Merge branch 'immediate-removal' into GLSL-master

Conflicts:
	Source/Core/VideoCommon/Src/PixelShaderGen.cpp
	Source/Plugins/Plugin_VideoSoftware/Src/SWRenderer.cpp

immediate-removal is a new created branch seperated from master but reverted the revert of immediate-removal
so we get less conflicts by merging

17 files changed, 788 insertions, 488 deletions

diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj b/Source/Core/AudioCommon/AudioCommon.vcxproj
index 0c5f256c19..abb5dbf505 100644
--- a/Source/Core/AudioCommon/AudioCommon.vcxproj
+++ b/Source/Core/AudioCommon/AudioCommon.vcxproj
@@ -109,78 +109,89 @@
   <PropertyGroup Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'" />
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
     </ClCompile>
     <Link>
       <GenerateDebugInformation>true</GenerateDebugInformation>
     </Link>
-    <Lib />
+    <Lib>
+      <AdditionalDependencies>SoundTouchD.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
+      <AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+    </Lib>
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
     <ClCompile>
-      <AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
     </ClCompile>
     <Link>
       <GenerateDebugInformation>true</GenerateDebugInformation>
     </Link>
-    <Lib />
+    <Lib>
+      <AdditionalDependencies>SoundTouchD.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
+      <AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+    </Lib>
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
     </ClCompile>
     <Link>
       <GenerateDebugInformation>true</GenerateDebugInformation>
       <EnableCOMDATFolding>true</EnableCOMDATFolding>
       <OptimizeReferences>true</OptimizeReferences>
     </Link>
-    <Lib />
+    <Lib>
+      <AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
+      <AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+    </Lib>
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='DebugFast|Win32'">
     <ClCompile>
-      <AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
     </ClCompile>
     <Link>
       <GenerateDebugInformation>true</GenerateDebugInformation>
       <EnableCOMDATFolding>true</EnableCOMDATFolding>
       <OptimizeReferences>true</OptimizeReferences>
     </Link>
-    <Lib />
+    <Lib>
+      <AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
+      <AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+    </Lib>
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
     <ClCompile>
-      <AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
     </ClCompile>
     <Link>
       <GenerateDebugInformation>true</GenerateDebugInformation>
       <EnableCOMDATFolding>true</EnableCOMDATFolding>
       <OptimizeReferences>true</OptimizeReferences>
     </Link>
-    <Lib />
+    <Lib>
+      <AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
+      <AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+    </Lib>
   </ItemDefinitionGroup>
   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'">
     <ClCompile>
-      <AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
+      <AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
     </ClCompile>
     <Link>
       <GenerateDebugInformation>true</GenerateDebugInformation>
       <EnableCOMDATFolding>true</EnableCOMDATFolding>
       <OptimizeReferences>true</OptimizeReferences>
     </Link>
-    <Lib />
+    <Lib>
+      <AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
+      <AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
+    </Lib>
   </ItemDefinitionGroup>
   <ItemGroup>
-    <ClCompile Include="Src\aldlist.cpp">
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|Win32'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|x64'">true</ExcludedFromBuild>
-    </ClCompile>
+    <ClCompile Include="Src\aldlist.cpp" />
     <ClCompile Include="Src\AOSoundStream.cpp" />
     <ClCompile Include="Src\AudioCommon.cpp" />
-    <ClCompile Include="Src\AudioCommonConfig.cpp" />
+    <ClCompile Include="Src\DPL2Decoder.cpp" />
     <ClCompile Include="Src\DSoundStream.cpp" />
     <ClCompile Include="Src\Mixer.cpp" />
     <ClCompile Include="Src\NullSoundStream.cpp" />
@@ -189,17 +200,10 @@
     <ClCompile Include="Src\XAudio2Stream.cpp" />
   </ItemGroup>
   <ItemGroup>
-    <ClInclude Include="Src\aldlist.h">
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|Win32'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</ExcludedFromBuild>
-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|x64'">true</ExcludedFromBuild>
-    </ClInclude>
+    <ClInclude Include="Src\aldlist.h" />
     <ClInclude Include="Src\AOSoundStream.h" />
     <ClInclude Include="Src\AudioCommon.h" />
-    <ClInclude Include="Src\AudioCommonConfig.h" />
+    <ClInclude Include="Src\DPL2Decoder.h" />
     <ClInclude Include="Src\DSoundStream.h" />
     <ClInclude Include="Src\Mixer.h" />
     <ClInclude Include="Src\NullSoundStream.h" />
@@ -212,6 +216,14 @@
     <None Include="CMakeLists.txt" />
   </ItemGroup>
   <ItemGroup>
+    <ProjectReference Include="..\..\..\Externals\SoundTouch\SoundTouch.vcxproj">
+      <Project>{68a5dd20-7057-448b-8fe0-b6ac8d205509}</Project>
+      <Private>true</Private>
+      <ReferenceOutputAssembly>true</ReferenceOutputAssembly>
+      <CopyLocalSatelliteAssemblies>false</CopyLocalSatelliteAssemblies>
+      <LinkLibraryDependencies>true</LinkLibraryDependencies>
+      <UseLibraryDependencyInputs>false</UseLibraryDependencyInputs>
+    </ProjectReference>
     <ProjectReference Include="..\Common\Common.vcxproj">
       <Project>{c87a4178-44f6-49b2-b7aa-c79af1b8c534}</Project>
       <Private>true</Private>
diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters
index 7e098ecb85..30ff35a073 100644
--- a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters
+++ b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters
@@ -3,7 +3,6 @@
   <ItemGroup>
     <ClCompile Include="Src\aldlist.cpp" />
     <ClCompile Include="Src\AudioCommon.cpp" />
-    <ClCompile Include="Src\AudioCommonConfig.cpp" />
     <ClCompile Include="Src\Mixer.cpp" />
     <ClCompile Include="Src\WaveFile.cpp" />
     <ClCompile Include="Src\AOSoundStream.cpp">
@@ -21,11 +20,11 @@
     <ClCompile Include="Src\NullSoundStream.cpp">
       <Filter>SoundStreams</Filter>
     </ClCompile>
+    <ClCompile Include="Src\DPL2Decoder.cpp" />
   </ItemGroup>
   <ItemGroup>
     <ClInclude Include="Src\aldlist.h" />
     <ClInclude Include="Src\AudioCommon.h" />
-    <ClInclude Include="Src\AudioCommonConfig.h" />
     <ClInclude Include="Src\Mixer.h" />
     <ClInclude Include="Src\SoundStream.h" />
     <ClInclude Include="Src\WaveFile.h" />
@@ -44,6 +43,7 @@
     <ClInclude Include="Src\XAudio2Stream.h">
       <Filter>SoundStreams</Filter>
     </ClInclude>
+    <ClInclude Include="Src\DPL2Decoder.h" />
   </ItemGroup>
   <ItemGroup>
     <None Include="CMakeLists.txt" />
diff --git a/Source/Core/AudioCommon/CMakeLists.txt b/Source/Core/AudioCommon/CMakeLists.txt
index be8c58d14a..3b6d0cea4d 100644
--- a/Source/Core/AudioCommon/CMakeLists.txt
+++ b/Source/Core/AudioCommon/CMakeLists.txt
@@ -1,5 +1,5 @@
 set(SRCS	Src/AudioCommon.cpp
-			Src/AudioCommonConfig.cpp
+			Src/DPL2Decoder.cpp
 			Src/Mixer.cpp
 			Src/WaveFile.cpp
 			Src/NullSoundStream.cpp)
@@ -18,7 +18,7 @@ endif(AO_FOUND)
 
 if(OPENAL_FOUND)
 	set(SRCS ${SRCS} Src/OpenALStream.cpp Src/aldlist.cpp)
-	set(LIBS ${LIBS} ${OPENAL_LIBRARY})
+	set(LIBS ${LIBS} ${OPENAL_LIBRARY} SoundTouch )
 endif(OPENAL_FOUND)
 
 if(PULSEAUDIO_FOUND)
diff --git a/Source/Core/AudioCommon/Src/AudioCommon.cpp b/Source/Core/AudioCommon/Src/AudioCommon.cpp
index 8a568448a3..e14b9ac45e 100644
--- a/Source/Core/AudioCommon/Src/AudioCommon.cpp
+++ b/Source/Core/AudioCommon/Src/AudioCommon.cpp
@@ -27,6 +27,10 @@
 #include "OpenALStream.h"
 #include "PulseAudioStream.h"
 #include "../../Core/Src/Movie.h"
+#include "../../Core/Src/ConfigManager.h"
+
+// This shouldn't be a global, at least not here.
+SoundStream *soundStream;
 
 namespace AudioCommon 
 {	
@@ -34,7 +38,7 @@ namespace AudioCommon
 	{
 		// TODO: possible memleak with mixer
 
-		std::string backend = ac_Config.sBackend;
+		std::string backend = SConfig::GetInstance().sBackend;
 		if (backend == BACKEND_OPENAL           && OpenALStream::isValid()) 
 			soundStream = new OpenALStream(mixer);
 		else if (backend == BACKEND_NULLSOUND   && NullSound::isValid()) 
@@ -54,10 +58,10 @@ namespace AudioCommon
 
 		if (soundStream != NULL)
 		{
-			ac_Config.Update();
+			UpdateSoundStream();
 			if (soundStream->Start())
 			{
-				if (ac_Config.m_DumpAudio)
+				if (SConfig::GetInstance().m_DumpAudio)
 				{
 					std::string audio_file_name = File::GetUserPath(D_DUMPAUDIO_IDX) + "audiodump.wav";
 					File::CreateFullPath(audio_file_name);
@@ -82,7 +86,7 @@ namespace AudioCommon
 		if (soundStream) 
 		{
 			soundStream->Stop();
-			if (ac_Config.m_DumpAudio)
+			if (SConfig::GetInstance().m_DumpAudio)
 				soundStream->GetMixer()->StopLogAudio();
 				//soundStream->StopLogAudio();
 			delete soundStream;
@@ -96,22 +100,22 @@ namespace AudioCommon
 	{
 		std::vector<std::string> backends;
 
-		if (NullSound::isValid())  
+		if (NullSound::isValid())
 			backends.push_back(BACKEND_NULLSOUND);
-		if (DSound::isValid())  
+		if (DSound::isValid())
 			backends.push_back(BACKEND_DIRECTSOUND);
-		if (XAudio2::isValid())  
+		if (XAudio2::isValid())
 			backends.push_back(BACKEND_XAUDIO2);
-		if (OpenALStream::isValid())
-			backends.push_back(BACKEND_OPENAL);
-		if (AOSound::isValid())   
+		if (AOSound::isValid())
 			backends.push_back(BACKEND_AOSOUND);
-		if (AlsaSound::isValid()) 
+		if (AlsaSound::isValid())
 			backends.push_back(BACKEND_ALSA);
-		if (CoreAudioSound::isValid())       
+		if (CoreAudioSound::isValid())
 			backends.push_back(BACKEND_COREAUDIO);
-		if (PulseAudio::isValid()) 
+		if (PulseAudio::isValid())
 			backends.push_back(BACKEND_PULSEAUDIO);
+		if (OpenALStream::isValid())
+			backends.push_back(BACKEND_OPENAL);
 	   
 		return backends;
 	}
@@ -122,7 +126,7 @@ namespace AudioCommon
 		{
 			return true;
 		}
-		return ac_Config.m_EnableJIT;
+		return SConfig::GetInstance().m_EnableJIT;
 	}
 
 	void PauseAndLock(bool doLock, bool unpauseOnUnlock)
@@ -143,4 +147,12 @@ namespace AudioCommon
 			}
 		}
 	}
+	void UpdateSoundStream()
+	{
+		if (soundStream)
+		{
+			soundStream->GetMixer()->SetThrottle(SConfig::GetInstance().m_Framelimit == 2);
+			soundStream->SetVolume(SConfig::GetInstance().m_Volume);
+		}
+	}
 }
diff --git a/Source/Core/AudioCommon/Src/AudioCommon.h b/Source/Core/AudioCommon/Src/AudioCommon.h
index e7a111b0a1..b7b32e7498 100644
--- a/Source/Core/AudioCommon/Src/AudioCommon.h
+++ b/Source/Core/AudioCommon/Src/AudioCommon.h
@@ -19,14 +19,12 @@
 #define _AUDIO_COMMON_H_
 
 #include "Common.h"
-#include "AudioCommonConfig.h"
 #include "SoundStream.h"
 
 
 class CMixer;
 
 extern SoundStream *soundStream;
-extern AudioCommonConfig ac_Config;
 
 // UDSPControl
 union UDSPControl
@@ -60,6 +58,7 @@ namespace AudioCommon
 	std::vector<std::string> GetSoundBackends();
 	bool UseJIT();
 	void PauseAndLock(bool doLock, bool unpauseOnUnlock=true);
+	void UpdateSoundStream();
 }
 
 #endif // _AUDIO_COMMON_H_
diff --git a/Source/Core/AudioCommon/Src/AudioCommonConfig.cpp b/Source/Core/AudioCommon/Src/AudioCommonConfig.cpp
deleted file mode 100644
index 08902e1c3f..0000000000
--- a/Source/Core/AudioCommon/Src/AudioCommonConfig.cpp
+++ /dev/null
@@ -1,70 +0,0 @@
-// Copyright (C) 2003 Dolphin Project.
-
-// 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, version 2.0.
-
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License 2.0 for more details.
-
-// A copy of the GPL 2.0 should have been included with the program.
-// If not, see http://www.gnu.org/licenses/
-
-// Official SVN repository and contact information can be found at
-// http://code.google.com/p/dolphin-emu/
-
-#include "AudioCommon.h"
-#include "CommonPaths.h"
-#include "FileUtil.h"
-#include "../../Core/Src/ConfigManager.h"
-
-AudioCommonConfig ac_Config;
-
-// This shouldn't be a global, at least not here.
-SoundStream *soundStream;
-
-// Load from given file
-void AudioCommonConfig::Load()
-{
-	IniFile file;
-	file.Load(File::GetUserPath(F_DSPCONFIG_IDX));
-
-	file.Get("Config", "EnableJIT", &m_EnableJIT, true);
-	file.Get("Config", "DumpAudio", &m_DumpAudio, false);
-#if defined __linux__ && HAVE_ALSA
-	file.Get("Config", "Backend", &sBackend, BACKEND_ALSA);
-#elif defined __APPLE__
-	file.Get("Config", "Backend", &sBackend, BACKEND_COREAUDIO);
-#elif defined _WIN32
-	file.Get("Config", "Backend", &sBackend, BACKEND_DIRECTSOUND);
-#else
-	file.Get("Config", "Backend", &sBackend, BACKEND_NULLSOUND);
-#endif
-	file.Get("Config", "Frequency", &iFrequency, 48000);
-	file.Get("Config", "Volume", &m_Volume, 100);
-}
-
-// Set the values for the file
-void AudioCommonConfig::SaveSettings()
-{
-	IniFile file;
-	file.Load(File::GetUserPath(F_DSPCONFIG_IDX));
-
-	file.Set("Config", "EnableJIT", m_EnableJIT);
-	file.Set("Config", "DumpAudio", m_DumpAudio);
-	file.Set("Config", "Backend", sBackend);
-	file.Set("Config", "Frequency", iFrequency);
-	file.Set("Config", "Volume", m_Volume);
-
-	file.Save(File::GetUserPath(F_DSPCONFIG_IDX));
-}
-
-// Update according to the values (stream/mixer)
-void AudioCommonConfig::Update() {
-	if (soundStream) {
-		soundStream->GetMixer()->SetThrottle(SConfig::GetInstance().m_Framelimit == 2);
-		soundStream->SetVolume(m_Volume);
-	}
-}
diff --git a/Source/Core/AudioCommon/Src/AudioCommonConfig.h b/Source/Core/AudioCommon/Src/AudioCommonConfig.h
deleted file mode 100644
index 76c50c6408..0000000000
--- a/Source/Core/AudioCommon/Src/AudioCommonConfig.h
+++ /dev/null
@@ -1,52 +0,0 @@
-// Copyright (C) 2003 Dolphin Project.
-
-// 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, version 2.0.
-
-// This program is distributed in the hope that it will be useful,
-// but WITHOUT ANY WARRANTY; without even the implied warranty of
-// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
-// GNU General Public License 2.0 for more details.
-
-// A copy of the GPL 2.0 should have been included with the program.
-// If not, see http://www.gnu.org/licenses/
-
-// Official SVN repository and contact information can be found at
-// http://code.google.com/p/dolphin-emu/
-
-#ifndef _AUDIO_COMMON_CONFIG_H_
-#define _AUDIO_COMMON_CONFIG_H_
-
-#include <string>
-#include "IniFile.h"
-
-// Backend Types
-#define BACKEND_NULLSOUND	"No audio output"
-#define BACKEND_ALSA		"ALSA"
-#define BACKEND_AOSOUND		"AOSound"
-#define BACKEND_COREAUDIO	"CoreAudio"
-#define BACKEND_DIRECTSOUND	"DSound"
-#define BACKEND_OPENAL		"OpenAL"
-#define BACKEND_PULSEAUDIO	"Pulse"
-#define BACKEND_XAUDIO2		"XAudio2"
-
-struct AudioCommonConfig 
-{
-	bool m_EnableJIT;
-	bool m_DumpAudio;
-	int m_Volume;
-	std::string sBackend;
-	int iFrequency;
-	
-	// Load from given file
-	void Load();
-	
-	// Self explanatory
-	void SaveSettings();
-
-	// Update according to the values (stream/mixer)
-	void Update();
-};
-
-#endif //AUDIO_COMMON_CONFIG
diff --git a/Source/Core/AudioCommon/Src/DPL2Decoder.cpp b/Source/Core/AudioCommon/Src/DPL2Decoder.cpp
new file mode 100644
index 0000000000..a5d2720878
--- /dev/null
+++ b/Source/Core/AudioCommon/Src/DPL2Decoder.cpp
@@ -0,0 +1,400 @@
+// Copyright (C) 2003 Dolphin Project.
+
+// 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, version 2.0.
+
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License 2.0 for more details.
+
+// A copy of the GPL 2.0 should have been included with the program.
+// If not, see http://www.gnu.org/licenses/
+
+// Official SVN repository and contact information can be found at
+// http://code.google.com/p/dolphin-emu/
+
+// Dolby Pro Logic 2 decoder from ffdshow-tryout
+//  * Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
+//  * Copyright (c) 2004-2006 Milan Cutka
+//	* based on mplayer HRTF plugin by ylai
+
+#include <functional>
+#include <vector>
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include "DPL2Decoder.h"
+
+#define M_PI 3.14159265358979323846
+#define M_SQRT1_2 0.70710678118654752440
+
+int olddelay = -1;
+unsigned int oldfreq = 0;
+unsigned int dlbuflen;
+int cyc_pos;
+float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
+std::vector<float> fwrbuf_l, fwrbuf_r;
+float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
+std::vector<float> lf, rf, lr, rr, cf, cr;
+float LFE_buf[256];
+unsigned int lfe_pos;
+float *filter_coefs_lfe;
+unsigned int len125;
+
+template<class T,class _ftype_t> static _ftype_t dotproduct(int count,const T *buf,const _ftype_t *coefficients)
+{
+	float sum0=0,sum1=0,sum2=0,sum3=0;
+	for (;count>=4;buf+=4,coefficients+=4,count-=4)
+	{
+		sum0+=buf[0]*coefficients[0];
+		sum1+=buf[1]*coefficients[1];
+		sum2+=buf[2]*coefficients[2];
+		sum3+=buf[3]*coefficients[3];
+	}
+	while (count--) sum0+= *buf++ * *coefficients++;
+	return sum0+sum1+sum2+sum3;
+}
+
+template<class T> static T firfilter(const T *buf, int pos, int len, int count, const float *coefficients)
+{
+	int count1, count2;
+
+	if (pos >= count)
+	{
+		pos -= count;
+		count1 = count; count2 = 0;
+	}
+	else
+	{
+		count2 = pos;
+		count1 = count - pos;
+		pos = len - count1;
+	}
+
+	// high part of window
+	const T *ptr = &buf[pos];
+
+	float r1=dotproduct(count1,ptr,coefficients);coefficients+=count1;
+	float r2=dotproduct(count2,buf,coefficients);
+	return T(r1+r2);
+}
+
+template<class T> inline const T& limit(const T& val, const T& min, const T& max)
+{
+	if (val < min) {
+		return min;
+	} else if (val > max) {
+		return max;
+	} else {
+		return val;
+	}
+}
+
+/*
+// Hamming
+//                        2*pi*k
+// w(k) = 0.54 - 0.46*cos(------), where 0 <= k < N
+//                         N-1
+//
+// n window length
+// w buffer for the window parameters
+*/
+void hamming(int n, float* w)
+{
+	int      i;
+	float k = float(2*M_PI/((float)(n-1))); // 2*pi/(N-1)
+
+	// Calculate window coefficients
+	for (i=0; i<n; i++)
+		*w++ = float(0.54 - 0.46*cos(k*(float)i));
+}
+
+/******************************************************************************
+*  FIR filter design
+******************************************************************************/
+
+/* Design FIR filter using the Window method
+
+n     filter length must be odd for HP and BS filters
+w     buffer for the filter taps (must be n long)
+fc    cutoff frequencies (1 for LP and HP, 2 for BP and BS)
+0 < fc < 1 where 1 <=> Fs/2
+flags window and filter type as defined in filter.h
+variables are ored together: i.e. LP|HAMMING will give a
+low pass filter designed using a hamming window
+opt   beta constant used only when designing using kaiser windows
+
+returns 0 if OK, -1 if fail
+*/
+float* design_fir(unsigned int *n, float* fc, float opt)
+{
+	unsigned int  o   = *n & 1;            // Indicator for odd filter length
+	unsigned int  end = ((*n + 1) >> 1) - o;       // Loop end
+	unsigned int  i;                      // Loop index
+
+	float k1 = 2 * float(M_PI);             // 2*pi*fc1
+	float k2 = 0.5f * (float)(1 - o);// Constant used if the filter has even length
+	float g  = 0.0f;                    // Gain
+	float t1;                     // Temporary variables
+	float fc1;                      // Cutoff frequencies
+
+	// Sanity check
+	if(*n==0) return NULL;
+	fc[0]=limit(fc[0],float(0.001),float(1));
+
+	float *w=(float*)calloc(sizeof(float),*n);
+
+	// Get window coefficients
+	hamming(*n,w);
+
+	fc1=*fc;
+	// Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
+	fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25f;
+	k1 *= fc1;
+
+	// Low pass filter
+
+	// If the filter length is odd, there is one point which is exactly
+	// in the middle. The value at this point is 2*fCutoff*sin(x)/x,
+	// where x is zero. To make sure nothing strange happens, we set this
+	// value separately.
+	if (o)
+	{
+		w[end] = fc1 * w[end] * 2.0f;
+		g=w[end];
+	}
+
+	// Create filter
+	for (i=0 ; i<end ; i++)
+	{
+		t1 = (float)(i+1) - k2;
+		w[end-i-1] = w[*n-end+i] = float(w[end-i-1] * sin(k1 * t1)/(M_PI * t1)); // Sinc
+		g += 2*w[end-i-1]; // Total gain in filter
+	}
+
+
+	// Normalize gain
+	g=1/g;
+	for (i=0; i<*n; i++)
+		w[i] *= g;
+
+	return w;
+}
+
+void onSeek(void)
+{
+	l_fwr = r_fwr = lpr_fwr = lmr_fwr = 0;
+	std::fill(fwrbuf_l.begin(), fwrbuf_l.end(), 0.0f);
+	std::fill(fwrbuf_r.begin(), fwrbuf_r.end(), 0.0f);
+	adapt_l_gain = adapt_r_gain = adapt_lpr_gain = adapt_lmr_gain = 0;
+	std::fill(lf.begin(), lf.end(), 0.0f);
+	std::fill(rf.begin(), rf.end(), 0.0f);
+	std::fill(lr.begin(), lr.end(), 0.0f);
+	std::fill(rr.begin(), rr.end(), 0.0f);
+	std::fill(cf.begin(), cf.end(), 0.0f);
+	std::fill(cr.begin(), cr.end(), 0.0f);
+	lfe_pos = 0;
+	memset(LFE_buf, 0, sizeof(LFE_buf));
+}
+
+void done(void)
+{
+	onSeek();
+	if (filter_coefs_lfe)
+	{
+		free(filter_coefs_lfe);
+	}
+	filter_coefs_lfe = NULL;
+}
+
+float* calc_coefficients_125Hz_lowpass(int rate)
+{
+	len125 = 256;
+	float f = 125.0f / (rate / 2);
+	float *coeffs = design_fir(&len125, &f, 0);
+	static const float M3_01DB = 0.7071067812f;
+	for (unsigned int i = 0; i < len125; i++)
+	{
+		coeffs[i] *= M3_01DB;
+	}
+	return coeffs;
+}
+
+float passive_lock(float x)
+{
+	static const float MATAGCLOCK = 0.2f;  /* AGC range (around 1) where the matrix behaves passively */
+	const float x1 = x - 1;
+	const float ax1s = fabs(x - 1) * (1.0f / MATAGCLOCK);
+	return x1 - x1 / (1 + ax1s * ax1s) + 1;
+}
+
+void matrix_decode(const float *in, const int k, const int il,
+	const int ir, bool decode_rear,
+	const int dlbuflen,
+	float l_fwr, float r_fwr,
+	float lpr_fwr, float lmr_fwr,
+	float *adapt_l_gain, float *adapt_r_gain,
+	float *adapt_lpr_gain, float *adapt_lmr_gain,
+	float *lf, float *rf, float *lr,
+	float *rr, float *cf)
+{
+	static const float M9_03DB = 0.3535533906f;
+	static const float MATAGCTRIG = 8.0f;  /* (Fuzzy) AGC trigger */
+	static const float MATAGCDECAY = 1.0f; /* AGC baseline decay rate (1/samp.) */
+	static const float MATCOMPGAIN = 0.37f; /* Cross talk compensation gain,  0.50 - 0.55 is full cancellation. */
+
+	const int kr = (k + olddelay) % dlbuflen;
+	float l_gain = (l_fwr + r_fwr) / (1 + l_fwr + l_fwr);
+	float r_gain = (l_fwr + r_fwr) / (1 + r_fwr + r_fwr);
+	/* The 2nd axis has strong gain fluctuations, and therefore require
+	limits.  The factor corresponds to the 1 / amplification of (Lt
+	- Rt) when (Lt, Rt) is strongly correlated. (e.g. during
+	dialogues).  It should be bigger than -12 dB to prevent
+	distortion. */
+	float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ? lmr_fwr : M9_03DB * lpr_fwr;
+	float lpr_gain = (lpr_fwr + lmr_lim_fwr) / (1 + lpr_fwr + lpr_fwr);
+	float lmr_gain = (lpr_fwr + lmr_lim_fwr) / (1 + lmr_lim_fwr + lmr_lim_fwr);
+	float lmr_unlim_gain = (lpr_fwr + lmr_fwr) / (1 + lmr_fwr + lmr_fwr);
+	float lpr, lmr;
+	float l_agc, r_agc, lpr_agc, lmr_agc;
+	float f, d_gain, c_gain, c_agc_cfk;
+
+	/*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
+	/* AGC adaption */
+	d_gain = (fabs(l_gain - *adapt_l_gain) + fabs(r_gain - *adapt_r_gain)) * 0.5f;
+	f = d_gain * (1.0f / MATAGCTRIG);
+	f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
+	*adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
+	*adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
+	/* Matrix */
+	l_agc = in[il] * passive_lock(*adapt_l_gain);
+	r_agc = in[ir] * passive_lock(*adapt_r_gain);
+	cf[k] = (l_agc + r_agc) * (float)M_SQRT1_2;
+	if (decode_rear)
+	{
+		lr[kr] = rr[kr] = (l_agc - r_agc) * (float)M_SQRT1_2;
+		/* Stereo rear channel is steered with the same AGC steering as
+		the decoding matrix. Note this requires a fast updating AGC
+		at the order of 20 ms (which is the case here). */
+		lr[kr] *= (l_fwr + l_fwr) / (1 + l_fwr + r_fwr);
+		rr[kr] *= (r_fwr + r_fwr) / (1 + l_fwr + r_fwr);
+	}
+
+	/*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
+	lpr = (in[il] + in[ir]) * (float)M_SQRT1_2;
+	lmr = (in[il] - in[ir]) * (float)M_SQRT1_2;
+	/* AGC adaption */
+	d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
+	f = d_gain * (1.0f / MATAGCTRIG);
+	f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
+	*adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
+	*adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
+	/* Matrix */
+	lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
+	lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
+	lf[k] = (lpr_agc + lmr_agc) * (float)M_SQRT1_2;
+	rf[k] = (lpr_agc - lmr_agc) * (float)M_SQRT1_2;
+
+	/*** CENTER FRONT CANCELLATION ***/
+	/* A heuristic approach exploits that Lt + Rt gain contains the
+	information about Lt, Rt correlation.  This effectively reshapes
+	the front and rear "cones" to concentrate Lt + Rt to C and
+	introduce Lt - Rt in L, R. */
+	/* 0.67677 is the empirical lower bound for lpr_gain. */
+	c_gain = 8 * (*adapt_lpr_gain - 0.67677f);
+	c_gain = c_gain > 0 ? c_gain : 0;
+	/* c_gain should not be too high, not even reaching full
+	cancellation (~ 0.50 - 0.55 at current AGC implementation), or
+	the center will sound too narrow. */
+	c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
+	c_agc_cfk = c_gain * cf[k];
+	lf[k] -= c_agc_cfk;
+	rf[k] -= c_agc_cfk;
+	cf[k] += c_agc_cfk + c_agc_cfk;
+}
+
+void dpl2decode(float *samples, int numsamples, float *out)
+{
+	static const unsigned int FWRDURATION = 240;    /* FWR average duration (samples) */
+	static const int cfg_delay = 0;
+	static const unsigned int fmt_freq = 48000;
+	static const unsigned int fmt_nchannels = 2; // input channels
+
+	int cur = 0;
+
+	if (olddelay != cfg_delay || oldfreq != fmt_freq)
+	{
+		done();
+		olddelay = cfg_delay;
+		oldfreq = fmt_freq;
+		dlbuflen = std::max(FWRDURATION, (fmt_freq * cfg_delay / 1000)); //+(len7000-1);
+		cyc_pos = dlbuflen - 1;
+		fwrbuf_l.resize(dlbuflen);
+		fwrbuf_r.resize(dlbuflen);
+		lf.resize(dlbuflen);
+		rf.resize(dlbuflen);
+		lr.resize(dlbuflen);
+		rr.resize(dlbuflen);
+		cf.resize(dlbuflen);
+		cr.resize(dlbuflen);
+		filter_coefs_lfe = calc_coefficients_125Hz_lowpass(fmt_freq);
+		lfe_pos = 0;
+		memset(LFE_buf, 0, sizeof(LFE_buf));
+	}
+
+	float *in = samples; // Input audio data
+	float *end = in + numsamples * fmt_nchannels; // Loop end
+
+	while (in < end)
+	{
+		const int k = cyc_pos;
+
+		const int fwr_pos = (k + FWRDURATION) % dlbuflen;
+		/* Update the full wave rectified total amplitude */
+		/* Input matrix decoder */
+		l_fwr += fabs(in[0]) - fabs(fwrbuf_l[fwr_pos]);
+		r_fwr += fabs(in[1]) - fabs(fwrbuf_r[fwr_pos]);
+		lpr_fwr += fabs(in[0] + in[1]) - fabs(fwrbuf_l[fwr_pos] + fwrbuf_r[fwr_pos]);
+		lmr_fwr += fabs(in[0] - in[1]) - fabs(fwrbuf_l[fwr_pos] - fwrbuf_r[fwr_pos]);
+
+		/* Matrix encoded 2 channel sources */
+		fwrbuf_l[k] = in[0];
+		fwrbuf_r[k] = in[1];
+		matrix_decode(in, k, 0, 1, true, dlbuflen,
+			l_fwr, r_fwr,
+			lpr_fwr, lmr_fwr,
+			&adapt_l_gain, &adapt_r_gain,
+			&adapt_lpr_gain, &adapt_lmr_gain,
+			&lf[0], &rf[0], &lr[0], &rr[0], &cf[0]);
+
+		out[cur + 0] = lf[k];
+		out[cur + 1] = rf[k];
+		out[cur + 2] = cf[k];
+		LFE_buf[lfe_pos] = (out[0] + out[1]) / 2;
+		out[cur + 3] = firfilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe);
+		lfe_pos++;
+		if (lfe_pos == len125)
+		{
+			lfe_pos = 0;
+		}
+		out[cur + 4] = lr[k];
+		out[cur + 5] = rr[k];
+		// Next sample...
+		in += 2;
+		cur += 6;
+		cyc_pos--;
+		if (cyc_pos < 0)
+		{
+			cyc_pos += dlbuflen;
+		}
+	}
+}
+
+void dpl2reset()
+{
+	olddelay = -1;
+	oldfreq = 0;
+	filter_coefs_lfe = NULL;
+}
diff --git a/Source/Core/AudioCommon/Src/DPL2Decoder.h b/Source/Core/AudioCommon/Src/DPL2Decoder.h
new file mode 100644
index 0000000000..eee8fbf15e
--- /dev/null
+++ b/Source/Core/AudioCommon/Src/DPL2Decoder.h
@@ -0,0 +1,24 @@
+// Copyright (C) 2003 Dolphin Project.
+
+// 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, version 2.0.
+
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License 2.0 for more details.
+
+// A copy of the GPL 2.0 should have been included with the program.
+// If not, see http://www.gnu.org/licenses/
+
+// Official SVN repository and contact information can be found at
+// http://code.google.com/p/dolphin-emu/
+
+#ifndef _DPL2DECODER_H_
+#define _DPL2DECODER_H_
+
+void dpl2decode(float *samples, int numsamples, float *out);
+void dpl2reset();
+
+#endif  // _DPL2DECODER_H_
diff --git a/Source/Core/AudioCommon/Src/Mixer.h b/Source/Core/AudioCommon/Src/Mixer.h
index da456d5bf7..c1a387cf49 100644
--- a/Source/Core/AudioCommon/Src/Mixer.h
+++ b/Source/Core/AudioCommon/Src/Mixer.h
@@ -92,6 +92,9 @@ public:
 
 	std::mutex& MixerCritical() { return m_csMixing; }
 
+	volatile float GetCurrentSpeed() const { return m_speed; }
+	void UpdateSpeed(volatile float val) { m_speed = val; }
+
 protected:
 	unsigned int m_sampleRate;
 	unsigned int m_aiSampleRate;
@@ -113,6 +116,8 @@ protected:
 
 	bool m_AIplaying;
 	std::mutex m_csMixing;
+
+	volatile float m_speed; // Current rate of the emulation (1.0 = 100% speed)
 private:
 
 };
diff --git a/Source/Core/AudioCommon/Src/OpenALStream.cpp b/Source/Core/AudioCommon/Src/OpenALStream.cpp
index 2abd357782..cfe6a40630 100644
--- a/Source/Core/AudioCommon/Src/OpenALStream.cpp
+++ b/Source/Core/AudioCommon/Src/OpenALStream.cpp
@@ -15,13 +15,14 @@
 // Official SVN repository and contact information can be found at
 // http://code.google.com/p/dolphin-emu/
 
-#include <functional>
-
 #include "aldlist.h"
 #include "OpenALStream.h"
+#include "DPL2Decoder.h"
 
 #if defined HAVE_OPENAL && HAVE_OPENAL
 
+soundtouch::SoundTouch soundTouch;
+
 //
 // AyuanX: Spec says OpenAL1.1 is thread safe already
 //
@@ -35,14 +36,21 @@ bool OpenALStream::Start()
 	pDeviceList = new ALDeviceList();
 	if ((pDeviceList) && (pDeviceList->GetNumDevices()))
 	{
-		char *defDevName = pDeviceList-> \
-			GetDeviceName(pDeviceList->GetDefaultDevice());
+		char *defDevName = pDeviceList->GetDeviceName(pDeviceList->GetDefaultDevice());
+
+		WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);
+
 		pDevice = alcOpenDevice(defDevName);
 		if (pDevice)
 		{
 			pContext = alcCreateContext(pDevice, NULL);
 			if (pContext)
 			{
+				// Used to determine an appropriate period size (2x period = total buffer size)
+				//ALCint refresh;
+				//alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
+				//period_size_in_millisec = 1000 / refresh;
+
 				alcMakeContextCurrent(pContext);
 				thread = std::thread(std::mem_fun(&OpenALStream::SoundLoop), this);
 				bReturn = true;
@@ -50,8 +58,7 @@ bool OpenALStream::Start()
 			else
 			{
 				alcCloseDevice(pDevice);
-				PanicAlertT("OpenAL: can't create context "
-					"for device %s", defDevName);
+				PanicAlertT("OpenAL: can't create context for device %s", defDevName);
 			}
 		}
 		else
@@ -65,6 +72,10 @@ bool OpenALStream::Start()
 		PanicAlertT("OpenAL: can't find sound devices");
 	}
 
+	// Initialise DPL2 parameters
+	dpl2reset();
+
+	soundTouch.clear();
 	return bReturn;
 }
 
@@ -74,6 +85,8 @@ void OpenALStream::Stop()
 	// kick the thread if it's waiting
 	soundSyncEvent.Set();
 
+	soundTouch.clear();
+
 	thread.join();
 
 	alSourceStop(uiSource);
@@ -82,7 +95,7 @@ void OpenALStream::Stop()
 	// Clean up buffers and sources
 	alDeleteSources(1, &uiSource);
 	uiSource = 0;
-	alDeleteBuffers(OAL_NUM_BUFFERS, uiBuffers);
+	alDeleteBuffers(numBuffers, uiBuffers);
 
 	ALCcontext *pContext = alcGetCurrentContext();
 	ALCdevice *pDevice = alcGetContextsDevice(pContext);
@@ -111,6 +124,7 @@ void OpenALStream::Clear(bool mute)
 
 	if(m_muted)
 	{
+		soundTouch.clear();
 		alSourceStop(uiSource);
 	}
 	else
@@ -124,20 +138,29 @@ void OpenALStream::SoundLoop()
 	Common::SetCurrentThreadName("Audio thread - openal");
 
 	u32 ulFrequency = m_mixer->GetSampleRate();
+	numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
 
-	memset(uiBuffers, 0, OAL_NUM_BUFFERS * sizeof(ALuint));
+	memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
 	uiSource = 0;
 
 	// Generate some AL Buffers for streaming
-	alGenBuffers(OAL_NUM_BUFFERS, (ALuint *)uiBuffers);
+	alGenBuffers(numBuffers, (ALuint *)uiBuffers);
 	// Generate a Source to playback the Buffers
 	alGenSources(1, &uiSource);
 
 	// Short Silence
+	memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
 	memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
-	for (int i = 0; i < OAL_NUM_BUFFERS; i++)
-		alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, OAL_MAX_SAMPLES, ulFrequency);
-	alSourceQueueBuffers(uiSource, OAL_NUM_BUFFERS, uiBuffers);
+	for (int i = 0; i < numBuffers; i++)
+	{
+#if !defined(__APPLE__)
+		if (Core::g_CoreStartupParameter.bDPL2Decoder)
+			alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
+		else
+#endif
+			alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
+	}
+	alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
 	alSourcePlay(uiSource);
 	
 	// Set the default sound volume as saved in the config file.
@@ -148,41 +171,175 @@ void OpenALStream::SoundLoop()
 
 	ALint iBuffersFilled = 0;
 	ALint iBuffersProcessed = 0;
-	ALuint uiBufferTemp[OAL_NUM_BUFFERS] = {0};
+	ALint iState = 0;
+	ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
+
+	soundTouch.setChannels(2);
+	soundTouch.setSampleRate(ulFrequency);
+	soundTouch.setTempo(1.0);
+	soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
+	soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
+	soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
+	soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
+	soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
+
+	bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
+#if defined(__APPLE__)
+	bool float32_capable = false;
+#else
+	bool float32_capable = true;
+#endif
 
 	while (!threadData) 
 	{
+		// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
+		const u32 stereo_16_bit_size = 4;
+		const u32 dma_length = 32;
+		const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
+		u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
+		u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
+
+		unsigned int numSamples = (unsigned int)num_samples_to_render;
+
+		numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
+		numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
+
+		// Convert the samples from short to float
+		float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
+		for (u32 i = 0; i < numSamples; ++i)
+		{
+			dest[i * 2 + 0] = (float)realtimeBuffer[i * 2 + 0] / (1 << 16);
+			dest[i * 2 + 1] = (float)realtimeBuffer[i * 2 + 1] / (1 << 16);
+		}
+
+		soundTouch.putSamples(dest, numSamples);
+
 		if (iBuffersProcessed == iBuffersFilled)
 		{
 			alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
 			iBuffersFilled = 0;
 		}
 
-		unsigned int numSamples = m_mixer->GetNumSamples();
-
-		if (iBuffersProcessed && (numSamples >= OAL_THRESHOLD))
+		if (iBuffersProcessed)
 		{
-			numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
-			// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
-			if (iBuffersFilled == 0)
-				alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
-
-			m_mixer->Mix(realtimeBuffer, numSamples);
-			alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, realtimeBuffer, numSamples * 4, ulFrequency);
-			alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
-			iBuffersFilled++;
-
-			if (iBuffersFilled == OAL_NUM_BUFFERS)
-				alSourcePlay(uiSource);
+			float rate = m_mixer->GetCurrentSpeed();
+			if (rate <= 0)
+			{
+				Core::RequestRefreshInfo();
+				rate = m_mixer->GetCurrentSpeed();
+			}
+
+			// Place a lower limit of 10% speed.  When a game boots up, there will be
+			// many silence samples.  These do not need to be timestretched.
+			if (rate > 0.10)
+			{
+				// Adjust SETTING_SEQUENCE_MS to balance between lag vs hollow audio
+				soundTouch.setSetting(SETTING_SEQUENCE_MS, (int)(1 / (rate * rate)));
+				soundTouch.setTempo(rate);
+			}
+			unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS);
+			if (nSamples > 0)
+			{
+				// Remove the Buffer from the Queue.  (uiBuffer contains the Buffer ID for the unqueued Buffer)
+				if (iBuffersFilled == 0)
+				{
+					alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
+					ALenum err = alGetError();
+					if (err != 0)
+					{
+						ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
+					}
+				}
+#if defined(__APPLE__)
+				// OSX does not have the alext AL_FORMAT_51CHN32 yet.
+				surround_capable = false;
+#else
+				if (surround_capable)
+				{
+					float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
+					dpl2decode(sampleBuffer, nSamples, dpl2);
+
+					alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
+					ALenum err = alGetError();
+					if (err == AL_INVALID_ENUM)
+					{
+						// 5.1 is not supported by the host, fallback to stereo
+						WARN_LOG(AUDIO, "Unable to set 5.1 surround mode.  Updating OpenAL Soft might fix this issue.");
+						surround_capable = false;
+					}
+					else if (err != 0)
+					{
+						ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
+					}
+				}
+#endif
+				if (!surround_capable)
+				{
+#if !defined(__APPLE__)
+					if (float32_capable)
+					{
+						alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, ulFrequency);
+						ALenum err = alGetError();
+						if (err == AL_INVALID_ENUM)
+						{
+							float32_capable = false;
+						}
+						else if (err != 0)
+						{
+							ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
+						}
+
+					}
+#endif
+					if (!float32_capable)
+					{
+						// Convert the samples from float to short
+						short stereo[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
+						for (u32 i = 0; i < nSamples; ++i)
+						{
+							stereo[i * 2 + 0] = (short)((float)sampleBuffer[i * 2 + 0] * (1 << 16));
+							stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16));
+						}
+						alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, ulFrequency);
+
+					}
+				}
+
+				alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
+				ALenum err = alGetError();
+				if (err != 0)
+				{
+					ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
+				}
+				iBuffersFilled++;
+
+				if (iBuffersFilled == numBuffers)
+				{
+					alSourcePlay(uiSource);
+					ALenum err = alGetError();
+					if (err != 0)
+					{
+						ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
+					}
+				}
+
+				alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
+				if (iState != AL_PLAYING)
+				{
+					// Buffer underrun occurred, resume playback
+					alSourcePlay(uiSource);
+					ALenum err = alGetError();
+					if (err != 0)
+					{
+						ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
+					}
+				}
+			}
 		}
-		else if (numSamples >= OAL_THRESHOLD)
+		else
 		{
-			ALint state = 0;
-			alGetSourcei(uiSource, AL_SOURCE_STATE, &state);
-			if (state == AL_STOPPED)
-				alSourcePlay(uiSource);
+			soundSyncEvent.Wait();
 		}
-		soundSyncEvent.Wait();
 	}
 }
 
diff --git a/Source/Core/AudioCommon/Src/OpenALStream.h b/Source/Core/AudioCommon/Src/OpenALStream.h
index 2c0d75b510..a808fcefc9 100644
--- a/Source/Core/AudioCommon/Src/OpenALStream.h
+++ b/Source/Core/AudioCommon/Src/OpenALStream.h
@@ -24,21 +24,30 @@
 
 #if defined HAVE_OPENAL && HAVE_OPENAL
 #ifdef _WIN32
-#include "../../../../Externals/OpenAL/include/al.h"
-#include "../../../../Externals/OpenAL/include/alc.h"
+#include <OpenAL/include/al.h>
+#include <OpenAL/include/alc.h>
+#include <OpenAL/include/alext.h>
 #elif defined __APPLE__
 #include <OpenAL/al.h>
 #include <OpenAL/alc.h>
 #else
 #include <AL/al.h>
 #include <AL/alc.h>
+#include <AL/alext.h>
 #endif
 
+#include "Core.h"
+#include "HW/SystemTimers.h"
+#include "HW/AudioInterface.h"
+#include <soundtouch/SoundTouch.h>
+#include <soundtouch/STTypes.h>
+
 // 16 bit Stereo
 #define SFX_MAX_SOURCE		1
-#define OAL_NUM_BUFFERS		16
-#define OAL_MAX_SAMPLES		512		// AyuanX: Don't make it too large, as larger buffer means longer delay
-#define OAL_THRESHOLD		128		// Some games are quite sensitive to delay
+#define OAL_MAX_BUFFERS		32
+#define OAL_MAX_SAMPLES		256
+#define SURROUND_CHANNELS	6	// number of channels in surround mode
+#define SIZE_FLOAT			4   // size of a float in bytes
 #endif
 
 class OpenALStream: public SoundStream
@@ -64,11 +73,14 @@ public:
 private:
 	std::thread thread;
 	Common::Event soundSyncEvent;
-	
+
 	short realtimeBuffer[OAL_MAX_SAMPLES * 2];
-	ALuint uiBuffers[OAL_NUM_BUFFERS];
+	soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
+	ALuint uiBuffers[OAL_MAX_BUFFERS];
 	ALuint uiSource;
 	ALfloat fVolume;
+
+	u8 numBuffers;
 #else
 public:
 	OpenALStream(CMixer *mixer, void *hWnd = NULL): SoundStream(mixer) {}
diff --git a/Source/Core/AudioCommon/Src/PulseAudioStream.cpp b/Source/Core/AudioCommon/Src/PulseAudioStream.cpp
index 4851b4bd12..faa94c22e0 100644
--- a/Source/Core/AudioCommon/Src/PulseAudioStream.cpp
+++ b/Source/Core/AudioCommon/Src/PulseAudioStream.cpp
@@ -22,31 +22,31 @@
 
 #include "PulseAudioStream.h"
 
-#define BUFFER_SIZE 4096
-#define BUFFER_SIZE_BYTES (BUFFER_SIZE * 4)
-
-PulseAudio::PulseAudio(CMixer *mixer)
-	: SoundStream(mixer), thread_running(false), mainloop(NULL)
-	, context(NULL), stream(NULL), iVolume(100)
+namespace
 {
-	mix_buffer = new u8[BUFFER_SIZE_BYTES];
+const size_t BUFFER_SAMPLES = 512;
+const size_t CHANNEL_COUNT = 2;
+const size_t BUFFER_SIZE = BUFFER_SAMPLES * CHANNEL_COUNT;
 }
 
-PulseAudio::~PulseAudio()
-{
-	delete [] mix_buffer;
-}
+PulseAudio::PulseAudio(CMixer *mixer)
+	: SoundStream(mixer)
+	, mix_buffer(BUFFER_SIZE)
+	, thread()
+	, run_thread()
+	, pa()
+{}
 
 bool PulseAudio::Start()
 {
-	thread_running = true;
+	run_thread = true;
 	thread = std::thread(std::mem_fun(&PulseAudio::SoundLoop), this);
 	return true;
 }
 
 void PulseAudio::Stop()
 {
-	thread_running = false;
+	run_thread = false;
 	thread.join();
 }
 
@@ -60,260 +60,53 @@ void PulseAudio::SoundLoop()
 {
 	Common::SetCurrentThreadName("Audio thread - pulse");
 
-	thread_running = PulseInit();
-
-	while (thread_running)
-	{
-		int frames_to_deliver = 512;
-		m_mixer->Mix((short *)mix_buffer, frames_to_deliver);
-		if (!Write(mix_buffer, frames_to_deliver * 4))
-			ERROR_LOG(AUDIO, "PulseAudio failure writing data");
-	}
-	PulseShutdown();
-}
-
-bool PulseAudio::PulseInit()
-{
-	// The Sample format to use
-	const pa_sample_spec ss =
-	{
-		PA_SAMPLE_S16LE,
-		m_mixer->GetSampleRate(),
-		2
-	};
-
-	mainloop = pa_threaded_mainloop_new();
-
-	context = pa_context_new(pa_threaded_mainloop_get_api(mainloop), "dolphin-emu");
-	pa_context_set_state_callback(context, ContextStateCB, this);
-
-	if (pa_context_connect(context, NULL, PA_CONTEXT_NOFLAGS, NULL) < 0)
+	if (PulseInit())
 	{
-		ERROR_LOG(AUDIO, "PulseAudio failed to connect context: %s",
-				pa_strerror(pa_context_errno(context)));
-		return false;
-	}
-
-	pa_threaded_mainloop_lock(mainloop);
-	pa_threaded_mainloop_start(mainloop);
-
-	for (;;)
-	{
-		pa_context_state_t state;
-
-		state = pa_context_get_state(context);
-
-		if (state == PA_CONTEXT_READY)
-			break;
-
-		if (!PA_CONTEXT_IS_GOOD(state))
+		while (run_thread)
 		{
-			ERROR_LOG(AUDIO, "PulseAudio context state failure: %s",
-					pa_strerror(pa_context_errno(context)));
-			pa_threaded_mainloop_unlock(mainloop);
-			return false;
+			m_mixer->Mix(&mix_buffer[0], mix_buffer.size() / CHANNEL_COUNT);
+			Write(&mix_buffer[0], mix_buffer.size() * sizeof(s16));
 		}
 
-		// Wait until the context is ready
-		pa_threaded_mainloop_wait(mainloop);
+		PulseShutdown();
 	}
-
-	if (!(stream = pa_stream_new(context, "emulator", &ss, NULL)))
-	{
-		ERROR_LOG(AUDIO, "PulseAudio failed to create playback stream: %s",
-				pa_strerror(pa_context_errno(context)));
-		pa_threaded_mainloop_unlock(mainloop);
-		return false;
-	}
-
-	// Set callbacks for the playback stream
-	pa_stream_set_state_callback(stream, StreamStateCB, this);
-	pa_stream_set_write_callback(stream, StreamWriteCB, this);
-
-	if (pa_stream_connect_playback(stream, NULL, NULL, PA_STREAM_NOFLAGS, NULL, NULL) < 0)
-	{
-		ERROR_LOG(AUDIO, "PulseAudio failed to connect playback stream: %s",
-				pa_strerror(pa_context_errno(context)));
-		pa_threaded_mainloop_unlock(mainloop);
-		return false;
-	}
-
-	for (;;)
-	{
-		pa_stream_state_t state;
-
-		state = pa_stream_get_state(stream);
-
-		if (state == PA_STREAM_READY)
-			break;
-
-		if (!PA_STREAM_IS_GOOD(state))
-		{
-			ERROR_LOG(AUDIO, "PulseAudio stream state failure: %s",
-					pa_strerror(pa_context_errno(context)));
-			pa_threaded_mainloop_unlock(mainloop);
-			return false;
-		}
-
-		// Wait until the stream is ready
-		pa_threaded_mainloop_wait(mainloop);
-	}
-
-	pa_threaded_mainloop_unlock(mainloop);
-
-	SetVolume(iVolume);
-
-	NOTICE_LOG(AUDIO, "Pulse successfully initialized.");
-	return true;
 }
 
-void PulseAudio::PulseShutdown()
+bool PulseAudio::PulseInit()
 {
-	if (mainloop)
-		pa_threaded_mainloop_stop(mainloop);
+	pa_sample_spec ss = {};
+	ss.format = PA_SAMPLE_S16LE;
+	ss.channels = 2;
+	ss.rate = m_mixer->GetSampleRate();
 
-	if (stream)
-		pa_stream_unref(stream);
+	int error;
+	pa = pa_simple_new(nullptr, "dolphin-emu", PA_STREAM_PLAYBACK,
+		nullptr, "audio", &ss, nullptr, nullptr, &error);
 
-	if (context)
+	if (!pa)
 	{
-		pa_context_disconnect(context);
-		pa_context_unref(context);
-	}
-
-	if (mainloop)
-		pa_threaded_mainloop_free(mainloop);
-}
-
-void PulseAudio::SignalMainLoop()
-{
-	pa_threaded_mainloop_signal(mainloop, 0);
-}
-
-void PulseAudio::ContextStateCB(pa_context *c, void *userdata)
-{
-	switch (pa_context_get_state(c))
-	{
-		case PA_CONTEXT_READY:
-		case PA_CONTEXT_TERMINATED:
-		case PA_CONTEXT_FAILED:
-			((PulseAudio *)userdata)->SignalMainLoop();
-			break;
-
-		default:
-			break;
+		ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s",
+			pa_strerror(error));
+		return false;
 	}
-}
-
-void PulseAudio::StreamStateCB(pa_stream *s, void * userdata)
-{
-	switch (pa_stream_get_state(s))
+	else
 	{
-		case PA_STREAM_READY:
-		case PA_STREAM_TERMINATED:
-		case PA_STREAM_FAILED:
-			((PulseAudio *)userdata)->SignalMainLoop();
-			break;
-
-		default:
-			break;
+		NOTICE_LOG(AUDIO, "Pulse successfully initialized.");
+		return true;
 	}
 }
 
-void PulseAudio::StreamWriteCB(pa_stream *s, size_t length, void *userdata)
-{
-	((PulseAudio *)userdata)->SignalMainLoop();
-}
-
-static bool StateIsGood(pa_context *context, pa_stream *stream)
+void PulseAudio::PulseShutdown()
 {
-	if (!context || !PA_CONTEXT_IS_GOOD(pa_context_get_state(context)) ||
-			!stream || !PA_STREAM_IS_GOOD(pa_stream_get_state(stream)))
-	{
-		if ((context && pa_context_get_state(context) == PA_CONTEXT_FAILED) ||
-				(stream && pa_stream_get_state(stream) == PA_STREAM_FAILED))
-		{
-			ERROR_LOG(AUDIO, "PulseAudio state failure: %s",
-					pa_strerror(pa_context_errno(context)));
-		}
-		else
-		{
-			ERROR_LOG(AUDIO, "PulseAudio state failure: %s",
-					pa_strerror(PA_ERR_BADSTATE));
-		}
-		return false;
-	}
-	return true;
+	pa_simple_free(pa);
 }
 
-bool PulseAudio::Write(const void *data, size_t length)
+void PulseAudio::Write(const void *data, size_t length)
 {
-	if (!data || length == 0 || !stream)
-		return false;
-
-	pa_threaded_mainloop_lock(mainloop);
-
-	if (!StateIsGood(context, stream))
+	int error;
+	if (pa_simple_write(pa, data, length, &error) < 0)
 	{
-		pa_threaded_mainloop_unlock(mainloop);
-		return false;
+		ERROR_LOG(AUDIO, "PulseAudio failed to write data: %s",
+			pa_strerror(error));
 	}
-
-	while (length > 0)
-	{
-		size_t l;
-		int r;
-
-		while (!(l = pa_stream_writable_size(stream)))
-		{
-			pa_threaded_mainloop_wait(mainloop);
-			if (!StateIsGood(context, stream))
-			{
-				pa_threaded_mainloop_unlock(mainloop);
-				return false;
-			}
-		}
-
-		if (l == (size_t)-1)
-		{
-			ERROR_LOG(AUDIO, "PulseAudio invalid stream:  %s",
-					pa_strerror(pa_context_errno(context)));
-			pa_threaded_mainloop_unlock(mainloop);
-			return false;
-		}
-
-		if (l > length)
-			l = length;
-
-		r = pa_stream_write(stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
-		if (r < 0)
-		{
-			ERROR_LOG(AUDIO, "PulseAudio error writing to stream:  %s",
-					pa_strerror(pa_context_errno(context)));
-			pa_threaded_mainloop_unlock(mainloop);
-			return false;
-		}
-
-		data = (const uint8_t*) data + l;
-		length -= l;
-	}
-
-	pa_threaded_mainloop_unlock(mainloop);
-	return true;
-}
-
-void PulseAudio::SetVolume(int volume)
-{
-	iVolume = volume;
-
-	if (!stream)
-		return;
-
-	pa_cvolume cvolume;
-	const pa_channel_map *channels = pa_stream_get_channel_map(stream);
-	pa_cvolume_set(&cvolume, channels->channels,
-			iVolume * (PA_VOLUME_NORM - PA_VOLUME_MUTED) / 100);
-
-	pa_context_set_sink_input_volume(context, pa_stream_get_index(stream),
-			&cvolume, NULL, this);
 }
diff --git a/Source/Core/AudioCommon/Src/PulseAudioStream.h b/Source/Core/AudioCommon/Src/PulseAudioStream.h
index f3a5a43e79..38399e2506 100644
--- a/Source/Core/AudioCommon/Src/PulseAudioStream.h
+++ b/Source/Core/AudioCommon/Src/PulseAudioStream.h
@@ -19,7 +19,8 @@
 #define _PULSE_AUDIO_STREAM_H
 
 #if defined(HAVE_PULSEAUDIO) && HAVE_PULSEAUDIO
-#include <pulse/pulseaudio.h>
+#include <pulse/simple.h>
+#include <pulse/error.h>
 #endif
 
 #include "Common.h"
@@ -27,16 +28,16 @@
 
 #include "Thread.h"
 
+#include <vector>
+
 class PulseAudio : public SoundStream
 {
 #if defined(HAVE_PULSEAUDIO) && HAVE_PULSEAUDIO
 public:
 	PulseAudio(CMixer *mixer);
-	virtual ~PulseAudio();
 
 	virtual bool Start();
 	virtual void Stop(); 
-	virtual void SetVolume(int volume);
 
 	static bool isValid() {return true;}
 
@@ -46,22 +47,16 @@ public:
 
 private:
 	virtual void SoundLoop();
+
 	bool PulseInit();
 	void PulseShutdown();
-	bool Write(const void *data, size_t bytes);
-	void SignalMainLoop();
-	static void ContextStateCB(pa_context *c, void *userdata);
-	static void StreamStateCB(pa_stream *s, void * userdata);
-	static void StreamWriteCB(pa_stream *s, size_t length, void *userdata);
+	void Write(const void *data, size_t bytes);
 
-	u8 *mix_buffer;
+	std::vector<s16> mix_buffer;
 	std::thread thread;
-	volatile bool thread_running;
+	volatile bool run_thread;
 
-	pa_threaded_mainloop *mainloop;
-	pa_context *context;
-	pa_stream *stream;
-	int iVolume;
+	pa_simple* pa;
 #else
 public:
 	PulseAudio(CMixer *mixer) : SoundStream(mixer) {}
diff --git a/Source/Core/AudioCommon/Src/WaveFile.cpp b/Source/Core/AudioCommon/Src/WaveFile.cpp
index 802f3f7417..d98348b8bf 100644
--- a/Source/Core/AudioCommon/Src/WaveFile.cpp
+++ b/Source/Core/AudioCommon/Src/WaveFile.cpp
@@ -100,39 +100,52 @@ void WaveFileWriter::Write4(const char *ptr)
 	file.WriteBytes(ptr, 4);
 }
 
-void WaveFileWriter::AddStereoSamples(const short *sample_data, int count)
+void WaveFileWriter::AddStereoSamples(const short *sample_data, u32 count)
 {
 	if (!file)
 		PanicAlertT("WaveFileWriter - file not open.");
-	if (skip_silence) {
+	
+	if (skip_silence)
+	{
 		bool all_zero = true;
-		for (int i = 0; i < count * 2; i++)
-			if (sample_data[i]) all_zero = false;
-		if (all_zero) return;
+		
+		for (u32 i = 0; i < count * 2; i++)
+		{
+			if (sample_data[i])
+				all_zero = false;
+		}
+		
+		if (all_zero)
+			return;
 	}
+	
 	file.WriteBytes(sample_data, count * 4);
 	audio_size += count * 4;
 }
 
-void WaveFileWriter::AddStereoSamplesBE(const short *sample_data, int count)
+void WaveFileWriter::AddStereoSamplesBE(const short *sample_data, u32 count)
 {
 	if (!file)
 		PanicAlertT("WaveFileWriter - file not open.");
 
 	if (count > BUF_SIZE * 2)
-		PanicAlert("WaveFileWriter - buffer too small (count = %i).", count);
+		PanicAlert("WaveFileWriter - buffer too small (count = %u).", count);
 
 	if (skip_silence) 
 	{
 		bool all_zero = true;
-		for (int i = 0; i < count * 2; i++)
+		
+		for (u32 i = 0; i < count * 2; i++)
+		{
 			if (sample_data[i])
 				all_zero = false;
+		}
+
 		if (all_zero)
 			return;
 	}
 
-	for (int i = 0; i < count * 2; i++)
+	for (u32 i = 0; i < count * 2; i++)
 		conv_buffer[i] = Common::swap16((u16)sample_data[i]);
 
 	file.WriteBytes(conv_buffer, count * 4);
diff --git a/Source/Core/AudioCommon/Src/WaveFile.h b/Source/Core/AudioCommon/Src/WaveFile.h
index 72a090814b..11fbb4ced5 100644
--- a/Source/Core/AudioCommon/Src/WaveFile.h
+++ b/Source/Core/AudioCommon/Src/WaveFile.h
@@ -50,8 +50,8 @@ public:
 
 	void SetSkipSilence(bool skip) { skip_silence = skip; }
 
-	void AddStereoSamples(const short *sample_data, int count);
-	void AddStereoSamplesBE(const short *sample_data, int count);  // big endian
+	void AddStereoSamples(const short *sample_data, u32 count);
+	void AddStereoSamplesBE(const short *sample_data, u32 count);  // big endian
 	u32 GetAudioSize() { return audio_size; }
 };
 
diff --git a/Source/Core/AudioCommon/Src/aldlist.cpp b/Source/Core/AudioCommon/Src/aldlist.cpp
index 69d42148ae..fa11fe3282 100644
--- a/Source/Core/AudioCommon/Src/aldlist.cpp
+++ b/Source/Core/AudioCommon/Src/aldlist.cpp
@@ -48,7 +48,7 @@ ALDeviceList::ALDeviceList()
 	const char *actualDeviceName = NULL;
 
 	// DeviceInfo vector stores, for each enumerated device, it's device name, selection status, spec version #, and extension support
-	vDeviceInfo.empty();
+	vDeviceInfo.clear();
 	vDeviceInfo.reserve(10);
 
 	defaultDeviceIndex = 0;
@@ -146,12 +146,12 @@ ALDeviceList::~ALDeviceList()
 {
 	for (u32 i = 0; i < vDeviceInfo.size(); i++) {
 		if (vDeviceInfo[i].pvstrExtensions) {
-			vDeviceInfo[i].pvstrExtensions->empty();
+			vDeviceInfo[i].pvstrExtensions->clear();
 			delete vDeviceInfo[i].pvstrExtensions;
 		}
 	}
 
-	vDeviceInfo.empty();
+	vDeviceInfo.clear();
 }
 
 /*