@@ -1340,26 +1344,49 @@
 /** Runs the BPM detection on every track in the TrackCollection that doesn't already have a BPM. */
 void Track::slotBatchBPMDetection()
 {
-/*
-    //Unfinished/WIP - Albert Jan 31/08
-    //General idea should work more or less, I suppose...
+    /* Possible use cases:
+       1) BPM detect whole library
+       2) BPM detect play queue or a specific play list
+    */
+
+    // TODO: Abort batch processing if either deck is playing and tell user...
+    int processed = 0;
+    for (int i =0; i < m_pTrackCollection->getSize(); i++) {
+      TrackInfoObject* cur_track = m_pTrackCollection->getTrack(i);
+
+      if (!cur_track->isValid()) continue; // Skip invalid track objects
+
+      // Batch Process Wave Summary
+      if (cur_track->getWaveSummary() == NULL || cur_track->getWaveSummary()->size() == 0) {
+        processed++;
+        // TODO: convert the qDebug statement to a status dialog box.
+        qDebug() << "Track #"<< i << "= Batch job #"<< processed << ":" << cur_track->getTitle() << "by" << cur_track->getArtist() << "has WaveSummary: false -- has BPM value:" << (cur_track->getBpm() > 0) << " ("<< m_pBpmDetector->queueCount() << "tracks in the BPM Detect Queue )";
+        // Open sound file
+        SoundSourceProxy * pSoundSource = new SoundSourceProxy(cur_track);
+	m_pWaveSummary->waveformSummaryGen(cur_track, pSoundSource, false);
+        delete pSoundSource;
+        pSoundSource = NULL;
+
+        // Batch Process BPM
+        if (cur_track->getBpm() == 0) {
+          cur_track->sendToBpmQueue();
+        }
+      }
 
-    //Iterate over each TrackInfoObject stored in m_pTrackCollection
-    TrackInfoObject* cur_track;
-    QListIterator<TrackCollection*> it(m_qPlaylists);
-    TrackPlaylist* current;
-    while (it.hasNext())
-    {
-        current = it.next();
-        if (current->getListName()==qName)
-            return current;
+      // FIXME: remove the song count limitation when BPM detect doesn't crash...
+      if (processed == 100) { qDebug() << "----- BPM detection/Waveform generation batch processing limit of " << processed << " songs has been reached. Rerun to detect more songs."; break; }
     }
-
-    //(TIO -> TrackInfoObject)
-    //Check to see if the TIO already has a non-zero BPM
-
-    //If not, run TIO->sendToBpmQueue().
-
-*/
+    // Block the UI thread until BPM detection queue is cleared.
+    while (m_pBpmDetector->queueCount()) {
+      // TODO: convert the qDebug statement to a status dialog box.
+      qDebug() << "---- waiting for BPM detection queue to empty... " << m_pBpmDetector->queueCount() << " to go.";
+#ifdef __WIN__
+      Sleep(3000);
+#else
+      sleep(3);
+#endif
+    }
+    // Save the track database to disk...
+    writeXML(m_pConfig->getValueString(ConfigKey("[Playlist]","Listfile")));
 }
 
diff -Nru mixxx-1.6.0/src/track.h mixxx-1.6.1/src/track.h
--- mixxx-1.6.0/src/track.h	2008-07-13 22:21:19.000000000 -0500
+++ mixxx-1.6.1/src/track.h	2008-08-14 00:24:33.000000000 -0500
@@ -155,12 +155,12 @@
     void slotSendToPlaylist(TrackPlaylist* playlist, TrackInfoObject* pTrackInfoObject);
     /** Slot for sending a track to a playlist */
     void slotSendToPlaylist(TrackPlaylist* playlist, QString filename);
-    
+
     /** Slot for showing a particular playlist in the track table view */
-    void slotShowPlaylist(TrackPlaylist* playlist); 
+    void slotShowPlaylist(TrackPlaylist* playlist);
     /** Runs the BPM detection on every track in the TrackCollection */
     void slotBatchBPMDetection();
-    
+
 signals:
     /** A new track has been loaded in player 1 */
     void newTrackPlayer1(TrackInfoObject *);
@@ -173,7 +173,7 @@
 private:
     /** Returns pointer to playlist by the given name */
     TrackPlaylist *getPlaylist(QString qName);
-    
+
     /** Creates the promo tracks playlist dynamically */
     void initPromoTracks();
     /** Load the special promo track metadata from an XML file */
@@ -191,7 +191,7 @@
     TrackPlaylist m_qPlayqueuePlaylist;
     /** The promo tracks playlist */
     TrackPlaylist m_qPromoPlaylist;
-    
+
     /** Data Model for Library */
     WTrackTableModel *m_pLibraryModel;
     /** Data Model for Playqueue*/
@@ -206,7 +206,7 @@
     WTrackTableModel *m_pPlaylistModel;
     /** Data Model containing the list of playlists */
     WPlaylistListModel *m_pPlaylistListModel;
-    
+
     /** Pointer to playlist for which a popup menu is currently displayed */
     TrackPlaylist *m_pActivePopupPlaylist;
     /** Pointer to TrackInfoObject for which a popup menu is currently displayed */
@@ -249,8 +249,8 @@
     unsigned int m_iPlayqueueIdx;
     /** Provides access to the config keys */
     ConfigObject<ConfigValue> *m_pConfig;
-    
-    int savedRowPosition;    
+
+    int savedRowPosition;
 };
 
 #endif
diff -Nru mixxx-1.6.0/src/trackinfoobject.cpp mixxx-1.6.1/src/trackinfoobject.cpp
--- mixxx-1.6.0/src/trackinfoobject.cpp	2008-08-04 13:12:52.000000000 -0500
+++ mixxx-1.6.1/src/trackinfoobject.cpp	2008-08-19 17:47:01.000000000 -0500
@@ -342,33 +342,18 @@
 
     if (m_sFilename.find('-') != -1)
     {
-        m_sArtist = m_sFilename.section('-',0,0); // Get the first part
+        m_sArtist = m_sFilename.section('-',0,0).trimmed(); // Get the first part
         m_sTitle = m_sFilename.section('-',1,1); // Get the second part
-        m_sTitle = m_sTitle.section('.',0,-2); // Remove the ending
-        m_sType = m_sFilename.section('.',-1); // Get the ending
+        m_sTitle = m_sTitle.section('.',0,-2).trimmed(); // Remove the ending
     }
     else
     {
-        m_sTitle = m_sFilename.section('.',0,-2); // Remove the ending;
-        m_sType = m_sFilename.section('.',-1); // Get the ending
+        m_sTitle = m_sFilename.section('.',0,-2).trimmed(); // Remove the ending;
+        m_sType = m_sFilename.section('.',-1).trimmed(); // Get the ending
     }
 
-    // Remove spaces from start and end of title and artist
-    while (m_sArtist.startsWith(" "))
-        m_sArtist = m_sArtist.right(m_sArtist.length()-1);
-    while (m_sArtist.endsWith(" "))
-        m_sArtist = m_sArtist.left(m_sArtist.length()-1);
-    while (m_sTitle.startsWith(" "))
-        m_sTitle = m_sTitle.right(m_sTitle.length()-1);
-    while (m_sTitle.endsWith(" "))
-        m_sTitle = m_sTitle.left(m_sTitle.length()-1);
-
-
-    // Sort out obviously wrong parsings:
-    if ((m_sArtist.length() < 3) || (m_sTitle.length() < 3))
-    {
-        m_sTitle = m_sFilename.section('.',0,-2);
-        m_sArtist = "";
+    if (m_sTitle.length() == 0) {
+        m_sTitle = m_sFilename.section('.',0,-2).trimmed();
     }
 
     // Find the length:
@@ -378,7 +363,7 @@
     m_sComment = QString("");
 
     // Find the type
-    m_sType = m_sFilename.section(".",-1).lower();
+    m_sType = m_sFilename.section(".",-1).lower().trimmed();
 
     m_qMutex.unlock();
 }
@@ -544,7 +529,7 @@
 QString TrackInfoObject::getInfo()  const
 {
     m_qMutex.lock();
-    QString artist = m_sArtist == "" ? "" : m_sArtist + ", ";
+    QString artist = m_sArtist.trimmed() == "" ? "" : m_sArtist + ", ";
     QString sInfo = artist + m_sTitle;
     m_qMutex.unlock();
 
@@ -587,7 +572,7 @@
 void TrackInfoObject::setTitle(QString s)
 {
     m_qMutex.lock();
-    m_sTitle = s;
+    m_sTitle = s.trimmed();
     m_qMutex.unlock();
 /*
     if (m_pTableItemTitle)
@@ -610,7 +595,7 @@
 void TrackInfoObject::setArtist(QString s)
 {
     m_qMutex.lock();
-    m_sArtist = s;
+    m_sArtist = s.trimmed();
     m_qMutex.unlock();
 /*
     if (m_pTableItemArtist)
@@ -899,14 +884,14 @@
     m_qMutex.unlock();
 }
 
-void TrackInfoObject::setWaveSummary(Q3MemArray<char> * pWave, Q3ValueList<long> * pSegmentation)
+void TrackInfoObject::setWaveSummary(Q3MemArray<char> * pWave, Q3ValueList<long> * pSegmentation, bool updateUI)
 {
     m_qMutex.lock();
     m_pWave = pWave;
     m_pSegmentation = pSegmentation;
     m_qMutex.unlock();
 
-    setOverviewWidget(m_pOverviewWidget);
+    if (updateUI) setOverviewWidget(m_pOverviewWidget);
 }
 
 //TODO: Get rid of these crappy methods. getNext() and getPrev() are terrible for modularity
diff -Nru mixxx-1.6.0/src/trackinfoobject.h mixxx-1.6.1/src/trackinfoobject.h
--- mixxx-1.6.0/src/trackinfoobject.h	2008-07-13 22:21:19.000000000 -0500
+++ mixxx-1.6.1/src/trackinfoobject.h	2008-08-14 00:23:43.000000000 -0500
@@ -52,7 +52,7 @@
     /** Creates a new track given information from the xml file. */
     TrackInfoObject(const QDomNode &, BpmDetector *bpmDetector);
     ~TrackInfoObject();
-    /** Returns true if the object contains valid information */    
+    /** Returns true if the object contains valid information */
     bool isValid() const;
     int parse();
     /** Checks if the file given in m_sFilename really exists on the disc, and
@@ -165,8 +165,9 @@
     void setVisualResampleRate(double dVisualResampleRate);
     double getVisualResampleRate();
 
-    /** Set pointer to waveform summary */
-    void setWaveSummary(Q3MemArray<char> *pWave, Q3ValueList<long> *pSegmentation);
+    /** Set pointer to waveform summary -- updates UI by default */
+    void setWaveSummary(Q3MemArray<char> *pWave, Q3ValueList<long> *pSegmentation, bool updateUI = true);
+
     /** Returns a pointer to waveform summary */
     Q3MemArray<char> *getWaveSummary();
     /** Returns a pointer to segmentation summary */
diff -Nru mixxx-1.6.0/src/waveform/waveformrenderbeat.cpp mixxx-1.6.1/src/waveform/waveformrenderbeat.cpp
--- mixxx-1.6.0/src/waveform/waveformrenderbeat.cpp	2008-07-21 23:19:17.000000000 -0500
+++ mixxx-1.6.1/src/waveform/waveformrenderbeat.cpp	2008-09-16 21:12:45.000000000 -0500
@@ -93,7 +93,7 @@
 }
 
 
-void WaveformRenderBeat::draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double dPlayPos) {
+void WaveformRenderBeat::draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double dPlayPos, double rateAdjust) {
 
     if(m_dBpm == -1 || m_dBpm == 0)
         return;
@@ -125,13 +125,14 @@
         m_dBeatLength = 60.0 * m_iSampleRate / m_dBpm;
     }
 
-    const int oversample = m_pParent->getSubpixelsPerPixel();
+    double subpixelsPerPixel = m_pParent->getSubpixelsPerPixel()*(1.0+rateAdjust);
+    const int oversample = (int)subpixelsPerPixel;
 
     pPainter->save();
-    pPainter->scale(1.0/oversample,1.0);
+    pPainter->scale(1.0/subpixelsPerPixel,1.0);
     pPainter->setPen(colorMarks);
     
-    double subpixelWidth = m_iWidth * oversample;
+    double subpixelWidth = m_iWidth * subpixelsPerPixel;
     double subpixelHalfWidth = subpixelWidth / 2.0;
     double halfw = m_iWidth/2;
     double halfh = m_iHeight/2;
@@ -140,8 +141,8 @@
     iCurPos = iCurPos >> 1;
 
     // basePos and endPos are in samples
-    double basePos = iCurPos - m_dSamplesPerPixel*halfw;
-    double endPos = basePos + m_iWidth*m_dSamplesPerPixel;
+    double basePos = iCurPos - m_dSamplesPerPixel*halfw*(1.0+rateAdjust);
+    double endPos = basePos + m_iWidth*m_dSamplesPerPixel*(1.0+rateAdjust);
 
     // snap to the first beat
     double curPos = ceilf(basePos/m_dBeatLength)*m_dBeatLength;
diff -Nru mixxx-1.6.0/src/waveform/waveformrenderbeat.h mixxx-1.6.1/src/waveform/waveformrenderbeat.h
--- mixxx-1.6.0/src/waveform/waveformrenderbeat.h	2008-07-21 23:19:17.000000000 -0500
+++ mixxx-1.6.1/src/waveform/waveformrenderbeat.h	2008-09-16 21:12:45.000000000 -0500
@@ -22,7 +22,7 @@
     void resize(int w, int h);
     void setup(QDomNode node);
     WaveformRenderBeat(const char *group, WaveformRenderer *parent);
-    void draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double playPos);
+    void draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double playPos, double rateAdjust);
     void newTrack(TrackInfoObject *pTrack);
 
 public slots:
diff -Nru mixxx-1.6.0/src/waveform/waveformrenderer.cpp mixxx-1.6.1/src/waveform/waveformrenderer.cpp
--- mixxx-1.6.0/src/waveform/waveformrenderer.cpp	2008-07-21 23:19:17.000000000 -0500
+++ mixxx-1.6.1/src/waveform/waveformrenderer.cpp	2008-09-16 21:12:45.000000000 -0500
@@ -54,6 +54,17 @@
     m_pRenderCue = new WaveformRenderMark(group, ConfigKey(group, "cue_point"), this);
 
     m_pCOVisualResample = new ControlObject(ConfigKey(group, "VisualResample"));
+
+    m_pRate = new ControlObjectThreadMain(ControlObject::getControl(ConfigKey(group, "rate")));
+    if(m_pRate != NULL) {
+        connect(m_pRate, SIGNAL(valueChanged(double)), this, SLOT(slotUpdateRate(double)));
+    }
+
+    m_pRateRange = new ControlObjectThreadMain(ControlObject::getControl(ConfigKey(group, "rateRange")));
+    if(m_pRateRange != NULL) {
+        connect(m_pRateRange, SIGNAL(valueChanged(double)), this, SLOT(slotUpdateRateRange(double)));
+    }
+
 }
 
 
@@ -62,6 +73,14 @@
         delete m_pCOVisualResample;
     m_pCOVisualResample = NULL;
 
+    if(m_pRate)
+        delete m_pRate;
+    m_pRate = NULL;
+    
+    if(m_pRateRange)
+        delete m_pRateRange;
+    m_pRateRange = NULL;
+
     if(m_pPlayPos)
         delete m_pPlayPos;
     m_pPlayPos = NULL;
@@ -83,6 +102,15 @@
     m_iPlayPosTime = clock();
 }
 
+void WaveformRenderer::slotUpdateRate(double v) {
+    m_dRate = v;
+}
+
+void WaveformRenderer::slotUpdateRateRange(double v) {
+    m_dRateRange = v;
+}
+
+
 void WaveformRenderer::resize(int w, int h) {
     m_iWidth = w;
     m_iHeight = h;
@@ -271,7 +299,7 @@
     
 
 
-void WaveformRenderer::drawSignalLines(QPainter *pPainter,double playpos) {
+void WaveformRenderer::drawSignalLines(QPainter *pPainter,double playpos, double rateAdjust) {
     
     if(m_pSampleBuffer == NULL) {
         return;
@@ -287,9 +315,18 @@
 
     pPainter->save();
 
-    int subpixelWidth = m_iWidth * m_iSubpixelsPerPixel;
+    double subpixelsPerPixel = m_iSubpixelsPerPixel * (1.0 + rateAdjust);
+    
+    int subpixelWidth = int(m_iWidth * subpixelsPerPixel);
+
+    pPainter->scale(1.0/subpixelsPerPixel,m_iHeight*0.40);
 
-    pPainter->scale(1.0/float(m_iSubpixelsPerPixel),m_iHeight*0.40);
+    // If the array is not large enough, expand it.
+    // Amortize the cost of this by requesting a factor of 2 more.
+    if(m_lines.size() < subpixelWidth) {
+        m_lines.resize(2*subpixelWidth);
+    }
+    
     int halfw = subpixelWidth/2;
     for(int i=0;i<subpixelWidth;i++) {
         // Start at curPos minus half the waveform viewer
@@ -303,7 +340,8 @@
         }
     }
 
-    pPainter->drawLines(m_lines);
+    // Only draw lines that we have provided
+    pPainter->drawLines(m_lines.data(), subpixelWidth);
 
     pPainter->restore();
 }
@@ -406,7 +444,7 @@
     }
     */
 
-    double playpos = m_dPlayPos + playposadjust;
+    double playpos = m_dPlayPos;
     
     if(m_bRepaintBackground) {
         generateBackgroundPixmap();
@@ -426,19 +464,23 @@
 
     // Translate our coordinate frame from (0,0) at top left
     // to (0,0) at left, center. All the subrenderers expect this.
-    
     pPainter->translate(0.0,m_iHeight/2.0);
+
     // Now scale so that positive-y points up.
     pPainter->scale(1.0,-1.0);
-
+    
     // Draw the center horizontal line under the signal.
     pPainter->drawLine(QLine(0,0,m_iWidth,0));
+
+    // Limit our rate adjustment to < 99%, "Bad Things" might happen otherwise.
+    double rateAdjust = math_min(0.99, m_dRate * m_dRateRange);
     
-    drawSignalLines(pPainter,playpos);
+    // Draw the signal
+    drawSignalLines(pPainter,playpos,rateAdjust);
 
     // Draw various markers.
-    m_pRenderBeat->draw(pPainter,pEvent, m_pSampleBuffer, playpos);
-    m_pRenderCue->draw(pPainter,pEvent, m_pSampleBuffer, playpos);
+    m_pRenderBeat->draw(pPainter,pEvent, m_pSampleBuffer, playpos, rateAdjust);
+    m_pRenderCue->draw(pPainter,pEvent, m_pSampleBuffer, playpos, rateAdjust);
     
     pPainter->setPen(colorMarker);
     
diff -Nru mixxx-1.6.0/src/waveform/waveformrenderer.h mixxx-1.6.1/src/waveform/waveformrenderer.h
--- mixxx-1.6.0/src/waveform/waveformrenderer.h	2008-07-21 23:19:17.000000000 -0500
+++ mixxx-1.6.1/src/waveform/waveformrenderer.h	2008-09-16 21:12:45.000000000 -0500
@@ -24,7 +24,7 @@
 
     void resize(int w, int h);
     void draw(QPainter* pPainter, QPaintEvent *pEvent);
-    void drawSignalLines(QPainter*, double playpos);
+    void drawSignalLines(QPainter*, double playpos, double rateAdjust);
     void drawSignalPixmap(QPainter* p);
     void newTrack(TrackInfoObject *pTrack);
     void setup(QDomNode node);
@@ -33,6 +33,8 @@
     int getPixelsPerSecond();
 public slots:
     void slotUpdatePlayPos(double playpos);
+    void slotUpdateRate(double rate);
+    void slotUpdateRateRange(double rate);
 
 private:
     void generateBackgroundPixmap();
@@ -43,7 +45,7 @@
     int m_iNumSamples;
 
     int m_iPlayPosTime, m_iPlayPosTimeOld;
-    double m_dPlayPos, m_dPlayPosOld;
+    double m_dPlayPos, m_dPlayPosOld, m_dRate, m_dRateRange;
 
     QVector<float> *m_pSampleBuffer;
     QVector<QLineF> m_lines;
@@ -53,7 +55,9 @@
     QImage m_pImage;
 
     ControlObjectThreadMain *m_pPlayPos;
-
+    ControlObjectThreadMain *m_pRate;
+    ControlObjectThreadMain *m_pRateRange;
+    
     ControlObject *m_pCOVisualResample;
 
     WaveformRenderBeat *m_pRenderBeat;
diff -Nru mixxx-1.6.0/src/waveform/waveformrendermark.cpp mixxx-1.6.1/src/waveform/waveformrendermark.cpp
--- mixxx-1.6.0/src/waveform/waveformrendermark.cpp	2008-07-21 23:19:17.000000000 -0500
+++ mixxx-1.6.1/src/waveform/waveformrendermark.cpp	2008-09-16 21:12:45.000000000 -0500
@@ -79,7 +79,7 @@
 }
 
 
-void WaveformRenderMark::draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double dPlayPos) {
+void WaveformRenderMark::draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double dPlayPos, double rateAdjust) {
 
     if(m_iSampleRate == -1 || m_iSampleRate == 0 || m_iNumSamples == 0)
         return;
@@ -87,14 +87,14 @@
     // necessary?
     if(buffer == NULL)
         return;
-    
-    const int oversample = m_pParent->getSubpixelsPerPixel();
+
+    double subpixelsPerPixel = m_pParent->getSubpixelsPerPixel()*(1.0+rateAdjust);
 
     pPainter->save();
-    pPainter->scale(1.0/oversample,1.0);
+    pPainter->scale(1.0/subpixelsPerPixel,1.0);
     pPainter->setPen(markColor);
     
-    double subpixelWidth = m_iWidth * oversample;
+    double subpixelWidth = m_iWidth * subpixelsPerPixel;
     double subpixelHalfWidth = subpixelWidth / 2.0;
     double halfh = m_iHeight/2;
     
diff -Nru mixxx-1.6.0/src/waveform/waveformrendermark.h mixxx-1.6.1/src/waveform/waveformrendermark.h
--- mixxx-1.6.0/src/waveform/waveformrendermark.h	2008-07-21 23:19:17.000000000 -0500
+++ mixxx-1.6.1/src/waveform/waveformrendermark.h	2008-09-16 21:12:45.000000000 -0500
@@ -24,7 +24,7 @@
     void resize(int w, int h);
     void setup(QDomNode node);
     WaveformRenderMark(const char *group, ConfigKey key, WaveformRenderer *parent);
-    void draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double playPos);
+    void draw(QPainter *pPainter, QPaintEvent *event, QVector<float> *buffer, double playPos, double rateAdjust);
     void newTrack(TrackInfoObject *pTrack);
 
 public slots:
diff -Nru mixxx-1.6.0/src/wavesummary.cpp mixxx-1.6.1/src/wavesummary.cpp
--- mixxx-1.6.0/src/wavesummary.cpp	2008-07-30 11:53:19.000000000 -0500
+++ mixxx-1.6.1/src/wavesummary.cpp	2008-09-07 04:36:04.000000000 -0500
@@ -87,7 +87,7 @@
         // Check if preview has been generated
         Q3MemArray<char> *p = pTrackInfoObject->getWaveSummary();
         if (!p || p->size()==0) {
-            waveformSummaryGen(pTrackInfoObject, pSoundSource);            
+            waveformSummaryGen(pTrackInfoObject, pSoundSource);
         }
 
         if(pTrackInfoObject->getVisualWaveform() == NULL) {
@@ -101,9 +101,8 @@
     qDebug() << "WaveSummary::run() exiting";
 }
 
+void WaveSummary::waveformSummaryGen(TrackInfoObject *pTrackInfoObject, SoundSourceProxy *pSoundSource, bool updateUI) {
 
-void WaveSummary::waveformSummaryGen(TrackInfoObject *pTrackInfoObject, SoundSourceProxy *pSoundSource) {
-    
     // Allocate temp buffer
     SAMPLE * pBuffer = new SAMPLE[kiBlockSize*2];
 
@@ -157,10 +156,10 @@
             liPos += pSoundSource->read(kiBlockSize, pBuffer);
         }
 
-    pTrackInfoObject->setWaveSummary(pData, 0);
+    pTrackInfoObject->setWaveSummary(pData, 0, updateUI);
 
     delete [] pBuffer;
-    pBuffer = NULL;            
+    pBuffer = NULL;
 
     qDebug() << "WaveSummary generation successful for " << pTrackInfoObject->getFilename();
 
@@ -177,18 +176,19 @@
     // => N = F / (MZ)
 
     int numSamples = pSoundSource->length();
+    if (numSamples <= 0) return;
     int f = pSoundSource->getSrate();
     double mz = pTrackInfoObject->getVisualResampleRate();
     double n = double(f) / mz;
-    
+
     int samplesPerDownsample = n;
     int numDownsamples = numSamples / samplesPerDownsample;
-    
+
     if(numDownsamples % 2 != 0)
         numDownsamples++;
 
     // Downsample from curSamples -> numDownsamples
-    
+
     QVector<float> *downsample = new QVector<float>(numDownsamples);
     float *downsampleVector = downsample->data();
     int i,j;
@@ -202,7 +202,7 @@
     // Allow the visual waveform to display this before we've populated it so
     // that it displays the wave as we work.
     pTrackInfoObject->setVisualWaveform(downsample);
-    
+
     qDebug() << "WaveSummary: f " << f << " samplesPerDownsample: " << samplesPerDownsample << " downsamples " << numDownsamples << " from " << numSamples;
 
     // readLen and bufRead need to be divisible by 2
@@ -249,8 +249,14 @@
             samplesAvailable -= bufRead;
         }
         int samplesRead = pSoundSource->read(readLen, pBuffer);
-        samplesAvailable += samplesRead;
-        filePos += samplesRead;
+        if(samplesRead) {
+            samplesAvailable += samplesRead;
+            filePos += samplesRead;
+        } else {
+            /* early EOF return, break out of the loop. */
+            qDebug() << "Waveform thread: got early EOF.";
+            filePos = numSamples;
+        }
     }
 
     qDebug() << "WaveSummary :: Waveform downsampling finished.";
diff -Nru mixxx-1.6.0/src/wavesummary.h mixxx-1.6.1/src/wavesummary.h
--- mixxx-1.6.0/src/wavesummary.h	2008-07-30 09:41:17.000000000 -0500
+++ mixxx-1.6.1/src/wavesummary.h	2008-08-14 00:23:43.000000000 -0500
@@ -52,7 +52,10 @@
     ~WaveSummary();
     /** Puts an TrackInfoObject into the queue of summary generation. Thread safe, blocking. */
     void enqueue(TrackInfoObject *pTrackInfoObject);
-	void extractBeat(TrackInfoObject *pTrackInfoObject);
+    void extractBeat(TrackInfoObject *pTrackInfoObject);
+
+    /** Generate a waveform summary data for the given track */
+    void waveformSummaryGen(TrackInfoObject *pTrackInfoObject, SoundSourceProxy *pSoundSource, bool updateUI = true);
 
 private:
     /** Main thread loop */
@@ -61,8 +64,7 @@
     void stop();
     /** Generate a visual waveform for the given track */
     void visualWaveformGen(TrackInfoObject *pTrackInfoObject, SoundSourceProxy *pSoundSource);
-    /** Generate a waveform summary for the given track */
-    void waveformSummaryGen(TrackInfoObject *pTrackInfoObject, SoundSourceProxy *pSoundSource);
+
     /** Queue holding files to generate a summary for */
     Q3PtrQueue<TrackInfoObject> m_qQueue;
     /** Mutex controlling access to m_qQueue */
@@ -71,7 +73,7 @@
     QWaitCondition m_qWait;
     /** Pointer to config object **/
     ConfigObject<ConfigValue> *m_Config;
-    
+
     ControlObjectThread *m_pControlVisualResample;
 
     bool m_bShouldExit;
diff -Nru mixxx-1.6.0/src/wvisualsimple.cpp mixxx-1.6.1/src/wvisualsimple.cpp
--- mixxx-1.6.0/src/wvisualsimple.cpp	2008-07-13 22:21:19.000000000 -0500
+++ mixxx-1.6.1/src/wvisualsimple.cpp	2008-09-15 13:58:32.000000000 -0500
@@ -52,7 +52,7 @@
   if (event->mimeData()->hasUrls()) {
     QList<QUrl> urls(event->mimeData()->urls());
     QUrl url = urls.first();
-    QString name = url.path();
+    QString name = url.toLocalFile();
 
     event->accept();
     emit(trackDropped(name));
@@ -96,7 +96,7 @@
     colorMarker = WSkinColor::getCorrectColor(colorMarker);
 }
 
-void WVisualSimple::slotNewTrack()
+void WVisualSimple::slotNewTrack(TrackInfoObject* track)
 {
 }
 
@@ -152,3 +152,6 @@
         paint.drawLine(m_qMarkerPos2, p2);
     }
 }
+
+
+
diff -Nru mixxx-1.6.0/src/wvisualsimple.h mixxx-1.6.1/src/wvisualsimple.h
--- mixxx-1.6.0/src/wvisualsimple.h	2008-07-13 22:21:19.000000000 -0500
+++ mixxx-1.6.1/src/wvisualsimple.h	2008-09-15 13:58:32.000000000 -0500
@@ -32,6 +32,7 @@
   */
 
 class WaveformRenderer;
+class TrackInfoObject;
 
 class WVisualSimple : public WWidget
 {
@@ -52,7 +53,7 @@
 
 public slots:
     void setValue(double) {};
-    void slotNewTrack();
+    void slotNewTrack(TrackInfoObject* track);
     
 protected:
     int m_iStartPosX, m_iValue;