legacy branch: merge r3209-r3382 from trunk (the translations aren't up to date with any implications this may have)legacy

git-svn-id: https://mpc-hc.svn.sourceforge.net/svnroot/mpc-hc/branches/legacy@3383 10f7b99b-c216-0410-bff0-8a66a9350fd8

27 files changed, 21 insertions, 7176 deletions

diff --git a/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.h b/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.h
index 4b515f4de..1f112c7df 100644
--- a/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.h
+++ b/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.h
@@ -34,7 +34,7 @@
 #include <Functiondiscoverykeys_devpkey.h>

 

 #include "MpcAudioRendererSettingsWnd.h"

-#include "SoundTouch/Include/SoundTouch.h"

+#include "SoundTouch.h"

 

 // REFERENCE_TIME time units per second and per millisecond

 #define REFTIMES_PER_SEC		10000000

diff --git a/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.rc b/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.rc
index 4e3f291e4..6761f8518 100644
--- a/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.rc
+++ b/src/filters/renderer/MpcAudioRenderer/MpcAudioRenderer.rc
@@ -1,7 +1,7 @@
 // Microsoft Visual C++ generated resource script.

 //

 #include "resource.h"

-#include "..\..\..\..\include\Version.h"

+#include "Version.h"

 

 #define APSTUDIO_READONLY_SYMBOLS

 /////////////////////////////////////////////////////////////////////////////

diff --git a/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcproj b/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcproj
deleted file mode 100644
index 18b7b74c2..000000000
--- a/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcproj
+++ /dev/null
@@ -1,835 +0,0 @@
-<?xml version="1.0" encoding="windows-1250"?>

-<VisualStudioProject

-	ProjectType="Visual C++"

-	Version="9,00"

-	Name="MpcAudioRendererFilter"

-	ProjectGUID="{D0620EF4-1313-40D5-9069-A82F6FE26994}"

-	RootNamespace="MpcAudioRenderer"

-	Keyword="Win32Proj"

-	TargetFrameworkVersion="131072"

-	>

-	<Platforms>

-		<Platform

-			Name="Win32"

-		/>

-		<Platform

-			Name="x64"

-		/>

-	</Platforms>

-	<ToolFiles>

-	</ToolFiles>

-	<Configurations>

-		<Configuration

-			Name="Debug Filter|Win32"

-			OutputDirectory="$(SolutionDir)bin\Filters_x86_Debug\"

-			ConfigurationType="2"

-			InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\debug.vsprops"

-			UseOfMFC="1"

-			CharacterSet="1"

-			>

-			<Tool

-				Name="VCPreBuildEventTool"

-			/>

-			<Tool

-				Name="VCCustomBuildTool"

-			/>

-			<Tool

-				Name="VCXMLDataGeneratorTool"

-			/>

-			<Tool

-				Name="VCWebServiceProxyGeneratorTool"

-			/>

-			<Tool

-				Name="VCMIDLTool"

-			/>

-			<Tool

-				Name="VCCLCompilerTool"

-				AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"

-				PreprocessorDefinitions="REGISTER_FILTER;WIN32;_DEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES"

-			/>

-			<Tool

-				Name="VCManagedResourceCompilerTool"

-			/>

-			<Tool

-				Name="VCResourceCompilerTool"

-				PreprocessorDefinitions="_UNICODE;UNICODE;WIN32"

-				Culture="1033"

-			/>

-			<Tool

-				Name="VCPreLinkEventTool"

-			/>

-			<Tool

-				Name="VCLinkerTool"

-				AdditionalDependencies="BaseClasses.lib DSUtil.lib dsound.lib Winmm.lib"

-				OutputFile="$(OutDir)\$(ProjectName).ax"

-				AdditionalLibraryDirectories="$(SolutionDir)bin\lib\Debug_$(PlatformName)"

-				IgnoreAllDefaultLibraries="false"

-				ModuleDefinitionFile="MpcAudioRenderer.def"

-				TargetMachine="1"

-			/>

-			<Tool

-				Name="VCALinkTool"

-			/>

-			<Tool

-				Name="VCManifestTool"

-			/>

-			<Tool

-				Name="VCXDCMakeTool"

-			/>

-			<Tool

-				Name="VCBscMakeTool"

-			/>

-			<Tool

-				Name="VCFxCopTool"

-			/>

-			<Tool

-				Name="VCAppVerifierTool"

-			/>

-			<Tool

-				Name="VCPostBuildEventTool"

-			/>

-		</Configuration>

-		<Configuration

-			Name="Debug Filter|x64"

-			OutputDirectory="$(SolutionDir)bin\Filters_x64_Debug\"

-			ConfigurationType="2"

-			InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\debug.vsprops"

-			UseOfMFC="1"

-			CharacterSet="1"

-			>

-			<Tool

-				Name="VCPreBuildEventTool"

-			/>

-			<Tool

-				Name="VCCustomBuildTool"

-			/>

-			<Tool

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-			/>

-			<Tool

-				Name="VCWebServiceProxyGeneratorTool"

-			/>

-			<Tool

-				Name="VCMIDLTool"

-				TargetEnvironment="3"

-			/>

-			<Tool

-				Name="VCCLCompilerTool"

-				AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"

-				PreprocessorDefinitions="REGISTER_FILTER;WIN32;_DEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES"

-				DebugInformationFormat="3"

-			/>

-			<Tool

-				Name="VCManagedResourceCompilerTool"

-			/>

-			<Tool

-				Name="VCResourceCompilerTool"

-				PreprocessorDefinitions="_UNICODE;UNICODE;_WIN64"

-				Culture="1033"

-			/>

-			<Tool

-				Name="VCPreLinkEventTool"

-			/>

-			<Tool

-				Name="VCLinkerTool"

-				AdditionalDependencies="BaseClasses.lib DSUtil.lib dsound.lib Winmm.lib"

-				OutputFile="$(OutDir)\$(ProjectName).ax"

-				AdditionalLibraryDirectories="$(SolutionDir)bin\lib\Debug_$(PlatformName)"

-				ModuleDefinitionFile="MpcAudioRenderer.def"

-				TargetMachine="17"

-			/>

-			<Tool

-				Name="VCALinkTool"

-			/>

-			<Tool

-				Name="VCManifestTool"

-			/>

-			<Tool

-				Name="VCXDCMakeTool"

-			/>

-			<Tool

-				Name="VCBscMakeTool"

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-			<Tool

-				Name="VCFxCopTool"

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-			<Tool

-				Name="VCAppVerifierTool"

-			/>

-			<Tool

-				Name="VCPostBuildEventTool"

-			/>

-		</Configuration>

-		<Configuration

-			Name="Release Filter|Win32"

-			OutputDirectory="$(SolutionDir)bin\Filters_x86\"

-			ConfigurationType="2"

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-			UseOfMFC="1"

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-			<Tool

-				Name="VCCLCompilerTool"

-				AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"

-				PreprocessorDefinitions="REGISTER_FILTER;WIN32;NDEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES"

-			/>

-			<Tool

-				Name="VCManagedResourceCompilerTool"

-			/>

-			<Tool

-				Name="VCResourceCompilerTool"

-				PreprocessorDefinitions="_UNICODE;UNICODE;WIN32"

-				Culture="1033"

-			/>

-			<Tool

-				Name="VCPreLinkEventTool"

-			/>

-			<Tool

-				Name="VCLinkerTool"

-				AdditionalDependencies="BaseClasses.lib DSUtil.lib dsound.lib Winmm.lib"

-				OutputFile="$(OutDir)\$(ProjectName).ax"

-				AdditionalLibraryDirectories="$(SolutionDir)bin\lib\Release_$(PlatformName)"

-				ModuleDefinitionFile="MpcAudioRenderer.def"

-				GenerateDebugInformation="true"

-				TargetMachine="1"

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-						ExcludedFromBuild="true"

-						>

-						<Tool

-							Name="VCCLCompilerTool"

-						/>

-					</FileConfiguration>

-					<FileConfiguration

-						Name="Release Filter|x64"

-						ExcludedFromBuild="true"

-						>

-						<Tool

-							Name="VCCLCompilerTool"

-						/>

-					</FileConfiguration>

-					<FileConfiguration

-						Name="Debug|x64"

-						ExcludedFromBuild="true"

-						>

-						<Tool

-							Name="VCCLCompilerTool"

-						/>

-					</FileConfiguration>

-					<FileConfiguration

-						Name="Release|x64"

-						ExcludedFromBuild="true"

-						>

-						<Tool

-							Name="VCCLCompilerTool"

-						/>

-					</FileConfiguration>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\FIFOSampleBuffer.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\FIRFilter.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\FIRFilter.h"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\mmx_optimized.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\PeakFinder.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\PeakFinder.h"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\RateTransposer.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\RateTransposer.h"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\SoundTouch.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\sse_optimized.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\TDStretch.cpp"

-					>

-				</File>

-				<File

-					RelativePath=".\SoundTouch\source\TDStretch.h"

-					>

-				</File>

-			</Filter>

-		</Filter>

-	</Files>

-	<Globals>

-		<Global

-			Name="DevPartner_IsInstrumented"

-			Value="0"

-		/>

-	</Globals>

-</VisualStudioProject>

diff --git a/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj b/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj
index 176eb6186..9330cd4a8 100644
--- a/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj
+++ b/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj
@@ -129,20 +129,18 @@
   </PropertyGroup>

   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug Filter|Win32'">

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>REGISTER_FILTER;WIN32;_DEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

-      <DisableSpecificWarnings>4127</DisableSpecificWarnings>

     </ClCompile>

     <Link>

-      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <OutputFile>$(OutDir)$(ProjectName)$(TargetExt)</OutputFile>

+      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;SoundTouch.lib;%(AdditionalDependencies)</AdditionalDependencies>

       <AdditionalLibraryDirectories>$(SolutionDir)bin10\lib\Debug_$(Platform);%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

-      <IgnoreAllDefaultLibraries>false</IgnoreAllDefaultLibraries>

       <ModuleDefinitionFile>MpcAudioRenderer.def</ModuleDefinitionFile>

       <TargetMachine>MachineX86</TargetMachine>

     </Link>

     <ResourceCompile>

       <PreprocessorDefinitions>_UNICODE;UNICODE;WIN32;%(PreprocessorDefinitions)</PreprocessorDefinitions>

+      <AdditionalIncludeDirectories>..\..\..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

     </ResourceCompile>

   </ItemDefinitionGroup>

   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug Filter|x64'">

@@ -150,37 +148,35 @@
       <TargetEnvironment>X64</TargetEnvironment>

     </Midl>

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>REGISTER_FILTER;WIN32;_DEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

       <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>

-      <DisableSpecificWarnings>4127</DisableSpecificWarnings>

     </ClCompile>

     <Link>

-      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <OutputFile>$(OutDir)$(ProjectName)$(TargetExt)</OutputFile>

+      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;SoundTouch.lib;%(AdditionalDependencies)</AdditionalDependencies>

       <AdditionalLibraryDirectories>$(SolutionDir)bin10\lib\Debug_$(Platform);%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

       <ModuleDefinitionFile>MpcAudioRenderer.def</ModuleDefinitionFile>

       <TargetMachine>MachineX64</TargetMachine>

     </Link>

     <ResourceCompile>

       <PreprocessorDefinitions>_UNICODE;UNICODE;_WIN64;%(PreprocessorDefinitions)</PreprocessorDefinitions>

+      <AdditionalIncludeDirectories>..\..\..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

     </ResourceCompile>

   </ItemDefinitionGroup>

   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release Filter|Win32'">

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>REGISTER_FILTER;WIN32;NDEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

-      <DisableSpecificWarnings>4127</DisableSpecificWarnings>

     </ClCompile>

     <Link>

-      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <OutputFile>$(OutDir)$(ProjectName)$(TargetExt)</OutputFile>

+      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;SoundTouch.lib;%(AdditionalDependencies)</AdditionalDependencies>

       <AdditionalLibraryDirectories>$(SolutionDir)bin10\lib\Release_$(Platform);%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

       <ModuleDefinitionFile>MpcAudioRenderer.def</ModuleDefinitionFile>

       <TargetMachine>MachineX86</TargetMachine>

     </Link>

     <ResourceCompile>

       <PreprocessorDefinitions>_UNICODE;UNICODE;WIN32;%(PreprocessorDefinitions)</PreprocessorDefinitions>

+      <AdditionalIncludeDirectories>..\..\..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

     </ResourceCompile>

   </ItemDefinitionGroup>

   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release Filter|x64'">

@@ -188,31 +184,28 @@
       <TargetEnvironment>X64</TargetEnvironment>

     </Midl>

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>_WIN64;REGISTER_FILTER;WIN32;NDEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

       <EnableEnhancedInstructionSet>NotSet</EnableEnhancedInstructionSet>

-      <DisableSpecificWarnings>4127</DisableSpecificWarnings>

     </ClCompile>

     <Link>

-      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <OutputFile>$(OutDir)$(ProjectName)$(TargetExt)</OutputFile>

+      <AdditionalDependencies>BaseClasses.lib;DSUtil.lib;dsound.lib;Winmm.lib;SoundTouch.lib;%(AdditionalDependencies)</AdditionalDependencies>

       <AdditionalLibraryDirectories>$(SolutionDir)bin10\lib\Release_$(Platform);%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

       <ModuleDefinitionFile>MpcAudioRenderer.def</ModuleDefinitionFile>

       <TargetMachine>MachineX64</TargetMachine>

     </Link>

     <ResourceCompile>

       <PreprocessorDefinitions>_UNICODE;UNICODE;_WIN64;%(PreprocessorDefinitions)</PreprocessorDefinitions>

+      <AdditionalIncludeDirectories>..\..\..\..\include;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

     </ResourceCompile>

   </ItemDefinitionGroup>

   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>WIN32;_DEBUG;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

-      <DisableSpecificWarnings>4127</DisableSpecificWarnings>

     </ClCompile>

     <Lib>

       <AdditionalDependencies>dsound.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

       <TargetMachine>MachineX86</TargetMachine>

     </Lib>

   </ItemDefinitionGroup>

@@ -221,26 +214,22 @@
       <TargetEnvironment>X64</TargetEnvironment>

     </Midl>

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>_WIN64;_DEBUG;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

       <DebugInformationFormat>ProgramDatabase</DebugInformationFormat>

-      <DisableSpecificWarnings>4127</DisableSpecificWarnings>

     </ClCompile>

     <Lib>

       <AdditionalDependencies>dsound.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

       <TargetMachine>MachineX64</TargetMachine>

     </Lib>

   </ItemDefinitionGroup>

   <ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>WIN32;NDEBUG;FLAC__NO_DLL;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

-      <DisableSpecificWarnings>4127;4244;%(DisableSpecificWarnings)</DisableSpecificWarnings>

     </ClCompile>

     <Lib>

       <AdditionalDependencies>dsound.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

       <TargetMachine>MachineX86</TargetMachine>

     </Lib>

   </ItemDefinitionGroup>

@@ -249,14 +238,12 @@
       <TargetEnvironment>X64</TargetEnvironment>

     </Midl>

     <ClCompile>

-      <AdditionalIncludeDirectories>..\..\..\..\include;SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

+      <AdditionalIncludeDirectories>..\..\..\..\include;..\..\..\thirdparty\SoundTouch\include;..\..\BaseClasses;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>

       <PreprocessorDefinitions>_WIN64;NDEBUG;FLAC__NO_DLL;SOUNDTOUCH_INTEGER_SAMPLES;%(PreprocessorDefinitions)</PreprocessorDefinitions>

       <EnableEnhancedInstructionSet>NotSet</EnableEnhancedInstructionSet>

-      <DisableSpecificWarnings>4127</DisableSpecificWarnings>

     </ClCompile>

     <Lib>

       <AdditionalDependencies>dsound.lib;%(AdditionalDependencies)</AdditionalDependencies>

-      <AdditionalLibraryDirectories>%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>

       <TargetMachine>MachineX64</TargetMachine>

     </Lib>

   </ItemDefinitionGroup>

@@ -286,22 +273,6 @@
       <PrecompiledHeader Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">Create</PrecompiledHeader>

       <PrecompiledHeader Condition="'$(Configuration)|$(Platform)'=='Release|x64'">Create</PrecompiledHeader>

     </ClCompile>

-    <ClCompile Include="SoundTouch\source\AAFilter.cpp" />

-    <ClCompile Include="SoundTouch\source\BPMDetect.cpp" />

-    <ClCompile Include="SoundTouch\source\cpu_detect_x86_win.cpp">

-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug Filter|x64'">true</ExcludedFromBuild>

-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</ExcludedFromBuild>

-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release Filter|x64'">true</ExcludedFromBuild>

-      <ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|x64'">true</ExcludedFromBuild>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\FIFOSampleBuffer.cpp" />

-    <ClCompile Include="SoundTouch\source\FIRFilter.cpp" />

-    <ClCompile Include="SoundTouch\source\mmx_optimized.cpp" />

-    <ClCompile Include="SoundTouch\source\PeakFinder.cpp" />

-    <ClCompile Include="SoundTouch\source\RateTransposer.cpp" />

-    <ClCompile Include="SoundTouch\source\SoundTouch.cpp" />

-    <ClCompile Include="SoundTouch\source\sse_optimized.cpp" />

-    <ClCompile Include="SoundTouch\source\TDStretch.cpp" />

   </ItemGroup>

   <ItemGroup>

     <None Include="MpcAudioRenderer.def" />

@@ -312,17 +283,6 @@
     <ClInclude Include="MpcAudioRendererSettingsWnd.h" />

     <ClInclude Include="resource.h" />

     <ClInclude Include="stdafx.h" />

-    <ClInclude Include="SoundTouch\include\BPMDetect.h" />

-    <ClInclude Include="SoundTouch\include\FIFOSampleBuffer.h" />

-    <ClInclude Include="SoundTouch\include\FIFOSamplePipe.h" />

-    <ClInclude Include="SoundTouch\include\SoundTouch.h" />

-    <ClInclude Include="SoundTouch\include\STTypes.h" />

-    <ClInclude Include="SoundTouch\source\AAFilter.h" />

-    <ClInclude Include="SoundTouch\source\cpu_detect.h" />

-    <ClInclude Include="SoundTouch\source\FIRFilter.h" />

-    <ClInclude Include="SoundTouch\source\PeakFinder.h" />

-    <ClInclude Include="SoundTouch\source\RateTransposer.h" />

-    <ClInclude Include="SoundTouch\source\TDStretch.h" />

   </ItemGroup>

   <ItemGroup>

     <ResourceCompile Include="MpcAudioRenderer.rc">

@@ -337,6 +297,9 @@
       <Project>{fc70988b-1ae5-4381-866d-4f405e28ac42}</Project>

       <ReferenceOutputAssembly>false</ReferenceOutputAssembly>

     </ProjectReference>

+    <ProjectReference Include="..\..\..\thirdparty\SoundTouch\source\SoundTouch.vcxproj">

+      <Project>{68a5dd20-7057-448b-8fe0-b6ac8d205509}</Project>

+    </ProjectReference>

     <ProjectReference Include="..\..\BaseClasses\BaseClasses.vcxproj">

       <Project>{e8a3f6fa-ae1c-4c8e-a0b6-9c8480324eaa}</Project>

       <ReferenceOutputAssembly>false</ReferenceOutputAssembly>

@@ -349,9 +312,4 @@
   <Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />

   <ImportGroup Label="ExtensionTargets">

   </ImportGroup>

-  <ProjectExtensions>

-    <VisualStudio>

-      <UserProperties DevPartner_IsInstrumented="0" />

-    </VisualStudio>

-  </ProjectExtensions>

 </Project>
\ No newline at end of file
diff --git a/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj.filters b/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj.filters
index 9610a5437..5808697a9 100644
--- a/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj.filters
+++ b/src/filters/renderer/MpcAudioRenderer/MpcAudioRendererFilter.vcxproj.filters
@@ -13,15 +13,6 @@
       <UniqueIdentifier>{addff0d6-0d94-4e4c-b5ea-b7d5b330fd45}</UniqueIdentifier>

       <Extensions>rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe</Extensions>

     </Filter>

-    <Filter Include="SoundTouch">

-      <UniqueIdentifier>{943d1fc2-8906-4d65-9623-b21a9dad5bf1}</UniqueIdentifier>

-    </Filter>

-    <Filter Include="SoundTouch\include">

-      <UniqueIdentifier>{03c6a7a0-ca37-40d3-b5f2-84aba4953caf}</UniqueIdentifier>

-    </Filter>

-    <Filter Include="SoundTouch\source">

-      <UniqueIdentifier>{2b74773f-75dc-4818-9c11-89e77d1ac369}</UniqueIdentifier>

-    </Filter>

   </ItemGroup>

   <ItemGroup>

     <ClCompile Include="..\..\FilterApp.cpp">

@@ -33,39 +24,6 @@
     <ClCompile Include="stdafx.cpp">

       <Filter>Source Files</Filter>

     </ClCompile>

-    <ClCompile Include="SoundTouch\source\AAFilter.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\BPMDetect.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\cpu_detect_x86_win.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\FIFOSampleBuffer.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\FIRFilter.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\mmx_optimized.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\PeakFinder.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\RateTransposer.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\SoundTouch.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\sse_optimized.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

-    <ClCompile Include="SoundTouch\source\TDStretch.cpp">

-      <Filter>SoundTouch\source</Filter>

-    </ClCompile>

     <ClCompile Include="MpcAudioRendererSettingsWnd.cpp">

       <Filter>Source Files</Filter>

     </ClCompile>

@@ -94,39 +52,6 @@
     <ClInclude Include="stdafx.h">

       <Filter>Header Files</Filter>

     </ClInclude>

-    <ClInclude Include="SoundTouch\include\BPMDetect.h">

-      <Filter>SoundTouch\include</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\include\FIFOSampleBuffer.h">

-      <Filter>SoundTouch\include</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\include\FIFOSamplePipe.h">

-      <Filter>SoundTouch\include</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\include\SoundTouch.h">

-      <Filter>SoundTouch\include</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\include\STTypes.h">

-      <Filter>SoundTouch\include</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\source\AAFilter.h">

-      <Filter>SoundTouch\source</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\source\cpu_detect.h">

-      <Filter>SoundTouch\source</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\source\FIRFilter.h">

-      <Filter>SoundTouch\source</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\source\PeakFinder.h">

-      <Filter>SoundTouch\source</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\source\RateTransposer.h">

-      <Filter>SoundTouch\source</Filter>

-    </ClInclude>

-    <ClInclude Include="SoundTouch\source\TDStretch.h">

-      <Filter>SoundTouch\source</Filter>

-    </ClInclude>

     <ClInclude Include="IMpcAudioRendererFilter.h">

       <Filter>Header Files</Filter>

     </ClInclude>

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/BPMDetect.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/BPMDetect.h
deleted file mode 100644
index ff1d3c44f..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/BPMDetect.h
+++ /dev/null
@@ -1,170 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// Beats-per-minute (BPM) detection routine.

-///

-/// The beat detection algorithm works as follows:

-/// - Use function 'inputSamples' to input a chunks of samples to the class for

-///   analysis. It's a good idea to enter a large sound file or stream in smallish

-///   chunks of around few kilosamples in order not to extinguish too much RAM memory.

-/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,

-///   which is basically ok as low (bass) frequencies mostly determine the beat rate.

-///   Simple averaging is used for anti-alias filtering because the resulting signal

-///   quality isn't of that high importance.

-/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by

-///   taking absolute value that's smoothed by sliding average. Signal levels that

-///   are below a couple of times the general RMS amplitude level are cut away to

-///   leave only notable peaks there.

-/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term 

-///   autocorrelation function of the enveloped signal.

-/// - After whole sound data file has been analyzed as above, the bpm level is 

-///   detected by function 'getBpm' that finds the highest peak of the autocorrelation 

-///   function, calculates it's precise location and converts this reading to bpm's.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef _BPMDetect_H_

-#define _BPMDetect_H_

-

-#include "STTypes.h"

-#include "FIFOSampleBuffer.h"

-

-namespace soundtouch

-{

-

-/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.

-#define MIN_BPM 29

-

-/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.

-#define MAX_BPM 230

-

-

-/// Class for calculating BPM rate for audio data.

-class BPMDetect

-{

-protected:

-    /// Auto-correlation accumulator bins.

-    float *xcorr;

-    

-    /// Amplitude envelope sliding average approximation level accumulator

-    double envelopeAccu;

-

-    /// RMS volume sliding average approximation level accumulator

-    double RMSVolumeAccu;

-

-    /// Level below which to cut off signals

-    double cutCoeff;

-

-    /// Accumulator for accounting what proportion of samples exceed cutCoeff level

-    double aboveCutAccu;

-

-    /// Accumulator for total samples to calculate proportion of samples that exceed cutCoeff level

-    double totalAccu;

-

-    /// Sample average counter.

-    int decimateCount;

-

-    /// Sample average accumulator for FIFO-like decimation.

-    soundtouch::LONG_SAMPLETYPE decimateSum;

-

-    /// Decimate sound by this coefficient to reach approx. 500 Hz.

-    int decimateBy;

-

-    /// Auto-correlation window length

-    int windowLen;

-

-    /// Number of channels (1 = mono, 2 = stereo)

-    int channels;

-

-    /// sample rate

-    int sampleRate;

-

-    /// Beginning of auto-correlation window: Autocorrelation isn't being updated for

-    /// the first these many correlation bins.

-    int windowStart;

- 

-    /// FIFO-buffer for decimated processing samples.

-    soundtouch::FIFOSampleBuffer *buffer;

-

-    /// Updates auto-correlation function for given number of decimated samples that 

-    /// are read from the internal 'buffer' pipe (samples aren't removed from the pipe 

-    /// though).

-    void updateXCorr(int process_samples      /// How many samples are processed.

-                     );

-

-    /// Decimates samples to approx. 500 Hz.

-    ///

-    /// \return Number of output samples.

-    int decimate(soundtouch::SAMPLETYPE *dest,      ///< Destination buffer

-                 const soundtouch::SAMPLETYPE *src, ///< Source sample buffer

-                 int numsamples                     ///< Number of source samples.

-                 );

-

-    /// Calculates amplitude envelope for the buffer of samples.

-    /// Result is output to 'samples'.

-    void calcEnvelope(soundtouch::SAMPLETYPE *samples,  ///< Pointer to input/output data buffer

-                      int numsamples                    ///< Number of samples in buffer

-                      );

-

-public:

-    /// Constructor.

-    BPMDetect(int numChannels,  ///< Number of channels in sample data.

-              int sampleRate    ///< Sample rate in Hz.

-              );

-

-    /// Destructor.

-    virtual ~BPMDetect();

-

-    /// Inputs a block of samples for analyzing: Envelopes the samples and then

-    /// updates the autocorrelation estimation. When whole song data has been input

-    /// in smaller blocks using this function, read the resulting bpm with 'getBpm' 

-    /// function. 

-    /// 

-    /// Notice that data in 'samples' array can be disrupted in processing.

-    void inputSamples(const soundtouch::SAMPLETYPE *samples,    ///< Pointer to input/working data buffer

-                      int numSamples                            ///< Number of samples in buffer

-                      );

-

-

-    /// Analyzes the results and returns the BPM rate. Use this function to read result

-    /// after whole song data has been input to the class by consecutive calls of

-    /// 'inputSamples' function.

-    ///

-    /// \return Beats-per-minute rate, or zero if detection failed.

-    float getBpm();

-};

-

-}

-

-#endif // _BPMDetect_H_

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/FIFOSampleBuffer.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/FIFOSampleBuffer.h
deleted file mode 100644
index e69918ac2..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/FIFOSampleBuffer.h
+++ /dev/null
@@ -1,174 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// A buffer class for temporarily storaging sound samples, operates as a 

-/// first-in-first-out pipe.

-///

-/// Samples are added to the end of the sample buffer with the 'putSamples' 

-/// function, and are received from the beginning of the buffer by calling

-/// the 'receiveSamples' function. The class automatically removes the 

-/// output samples from the buffer as well as grows the storage size 

-/// whenever necessary.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef FIFOSampleBuffer_H

-#define FIFOSampleBuffer_H

-

-#include "FIFOSamplePipe.h"

-

-namespace soundtouch

-{

-

-/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes

-/// care of storage size adjustment and data moving during input/output operations.

-///

-/// Notice that in case of stereo audio, one sample is considered to consist of 

-/// both channel data.

-class FIFOSampleBuffer : public FIFOSamplePipe

-{

-private:

-    /// Sample buffer.

-    SAMPLETYPE *buffer;

-

-    // Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first

-    // 16-byte aligned location of this buffer

-    SAMPLETYPE *bufferUnaligned;

-

-    /// Sample buffer size in bytes

-    uint sizeInBytes;

-

-    /// How many samples are currently in buffer.

-    uint samplesInBuffer;

-

-    /// Channels, 1=mono, 2=stereo.

-    uint channels;

-

-    /// Current position pointer to the buffer. This pointer is increased when samples are 

-    /// removed from the pipe so that it's necessary to actually rewind buffer (move data)

-    /// only new data when is put to the pipe.

-    uint bufferPos;

-

-    /// Rewind the buffer by moving data from position pointed by 'bufferPos' to real 

-    /// beginning of the buffer.

-    void rewind();

-

-    /// Ensures that the buffer has capacity for at least this many samples.

-    void ensureCapacity(uint capacityRequirement);

-

-    /// Returns current capacity.

-    uint getCapacity() const;

- 

-public:

-

-    /// Constructor

-    FIFOSampleBuffer(int numChannels = 2     ///< Number of channels, 1=mono, 2=stereo.

-                                              ///< Default is stereo.

-                     );

-

-    /// destructor

-    ~FIFOSampleBuffer();

-

-    /// Returns a pointer to the beginning of the output samples. 

-    /// This function is provided for accessing the output samples directly. 

-    /// Please be careful for not to corrupt the book-keeping!

-    ///

-    /// When using this function to output samples, also remember to 'remove' the

-    /// output samples from the buffer by calling the 

-    /// 'receiveSamples(numSamples)' function

-    virtual SAMPLETYPE *ptrBegin();

-

-    /// Returns a pointer to the end of the used part of the sample buffer (i.e. 

-    /// where the new samples are to be inserted). This function may be used for 

-    /// inserting new samples into the sample buffer directly. Please be careful

-    /// not corrupt the book-keeping!

-    ///

-    /// When using this function as means for inserting new samples, also remember 

-    /// to increase the sample count afterwards, by calling  the 

-    /// 'putSamples(numSamples)' function.

-    SAMPLETYPE *ptrEnd(

-                uint slackCapacity   ///< How much free capacity (in samples) there _at least_ 

-                                     ///< should be so that the caller can succesfully insert the 

-                                     ///< desired samples to the buffer. If necessary, the function 

-                                     ///< grows the buffer size to comply with this requirement.

-                );

-

-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position to

-    /// the sample buffer.

-    virtual void putSamples(const SAMPLETYPE *samples,  ///< Pointer to samples.

-                            uint numSamples                         ///< Number of samples to insert.

-                            );

-

-    /// Adjusts the book-keeping to increase number of samples in the buffer without 

-    /// copying any actual samples.

-    ///

-    /// This function is used to update the number of samples in the sample buffer

-    /// when accessing the buffer directly with 'ptrEnd' function. Please be 

-    /// careful though!

-    virtual void putSamples(uint numSamples   ///< Number of samples been inserted.

-                            );

-

-    /// Output samples from beginning of the sample buffer. Copies requested samples to 

-    /// output buffer and removes them from the sample buffer. If there are less than 

-    /// 'numsample' samples in the buffer, returns all that available.

-    ///

-    /// \return Number of samples returned.

-    virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.

-                                uint maxSamples                 ///< How many samples to receive at max.

-                                );

-

-    /// Adjusts book-keeping so that given number of samples are removed from beginning of the 

-    /// sample buffer without copying them anywhere. 

-    ///

-    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly

-    /// with 'ptrBegin' function.

-    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.

-                                );

-

-    /// Returns number of samples currently available.

-    virtual uint numSamples() const;

-

-    /// Sets number of channels, 1 = mono, 2 = stereo.

-    void setChannels(int numChannels);

-

-    /// Returns nonzero if there aren't any samples available for outputting.

-    virtual int isEmpty() const;

-

-    /// Clears all the samples.

-    virtual void clear();

-};

-

-}

-

-#endif

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/FIFOSamplePipe.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/FIFOSamplePipe.h
deleted file mode 100644
index ad982cbba..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/FIFOSamplePipe.h
+++ /dev/null
@@ -1,221 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound

-/// samples by operating like a first-in-first-out pipe: New samples are fed

-/// into one end of the pipe with the 'putSamples' function, and the processed

-/// samples are received from the other end with the 'receiveSamples' function.

-///

-/// 'FIFOProcessor' : A base class for classes the do signal processing with 

-/// the samples while operating like a first-in-first-out pipe. When samples

-/// are input with the 'putSamples' function, the class processes them

-/// and moves the processed samples to the given 'output' pipe object, which

-/// may be either another processing stage, or a fifo sample buffer object.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef FIFOSamplePipe_H

-#define FIFOSamplePipe_H

-

-#include <assert.h>

-#include <stdlib.h>

-#include "STTypes.h"

-

-namespace soundtouch

-{

-

-/// Abstract base class for FIFO (first-in-first-out) sample processing classes.

-class FIFOSamplePipe

-{

-public:

-    // virtual default destructor

-    virtual ~FIFOSamplePipe() {}

-

-

-    /// Returns a pointer to the beginning of the output samples. 

-    /// This function is provided for accessing the output samples directly. 

-    /// Please be careful for not to corrupt the book-keeping!

-    ///

-    /// When using this function to output samples, also remember to 'remove' the

-    /// output samples from the buffer by calling the 

-    /// 'receiveSamples(numSamples)' function

-    virtual SAMPLETYPE *ptrBegin() = 0;

-

-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position to

-    /// the sample buffer.

-    virtual void putSamples(const SAMPLETYPE *samples,  ///< Pointer to samples.

-                            uint numSamples             ///< Number of samples to insert.

-                            ) = 0;

-

-

-    // Moves samples from the 'other' pipe instance to this instance.

-    void moveSamples(FIFOSamplePipe &other  ///< Other pipe instance where from the receive the data.

-         )

-    {

-        int oNumSamples = other.numSamples();

-

-        putSamples(other.ptrBegin(), oNumSamples);

-        other.receiveSamples(oNumSamples);

-    };

-

-    /// Output samples from beginning of the sample buffer. Copies requested samples to 

-    /// output buffer and removes them from the sample buffer. If there are less than 

-    /// 'numsample' samples in the buffer, returns all that available.

-    ///

-    /// \return Number of samples returned.

-    virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.

-                                uint maxSamples                 ///< How many samples to receive at max.

-                                ) = 0;

-

-    /// Adjusts book-keeping so that given number of samples are removed from beginning of the 

-    /// sample buffer without copying them anywhere. 

-    ///

-    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly

-    /// with 'ptrBegin' function.

-    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.

-                                ) = 0;

-

-    /// Returns number of samples currently available.

-    virtual uint numSamples() const = 0;

-

-    // Returns nonzero if there aren't any samples available for outputting.

-    virtual int isEmpty() const = 0;

-

-    /// Clears all the samples.

-    virtual void clear() = 0;

-};

-

-

-

-/// Base-class for sound processing routines working in FIFO principle. With this base 

-/// class it's easy to implement sound processing stages that can be chained together,

-/// so that samples that are fed into beginning of the pipe automatically go through 

-/// all the processing stages.

-///

-/// When samples are input to this class, they're first processed and then put to 

-/// the FIFO pipe that's defined as output of this class. This output pipe can be

-/// either other processing stage or a FIFO sample buffer.

-class FIFOProcessor :public FIFOSamplePipe

-{

-protected:

-    /// Internal pipe where processed samples are put.

-    FIFOSamplePipe *output;

-

-    /// Sets output pipe.

-    void setOutPipe(FIFOSamplePipe *pOutput)

-    {

-        assert(output == NULL);

-        assert(pOutput != NULL);

-        output = pOutput;

-    }

-

-

-    /// Constructor. Doesn't define output pipe; it has to be set be 

-    /// 'setOutPipe' function.

-    FIFOProcessor()

-    {

-        output = NULL;

-    }

-

-

-    /// Constructor. Configures output pipe.

-    FIFOProcessor(FIFOSamplePipe *pOutput   ///< Output pipe.

-                 )

-    {

-        output = pOutput;

-    }

-

-

-    /// Destructor.

-    virtual ~FIFOProcessor()

-    {

-    }

-

-

-    /// Returns a pointer to the beginning of the output samples. 

-    /// This function is provided for accessing the output samples directly. 

-    /// Please be careful for not to corrupt the book-keeping!

-    ///

-    /// When using this function to output samples, also remember to 'remove' the

-    /// output samples from the buffer by calling the 

-    /// 'receiveSamples(numSamples)' function

-    virtual SAMPLETYPE *ptrBegin()

-    {

-        return output->ptrBegin();

-    }

-

-public:

-

-    /// Output samples from beginning of the sample buffer. Copies requested samples to 

-    /// output buffer and removes them from the sample buffer. If there are less than 

-    /// 'numsample' samples in the buffer, returns all that available.

-    ///

-    /// \return Number of samples returned.

-    virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.

-                                uint maxSamples                    ///< How many samples to receive at max.

-                                )

-    {

-        return output->receiveSamples(outBuffer, maxSamples);

-    }

-

-

-    /// Adjusts book-keeping so that given number of samples are removed from beginning of the 

-    /// sample buffer without copying them anywhere. 

-    ///

-    /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly

-    /// with 'ptrBegin' function.

-    virtual uint receiveSamples(uint maxSamples   ///< Remove this many samples from the beginning of pipe.

-                                )

-    {

-        return output->receiveSamples(maxSamples);

-    }

-

-

-    /// Returns number of samples currently available.

-    virtual uint numSamples() const

-    {

-        return output->numSamples();

-    }

-

-

-    /// Returns nonzero if there aren't any samples available for outputting.

-    virtual int isEmpty() const

-    {

-        return output->isEmpty();

-    }

-};

-

-}

-

-#endif

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/STTypes.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/STTypes.h
deleted file mode 100644
index 98186a552..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/STTypes.h
+++ /dev/null
@@ -1,145 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// Common type definitions for SoundTouch audio processing library.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 3 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef STTypes_H

-#define STTypes_H

-

-typedef unsigned int    uint;

-typedef unsigned long   ulong;

-

-#ifdef __GNUC__

-    // In GCC, include soundtouch_config.h made by config scritps

-    #include "soundtouch_config.h"

-#endif

-

-#ifndef _WINDEF_

-    // if these aren't defined already by Windows headers, define now

-

-    typedef int BOOL;

-

-    #define FALSE   0

-    #define TRUE    1

-

-#endif  // _WINDEF_

-

-

-namespace soundtouch

-{

-

-/// Activate these undef's to overrule the possible sampletype 

-/// setting inherited from some other header file:

-//#undef SOUNDTOUCH_INTEGER_SAMPLES

-//#undef SOUNDTOUCH_FLOAT_SAMPLES

-

-#if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES)

-   

-    /// Choose either 32bit floating point or 16bit integer sampletype

-    /// by choosing one of the following defines, unless this selection 

-    /// has already been done in some other file.

-    ////

-    /// Notes:

-    /// - In Windows environment, choose the sample format with the

-    ///   following defines.

-    /// - In GNU environment, the floating point samples are used by 

-    ///   default, but integer samples can be chosen by giving the 

-    ///   following switch to the configure script:

-    ///       ./configure --enable-integer-samples

-    ///   However, if you still prefer to select the sample format here 

-    ///   also in GNU environment, then please #undef the INTEGER_SAMPLE

-    ///   and FLOAT_SAMPLE defines first as in comments above.

-    //#define SOUNDTOUCH_INTEGER_SAMPLES     1    //< 16bit integer samples

-    #define SOUNDTOUCH_FLOAT_SAMPLES       1    //< 32bit float samples

- 

- #endif

-

-    #ifndef _WIN64 //mpc custom code

-        /// Define this to allow X86-specific assembler/intrinsic optimizations. 

-        /// Notice that library contains also usual C++ versions of each of these

-        /// these routines, so if you're having difficulties getting the optimized 

-        /// routines compiled for whatever reason, you may disable these optimizations 

-        /// to make the library compile.

-

-        #define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS     1

-

-    #endif

-

-    // If defined, allows the SIMD-optimized routines to take minor shortcuts 

-    // for improved performance. Undefine to require faithfully similar SIMD 

-    // calculations as in normal C implementation.

-    #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION    1

-

-

-    #ifdef SOUNDTOUCH_INTEGER_SAMPLES

-        // 16bit integer sample type

-        typedef short SAMPLETYPE;

-        // data type for sample accumulation: Use 32bit integer to prevent overflows

-        typedef long  LONG_SAMPLETYPE;

-

-        #ifdef SOUNDTOUCH_FLOAT_SAMPLES

-            // check that only one sample type is defined

-            #error "conflicting sample types defined"

-        #endif // SOUNDTOUCH_FLOAT_SAMPLES

-

-        #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS

-            // Allow MMX optimizations

-            #define SOUNDTOUCH_ALLOW_MMX   1

-        #endif

-

-    #else

-

-        // floating point samples

-        typedef float  SAMPLETYPE;

-        // data type for sample accumulation: Use double to utilize full precision.

-        typedef double LONG_SAMPLETYPE;

-

-        #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS

-            // Allow SSE optimizations

-            #define SOUNDTOUCH_ALLOW_SSE       1

-        #endif

-

-    #endif  // SOUNDTOUCH_INTEGER_SAMPLES

-};

-

-

-// When this #define is active, eliminates a clicking sound when the "rate" or "pitch" 

-// parameter setting crosses from value <1 to >=1 or vice versa during processing. 

-// Default is off as such crossover is untypical case and involves a slight sound 

-// quality compromise.

-//#define SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER   1

-

-#endif

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/SoundTouch.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/SoundTouch.h
deleted file mode 100644
index 203d84fc4..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/Include/SoundTouch.h
+++ /dev/null
@@ -1,277 +0,0 @@
-//////////////////////////////////////////////////////////////////////////////

-///

-/// SoundTouch - main class for tempo/pitch/rate adjusting routines. 

-///

-/// Notes:

-/// - Initialize the SoundTouch object instance by setting up the sound stream 

-///   parameters with functions 'setSampleRate' and 'setChannels', then set 

-///   desired tempo/pitch/rate settings with the corresponding functions.

-///

-/// - The SoundTouch class behaves like a first-in-first-out pipeline: The 

-///   samples that are to be processed are fed into one of the pipe by calling

-///   function 'putSamples', while the ready processed samples can be read 

-///   from the other end of the pipeline with function 'receiveSamples'.

-/// 

-/// - The SoundTouch processing classes require certain sized 'batches' of 

-///   samples in order to process the sound. For this reason the classes buffer 

-///   incoming samples until there are enough of samples available for 

-///   processing, then they carry out the processing step and consequently

-///   make the processed samples available for outputting.

-/// 

-/// - For the above reason, the processing routines introduce a certain 

-///   'latency' between the input and output, so that the samples input to

-///   SoundTouch may not be immediately available in the output, and neither 

-///   the amount of outputtable samples may not immediately be in direct 

-///   relationship with the amount of previously input samples.

-///

-/// - The tempo/pitch/rate control parameters can be altered during processing.

-///   Please notice though that they aren't currently protected by semaphores,

-///   so in multi-thread application external semaphore protection may be

-///   required.

-///

-/// - This class utilizes classes 'TDStretch' for tempo change (without modifying

-///   pitch) and 'RateTransposer' for changing the playback rate (that is, both 

-///   tempo and pitch in the same ratio) of the sound. The third available control 

-///   'pitch' (change pitch but maintain tempo) is produced by a combination of

-///   combining the two other controls.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef SoundTouch_H

-#define SoundTouch_H

-

-#include "FIFOSamplePipe.h"

-#include "STTypes.h"

-

-namespace soundtouch

-{

-

-/// Soundtouch library version string

-#define SOUNDTOUCH_VERSION          "1.5.1pre"

-

-/// SoundTouch library version id

-#define SOUNDTOUCH_VERSION_ID       (10509)

-

-//

-// Available setting IDs for the 'setSetting' & 'get_setting' functions:

-

-/// Enable/disable anti-alias filter in pitch transposer (0 = disable)

-#define SETTING_USE_AA_FILTER       0

-

-/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)

-#define SETTING_AA_FILTER_LENGTH    1

-

-/// Enable/disable quick seeking algorithm in tempo changer routine

-/// (enabling quick seeking lowers CPU utilization but causes a minor sound

-///  quality compromising)

-#define SETTING_USE_QUICKSEEK       2

-

-/// Time-stretch algorithm single processing sequence length in milliseconds. This determines 

-/// to how long sequences the original sound is chopped in the time-stretch algorithm. 

-/// See "STTypes.h" or README for more information.

-#define SETTING_SEQUENCE_MS         3

-

-/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the 

-/// best possible overlapping location. This determines from how wide window the algorithm 

-/// may look for an optimal joining location when mixing the sound sequences back together. 

-/// See "STTypes.h" or README for more information.

-#define SETTING_SEEKWINDOW_MS       4

-

-/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences 

-/// are mixed back together, to form a continuous sound stream, this parameter defines over 

-/// how long period the two consecutive sequences are let to overlap each other. 

-/// See "STTypes.h" or README for more information.

-#define SETTING_OVERLAP_MS          5

-

-

-/// Call "getSetting" with this ID to query nominal average processing sequence

-/// size in samples. This value tells approcimate value how many input samples 

-/// SoundTouch needs to gather before it does DSP processing run for the sample batch.

-///

-/// Notices: 

-/// - This is read-only parameter, i.e. setSetting ignores this parameter

-/// - Returned value is approximate average value, exact processing batch

-///   size may wary from time to time

-/// - This parameter value is not constant but may change depending on 

-///   tempo/pitch/rate/samplerate settings.

-#define SETTING_NOMINAL_INPUT_SEQUENCE		6

-

-

-/// Call "getSetting" with this ID to query nominal average processing output 

-/// size in samples. This value tells approcimate value how many output samples 

-/// SoundTouch outputs once it does DSP processing run for a batch of input samples.

-///	

-/// Notices: 

-/// - This is read-only parameter, i.e. setSetting ignores this parameter

-/// - Returned value is approximate average value, exact processing batch

-///   size may wary from time to time

-/// - This parameter value is not constant but may change depending on 

-///   tempo/pitch/rate/samplerate settings.

-#define SETTING_NOMINAL_OUTPUT_SEQUENCE		7

-

-class SoundTouch : public FIFOProcessor

-{

-private:

-    /// Rate transposer class instance

-    class RateTransposer *pRateTransposer;

-

-    /// Time-stretch class instance

-    class TDStretch *pTDStretch;

-

-    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.

-    float virtualRate;

-

-    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.

-    float virtualTempo;

-

-    /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.

-    float virtualPitch;

-

-    /// Flag: Has sample rate been set?

-    BOOL  bSrateSet;

-

-    /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and 

-    /// 'virtualPitch' parameters.

-    void calcEffectiveRateAndTempo();

-

-protected :

-    /// Number of channels

-    uint  channels;

-

-    /// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'

-    float rate;

-

-    /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'

-    float tempo;

-

-public:

-    SoundTouch();

-    virtual ~SoundTouch();

-

-    /// Get SoundTouch library version string

-    static const char *getVersionString();

-

-    /// Get SoundTouch library version Id

-    static uint getVersionId();

-

-    /// Sets new rate control value. Normal rate = 1.0, smaller values

-    /// represent slower rate, larger faster rates.

-    void setRate(float newRate);

-

-    /// Sets new tempo control value. Normal tempo = 1.0, smaller values

-    /// represent slower tempo, larger faster tempo.

-    void setTempo(float newTempo);

-

-    /// Sets new rate control value as a difference in percents compared

-    /// to the original rate (-50 .. +100 %)

-    void setRateChange(float newRate);

-

-    /// Sets new tempo control value as a difference in percents compared

-    /// to the original tempo (-50 .. +100 %)

-    void setTempoChange(float newTempo);

-

-    /// Sets new pitch control value. Original pitch = 1.0, smaller values

-    /// represent lower pitches, larger values higher pitch.

-    void setPitch(float newPitch);

-

-    /// Sets pitch change in octaves compared to the original pitch  

-    /// (-1.00 .. +1.00)

-    void setPitchOctaves(float newPitch);

-

-    /// Sets pitch change in semi-tones compared to the original pitch

-    /// (-12 .. +12)

-    void setPitchSemiTones(int newPitch);

-    void setPitchSemiTones(float newPitch);

-

-    /// Sets the number of channels, 1 = mono, 2 = stereo

-    void setChannels(uint numChannels);

-

-    /// Sets sample rate.

-    void setSampleRate(uint srate);

-

-    /// Flushes the last samples from the processing pipeline to the output.

-    /// Clears also the internal processing buffers.

-    //

-    /// Note: This function is meant for extracting the last samples of a sound

-    /// stream. This function may introduce additional blank samples in the end

-    /// of the sound stream, and thus it's not recommended to call this function

-    /// in the middle of a sound stream.

-    void flush();

-

-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position into

-    /// the input of the object. Notice that sample rate _has_to_ be set before

-    /// calling this function, otherwise throws a runtime_error exception.

-    virtual void putSamples(

-            const SAMPLETYPE *samples,  ///< Pointer to sample buffer.

-            uint numSamples                         ///< Number of samples in buffer. Notice

-                                                    ///< that in case of stereo-sound a single sample

-                                                    ///< contains data for both channels.

-            );

-

-    /// Clears all the samples in the object's output and internal processing

-    /// buffers.

-    virtual void clear();

-

-    /// Changes a setting controlling the processing system behaviour. See the

-    /// 'SETTING_...' defines for available setting ID's.

-    /// 

-    /// \return 'TRUE' if the setting was succesfully changed

-    BOOL setSetting(int settingId,   ///< Setting ID number. see SETTING_... defines.

-                    int value        ///< New setting value.

-                    );

-

-    /// Reads a setting controlling the processing system behaviour. See the

-    /// 'SETTING_...' defines for available setting ID's.

-    ///

-    /// \return the setting value.

-    int getSetting(int settingId    ///< Setting ID number, see SETTING_... defines.

-                   ) const;

-

-    /// Returns number of samples currently unprocessed.

-    virtual uint numUnprocessedSamples() const;

-

-

-    /// Other handy functions that are implemented in the ancestor classes (see

-    /// classes 'FIFOProcessor' and 'FIFOSamplePipe')

-    ///

-    /// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.

-    /// - numSamples()     : Get number of 'ready' samples that can be received with 

-    ///                      function 'receiveSamples()'

-    /// - isEmpty()        : Returns nonzero if there aren't any 'ready' samples.

-    /// - clear()          : Clears all samples from ready/processing buffers.

-};

-

-}

-#endif

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/AAFilter.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/AAFilter.cpp
deleted file mode 100644
index 191b97b13..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/AAFilter.cpp
+++ /dev/null
@@ -1,184 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// FIR low-pass (anti-alias) filter with filter coefficient design routine and

-/// MMX optimization. 

-/// 

-/// Anti-alias filter is used to prevent folding of high frequencies when 

-/// transposing the sample rate with interpolation.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <memory.h>

-#include <assert.h>

-#include <math.h>

-#include <stdlib.h>

-#include "AAFilter.h"

-#include "FIRFilter.h"

-

-using namespace soundtouch;

-

-#define PI        3.141592655357989

-#define TWOPI    (2 * PI)

-

-/*****************************************************************************

- *

- * Implementation of the class 'AAFilter'

- *

- *****************************************************************************/

-

-AAFilter::AAFilter(uint len)

-{

-    pFIR = FIRFilter::newInstance();

-    cutoffFreq = 0.5;

-    setLength(len);

-}

-

-

-

-AAFilter::~AAFilter()

-{

-    delete pFIR;

-}

-

-

-

-// Sets new anti-alias filter cut-off edge frequency, scaled to

-// sampling frequency (nyquist frequency = 0.5).

-// The filter will cut frequencies higher than the given frequency.

-void AAFilter::setCutoffFreq(double newCutoffFreq)

-{

-    cutoffFreq = newCutoffFreq;

-    calculateCoeffs();

-}

-

-

-

-// Sets number of FIR filter taps

-void AAFilter::setLength(uint newLength)

-{

-    length = newLength;

-    calculateCoeffs();

-}

-

-

-

-// Calculates coefficients for a low-pass FIR filter using Hamming window

-void AAFilter::calculateCoeffs()

-{

-    uint i;

-    double cntTemp, temp, tempCoeff,h, w;

-    double fc2, wc;

-    double scaleCoeff, sum;

-    double *work;

-    SAMPLETYPE *coeffs;

-

-    assert(length >= 2);

-    assert(length % 4 == 0);

-    assert(cutoffFreq >= 0);

-    assert(cutoffFreq <= 0.5);

-

-    work = new double[length];

-    coeffs = new SAMPLETYPE[length];

-

-    fc2 = 2.0 * cutoffFreq; 

-    wc = PI * fc2;

-    tempCoeff = TWOPI / (double)length;

-

-    sum = 0;

-    for (i = 0; i < length; i ++) 

-    {

-        cntTemp = (double)i - (double)(length / 2);

-

-        temp = cntTemp * wc;

-        if (temp != 0) 

-        {

-            h = fc2 * sin(temp) / temp;                     // sinc function

-        } 

-        else 

-        {

-            h = 1.0;

-        }

-        w = 0.54 + 0.46 * cos(tempCoeff * cntTemp);       // hamming window

-

-        temp = w * h;

-        work[i] = temp;

-

-        // calc net sum of coefficients 

-        sum += temp;

-    }

-

-    // ensure the sum of coefficients is larger than zero

-    assert(sum > 0);

-

-    // ensure we've really designed a lowpass filter...

-    assert(work[length/2] > 0);

-    assert(work[length/2 + 1] > -1e-6);

-    assert(work[length/2 - 1] > -1e-6);

-

-    // Calculate a scaling coefficient in such a way that the result can be

-    // divided by 16384

-    scaleCoeff = 16384.0f / sum;

-

-    for (i = 0; i < length; i ++) 

-    {

-        // scale & round to nearest integer

-        temp = work[i] * scaleCoeff;

-        temp += (temp >= 0) ? 0.5 : -0.5;

-        // ensure no overfloods

-        assert(temp >= -32768 && temp <= 32767);

-        coeffs[i] = (SAMPLETYPE)temp;

-    }

-

-    // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384

-    pFIR->setCoefficients(coeffs, length, 14);

-

-    delete[] work;

-    delete[] coeffs;

-}

-

-

-// Applies the filter to the given sequence of samples. 

-// Note : The amount of outputted samples is by value of 'filter length' 

-// smaller than the amount of input samples.

-uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const

-{

-    return pFIR->evaluate(dest, src, numSamples, numChannels);

-}

-

-

-uint AAFilter::getLength() const

-{

-    return pFIR->getLength();

-}

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/AAFilter.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/AAFilter.h
deleted file mode 100644
index 0ab97dde1..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/AAFilter.h
+++ /dev/null
@@ -1,91 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo 

-/// while maintaining the original pitch by using a time domain WSOLA-like method 

-/// with several performance-increasing tweaks.

-///

-/// Anti-alias filter is used to prevent folding of high frequencies when 

-/// transposing the sample rate with interpolation.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef AAFilter_H

-#define AAFilter_H

-

-#include "STTypes.h"

-

-namespace soundtouch

-{

-

-class AAFilter

-{

-protected:

-    class FIRFilter *pFIR;

-

-    /// Low-pass filter cut-off frequency, negative = invalid

-    double cutoffFreq;

-

-    /// num of filter taps

-    uint length;

-

-    /// Calculate the FIR coefficients realizing the given cutoff-frequency

-    void calculateCoeffs();

-public:

-    AAFilter(uint length);

-

-    ~AAFilter();

-

-    /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling 

-    /// frequency (nyquist frequency = 0.5). The filter will cut off the 

-    /// frequencies than that.

-    void setCutoffFreq(double newCutoffFreq);

-

-    /// Sets number of FIR filter taps, i.e. ~filter complexity

-    void setLength(uint newLength);

-

-    uint getLength() const;

-

-    /// Applies the filter to the given sequence of samples. 

-    /// Note : The amount of outputted samples is by value of 'filter length' 

-    /// smaller than the amount of input samples.

-    uint evaluate(SAMPLETYPE *dest, 

-                  const SAMPLETYPE *src, 

-                  uint numSamples, 

-                  uint numChannels) const;

-};

-

-}

-

-#endif

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/BPMDetect.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/BPMDetect.cpp
deleted file mode 100644
index 4faa29409..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/BPMDetect.cpp
+++ /dev/null
@@ -1,339 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// Beats-per-minute (BPM) detection routine.

-///

-/// The beat detection algorithm works as follows:

-/// - Use function 'inputSamples' to input a chunks of samples to the class for

-///   analysis. It's a good idea to enter a large sound file or stream in smallish

-///   chunks of around few kilosamples in order not to extinguish too much RAM memory.

-/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,

-///   which is basically ok as low (bass) frequencies mostly determine the beat rate.

-///   Simple averaging is used for anti-alias filtering because the resulting signal

-///   quality isn't of that high importance.

-/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by

-///   taking absolute value that's smoothed by sliding average. Signal levels that

-///   are below a couple of times the general RMS amplitude level are cut away to

-///   leave only notable peaks there.

-/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term 

-///   autocorrelation function of the enveloped signal.

-/// - After whole sound data file has been analyzed as above, the bpm level is 

-///   detected by function 'getBpm' that finds the highest peak of the autocorrelation 

-///   function, calculates it's precise location and converts this reading to bpm's.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <math.h>

-#include <assert.h>

-#include <string.h>

-#include "FIFOSampleBuffer.h"

-#include "PeakFinder.h"

-#include "BPMDetect.h"

-

-using namespace soundtouch;

-

-#define INPUT_BLOCK_SAMPLES       2048

-#define DECIMATED_BLOCK_SAMPLES   256

-

-/// decay constant for calculating RMS volume sliding average approximation 

-/// (time constant is about 10 sec)

-const float avgdecay = 0.99986f;

-

-/// Normalization coefficient for calculating RMS sliding average approximation.

-const float avgnorm = (1 - avgdecay);

-

-

-

-BPMDetect::BPMDetect(int numChannels, int aSampleRate)

-{

-    this->sampleRate = aSampleRate;

-    this->channels = numChannels;

-

-    decimateSum = 0;

-    decimateCount = 0;

-

-    envelopeAccu = 0;

-

-    // Initialize RMS volume accumulator to RMS level of 3000 (out of 32768) that's

-    // a typical RMS signal level value for song data. This value is then adapted

-    // to the actual level during processing.

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-    // integer samples

-    RMSVolumeAccu = (3000 * 3000) / avgnorm;

-#else

-    // float samples, scaled to range [-1..+1[

-    RMSVolumeAccu = (0.092f * 0.092f) / avgnorm;

-#endif

-

-    cutCoeff = 1.75;

-    aboveCutAccu = 0;

-    totalAccu = 0;

-

-    // choose decimation factor so that result is approx. 500 Hz

-    decimateBy = sampleRate / 500;

-    assert(decimateBy > 0);

-    assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);

-

-    // Calculate window length & starting item according to desired min & max bpms

-    windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);

-    windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);

-

-    assert(windowLen > windowStart);

-

-    // allocate new working objects

-    xcorr = new float[windowLen];

-    memset(xcorr, 0, windowLen * sizeof(float));

-

-    // allocate processing buffer

-    buffer = new FIFOSampleBuffer();

-    // we do processing in mono mode

-    buffer->setChannels(1);

-    buffer->clear();

-}

-

-

-

-BPMDetect::~BPMDetect()

-{

-    delete[] xcorr;

-    delete buffer;

-}

-

-

-

-/// convert to mono, low-pass filter & decimate to about 500 Hz. 

-/// return number of outputted samples.

-///

-/// Decimation is used to remove the unnecessary frequencies and thus to reduce 

-/// the amount of data needed to be processed as calculating autocorrelation 

-/// function is a very-very heavy operation.

-///

-/// Anti-alias filtering is done simply by averaging the samples. This is really a 

-/// poor-man's anti-alias filtering, but it's not so critical in this kind of application

-/// (it'd also be difficult to design a high-quality filter with steep cut-off at very 

-/// narrow band)

-int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)

-{

-    int count, outcount;

-    LONG_SAMPLETYPE out;

-

-    assert(channels > 0);

-    assert(decimateBy > 0);

-    outcount = 0;

-    for (count = 0; count < numsamples; count ++) 

-    {

-        int j;

-

-        // convert to mono and accumulate

-        for (j = 0; j < channels; j ++)

-        {

-            decimateSum += src[j];

-        }

-        src += j;

-

-        decimateCount ++;

-        if (decimateCount >= decimateBy) 

-        {

-            // Store every Nth sample only

-            out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));

-            decimateSum = 0;

-            decimateCount = 0;

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-            // check ranges for sure (shouldn't actually be necessary)

-            if (out > 32767) 

-            {

-                out = 32767;

-            } 

-            else if (out < -32768) 

-            {

-                out = -32768;

-            }

-#endif // SOUNDTOUCH_INTEGER_SAMPLES

-            dest[outcount] = (SAMPLETYPE)out;

-            outcount ++;

-        }

-    }

-    return outcount;

-}

-

-

-

-// Calculates autocorrelation function of the sample history buffer

-void BPMDetect::updateXCorr(int process_samples)

-{

-    int offs;

-    SAMPLETYPE *pBuffer;

-    

-    assert(buffer->numSamples() >= (uint)(process_samples + windowLen));

-

-    pBuffer = buffer->ptrBegin();

-    for (offs = windowStart; offs < windowLen; offs ++) 

-    {

-        LONG_SAMPLETYPE sum;

-        int i;

-

-        sum = 0;

-        for (i = 0; i < process_samples; i ++) 

-        {

-            sum += pBuffer[i] * pBuffer[i + offs];    // scaling the sub-result shouldn't be necessary

-        }

-//        xcorr[offs] *= xcorr_decay;   // decay 'xcorr' here with suitable coefficients 

-                                        // if it's desired that the system adapts automatically to

-                                        // various bpms, e.g. in processing continouos music stream.

-                                        // The 'xcorr_decay' should be a value that's smaller than but 

-                                        // close to one, and should also depend on 'process_samples' value.

-

-        xcorr[offs] += (float)sum;

-    }

-}

-

-

-// Calculates envelope of the sample data

-void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples) 

-{

-    const static double decay = 0.7f;               // decay constant for smoothing the envelope

-    const static double norm = (1 - decay);

-

-    int i;

-    LONG_SAMPLETYPE out;

-    double val;

-

-    for (i = 0; i < numsamples; i ++) 

-    {

-        // calc average RMS volume

-        RMSVolumeAccu *= avgdecay;

-        val = (float)fabs((float)samples[i]);

-        RMSVolumeAccu += val * val;

-

-        // cut amplitudes that are below cutoff ~2 times RMS volume

-        // (we're interested in peak values, not the silent moments)

-        val -= cutCoeff * sqrt(RMSVolumeAccu * avgnorm);

-        if (val > 0)

-        {

-            aboveCutAccu += 1.0;  // sample above threshold

-        }

-        else

-        {

-            val = 0;

-        }

-

-        totalAccu += 1.0;

-

-        // maintain sliding statistic what proportion of 'val' samples is

-        // above cutoff threshold

-        aboveCutAccu *= 0.99931;  // 2 sec time constant

-        totalAccu *= 0.99931;

-

-        if (totalAccu > 500)

-        {

-            // after initial settling, auto-adjust cutoff level so that ~8% of 

-            // values are above the threshold

-            double d = (aboveCutAccu / totalAccu) - 0.08;

-            cutCoeff += 0.001 * d;

-        }

-

-        // smooth amplitude envelope

-        envelopeAccu *= decay;

-        envelopeAccu += val;

-        out = (LONG_SAMPLETYPE)(envelopeAccu * norm);

-

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-        // cut peaks (shouldn't be necessary though)

-        if (out > 32767) out = 32767;

-#endif // SOUNDTOUCH_INTEGER_SAMPLES

-        samples[i] = (SAMPLETYPE)out;

-    }

-

-    // check that cutoff doesn't get too small - it can be just silent sequence!

-    if (cutCoeff < 1.5) 

-    {

-        cutCoeff = 1.5;

-    }

-}

-

-

-

-void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)

-{

-    SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];

-

-    // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration

-    while (numSamples > 0)

-    {

-        int block;

-        int decSamples;

-

-        block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;

-

-        // decimate. note that converts to mono at the same time

-        decSamples = decimate(decimated, samples, block);

-        samples += block * channels;

-        numSamples -= block;

-

-        // envelope new samples and add them to buffer

-        calcEnvelope(decimated, decSamples);

-        buffer->putSamples(decimated, decSamples);

-    }

-

-    // when the buffer has enought samples for processing...

-    if ((int)buffer->numSamples() > windowLen) 

-    {

-        int processLength;

-

-        // how many samples are processed

-        processLength = (int)buffer->numSamples() - windowLen;

-

-        // ... calculate autocorrelations for oldest samples...

-        updateXCorr(processLength);

-        // ... and remove them from the buffer

-        buffer->receiveSamples(processLength);

-    }

-}

-

-

-

-float BPMDetect::getBpm()

-{

-    double peakPos;

-    PeakFinder peakFinder;

-

-    // find peak position

-    peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);

-

-    assert(decimateBy != 0);

-    if (peakPos < 1e-9) return 0.0; // detection failed.

-

-    // calculate BPM

-    return (float)(60.0 * (((double)sampleRate / (double)decimateBy) / peakPos));

-}

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIFOSampleBuffer.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIFOSampleBuffer.cpp
deleted file mode 100644
index 8393f7b0d..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIFOSampleBuffer.cpp
+++ /dev/null
@@ -1,262 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// A buffer class for temporarily storaging sound samples, operates as a 

-/// first-in-first-out pipe.

-///

-/// Samples are added to the end of the sample buffer with the 'putSamples' 

-/// function, and are received from the beginning of the buffer by calling

-/// the 'receiveSamples' function. The class automatically removes the 

-/// outputted samples from the buffer, as well as grows the buffer size 

-/// whenever necessary.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <stdlib.h>

-#include <memory.h>

-#include <string.h>

-#include <assert.h>

-#include <stdexcept>

-

-#include "FIFOSampleBuffer.h"

-

-using namespace soundtouch;

-

-// Constructor

-FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)

-{

-    assert(numChannels > 0);

-    sizeInBytes = 0; // reasonable initial value

-    buffer = NULL;

-    bufferUnaligned = NULL;

-    samplesInBuffer = 0;

-    bufferPos = 0;

-    channels = (uint)numChannels;

-    ensureCapacity(32);     // allocate initial capacity 

-}

-

-

-// destructor

-FIFOSampleBuffer::~FIFOSampleBuffer()

-{

-    delete[] bufferUnaligned;

-    bufferUnaligned = NULL;

-    buffer = NULL;

-}

-

-

-// Sets number of channels, 1 = mono, 2 = stereo

-void FIFOSampleBuffer::setChannels(int numChannels)

-{

-    uint usedBytes;

-

-    assert(numChannels > 0);

-    usedBytes = channels * samplesInBuffer;

-    channels = (uint)numChannels;

-    samplesInBuffer = usedBytes / channels;

-}

-

-

-// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and

-// zeroes this pointer by copying samples from the 'bufferPos' pointer 

-// location on to the beginning of the buffer.

-void FIFOSampleBuffer::rewind()

-{

-    if (buffer && bufferPos) 

-    {

-        memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);

-        bufferPos = 0;

-    }

-}

-

-

-// Adds 'numSamples' pcs of samples from the 'samples' memory position to 

-// the sample buffer.

-void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)

-{

-    memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);

-    samplesInBuffer += nSamples;

-}

-

-

-// Increases the number of samples in the buffer without copying any actual

-// samples.

-//

-// This function is used to update the number of samples in the sample buffer

-// when accessing the buffer directly with 'ptrEnd' function. Please be 

-// careful though!

-void FIFOSampleBuffer::putSamples(uint nSamples)

-{

-    uint req;

-

-    req = samplesInBuffer + nSamples;

-    ensureCapacity(req);

-    samplesInBuffer += nSamples;

-}

-

-

-// Returns a pointer to the end of the used part of the sample buffer (i.e. 

-// where the new samples are to be inserted). This function may be used for 

-// inserting new samples into the sample buffer directly. Please be careful! 

-//

-// Parameter 'slackCapacity' tells the function how much free capacity (in

-// terms of samples) there _at least_ should be, in order to the caller to

-// succesfully insert all the required samples to the buffer. When necessary, 

-// the function grows the buffer size to comply with this requirement.

-//

-// When using this function as means for inserting new samples, also remember 

-// to increase the sample count afterwards, by calling  the 

-// 'putSamples(numSamples)' function.

-SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity) 

-{

-    ensureCapacity(samplesInBuffer + slackCapacity);

-    return buffer + samplesInBuffer * channels;

-}

-

-

-// Returns a pointer to the beginning of the currently non-outputted samples. 

-// This function is provided for accessing the output samples directly. 

-// Please be careful!

-//

-// When using this function to output samples, also remember to 'remove' the

-// outputted samples from the buffer by calling the 

-// 'receiveSamples(numSamples)' function

-SAMPLETYPE *FIFOSampleBuffer::ptrBegin()

-{

-    assert(buffer);

-    return buffer + bufferPos * channels;

-}

-

-

-// Ensures that the buffer has enought capacity, i.e. space for _at least_

-// 'capacityRequirement' number of samples. The buffer is grown in steps of

-// 4 kilobytes to eliminate the need for frequently growing up the buffer,

-// as well as to round the buffer size up to the virtual memory page size.

-void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)

-{

-    SAMPLETYPE *tempUnaligned, *temp;

-

-    if (capacityRequirement > getCapacity()) 

-    {

-        // enlarge the buffer in 4kbyte steps (round up to next 4k boundary)

-        sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;

-        assert(sizeInBytes % 2 == 0);

-        tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];

-        if (tempUnaligned == NULL)

-        {

-            throw std::runtime_error("Couldn't allocate memory!\n");

-        }

-        // Align the buffer to begin at 16byte cache line boundary for optimal performance

-        temp = (SAMPLETYPE *)(((ulong)tempUnaligned + 15) & (ulong)-16);

-        if (samplesInBuffer)

-        {

-            memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));

-        }

-        delete[] bufferUnaligned;

-        buffer = temp;

-        bufferUnaligned = tempUnaligned;

-        bufferPos = 0;

-    } 

-    else 

-    {

-        // simply rewind the buffer (if necessary)

-        rewind();

-    }

-}

-

-

-// Returns the current buffer capacity in terms of samples

-uint FIFOSampleBuffer::getCapacity() const

-{

-    return sizeInBytes / (channels * sizeof(SAMPLETYPE));

-}

-

-

-// Returns the number of samples currently in the buffer

-uint FIFOSampleBuffer::numSamples() const

-{

-    return samplesInBuffer;

-}

-

-

-// Output samples from beginning of the sample buffer. Copies demanded number

-// of samples to output and removes them from the sample buffer. If there

-// are less than 'numsample' samples in the buffer, returns all available.

-//

-// Returns number of samples copied.

-uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)

-{

-    uint num;

-

-    num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;

-

-    memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);

-    return receiveSamples(num);

-}

-

-

-// Removes samples from the beginning of the sample buffer without copying them

-// anywhere. Used to reduce the number of samples in the buffer, when accessing

-// the sample buffer with the 'ptrBegin' function.

-uint FIFOSampleBuffer::receiveSamples(uint maxSamples)

-{

-    if (maxSamples >= samplesInBuffer)

-    {

-        uint temp;

-

-        temp = samplesInBuffer;

-        samplesInBuffer = 0;

-        return temp;

-    }

-

-    samplesInBuffer -= maxSamples;

-    bufferPos += maxSamples;

-

-    return maxSamples;

-}

-

-

-// Returns nonzero if the sample buffer is empty

-int FIFOSampleBuffer::isEmpty() const

-{

-    return (samplesInBuffer == 0) ? 1 : 0;

-}

-

-

-// Clears the sample buffer

-void FIFOSampleBuffer::clear()

-{

-    samplesInBuffer = 0;

-    bufferPos = 0;

-}

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIRFilter.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIRFilter.cpp
deleted file mode 100644
index 50c4f62ee..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIRFilter.cpp
+++ /dev/null
@@ -1,263 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// General FIR digital filter routines with MMX optimization. 

-///

-/// Note : MMX optimized functions reside in a separate, platform-specific file, 

-/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <memory.h>

-#include <assert.h>

-#include <math.h>

-#include <stdlib.h>

-#include <stdexcept>

-#include "FIRFilter.h"

-#include "cpu_detect.h"

-

-using namespace soundtouch;

-

-/*****************************************************************************

- *

- * Implementation of the class 'FIRFilter'

- *

- *****************************************************************************/

-

-FIRFilter::FIRFilter()

-{

-    resultDivFactor = 0;

-    resultDivider = 0;

-    length = 0;

-    lengthDiv8 = 0;

-    filterCoeffs = NULL;

-}

-

-

-FIRFilter::~FIRFilter()

-{

-    delete[] filterCoeffs;

-}

-

-// Usual C-version of the filter routine for stereo sound

-uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const

-{

-    uint i, j, end;

-    LONG_SAMPLETYPE suml, sumr;

-#ifdef SOUNDTOUCH_FLOAT_SAMPLES

-    // when using floating point samples, use a scaler instead of a divider

-    // because division is much slower operation than multiplying.

-    double dScaler = 1.0 / (double)resultDivider;

-#endif

-

-    assert(length != 0);

-    assert(src != NULL);

-    assert(dest != NULL);

-    assert(filterCoeffs != NULL);

-

-    end = 2 * (numSamples - length);

-

-    for (j = 0; j < end; j += 2) 

-    {

-        const SAMPLETYPE *ptr;

-

-        suml = sumr = 0;

-        ptr = src + j;

-

-        for (i = 0; i < length; i += 4) 

-        {

-            // loop is unrolled by factor of 4 here for efficiency

-            suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +

-                    ptr[2 * i + 2] * filterCoeffs[i + 1] +

-                    ptr[2 * i + 4] * filterCoeffs[i + 2] +

-                    ptr[2 * i + 6] * filterCoeffs[i + 3];

-            sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +

-                    ptr[2 * i + 3] * filterCoeffs[i + 1] +

-                    ptr[2 * i + 5] * filterCoeffs[i + 2] +

-                    ptr[2 * i + 7] * filterCoeffs[i + 3];

-        }

-

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-        suml >>= resultDivFactor;

-        sumr >>= resultDivFactor;

-        // saturate to 16 bit integer limits

-        suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;

-        // saturate to 16 bit integer limits

-        sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;

-#else

-        suml *= dScaler;

-        sumr *= dScaler;

-#endif // SOUNDTOUCH_INTEGER_SAMPLES

-        dest[j] = (SAMPLETYPE)suml;

-        dest[j + 1] = (SAMPLETYPE)sumr;

-    }

-    return numSamples - length;

-}

-

-

-

-

-// Usual C-version of the filter routine for mono sound

-uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const

-{

-    uint i, j, end;

-    LONG_SAMPLETYPE sum;

-#ifdef SOUNDTOUCH_FLOAT_SAMPLES

-    // when using floating point samples, use a scaler instead of a divider

-    // because division is much slower operation than multiplying.

-    double dScaler = 1.0 / (double)resultDivider;

-#endif

-

-

-    assert(length != 0);

-

-    end = numSamples - length;

-    for (j = 0; j < end; j ++) 

-    {

-        sum = 0;

-        for (i = 0; i < length; i += 4) 

-        {

-            // loop is unrolled by factor of 4 here for efficiency

-            sum += src[i + 0] * filterCoeffs[i + 0] + 

-                   src[i + 1] * filterCoeffs[i + 1] + 

-                   src[i + 2] * filterCoeffs[i + 2] + 

-                   src[i + 3] * filterCoeffs[i + 3];

-        }

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-        sum >>= resultDivFactor;

-        // saturate to 16 bit integer limits

-        sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;

-#else

-        sum *= dScaler;

-#endif // SOUNDTOUCH_INTEGER_SAMPLES

-        dest[j] = (SAMPLETYPE)sum;

-        src ++;

-    }

-    return end;

-}

-

-

-// Set filter coeffiecients and length.

-//

-// Throws an exception if filter length isn't divisible by 8

-void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)

-{

-    assert(newLength > 0);

-    if (newLength % 8) throw std::runtime_error("FIR filter length not divisible by 8");

-

-    lengthDiv8 = newLength / 8;

-    length = lengthDiv8 * 8;

-    assert(length == newLength);

-

-    resultDivFactor = uResultDivFactor;

-    resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);

-

-    delete[] filterCoeffs;

-    filterCoeffs = new SAMPLETYPE[length];

-    memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));

-}

-

-

-uint FIRFilter::getLength() const

-{

-    return length;

-}

-

-

-

-// Applies the filter to the given sequence of samples. 

-//

-// Note : The amount of outputted samples is by value of 'filter_length' 

-// smaller than the amount of input samples.

-uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const

-{

-    assert(numChannels == 1 || numChannels == 2);

-

-    assert(length > 0);

-    assert(lengthDiv8 * 8 == length);

-    if (numSamples < length) return 0;

-    if (numChannels == 2) 

-    {

-        return evaluateFilterStereo(dest, src, numSamples);

-    } else {

-        return evaluateFilterMono(dest, src, numSamples);

-    }

-}

-

-

-

-// Operator 'new' is overloaded so that it automatically creates a suitable instance 

-// depending on if we've a MMX-capable CPU available or not.

-void * FIRFilter::operator new(size_t /*s*/)

-{

-    // Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!

-    throw std::runtime_error("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");

-    return NULL;

-}

-

-

-FIRFilter * FIRFilter::newInstance()

-{

-#ifndef _WIN64 //mpc custom code

-    uint uExtensions;

-

-    uExtensions = detectCPUextensions();

-

-    // Check if MMX/SSE instruction set extensions supported by CPU

-

-#ifdef SOUNDTOUCH_ALLOW_MMX

-    // MMX routines available only with integer sample types

-    if (uExtensions & SUPPORT_MMX)

-    {

-        return ::new FIRFilterMMX;

-    }

-    else

-#endif // SOUNDTOUCH_ALLOW_MMX

-

-#ifdef SOUNDTOUCH_ALLOW_SSE

-    if (uExtensions & SUPPORT_SSE)

-    {

-        // SSE support

-        return ::new FIRFilterSSE;

-    }

-    else

-#endif // SOUNDTOUCH_ALLOW_SSE

-

-#endif // _WIN64 mpc custom code

-

-    {

-        // ISA optimizations not supported, use plain C version

-        return ::new FIRFilter;

-    }

-}

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIRFilter.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIRFilter.h
deleted file mode 100644
index fcee72ac1..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/FIRFilter.h
+++ /dev/null
@@ -1,145 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// General FIR digital filter routines with MMX optimization. 

-///

-/// Note : MMX optimized functions reside in a separate, platform-specific file, 

-/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef FIRFilter_H

-#define FIRFilter_H

-

-#include <stddef.h>

-#include "STTypes.h"

-

-namespace soundtouch

-{

-

-class FIRFilter 

-{

-protected:

-    // Number of FIR filter taps

-    uint length;    

-    // Number of FIR filter taps divided by 8

-    uint lengthDiv8;

-

-    // Result divider factor in 2^k format

-    uint resultDivFactor;

-

-    // Result divider value.

-    SAMPLETYPE resultDivider;

-

-    // Memory for filter coefficients

-    SAMPLETYPE *filterCoeffs;

-

-    virtual uint evaluateFilterStereo(SAMPLETYPE *dest, 

-                                      const SAMPLETYPE *src, 

-                                      uint numSamples) const;

-    virtual uint evaluateFilterMono(SAMPLETYPE *dest, 

-                                    const SAMPLETYPE *src, 

-                                    uint numSamples) const;

-

-public:

-    FIRFilter();

-    virtual ~FIRFilter();

-

-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 

-    /// depending on if we've a MMX-capable CPU available or not.

-    static void * operator new(size_t s);

-

-    static FIRFilter *newInstance();

-

-    /// Applies the filter to the given sequence of samples. 

-    /// Note : The amount of outputted samples is by value of 'filter_length' 

-    /// smaller than the amount of input samples.

-    ///

-    /// \return Number of samples copied to 'dest'.

-    uint evaluate(SAMPLETYPE *dest, 

-                  const SAMPLETYPE *src, 

-                  uint numSamples, 

-                  uint numChannels) const;

-

-    uint getLength() const;

-

-    virtual void setCoefficients(const SAMPLETYPE *coeffs, 

-                                 uint newLength, 

-                                 uint uResultDivFactor);

-};

-

-

-// Optional subclasses that implement CPU-specific optimizations:

-

-#ifdef SOUNDTOUCH_ALLOW_MMX

-

-/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.

-    class FIRFilterMMX : public FIRFilter

-    {

-    protected:

-        short *filterCoeffsUnalign;

-        short *filterCoeffsAlign;

-

-        virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;

-    public:

-        FIRFilterMMX();

-        ~FIRFilterMMX();

-

-        virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);

-    };

-

-#endif // SOUNDTOUCH_ALLOW_MMX

-

-

-#ifdef SOUNDTOUCH_ALLOW_SSE

-    /// Class that implements SSE optimized functions exclusive for floating point samples type.

-    class FIRFilterSSE : public FIRFilter

-    {

-    protected:

-        float *filterCoeffsUnalign;

-        float *filterCoeffsAlign;

-

-        virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;

-    public:

-        FIRFilterSSE();

-        ~FIRFilterSSE();

-

-        virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);

-    };

-

-#endif // SOUNDTOUCH_ALLOW_SSE

-

-}

-

-#endif  // FIRFilter_H

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/PeakFinder.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/PeakFinder.cpp
deleted file mode 100644
index 9ad601cd9..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/PeakFinder.cpp
+++ /dev/null
@@ -1,239 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// Peak detection routine. 

-///

-/// The routine detects highest value on an array of values and calculates the 

-/// precise peak location as a mass-center of the 'hump' around the peak value.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <math.h>

-#include <assert.h>

-

-#include "PeakFinder.h"

-

-using namespace soundtouch;

-

-#define max(x, y) (((x) > (y)) ? (x) : (y))

-

-

-PeakFinder::PeakFinder()

-{

-    minPos = maxPos = 0;

-}

-

-

-// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding

-// to direction defined by 'direction' until next 'hump' after minimum value will 

-// begin

-int PeakFinder::findGround(const float *data, int peakpos, int direction) const

-{

-    float refvalue;

-    int lowpos;

-    int pos;

-    int climb_count;

-    float delta;

-

-    climb_count = 0;

-    refvalue = data[peakpos];

-    lowpos = peakpos;

-

-    pos = peakpos;

-

-    while ((pos > minPos) && (pos < maxPos))

-    {

-        int prevpos;

-

-        prevpos = pos;

-        pos += direction;

-

-        // calculate derivate

-        delta = data[pos] - data[prevpos];

-        if (delta <= 0)

-        {

-            // going downhill, ok

-            if (climb_count)

-            {

-                climb_count --;  // decrease climb count

-            }

-

-            // check if new minimum found

-            if (data[pos] < refvalue)

-            {

-                // new minimum found

-                lowpos = pos;

-                refvalue = data[pos];

-            }

-        }

-        else

-        {

-            // going uphill, increase climbing counter

-            climb_count ++;

-            if (climb_count > 5) break;    // we've been climbing too long => it's next uphill => quit

-        }

-    }

-    return lowpos;

-}

-

-

-// Find offset where the value crosses the given level, when starting from 'peakpos' and

-// proceeds to direction defined in 'direction'

-int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const

-{

-    float peaklevel;

-    int pos;

-

-    peaklevel = data[peakpos];

-    assert(peaklevel >= level);

-    pos = peakpos;

-    while ((pos >= minPos) && (pos < maxPos))

-    {

-        if (data[pos + direction] < level) return pos;   // crossing found

-        pos += direction;

-    }

-    return -1;  // not found

-}

-

-

-// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'

-double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const

-{

-    int i;

-    float sum;

-    float wsum;

-

-    sum = 0;

-    wsum = 0;

-    for (i = firstPos; i <= lastPos; i ++)

-    {

-        sum += (float)i * data[i];

-        wsum += data[i];

-    }

-

-    if (wsum < 1e-6) return 0;

-    return sum / wsum;

-}

-

-

-

-/// get exact center of peak near given position by calculating local mass of center

-double PeakFinder::getPeakCenter(const float *data, int peakpos) const

-{

-    float peakLevel;            // peak level

-    int crosspos1, crosspos2;   // position where the peak 'hump' crosses cutting level

-    float cutLevel;             // cutting value

-    float groundLevel;          // ground level of the peak

-    int gp1, gp2;               // bottom positions of the peak 'hump'

-

-    // find ground positions.

-    gp1 = findGround(data, peakpos, -1);

-    gp2 = findGround(data, peakpos, 1);

-

-    groundLevel = max(data[gp1], data[gp2]);

-    peakLevel = data[peakpos];

-

-    if (groundLevel < 1e-9) return 0;                // ground level too small => detection failed

-    if ((peakLevel / groundLevel) < 1.3) return 0;   // peak less than 30% of the ground level => no good peak detected

-

-    // calculate 70%-level of the peak

-    cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;

-    // find mid-level crossings

-    crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);

-    crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);

-

-    if ((crosspos1 < 0) || (crosspos2 < 0)) return 0;   // no crossing, no peak..

-

-    // calculate mass center of the peak surroundings

-    return calcMassCenter(data, crosspos1, crosspos2);

-}

-

-

-

-double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos) 

-{

-

-    int i;

-    int peakpos;                // position of peak level

-    double highPeak, peak;

-

-    this->minPos = aminPos;

-    this->maxPos = amaxPos;

-

-    // find absolute peak

-    peakpos = minPos;

-    peak = data[minPos];

-    for (i = minPos + 1; i < maxPos; i ++)

-    {

-        if (data[i] > peak) 

-        {

-            peak = data[i];

-            peakpos = i;

-        }

-    }

-    

-    // Calculate exact location of the highest peak mass center

-    highPeak = getPeakCenter(data, peakpos);

-    peak = highPeak;

-

-    // Now check if the highest peak were in fact harmonic of the true base beat peak 

-    // - sometimes the highest peak can be Nth harmonic of the true base peak yet 

-    // just a slightly higher than the true base

-    for (i = 2; i < 10; i ++)

-    {

-        double peaktmp, tmp;

-        int i1,i2;

-

-        peakpos = (int)(highPeak / (double)i + 0.5f);

-        if (peakpos < minPos) break;

-

-        // calculate mass-center of possible base peak

-        peaktmp = getPeakCenter(data, peakpos);

-

-        // now compare to highest detected peak

-        i1 = (int)(highPeak + 0.5);

-        i2 = (int)(peaktmp + 0.5);

-        tmp = 2 * (data[i2] - data[i1]) / (data[i2] + data[i1]);

-        if (fabs(tmp) < 0.1)

-        {

-            // The highest peak is harmonic of almost as high base peak,

-            // thus use the base peak instead

-            peak = peaktmp;

-        }

-    }

-

-    return peak;

-}

-

-

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/PeakFinder.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/PeakFinder.h
deleted file mode 100644
index a72b24f28..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/PeakFinder.h
+++ /dev/null
@@ -1,93 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// The routine detects highest value on an array of values and calculates the 

-/// precise peak location as a mass-center of the 'hump' around the peak value.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef _PeakFinder_H_

-#define _PeakFinder_H_

-

-namespace soundtouch

-{

-

-class PeakFinder

-{

-protected:

-    /// Min, max allowed peak positions within the data vector

-    int minPos, maxPos;

-

-    /// Calculates the mass center between given vector items.

-    double calcMassCenter(const float *data, ///< Data vector.

-                         int firstPos,      ///< Index of first vector item beloging to the peak.

-                         int lastPos        ///< Index of last vector item beloging to the peak.

-                         ) const;

-

-    /// Finds the data vector index where the monotoniously decreasing signal crosses the

-    /// given level.

-    int   findCrossingLevel(const float *data,  ///< Data vector.

-                            float level,        ///< Goal crossing level.

-                            int peakpos,        ///< Peak position index within the data vector.

-                            int direction       /// Direction where to proceed from the peak: 1 = right, -1 = left.

-                            ) const;

-

-    /// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right- 

-    /// or left-hand side of the given peak position.

-    int   findGround(const float *data,     /// Data vector.

-                     int peakpos,           /// Peak position index within the data vector.

-                     int direction          /// Direction where to proceed from the peak: 1 = right, -1 = left.

-                     ) const;

-

-    /// get exact center of peak near given position by calculating local mass of center

-    double getPeakCenter(const float *data, int peakpos) const;

-

-public:

-    /// Constructor. 

-    PeakFinder();

-

-    /// Detect exact peak position of the data vector by finding the largest peak 'hump'

-    /// and calculating the mass-center location of the peak hump.

-    ///

-    /// \return The location of the largest base harmonic peak hump.

-    double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has

-                                        /// to be at least 'maxPos' items long.

-                     int minPos,        ///< Min allowed peak location within the vector data.

-                     int maxPos         ///< Max allowed peak location within the vector data.

-                     );

-};

-

-}

-

-#endif // _PeakFinder_H_

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/RateTransposer.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/RateTransposer.cpp
deleted file mode 100644
index c288fddb9..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/RateTransposer.cpp
+++ /dev/null
@@ -1,628 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-/// 

-/// Sample rate transposer. Changes sample rate by using linear interpolation 

-/// together with anti-alias filtering (first order interpolation with anti-

-/// alias filtering should be quite adequate for this application)

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <memory.h>

-#include <assert.h>

-#include <stdlib.h>

-#include <stdio.h>

-#include <stdexcept>

-#include "RateTransposer.h"

-#include "AAFilter.h"

-

-using namespace std;

-using namespace soundtouch;

-

-

-/// A linear samplerate transposer class that uses integer arithmetics.

-/// for the transposing.

-class RateTransposerInteger : public RateTransposer

-{

-protected:

-    int iSlopeCount;

-    int iRate;

-    SAMPLETYPE sPrevSampleL, sPrevSampleR;

-

-    virtual void resetRegisters();

-

-    virtual uint transposeStereo(SAMPLETYPE *dest, 

-                         const SAMPLETYPE *src, 

-                         uint numSamples);

-    virtual uint transposeMono(SAMPLETYPE *dest, 

-                       const SAMPLETYPE *src, 

-                       uint numSamples);

-

-public:

-    RateTransposerInteger();

-    virtual ~RateTransposerInteger();

-

-    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 

-    /// rate, larger faster rates.

-    virtual void setRate(float newRate);

-

-};

-

-

-/// A linear samplerate transposer class that uses floating point arithmetics

-/// for the transposing.

-class RateTransposerFloat : public RateTransposer

-{

-protected:

-    float fSlopeCount;

-    SAMPLETYPE sPrevSampleL, sPrevSampleR;

-

-    virtual void resetRegisters();

-

-    virtual uint transposeStereo(SAMPLETYPE *dest, 

-                         const SAMPLETYPE *src, 

-                         uint numSamples);

-    virtual uint transposeMono(SAMPLETYPE *dest, 

-                       const SAMPLETYPE *src, 

-                       uint numSamples);

-

-public:

-    RateTransposerFloat();

-    virtual ~RateTransposerFloat();

-};

-

-

-

-

-// Operator 'new' is overloaded so that it automatically creates a suitable instance 

-// depending on if we've a MMX/SSE/etc-capable CPU available or not.

-void * RateTransposer::operator new(size_t /*s*/)

-{

-    throw runtime_error("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");

-    return NULL;

-}

-

-

-RateTransposer *RateTransposer::newInstance()

-{

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-    return ::new RateTransposerInteger;

-#else

-    return ::new RateTransposerFloat;

-#endif

-}

-

-

-// Constructor

-RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)

-{

-    numChannels = 2;

-    bUseAAFilter = TRUE;

-    fRate = 0;

-

-    // Instantiates the anti-alias filter with default tap length

-    // of 32

-    pAAFilter = new AAFilter(32);

-}

-

-

-

-RateTransposer::~RateTransposer()

-{

-    delete pAAFilter;

-}

-

-

-

-/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable

-void RateTransposer::enableAAFilter(BOOL newMode)

-{

-    bUseAAFilter = newMode;

-}

-

-

-/// Returns nonzero if anti-alias filter is enabled.

-BOOL RateTransposer::isAAFilterEnabled() const

-{

-    return bUseAAFilter;

-}

-

-

-AAFilter *RateTransposer::getAAFilter()

-{

-    return pAAFilter;

-}

-

-

-

-// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 

-// iRate, larger faster iRates.

-void RateTransposer::setRate(float newRate)

-{

-    double fCutoff;

-

-    fRate = newRate;

-

-    // design a new anti-alias filter

-    if (newRate > 1.0f) 

-    {

-        fCutoff = 0.5f / newRate;

-    } 

-    else 

-    {

-        fCutoff = 0.5f * newRate;

-    }

-    pAAFilter->setCutoffFreq(fCutoff);

-}

-

-

-// Outputs as many samples of the 'outputBuffer' as possible, and if there's

-// any room left, outputs also as many of the incoming samples as possible.

-// The goal is to drive the outputBuffer empty.

-//

-// It's allowed for 'output' and 'input' parameters to point to the same

-// memory position.

-/*

-void RateTransposer::flushStoreBuffer()

-{

-    if (storeBuffer.isEmpty()) return;

-

-    outputBuffer.moveSamples(storeBuffer);

-}

-*/

-

-

-// Adds 'nSamples' pcs of samples from the 'samples' memory position into

-// the input of the object.

-void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)

-{

-    processSamples(samples, nSamples);

-}

-

-

-

-// Transposes up the sample rate, causing the observed playback 'rate' of the

-// sound to decrease

-void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)

-{

-    uint count, sizeTemp, num;

-

-    // If the parameter 'uRate' value is smaller than 'SCALE', first transpose

-    // the samples and then apply the anti-alias filter to remove aliasing.

-

-    // First check that there's enough room in 'storeBuffer' 

-    // (+16 is to reserve some slack in the destination buffer)

-    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);

-

-    // Transpose the samples, store the result into the end of "storeBuffer"

-    count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);

-    storeBuffer.putSamples(count);

-

-    // Apply the anti-alias filter to samples in "store output", output the

-    // result to "dest"

-    num = storeBuffer.numSamples();

-    count = pAAFilter->evaluate(outputBuffer.ptrEnd(num), 

-        storeBuffer.ptrBegin(), num, (uint)numChannels);

-    outputBuffer.putSamples(count);

-

-    // Remove the processed samples from "storeBuffer"

-    storeBuffer.receiveSamples(count);

-}

-

-

-// Transposes down the sample rate, causing the observed playback 'rate' of the

-// sound to increase

-void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)

-{

-    uint count, sizeTemp;

-

-    // If the parameter 'uRate' value is larger than 'SCALE', first apply the

-    // anti-alias filter to remove high frequencies (prevent them from folding

-    // over the lover frequencies), then transpose.

-

-    // Add the new samples to the end of the storeBuffer

-    storeBuffer.putSamples(src, nSamples);

-

-    // Anti-alias filter the samples to prevent folding and output the filtered 

-    // data to tempBuffer. Note : because of the FIR filter length, the

-    // filtering routine takes in 'filter_length' more samples than it outputs.

-    assert(tempBuffer.isEmpty());

-    sizeTemp = storeBuffer.numSamples();

-

-    count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp), 

-        storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);

-

-	if (count == 0) return;

-

-    // Remove the filtered samples from 'storeBuffer'

-    storeBuffer.receiveSamples(count);

-

-    // Transpose the samples (+16 is to reserve some slack in the destination buffer)

-    sizeTemp = (uint)((float)nSamples / fRate + 16.0f);

-    count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);

-    outputBuffer.putSamples(count);

-}

-

-

-// Transposes sample rate by applying anti-alias filter to prevent folding. 

-// Returns amount of samples returned in the "dest" buffer.

-// The maximum amount of samples that can be returned at a time is set by

-// the 'set_returnBuffer_size' function.

-void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)

-{

-    uint count;

-    uint sizeReq;

-

-    if (nSamples == 0) return;

-    assert(pAAFilter);

-

-    // If anti-alias filter is turned off, simply transpose without applying

-    // the filter

-    if (bUseAAFilter == FALSE) 

-    {

-        sizeReq = (uint)((float)nSamples / fRate + 1.0f);

-        count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);

-        outputBuffer.putSamples(count);

-        return;

-    }

-

-    // Transpose with anti-alias filter

-    if (fRate < 1.0f) 

-    {

-        upsample(src, nSamples);

-    } 

-    else  

-    {

-        downsample(src, nSamples);

-    }

-}

-

-

-// Transposes the sample rate of the given samples using linear interpolation. 

-// Returns the number of samples returned in the "dest" buffer

-inline uint RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)

-{

-    if (numChannels == 2) 

-    {

-        return transposeStereo(dest, src, nSamples);

-    } 

-    else 

-    {

-        return transposeMono(dest, src, nSamples);

-    }

-}

-

-

-// Sets the number of channels, 1 = mono, 2 = stereo

-void RateTransposer::setChannels(int nChannels)

-{

-    assert(nChannels > 0);

-    if (numChannels == nChannels) return;

-

-    assert(nChannels == 1 || nChannels == 2);

-    numChannels = nChannels;

-

-    storeBuffer.setChannels(numChannels);

-    tempBuffer.setChannels(numChannels);

-    outputBuffer.setChannels(numChannels);

-

-    // Inits the linear interpolation registers

-    resetRegisters();

-}

-

-

-// Clears all the samples in the object

-void RateTransposer::clear()

-{

-    outputBuffer.clear();

-    storeBuffer.clear();

-}

-

-

-// Returns nonzero if there aren't any samples available for outputting.

-int RateTransposer::isEmpty() const

-{

-    int res;

-

-    res = FIFOProcessor::isEmpty();

-    if (res == 0) return 0;

-    return storeBuffer.isEmpty();

-}

-

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// RateTransposerInteger - integer arithmetic implementation

-// 

-

-/// fixed-point interpolation routine precision

-#define SCALE    65536

-

-// Constructor

-RateTransposerInteger::RateTransposerInteger() : RateTransposer()

-{

-    // Notice: use local function calling syntax for sake of clarity, 

-    // to indicate the fact that C++ constructor can't call virtual functions.

-    RateTransposerInteger::resetRegisters();

-    RateTransposerInteger::setRate(1.0f);

-}

-

-

-RateTransposerInteger::~RateTransposerInteger()

-{

-}

-

-

-void RateTransposerInteger::resetRegisters()

-{

-    iSlopeCount = 0;

-    sPrevSampleL = 

-    sPrevSampleR = 0;

-}

-

-

-

-// Transposes the sample rate of the given samples using linear interpolation. 

-// 'Mono' version of the routine. Returns the number of samples returned in 

-// the "dest" buffer

-uint RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)

-{

-    unsigned int i, used;

-    LONG_SAMPLETYPE temp, vol1;

-

-    if (nSamples == 0) return 0;  // no samples, no work

-

-	used = 0;    

-    i = 0;

-

-    // Process the last sample saved from the previous call first...

-    while (iSlopeCount <= SCALE) 

-    {

-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);

-        temp = vol1 * sPrevSampleL + iSlopeCount * src[0];

-        dest[i] = (SAMPLETYPE)(temp / SCALE);

-        i++;

-        iSlopeCount += iRate;

-    }

-    // now always (iSlopeCount > SCALE)

-    iSlopeCount -= SCALE;

-

-    while (1)

-    {

-        while (iSlopeCount > SCALE) 

-        {

-            iSlopeCount -= SCALE;

-            used ++;

-            if (used >= nSamples - 1) goto end;

-        }

-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);

-        temp = src[used] * vol1 + iSlopeCount * src[used + 1];

-        dest[i] = (SAMPLETYPE)(temp / SCALE);

-

-        i++;

-        iSlopeCount += iRate;

-    }

-end:

-    // Store the last sample for the next round

-    sPrevSampleL = src[nSamples - 1];

-

-    return i;

-}

-

-

-// Transposes the sample rate of the given samples using linear interpolation. 

-// 'Stereo' version of the routine. Returns the number of samples returned in 

-// the "dest" buffer

-uint RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)

-{

-    unsigned int srcPos, i, used;

-    LONG_SAMPLETYPE temp, vol1;

-

-    if (nSamples == 0) return 0;  // no samples, no work

-

-    used = 0;    

-    i = 0;

-

-    // Process the last sample saved from the sPrevSampleLious call first...

-    while (iSlopeCount <= SCALE) 

-    {

-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);

-        temp = vol1 * sPrevSampleL + iSlopeCount * src[0];

-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);

-        temp = vol1 * sPrevSampleR + iSlopeCount * src[1];

-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);

-        i++;

-        iSlopeCount += iRate;

-    }

-    // now always (iSlopeCount > SCALE)

-    iSlopeCount -= SCALE;

-

-    while (1)

-    {

-        while (iSlopeCount > SCALE) 

-        {

-            iSlopeCount -= SCALE;

-            used ++;

-            if (used >= nSamples - 1) goto end;

-        }

-        srcPos = 2 * used;

-        vol1 = (LONG_SAMPLETYPE)(SCALE - iSlopeCount);

-        temp = src[srcPos] * vol1 + iSlopeCount * src[srcPos + 2];

-        dest[2 * i] = (SAMPLETYPE)(temp / SCALE);

-        temp = src[srcPos + 1] * vol1 + iSlopeCount * src[srcPos + 3];

-        dest[2 * i + 1] = (SAMPLETYPE)(temp / SCALE);

-

-        i++;

-        iSlopeCount += iRate;

-    }

-end:

-    // Store the last sample for the next round

-    sPrevSampleL = src[2 * nSamples - 2];

-    sPrevSampleR = src[2 * nSamples - 1];

-

-    return i;

-}

-

-

-// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 

-// iRate, larger faster iRates.

-void RateTransposerInteger::setRate(float newRate)

-{

-    iRate = (int)(newRate * SCALE + 0.5f);

-    RateTransposer::setRate(newRate);

-}

-

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// RateTransposerFloat - floating point arithmetic implementation

-// 

-//////////////////////////////////////////////////////////////////////////////

-

-// Constructor

-RateTransposerFloat::RateTransposerFloat() : RateTransposer()

-{

-    // Notice: use local function calling syntax for sake of clarity, 

-    // to indicate the fact that C++ constructor can't call virtual functions.

-    RateTransposerFloat::resetRegisters();

-    RateTransposerFloat::setRate(1.0f);

-}

-

-

-RateTransposerFloat::~RateTransposerFloat()

-{

-}

-

-

-void RateTransposerFloat::resetRegisters()

-{

-    fSlopeCount = 0;

-    sPrevSampleL = 

-    sPrevSampleR = 0;

-}

-

-

-

-// Transposes the sample rate of the given samples using linear interpolation. 

-// 'Mono' version of the routine. Returns the number of samples returned in 

-// the "dest" buffer

-uint RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)

-{

-    unsigned int i, used;

-

-    used = 0;    

-    i = 0;

-

-    // Process the last sample saved from the previous call first...

-    while (fSlopeCount <= 1.0f) 

-    {

-        dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);

-        i++;

-        fSlopeCount += fRate;

-    }

-    fSlopeCount -= 1.0f;

-

-    if (nSamples > 1)

-    {

-        while (1)

-        {

-            while (fSlopeCount > 1.0f) 

-            {

-                fSlopeCount -= 1.0f;

-                used ++;

-                if (used >= nSamples - 1) goto end;

-            }

-            dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[used] + fSlopeCount * src[used + 1]);

-            i++;

-            fSlopeCount += fRate;

-        }

-    }

-end:

-    // Store the last sample for the next round

-    sPrevSampleL = src[nSamples - 1];

-

-    return i;

-}

-

-

-// Transposes the sample rate of the given samples using linear interpolation. 

-// 'Mono' version of the routine. Returns the number of samples returned in 

-// the "dest" buffer

-uint RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)

-{

-    unsigned int srcPos, i, used;

-

-    if (nSamples == 0) return 0;  // no samples, no work

-

-    used = 0;    

-    i = 0;

-

-    // Process the last sample saved from the sPrevSampleLious call first...

-    while (fSlopeCount <= 1.0f) 

-    {

-        dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleL + fSlopeCount * src[0]);

-        dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSampleR + fSlopeCount * src[1]);

-        i++;

-        fSlopeCount += fRate;

-    }

-    // now always (iSlopeCount > 1.0f)

-    fSlopeCount -= 1.0f;

-

-    if (nSamples > 1)

-    {

-        while (1)

-        {

-            while (fSlopeCount > 1.0f) 

-            {

-                fSlopeCount -= 1.0f;

-                used ++;

-                if (used >= nSamples - 1) goto end;

-            }

-            srcPos = 2 * used;

-

-            dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos] 

-                + fSlopeCount * src[srcPos + 2]);

-            dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[srcPos + 1] 

-                + fSlopeCount * src[srcPos + 3]);

-

-            i++;

-            fSlopeCount += fRate;

-        }

-    }

-end:

-    // Store the last sample for the next round

-    sPrevSampleL = src[2 * nSamples - 2];

-    sPrevSampleR = src[2 * nSamples - 1];

-

-    return i;

-}

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/RateTransposer.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/RateTransposer.h
deleted file mode 100644
index 9cd1c6f06..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/RateTransposer.h
+++ /dev/null
@@ -1,159 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-/// 

-/// Sample rate transposer. Changes sample rate by using linear interpolation 

-/// together with anti-alias filtering (first order interpolation with anti-

-/// alias filtering should be quite adequate for this application).

-///

-/// Use either of the derived classes of 'RateTransposerInteger' or 

-/// 'RateTransposerFloat' for corresponding integer/floating point tranposing

-/// algorithm implementation.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef RateTransposer_H

-#define RateTransposer_H

-

-#include <stddef.h>

-#include "AAFilter.h"

-#include "FIFOSamplePipe.h"

-#include "FIFOSampleBuffer.h"

-

-#include "STTypes.h"

-

-namespace soundtouch

-{

-

-/// A common linear samplerate transposer class.

-///

-/// Note: Use function "RateTransposer::newInstance()" to create a new class 

-/// instance instead of the "new" operator; that function automatically 

-/// chooses a correct implementation depending on if integer or floating 

-/// arithmetics are to be used.

-class RateTransposer : public FIFOProcessor

-{

-protected:

-    /// Anti-alias filter object

-    AAFilter *pAAFilter;

-

-    float fRate;

-

-    int numChannels;

-

-    /// Buffer for collecting samples to feed the anti-alias filter between

-    /// two batches

-    FIFOSampleBuffer storeBuffer;

-

-    /// Buffer for keeping samples between transposing & anti-alias filter

-    FIFOSampleBuffer tempBuffer;

-

-    /// Output sample buffer

-    FIFOSampleBuffer outputBuffer;

-

-    BOOL bUseAAFilter;

-

-    virtual void resetRegisters() = 0;

-

-    virtual uint transposeStereo(SAMPLETYPE *dest, 

-                         const SAMPLETYPE *src, 

-                         uint numSamples) = 0;

-    virtual uint transposeMono(SAMPLETYPE *dest, 

-                       const SAMPLETYPE *src, 

-                       uint numSamples) = 0;

-    inline uint transpose(SAMPLETYPE *dest, 

-                   const SAMPLETYPE *src, 

-                   uint numSamples);

-

-    void downsample(const SAMPLETYPE *src, 

-                    uint numSamples);

-    void upsample(const SAMPLETYPE *src, 

-                 uint numSamples);

-

-    /// Transposes sample rate by applying anti-alias filter to prevent folding. 

-    /// Returns amount of samples returned in the "dest" buffer.

-    /// The maximum amount of samples that can be returned at a time is set by

-    /// the 'set_returnBuffer_size' function.

-    void processSamples(const SAMPLETYPE *src, 

-                        uint numSamples);

-

-

-public:

-    RateTransposer();

-    virtual ~RateTransposer();

-

-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 

-    /// depending on if we're to use integer or floating point arithmetics.

-    static void *operator new(size_t s);

-

-    /// Use this function instead of "new" operator to create a new instance of this class. 

-    /// This function automatically chooses a correct implementation, depending on if 

-    /// integer ot floating point arithmetics are to be used.

-    static RateTransposer *newInstance();

-

-    /// Returns the output buffer object

-    FIFOSamplePipe *getOutput() { return &outputBuffer; };

-

-    /// Returns the store buffer object

-    FIFOSamplePipe *getStore() { return &storeBuffer; };

-

-    /// Return anti-alias filter object

-    AAFilter *getAAFilter();

-

-    /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable

-    void enableAAFilter(BOOL newMode);

-

-    /// Returns nonzero if anti-alias filter is enabled.

-    BOOL isAAFilterEnabled() const;

-

-    /// Sets new target rate. Normal rate = 1.0, smaller values represent slower 

-    /// rate, larger faster rates.

-    virtual void setRate(float newRate);

-

-    /// Sets the number of channels, 1 = mono, 2 = stereo

-    void setChannels(int channels);

-

-    /// Adds 'numSamples' pcs of samples from the 'samples' memory position into

-    /// the input of the object.

-    void putSamples(const SAMPLETYPE *samples, uint numSamples);

-

-    /// Clears all the samples in the object

-    void clear();

-

-    /// Returns nonzero if there aren't any samples available for outputting.

-    int isEmpty() const;

-};

-

-}

-

-#endif

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/SoundTouch.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/SoundTouch.cpp
deleted file mode 100644
index 6f7b9a894..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/SoundTouch.cpp
+++ /dev/null
@@ -1,486 +0,0 @@
-//////////////////////////////////////////////////////////////////////////////

-///

-/// SoundTouch - main class for tempo/pitch/rate adjusting routines. 

-///

-/// Notes:

-/// - Initialize the SoundTouch object instance by setting up the sound stream 

-///   parameters with functions 'setSampleRate' and 'setChannels', then set 

-///   desired tempo/pitch/rate settings with the corresponding functions.

-///

-/// - The SoundTouch class behaves like a first-in-first-out pipeline: The 

-///   samples that are to be processed are fed into one of the pipe by calling

-///   function 'putSamples', while the ready processed samples can be read 

-///   from the other end of the pipeline with function 'receiveSamples'.

-/// 

-/// - The SoundTouch processing classes require certain sized 'batches' of 

-///   samples in order to process the sound. For this reason the classes buffer 

-///   incoming samples until there are enough of samples available for 

-///   processing, then they carry out the processing step and consequently

-///   make the processed samples available for outputting.

-/// 

-/// - For the above reason, the processing routines introduce a certain 

-///   'latency' between the input and output, so that the samples input to

-///   SoundTouch may not be immediately available in the output, and neither 

-///   the amount of outputtable samples may not immediately be in direct 

-///   relationship with the amount of previously input samples.

-///

-/// - The tempo/pitch/rate control parameters can be altered during processing.

-///   Please notice though that they aren't currently protected by semaphores,

-///   so in multi-thread application external semaphore protection may be

-///   required.

-///

-/// - This class utilizes classes 'TDStretch' for tempo change (without modifying

-///   pitch) and 'RateTransposer' for changing the playback rate (that is, both 

-///   tempo and pitch in the same ratio) of the sound. The third available control 

-///   'pitch' (change pitch but maintain tempo) is produced by a combination of

-///   combining the two other controls.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <assert.h>

-#include <stdlib.h>

-#include <memory.h>

-#include <math.h>

-#include <stdexcept>

-#include <stdio.h>

-

-#include "SoundTouch.h"

-#include "TDStretch.h"

-#include "RateTransposer.h"

-#include "cpu_detect.h"

-

-using namespace soundtouch;

-    

-/// test if two floating point numbers are equal

-#define TEST_FLOAT_EQUAL(a, b)  (fabs(a - b) < 1e-10)

-

-

-/// Print library version string for autoconf

-extern "C" void soundtouch_ac_test()

-{

-    printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);

-} 

-

-

-SoundTouch::SoundTouch()

-{

-    // Initialize rate transposer and tempo changer instances

-

-    pRateTransposer = RateTransposer::newInstance();

-    pTDStretch = TDStretch::newInstance();

-

-    setOutPipe(pTDStretch);

-

-    rate = tempo = 0;

-

-    virtualPitch = 

-    virtualRate = 

-    virtualTempo = 1.0;

-

-    calcEffectiveRateAndTempo();

-

-    channels = 0;

-    bSrateSet = FALSE;

-}

-

-

-

-SoundTouch::~SoundTouch()

-{

-    delete pRateTransposer;

-    delete pTDStretch;

-}

-

-

-

-/// Get SoundTouch library version string

-const char *SoundTouch::getVersionString()

-{

-    static const char *_version = SOUNDTOUCH_VERSION;

-

-    return _version;

-}

-

-

-/// Get SoundTouch library version Id

-uint SoundTouch::getVersionId()

-{

-    return SOUNDTOUCH_VERSION_ID;

-}

-

-

-// Sets the number of channels, 1 = mono, 2 = stereo

-void SoundTouch::setChannels(uint numChannels)

-{

-    if (numChannels != 1 && numChannels != 2) 

-    {

-        throw std::runtime_error("Illegal number of channels");

-    }

-    channels = numChannels;

-    pRateTransposer->setChannels((int)numChannels);

-    pTDStretch->setChannels((int)numChannels);

-}

-

-

-

-// Sets new rate control value. Normal rate = 1.0, smaller values

-// represent slower rate, larger faster rates.

-void SoundTouch::setRate(float newRate)

-{

-    virtualRate = newRate;

-    calcEffectiveRateAndTempo();

-}

-

-

-

-// Sets new rate control value as a difference in percents compared

-// to the original rate (-50 .. +100 %)

-void SoundTouch::setRateChange(float newRate)

-{

-    virtualRate = 1.0f + 0.01f * newRate;

-    calcEffectiveRateAndTempo();

-}

-

-

-

-// Sets new tempo control value. Normal tempo = 1.0, smaller values

-// represent slower tempo, larger faster tempo.

-void SoundTouch::setTempo(float newTempo)

-{

-    virtualTempo = newTempo;

-    calcEffectiveRateAndTempo();

-}

-

-

-

-// Sets new tempo control value as a difference in percents compared

-// to the original tempo (-50 .. +100 %)

-void SoundTouch::setTempoChange(float newTempo)

-{

-    virtualTempo = 1.0f + 0.01f * newTempo;

-    calcEffectiveRateAndTempo();

-}

-

-

-

-// Sets new pitch control value. Original pitch = 1.0, smaller values

-// represent lower pitches, larger values higher pitch.

-void SoundTouch::setPitch(float newPitch)

-{

-    virtualPitch = newPitch;

-    calcEffectiveRateAndTempo();

-}

-

-

-

-// Sets pitch change in octaves compared to the original pitch

-// (-1.00 .. +1.00)

-void SoundTouch::setPitchOctaves(float newPitch)

-{

-    virtualPitch = (float)exp(0.69314718056f * newPitch);

-    calcEffectiveRateAndTempo();

-}

-

-

-

-// Sets pitch change in semi-tones compared to the original pitch

-// (-12 .. +12)

-void SoundTouch::setPitchSemiTones(int newPitch)

-{

-    setPitchOctaves((float)newPitch / 12.0f);

-}

-

-

-

-void SoundTouch::setPitchSemiTones(float newPitch)

-{

-    setPitchOctaves(newPitch / 12.0f);

-}

-

-

-// Calculates 'effective' rate and tempo values from the

-// nominal control values.

-void SoundTouch::calcEffectiveRateAndTempo()

-{

-    float oldTempo = tempo;

-    float oldRate = rate;

-

-    tempo = virtualTempo / virtualPitch;

-    rate = virtualPitch * virtualRate;

-

-    if (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate);

-    if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo);

-

-#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER

-    if (rate <= 1.0f) 

-    {

-        if (output != pTDStretch) 

-        {

-            FIFOSamplePipe *tempoOut;

-

-            assert(output == pRateTransposer);

-            // move samples in the current output buffer to the output of pTDStretch

-            tempoOut = pTDStretch->getOutput();

-            tempoOut->moveSamples(*output);

-            // move samples in pitch transposer's store buffer to tempo changer's input

-            pTDStretch->moveSamples(*pRateTransposer->getStore());

-

-            output = pTDStretch;

-        }

-    }

-    else

-#endif

-    {

-        if (output != pRateTransposer) 

-        {

-            FIFOSamplePipe *transOut;

-

-            assert(output == pTDStretch);

-            // move samples in the current output buffer to the output of pRateTransposer

-            transOut = pRateTransposer->getOutput();

-            transOut->moveSamples(*output);

-            // move samples in tempo changer's input to pitch transposer's input

-            pRateTransposer->moveSamples(*pTDStretch->getInput());

-

-            output = pRateTransposer;

-        }

-    } 

-}

-

-

-// Sets sample rate.

-void SoundTouch::setSampleRate(uint srate)

-{

-    bSrateSet = TRUE;

-    // set sample rate, leave other tempo changer parameters as they are.

-    pTDStretch->setParameters((int)srate);

-}

-

-

-// Adds 'numSamples' pcs of samples from the 'samples' memory position into

-// the input of the object.

-void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)

-{

-    if (bSrateSet == FALSE) 

-    {

-        throw std::runtime_error("SoundTouch : Sample rate not defined");

-    } 

-    else if (channels == 0) 

-    {

-        throw std::runtime_error("SoundTouch : Number of channels not defined");

-    }

-

-    // Transpose the rate of the new samples if necessary

-    /* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...

-    if (rate == 1.0f) 

-    {

-        // The rate value is same as the original, simply evaluate the tempo changer. 

-        assert(output == pTDStretch);

-        if (pRateTransposer->isEmpty() == 0) 

-        {

-            // yet flush the last samples in the pitch transposer buffer

-            // (may happen if 'rate' changes from a non-zero value to zero)

-            pTDStretch->moveSamples(*pRateTransposer);

-        }

-        pTDStretch->putSamples(samples, nSamples);

-    } 

-    */

-#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER

-    else if (rate <= 1.0f) 

-    {

-        // transpose the rate down, output the transposed sound to tempo changer buffer

-        assert(output == pTDStretch);

-        pRateTransposer->putSamples(samples, nSamples);

-        pTDStretch->moveSamples(*pRateTransposer);

-    } 

-    else 

-#endif

-    {

-        // evaluate the tempo changer, then transpose the rate up, 

-        assert(output == pRateTransposer);

-        pTDStretch->putSamples(samples, nSamples);

-        pRateTransposer->moveSamples(*pTDStretch);

-    }

-}

-

-

-// Flushes the last samples from the processing pipeline to the output.

-// Clears also the internal processing buffers.

-//

-// Note: This function is meant for extracting the last samples of a sound

-// stream. This function may introduce additional blank samples in the end

-// of the sound stream, and thus it's not recommended to call this function

-// in the middle of a sound stream.

-void SoundTouch::flush()

-{

-    int i;

-    uint nOut;

-    SAMPLETYPE buff[128];

-

-    nOut = numSamples();

-

-    memset(buff, 0, 128 * sizeof(SAMPLETYPE));

-    // "Push" the last active samples out from the processing pipeline by

-    // feeding blank samples into the processing pipeline until new, 

-    // processed samples appear in the output (not however, more than 

-    // 8ksamples in any case)

-    for (i = 0; i < 128; i ++) 

-    {

-        putSamples(buff, 64);

-        if (numSamples() != nOut) break;  // new samples have appeared in the output!

-    }

-

-    // Clear working buffers

-    pRateTransposer->clear();

-    pTDStretch->clearInput();

-    // yet leave the 'tempoChanger' output intouched as that's where the

-    // flushed samples are!

-}

-

-

-// Changes a setting controlling the processing system behaviour. See the

-// 'SETTING_...' defines for available setting ID's.

-BOOL SoundTouch::setSetting(int settingId, int value)

-{

-    int sampleRate, sequenceMs, seekWindowMs, overlapMs;

-

-    // read current tdstretch routine parameters

-    pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);

-

-    switch (settingId) 

-    {

-        case SETTING_USE_AA_FILTER :

-            // enables / disabless anti-alias filter

-            pRateTransposer->enableAAFilter((value != 0) ? TRUE : FALSE);

-            return TRUE;

-

-        case SETTING_AA_FILTER_LENGTH :

-            // sets anti-alias filter length

-            pRateTransposer->getAAFilter()->setLength(value);

-            return TRUE;

-

-        case SETTING_USE_QUICKSEEK :

-            // enables / disables tempo routine quick seeking algorithm

-            pTDStretch->enableQuickSeek((value != 0) ? TRUE : FALSE);

-            return TRUE;

-

-        case SETTING_SEQUENCE_MS:

-            // change time-stretch sequence duration parameter

-            pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);

-            return TRUE;

-

-        case SETTING_SEEKWINDOW_MS:

-            // change time-stretch seek window length parameter

-            pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);

-            return TRUE;

-

-        case SETTING_OVERLAP_MS:

-            // change time-stretch overlap length parameter

-            pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);

-            return TRUE;

-

-        default :

-            return FALSE;

-    }

-}

-

-

-// Reads a setting controlling the processing system behaviour. See the

-// 'SETTING_...' defines for available setting ID's.

-//

-// Returns the setting value.

-int SoundTouch::getSetting(int settingId) const

-{

-    int temp;

-

-    switch (settingId) 

-    {

-        case SETTING_USE_AA_FILTER :

-            return (uint)pRateTransposer->isAAFilterEnabled();

-

-        case SETTING_AA_FILTER_LENGTH :

-            return pRateTransposer->getAAFilter()->getLength();

-

-        case SETTING_USE_QUICKSEEK :

-            return (uint)   pTDStretch->isQuickSeekEnabled();

-

-        case SETTING_SEQUENCE_MS:

-            pTDStretch->getParameters(NULL, &temp, NULL, NULL);

-            return temp;

-

-        case SETTING_SEEKWINDOW_MS:

-            pTDStretch->getParameters(NULL, NULL, &temp, NULL);

-            return temp;

-

-        case SETTING_OVERLAP_MS:

-            pTDStretch->getParameters(NULL, NULL, NULL, &temp);

-            return temp;

-

-		case SETTING_NOMINAL_INPUT_SEQUENCE :

-			return pTDStretch->getInputSampleReq();

-

-		case SETTING_NOMINAL_OUTPUT_SEQUENCE :

-			return pTDStretch->getOutputBatchSize();

-

-		default :

-            return 0;

-    }

-}

-

-

-// Clears all the samples in the object's output and internal processing

-// buffers.

-void SoundTouch::clear()

-{

-    pRateTransposer->clear();

-    pTDStretch->clear();

-}

-

-

-

-/// Returns number of samples currently unprocessed.

-uint SoundTouch::numUnprocessedSamples() const

-{

-    FIFOSamplePipe * psp;

-    if (pTDStretch)

-    {

-        psp = pTDStretch->getInput();

-        if (psp)

-        {

-            return psp->numSamples();

-        }

-    }

-    return 0;

-}

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/TDStretch.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/TDStretch.cpp
deleted file mode 100644
index dd7a43eb5..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/TDStretch.cpp
+++ /dev/null
@@ -1,1029 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-/// 

-/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo 

-/// while maintaining the original pitch by using a time domain WSOLA-like 

-/// method with several performance-increasing tweaks.

-///

-/// Note : MMX optimized functions reside in a separate, platform-specific 

-/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 1.12 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include <string.h>

-#include <limits.h>

-#include <assert.h>

-#include <math.h>

-#include <float.h>

-#include <stdexcept>

-

-#include "STTypes.h"

-#include "cpu_detect.h"

-#include "TDStretch.h"

-

-#include <stdio.h>

-

-using namespace soundtouch;

-

-#define max(x, y) (((x) > (y)) ? (x) : (y))

-

-

-/*****************************************************************************

- *

- * Constant definitions

- *

- *****************************************************************************/

-

-// Table for the hierarchical mixing position seeking algorithm

-static const short _scanOffsets[5][24]={

-    { 124,  186,  248,  310,  372,  434,  496,  558,  620,  682,  744, 806,

-      868,  930,  992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488,   0},

-    {-100,  -75,  -50,  -25,   25,   50,   75,  100,    0,    0,    0,   0,

-        0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,   0},

-    { -20,  -15,  -10,   -5,    5,   10,   15,   20,    0,    0,    0,   0,

-        0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,   0},

-    {  -4,   -3,   -2,   -1,    1,    2,    3,    4,    0,    0,    0,   0,

-        0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,   0},

-    { 121,  114,   97,  114,   98,  105,  108,   32,  104,   99,  117,  111,

-      116,  100,  110,  117,  111,  115,    0,    0,    0,    0,    0,   0}};

-

-/*****************************************************************************

- *

- * Implementation of the class 'TDStretch'

- *

- *****************************************************************************/

-

-

-TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)

-{

-    bQuickSeek = FALSE;

-    channels = 2;

-

-    pMidBuffer = NULL;

-    pRefMidBufferUnaligned = NULL;

-    overlapLength = 0;

-

-    bAutoSeqSetting = TRUE;

-    bAutoSeekSetting = TRUE;

-

-//    outDebt = 0;

-    skipFract = 0;

-

-    tempo = 1.0f;

-    setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);

-    setTempo(1.0f);

-

-    clear();

-}

-

-

-

-TDStretch::~TDStretch()

-{

-    delete[] pMidBuffer;

-    delete[] pRefMidBufferUnaligned;

-}

-

-

-

-// Sets routine control parameters. These control are certain time constants

-// defining how the sound is stretched to the desired duration.

-//

-// 'sampleRate' = sample rate of the sound

-// 'sequenceMS' = one processing sequence length in milliseconds (default = 82 ms)

-// 'seekwindowMS' = seeking window length for scanning the best overlapping 

-//      position (default = 28 ms)

-// 'overlapMS' = overlapping length (default = 12 ms)

-

-void TDStretch::setParameters(int aSampleRate, int aSequenceMS, 

-                              int aSeekWindowMS, int aOverlapMS)

-{

-    // accept only positive parameter values - if zero or negative, use old values instead

-    if (aSampleRate > 0)   this->sampleRate = aSampleRate;

-    if (aOverlapMS > 0)    this->overlapMs = aOverlapMS;

-

-    if (aSequenceMS > 0)

-    {

-        this->sequenceMs = aSequenceMS;

-        bAutoSeqSetting = FALSE;

-    } 

-    else if (aSequenceMS == 0)

-    {

-        // if zero, use automatic setting

-        bAutoSeqSetting = TRUE;

-    }

-

-    if (aSeekWindowMS > 0) 

-    {

-        this->seekWindowMs = aSeekWindowMS;

-        bAutoSeekSetting = FALSE;

-    } 

-    else if (aSeekWindowMS == 0) 

-    {

-        // if zero, use automatic setting

-        bAutoSeekSetting = TRUE;

-    }

-

-    calcSeqParameters();

-

-    calculateOverlapLength(overlapMs);

-

-    // set tempo to recalculate 'sampleReq'

-    setTempo(tempo);

-

-}

-

-

-

-/// Get routine control parameters, see setParameters() function.

-/// Any of the parameters to this function can be NULL, in such case corresponding parameter

-/// value isn't returned.

-void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const

-{

-    if (pSampleRate)

-    {

-        *pSampleRate = sampleRate;

-    }

-

-    if (pSequenceMs)

-    {

-        *pSequenceMs = (bAutoSeqSetting) ? (USE_AUTO_SEQUENCE_LEN) : sequenceMs;

-    }

-

-    if (pSeekWindowMs)

-    {

-        *pSeekWindowMs = (bAutoSeekSetting) ? (USE_AUTO_SEEKWINDOW_LEN) : seekWindowMs;

-    }

-

-    if (pOverlapMs)

-    {

-        *pOverlapMs = overlapMs;

-    }

-}

-

-

-// Overlaps samples in 'midBuffer' with the samples in 'pInput'

-void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const

-{

-    int i, itemp;

-

-    for (i = 0; i < overlapLength ; i ++) 

-    {

-        itemp = overlapLength - i;

-        pOutput[i] = (pInput[i] * i + pMidBuffer[i] * itemp ) / overlapLength;    // >> overlapDividerBits;

-    }

-}

-

-

-

-void TDStretch::clearMidBuffer()

-{

-    memset(pMidBuffer, 0, 2 * sizeof(SAMPLETYPE) * overlapLength);

-}

-

-

-void TDStretch::clearInput()

-{

-    inputBuffer.clear();

-    clearMidBuffer();

-}

-

-

-// Clears the sample buffers

-void TDStretch::clear()

-{

-    outputBuffer.clear();

-    clearInput();

-}

-

-

-

-// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero

-// to enable

-void TDStretch::enableQuickSeek(BOOL enable)

-{

-    bQuickSeek = enable;

-}

-

-

-// Returns nonzero if the quick seeking algorithm is enabled.

-BOOL TDStretch::isQuickSeekEnabled() const

-{

-    return bQuickSeek;

-}

-

-

-// Seeks for the optimal overlap-mixing position.

-int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos)

-{

-    if (channels == 2) 

-    {

-        // stereo sound

-        if (bQuickSeek) 

-        {

-            return seekBestOverlapPositionStereoQuick(refPos);

-        } 

-        else 

-        {

-            return seekBestOverlapPositionStereo(refPos);

-        }

-    } 

-    else 

-    {

-        // mono sound

-        if (bQuickSeek) 

-        {

-            return seekBestOverlapPositionMonoQuick(refPos);

-        } 

-        else 

-        {

-            return seekBestOverlapPositionMono(refPos);

-        }

-    }

-}

-

-

-

-

-// Overlaps samples in 'midBuffer' with the samples in 'pInputBuffer' at position

-// of 'ovlPos'.

-inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const

-{

-    if (channels == 2) 

-    {

-        // stereo sound

-        overlapStereo(pOutput, pInput + 2 * ovlPos);

-    } else {

-        // mono sound.

-        overlapMono(pOutput, pInput + ovlPos);

-    }

-}

-

-

-

-

-// Seeks for the optimal overlap-mixing position. The 'stereo' version of the

-// routine

-//

-// The best position is determined as the position where the two overlapped

-// sample sequences are 'most alike', in terms of the highest cross-correlation

-// value over the overlapping period

-int TDStretch::seekBestOverlapPositionStereo(const SAMPLETYPE *refPos) 

-{

-    int bestOffs;

-    double bestCorr, corr;

-    int i;

-

-    // Slopes the amplitudes of the 'midBuffer' samples

-    precalcCorrReferenceStereo();

-

-    bestCorr = FLT_MIN;

-    bestOffs = 0;

-

-    // Scans for the best correlation value by testing each possible position

-    // over the permitted range.

-    for (i = 0; i < seekLength; i ++) 

-    {

-        // Calculates correlation value for the mixing position corresponding

-        // to 'i'

-        corr = (double)calcCrossCorrStereo(refPos + 2 * i, pRefMidBuffer);

-        // heuristic rule to slightly favour values close to mid of the range

-        double tmp = (double)(2 * i - seekLength) / (double)seekLength;

-        corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));

-

-        // Checks for the highest correlation value

-        if (corr > bestCorr) 

-        {

-            bestCorr = corr;

-            bestOffs = i;

-        }

-    }

-    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).

-    clearCrossCorrState();

-

-    return bestOffs;

-}

-

-

-// Seeks for the optimal overlap-mixing position. The 'stereo' version of the

-// routine

-//

-// The best position is determined as the position where the two overlapped

-// sample sequences are 'most alike', in terms of the highest cross-correlation

-// value over the overlapping period

-int TDStretch::seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos) 

-{

-    int j;

-    int bestOffs;

-    double bestCorr, corr;

-    int scanCount, corrOffset, tempOffset;

-

-    // Slopes the amplitude of the 'midBuffer' samples

-    precalcCorrReferenceStereo();

-

-    bestCorr = FLT_MIN;

-    bestOffs = _scanOffsets[0][0];

-    corrOffset = 0;

-    tempOffset = 0;

-

-    // Scans for the best correlation value using four-pass hierarchical search.

-    //

-    // The look-up table 'scans' has hierarchical position adjusting steps.

-    // In first pass the routine searhes for the highest correlation with 

-    // relatively coarse steps, then rescans the neighbourhood of the highest

-    // correlation with better resolution and so on.

-    for (scanCount = 0;scanCount < 4; scanCount ++) 

-    {

-        j = 0;

-        while (_scanOffsets[scanCount][j]) 

-        {

-            tempOffset = corrOffset + _scanOffsets[scanCount][j];

-            if (tempOffset >= seekLength) break;

-

-            // Calculates correlation value for the mixing position corresponding

-            // to 'tempOffset'

-            corr = (double)calcCrossCorrStereo(refPos + 2 * tempOffset, pRefMidBuffer);

-            // heuristic rule to slightly favour values close to mid of the range

-            double tmp = (double)(2 * tempOffset - seekLength) / seekLength;

-            corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));

-

-            // Checks for the highest correlation value

-            if (corr > bestCorr) 

-            {

-                bestCorr = corr;

-                bestOffs = tempOffset;

-            }

-            j ++;

-        }

-        corrOffset = bestOffs;

-    }

-    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).

-    clearCrossCorrState();

-

-    return bestOffs;

-}

-

-

-

-// Seeks for the optimal overlap-mixing position. The 'mono' version of the

-// routine

-//

-// The best position is determined as the position where the two overlapped

-// sample sequences are 'most alike', in terms of the highest cross-correlation

-// value over the overlapping period

-int TDStretch::seekBestOverlapPositionMono(const SAMPLETYPE *refPos) 

-{

-    int bestOffs;

-    double bestCorr, corr;

-    int tempOffset;

-    const SAMPLETYPE *compare;

-

-    // Slopes the amplitude of the 'midBuffer' samples

-    precalcCorrReferenceMono();

-

-    bestCorr = FLT_MIN;

-    bestOffs = 0;

-

-    // Scans for the best correlation value by testing each possible position

-    // over the permitted range.

-    for (tempOffset = 0; tempOffset < seekLength; tempOffset ++) 

-    {

-        compare = refPos + tempOffset;

-

-        // Calculates correlation value for the mixing position corresponding

-        // to 'tempOffset'

-        corr = (double)calcCrossCorrMono(pRefMidBuffer, compare);

-        // heuristic rule to slightly favour values close to mid of the range

-        double tmp = (double)(2 * tempOffset - seekLength) / seekLength;

-        corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));

-

-        // Checks for the highest correlation value

-        if (corr > bestCorr) 

-        {

-            bestCorr = corr;

-            bestOffs = tempOffset;

-        }

-    }

-    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).

-    clearCrossCorrState();

-

-    return bestOffs;

-}

-

-

-// Seeks for the optimal overlap-mixing position. The 'mono' version of the

-// routine

-//

-// The best position is determined as the position where the two overlapped

-// sample sequences are 'most alike', in terms of the highest cross-correlation

-// value over the overlapping period

-int TDStretch::seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos) 

-{

-    int j;

-    int bestOffs;

-    double bestCorr, corr;

-    int scanCount, corrOffset, tempOffset;

-

-    // Slopes the amplitude of the 'midBuffer' samples

-    precalcCorrReferenceMono();

-

-    bestCorr = FLT_MIN;

-    bestOffs = _scanOffsets[0][0];

-    corrOffset = 0;

-    tempOffset = 0;

-

-    // Scans for the best correlation value using four-pass hierarchical search.

-    //

-    // The look-up table 'scans' has hierarchical position adjusting steps.

-    // In first pass the routine searhes for the highest correlation with 

-    // relatively coarse steps, then rescans the neighbourhood of the highest

-    // correlation with better resolution and so on.

-    for (scanCount = 0;scanCount < 4; scanCount ++) 

-    {

-        j = 0;

-        while (_scanOffsets[scanCount][j]) 

-        {

-            tempOffset = corrOffset + _scanOffsets[scanCount][j];

-            if (tempOffset >= seekLength) break;

-

-            // Calculates correlation value for the mixing position corresponding

-            // to 'tempOffset'

-            corr = (double)calcCrossCorrMono(refPos + tempOffset, pRefMidBuffer);

-            // heuristic rule to slightly favour values close to mid of the range

-            double tmp = (double)(2 * tempOffset - seekLength) / seekLength;

-            corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp));

-

-            // Checks for the highest correlation value

-            if (corr > bestCorr) 

-            {

-                bestCorr = corr;

-                bestOffs = tempOffset;

-            }

-            j ++;

-        }

-        corrOffset = bestOffs;

-    }

-    // clear cross correlation routine state if necessary (is so e.g. in MMX routines).

-    clearCrossCorrState();

-

-    return bestOffs;

-}

-

-

-/// clear cross correlation routine state if necessary 

-void TDStretch::clearCrossCorrState()

-{

-    // default implementation is empty.

-}

-

-

-/// Calculates processing sequence length according to tempo setting

-void TDStretch::calcSeqParameters()

-{

-    // Adjust tempo param according to tempo, so that variating processing sequence length is used

-    // at varius tempo settings, between the given low...top limits

-    #define AUTOSEQ_TEMPO_LOW   0.5     // auto setting low tempo range (-50%)

-    #define AUTOSEQ_TEMPO_TOP   2.0     // auto setting top tempo range (+100%)

-

-    // sequence-ms setting values at above low & top tempo

-    #define AUTOSEQ_AT_MIN      125.0

-    #define AUTOSEQ_AT_MAX      50.0

-    #define AUTOSEQ_K           ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))

-    #define AUTOSEQ_C           (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW))

-

-    // seek-window-ms setting values at above low & top tempo

-    #define AUTOSEEK_AT_MIN     25.0

-    #define AUTOSEEK_AT_MAX     15.0

-    #define AUTOSEEK_K          ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))

-    #define AUTOSEEK_C          (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW))

-

-    #define CHECK_LIMITS(x, mi, ma) (((x) < (mi)) ? (mi) : (((x) > (ma)) ? (ma) : (x)))

-

-    double seq, seek;

-    

-    if (bAutoSeqSetting)

-    {

-        seq = AUTOSEQ_C + AUTOSEQ_K * tempo;

-        seq = CHECK_LIMITS(seq, AUTOSEQ_AT_MAX, AUTOSEQ_AT_MIN);

-        sequenceMs = (int)(seq + 0.5);

-    }

-

-    if (bAutoSeekSetting)

-    {

-        seek = AUTOSEEK_C + AUTOSEEK_K * tempo;

-        seek = CHECK_LIMITS(seek, AUTOSEEK_AT_MAX, AUTOSEEK_AT_MIN);

-        seekWindowMs = (int)(seek + 0.5);

-    }

-

-    // Update seek window lengths

-    seekWindowLength = (sampleRate * sequenceMs) / 1000;

-    if (seekWindowLength < 2 * overlapLength) 

-    {

-        seekWindowLength = 2 * overlapLength;

-    }

-    seekLength = (sampleRate * seekWindowMs) / 1000;

-}

-

-

-

-// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower 

-// tempo, larger faster tempo.

-void TDStretch::setTempo(float newTempo)

-{

-    int intskip;

-

-    tempo = newTempo;

-

-    // Calculate new sequence duration

-    calcSeqParameters();

-

-    // Calculate ideal skip length (according to tempo value) 

-    nominalSkip = tempo * (seekWindowLength - overlapLength);

-    intskip = (int)(nominalSkip + 0.5f);

-

-    // Calculate how many samples are needed in the 'inputBuffer' to 

-    // process another batch of samples

-    //sampleReq = max(intskip + overlapLength, seekWindowLength) + seekLength / 2;

-    sampleReq = max(intskip + overlapLength, seekWindowLength) + seekLength;

-}

-

-

-

-// Sets the number of channels, 1 = mono, 2 = stereo

-void TDStretch::setChannels(int numChannels)

-{

-    assert(numChannels > 0);

-    if (channels == numChannels) return;

-    assert(numChannels == 1 || numChannels == 2);

-

-    channels = numChannels;

-    inputBuffer.setChannels(channels);

-    outputBuffer.setChannels(channels);

-}

-

-

-// nominal tempo, no need for processing, just pass the samples through

-// to outputBuffer

-/*

-void TDStretch::processNominalTempo()

-{

-    assert(tempo == 1.0f);

-

-    if (bMidBufferDirty) 

-    {

-        // If there are samples in pMidBuffer waiting for overlapping,

-        // do a single sliding overlapping with them in order to prevent a 

-        // clicking distortion in the output sound

-        if (inputBuffer.numSamples() < overlapLength) 

-        {

-            // wait until we've got overlapLength input samples

-            return;

-        }

-        // Mix the samples in the beginning of 'inputBuffer' with the 

-        // samples in 'midBuffer' using sliding overlapping 

-        overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), 0);

-        outputBuffer.putSamples(overlapLength);

-        inputBuffer.receiveSamples(overlapLength);

-        clearMidBuffer();

-        // now we've caught the nominal sample flow and may switch to

-        // bypass mode

-    }

-

-    // Simply bypass samples from input to output

-    outputBuffer.moveSamples(inputBuffer);

-}

-*/

-

-#include <stdio.h>

-

-// Processes as many processing frames of the samples 'inputBuffer', store

-// the result into 'outputBuffer'

-void TDStretch::processSamples()

-{

-    int ovlSkip, offset;

-    int temp;

-

-    /* Removed this small optimization - can introduce a click to sound when tempo setting

-       crosses the nominal value

-    if (tempo == 1.0f) 

-    {

-        // tempo not changed from the original, so bypass the processing

-        processNominalTempo();

-        return;

-    }

-    */

-

-    // Process samples as long as there are enough samples in 'inputBuffer'

-    // to form a processing frame.

-    while ((int)inputBuffer.numSamples() >= sampleReq) 

-    {

-        // If tempo differs from the normal ('SCALE'), scan for the best overlapping

-        // position

-        offset = seekBestOverlapPosition(inputBuffer.ptrBegin());

-

-        // Mix the samples in the 'inputBuffer' at position of 'offset' with the 

-        // samples in 'midBuffer' using sliding overlapping

-        // ... first partially overlap with the end of the previous sequence

-        // (that's in 'midBuffer')

-        overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);

-        outputBuffer.putSamples((uint)overlapLength);

-

-        // ... then copy sequence samples from 'inputBuffer' to output:

-

-        // length of sequence

-        temp = (seekWindowLength - 2 * overlapLength);

-

-        // crosscheck that we don't have buffer overflow...

-        if ((int)inputBuffer.numSamples() < (offset + temp + overlapLength * 2))

-        {

-            continue;    // just in case, shouldn't really happen

-        }

-

-        outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), (uint)temp);

-

-        // Copies the end of the current sequence from 'inputBuffer' to 

-        // 'midBuffer' for being mixed with the beginning of the next 

-        // processing sequence and so on

-        assert((offset + temp + overlapLength * 2) <= (int)inputBuffer.numSamples());

-        memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp + overlapLength), 

-            channels * sizeof(SAMPLETYPE) * overlapLength);

-

-        // Remove the processed samples from the input buffer. Update

-        // the difference between integer & nominal skip step to 'skipFract'

-        // in order to prevent the error from accumulating over time.

-        skipFract += nominalSkip;   // real skip size

-        ovlSkip = (int)skipFract;   // rounded to integer skip

-        skipFract -= ovlSkip;       // maintain the fraction part, i.e. real vs. integer skip

-        inputBuffer.receiveSamples((uint)ovlSkip);

-    }

-}

-

-

-// Adds 'numsamples' pcs of samples from the 'samples' memory position into

-// the input of the object.

-void TDStretch::putSamples(const SAMPLETYPE *samples, uint nSamples)

-{

-    // Add the samples into the input buffer

-    inputBuffer.putSamples(samples, nSamples);

-    // Process the samples in input buffer

-    processSamples();

-}

-

-

-

-/// Set new overlap length parameter & reallocate RefMidBuffer if necessary.

-void TDStretch::acceptNewOverlapLength(int newOverlapLength)

-{

-    int prevOvl;

-

-    assert(newOverlapLength >= 0);

-    prevOvl = overlapLength;

-    overlapLength = newOverlapLength;

-

-    if (overlapLength > prevOvl)

-    {

-        delete[] pMidBuffer;

-        delete[] pRefMidBufferUnaligned;

-

-        pMidBuffer = new SAMPLETYPE[overlapLength * 2];

-        clearMidBuffer();

-

-        pRefMidBufferUnaligned = new SAMPLETYPE[2 * overlapLength + 16 / sizeof(SAMPLETYPE)];

-        // ensure that 'pRefMidBuffer' is aligned to 16 byte boundary for efficiency

-        pRefMidBuffer = (SAMPLETYPE *)((((ulong)pRefMidBufferUnaligned) + 15) & (ulong)-16);

-    }

-}

-

-

-// Operator 'new' is overloaded so that it automatically creates a suitable instance 

-// depending on if we've a MMX/SSE/etc-capable CPU available or not.

-void * TDStretch::operator new(size_t /*s*/)

-{

-    // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!

-    throw std::runtime_error("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");

-    return NULL;

-}

-

-

-TDStretch * TDStretch::newInstance()

-{

-#ifndef _WIN64 //mpc custom code

-    uint uExtensions;

-

-    uExtensions = detectCPUextensions();

-

-    // Check if MMX/SSE instruction set extensions supported by CPU

-

-#ifdef SOUNDTOUCH_ALLOW_MMX

-    // MMX routines available only with integer sample types

-    if (uExtensions & SUPPORT_MMX)

-    {

-        return ::new TDStretchMMX;

-    }

-    else

-#endif // SOUNDTOUCH_ALLOW_MMX

-

-

-#ifdef SOUNDTOUCH_ALLOW_SSE

-    if (uExtensions & SUPPORT_SSE)

-    {

-        // SSE support

-        return ::new TDStretchSSE;

-    }

-    else

-#endif // SOUNDTOUCH_ALLOW_SSE

-

-#endif // _WIN64 mpc custom code

-

-    {

-        // ISA optimizations not supported, use plain C version

-        return ::new TDStretch;

-    }

-}

-

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// Integer arithmetics specific algorithm implementations.

-//

-//////////////////////////////////////////////////////////////////////////////

-

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-

-// Slopes the amplitude of the 'midBuffer' samples so that cross correlation

-// is faster to calculate

-void TDStretch::precalcCorrReferenceStereo()

-{

-    int i, cnt2;

-    int temp, temp2;

-

-    for (i=0 ; i < (int)overlapLength ;i ++) 

-    {

-        temp = i * (overlapLength - i);

-        cnt2 = i * 2;

-

-        temp2 = (pMidBuffer[cnt2] * temp) / slopingDivider;

-        pRefMidBuffer[cnt2] = (short)(temp2);

-        temp2 = (pMidBuffer[cnt2 + 1] * temp) / slopingDivider;

-        pRefMidBuffer[cnt2 + 1] = (short)(temp2);

-    }

-}

-

-

-// Slopes the amplitude of the 'midBuffer' samples so that cross correlation

-// is faster to calculate

-void TDStretch::precalcCorrReferenceMono()

-{

-    int i;

-    long temp;

-    long temp2;

-

-    for (i=0 ; i < (int)overlapLength ;i ++) 

-    {

-        temp = i * (overlapLength - i);

-        temp2 = (pMidBuffer[i] * temp) / slopingDivider;

-        pRefMidBuffer[i] = (short)temp2;

-    }

-}

-

-

-// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo' 

-// version of the routine.

-void TDStretch::overlapStereo(short *poutput, const short *input) const

-{

-    int i;

-    short temp;

-    int cnt2;

-

-    for (i = 0; i < overlapLength ; i ++) 

-    {

-        temp = (short)(overlapLength - i);

-        cnt2 = 2 * i;

-        poutput[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp )  / overlapLength;

-        poutput[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength;

-    }

-}

-

-// Calculates the x having the closest 2^x value for the given value

-static int _getClosest2Power(double value)

-{

-    return (int)(log(value) / log(2.0) + 0.5);

-}

-

-

-/// Calculates overlap period length in samples.

-/// Integer version rounds overlap length to closest power of 2

-/// for a divide scaling operation.

-void TDStretch::calculateOverlapLength(int aoverlapMs)

-{

-    int newOvl;

-

-    assert(aoverlapMs >= 0);

-

-    // calculate overlap length so that it's power of 2 - thus it's easy to do

-    // integer division by right-shifting. Term "-1" at end is to account for 

-    // the extra most significatnt bit left unused in result by signed multiplication 

-    overlapDividerBits = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1;

-    if (overlapDividerBits > 9) overlapDividerBits = 9;

-    if (overlapDividerBits < 3) overlapDividerBits = 3;

-    newOvl = (int)pow(2.0, (int)overlapDividerBits + 1);    // +1 => account for -1 above

-

-    acceptNewOverlapLength(newOvl);

-

-    // calculate sloping divider so that crosscorrelation operation won't 

-    // overflow 32-bit register. Max. sum of the crosscorrelation sum without 

-    // divider would be 2^30*(N^3-N)/3, where N = overlap length

-    slopingDivider = (newOvl * newOvl - 1) / 3;

-}

-

-

-long TDStretch::calcCrossCorrMono(const short *mixingPos, const short *compare) const

-{

-    long corr;

-    long norm;

-    int i;

-

-    corr = norm = 0;

-    for (i = 1; i < overlapLength; i ++) 

-    {

-        corr += (mixingPos[i] * compare[i]) >> overlapDividerBits;

-        norm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBits;

-    }

-

-    // Normalize result by dividing by sqrt(norm) - this step is easiest 

-    // done using floating point operation

-    if (norm == 0) norm = 1;    // to avoid div by zero

-    return (long)((double)corr * SHRT_MAX / sqrt((double)norm));

-}

-

-

-long TDStretch::calcCrossCorrStereo(const short *mixingPos, const short *compare) const

-{

-    long corr;

-    long norm;

-    int i;

-

-    corr = norm = 0;

-    for (i = 2; i < 2 * overlapLength; i += 2) 

-    {

-        corr += (mixingPos[i] * compare[i] +

-                 mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBits;

-        norm += (mixingPos[i] * mixingPos[i] + mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBits;

-    }

-

-    // Normalize result by dividing by sqrt(norm) - this step is easiest 

-    // done using floating point operation

-    if (norm == 0) norm = 1;    // to avoid div by zero

-    return (long)((double)corr * SHRT_MAX / sqrt((double)norm));

-}

-

-#endif // SOUNDTOUCH_INTEGER_SAMPLES

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// Floating point arithmetics specific algorithm implementations.

-//

-

-#ifdef SOUNDTOUCH_FLOAT_SAMPLES

-

-

-// Slopes the amplitude of the 'midBuffer' samples so that cross correlation

-// is faster to calculate

-void TDStretch::precalcCorrReferenceStereo()

-{

-    int i, cnt2;

-    float temp;

-

-    for (i=0 ; i < (int)overlapLength ;i ++) 

-    {

-        temp = (float)i * (float)(overlapLength - i);

-        cnt2 = i * 2;

-        pRefMidBuffer[cnt2] = (float)(pMidBuffer[cnt2] * temp);

-        pRefMidBuffer[cnt2 + 1] = (float)(pMidBuffer[cnt2 + 1] * temp);

-    }

-}

-

-

-// Slopes the amplitude of the 'midBuffer' samples so that cross correlation

-// is faster to calculate

-void TDStretch::precalcCorrReferenceMono()

-{

-    int i;

-    float temp;

-

-    for (i=0 ; i < (int)overlapLength ;i ++) 

-    {

-        temp = (float)i * (float)(overlapLength - i);

-        pRefMidBuffer[i] = (float)(pMidBuffer[i] * temp);

-    }

-}

-

-

-// Overlaps samples in 'midBuffer' with the samples in 'pInput'

-void TDStretch::overlapStereo(float *pOutput, const float *pInput) const

-{

-    int i;

-    int cnt2;

-    float fTemp;

-    float fScale;

-    float fi;

-

-    fScale = 1.0f / (float)overlapLength;

-

-    for (i = 0; i < (int)overlapLength ; i ++) 

-    {

-        fTemp = (float)(overlapLength - i) * fScale;

-        fi = (float)i * fScale;

-        cnt2 = 2 * i;

-        pOutput[cnt2 + 0] = pInput[cnt2 + 0] * fi + pMidBuffer[cnt2 + 0] * fTemp;

-        pOutput[cnt2 + 1] = pInput[cnt2 + 1] * fi + pMidBuffer[cnt2 + 1] * fTemp;

-    }

-}

-

-

-/// Calculates overlapInMsec period length in samples.

-void TDStretch::calculateOverlapLength(int overlapInMsec)

-{

-    int newOvl;

-

-    assert(overlapInMsec >= 0);

-    newOvl = (sampleRate * overlapInMsec) / 1000;

-    if (newOvl < 16) newOvl = 16;

-

-    // must be divisible by 8

-    newOvl -= newOvl % 8;

-

-    acceptNewOverlapLength(newOvl);

-}

-

-

-

-double TDStretch::calcCrossCorrMono(const float *mixingPos, const float *compare) const

-{

-    double corr;

-    double norm;

-    int i;

-

-    corr = norm = 0;

-    for (i = 1; i < overlapLength; i ++) 

-    {

-        corr += mixingPos[i] * compare[i];

-        norm += mixingPos[i] * mixingPos[i];

-    }

-

-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero

-    return corr / sqrt(norm);

-}

-

-

-double TDStretch::calcCrossCorrStereo(const float *mixingPos, const float *compare) const

-{

-    double corr;

-    double norm;

-    int i;

-

-    corr = norm = 0;

-    for (i = 2; i < 2 * overlapLength; i += 2) 

-    {

-        corr += mixingPos[i] * compare[i] +

-                mixingPos[i + 1] * compare[i + 1];

-        norm += mixingPos[i] * mixingPos[i] + 

-                mixingPos[i + 1] * mixingPos[i + 1];

-    }

-

-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero

-    return corr / sqrt(norm);

-}

-

-#endif // SOUNDTOUCH_FLOAT_SAMPLES

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/TDStretch.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/TDStretch.h
deleted file mode 100644
index c236aa4e7..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/TDStretch.h
+++ /dev/null
@@ -1,277 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-/// 

-/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo 

-/// while maintaining the original pitch by using a time domain WSOLA-like method 

-/// with several performance-increasing tweaks.

-///

-/// Note : MMX/SSE optimized functions reside in separate, platform-specific files 

-/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef TDStretch_H

-#define TDStretch_H

-

-#include <stddef.h>

-#include "STTypes.h"

-#include "RateTransposer.h"

-#include "FIFOSamplePipe.h"

-

-namespace soundtouch

-{

-

-/// Default values for sound processing parameters:

-/// Notice that the default parameters are tuned for contemporary popular music 

-/// processing. For speech processing applications these parameters suit better:

-///     #define DEFAULT_SEQUENCE_MS     40

-///     #define DEFAULT_SEEKWINDOW_MS   15

-///     #define DEFAULT_OVERLAP_MS      8

-///

-

-/// Default length of a single processing sequence, in milliseconds. This determines to how 

-/// long sequences the original sound is chopped in the time-stretch algorithm.

-///

-/// The larger this value is, the lesser sequences are used in processing. In principle

-/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo

-/// and vice versa.

-///

-/// Increasing this value reduces computational burden & vice versa.

-//#define DEFAULT_SEQUENCE_MS         40

-#define DEFAULT_SEQUENCE_MS         USE_AUTO_SEQUENCE_LEN

-

-/// Giving this value for the sequence length sets automatic parameter value

-/// according to tempo setting (recommended)

-#define USE_AUTO_SEQUENCE_LEN       0

-

-/// Seeking window default length in milliseconds for algorithm that finds the best possible 

-/// overlapping location. This determines from how wide window the algorithm may look for an 

-/// optimal joining location when mixing the sound sequences back together. 

-///

-/// The bigger this window setting is, the higher the possibility to find a better mixing

-/// position will become, but at the same time large values may cause a "drifting" artifact

-/// because consequent sequences will be taken at more uneven intervals.

-///

-/// If there's a disturbing artifact that sounds as if a constant frequency was drifting 

-/// around, try reducing this setting.

-///

-/// Increasing this value increases computational burden & vice versa.

-//#define DEFAULT_SEEKWINDOW_MS       15

-#define DEFAULT_SEEKWINDOW_MS       USE_AUTO_SEEKWINDOW_LEN

-

-/// Giving this value for the seek window length sets automatic parameter value

-/// according to tempo setting (recommended)

-#define USE_AUTO_SEEKWINDOW_LEN     0

-

-/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together, 

-/// to form a continuous sound stream, this parameter defines over how long period the two 

-/// consecutive sequences are let to overlap each other. 

-///

-/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting 

-/// by a large amount, you might wish to try a smaller value on this.

-///

-/// Increasing this value increases computational burden & vice versa.

-#define DEFAULT_OVERLAP_MS      8

-

-

-/// Class that does the time-stretch (tempo change) effect for the processed

-/// sound.

-class TDStretch : public FIFOProcessor

-{

-protected:

-    int channels;

-    int sampleReq;

-    float tempo;

-

-    SAMPLETYPE *pMidBuffer;

-    SAMPLETYPE *pRefMidBuffer;

-    SAMPLETYPE *pRefMidBufferUnaligned;

-    int overlapLength;

-    int seekLength;

-    int seekWindowLength;

-    int overlapDividerBits;

-    int slopingDivider;

-    float nominalSkip;

-    float skipFract;

-    FIFOSampleBuffer outputBuffer;

-    FIFOSampleBuffer inputBuffer;

-    BOOL bQuickSeek;

-//    int outDebt;

-//    BOOL bMidBufferDirty;

-

-    int sampleRate;

-    int sequenceMs;

-    int seekWindowMs;

-    int overlapMs;

-    BOOL bAutoSeqSetting;

-    BOOL bAutoSeekSetting;

-

-    void acceptNewOverlapLength(int newOverlapLength);

-

-    virtual void clearCrossCorrState();

-    void calculateOverlapLength(int overlapMs);

-

-    virtual LONG_SAMPLETYPE calcCrossCorrStereo(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;

-    virtual LONG_SAMPLETYPE calcCrossCorrMono(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;

-

-    virtual int seekBestOverlapPositionStereo(const SAMPLETYPE *refPos);

-    virtual int seekBestOverlapPositionStereoQuick(const SAMPLETYPE *refPos);

-    virtual int seekBestOverlapPositionMono(const SAMPLETYPE *refPos);

-    virtual int seekBestOverlapPositionMonoQuick(const SAMPLETYPE *refPos);

-    int seekBestOverlapPosition(const SAMPLETYPE *refPos);

-

-    virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;

-    virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;

-

-    void clearMidBuffer();

-    void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;

-

-    void precalcCorrReferenceMono();

-    void precalcCorrReferenceStereo();

-

-    void calcSeqParameters();

-

-    /// Changes the tempo of the given sound samples.

-    /// Returns amount of samples returned in the "output" buffer.

-    /// The maximum amount of samples that can be returned at a time is set by

-    /// the 'set_returnBuffer_size' function.

-    void processSamples();

-    

-public:

-    TDStretch();

-    virtual ~TDStretch();

-

-    /// Operator 'new' is overloaded so that it automatically creates a suitable instance 

-    /// depending on if we've a MMX/SSE/etc-capable CPU available or not.

-    static void *operator new(size_t s);

-

-    /// Use this function instead of "new" operator to create a new instance of this class. 

-    /// This function automatically chooses a correct feature set depending on if the CPU

-    /// supports MMX/SSE/etc extensions.

-    static TDStretch *newInstance();

-    

-    /// Returns the output buffer object

-    FIFOSamplePipe *getOutput() { return &outputBuffer; };

-

-    /// Returns the input buffer object

-    FIFOSamplePipe *getInput() { return &inputBuffer; };

-

-    /// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower 

-    /// tempo, larger faster tempo.

-    void setTempo(float newTempo);

-

-    /// Returns nonzero if there aren't any samples available for outputting.

-    virtual void clear();

-

-    /// Clears the input buffer

-    void clearInput();

-

-    /// Sets the number of channels, 1 = mono, 2 = stereo

-    void setChannels(int numChannels);

-

-    /// Enables/disables the quick position seeking algorithm. Zero to disable, 

-    /// nonzero to enable

-    void enableQuickSeek(BOOL enable);

-

-    /// Returns nonzero if the quick seeking algorithm is enabled.

-    BOOL isQuickSeekEnabled() const;

-

-    /// Sets routine control parameters. These control are certain time constants

-    /// defining how the sound is stretched to the desired duration.

-    //

-    /// 'sampleRate' = sample rate of the sound

-    /// 'sequenceMS' = one processing sequence length in milliseconds

-    /// 'seekwindowMS' = seeking window length for scanning the best overlapping 

-    ///      position

-    /// 'overlapMS' = overlapping length

-    void setParameters(int sampleRate,          ///< Samplerate of sound being processed (Hz)

-                       int sequenceMS = -1,     ///< Single processing sequence length (ms)

-                       int seekwindowMS = -1,   ///< Offset seeking window length (ms)

-                       int overlapMS = -1       ///< Sequence overlapping length (ms)

-                       );

-

-    /// Get routine control parameters, see setParameters() function.

-    /// Any of the parameters to this function can be NULL, in such case corresponding parameter

-    /// value isn't returned.

-    void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;

-

-    /// Adds 'numsamples' pcs of samples from the 'samples' memory position into

-    /// the input of the object.

-    virtual void putSamples(

-            const SAMPLETYPE *samples,  ///< Input sample data

-            uint numSamples                         ///< Number of samples in 'samples' so that one sample

-                                                    ///< contains both channels if stereo

-            );

-

-	/// return nominal input sample requirement for triggering a processing batch

-	int getInputSampleReq() const

-	{

-		return (int)(nominalSkip + 0.5);

-	}

-

-	/// return nominal output sample amount when running a processing batch

-	int getOutputBatchSize() const

-	{

-		return seekWindowLength - overlapLength;

-	}

-};

-

-

-

-// Implementation-specific class declarations:

-

-#ifdef SOUNDTOUCH_ALLOW_MMX

-    /// Class that implements MMX optimized routines for 16bit integer samples type.

-    class TDStretchMMX : public TDStretch

-    {

-    protected:

-        long calcCrossCorrStereo(const short *mixingPos, const short *compare) const;

-        virtual void overlapStereo(short *output, const short *input) const;

-        virtual void clearCrossCorrState();

-    };

-#endif /// SOUNDTOUCH_ALLOW_MMX

-

-

-#ifdef SOUNDTOUCH_ALLOW_SSE

-    /// Class that implements SSE optimized routines for floating point samples type.

-    class TDStretchSSE : public TDStretch

-    {

-    protected:

-        double calcCrossCorrStereo(const float *mixingPos, const float *compare) const;

-    };

-

-#endif /// SOUNDTOUCH_ALLOW_SSE

-

-}

-#endif  /// TDStretch_H

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/cpu_detect.h b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/cpu_detect.h
deleted file mode 100644
index 900eb3ce0..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/cpu_detect.h
+++ /dev/null
@@ -1,62 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// A header file for detecting the Intel MMX instructions set extension.

-///

-/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the 

-/// routine implementations for x86 Windows, x86 gnu version and non-x86 

-/// platforms, respectively.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#ifndef _CPU_DETECT_H_

-#define _CPU_DETECT_H_

-

-#include "STTypes.h"

-

-#define SUPPORT_MMX         0x0001

-#define SUPPORT_3DNOW       0x0002

-#define SUPPORT_ALTIVEC     0x0004

-#define SUPPORT_SSE         0x0008

-#define SUPPORT_SSE2        0x0010

-

-/// Checks which instruction set extensions are supported by the CPU.

-///

-/// \return A bitmask of supported extensions, see SUPPORT_... defines.

-uint detectCPUextensions(void);

-

-/// Disables given set of instruction extensions. See SUPPORT_... defines.

-void disableExtensions(uint wDisableMask);

-

-#endif  // _CPU_DETECT_H_

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/cpu_detect_x86_win.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/cpu_detect_x86_win.cpp
deleted file mode 100644
index 98ff02254..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/cpu_detect_x86_win.cpp
+++ /dev/null
@@ -1,129 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// Win32 version of the x86 CPU detect routine.

-///

-/// This file is to be compiled in Windows platform with Microsoft Visual C++ 

-/// Compiler. Please see 'cpu_detect_x86_gcc.cpp' for the gcc compiler version 

-/// for all GNU platforms.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include "cpu_detect.h"

-

-#ifndef WIN32

-#error wrong platform - this source code file is exclusively for Win32 platform

-#endif

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// processor instructions extension detection routines

-//

-//////////////////////////////////////////////////////////////////////////////

-

-// Flag variable indicating whick ISA extensions are disabled (for debugging)

-static uint _dwDisabledISA = 0x00;      // 0xffffffff; //<- use this to disable all extensions

-

-

-// Disables given set of instruction extensions. See SUPPORT_... defines.

-void disableExtensions(uint dwDisableMask)

-{

-    _dwDisabledISA = dwDisableMask;

-}

-

-

-

-/// Checks which instruction set extensions are supported by the CPU.

-uint detectCPUextensions(void)

-{

-    uint res = 0;

-

-    if (_dwDisabledISA == 0xffffffff) return 0;

-

-    _asm 

-    {

-        ; check if 'cpuid' instructions is available by toggling eflags bit 21

-        ;

-        xor     esi, esi            ; clear esi = result register

-

-        pushfd                      ; save eflags to stack

-        mov     eax,dword ptr [esp] ; load eax from stack (with eflags)

-        mov     ecx, eax            ; save the original eflags values to ecx

-        xor     eax, 0x00200000     ; toggle bit 21

-        mov     dword ptr [esp],eax ; store toggled eflags to stack

-        popfd                       ; load eflags from stack

-

-        pushfd                      ; save updated eflags to stack

-        mov     eax,dword ptr [esp] ; load eax from stack

-        popfd                       ; pop stack to restore stack pointer

-

-        xor     edx, edx            ; clear edx for defaulting no mmx

-        cmp     eax, ecx            ; compare to original eflags values

-        jz      end                 ; jumps to 'end' if cpuid not present

-

-        ; cpuid instruction available, test for presence of mmx instructions 

-        mov     eax, 1

-        cpuid

-        test    edx, 0x00800000

-        jz      end                 ; branch if MMX not available

-

-        or      esi, SUPPORT_MMX    ; otherwise add MMX support bit

-

-        test    edx, 0x02000000

-        jz      test3DNow           ; branch if SSE not available

-

-        or      esi, SUPPORT_SSE    ; otherwise add SSE support bit

-

-    test3DNow:

-        ; test for precense of AMD extensions

-        mov     eax, 0x80000000

-        cpuid

-        cmp     eax, 0x80000000

-        jbe     end                ; branch if no AMD extensions detected

-

-        ; test for precense of 3DNow! extension

-        mov     eax, 0x80000001

-        cpuid

-        test    edx, 0x80000000

-        jz      end                 ; branch if 3DNow! not detected

-

-        or      esi, SUPPORT_3DNOW  ; otherwise add 3DNow support bit

-

-    end:

-

-        mov     res, esi

-    }

-

-    return res & ~_dwDisabledISA;

-}

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/mmx_optimized.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/mmx_optimized.cpp
deleted file mode 100644
index 495599c9a..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/mmx_optimized.cpp
+++ /dev/null
@@ -1,320 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// MMX optimized routines. All MMX optimized functions have been gathered into 

-/// this single source code file, regardless to their class or original source 

-/// code file, in order to ease porting the library to other compiler and 

-/// processor platforms.

-///

-/// The MMX-optimizations are programmed using MMX compiler intrinsics that

-/// are supported both by Microsoft Visual C++ and GCC compilers, so this file

-/// should compile with both toolsets.

-///

-/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++ 

-/// 6.0 processor pack" update to support compiler intrinsic syntax. The update

-/// is available for download at Microsoft Developers Network, see here:

-/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include "STTypes.h"

-

-#ifdef SOUNDTOUCH_ALLOW_MMX

-// MMX routines available only with integer sample type

-

-#if !(WIN32 || __i386__ || __x86_64__)

-#error "wrong platform - this source code file is exclusively for x86 platforms"

-#endif

-

-using namespace soundtouch;

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// implementation of MMX optimized functions of class 'TDStretchMMX'

-//

-//////////////////////////////////////////////////////////////////////////////

-

-#include "TDStretch.h"

-#include <mmintrin.h>

-#include <limits.h>

-#include <math.h>

-

-

-// Calculates cross correlation of two buffers

-long TDStretchMMX::calcCrossCorrStereo(const short *pV1, const short *pV2) const

-{

-    const __m64 *pVec1, *pVec2;

-    __m64 shifter;

-    __m64 accu, normaccu;

-    long corr, norm;

-    int i;

-   

-    pVec1 = (__m64*)pV1;

-    pVec2 = (__m64*)pV2;

-

-    shifter = _m_from_int(overlapDividerBits);

-    normaccu = accu = _mm_setzero_si64();

-

-    // Process 4 parallel sets of 2 * stereo samples each during each 

-    // round to improve CPU-level parallellization.

-    for (i = 0; i < overlapLength / 8; i ++)

-    {

-        __m64 temp, temp2;

-

-        // dictionary of instructions:

-        // _m_pmaddwd   : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]

-        // _mm_add_pi32 : 2*32bit add

-        // _m_psrad     : 32bit right-shift

-

-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]),

-                            _mm_madd_pi16(pVec1[1], pVec2[1]));

-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]),

-                             _mm_madd_pi16(pVec1[1], pVec1[1]));

-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));

-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));

-

-        temp = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]),

-                            _mm_madd_pi16(pVec1[3], pVec2[3]));

-        temp2 = _mm_add_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]),

-                             _mm_madd_pi16(pVec1[3], pVec1[3]));

-        accu = _mm_add_pi32(accu, _mm_sra_pi32(temp, shifter));

-        normaccu = _mm_add_pi32(normaccu, _mm_sra_pi32(temp2, shifter));

-

-        pVec1 += 4;

-        pVec2 += 4;

-    }

-

-    // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1

-    // and finally store the result into the variable "corr"

-

-    accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));

-    corr = _m_to_int(accu);

-

-    normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32));

-    norm = _m_to_int(normaccu);

-

-    // Clear MMS state

-    _m_empty();

-

-    // Normalize result by dividing by sqrt(norm) - this step is easiest 

-    // done using floating point operation

-    if (norm == 0) norm = 1;    // to avoid div by zero

-    return (long)((double)corr * USHRT_MAX / sqrt((double)norm));

-    // Note: Warning about the missing EMMS instruction is harmless

-    // as it'll be called elsewhere.

-}

-

-

-

-void TDStretchMMX::clearCrossCorrState()

-{

-    // Clear MMS state

-    _m_empty();

-    //_asm EMMS;

-}

-

-

-

-// MMX-optimized version of the function overlapStereo

-void TDStretchMMX::overlapStereo(short *output, const short *input) const

-{

-    const __m64 *pVinput, *pVMidBuf;

-    __m64 *pVdest;

-    __m64 mix1, mix2, adder, shifter;

-    int i;

-

-    pVinput  = (const __m64*)input;

-    pVMidBuf = (const __m64*)pMidBuffer;

-    pVdest   = (__m64*)output;

-

-    // mix1  = mixer values for 1st stereo sample

-    // mix1  = mixer values for 2nd stereo sample

-    // adder = adder for updating mixer values after each round

-    

-    mix1  = _mm_set_pi16(0, overlapLength,   0, overlapLength);

-    adder = _mm_set_pi16(1, -1, 1, -1);

-    mix2  = _mm_add_pi16(mix1, adder);

-    adder = _mm_add_pi16(adder, adder);

-

-    // Overlaplength-division by shifter. "+1" is to account for "-1" deduced in

-    // overlapDividerBits calculation earlier.

-    shifter = _m_from_int(overlapDividerBits + 1);

-

-    for (i = 0; i < overlapLength / 4; i ++)

-    {

-        __m64 temp1, temp2;

-                

-        // load & shuffle data so that input & mixbuffer data samples are paired

-        temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]);     // = i0l m0l i0r m0r

-        temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]);     // = i1l m1l i1r m1r

-

-        // temp = (temp .* mix) >> shifter

-        temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);

-        temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);

-        pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit

-

-        // update mix += adder

-        mix1 = _mm_add_pi16(mix1, adder);

-        mix2 = _mm_add_pi16(mix2, adder);

-

-        // --- second round begins here ---

-

-        // load & shuffle data so that input & mixbuffer data samples are paired

-        temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]);       // = i2l m2l i2r m2r

-        temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]);       // = i3l m3l i3r m3r

-

-        // temp = (temp .* mix) >> shifter

-        temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);

-        temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);

-        pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit

-

-        // update mix += adder

-        mix1 = _mm_add_pi16(mix1, adder);

-        mix2 = _mm_add_pi16(mix2, adder);

-

-        pVinput  += 2;

-        pVMidBuf += 2;

-        pVdest   += 2;

-    }

-

-    _m_empty(); // clear MMS state

-}

-

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// implementation of MMX optimized functions of class 'FIRFilter'

-//

-//////////////////////////////////////////////////////////////////////////////

-

-#include "FIRFilter.h"

-

-

-FIRFilterMMX::FIRFilterMMX() : FIRFilter()

-{

-    filterCoeffsUnalign = NULL;

-}

-

-

-FIRFilterMMX::~FIRFilterMMX()

-{

-    delete[] filterCoeffsUnalign;

-}

-

-

-// (overloaded) Calculates filter coefficients for MMX routine

-void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)

-{

-    uint i;

-    FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);

-

-    // Ensure that filter coeffs array is aligned to 16-byte boundary

-    delete[] filterCoeffsUnalign;

-    filterCoeffsUnalign = new short[2 * newLength + 8];

-    filterCoeffsAlign = (short *)(((ulong)filterCoeffsUnalign + 15) & -16);

-

-    // rearrange the filter coefficients for mmx routines 

-    for (i = 0;i < length; i += 4) 

-    {

-        filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];

-        filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];

-        filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];

-        filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];

-

-        filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];

-        filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];

-        filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];

-        filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];

-    }

-}

-

-

-

-// mmx-optimized version of the filter routine for stereo sound

-uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const

-{

-    // Create stack copies of the needed member variables for asm routines :

-    uint i, j;

-    __m64 *pVdest = (__m64*)dest;

-

-    if (length < 2) return 0;

-

-    for (i = 0; i < (numSamples - length) / 2; i ++)

-    {

-        __m64 accu1;

-        __m64 accu2;

-        const __m64 *pVsrc = (const __m64*)src;

-        const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;

-

-        accu1 = accu2 = _mm_setzero_si64();

-        for (j = 0; j < lengthDiv8 * 2; j ++)

-        {

-            __m64 temp1, temp2;

-

-            temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]);  // = l2 l0 r2 r0

-            temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]);  // = l3 l1 r3 r1

-

-            accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0]));  // += l2*f2+l0*f0 r2*f2+r0*f0

-            accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1]));  // += l3*f3+l1*f1 r3*f3+r1*f1

-

-            temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]);  // = l4 l2 r4 r2

-

-            accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0]));  // += l3*f2+l1*f0 r3*f2+r1*f0

-            accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1]));  // += l4*f3+l2*f1 r4*f3+r2*f1

-

-            // accu1 += l2*f2+l0*f0 r2*f2+r0*f0

-            //       += l3*f3+l1*f1 r3*f3+r1*f1

-

-            // accu2 += l3*f2+l1*f0 r3*f2+r1*f0

-            //          l4*f3+l2*f1 r4*f3+r2*f1

-

-            pVfilter += 2;

-            pVsrc += 2;

-        }

-        // accu >>= resultDivFactor

-        accu1 = _mm_srai_pi32(accu1, resultDivFactor);

-        accu2 = _mm_srai_pi32(accu2, resultDivFactor);

-

-        // pack 2*2*32bits => 4*16 bits

-        pVdest[0] = _mm_packs_pi32(accu1, accu2);

-        src += 4;

-        pVdest ++;

-    }

-

-   _m_empty();  // clear emms state

-

-    return (numSamples & 0xfffffffe) - length;

-}

-

-#endif  // SOUNDTOUCH_ALLOW_MMX

diff --git a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/sse_optimized.cpp b/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/sse_optimized.cpp
deleted file mode 100644
index a1f318b5f..000000000
--- a/src/filters/renderer/MpcAudioRenderer/SoundTouch/source/sse_optimized.cpp
+++ /dev/null
@@ -1,510 +0,0 @@
-////////////////////////////////////////////////////////////////////////////////

-///

-/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE 

-/// optimized functions have been gathered into this single source 

-/// code file, regardless to their class or original source code file, in order 

-/// to ease porting the library to other compiler and processor platforms.

-///

-/// The SSE-optimizations are programmed using SSE compiler intrinsics that

-/// are supported both by Microsoft Visual C++ and GCC compilers, so this file

-/// should compile with both toolsets.

-///

-/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++ 

-/// 6.0 processor pack" update to support SSE instruction set. The update is 

-/// available for download at Microsoft Developers Network, see here:

-/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx

-///

-/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and 

-/// perform a search with keywords "processor pack".

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date$

-// File revision : $Revision: 4 $

-//

-// $Id$

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library is free software; you can redistribute it and/or

-//  modify it under the terms of the GNU Lesser General Public

-//  License as published by the Free Software Foundation; either

-//  version 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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

-//  Lesser General Public License for more details.

-//

-//  You should have received a copy of the GNU Lesser General Public

-//  License along with this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#include "cpu_detect.h"

-#include "STTypes.h"

-

-using namespace soundtouch;

-

-#ifdef SOUNDTOUCH_ALLOW_SSE

-

-// SSE routines available only with float sample type    

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// implementation of SSE optimized functions of class 'TDStretchSSE'

-//

-//////////////////////////////////////////////////////////////////////////////

-

-#include "TDStretch.h"

-#include <xmmintrin.h>

-#include <math.h>

-

-// Calculates cross correlation of two buffers

-double TDStretchSSE::calcCrossCorrStereo(const float *pV1, const float *pV2) const

-{

-    int i;

-    const float *pVec1;

-    const __m128 *pVec2;

-    __m128 vSum, vNorm;

-

-    // Note. It means a major slow-down if the routine needs to tolerate 

-    // unaligned __m128 memory accesses. It's way faster if we can skip 

-    // unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.

-    // This can mean up to ~ 10-fold difference (incl. part of which is

-    // due to skipping every second round for stereo sound though).

-    //

-    // Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided

-    // for choosing if this little cheating is allowed.

-

-#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION

-    // Little cheating allowed, return valid correlation only for 

-    // aligned locations, meaning every second round for stereo sound.

-

-    #define _MM_LOAD    _mm_load_ps

-

-    if (((ulong)pV1) & 15) return -1e50;    // skip unaligned locations

-

-#else

-    // No cheating allowed, use unaligned load & take the resulting

-    // performance hit.

-    #define _MM_LOAD    _mm_loadu_ps

-#endif 

-

-    // ensure overlapLength is divisible by 8

-    assert((overlapLength % 8) == 0);

-

-    // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors

-    // Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.

-    pVec1 = (const float*)pV1;

-    pVec2 = (const __m128*)pV2;

-    vSum = vNorm = _mm_setzero_ps();

-

-    // Unroll the loop by factor of 4 * 4 operations

-    for (i = 0; i < overlapLength / 8; i ++) 

-    {

-        __m128 vTemp;

-        // vSum += pV1[0..3] * pV2[0..3]

-        vTemp = _MM_LOAD(pVec1);

-        vSum  = _mm_add_ps(vSum,  _mm_mul_ps(vTemp ,pVec2[0]));

-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));

-

-        // vSum += pV1[4..7] * pV2[4..7]

-        vTemp = _MM_LOAD(pVec1 + 4);

-        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));

-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));

-

-        // vSum += pV1[8..11] * pV2[8..11]

-        vTemp = _MM_LOAD(pVec1 + 8);

-        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));

-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));

-

-        // vSum += pV1[12..15] * pV2[12..15]

-        vTemp = _MM_LOAD(pVec1 + 12);

-        vSum  = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));

-        vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));

-

-        pVec1 += 16;

-        pVec2 += 4;

-    }

-

-    // return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]

-    float *pvNorm = (float*)&vNorm;

-    double norm = sqrt(pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);

-    if (norm < 1e-9) norm = 1.0;    // to avoid div by zero

-

-    float *pvSum = (float*)&vSum;

-    return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / norm;

-

-    /* This is approximately corresponding routine in C-language yet without normalization:

-    double corr, norm;

-    uint i;

-

-    // Calculates the cross-correlation value between 'pV1' and 'pV2' vectors

-    corr = norm = 0.0;

-    for (i = 0; i < overlapLength / 8; i ++) 

-    {

-        corr += pV1[0] * pV2[0] +

-                pV1[1] * pV2[1] +

-                pV1[2] * pV2[2] +

-                pV1[3] * pV2[3] +

-                pV1[4] * pV2[4] +

-                pV1[5] * pV2[5] +

-                pV1[6] * pV2[6] +

-                pV1[7] * pV2[7] +

-                pV1[8] * pV2[8] +

-                pV1[9] * pV2[9] +

-                pV1[10] * pV2[10] +

-                pV1[11] * pV2[11] +

-                pV1[12] * pV2[12] +

-                pV1[13] * pV2[13] +

-                pV1[14] * pV2[14] +

-                pV1[15] * pV2[15];

-

-	for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];

-

-        pV1 += 16;

-        pV2 += 16;

-    }

-    return corr / sqrt(norm);

-    */

-

-    /* This is a bit outdated, corresponding routine in assembler. This may be teeny-weeny bit

-       faster than intrinsic version, but more difficult to maintain & get compiled on multiple

-       platforms.

-

-    uint overlapLengthLocal = overlapLength;

-    float corr;

-

-    _asm 

-    {

-        // Very important note: data in 'pV2' _must_ be aligned to 

-        // 16-byte boundary!

-

-        // give prefetch hints to CPU of what data are to be needed soonish

-        // give more aggressive hints on pV1 as that changes while pV2 stays

-        // same between runs

-        prefetcht0 [pV1]

-        prefetcht0 [pV2]

-        prefetcht0 [pV1 + 32]

-

-        mov     eax, dword ptr pV1

-        mov     ebx, dword ptr pV2

-

-        xorps   xmm0, xmm0

-

-        mov     ecx, overlapLengthLocal

-        shr     ecx, 3  // div by eight

-

-    loop1:

-        prefetcht0 [eax + 64]     // give a prefetch hint to CPU what data are to be needed soonish

-        prefetcht0 [ebx + 32]     // give a prefetch hint to CPU what data are to be needed soonish

-        movups  xmm1, [eax]

-        mulps   xmm1, [ebx]

-        addps   xmm0, xmm1

-

-        movups  xmm2, [eax + 16]

-        mulps   xmm2, [ebx + 16]

-        addps   xmm0, xmm2

-

-        prefetcht0 [eax + 96]     // give a prefetch hint to CPU what data are to be needed soonish

-        prefetcht0 [ebx + 64]     // give a prefetch hint to CPU what data are to be needed soonish

-

-        movups  xmm3, [eax + 32]

-        mulps   xmm3, [ebx + 32]

-        addps   xmm0, xmm3

-

-        movups  xmm4, [eax + 48]

-        mulps   xmm4, [ebx + 48]

-        addps   xmm0, xmm4

-

-        add     eax, 64

-        add     ebx, 64

-

-        dec     ecx

-        jnz     loop1

-

-        // add the four floats of xmm0 together and return the result. 

-

-        movhlps xmm1, xmm0          // move 3 & 4 of xmm0 to 1 & 2 of xmm1

-        addps   xmm1, xmm0

-        movaps  xmm2, xmm1

-        shufps  xmm2, xmm2, 0x01    // move 2 of xmm2 as 1 of xmm2

-        addss   xmm2, xmm1

-        movss   corr, xmm2

-    }

-

-    return (double)corr;

-    */

-}

-

-

-//////////////////////////////////////////////////////////////////////////////

-//

-// implementation of SSE optimized functions of class 'FIRFilter'

-//

-//////////////////////////////////////////////////////////////////////////////

-

-#include "FIRFilter.h"

-

-FIRFilterSSE::FIRFilterSSE() : FIRFilter()

-{

-    filterCoeffsAlign = NULL;

-    filterCoeffsUnalign = NULL;

-}

-

-

-FIRFilterSSE::~FIRFilterSSE()

-{

-    delete[] filterCoeffsUnalign;

-    filterCoeffsAlign = NULL;

-    filterCoeffsUnalign = NULL;

-}

-

-

-// (overloaded) Calculates filter coefficients for SSE routine

-void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)

-{

-    uint i;

-    float fDivider;

-

-    FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);

-

-    // Scale the filter coefficients so that it won't be necessary to scale the filtering result

-    // also rearrange coefficients suitably for SSE

-    // Ensure that filter coeffs array is aligned to 16-byte boundary

-    delete[] filterCoeffsUnalign;

-    filterCoeffsUnalign = new float[2 * newLength + 4];

-    filterCoeffsAlign = (float *)(((unsigned long)filterCoeffsUnalign + 15) & (ulong)-16);

-

-    fDivider = (float)resultDivider;

-

-    // rearrange the filter coefficients for mmx routines 

-    for (i = 0; i < newLength; i ++)

-    {

-        filterCoeffsAlign[2 * i + 0] =

-        filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;

-    }

-}

-

-

-

-// SSE-optimized version of the filter routine for stereo sound

-uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const

-{

-    int count = (int)((numSamples - length) & (uint)-2);

-    int j;

-

-    assert(count % 2 == 0);

-

-    if (count < 2) return 0;

-

-    assert(source != NULL);

-    assert(dest != NULL);

-    assert((length % 8) == 0);

-    assert(filterCoeffsAlign != NULL);

-    assert(((ulong)filterCoeffsAlign) % 16 == 0);

-

-    // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'

-    for (j = 0; j < count; j += 2)

-    {

-        const float *pSrc;

-        const __m128 *pFil;

-        __m128 sum1, sum2;

-        uint i;

-

-        pSrc = (const float*)source;              // source audio data

-        pFil = (const __m128*)filterCoeffsAlign;  // filter coefficients. NOTE: Assumes coefficients 

-                                                  // are aligned to 16-byte boundary

-        sum1 = sum2 = _mm_setzero_ps();

-

-        for (i = 0; i < length / 8; i ++) 

-        {

-            // Unroll loop for efficiency & calculate filter for 2*2 stereo samples 

-            // at each pass

-

-            // sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset

-            // sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.

-

-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc)    , pFil[0]));

-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));

-

-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));

-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));

-

-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) ,  pFil[2]));

-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));

-

-            sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));

-            sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));

-

-            pSrc += 16;

-            pFil += 4;

-        }

-

-        // Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need

-        // to sum the two hi- and lo-floats of these registers together.

-

-        // post-shuffle & add the filtered values and store to dest.

-        _mm_storeu_ps(dest, _mm_add_ps(

-                    _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)),   // s2_1 s2_0 s1_3 s1_2

-                    _mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0))    // s2_3 s2_2 s1_1 s1_0

-                    ));

-        source += 4;

-        dest += 4;

-    }

-

-    // Ideas for further improvement:

-    // 1. If it could be guaranteed that 'source' were always aligned to 16-byte 

-    //    boundary, a faster aligned '_mm_load_ps' instruction could be used.

-    // 2. If it could be guaranteed that 'dest' were always aligned to 16-byte 

-    //    boundary, a faster '_mm_store_ps' instruction could be used.

-

-    return (uint)count;

-

-    /* original routine in C-language. please notice the C-version has differently 

-       organized coefficients though.

-    double suml1, suml2;

-    double sumr1, sumr2;

-    uint i, j;

-

-    for (j = 0; j < count; j += 2)

-    {

-        const float *ptr;

-        const float *pFil;

-

-        suml1 = sumr1 = 0.0;

-        suml2 = sumr2 = 0.0;

-        ptr = src;

-        pFil = filterCoeffs;

-        for (i = 0; i < lengthLocal; i ++) 

-        {

-            // unroll loop for efficiency.

-

-            suml1 += ptr[0] * pFil[0] + 

-                     ptr[2] * pFil[2] +

-                     ptr[4] * pFil[4] +

-                     ptr[6] * pFil[6];

-

-            sumr1 += ptr[1] * pFil[1] + 

-                     ptr[3] * pFil[3] +

-                     ptr[5] * pFil[5] +

-                     ptr[7] * pFil[7];

-

-            suml2 += ptr[8] * pFil[0] + 

-                     ptr[10] * pFil[2] +

-                     ptr[12] * pFil[4] +

-                     ptr[14] * pFil[6];

-

-            sumr2 += ptr[9] * pFil[1] + 

-                     ptr[11] * pFil[3] +

-                     ptr[13] * pFil[5] +

-                     ptr[15] * pFil[7];

-

-            ptr += 16;

-            pFil += 8;

-        }

-        dest[0] = (float)suml1;

-        dest[1] = (float)sumr1;

-        dest[2] = (float)suml2;

-        dest[3] = (float)sumr2;

-

-        src += 4;

-        dest += 4;

-    }

-    */

-

-

-    /* Similar routine in assembly, again obsoleted due to maintainability

-    _asm

-    {

-        // Very important note: data in 'src' _must_ be aligned to 

-        // 16-byte boundary!

-        mov     edx, count

-        mov     ebx, dword ptr src

-        mov     eax, dword ptr dest

-        shr     edx, 1

-

-    loop1:

-        // "outer loop" : during each round 2*2 output samples are calculated

-

-        // give prefetch hints to CPU of what data are to be needed soonish

-        prefetcht0 [ebx]

-        prefetcht0 [filterCoeffsLocal]

-

-        mov     esi, ebx

-        mov     edi, filterCoeffsLocal

-        xorps   xmm0, xmm0

-        xorps   xmm1, xmm1

-        mov     ecx, lengthLocal

-

-    loop2:

-        // "inner loop" : during each round eight FIR filter taps are evaluated for 2*2 samples

-        prefetcht0 [esi + 32]     // give a prefetch hint to CPU what data are to be needed soonish

-        prefetcht0 [edi + 32]     // give a prefetch hint to CPU what data are to be needed soonish

-

-        movups  xmm2, [esi]         // possibly unaligned load

-        movups  xmm3, [esi + 8]     // possibly unaligned load

-        mulps   xmm2, [edi]

-        mulps   xmm3, [edi]

-        addps   xmm0, xmm2

-        addps   xmm1, xmm3

-

-        movups  xmm4, [esi + 16]    // possibly unaligned load

-        movups  xmm5, [esi + 24]    // possibly unaligned load

-        mulps   xmm4, [edi + 16]

-        mulps   xmm5, [edi + 16]

-        addps   xmm0, xmm4

-        addps   xmm1, xmm5

-

-        prefetcht0 [esi + 64]     // give a prefetch hint to CPU what data are to be needed soonish

-        prefetcht0 [edi + 64]     // give a prefetch hint to CPU what data are to be needed soonish

-

-        movups  xmm6, [esi + 32]    // possibly unaligned load

-        movups  xmm7, [esi + 40]    // possibly unaligned load

-        mulps   xmm6, [edi + 32]

-        mulps   xmm7, [edi + 32]

-        addps   xmm0, xmm6

-        addps   xmm1, xmm7

-

-        movups  xmm4, [esi + 48]    // possibly unaligned load

-        movups  xmm5, [esi + 56]    // possibly unaligned load

-        mulps   xmm4, [edi + 48]

-        mulps   xmm5, [edi + 48]

-        addps   xmm0, xmm4

-        addps   xmm1, xmm5

-

-        add     esi, 64

-        add     edi, 64

-        dec     ecx

-        jnz     loop2

-

-        // Now xmm0 and xmm1 both have a filtered 2-channel sample each, but we still need

-        // to sum the two hi- and lo-floats of these registers together.

-

-        movhlps xmm2, xmm0          // xmm2 = xmm2_3 xmm2_2 xmm0_3 xmm0_2

-        movlhps xmm2, xmm1          // xmm2 = xmm1_1 xmm1_0 xmm0_3 xmm0_2

-        shufps  xmm0, xmm1, 0xe4    // xmm0 = xmm1_3 xmm1_2 xmm0_1 xmm0_0

-        addps   xmm0, xmm2

-

-        movaps  [eax], xmm0

-        add     ebx, 16

-        add     eax, 16

-

-        dec     edx

-        jnz     loop1

-    }

-    */

-}

-

-#endif  // SOUNDTOUCH_ALLOW_SSE