diff options
author | XhmikosR <xhmikosr@users.sourceforge.net> | 2011-07-14 19:53:11 +0400 |
---|---|---|
committer | XhmikosR <xhmikosr@users.sourceforge.net> | 2011-07-14 19:53:11 +0400 |
commit | 4b616e3b7c79b34ea29d62caa2116ed274713a38 (patch) | |
tree | f9083c4bf19b5df59324e27db306b0e2f7009649 /src/filters/renderer/MpcAudioRenderer | |
parent | 9d042ba4e8c0ee43a173134aa755317b5f3e3629 (diff) |
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
Diffstat (limited to 'src/filters/renderer/MpcAudioRenderer')
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="$(SolutionDir)bin\Filters_x64_Debug\"
- ConfigurationType="2"
- InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\debug.vsprops"
- UseOfMFC="1"
- CharacterSet="1"
- >
- <Tool
- Name="VCPreBuildEventTool"
- />
- <Tool
- Name="VCCustomBuildTool"
- />
- <Tool
- Name="VCXMLDataGeneratorTool"
- />
- <Tool
- Name="VCWebServiceProxyGeneratorTool"
- />
- <Tool
- Name="VCMIDLTool"
- TargetEnvironment="3"
- />
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"
- PreprocessorDefinitions="REGISTER_FILTER;WIN32;_DEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES"
- DebugInformationFormat="3"
- />
- <Tool
- Name="VCManagedResourceCompilerTool"
- />
- <Tool
- Name="VCResourceCompilerTool"
- PreprocessorDefinitions="_UNICODE;UNICODE;_WIN64"
- Culture="1033"
- />
- <Tool
- Name="VCPreLinkEventTool"
- />
- <Tool
- Name="VCLinkerTool"
- AdditionalDependencies="BaseClasses.lib DSUtil.lib dsound.lib Winmm.lib"
- OutputFile="$(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"
- />
- <Tool
- Name="VCFxCopTool"
- />
- <Tool
- Name="VCAppVerifierTool"
- />
- <Tool
- Name="VCPostBuildEventTool"
- />
- </Configuration>
- <Configuration
- Name="Release Filter|Win32"
- OutputDirectory="$(SolutionDir)bin\Filters_x86\"
- ConfigurationType="2"
- InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\release.vsprops"
- UseOfMFC="1"
- CharacterSet="1"
- >
- <Tool
- Name="VCPreBuildEventTool"
- />
- <Tool
- Name="VCCustomBuildTool"
- />
- <Tool
- Name="VCXMLDataGeneratorTool"
- />
- <Tool
- Name="VCWebServiceProxyGeneratorTool"
- />
- <Tool
- Name="VCMIDLTool"
- />
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"
- PreprocessorDefinitions="REGISTER_FILTER;WIN32;NDEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES"
- />
- <Tool
- Name="VCManagedResourceCompilerTool"
- />
- <Tool
- Name="VCResourceCompilerTool"
- PreprocessorDefinitions="_UNICODE;UNICODE;WIN32"
- Culture="1033"
- />
- <Tool
- Name="VCPreLinkEventTool"
- />
- <Tool
- Name="VCLinkerTool"
- AdditionalDependencies="BaseClasses.lib DSUtil.lib dsound.lib Winmm.lib"
- OutputFile="$(OutDir)\$(ProjectName).ax"
- AdditionalLibraryDirectories="$(SolutionDir)bin\lib\Release_$(PlatformName)"
- ModuleDefinitionFile="MpcAudioRenderer.def"
- GenerateDebugInformation="true"
- TargetMachine="1"
- />
- <Tool
- Name="VCALinkTool"
- />
- <Tool
- Name="VCManifestTool"
- />
- <Tool
- Name="VCXDCMakeTool"
- />
- <Tool
- Name="VCBscMakeTool"
- />
- <Tool
- Name="VCFxCopTool"
- />
- <Tool
- Name="VCAppVerifierTool"
- />
- <Tool
- Name="VCPostBuildEventTool"
- />
- </Configuration>
- <Configuration
- Name="Release Filter|x64"
- OutputDirectory="$(SolutionDir)bin\Filters_x64\"
- ConfigurationType="2"
- InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\release.vsprops"
- UseOfMFC="1"
- CharacterSet="1"
- >
- <Tool
- Name="VCPreBuildEventTool"
- />
- <Tool
- Name="VCCustomBuildTool"
- />
- <Tool
- Name="VCXMLDataGeneratorTool"
- />
- <Tool
- Name="VCWebServiceProxyGeneratorTool"
- />
- <Tool
- Name="VCMIDLTool"
- TargetEnvironment="3"
- />
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"
- PreprocessorDefinitions="_WIN64;REGISTER_FILTER;WIN32;NDEBUG;_USRDLL;SOUNDTOUCH_INTEGER_SAMPLES"
- EnableEnhancedInstructionSet="0"
- />
- <Tool
- Name="VCManagedResourceCompilerTool"
- />
- <Tool
- Name="VCResourceCompilerTool"
- PreprocessorDefinitions="_UNICODE;UNICODE;_WIN64"
- Culture="1033"
- />
- <Tool
- Name="VCPreLinkEventTool"
- />
- <Tool
- Name="VCLinkerTool"
- AdditionalDependencies="BaseClasses.lib DSUtil.lib dsound.lib Winmm.lib"
- OutputFile="$(OutDir)\$(ProjectName).ax"
- AdditionalLibraryDirectories="$(SolutionDir)bin\lib\Release_$(PlatformName)"
- ModuleDefinitionFile="MpcAudioRenderer.def"
- GenerateDebugInformation="true"
- TargetMachine="17"
- />
- <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|Win32"
- ConfigurationType="4"
- 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="WIN32;_DEBUG;SOUNDTOUCH_INTEGER_SAMPLES"
- />
- <Tool
- Name="VCManagedResourceCompilerTool"
- />
- <Tool
- Name="VCResourceCompilerTool"
- />
- <Tool
- Name="VCPreLinkEventTool"
- />
- <Tool
- Name="VCLibrarianTool"
- AdditionalDependencies="dsound.lib "
- AdditionalLibraryDirectories=""
- />
- <Tool
- Name="VCALinkTool"
- />
- <Tool
- Name="VCXDCMakeTool"
- />
- <Tool
- Name="VCBscMakeTool"
- />
- <Tool
- Name="VCFxCopTool"
- />
- <Tool
- Name="VCPostBuildEventTool"
- />
- </Configuration>
- <Configuration
- Name="Debug|x64"
- ConfigurationType="4"
- InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\debug.vsprops"
- UseOfMFC="1"
- CharacterSet="1"
- >
- <Tool
- Name="VCPreBuildEventTool"
- />
- <Tool
- Name="VCCustomBuildTool"
- />
- <Tool
- Name="VCXMLDataGeneratorTool"
- />
- <Tool
- Name="VCWebServiceProxyGeneratorTool"
- />
- <Tool
- Name="VCMIDLTool"
- TargetEnvironment="3"
- />
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"
- PreprocessorDefinitions="_WIN64;_DEBUG;SOUNDTOUCH_INTEGER_SAMPLES"
- DebugInformationFormat="3"
- />
- <Tool
- Name="VCManagedResourceCompilerTool"
- />
- <Tool
- Name="VCResourceCompilerTool"
- />
- <Tool
- Name="VCPreLinkEventTool"
- />
- <Tool
- Name="VCLibrarianTool"
- AdditionalDependencies="dsound.lib"
- AdditionalLibraryDirectories=""
- />
- <Tool
- Name="VCALinkTool"
- />
- <Tool
- Name="VCXDCMakeTool"
- />
- <Tool
- Name="VCBscMakeTool"
- />
- <Tool
- Name="VCFxCopTool"
- />
- <Tool
- Name="VCPostBuildEventTool"
- />
- </Configuration>
- <Configuration
- Name="Release|Win32"
- ConfigurationType="4"
- InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\release.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="WIN32;NDEBUG;FLAC__NO_DLL;SOUNDTOUCH_INTEGER_SAMPLES"
- />
- <Tool
- Name="VCManagedResourceCompilerTool"
- />
- <Tool
- Name="VCResourceCompilerTool"
- />
- <Tool
- Name="VCPreLinkEventTool"
- />
- <Tool
- Name="VCLibrarianTool"
- AdditionalDependencies="dsound.lib "
- AdditionalLibraryDirectories=""
- />
- <Tool
- Name="VCALinkTool"
- />
- <Tool
- Name="VCXDCMakeTool"
- />
- <Tool
- Name="VCBscMakeTool"
- />
- <Tool
- Name="VCFxCopTool"
- />
- <Tool
- Name="VCPostBuildEventTool"
- />
- </Configuration>
- <Configuration
- Name="Release|x64"
- ConfigurationType="4"
- InheritedPropertySheets="..\..\..\common.vsprops;..\..\..\release.vsprops"
- UseOfMFC="1"
- CharacterSet="1"
- >
- <Tool
- Name="VCPreBuildEventTool"
- />
- <Tool
- Name="VCCustomBuildTool"
- />
- <Tool
- Name="VCXMLDataGeneratorTool"
- />
- <Tool
- Name="VCWebServiceProxyGeneratorTool"
- />
- <Tool
- Name="VCMIDLTool"
- TargetEnvironment="3"
- />
- <Tool
- Name="VCCLCompilerTool"
- AdditionalIncludeDirectories="..\..\..\..\include;SoundTouch\include;..\..\BaseClasses"
- PreprocessorDefinitions="_WIN64;NDEBUG;FLAC__NO_DLL;SOUNDTOUCH_INTEGER_SAMPLES"
- EnableEnhancedInstructionSet="0"
- />
- <Tool
- Name="VCManagedResourceCompilerTool"
- />
- <Tool
- Name="VCResourceCompilerTool"
- />
- <Tool
- Name="VCPreLinkEventTool"
- />
- <Tool
- Name="VCLibrarianTool"
- AdditionalDependencies="dsound.lib "
- AdditionalLibraryDirectories=""
- />
- <Tool
- Name="VCALinkTool"
- />
- <Tool
- Name="VCXDCMakeTool"
- />
- <Tool
- Name="VCBscMakeTool"
- />
- <Tool
- Name="VCFxCopTool"
- />
- <Tool
- Name="VCPostBuildEventTool"
- />
- </Configuration>
- </Configurations>
- <References>
- </References>
- <Files>
- <Filter
- Name="Source Files"
- Filter="cpp;c;cxx;def;odl;idl;hpj;bat;asm"
- >
- <File
- RelativePath="..\..\FilterApp.cpp"
- >
- </File>
- <File
- RelativePath=".\MpcAudioRenderer.cpp"
- >
- </File>
- <File
- RelativePath=".\MpcAudioRenderer.def"
- >
- </File>
- <File
- RelativePath=".\MpcAudioRendererSettingsWnd.cpp"
- >
- </File>
- <File
- RelativePath=".\stdafx.cpp"
- >
- <FileConfiguration
- Name="Debug Filter|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Debug Filter|x64"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release Filter|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release Filter|x64"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Debug|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Debug|x64"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|Win32"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|x64"
- >
- <Tool
- Name="VCCLCompilerTool"
- UsePrecompiledHeader="1"
- />
- </FileConfiguration>
- </File>
- </Filter>
- <Filter
- Name="Header Files"
- Filter="h;hpp;hxx;hm;inl;inc"
- >
- <File
- RelativePath=".\IMpcAudioRendererFilter.h"
- >
- </File>
- <File
- RelativePath=".\MpcAudioRenderer.h"
- >
- </File>
- <File
- RelativePath=".\MpcAudioRendererSettingsWnd.h"
- >
- </File>
- <File
- RelativePath=".\resource.h"
- >
- </File>
- <File
- RelativePath=".\stdafx.h"
- >
- </File>
- </Filter>
- <Filter
- Name="Resource Files"
- Filter="rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe"
- >
- <File
- RelativePath=".\MpcAudioRenderer.rc"
- >
- <FileConfiguration
- Name="Debug|Win32"
- ExcludedFromBuild="true"
- >
- <Tool
- Name="VCResourceCompilerTool"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Debug|x64"
- ExcludedFromBuild="true"
- >
- <Tool
- Name="VCResourceCompilerTool"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|Win32"
- ExcludedFromBuild="true"
- >
- <Tool
- Name="VCResourceCompilerTool"
- />
- </FileConfiguration>
- <FileConfiguration
- Name="Release|x64"
- ExcludedFromBuild="true"
- >
- <Tool
- Name="VCResourceCompilerTool"
- />
- </FileConfiguration>
- </File>
- </Filter>
- <Filter
- Name="SoundTouch"
- >
- <Filter
- Name="include"
- >
- <File
- RelativePath=".\SoundTouch\include\BPMDetect.h"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\include\FIFOSampleBuffer.h"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\include\FIFOSamplePipe.h"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\include\SoundTouch.h"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\include\STTypes.h"
- >
- </File>
- </Filter>
- <Filter
- Name="source"
- >
- <File
- RelativePath=".\SoundTouch\source\AAFilter.cpp"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\source\AAFilter.h"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\source\BPMDetect.cpp"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\source\cpu_detect.h"
- >
- </File>
- <File
- RelativePath=".\SoundTouch\source\cpu_detect_x86_win.cpp"
- >
- <FileConfiguration
- Name="Debug Filter|x64"
- 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>
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@@ -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>
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<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" />
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- <ClInclude Include="SoundTouch\source\FIRFilter.h" />
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- <ClInclude Include="SoundTouch\source\RateTransposer.h" />
- <ClInclude Include="SoundTouch\source\TDStretch.h" />
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<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>
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- <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>
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- <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
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