Merge branch 'OpenAL'
* OpenAL: Changed SoundTouch to use float samples, allowing SSE to be used. Made the DPL2 decoder disabled by default. Re-added the audio hack used by the Accurate VBeam emulation option. Added a latency setting to the audio settings. Removed the Sample Rate setting. It is now hardcoded to 48000hz (accurate audio timing). Skipped timestretching if the emulator is running below 10% speed to prevent buffer overflows. Removed the synchronisation between the CPU thread and the audio thread. Added code to detect and resume from buffer underruns. Disabled the ability to change the DPL2 option after the game has started. Fixed a memory leak that occurred in the DPL2 decoder. Fixed the OSX build. Build fix Added a Dolby Pro Logic II (DPL2) decoder in the OpenAL backend. DPL2 audio is decoded to 5.1. Code adapted from ffdshow. Added an option in the DSP settings to disable the DPL2 decoder in case Dolphin incorrectly detects a 5.1 audio system. Updated the OpenAL files to OpenAL Soft 1.15.1 in the Windows build. Removed the system timing hack which was activated when the Accurate VBeam option was enabled. Fixed the include directories in Audio Common for the Windows 32bit build. Fixed the include directories in Audio Common for the Windows build. Messed up the static include line Fix include paths and compiling in Linux. Externals soundtouch is 1.7.1, while Ubuntu 12.10 is 1.6.x. Externals soundtouch is compiled with integer samples, while ubuntu is compiled with float samples. Float samples is probably the more common route. If you're going to use soundtouch, you should probably use SAMPLETYPE instead of explicitly choosing short. This probably breaks the windows build since its includes aren't setup. OSX: typedef signed char BOOL OSX build fix Build fix Added audio time stretching by using the SoundTouch library. Implemented correct audio timing. OpenAL for Windows initial commit
This commit is contained in:
skidau
commit
0a4272c96b
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@ -405,6 +405,19 @@ else()
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set(LZO lzo2)
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endif()
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if(OPENAL_FOUND)
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if(NOT ${CMAKE_SYSTEM_NAME} MATCHES "Darwin")
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check_lib(SOUNDTOUCH SoundTouch soundtouch/soundtouch.h QUIET)
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endif()
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if (SOUNDTOUCH_FOUND)
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message("Using shared soundtouch")
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else()
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message("Using static soundtouch from Externals")
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add_subdirectory(Externals/soundtouch)
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include_directories(Externals)
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endif()
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endif()
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if(NOT ${CMAKE_SYSTEM_NAME} MATCHES "Darwin")
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include(FindSDL2 OPTIONAL)
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endif()
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@ -0,0 +1,422 @@
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/*******************************************************************\
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* *
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* EFX-UTIL.H - EFX Utilities functions and Reverb Presets *
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* *
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* File revision 1.0 *
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* *
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\*******************************************************************/
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#ifndef EAXVECTOR_DEFINED
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#define EAXVECTOR_DEFINED
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typedef struct _EAXVECTOR {
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float x;
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float y;
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float z;
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} EAXVECTOR;
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#endif
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#ifndef EAXREVERBPROPERTIES_DEFINED
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#define EAXREVERBPROPERTIES_DEFINED
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typedef struct _EAXREVERBPROPERTIES
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{
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unsigned long ulEnvironment;
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float flEnvironmentSize;
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float flEnvironmentDiffusion;
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long lRoom;
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long lRoomHF;
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long lRoomLF;
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float flDecayTime;
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float flDecayHFRatio;
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float flDecayLFRatio;
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long lReflections;
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float flReflectionsDelay;
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EAXVECTOR vReflectionsPan;
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long lReverb;
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float flReverbDelay;
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EAXVECTOR vReverbPan;
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float flEchoTime;
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float flEchoDepth;
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float flModulationTime;
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float flModulationDepth;
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float flAirAbsorptionHF;
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float flHFReference;
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float flLFReference;
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float flRoomRolloffFactor;
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unsigned long ulFlags;
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} EAXREVERBPROPERTIES, *LPEAXREVERBPROPERTIES;
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#endif
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#ifndef EFXEAXREVERBPROPERTIES_DEFINED
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#define EFXEAXREVERBPROPERTIES_DEFINED
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typedef struct
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{
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float flDensity;
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float flDiffusion;
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float flGain;
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float flGainHF;
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float flGainLF;
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float flDecayTime;
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float flDecayHFRatio;
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float flDecayLFRatio;
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float flReflectionsGain;
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float flReflectionsDelay;
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float flReflectionsPan[3];
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float flLateReverbGain;
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float flLateReverbDelay;
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float flLateReverbPan[3];
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float flEchoTime;
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float flEchoDepth;
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float flModulationTime;
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float flModulationDepth;
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float flAirAbsorptionGainHF;
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float flHFReference;
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float flLFReference;
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float flRoomRolloffFactor;
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int iDecayHFLimit;
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} EFXEAXREVERBPROPERTIES, *LPEFXEAXREVERBPROPERTIES;
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#endif
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#ifndef EAXOBSTRUCTIONPROPERTIES_DEFINED
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#define EAXOBSTRUCTIONPROPERTIES_DEFINED
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typedef struct _EAXOBSTRUCTIONPROPERTIES
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{
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long lObstruction;
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float flObstructionLFRatio;
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} EAXOBSTRUCTIONPROPERTIES, *LPEAXOBSTRUCTIONPROPERTIES;
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#endif
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#ifndef EAXOCCLUSIONPROPERTIES_DEFINED
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#define EAXOCCLUSIONPROPERTIES_DEFINED
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typedef struct _EAXOCCLUSIONPROPERTIES
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{
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long lOcclusion;
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float flOcclusionLFRatio;
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float flOcclusionRoomRatio;
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float flOcclusionDirectRatio;
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} EAXOCCLUSIONPROPERTIES, *LPEAXOCCLUSIONPROPERTIES;
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#endif
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#ifndef EAXEXCLUSIONPROPERTIES_DEFINED
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#define EAXEXCLUSIONPROPERTIES_DEFINED
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typedef struct _EAXEXCLUSIONPROPERTIES
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{
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long lExclusion;
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float flExclusionLFRatio;
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} EAXEXCLUSIONPROPERTIES, *LPEAXEXCLUSIONPROPERTIES;
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#endif
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#ifndef EFXLOWPASSFILTER_DEFINED
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#define EFXLOWPASSFILTER_DEFINED
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typedef struct _EFXLOWPASSFILTER
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{
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float flGain;
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float flGainHF;
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} EFXLOWPASSFILTER, *LPEFXLOWPASSFILTER;
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#endif
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void ConvertReverbParameters(EAXREVERBPROPERTIES *pEAXProp, EFXEAXREVERBPROPERTIES *pEFXEAXReverb);
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void ConvertObstructionParameters(EAXOBSTRUCTIONPROPERTIES *pObProp, EFXLOWPASSFILTER *pDirectLowPassFilter);
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void ConvertExclusionParameters(EAXEXCLUSIONPROPERTIES *pExProp, EFXLOWPASSFILTER *pSendLowPassFilter);
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void ConvertOcclusionParameters(EAXOCCLUSIONPROPERTIES *pOcProp, EFXLOWPASSFILTER *pDirectLowPassFilter, EFXLOWPASSFILTER *pSendLowPassFilter);
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/***********************************************************************************************\
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*
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* EAX Reverb Presets in legacy format - use ConvertReverbParameters() to convert to
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* EFX EAX Reverb Presets for use with the OpenAL Effects Extension.
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*
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************************************************************************************************/
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// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
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#define REVERB_PRESET_GENERIC \
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{0, 7.5f, 1.000f, -1000, -100, 0, 1.49f, 0.83f, 1.00f, -2602, 0.007f, 0.00f,0.00f,0.00f, 200, 0.011f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_PADDEDCELL \
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{1, 1.4f, 1.000f, -1000, -6000, 0, 0.17f, 0.10f, 1.00f, -1204, 0.001f, 0.00f,0.00f,0.00f, 207, 0.002f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_ROOM \
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{2, 1.9f, 1.000f, -1000, -454, 0, 0.40f, 0.83f, 1.00f, -1646, 0.002f, 0.00f,0.00f,0.00f, 53, 0.003f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_BATHROOM \
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{3, 1.4f, 1.000f, -1000, -1200, 0, 1.49f, 0.54f, 1.00f, -370, 0.007f, 0.00f,0.00f,0.00f, 1030, 0.011f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_LIVINGROOM \
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{4, 2.5f, 1.000f, -1000, -6000, 0, 0.50f, 0.10f, 1.00f, -1376, 0.003f, 0.00f,0.00f,0.00f, -1104, 0.004f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_STONEROOM \
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{5, 11.6f, 1.000f, -1000, -300, 0, 2.31f, 0.64f, 1.00f, -711, 0.012f, 0.00f,0.00f,0.00f, 83, 0.017f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_AUDITORIUM \
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{6, 21.6f, 1.000f, -1000, -476, 0, 4.32f, 0.59f, 1.00f, -789, 0.020f, 0.00f,0.00f,0.00f, -289, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_CONCERTHALL \
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{7, 19.6f, 1.000f, -1000, -500, 0, 3.92f, 0.70f, 1.00f, -1230, 0.020f, 0.00f,0.00f,0.00f, -02, 0.029f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_CAVE \
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{8, 14.6f, 1.000f, -1000, 0, 0, 2.91f, 1.30f, 1.00f, -602, 0.015f, 0.00f,0.00f,0.00f, -302, 0.022f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
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#define REVERB_PRESET_ARENA \
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{9, 36.2f, 1.000f, -1000, -698, 0, 7.24f, 0.33f, 1.00f, -1166, 0.020f, 0.00f,0.00f,0.00f, 16, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_HANGAR \
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{10, 50.3f, 1.000f, -1000, -1000, 0, 10.05f, 0.23f, 1.00f, -602, 0.020f, 0.00f,0.00f,0.00f, 198, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_CARPETTEDHALLWAY \
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{11, 1.9f, 1.000f, -1000, -4000, 0, 0.30f, 0.10f, 1.00f, -1831, 0.002f, 0.00f,0.00f,0.00f, -1630, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_HALLWAY \
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{12, 1.8f, 1.000f, -1000, -300, 0, 1.49f, 0.59f, 1.00f, -1219, 0.007f, 0.00f,0.00f,0.00f, 441, 0.011f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_STONECORRIDOR \
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{13, 13.5f, 1.000f, -1000, -237, 0, 2.70f, 0.79f, 1.00f, -1214, 0.013f, 0.00f,0.00f,0.00f, 395, 0.020f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_ALLEY \
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||||
{14, 7.5f, 0.300f, -1000, -270, 0, 1.49f, 0.86f, 1.00f, -1204, 0.007f, 0.00f,0.00f,0.00f, -4, 0.011f, 0.00f,0.00f,0.00f, 0.125f, 0.950f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_FOREST \
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{15, 38.0f, 0.300f, -1000, -3300, 0, 1.49f, 0.54f, 1.00f, -2560, 0.162f, 0.00f,0.00f,0.00f, -229, 0.088f, 0.00f,0.00f,0.00f, 0.125f, 1.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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||||
#define REVERB_PRESET_CITY \
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||||
{16, 7.5f, 0.500f, -1000, -800, 0, 1.49f, 0.67f, 1.00f, -2273, 0.007f, 0.00f,0.00f,0.00f, -1691, 0.011f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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#define REVERB_PRESET_MOUNTAINS \
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||||
{17, 100.0f, 0.270f, -1000, -2500, 0, 1.49f, 0.21f, 1.00f, -2780, 0.300f, 0.00f,0.00f,0.00f, -1434, 0.100f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
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||||
#define REVERB_PRESET_QUARRY \
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||||
{18, 17.5f, 1.000f, -1000, -1000, 0, 1.49f, 0.83f, 1.00f, -10000, 0.061f, 0.00f,0.00f,0.00f, 500, 0.025f, 0.00f,0.00f,0.00f, 0.125f, 0.700f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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||||
#define REVERB_PRESET_PLAIN \
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||||
{19, 42.5f, 0.210f, -1000, -2000, 0, 1.49f, 0.50f, 1.00f, -2466, 0.179f, 0.00f,0.00f,0.00f, -1926, 0.100f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
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||||
#define REVERB_PRESET_PARKINGLOT \
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||||
{20, 8.3f, 1.000f, -1000, 0, 0, 1.65f, 1.50f, 1.00f, -1363, 0.008f, 0.00f,0.00f,0.00f, -1153, 0.012f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
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||||
#define REVERB_PRESET_SEWERPIPE \
|
||||
{21, 1.7f, 0.800f, -1000, -1000, 0, 2.81f, 0.14f, 1.00f, 429, 0.014f, 0.00f,0.00f,0.00f, 1023, 0.021f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_UNDERWATER \
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||||
{22, 1.8f, 1.000f, -1000, -4000, 0, 1.49f, 0.10f, 1.00f, -449, 0.007f, 0.00f,0.00f,0.00f, 1700, 0.011f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 1.180f, 0.348f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_DRUGGED \
|
||||
{23, 1.9f, 0.500f, -1000, 0, 0, 8.39f, 1.39f, 1.00f, -115, 0.002f, 0.00f,0.00f,0.00f, 985, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 1.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
|
||||
#define REVERB_PRESET_DIZZY \
|
||||
{24, 1.8f, 0.600f, -1000, -400, 0, 17.23f, 0.56f, 1.00f, -1713, 0.020f, 0.00f,0.00f,0.00f, -613, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.810f, 0.310f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
|
||||
#define REVERB_PRESET_PSYCHOTIC \
|
||||
{25, 1.0f, 0.500f, -1000, -151, 0, 7.56f, 0.91f, 1.00f, -626, 0.020f, 0.00f,0.00f,0.00f, 774, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 4.000f, 1.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
|
||||
|
||||
|
||||
// CASTLE PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_CASTLE_SMALLROOM \
|
||||
{ 26, 8.3f, 0.890f, -1000, -800, -2000, 1.22f, 0.83f, 0.31f, -100, 0.022f, 0.00f,0.00f,0.00f, 600, 0.011f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CASTLE_SHORTPASSAGE \
|
||||
{ 26, 8.3f, 0.890f, -1000, -1000, -2000, 2.32f, 0.83f, 0.31f, -100, 0.007f, 0.00f,0.00f,0.00f, 200, 0.023f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CASTLE_MEDIUMROOM \
|
||||
{ 26, 8.3f, 0.930f, -1000, -1100, -2000, 2.04f, 0.83f, 0.46f, -400, 0.022f, 0.00f,0.00f,0.00f, 400, 0.011f, 0.00f,0.00f,0.00f, 0.155f, 0.030f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CASTLE_LONGPASSAGE \
|
||||
{ 26, 8.3f, 0.890f, -1000, -800, -2000, 3.42f, 0.83f, 0.31f, -100, 0.007f, 0.00f,0.00f,0.00f, 300, 0.023f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CASTLE_LARGEROOM \
|
||||
{ 26, 8.3f, 0.820f, -1000, -1100, -1800, 2.53f, 0.83f, 0.50f, -700, 0.034f, 0.00f,0.00f,0.00f, 200, 0.016f, 0.00f,0.00f,0.00f, 0.185f, 0.070f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CASTLE_HALL \
|
||||
{ 26, 8.3f, 0.810f, -1000, -1100, -1500, 3.14f, 0.79f, 0.62f, -1500, 0.056f, 0.00f,0.00f,0.00f, 100, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CASTLE_CUPBOARD \
|
||||
{ 26, 8.3f, 0.890f, -1000, -1100, -2000, 0.67f, 0.87f, 0.31f, 300, 0.010f, 0.00f,0.00f,0.00f, 1100, 0.007f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CASTLE_COURTYARD \
|
||||
{ 26, 8.3f, 0.420f, -1000, -700, -1400, 2.13f, 0.61f, 0.23f, -1300, 0.160f, 0.00f,0.00f,0.00f, -300, 0.036f, 0.00f,0.00f,0.00f, 0.250f, 0.370f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
|
||||
#define REVERB_PRESET_CASTLE_ALCOVE \
|
||||
{ 26, 8.3f, 0.890f, -1000, -600, -2000, 1.64f, 0.87f, 0.31f, 00, 0.007f, 0.00f,0.00f,0.00f, 300, 0.034f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 }
|
||||
|
||||
|
||||
// FACTORY PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_FACTORY_ALCOVE \
|
||||
{ 26, 1.8f, 0.590f, -1200, -200, -600, 3.14f, 0.65f, 1.31f, 300, 0.010f, 0.00f,0.00f,0.00f, 000, 0.038f, 0.00f,0.00f,0.00f, 0.114f, 0.100f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_SHORTPASSAGE \
|
||||
{ 26, 1.8f, 0.640f, -1200, -200, -600, 2.53f, 0.65f, 1.31f, 0, 0.010f, 0.00f,0.00f,0.00f, 200, 0.038f, 0.00f,0.00f,0.00f, 0.135f, 0.230f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_MEDIUMROOM \
|
||||
{ 26, 1.9f, 0.820f, -1200, -200, -600, 2.76f, 0.65f, 1.31f, -1100, 0.022f, 0.00f,0.00f,0.00f, 300, 0.023f, 0.00f,0.00f,0.00f, 0.174f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_LONGPASSAGE \
|
||||
{ 26, 1.8f, 0.640f, -1200, -200, -600, 4.06f, 0.65f, 1.31f, 0, 0.020f, 0.00f,0.00f,0.00f, 200, 0.037f, 0.00f,0.00f,0.00f, 0.135f, 0.230f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_LARGEROOM \
|
||||
{ 26, 1.9f, 0.750f, -1200, -300, -400, 4.24f, 0.51f, 1.31f, -1500, 0.039f, 0.00f,0.00f,0.00f, 100, 0.023f, 0.00f,0.00f,0.00f, 0.231f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_HALL \
|
||||
{ 26, 1.9f, 0.750f, -1000, -300, -400, 7.43f, 0.51f, 1.31f, -2400, 0.073f, 0.00f,0.00f,0.00f, -100, 0.027f, 0.00f,0.00f,0.00f, 0.250f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_CUPBOARD \
|
||||
{ 26, 1.7f, 0.630f, -1200, -200, -600, 0.49f, 0.65f, 1.31f, 200, 0.010f, 0.00f,0.00f,0.00f, 600, 0.032f, 0.00f,0.00f,0.00f, 0.107f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_COURTYARD \
|
||||
{ 26, 1.7f, 0.570f, -1000, -1000, -400, 2.32f, 0.29f, 0.56f, -1300, 0.140f, 0.00f,0.00f,0.00f, -800, 0.039f, 0.00f,0.00f,0.00f, 0.250f, 0.290f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_FACTORY_SMALLROOM \
|
||||
{ 26, 1.8f, 0.820f, -1000, -200, -600, 1.72f, 0.65f, 1.31f, -300, 0.010f, 0.00f,0.00f,0.00f, 500, 0.024f, 0.00f,0.00f,0.00f, 0.119f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 }
|
||||
|
||||
|
||||
// ICE PALACE PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_ICEPALACE_ALCOVE \
|
||||
{ 26, 2.7f, 0.840f, -1000, -500, -1100, 2.76f, 1.46f, 0.28f, 100, 0.010f, 0.00f,0.00f,0.00f, -100, 0.030f, 0.00f,0.00f,0.00f, 0.161f, 0.090f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_SHORTPASSAGE \
|
||||
{ 26, 2.7f, 0.750f, -1000, -500, -1100, 1.79f, 1.46f, 0.28f, -600, 0.010f, 0.00f,0.00f,0.00f, 100, 0.019f, 0.00f,0.00f,0.00f, 0.177f, 0.090f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_MEDIUMROOM \
|
||||
{ 26, 2.7f, 0.870f, -1000, -500, -700, 2.22f, 1.53f, 0.32f, -800, 0.039f, 0.00f,0.00f,0.00f, 100, 0.027f, 0.00f,0.00f,0.00f, 0.186f, 0.120f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_LONGPASSAGE \
|
||||
{ 26, 2.7f, 0.770f, -1000, -500, -800, 3.01f, 1.46f, 0.28f, -200, 0.012f, 0.00f,0.00f,0.00f, 200, 0.025f, 0.00f,0.00f,0.00f, 0.186f, 0.040f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_LARGEROOM \
|
||||
{ 26, 2.9f, 0.810f, -1000, -500, -700, 3.14f, 1.53f, 0.32f, -1200, 0.039f, 0.00f,0.00f,0.00f, 000, 0.027f, 0.00f,0.00f,0.00f, 0.214f, 0.110f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_HALL \
|
||||
{ 26, 2.9f, 0.760f, -1000, -700, -500, 5.49f, 1.53f, 0.38f, -1900, 0.054f, 0.00f,0.00f,0.00f, -400, 0.052f, 0.00f,0.00f,0.00f, 0.226f, 0.110f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_CUPBOARD \
|
||||
{ 26, 2.7f, 0.830f, -1000, -600, -1300, 0.76f, 1.53f, 0.26f, 100, 0.012f, 0.00f,0.00f,0.00f, 600, 0.016f, 0.00f,0.00f,0.00f, 0.143f, 0.080f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_COURTYARD \
|
||||
{ 26, 2.9f, 0.590f, -1000, -1100, -1000, 2.04f, 1.20f, 0.38f, -1000, 0.173f, 0.00f,0.00f,0.00f, -1000, 0.043f, 0.00f,0.00f,0.00f, 0.235f, 0.480f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_ICEPALACE_SMALLROOM \
|
||||
{ 26, 2.7f, 0.840f, -1000, -500, -1100, 1.51f, 1.53f, 0.27f, -100, 0.010f, 0.00f,0.00f,0.00f, 300, 0.011f, 0.00f,0.00f,0.00f, 0.164f, 0.140f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 }
|
||||
|
||||
|
||||
// SPACE STATION PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_SPACESTATION_ALCOVE \
|
||||
{ 26, 1.5f, 0.780f, -1000, -300, -100, 1.16f, 0.81f, 0.55f, 300, 0.007f, 0.00f,0.00f,0.00f, 000, 0.018f, 0.00f,0.00f,0.00f, 0.192f, 0.210f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPACESTATION_MEDIUMROOM \
|
||||
{ 26, 1.5f, 0.750f, -1000, -400, -100, 3.01f, 0.50f, 0.55f, -800, 0.034f, 0.00f,0.00f,0.00f, 100, 0.035f, 0.00f,0.00f,0.00f, 0.209f, 0.310f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPACESTATION_SHORTPASSAGE \
|
||||
{ 26, 1.5f, 0.870f, -1000, -400, -100, 3.57f, 0.50f, 0.55f, 0, 0.012f, 0.00f,0.00f,0.00f, 100, 0.016f, 0.00f,0.00f,0.00f, 0.172f, 0.200f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPACESTATION_LONGPASSAGE \
|
||||
{ 26, 1.9f, 0.820f, -1000, -400, -100, 4.62f, 0.62f, 0.55f, 0, 0.012f, 0.00f,0.00f,0.00f, 200, 0.031f, 0.00f,0.00f,0.00f, 0.250f, 0.230f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPACESTATION_LARGEROOM \
|
||||
{ 26, 1.8f, 0.810f, -1000, -400, -100, 3.89f, 0.38f, 0.61f, -1000, 0.056f, 0.00f,0.00f,0.00f, -100, 0.035f, 0.00f,0.00f,0.00f, 0.233f, 0.280f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPACESTATION_HALL \
|
||||
{ 26, 1.9f, 0.870f, -1000, -400, -100, 7.11f, 0.38f, 0.61f, -1500, 0.100f, 0.00f,0.00f,0.00f, -400, 0.047f, 0.00f,0.00f,0.00f, 0.250f, 0.250f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPACESTATION_CUPBOARD \
|
||||
{ 26, 1.4f, 0.560f, -1000, -300, -100, 0.79f, 0.81f, 0.55f, 300, 0.007f, 0.00f,0.00f,0.00f, 500, 0.018f, 0.00f,0.00f,0.00f, 0.181f, 0.310f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPACESTATION_SMALLROOM \
|
||||
{ 26, 1.5f, 0.700f, -1000, -300, -100, 1.72f, 0.82f, 0.55f, -200, 0.007f, 0.00f,0.00f,0.00f, 300, 0.013f, 0.00f,0.00f,0.00f, 0.188f, 0.260f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 }
|
||||
|
||||
|
||||
// WOODEN GALLEON PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_WOODEN_ALCOVE \
|
||||
{ 26, 7.5f, 1.000f, -1000, -1800, -1000, 1.22f, 0.62f, 0.91f, 100, 0.012f, 0.00f,0.00f,0.00f, -300, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_SHORTPASSAGE \
|
||||
{ 26, 7.5f, 1.000f, -1000, -1800, -1000, 1.75f, 0.50f, 0.87f, -100, 0.012f, 0.00f,0.00f,0.00f, -400, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_MEDIUMROOM \
|
||||
{ 26, 7.5f, 1.000f, -1000, -2000, -1100, 1.47f, 0.42f, 0.82f, -100, 0.049f, 0.00f,0.00f,0.00f, -100, 0.029f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_LONGPASSAGE \
|
||||
{ 26, 7.5f, 1.000f, -1000, -2000, -1000, 1.99f, 0.40f, 0.79f, 000, 0.020f, 0.00f,0.00f,0.00f, -700, 0.036f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_LARGEROOM \
|
||||
{ 26, 7.5f, 1.000f, -1000, -2100, -1100, 2.65f, 0.33f, 0.82f, -100, 0.066f, 0.00f,0.00f,0.00f, -200, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_HALL \
|
||||
{ 26, 7.5f, 1.000f, -1000, -2200, -1100, 3.45f, 0.30f, 0.82f, -100, 0.088f, 0.00f,0.00f,0.00f, -200, 0.063f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_CUPBOARD \
|
||||
{ 26, 7.5f, 1.000f, -1000, -1700, -1000, 0.56f, 0.46f, 0.91f, 100, 0.012f, 0.00f,0.00f,0.00f, 100, 0.028f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_SMALLROOM \
|
||||
{ 26, 7.5f, 1.000f, -1000, -1900, -1000, 0.79f, 0.32f, 0.87f, 00, 0.032f, 0.00f,0.00f,0.00f, -100, 0.029f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_WOODEN_COURTYARD \
|
||||
{ 26, 7.5f, 0.650f, -1000, -2200, -1000, 1.79f, 0.35f, 0.79f, -500, 0.123f, 0.00f,0.00f,0.00f, -2000, 0.032f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f }
|
||||
|
||||
|
||||
// SPORTS PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_SPORT_EMPTYSTADIUM \
|
||||
{ 26, 7.2f, 1.000f, -1000, -700, -200, 6.26f, 0.51f, 1.10f, -2400, 0.183f, 0.00f,0.00f,0.00f, -800, 0.038f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPORT_SQUASHCOURT \
|
||||
{ 26, 7.5f, 0.750f, -1000, -1000, -200, 2.22f, 0.91f, 1.16f, -700, 0.007f, 0.00f,0.00f,0.00f, -200, 0.011f, 0.00f,0.00f,0.00f, 0.126f, 0.190f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPORT_SMALLSWIMMINGPOOL \
|
||||
{ 26, 36.2f, 0.700f, -1000, -200, -100, 2.76f, 1.25f, 1.14f, -400, 0.020f, 0.00f,0.00f,0.00f, -200, 0.030f, 0.00f,0.00f,0.00f, 0.179f, 0.150f, 0.895f, 0.190f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_SPORT_LARGESWIMMINGPOOL\
|
||||
{ 26, 36.2f, 0.820f, -1000, -200, 0, 5.49f, 1.31f, 1.14f, -700, 0.039f, 0.00f,0.00f,0.00f, -600, 0.049f, 0.00f,0.00f,0.00f, 0.222f, 0.550f, 1.159f, 0.210f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_SPORT_GYMNASIUM \
|
||||
{ 26, 7.5f, 0.810f, -1000, -700, -100, 3.14f, 1.06f, 1.35f, -800, 0.029f, 0.00f,0.00f,0.00f, -500, 0.045f, 0.00f,0.00f,0.00f, 0.146f, 0.140f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPORT_FULLSTADIUM \
|
||||
{ 26, 7.2f, 1.000f, -1000, -2300, -200, 5.25f, 0.17f, 0.80f, -2000, 0.188f, 0.00f,0.00f,0.00f, -1100, 0.038f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_SPORT_STADIUMTANNOY \
|
||||
{ 26, 3.0f, 0.780f, -1000, -500, -600, 2.53f, 0.88f, 0.68f, -1100, 0.230f, 0.00f,0.00f,0.00f, -600, 0.063f, 0.00f,0.00f,0.00f, 0.250f, 0.200f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
|
||||
|
||||
// PREFAB PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_PREFAB_WORKSHOP \
|
||||
{ 26, 1.9f, 1.000f, -1000, -1700, -800, 0.76f, 1.00f, 1.00f, 0, 0.012f, 0.00f,0.00f,0.00f, 100, 0.012f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_PREFAB_SCHOOLROOM \
|
||||
{ 26, 1.86f, 0.690f, -1000, -400, -600, 0.98f, 0.45f, 0.18f, 300, 0.017f, 0.00f,0.00f,0.00f, 300, 0.015f, 0.00f,0.00f,0.00f, 0.095f, 0.140f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_PREFAB_PRACTISEROOM \
|
||||
{ 26, 1.86f, 0.870f, -1000, -800, -600, 1.12f, 0.56f, 0.18f, 200, 0.010f, 0.00f,0.00f,0.00f, 300, 0.011f, 0.00f,0.00f,0.00f, 0.095f, 0.140f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_PREFAB_OUTHOUSE \
|
||||
{ 26, 80.3f, 0.820f, -1000, -1900, -1600, 1.38f, 0.38f, 0.35f, -100, 0.024f, 0.00f,0.00f,-0.00f, -400, 0.044f, 0.00f,0.00f,0.00f, 0.121f, 0.170f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_PREFAB_CARAVAN \
|
||||
{ 26, 8.3f, 1.000f, -1000, -2100, -1800, 0.43f, 1.50f, 1.00f, 0, 0.012f, 0.00f,0.00f,0.00f, 600, 0.012f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f }
|
||||
// for US developers, a caravan is the same as a trailer =o)
|
||||
|
||||
|
||||
// DOME AND PIPE PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_DOME_TOMB \
|
||||
{ 26, 51.8f, 0.790f, -1000, -900, -1300, 4.18f, 0.21f, 0.10f, -825, 0.030f, 0.00f,0.00f,0.00f, 450, 0.022f, 0.00f,0.00f,0.00f, 0.177f, 0.190f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_PIPE_SMALL \
|
||||
{ 26, 50.3f, 1.000f, -1000, -900, -1300, 5.04f, 0.10f, 0.10f, -600, 0.032f, 0.00f,0.00f,0.00f, 800, 0.015f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_DOME_SAINTPAULS \
|
||||
{ 26, 50.3f, 0.870f, -1000, -900, -1300, 10.48f, 0.19f, 0.10f, -1500, 0.090f, 0.00f,0.00f,0.00f, 200, 0.042f, 0.00f,0.00f,0.00f, 0.250f, 0.120f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_PIPE_LONGTHIN \
|
||||
{ 26, 1.6f, 0.910f, -1000, -700, -1100, 9.21f, 0.18f, 0.10f, -300, 0.010f, 0.00f,0.00f,0.00f, -300, 0.022f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_PIPE_LARGE \
|
||||
{ 26, 50.3f, 1.000f, -1000, -900, -1300, 8.45f, 0.10f, 0.10f, -800, 0.046f, 0.00f,0.00f,0.00f, 400, 0.032f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_PIPE_RESONANT \
|
||||
{ 26, 1.3f, 0.910f, -1000, -700, -1100, 6.81f, 0.18f, 0.10f, -300, 0.010f, 0.00f,0.00f,0.00f, 00, 0.022f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x0 }
|
||||
|
||||
|
||||
// OUTDOORS PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_OUTDOORS_BACKYARD \
|
||||
{ 26, 80.3f, 0.450f, -1000, -1200, -600, 1.12f, 0.34f, 0.46f, -700, 0.069f, 0.00f,0.00f,-0.00f, -300, 0.023f, 0.00f,0.00f,0.00f, 0.218f, 0.340f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_OUTDOORS_ROLLINGPLAINS \
|
||||
{ 26, 80.3f, 0.000f, -1000, -3900, -400, 2.13f, 0.21f, 0.46f, -1500, 0.300f, 0.00f,0.00f,-0.00f, -700, 0.019f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_OUTDOORS_DEEPCANYON \
|
||||
{ 26, 80.3f, 0.740f, -1000, -1500, -400, 3.89f, 0.21f, 0.46f, -1000, 0.223f, 0.00f,0.00f,-0.00f, -900, 0.019f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_OUTDOORS_CREEK \
|
||||
{ 26, 80.3f, 0.350f, -1000, -1500, -600, 2.13f, 0.21f, 0.46f, -800, 0.115f, 0.00f,0.00f,-0.00f, -1400, 0.031f, 0.00f,0.00f,0.00f, 0.218f, 0.340f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_OUTDOORS_VALLEY \
|
||||
{ 26, 80.3f, 0.280f, -1000, -3100, -1600, 2.88f, 0.26f, 0.35f, -1700, 0.263f, 0.00f,0.00f,-0.00f, -800, 0.100f, 0.00f,0.00f,0.00f, 0.250f, 0.340f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 }
|
||||
|
||||
|
||||
// MOOD PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_MOOD_HEAVEN \
|
||||
{ 26, 19.6f, 0.940f, -1000, -200, -700, 5.04f, 1.12f, 0.56f, -1230, 0.020f, 0.00f,0.00f,0.00f, 200, 0.029f, 0.00f,0.00f,0.00f, 0.250f, 0.080f, 2.742f, 0.050f, -2.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_MOOD_HELL \
|
||||
{ 26, 100.0f, 0.570f, -1000, -900, -700, 3.57f, 0.49f, 2.00f, -10000, 0.020f, 0.00f,0.00f,0.00f, 300, 0.030f, 0.00f,0.00f,0.00f, 0.110f, 0.040f, 2.109f, 0.520f, -5.0f, 5000.0f, 139.5f, 0.00f, 0x40 }
|
||||
#define REVERB_PRESET_MOOD_MEMORY \
|
||||
{ 26, 8.0f, 0.850f, -1000, -400, -900, 4.06f, 0.82f, 0.56f, -2800, 0.000f, 0.00f,0.00f,0.00f, 100, 0.000f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.474f, 0.450f, -10.0f, 5000.0f, 250.0f, 0.00f, 0x0 }
|
||||
|
||||
|
||||
// DRIVING SIMULATION PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_DRIVING_COMMENTATOR \
|
||||
{ 26, 3.0f, 0.000f, 1000, -500, -600, 2.42f, 0.88f, 0.68f, -1400, 0.093f, 0.00f,0.00f,0.00f, -1200, 0.017f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -10.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_DRIVING_PITGARAGE \
|
||||
{ 26, 1.9f, 0.590f, -1000, -300, -500, 1.72f, 0.93f, 0.87f, -500, 0.000f, 0.00f,0.00f,0.00f, 200, 0.016f, 0.00f,0.00f,0.00f, 0.250f, 0.110f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_DRIVING_INCAR_RACER \
|
||||
{ 26, 1.1f, 0.800f, -1000, 0, -200, 0.17f, 2.00f, 0.41f, 500, 0.007f, 0.00f,0.00f,0.00f, -300, 0.015f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10268.2f, 251.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_DRIVING_INCAR_SPORTS \
|
||||
{ 26, 1.1f, 0.800f, -1000, -400, 0, 0.17f, 0.75f, 0.41f, 0, 0.010f, 0.00f,0.00f,0.00f, -500, 0.000f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10268.2f, 251.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_DRIVING_INCAR_LUXURY \
|
||||
{ 26, 1.6f, 1.000f, -1000, -2000, -600, 0.13f, 0.41f, 0.46f, -200, 0.010f, 0.00f,0.00f,0.00f, 400, 0.010f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10268.2f, 251.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_DRIVING_FULLGRANDSTAND \
|
||||
{ 26, 8.3f, 1.000f, -1000, -1100, -400, 3.01f, 1.37f, 1.28f, -900, 0.090f, 0.00f,0.00f,0.00f, -1500, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10420.2f, 250.0f, 0.00f, 0x1f }
|
||||
#define REVERB_PRESET_DRIVING_EMPTYGRANDSTAND \
|
||||
{ 26, 8.3f, 1.000f, -1000, 0, -200, 4.62f, 1.75f, 1.40f, -1363, 0.090f, 0.00f,0.00f,0.00f, -1200, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10420.2f, 250.0f, 0.00f, 0x1f }
|
||||
#define REVERB_PRESET_DRIVING_TUNNEL \
|
||||
{ 26, 3.1f, 0.810f, -1000, -800, -100, 3.42f, 0.94f, 1.31f, -300, 0.051f, 0.00f,0.00f,0.00f, -300, 0.047f, 0.00f,0.00f,0.00f, 0.214f, 0.050f, 0.250f, 0.000f, -5.0f, 5000.0f, 155.3f, 0.00f, 0x20 }
|
||||
|
||||
|
||||
// CITY PRESETS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_CITY_STREETS \
|
||||
{ 26, 3.0f, 0.780f, -1000, -300, -100, 1.79f, 1.12f, 0.91f, -1100, 0.046f, 0.00f,0.00f,0.00f, -1400, 0.028f, 0.00f,0.00f,0.00f, 0.250f, 0.200f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CITY_SUBWAY \
|
||||
{ 26, 3.0f, 0.740f, -1000, -300, -100, 3.01f, 1.23f, 0.91f, -300, 0.046f, 0.00f,0.00f,0.00f, 200, 0.028f, 0.00f,0.00f,0.00f, 0.125f, 0.210f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CITY_MUSEUM \
|
||||
{ 26, 80.3f, 0.820f, -1000, -1500, -1500, 3.28f, 1.40f, 0.57f, -1200, 0.039f, 0.00f,0.00f,-0.00f, -100, 0.034f, 0.00f,0.00f,0.00f, 0.130f, 0.170f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_CITY_LIBRARY \
|
||||
{ 26, 80.3f, 0.820f, -1000, -1100, -2100, 2.76f, 0.89f, 0.41f, -900, 0.029f, 0.00f,0.00f,-0.00f, -100, 0.020f, 0.00f,0.00f,0.00f, 0.130f, 0.170f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 }
|
||||
#define REVERB_PRESET_CITY_UNDERPASS \
|
||||
{ 26, 3.0f, 0.820f, -1000, -700, -100, 3.57f, 1.12f, 0.91f, -800, 0.059f, 0.00f,0.00f,0.00f, -100, 0.037f, 0.00f,0.00f,0.00f, 0.250f, 0.140f, 0.250f, 0.000f, -7.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CITY_ABANDONED \
|
||||
{ 26, 3.0f, 0.690f, -1000, -200, -100, 3.28f, 1.17f, 0.91f, -700, 0.044f, 0.00f,0.00f,0.00f, -1100, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.200f, 0.250f, 0.000f, -3.0f, 5000.0f, 250.0f, 0.00f, 0x20 }
|
||||
|
||||
|
||||
// MISC ROOMS
|
||||
|
||||
// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS
|
||||
#define REVERB_PRESET_DUSTYROOM \
|
||||
{ 26, 1.8f, 0.560f, -1000, -200, -300, 1.79f, 0.38f, 0.21f, -600, 0.002f, 0.00f,0.00f,0.00f, 200, 0.006f, 0.00f,0.00f,0.00f, 0.202f, 0.050f, 0.250f, 0.000f, -10.0f, 13046.0f, 163.3f, 0.00f, 0x20 }
|
||||
#define REVERB_PRESET_CHAPEL \
|
||||
{ 26, 19.6f, 0.840f, -1000, -500, 0, 4.62f, 0.64f, 1.23f, -700, 0.032f, 0.00f,0.00f,0.00f, -200, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.110f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f }
|
||||
#define REVERB_PRESET_SMALLWATERROOM \
|
||||
{ 26, 36.2f, 0.700f, -1000, -698, 0, 1.51f, 1.25f, 1.14f, -100, 0.020f, 0.00f,0.00f,0.00f, 300, 0.030f, 0.00f,0.00f,0.00f, 0.179f, 0.150f, 0.895f, 0.190f, -7.0f, 5000.0f, 250.0f, 0.00f, 0x0 }
|
||||
|
|
@ -0,0 +1,656 @@
|
|||
#ifndef AL_AL_H
|
||||
#define AL_AL_H
|
||||
|
||||
#if defined(__cplusplus)
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
#ifndef AL_API
|
||||
#if defined(AL_LIBTYPE_STATIC)
|
||||
#define AL_API
|
||||
#elif defined(_WIN32)
|
||||
#define AL_API __declspec(dllimport)
|
||||
#else
|
||||
#define AL_API extern
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#if defined(_WIN32)
|
||||
#define AL_APIENTRY __cdecl
|
||||
#else
|
||||
#define AL_APIENTRY
|
||||
#endif
|
||||
|
||||
|
||||
/** Deprecated macro. */
|
||||
#define OPENAL
|
||||
#define ALAPI AL_API
|
||||
#define ALAPIENTRY AL_APIENTRY
|
||||
#define AL_INVALID (-1)
|
||||
#define AL_ILLEGAL_ENUM AL_INVALID_ENUM
|
||||
#define AL_ILLEGAL_COMMAND AL_INVALID_OPERATION
|
||||
|
||||
/** Supported AL version. */
|
||||
#define AL_VERSION_1_0
|
||||
#define AL_VERSION_1_1
|
||||
|
||||
/** 8-bit boolean */
|
||||
typedef char ALboolean;
|
||||
|
||||
/** character */
|
||||
typedef char ALchar;
|
||||
|
||||
/** signed 8-bit 2's complement integer */
|
||||
typedef signed char ALbyte;
|
||||
|
||||
/** unsigned 8-bit integer */
|
||||
typedef unsigned char ALubyte;
|
||||
|
||||
/** signed 16-bit 2's complement integer */
|
||||
typedef short ALshort;
|
||||
|
||||
/** unsigned 16-bit integer */
|
||||
typedef unsigned short ALushort;
|
||||
|
||||
/** signed 32-bit 2's complement integer */
|
||||
typedef int ALint;
|
||||
|
||||
/** unsigned 32-bit integer */
|
||||
typedef unsigned int ALuint;
|
||||
|
||||
/** non-negative 32-bit binary integer size */
|
||||
typedef int ALsizei;
|
||||
|
||||
/** enumerated 32-bit value */
|
||||
typedef int ALenum;
|
||||
|
||||
/** 32-bit IEEE754 floating-point */
|
||||
typedef float ALfloat;
|
||||
|
||||
/** 64-bit IEEE754 floating-point */
|
||||
typedef double ALdouble;
|
||||
|
||||
/** void type (for opaque pointers only) */
|
||||
typedef void ALvoid;
|
||||
|
||||
|
||||
/* Enumerant values begin at column 50. No tabs. */
|
||||
|
||||
/** "no distance model" or "no buffer" */
|
||||
#define AL_NONE 0
|
||||
|
||||
/** Boolean False. */
|
||||
#define AL_FALSE 0
|
||||
|
||||
/** Boolean True. */
|
||||
#define AL_TRUE 1
|
||||
|
||||
|
||||
/**
|
||||
* Relative source.
|
||||
* Type: ALboolean
|
||||
* Range: [AL_TRUE, AL_FALSE]
|
||||
* Default: AL_FALSE
|
||||
*
|
||||
* Specifies if the Source has relative coordinates.
|
||||
*/
|
||||
#define AL_SOURCE_RELATIVE 0x202
|
||||
|
||||
|
||||
/**
|
||||
* Inner cone angle, in degrees.
|
||||
* Type: ALint, ALfloat
|
||||
* Range: [0 - 360]
|
||||
* Default: 360
|
||||
*
|
||||
* The angle covered by the inner cone, where the source will not attenuate.
|
||||
*/
|
||||
#define AL_CONE_INNER_ANGLE 0x1001
|
||||
|
||||
/**
|
||||
* Outer cone angle, in degrees.
|
||||
* Range: [0 - 360]
|
||||
* Default: 360
|
||||
*
|
||||
* The angle covered by the outer cone, where the source will be fully
|
||||
* attenuated.
|
||||
*/
|
||||
#define AL_CONE_OUTER_ANGLE 0x1002
|
||||
|
||||
/**
|
||||
* Source pitch.
|
||||
* Type: ALfloat
|
||||
* Range: [0.5 - 2.0]
|
||||
* Default: 1.0
|
||||
*
|
||||
* A multiplier for the frequency (sample rate) of the source's buffer.
|
||||
*/
|
||||
#define AL_PITCH 0x1003
|
||||
|
||||
/**
|
||||
* Source or listener position.
|
||||
* Type: ALfloat[3], ALint[3]
|
||||
* Default: {0, 0, 0}
|
||||
*
|
||||
* The source or listener location in three dimensional space.
|
||||
*
|
||||
* OpenAL, like OpenGL, uses a right handed coordinate system, where in a
|
||||
* frontal default view X (thumb) points right, Y points up (index finger), and
|
||||
* Z points towards the viewer/camera (middle finger).
|
||||
*
|
||||
* To switch from a left handed coordinate system, flip the sign on the Z
|
||||
* coordinate.
|
||||
*/
|
||||
#define AL_POSITION 0x1004
|
||||
|
||||
/**
|
||||
* Source direction.
|
||||
* Type: ALfloat[3], ALint[3]
|
||||
* Default: {0, 0, 0}
|
||||
*
|
||||
* Specifies the current direction in local space.
|
||||
* A zero-length vector specifies an omni-directional source (cone is ignored).
|
||||
*/
|
||||
#define AL_DIRECTION 0x1005
|
||||
|
||||
/**
|
||||
* Source or listener velocity.
|
||||
* Type: ALfloat[3], ALint[3]
|
||||
* Default: {0, 0, 0}
|
||||
*
|
||||
* Specifies the current velocity in local space.
|
||||
*/
|
||||
#define AL_VELOCITY 0x1006
|
||||
|
||||
/**
|
||||
* Source looping.
|
||||
* Type: ALboolean
|
||||
* Range: [AL_TRUE, AL_FALSE]
|
||||
* Default: AL_FALSE
|
||||
*
|
||||
* Specifies whether source is looping.
|
||||
*/
|
||||
#define AL_LOOPING 0x1007
|
||||
|
||||
/**
|
||||
* Source buffer.
|
||||
* Type: ALuint
|
||||
* Range: any valid Buffer.
|
||||
*
|
||||
* Specifies the buffer to provide sound samples.
|
||||
*/
|
||||
#define AL_BUFFER 0x1009
|
||||
|
||||
/**
|
||||
* Source or listener gain.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - ]
|
||||
*
|
||||
* A value of 1.0 means unattenuated. Each division by 2 equals an attenuation
|
||||
* of about -6dB. Each multiplicaton by 2 equals an amplification of about
|
||||
* +6dB.
|
||||
*
|
||||
* A value of 0.0 is meaningless with respect to a logarithmic scale; it is
|
||||
* silent.
|
||||
*/
|
||||
#define AL_GAIN 0x100A
|
||||
|
||||
/**
|
||||
* Minimum source gain.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - 1.0]
|
||||
*
|
||||
* The minimum gain allowed for a source, after distance and cone attenation is
|
||||
* applied (if applicable).
|
||||
*/
|
||||
#define AL_MIN_GAIN 0x100D
|
||||
|
||||
/**
|
||||
* Maximum source gain.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - 1.0]
|
||||
*
|
||||
* The maximum gain allowed for a source, after distance and cone attenation is
|
||||
* applied (if applicable).
|
||||
*/
|
||||
#define AL_MAX_GAIN 0x100E
|
||||
|
||||
/**
|
||||
* Listener orientation.
|
||||
* Type: ALfloat[6]
|
||||
* Default: {0.0, 0.0, -1.0, 0.0, 1.0, 0.0}
|
||||
*
|
||||
* Effectively two three dimensional vectors. The first vector is the front (or
|
||||
* "at") and the second is the top (or "up").
|
||||
*
|
||||
* Both vectors are in local space.
|
||||
*/
|
||||
#define AL_ORIENTATION 0x100F
|
||||
|
||||
/**
|
||||
* Source state (query only).
|
||||
* Type: ALint
|
||||
* Range: [AL_INITIAL, AL_PLAYING, AL_PAUSED, AL_STOPPED]
|
||||
*/
|
||||
#define AL_SOURCE_STATE 0x1010
|
||||
|
||||
/** Source state value. */
|
||||
#define AL_INITIAL 0x1011
|
||||
#define AL_PLAYING 0x1012
|
||||
#define AL_PAUSED 0x1013
|
||||
#define AL_STOPPED 0x1014
|
||||
|
||||
/**
|
||||
* Source Buffer Queue size (query only).
|
||||
* Type: ALint
|
||||
*
|
||||
* The number of buffers queued using alSourceQueueBuffers, minus the buffers
|
||||
* removed with alSourceUnqueueBuffers.
|
||||
*/
|
||||
#define AL_BUFFERS_QUEUED 0x1015
|
||||
|
||||
/**
|
||||
* Source Buffer Queue processed count (query only).
|
||||
* Type: ALint
|
||||
*
|
||||
* The number of queued buffers that have been fully processed, and can be
|
||||
* removed with alSourceUnqueueBuffers.
|
||||
*
|
||||
* Looping sources will never fully process buffers because they will be set to
|
||||
* play again for when the source loops.
|
||||
*/
|
||||
#define AL_BUFFERS_PROCESSED 0x1016
|
||||
|
||||
/**
|
||||
* Source reference distance.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - ]
|
||||
* Default: 1.0
|
||||
*
|
||||
* The distance in units that no attenuation occurs.
|
||||
*
|
||||
* At 0.0, no distance attenuation ever occurs on non-linear attenuation models.
|
||||
*/
|
||||
#define AL_REFERENCE_DISTANCE 0x1020
|
||||
|
||||
/**
|
||||
* Source rolloff factor.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - ]
|
||||
* Default: 1.0
|
||||
*
|
||||
* Multiplier to exaggerate or diminish distance attenuation.
|
||||
*
|
||||
* At 0.0, no distance attenuation ever occurs.
|
||||
*/
|
||||
#define AL_ROLLOFF_FACTOR 0x1021
|
||||
|
||||
/**
|
||||
* Outer cone gain.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - 1.0]
|
||||
* Default: 0.0
|
||||
*
|
||||
* The gain attenuation applied when the listener is outside of the source's
|
||||
* outer cone.
|
||||
*/
|
||||
#define AL_CONE_OUTER_GAIN 0x1022
|
||||
|
||||
/**
|
||||
* Source maximum distance.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - ]
|
||||
* Default: +inf
|
||||
*
|
||||
* The distance above which the source is not attenuated any further with a
|
||||
* clamped distance model, or where attenuation reaches 0.0 gain for linear
|
||||
* distance models with a default rolloff factor.
|
||||
*/
|
||||
#define AL_MAX_DISTANCE 0x1023
|
||||
|
||||
/** Source buffer position, in seconds */
|
||||
#define AL_SEC_OFFSET 0x1024
|
||||
/** Source buffer position, in sample frames */
|
||||
#define AL_SAMPLE_OFFSET 0x1025
|
||||
/** Source buffer position, in bytes */
|
||||
#define AL_BYTE_OFFSET 0x1026
|
||||
|
||||
/**
|
||||
* Source type (query only).
|
||||
* Type: ALint
|
||||
* Range: [AL_STATIC, AL_STREAMING, AL_UNDETERMINED]
|
||||
*
|
||||
* A Source is Static if a Buffer has been attached using AL_BUFFER.
|
||||
*
|
||||
* A Source is Streaming if one or more Buffers have been attached using
|
||||
* alSourceQueueBuffers.
|
||||
*
|
||||
* A Source is Undetermined when it has the NULL buffer attached using
|
||||
* AL_BUFFER.
|
||||
*/
|
||||
#define AL_SOURCE_TYPE 0x1027
|
||||
|
||||
/** Source type value. */
|
||||
#define AL_STATIC 0x1028
|
||||
#define AL_STREAMING 0x1029
|
||||
#define AL_UNDETERMINED 0x1030
|
||||
|
||||
/** Buffer format specifier. */
|
||||
#define AL_FORMAT_MONO8 0x1100
|
||||
#define AL_FORMAT_MONO16 0x1101
|
||||
#define AL_FORMAT_STEREO8 0x1102
|
||||
#define AL_FORMAT_STEREO16 0x1103
|
||||
|
||||
/** Buffer frequency (query only). */
|
||||
#define AL_FREQUENCY 0x2001
|
||||
/** Buffer bits per sample (query only). */
|
||||
#define AL_BITS 0x2002
|
||||
/** Buffer channel count (query only). */
|
||||
#define AL_CHANNELS 0x2003
|
||||
/** Buffer data size (query only). */
|
||||
#define AL_SIZE 0x2004
|
||||
|
||||
/**
|
||||
* Buffer state.
|
||||
*
|
||||
* Not for public use.
|
||||
*/
|
||||
#define AL_UNUSED 0x2010
|
||||
#define AL_PENDING 0x2011
|
||||
#define AL_PROCESSED 0x2012
|
||||
|
||||
|
||||
/** No error. */
|
||||
#define AL_NO_ERROR 0
|
||||
|
||||
/** Invalid name paramater passed to AL call. */
|
||||
#define AL_INVALID_NAME 0xA001
|
||||
|
||||
/** Invalid enum parameter passed to AL call. */
|
||||
#define AL_INVALID_ENUM 0xA002
|
||||
|
||||
/** Invalid value parameter passed to AL call. */
|
||||
#define AL_INVALID_VALUE 0xA003
|
||||
|
||||
/** Illegal AL call. */
|
||||
#define AL_INVALID_OPERATION 0xA004
|
||||
|
||||
/** Not enough memory. */
|
||||
#define AL_OUT_OF_MEMORY 0xA005
|
||||
|
||||
|
||||
/** Context string: Vendor ID. */
|
||||
#define AL_VENDOR 0xB001
|
||||
/** Context string: Version. */
|
||||
#define AL_VERSION 0xB002
|
||||
/** Context string: Renderer ID. */
|
||||
#define AL_RENDERER 0xB003
|
||||
/** Context string: Space-separated extension list. */
|
||||
#define AL_EXTENSIONS 0xB004
|
||||
|
||||
|
||||
/**
|
||||
* Doppler scale.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0 - ]
|
||||
* Default: 1.0
|
||||
*
|
||||
* Scale for source and listener velocities.
|
||||
*/
|
||||
#define AL_DOPPLER_FACTOR 0xC000
|
||||
AL_API void AL_APIENTRY alDopplerFactor(ALfloat value);
|
||||
|
||||
/**
|
||||
* Doppler velocity (deprecated).
|
||||
*
|
||||
* A multiplier applied to the Speed of Sound.
|
||||
*/
|
||||
#define AL_DOPPLER_VELOCITY 0xC001
|
||||
AL_API void AL_APIENTRY alDopplerVelocity(ALfloat value);
|
||||
|
||||
/**
|
||||
* Speed of Sound, in units per second.
|
||||
* Type: ALfloat
|
||||
* Range: [0.0001 - ]
|
||||
* Default: 343.3
|
||||
*
|
||||
* The speed at which sound waves are assumed to travel, when calculating the
|
||||
* doppler effect.
|
||||
*/
|
||||
#define AL_SPEED_OF_SOUND 0xC003
|
||||
AL_API void AL_APIENTRY alSpeedOfSound(ALfloat value);
|
||||
|
||||
/**
|
||||
* Distance attenuation model.
|
||||
* Type: ALint
|
||||
* Range: [AL_NONE, AL_INVERSE_DISTANCE, AL_INVERSE_DISTANCE_CLAMPED,
|
||||
* AL_LINEAR_DISTANCE, AL_LINEAR_DISTANCE_CLAMPED,
|
||||
* AL_EXPONENT_DISTANCE, AL_EXPONENT_DISTANCE_CLAMPED]
|
||||
* Default: AL_INVERSE_DISTANCE_CLAMPED
|
||||
*
|
||||
* The model by which sources attenuate with distance.
|
||||
*
|
||||
* None - No distance attenuation.
|
||||
* Inverse - Doubling the distance halves the source gain.
|
||||
* Linear - Linear gain scaling between the reference and max distances.
|
||||
* Exponent - Exponential gain dropoff.
|
||||
*
|
||||
* Clamped variations work like the non-clamped counterparts, except the
|
||||
* distance calculated is clamped between the reference and max distances.
|
||||
*/
|
||||
#define AL_DISTANCE_MODEL 0xD000
|
||||
AL_API void AL_APIENTRY alDistanceModel(ALenum distanceModel);
|
||||
|
||||
/** Distance model value. */
|
||||
#define AL_INVERSE_DISTANCE 0xD001
|
||||
#define AL_INVERSE_DISTANCE_CLAMPED 0xD002
|
||||
#define AL_LINEAR_DISTANCE 0xD003
|
||||
#define AL_LINEAR_DISTANCE_CLAMPED 0xD004
|
||||
#define AL_EXPONENT_DISTANCE 0xD005
|
||||
#define AL_EXPONENT_DISTANCE_CLAMPED 0xD006
|
||||
|
||||
/** Renderer State management. */
|
||||
AL_API void AL_APIENTRY alEnable(ALenum capability);
|
||||
AL_API void AL_APIENTRY alDisable(ALenum capability);
|
||||
AL_API ALboolean AL_APIENTRY alIsEnabled(ALenum capability);
|
||||
|
||||
/** State retrieval. */
|
||||
AL_API const ALchar* AL_APIENTRY alGetString(ALenum param);
|
||||
AL_API void AL_APIENTRY alGetBooleanv(ALenum param, ALboolean *values);
|
||||
AL_API void AL_APIENTRY alGetIntegerv(ALenum param, ALint *values);
|
||||
AL_API void AL_APIENTRY alGetFloatv(ALenum param, ALfloat *values);
|
||||
AL_API void AL_APIENTRY alGetDoublev(ALenum param, ALdouble *values);
|
||||
AL_API ALboolean AL_APIENTRY alGetBoolean(ALenum param);
|
||||
AL_API ALint AL_APIENTRY alGetInteger(ALenum param);
|
||||
AL_API ALfloat AL_APIENTRY alGetFloat(ALenum param);
|
||||
AL_API ALdouble AL_APIENTRY alGetDouble(ALenum param);
|
||||
|
||||
/**
|
||||
* Error retrieval.
|
||||
*
|
||||
* Obtain the first error generated in the AL context since the last check.
|
||||
*/
|
||||
AL_API ALenum AL_APIENTRY alGetError(void);
|
||||
|
||||
/**
|
||||
* Extension support.
|
||||
*
|
||||
* Query for the presence of an extension, and obtain any appropriate function
|
||||
* pointers and enum values.
|
||||
*/
|
||||
AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extname);
|
||||
AL_API void* AL_APIENTRY alGetProcAddress(const ALchar *fname);
|
||||
AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *ename);
|
||||
|
||||
|
||||
/** Set Listener parameters */
|
||||
AL_API void AL_APIENTRY alListenerf(ALenum param, ALfloat value);
|
||||
AL_API void AL_APIENTRY alListener3f(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3);
|
||||
AL_API void AL_APIENTRY alListenerfv(ALenum param, const ALfloat *values);
|
||||
AL_API void AL_APIENTRY alListeneri(ALenum param, ALint value);
|
||||
AL_API void AL_APIENTRY alListener3i(ALenum param, ALint value1, ALint value2, ALint value3);
|
||||
AL_API void AL_APIENTRY alListeneriv(ALenum param, const ALint *values);
|
||||
|
||||
/** Get Listener parameters */
|
||||
AL_API void AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value);
|
||||
AL_API void AL_APIENTRY alGetListener3f(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3);
|
||||
AL_API void AL_APIENTRY alGetListenerfv(ALenum param, ALfloat *values);
|
||||
AL_API void AL_APIENTRY alGetListeneri(ALenum param, ALint *value);
|
||||
AL_API void AL_APIENTRY alGetListener3i(ALenum param, ALint *value1, ALint *value2, ALint *value3);
|
||||
AL_API void AL_APIENTRY alGetListeneriv(ALenum param, ALint *values);
|
||||
|
||||
|
||||
/** Create Source objects. */
|
||||
AL_API void AL_APIENTRY alGenSources(ALsizei n, ALuint *sources);
|
||||
/** Delete Source objects. */
|
||||
AL_API void AL_APIENTRY alDeleteSources(ALsizei n, const ALuint *sources);
|
||||
/** Verify a handle is a valid Source. */
|
||||
AL_API ALboolean AL_APIENTRY alIsSource(ALuint source);
|
||||
|
||||
/** Set Source parameters. */
|
||||
AL_API void AL_APIENTRY alSourcef(ALuint source, ALenum param, ALfloat value);
|
||||
AL_API void AL_APIENTRY alSource3f(ALuint source, ALenum param, ALfloat value1, ALfloat value2, ALfloat value3);
|
||||
AL_API void AL_APIENTRY alSourcefv(ALuint source, ALenum param, const ALfloat *values);
|
||||
AL_API void AL_APIENTRY alSourcei(ALuint source, ALenum param, ALint value);
|
||||
AL_API void AL_APIENTRY alSource3i(ALuint source, ALenum param, ALint value1, ALint value2, ALint value3);
|
||||
AL_API void AL_APIENTRY alSourceiv(ALuint source, ALenum param, const ALint *values);
|
||||
|
||||
/** Get Source parameters. */
|
||||
AL_API void AL_APIENTRY alGetSourcef(ALuint source, ALenum param, ALfloat *value);
|
||||
AL_API void AL_APIENTRY alGetSource3f(ALuint source, ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3);
|
||||
AL_API void AL_APIENTRY alGetSourcefv(ALuint source, ALenum param, ALfloat *values);
|
||||
AL_API void AL_APIENTRY alGetSourcei(ALuint source, ALenum param, ALint *value);
|
||||
AL_API void AL_APIENTRY alGetSource3i(ALuint source, ALenum param, ALint *value1, ALint *value2, ALint *value3);
|
||||
AL_API void AL_APIENTRY alGetSourceiv(ALuint source, ALenum param, ALint *values);
|
||||
|
||||
|
||||
/** Play, replay, or resume (if paused) a list of Sources */
|
||||
AL_API void AL_APIENTRY alSourcePlayv(ALsizei n, const ALuint *sources);
|
||||
/** Stop a list of Sources */
|
||||
AL_API void AL_APIENTRY alSourceStopv(ALsizei n, const ALuint *sources);
|
||||
/** Rewind a list of Sources */
|
||||
AL_API void AL_APIENTRY alSourceRewindv(ALsizei n, const ALuint *sources);
|
||||
/** Pause a list of Sources */
|
||||
AL_API void AL_APIENTRY alSourcePausev(ALsizei n, const ALuint *sources);
|
||||
|
||||
/** Play, replay, or resume a Source */
|
||||
AL_API void AL_APIENTRY alSourcePlay(ALuint source);
|
||||
/** Stop a Source */
|
||||
AL_API void AL_APIENTRY alSourceStop(ALuint source);
|
||||
/** Rewind a Source (set playback postiton to beginning) */
|
||||
AL_API void AL_APIENTRY alSourceRewind(ALuint source);
|
||||
/** Pause a Source */
|
||||
AL_API void AL_APIENTRY alSourcePause(ALuint source);
|
||||
|
||||
/** Queue buffers onto a source */
|
||||
AL_API void AL_APIENTRY alSourceQueueBuffers(ALuint source, ALsizei nb, const ALuint *buffers);
|
||||
/** Unqueue processed buffers from a source */
|
||||
AL_API void AL_APIENTRY alSourceUnqueueBuffers(ALuint source, ALsizei nb, ALuint *buffers);
|
||||
|
||||
|
||||
/** Create Buffer objects */
|
||||
AL_API void AL_APIENTRY alGenBuffers(ALsizei n, ALuint *buffers);
|
||||
/** Delete Buffer objects */
|
||||
AL_API void AL_APIENTRY alDeleteBuffers(ALsizei n, const ALuint *buffers);
|
||||
/** Verify a handle is a valid Buffer */
|
||||
AL_API ALboolean AL_APIENTRY alIsBuffer(ALuint buffer);
|
||||
|
||||
/** Specifies the data to be copied into a buffer */
|
||||
AL_API void AL_APIENTRY alBufferData(ALuint buffer, ALenum format, const ALvoid *data, ALsizei size, ALsizei freq);
|
||||
|
||||
/** Set Buffer parameters, */
|
||||
AL_API void AL_APIENTRY alBufferf(ALuint buffer, ALenum param, ALfloat value);
|
||||
AL_API void AL_APIENTRY alBuffer3f(ALuint buffer, ALenum param, ALfloat value1, ALfloat value2, ALfloat value3);
|
||||
AL_API void AL_APIENTRY alBufferfv(ALuint buffer, ALenum param, const ALfloat *values);
|
||||
AL_API void AL_APIENTRY alBufferi(ALuint buffer, ALenum param, ALint value);
|
||||
AL_API void AL_APIENTRY alBuffer3i(ALuint buffer, ALenum param, ALint value1, ALint value2, ALint value3);
|
||||
AL_API void AL_APIENTRY alBufferiv(ALuint buffer, ALenum param, const ALint *values);
|
||||
|
||||
/** Get Buffer parameters. */
|
||||
AL_API void AL_APIENTRY alGetBufferf(ALuint buffer, ALenum param, ALfloat *value);
|
||||
AL_API void AL_APIENTRY alGetBuffer3f(ALuint buffer, ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3);
|
||||
AL_API void AL_APIENTRY alGetBufferfv(ALuint buffer, ALenum param, ALfloat *values);
|
||||
AL_API void AL_APIENTRY alGetBufferi(ALuint buffer, ALenum param, ALint *value);
|
||||
AL_API void AL_APIENTRY alGetBuffer3i(ALuint buffer, ALenum param, ALint *value1, ALint *value2, ALint *value3);
|
||||
AL_API void AL_APIENTRY alGetBufferiv(ALuint buffer, ALenum param, ALint *values);
|
||||
|
||||
/** Pointer-to-function type, useful for dynamically getting AL entry points. */
|
||||
typedef void (AL_APIENTRY *LPALENABLE)(ALenum capability);
|
||||
typedef void (AL_APIENTRY *LPALDISABLE)(ALenum capability);
|
||||
typedef ALboolean (AL_APIENTRY *LPALISENABLED)(ALenum capability);
|
||||
typedef const ALchar* (AL_APIENTRY *LPALGETSTRING)(ALenum param);
|
||||
typedef void (AL_APIENTRY *LPALGETBOOLEANV)(ALenum param, ALboolean *values);
|
||||
typedef void (AL_APIENTRY *LPALGETINTEGERV)(ALenum param, ALint *values);
|
||||
typedef void (AL_APIENTRY *LPALGETFLOATV)(ALenum param, ALfloat *values);
|
||||
typedef void (AL_APIENTRY *LPALGETDOUBLEV)(ALenum param, ALdouble *values);
|
||||
typedef ALboolean (AL_APIENTRY *LPALGETBOOLEAN)(ALenum param);
|
||||
typedef ALint (AL_APIENTRY *LPALGETINTEGER)(ALenum param);
|
||||
typedef ALfloat (AL_APIENTRY *LPALGETFLOAT)(ALenum param);
|
||||
typedef ALdouble (AL_APIENTRY *LPALGETDOUBLE)(ALenum param);
|
||||
typedef ALenum (AL_APIENTRY *LPALGETERROR)(void);
|
||||
typedef ALboolean (AL_APIENTRY *LPALISEXTENSIONPRESENT)(const ALchar *extname);
|
||||
typedef void* (AL_APIENTRY *LPALGETPROCADDRESS)(const ALchar *fname);
|
||||
typedef ALenum (AL_APIENTRY *LPALGETENUMVALUE)(const ALchar *ename);
|
||||
typedef void (AL_APIENTRY *LPALLISTENERF)(ALenum param, ALfloat value);
|
||||
typedef void (AL_APIENTRY *LPALLISTENER3F)(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3);
|
||||
typedef void (AL_APIENTRY *LPALLISTENERFV)(ALenum param, const ALfloat *values);
|
||||
typedef void (AL_APIENTRY *LPALLISTENERI)(ALenum param, ALint value);
|
||||
typedef void (AL_APIENTRY *LPALLISTENER3I)(ALenum param, ALint value1, ALint value2, ALint value3);
|
||||
typedef void (AL_APIENTRY *LPALLISTENERIV)(ALenum param, const ALint *values);
|
||||
typedef void (AL_APIENTRY *LPALGETLISTENERF)(ALenum param, ALfloat *value);
|
||||
typedef void (AL_APIENTRY *LPALGETLISTENER3F)(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3);
|
||||
typedef void (AL_APIENTRY *LPALGETLISTENERFV)(ALenum param, ALfloat *values);
|
||||
typedef void (AL_APIENTRY *LPALGETLISTENERI)(ALenum param, ALint *value);
|
||||
typedef void (AL_APIENTRY *LPALGETLISTENER3I)(ALenum param, ALint *value1, ALint *value2, ALint *value3);
|
||||
typedef void (AL_APIENTRY *LPALGETLISTENERIV)(ALenum param, ALint *values);
|
||||
typedef void (AL_APIENTRY *LPALGENSOURCES)(ALsizei n, ALuint *sources);
|
||||
typedef void (AL_APIENTRY *LPALDELETESOURCES)(ALsizei n, const ALuint *sources);
|
||||
typedef ALboolean (AL_APIENTRY *LPALISSOURCE)(ALuint source);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEF)(ALuint source, ALenum param, ALfloat value);
|
||||
typedef void (AL_APIENTRY *LPALSOURCE3F)(ALuint source, ALenum param, ALfloat value1, ALfloat value2, ALfloat value3);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEFV)(ALuint source, ALenum param, const ALfloat *values);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEI)(ALuint source, ALenum param, ALint value);
|
||||
typedef void (AL_APIENTRY *LPALSOURCE3I)(ALuint source, ALenum param, ALint value1, ALint value2, ALint value3);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEIV)(ALuint source, ALenum param, const ALint *values);
|
||||
typedef void (AL_APIENTRY *LPALGETSOURCEF)(ALuint source, ALenum param, ALfloat *value);
|
||||
typedef void (AL_APIENTRY *LPALGETSOURCE3F)(ALuint source, ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3);
|
||||
typedef void (AL_APIENTRY *LPALGETSOURCEFV)(ALuint source, ALenum param, ALfloat *values);
|
||||
typedef void (AL_APIENTRY *LPALGETSOURCEI)(ALuint source, ALenum param, ALint *value);
|
||||
typedef void (AL_APIENTRY *LPALGETSOURCE3I)(ALuint source, ALenum param, ALint *value1, ALint *value2, ALint *value3);
|
||||
typedef void (AL_APIENTRY *LPALGETSOURCEIV)(ALuint source, ALenum param, ALint *values);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEPLAYV)(ALsizei n, const ALuint *sources);
|
||||
typedef void (AL_APIENTRY *LPALSOURCESTOPV)(ALsizei n, const ALuint *sources);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEREWINDV)(ALsizei n, const ALuint *sources);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEPAUSEV)(ALsizei n, const ALuint *sources);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEPLAY)(ALuint source);
|
||||
typedef void (AL_APIENTRY *LPALSOURCESTOP)(ALuint source);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEREWIND)(ALuint source);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEPAUSE)(ALuint source);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEQUEUEBUFFERS)(ALuint source, ALsizei nb, const ALuint *buffers);
|
||||
typedef void (AL_APIENTRY *LPALSOURCEUNQUEUEBUFFERS)(ALuint source, ALsizei nb, ALuint *buffers);
|
||||
typedef void (AL_APIENTRY *LPALGENBUFFERS)(ALsizei n, ALuint *buffers);
|
||||
typedef void (AL_APIENTRY *LPALDELETEBUFFERS)(ALsizei n, const ALuint *buffers);
|
||||
typedef ALboolean (AL_APIENTRY *LPALISBUFFER)(ALuint buffer);
|
||||
typedef void (AL_APIENTRY *LPALBUFFERDATA)(ALuint buffer, ALenum format, const ALvoid *data, ALsizei size, ALsizei freq);
|
||||
typedef void (AL_APIENTRY *LPALBUFFERF)(ALuint buffer, ALenum param, ALfloat value);
|
||||
typedef void (AL_APIENTRY *LPALBUFFER3F)(ALuint buffer, ALenum param, ALfloat value1, ALfloat value2, ALfloat value3);
|
||||
typedef void (AL_APIENTRY *LPALBUFFERFV)(ALuint buffer, ALenum param, const ALfloat *values);
|
||||
typedef void (AL_APIENTRY *LPALBUFFERI)(ALuint buffer, ALenum param, ALint value);
|
||||
typedef void (AL_APIENTRY *LPALBUFFER3I)(ALuint buffer, ALenum param, ALint value1, ALint value2, ALint value3);
|
||||
typedef void (AL_APIENTRY *LPALBUFFERIV)(ALuint buffer, ALenum param, const ALint *values);
|
||||
typedef void (AL_APIENTRY *LPALGETBUFFERF)(ALuint buffer, ALenum param, ALfloat *value);
|
||||
typedef void (AL_APIENTRY *LPALGETBUFFER3F)(ALuint buffer, ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3);
|
||||
typedef void (AL_APIENTRY *LPALGETBUFFERFV)(ALuint buffer, ALenum param, ALfloat *values);
|
||||
typedef void (AL_APIENTRY *LPALGETBUFFERI)(ALuint buffer, ALenum param, ALint *value);
|
||||
typedef void (AL_APIENTRY *LPALGETBUFFER3I)(ALuint buffer, ALenum param, ALint *value1, ALint *value2, ALint *value3);
|
||||
typedef void (AL_APIENTRY *LPALGETBUFFERIV)(ALuint buffer, ALenum param, ALint *values);
|
||||
typedef void (AL_APIENTRY *LPALDOPPLERFACTOR)(ALfloat value);
|
||||
typedef void (AL_APIENTRY *LPALDOPPLERVELOCITY)(ALfloat value);
|
||||
typedef void (AL_APIENTRY *LPALSPEEDOFSOUND)(ALfloat value);
|
||||
typedef void (AL_APIENTRY *LPALDISTANCEMODEL)(ALenum distanceModel);
|
||||
|
||||
#if defined(__cplusplus)
|
||||
} /* extern "C" */
|
||||
#endif
|
||||
|
||||
#endif /* AL_AL_H */
|
||||
|
|
@ -0,0 +1,237 @@
|
|||
#ifndef AL_ALC_H
|
||||
#define AL_ALC_H
|
||||
|
||||
#if defined(__cplusplus)
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
#ifndef ALC_API
|
||||
#if defined(AL_LIBTYPE_STATIC)
|
||||
#define ALC_API
|
||||
#elif defined(_WIN32)
|
||||
#define ALC_API __declspec(dllimport)
|
||||
#else
|
||||
#define ALC_API extern
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#if defined(_WIN32)
|
||||
#define ALC_APIENTRY __cdecl
|
||||
#else
|
||||
#define ALC_APIENTRY
|
||||
#endif
|
||||
|
||||
|
||||
/** Deprecated macro. */
|
||||
#define ALCAPI ALC_API
|
||||
#define ALCAPIENTRY ALC_APIENTRY
|
||||
#define ALC_INVALID 0
|
||||
|
||||
/** Supported ALC version? */
|
||||
#define ALC_VERSION_0_1 1
|
||||
|
||||
/** Opaque device handle */
|
||||
typedef struct ALCdevice_struct ALCdevice;
|
||||
/** Opaque context handle */
|
||||
typedef struct ALCcontext_struct ALCcontext;
|
||||
|
||||
/** 8-bit boolean */
|
||||
typedef char ALCboolean;
|
||||
|
||||
/** character */
|
||||
typedef char ALCchar;
|
||||
|
||||
/** signed 8-bit 2's complement integer */
|
||||
typedef signed char ALCbyte;
|
||||
|
||||
/** unsigned 8-bit integer */
|
||||
typedef unsigned char ALCubyte;
|
||||
|
||||
/** signed 16-bit 2's complement integer */
|
||||
typedef short ALCshort;
|
||||
|
||||
/** unsigned 16-bit integer */
|
||||
typedef unsigned short ALCushort;
|
||||
|
||||
/** signed 32-bit 2's complement integer */
|
||||
typedef int ALCint;
|
||||
|
||||
/** unsigned 32-bit integer */
|
||||
typedef unsigned int ALCuint;
|
||||
|
||||
/** non-negative 32-bit binary integer size */
|
||||
typedef int ALCsizei;
|
||||
|
||||
/** enumerated 32-bit value */
|
||||
typedef int ALCenum;
|
||||
|
||||
/** 32-bit IEEE754 floating-point */
|
||||
typedef float ALCfloat;
|
||||
|
||||
/** 64-bit IEEE754 floating-point */
|
||||
typedef double ALCdouble;
|
||||
|
||||
/** void type (for opaque pointers only) */
|
||||
typedef void ALCvoid;
|
||||
|
||||
|
||||
/* Enumerant values begin at column 50. No tabs. */
|
||||
|
||||
/** Boolean False. */
|
||||
#define ALC_FALSE 0
|
||||
|
||||
/** Boolean True. */
|
||||
#define ALC_TRUE 1
|
||||
|
||||
/** Context attribute: <int> Hz. */
|
||||
#define ALC_FREQUENCY 0x1007
|
||||
|
||||
/** Context attribute: <int> Hz. */
|
||||
#define ALC_REFRESH 0x1008
|
||||
|
||||
/** Context attribute: AL_TRUE or AL_FALSE. */
|
||||
#define ALC_SYNC 0x1009
|
||||
|
||||
/** Context attribute: <int> requested Mono (3D) Sources. */
|
||||
#define ALC_MONO_SOURCES 0x1010
|
||||
|
||||
/** Context attribute: <int> requested Stereo Sources. */
|
||||
#define ALC_STEREO_SOURCES 0x1011
|
||||
|
||||
/** No error. */
|
||||
#define ALC_NO_ERROR 0
|
||||
|
||||
/** Invalid device handle. */
|
||||
#define ALC_INVALID_DEVICE 0xA001
|
||||
|
||||
/** Invalid context handle. */
|
||||
#define ALC_INVALID_CONTEXT 0xA002
|
||||
|
||||
/** Invalid enum parameter passed to an ALC call. */
|
||||
#define ALC_INVALID_ENUM 0xA003
|
||||
|
||||
/** Invalid value parameter passed to an ALC call. */
|
||||
#define ALC_INVALID_VALUE 0xA004
|
||||
|
||||
/** Out of memory. */
|
||||
#define ALC_OUT_OF_MEMORY 0xA005
|
||||
|
||||
|
||||
/** Runtime ALC version. */
|
||||
#define ALC_MAJOR_VERSION 0x1000
|
||||
#define ALC_MINOR_VERSION 0x1001
|
||||
|
||||
/** Context attribute list properties. */
|
||||
#define ALC_ATTRIBUTES_SIZE 0x1002
|
||||
#define ALC_ALL_ATTRIBUTES 0x1003
|
||||
|
||||
/** String for the default device specifier. */
|
||||
#define ALC_DEFAULT_DEVICE_SPECIFIER 0x1004
|
||||
/**
|
||||
* String for the given device's specifier.
|
||||
*
|
||||
* If device handle is NULL, it is instead a null-char separated list of
|
||||
* strings of known device specifiers (list ends with an empty string).
|
||||
*/
|
||||
#define ALC_DEVICE_SPECIFIER 0x1005
|
||||
/** String for space-separated list of ALC extensions. */
|
||||
#define ALC_EXTENSIONS 0x1006
|
||||
|
||||
|
||||
/** Capture extension */
|
||||
#define ALC_EXT_CAPTURE 1
|
||||
/**
|
||||
* String for the given capture device's specifier.
|
||||
*
|
||||
* If device handle is NULL, it is instead a null-char separated list of
|
||||
* strings of known capture device specifiers (list ends with an empty string).
|
||||
*/
|
||||
#define ALC_CAPTURE_DEVICE_SPECIFIER 0x310
|
||||
/** String for the default capture device specifier. */
|
||||
#define ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER 0x311
|
||||
/** Number of sample frames available for capture. */
|
||||
#define ALC_CAPTURE_SAMPLES 0x312
|
||||
|
||||
|
||||
/** Enumerate All extension */
|
||||
#define ALC_ENUMERATE_ALL_EXT 1
|
||||
/** String for the default extended device specifier. */
|
||||
#define ALC_DEFAULT_ALL_DEVICES_SPECIFIER 0x1012
|
||||
/**
|
||||
* String for the given extended device's specifier.
|
||||
*
|
||||
* If device handle is NULL, it is instead a null-char separated list of
|
||||
* strings of known extended device specifiers (list ends with an empty string).
|
||||
*/
|
||||
#define ALC_ALL_DEVICES_SPECIFIER 0x1013
|
||||
|
||||
|
||||
/** Context management. */
|
||||
ALC_API ALCcontext* ALC_APIENTRY alcCreateContext(ALCdevice *device, const ALCint* attrlist);
|
||||
ALC_API ALCboolean ALC_APIENTRY alcMakeContextCurrent(ALCcontext *context);
|
||||
ALC_API void ALC_APIENTRY alcProcessContext(ALCcontext *context);
|
||||
ALC_API void ALC_APIENTRY alcSuspendContext(ALCcontext *context);
|
||||
ALC_API void ALC_APIENTRY alcDestroyContext(ALCcontext *context);
|
||||
ALC_API ALCcontext* ALC_APIENTRY alcGetCurrentContext(void);
|
||||
ALC_API ALCdevice* ALC_APIENTRY alcGetContextsDevice(ALCcontext *context);
|
||||
|
||||
/** Device management. */
|
||||
ALC_API ALCdevice* ALC_APIENTRY alcOpenDevice(const ALCchar *devicename);
|
||||
ALC_API ALCboolean ALC_APIENTRY alcCloseDevice(ALCdevice *device);
|
||||
|
||||
|
||||
/**
|
||||
* Error support.
|
||||
*
|
||||
* Obtain the most recent Device error.
|
||||
*/
|
||||
ALC_API ALCenum ALC_APIENTRY alcGetError(ALCdevice *device);
|
||||
|
||||
/**
|
||||
* Extension support.
|
||||
*
|
||||
* Query for the presence of an extension, and obtain any appropriate
|
||||
* function pointers and enum values.
|
||||
*/
|
||||
ALC_API ALCboolean ALC_APIENTRY alcIsExtensionPresent(ALCdevice *device, const ALCchar *extname);
|
||||
ALC_API void* ALC_APIENTRY alcGetProcAddress(ALCdevice *device, const ALCchar *funcname);
|
||||
ALC_API ALCenum ALC_APIENTRY alcGetEnumValue(ALCdevice *device, const ALCchar *enumname);
|
||||
|
||||
/** Query function. */
|
||||
ALC_API const ALCchar* ALC_APIENTRY alcGetString(ALCdevice *device, ALCenum param);
|
||||
ALC_API void ALC_APIENTRY alcGetIntegerv(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values);
|
||||
|
||||
/** Capture function. */
|
||||
ALC_API ALCdevice* ALC_APIENTRY alcCaptureOpenDevice(const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize);
|
||||
ALC_API ALCboolean ALC_APIENTRY alcCaptureCloseDevice(ALCdevice *device);
|
||||
ALC_API void ALC_APIENTRY alcCaptureStart(ALCdevice *device);
|
||||
ALC_API void ALC_APIENTRY alcCaptureStop(ALCdevice *device);
|
||||
ALC_API void ALC_APIENTRY alcCaptureSamples(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);
|
||||
|
||||
/** Pointer-to-function type, useful for dynamically getting ALC entry points. */
|
||||
typedef ALCcontext* (ALC_APIENTRY *LPALCCREATECONTEXT)(ALCdevice *device, const ALCint *attrlist);
|
||||
typedef ALCboolean (ALC_APIENTRY *LPALCMAKECONTEXTCURRENT)(ALCcontext *context);
|
||||
typedef void (ALC_APIENTRY *LPALCPROCESSCONTEXT)(ALCcontext *context);
|
||||
typedef void (ALC_APIENTRY *LPALCSUSPENDCONTEXT)(ALCcontext *context);
|
||||
typedef void (ALC_APIENTRY *LPALCDESTROYCONTEXT)(ALCcontext *context);
|
||||
typedef ALCcontext* (ALC_APIENTRY *LPALCGETCURRENTCONTEXT)(void);
|
||||
typedef ALCdevice* (ALC_APIENTRY *LPALCGETCONTEXTSDEVICE)(ALCcontext *context);
|
||||
typedef ALCdevice* (ALC_APIENTRY *LPALCOPENDEVICE)(const ALCchar *devicename);
|
||||
typedef ALCboolean (ALC_APIENTRY *LPALCCLOSEDEVICE)(ALCdevice *device);
|
||||
typedef ALCenum (ALC_APIENTRY *LPALCGETERROR)(ALCdevice *device);
|
||||
typedef ALCboolean (ALC_APIENTRY *LPALCISEXTENSIONPRESENT)(ALCdevice *device, const ALCchar *extname);
|
||||
typedef void* (ALC_APIENTRY *LPALCGETPROCADDRESS)(ALCdevice *device, const ALCchar *funcname);
|
||||
typedef ALCenum (ALC_APIENTRY *LPALCGETENUMVALUE)(ALCdevice *device, const ALCchar *enumname);
|
||||
typedef const ALCchar* (ALC_APIENTRY *LPALCGETSTRING)(ALCdevice *device, ALCenum param);
|
||||
typedef void (ALC_APIENTRY *LPALCGETINTEGERV)(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values);
|
||||
typedef ALCdevice* (ALC_APIENTRY *LPALCCAPTUREOPENDEVICE)(const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize);
|
||||
typedef ALCboolean (ALC_APIENTRY *LPALCCAPTURECLOSEDEVICE)(ALCdevice *device);
|
||||
typedef void (ALC_APIENTRY *LPALCCAPTURESTART)(ALCdevice *device);
|
||||
typedef void (ALC_APIENTRY *LPALCCAPTURESTOP)(ALCdevice *device);
|
||||
typedef void (ALC_APIENTRY *LPALCCAPTURESAMPLES)(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);
|
||||
|
||||
#if defined(__cplusplus)
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif /* AL_ALC_H */
|
||||
|
|
@ -0,0 +1,355 @@
|
|||
/**
|
||||
* OpenAL cross platform audio library
|
||||
* Copyright (C) 2008 by authors.
|
||||
* This library is free software; you can redistribute it and/or
|
||||
* modify it under the terms of the GNU Library General Public
|
||||
* License as published by the Free Software Foundation; either
|
||||
* version 2 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
|
||||
* Library General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU Library 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.
|
||||
* Or go to http://www.gnu.org/copyleft/lgpl.html
|
||||
*/
|
||||
|
||||
#ifndef AL_ALEXT_H
|
||||
#define AL_ALEXT_H
|
||||
|
||||
#include <stddef.h>
|
||||
/* Define int64_t and uint64_t types */
|
||||
#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
|
||||
#include <inttypes.h>
|
||||
#elif defined(_WIN32) && defined(__GNUC__)
|
||||
#include <stdint.h>
|
||||
#elif defined(_WIN32)
|
||||
typedef __int64 int64_t;
|
||||
typedef unsigned __int64 uint64_t;
|
||||
#else
|
||||
/* Fallback if nothing above works */
|
||||
#include <inttypes.h>
|
||||
#endif
|
||||
|
||||
#include "alc.h"
|
||||
#include "al.h"
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
#ifndef AL_LOKI_IMA_ADPCM_format
|
||||
#define AL_LOKI_IMA_ADPCM_format 1
|
||||
#define AL_FORMAT_IMA_ADPCM_MONO16_EXT 0x10000
|
||||
#define AL_FORMAT_IMA_ADPCM_STEREO16_EXT 0x10001
|
||||
#endif
|
||||
|
||||
#ifndef AL_LOKI_WAVE_format
|
||||
#define AL_LOKI_WAVE_format 1
|
||||
#define AL_FORMAT_WAVE_EXT 0x10002
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_vorbis
|
||||
#define AL_EXT_vorbis 1
|
||||
#define AL_FORMAT_VORBIS_EXT 0x10003
|
||||
#endif
|
||||
|
||||
#ifndef AL_LOKI_quadriphonic
|
||||
#define AL_LOKI_quadriphonic 1
|
||||
#define AL_FORMAT_QUAD8_LOKI 0x10004
|
||||
#define AL_FORMAT_QUAD16_LOKI 0x10005
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_float32
|
||||
#define AL_EXT_float32 1
|
||||
#define AL_FORMAT_MONO_FLOAT32 0x10010
|
||||
#define AL_FORMAT_STEREO_FLOAT32 0x10011
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_double
|
||||
#define AL_EXT_double 1
|
||||
#define AL_FORMAT_MONO_DOUBLE_EXT 0x10012
|
||||
#define AL_FORMAT_STEREO_DOUBLE_EXT 0x10013
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_MULAW
|
||||
#define AL_EXT_MULAW 1
|
||||
#define AL_FORMAT_MONO_MULAW_EXT 0x10014
|
||||
#define AL_FORMAT_STEREO_MULAW_EXT 0x10015
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_ALAW
|
||||
#define AL_EXT_ALAW 1
|
||||
#define AL_FORMAT_MONO_ALAW_EXT 0x10016
|
||||
#define AL_FORMAT_STEREO_ALAW_EXT 0x10017
|
||||
#endif
|
||||
|
||||
#ifndef ALC_LOKI_audio_channel
|
||||
#define ALC_LOKI_audio_channel 1
|
||||
#define ALC_CHAN_MAIN_LOKI 0x500001
|
||||
#define ALC_CHAN_PCM_LOKI 0x500002
|
||||
#define ALC_CHAN_CD_LOKI 0x500003
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_MCFORMATS
|
||||
#define AL_EXT_MCFORMATS 1
|
||||
#define AL_FORMAT_QUAD8 0x1204
|
||||
#define AL_FORMAT_QUAD16 0x1205
|
||||
#define AL_FORMAT_QUAD32 0x1206
|
||||
#define AL_FORMAT_REAR8 0x1207
|
||||
#define AL_FORMAT_REAR16 0x1208
|
||||
#define AL_FORMAT_REAR32 0x1209
|
||||
#define AL_FORMAT_51CHN8 0x120A
|
||||
#define AL_FORMAT_51CHN16 0x120B
|
||||
#define AL_FORMAT_51CHN32 0x120C
|
||||
#define AL_FORMAT_61CHN8 0x120D
|
||||
#define AL_FORMAT_61CHN16 0x120E
|
||||
#define AL_FORMAT_61CHN32 0x120F
|
||||
#define AL_FORMAT_71CHN8 0x1210
|
||||
#define AL_FORMAT_71CHN16 0x1211
|
||||
#define AL_FORMAT_71CHN32 0x1212
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_MULAW_MCFORMATS
|
||||
#define AL_EXT_MULAW_MCFORMATS 1
|
||||
#define AL_FORMAT_MONO_MULAW 0x10014
|
||||
#define AL_FORMAT_STEREO_MULAW 0x10015
|
||||
#define AL_FORMAT_QUAD_MULAW 0x10021
|
||||
#define AL_FORMAT_REAR_MULAW 0x10022
|
||||
#define AL_FORMAT_51CHN_MULAW 0x10023
|
||||
#define AL_FORMAT_61CHN_MULAW 0x10024
|
||||
#define AL_FORMAT_71CHN_MULAW 0x10025
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_IMA4
|
||||
#define AL_EXT_IMA4 1
|
||||
#define AL_FORMAT_MONO_IMA4 0x1300
|
||||
#define AL_FORMAT_STEREO_IMA4 0x1301
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_STATIC_BUFFER
|
||||
#define AL_EXT_STATIC_BUFFER 1
|
||||
typedef ALvoid (AL_APIENTRY*PFNALBUFFERDATASTATICPROC)(const ALint,ALenum,ALvoid*,ALsizei,ALsizei);
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
AL_API ALvoid AL_APIENTRY alBufferDataStatic(const ALint buffer, ALenum format, ALvoid *data, ALsizei len, ALsizei freq);
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifndef ALC_EXT_EFX
|
||||
#define ALC_EXT_EFX 1
|
||||
#include "efx.h"
|
||||
#endif
|
||||
|
||||
#ifndef ALC_EXT_disconnect
|
||||
#define ALC_EXT_disconnect 1
|
||||
#define ALC_CONNECTED 0x313
|
||||
#endif
|
||||
|
||||
#ifndef ALC_EXT_thread_local_context
|
||||
#define ALC_EXT_thread_local_context 1
|
||||
typedef ALCboolean (ALC_APIENTRY*PFNALCSETTHREADCONTEXTPROC)(ALCcontext *context);
|
||||
typedef ALCcontext* (ALC_APIENTRY*PFNALCGETTHREADCONTEXTPROC)(void);
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
ALC_API ALCboolean ALC_APIENTRY alcSetThreadContext(ALCcontext *context);
|
||||
ALC_API ALCcontext* ALC_APIENTRY alcGetThreadContext(void);
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_source_distance_model
|
||||
#define AL_EXT_source_distance_model 1
|
||||
#define AL_SOURCE_DISTANCE_MODEL 0x200
|
||||
#endif
|
||||
|
||||
#ifndef AL_SOFT_buffer_sub_data
|
||||
#define AL_SOFT_buffer_sub_data 1
|
||||
#define AL_BYTE_RW_OFFSETS_SOFT 0x1031
|
||||
#define AL_SAMPLE_RW_OFFSETS_SOFT 0x1032
|
||||
typedef ALvoid (AL_APIENTRY*PFNALBUFFERSUBDATASOFTPROC)(ALuint,ALenum,const ALvoid*,ALsizei,ALsizei);
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
AL_API ALvoid AL_APIENTRY alBufferSubDataSOFT(ALuint buffer,ALenum format,const ALvoid *data,ALsizei offset,ALsizei length);
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifndef AL_SOFT_loop_points
|
||||
#define AL_SOFT_loop_points 1
|
||||
#define AL_LOOP_POINTS_SOFT 0x2015
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_FOLDBACK
|
||||
#define AL_EXT_FOLDBACK 1
|
||||
#define AL_EXT_FOLDBACK_NAME "AL_EXT_FOLDBACK"
|
||||
#define AL_FOLDBACK_EVENT_BLOCK 0x4112
|
||||
#define AL_FOLDBACK_EVENT_START 0x4111
|
||||
#define AL_FOLDBACK_EVENT_STOP 0x4113
|
||||
#define AL_FOLDBACK_MODE_MONO 0x4101
|
||||
#define AL_FOLDBACK_MODE_STEREO 0x4102
|
||||
typedef void (AL_APIENTRY*LPALFOLDBACKCALLBACK)(ALenum,ALsizei);
|
||||
typedef void (AL_APIENTRY*LPALREQUESTFOLDBACKSTART)(ALenum,ALsizei,ALsizei,ALfloat*,LPALFOLDBACKCALLBACK);
|
||||
typedef void (AL_APIENTRY*LPALREQUESTFOLDBACKSTOP)(void);
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
AL_API void AL_APIENTRY alRequestFoldbackStart(ALenum mode,ALsizei count,ALsizei length,ALfloat *mem,LPALFOLDBACKCALLBACK callback);
|
||||
AL_API void AL_APIENTRY alRequestFoldbackStop(void);
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifndef ALC_EXT_DEDICATED
|
||||
#define ALC_EXT_DEDICATED 1
|
||||
#define AL_DEDICATED_GAIN 0x0001
|
||||
#define AL_EFFECT_DEDICATED_DIALOGUE 0x9001
|
||||
#define AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT 0x9000
|
||||
#endif
|
||||
|
||||
#ifndef AL_SOFT_buffer_samples
|
||||
#define AL_SOFT_buffer_samples 1
|
||||
/* Channel configurations */
|
||||
#define AL_MONO_SOFT 0x1500
|
||||
#define AL_STEREO_SOFT 0x1501
|
||||
#define AL_REAR_SOFT 0x1502
|
||||
#define AL_QUAD_SOFT 0x1503
|
||||
#define AL_5POINT1_SOFT 0x1504
|
||||
#define AL_6POINT1_SOFT 0x1505
|
||||
#define AL_7POINT1_SOFT 0x1506
|
||||
|
||||
/* Sample types */
|
||||
#define AL_BYTE_SOFT 0x1400
|
||||
#define AL_UNSIGNED_BYTE_SOFT 0x1401
|
||||
#define AL_SHORT_SOFT 0x1402
|
||||
#define AL_UNSIGNED_SHORT_SOFT 0x1403
|
||||
#define AL_INT_SOFT 0x1404
|
||||
#define AL_UNSIGNED_INT_SOFT 0x1405
|
||||
#define AL_FLOAT_SOFT 0x1406
|
||||
#define AL_DOUBLE_SOFT 0x1407
|
||||
#define AL_BYTE3_SOFT 0x1408
|
||||
#define AL_UNSIGNED_BYTE3_SOFT 0x1409
|
||||
|
||||
/* Storage formats */
|
||||
#define AL_MONO8_SOFT 0x1100
|
||||
#define AL_MONO16_SOFT 0x1101
|
||||
#define AL_MONO32F_SOFT 0x10010
|
||||
#define AL_STEREO8_SOFT 0x1102
|
||||
#define AL_STEREO16_SOFT 0x1103
|
||||
#define AL_STEREO32F_SOFT 0x10011
|
||||
#define AL_QUAD8_SOFT 0x1204
|
||||
#define AL_QUAD16_SOFT 0x1205
|
||||
#define AL_QUAD32F_SOFT 0x1206
|
||||
#define AL_REAR8_SOFT 0x1207
|
||||
#define AL_REAR16_SOFT 0x1208
|
||||
#define AL_REAR32F_SOFT 0x1209
|
||||
#define AL_5POINT1_8_SOFT 0x120A
|
||||
#define AL_5POINT1_16_SOFT 0x120B
|
||||
#define AL_5POINT1_32F_SOFT 0x120C
|
||||
#define AL_6POINT1_8_SOFT 0x120D
|
||||
#define AL_6POINT1_16_SOFT 0x120E
|
||||
#define AL_6POINT1_32F_SOFT 0x120F
|
||||
#define AL_7POINT1_8_SOFT 0x1210
|
||||
#define AL_7POINT1_16_SOFT 0x1211
|
||||
#define AL_7POINT1_32F_SOFT 0x1212
|
||||
|
||||
/* Buffer attributes */
|
||||
#define AL_INTERNAL_FORMAT_SOFT 0x2008
|
||||
#define AL_BYTE_LENGTH_SOFT 0x2009
|
||||
#define AL_SAMPLE_LENGTH_SOFT 0x200A
|
||||
#define AL_SEC_LENGTH_SOFT 0x200B
|
||||
|
||||
typedef void (AL_APIENTRY*LPALBUFFERSAMPLESSOFT)(ALuint,ALuint,ALenum,ALsizei,ALenum,ALenum,const ALvoid*);
|
||||
typedef void (AL_APIENTRY*LPALBUFFERSUBSAMPLESSOFT)(ALuint,ALsizei,ALsizei,ALenum,ALenum,const ALvoid*);
|
||||
typedef void (AL_APIENTRY*LPALGETBUFFERSAMPLESSOFT)(ALuint,ALsizei,ALsizei,ALenum,ALenum,ALvoid*);
|
||||
typedef ALboolean (AL_APIENTRY*LPALISBUFFERFORMATSUPPORTEDSOFT)(ALenum);
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
AL_API void AL_APIENTRY alBufferSamplesSOFT(ALuint buffer, ALuint samplerate, ALenum internalformat, ALsizei samples, ALenum channels, ALenum type, const ALvoid *data);
|
||||
AL_API void AL_APIENTRY alBufferSubSamplesSOFT(ALuint buffer, ALsizei offset, ALsizei samples, ALenum channels, ALenum type, const ALvoid *data);
|
||||
AL_API void AL_APIENTRY alGetBufferSamplesSOFT(ALuint buffer, ALsizei offset, ALsizei samples, ALenum channels, ALenum type, ALvoid *data);
|
||||
AL_API ALboolean AL_APIENTRY alIsBufferFormatSupportedSOFT(ALenum format);
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifndef AL_SOFT_direct_channels
|
||||
#define AL_SOFT_direct_channels 1
|
||||
#define AL_DIRECT_CHANNELS_SOFT 0x1033
|
||||
#endif
|
||||
|
||||
#ifndef ALC_SOFT_loopback
|
||||
#define ALC_SOFT_loopback 1
|
||||
#define ALC_FORMAT_CHANNELS_SOFT 0x1990
|
||||
#define ALC_FORMAT_TYPE_SOFT 0x1991
|
||||
|
||||
/* Sample types */
|
||||
#define ALC_BYTE_SOFT 0x1400
|
||||
#define ALC_UNSIGNED_BYTE_SOFT 0x1401
|
||||
#define ALC_SHORT_SOFT 0x1402
|
||||
#define ALC_UNSIGNED_SHORT_SOFT 0x1403
|
||||
#define ALC_INT_SOFT 0x1404
|
||||
#define ALC_UNSIGNED_INT_SOFT 0x1405
|
||||
#define ALC_FLOAT_SOFT 0x1406
|
||||
|
||||
/* Channel configurations */
|
||||
#define ALC_MONO_SOFT 0x1500
|
||||
#define ALC_STEREO_SOFT 0x1501
|
||||
#define ALC_QUAD_SOFT 0x1503
|
||||
#define ALC_5POINT1_SOFT 0x1504
|
||||
#define ALC_6POINT1_SOFT 0x1505
|
||||
#define ALC_7POINT1_SOFT 0x1506
|
||||
|
||||
typedef ALCdevice* (ALC_APIENTRY*LPALCLOOPBACKOPENDEVICESOFT)(const ALCchar*);
|
||||
typedef ALCboolean (ALC_APIENTRY*LPALCISRENDERFORMATSUPPORTEDSOFT)(ALCdevice*,ALCsizei,ALCenum,ALCenum);
|
||||
typedef void (ALC_APIENTRY*LPALCRENDERSAMPLESSOFT)(ALCdevice*,ALCvoid*,ALCsizei);
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
ALC_API ALCdevice* ALC_APIENTRY alcLoopbackOpenDeviceSOFT(const ALCchar *deviceName);
|
||||
ALC_API ALCboolean ALC_APIENTRY alcIsRenderFormatSupportedSOFT(ALCdevice *device, ALCsizei freq, ALCenum channels, ALCenum type);
|
||||
ALC_API void ALC_APIENTRY alcRenderSamplesSOFT(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_STEREO_ANGLES
|
||||
#define AL_EXT_STEREO_ANGLES 1
|
||||
#define AL_STEREO_ANGLES 0x1030
|
||||
#endif
|
||||
|
||||
#ifndef AL_EXT_SOURCE_RADIUS
|
||||
#define AL_EXT_SOURCE_RADIUS 1
|
||||
#define AL_SOURCE_RADIUS 0x1031
|
||||
#endif
|
||||
|
||||
#ifndef AL_SOFT_source_latency
|
||||
#define AL_SOFT_source_latency 1
|
||||
#define AL_SAMPLE_OFFSET_LATENCY_SOFT 0x1200
|
||||
#define AL_SEC_OFFSET_LATENCY_SOFT 0x1201
|
||||
typedef int64_t ALint64SOFT;
|
||||
typedef uint64_t ALuint64SOFT;
|
||||
typedef void (AL_APIENTRY*LPALSOURCEDSOFT)(ALuint,ALenum,ALdouble);
|
||||
typedef void (AL_APIENTRY*LPALSOURCE3DSOFT)(ALuint,ALenum,ALdouble,ALdouble,ALdouble);
|
||||
typedef void (AL_APIENTRY*LPALSOURCEDVSOFT)(ALuint,ALenum,const ALdouble*);
|
||||
typedef void (AL_APIENTRY*LPALGETSOURCEDSOFT)(ALuint,ALenum,ALdouble*);
|
||||
typedef void (AL_APIENTRY*LPALGETSOURCE3DSOFT)(ALuint,ALenum,ALdouble*,ALdouble*,ALdouble*);
|
||||
typedef void (AL_APIENTRY*LPALGETSOURCEDVSOFT)(ALuint,ALenum,ALdouble*);
|
||||
typedef void (AL_APIENTRY*LPALSOURCEI64SOFT)(ALuint,ALenum,ALint64SOFT);
|
||||
typedef void (AL_APIENTRY*LPALSOURCE3I64SOFT)(ALuint,ALenum,ALint64SOFT,ALint64SOFT,ALint64SOFT);
|
||||
typedef void (AL_APIENTRY*LPALSOURCEI64VSOFT)(ALuint,ALenum,const ALint64SOFT*);
|
||||
typedef void (AL_APIENTRY*LPALGETSOURCEI64SOFT)(ALuint,ALenum,ALint64SOFT*);
|
||||
typedef void (AL_APIENTRY*LPALGETSOURCE3I64SOFT)(ALuint,ALenum,ALint64SOFT*,ALint64SOFT*,ALint64SOFT*);
|
||||
typedef void (AL_APIENTRY*LPALGETSOURCEI64VSOFT)(ALuint,ALenum,ALint64SOFT*);
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
AL_API void AL_APIENTRY alSourcedSOFT(ALuint source, ALenum param, ALdouble value);
|
||||
AL_API void AL_APIENTRY alSource3dSOFT(ALuint source, ALenum param, ALdouble value1, ALdouble value2, ALdouble value3);
|
||||
AL_API void AL_APIENTRY alSourcedvSOFT(ALuint source, ALenum param, const ALdouble *values);
|
||||
AL_API void AL_APIENTRY alGetSourcedSOFT(ALuint source, ALenum param, ALdouble *value);
|
||||
AL_API void AL_APIENTRY alGetSource3dSOFT(ALuint source, ALenum param, ALdouble *value1, ALdouble *value2, ALdouble *value3);
|
||||
AL_API void AL_APIENTRY alGetSourcedvSOFT(ALuint source, ALenum param, ALdouble *values);
|
||||
AL_API void AL_APIENTRY alSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT value);
|
||||
AL_API void AL_APIENTRY alSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT value1, ALint64SOFT value2, ALint64SOFT value3);
|
||||
AL_API void AL_APIENTRY alSourcei64vSOFT(ALuint source, ALenum param, const ALint64SOFT *values);
|
||||
AL_API void AL_APIENTRY alGetSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT *value);
|
||||
AL_API void AL_APIENTRY alGetSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT *value1, ALint64SOFT *value2, ALint64SOFT *value3);
|
||||
AL_API void AL_APIENTRY alGetSourcei64vSOFT(ALuint source, ALenum param, ALint64SOFT *values);
|
||||
#endif
|
||||
#endif
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
#endif
|
||||
|
|
@ -0,0 +1,3 @@
|
|||
/* The tokens that would be defined here are already defined in efx.h. This
|
||||
* empty file is here to provide compatibility with Windows-based projects
|
||||
* that would include it. */
|
||||
|
|
@ -0,0 +1,402 @@
|
|||
/* Reverb presets for EFX */
|
||||
|
||||
#ifndef EFX_PRESETS_H
|
||||
#define EFX_PRESETS_H
|
||||
|
||||
#ifndef EFXEAXREVERBPROPERTIES_DEFINED
|
||||
#define EFXEAXREVERBPROPERTIES_DEFINED
|
||||
typedef struct {
|
||||
float flDensity;
|
||||
float flDiffusion;
|
||||
float flGain;
|
||||
float flGainHF;
|
||||
float flGainLF;
|
||||
float flDecayTime;
|
||||
float flDecayHFRatio;
|
||||
float flDecayLFRatio;
|
||||
float flReflectionsGain;
|
||||
float flReflectionsDelay;
|
||||
float flReflectionsPan[3];
|
||||
float flLateReverbGain;
|
||||
float flLateReverbDelay;
|
||||
float flLateReverbPan[3];
|
||||
float flEchoTime;
|
||||
float flEchoDepth;
|
||||
float flModulationTime;
|
||||
float flModulationDepth;
|
||||
float flAirAbsorptionGainHF;
|
||||
float flHFReference;
|
||||
float flLFReference;
|
||||
float flRoomRolloffFactor;
|
||||
int iDecayHFLimit;
|
||||
} EFXEAXREVERBPROPERTIES, *LPEFXEAXREVERBPROPERTIES;
|
||||
#endif
|
||||
|
||||
/* Default Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_GENERIC \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.8913f, 1.0000f, 1.4900f, 0.8300f, 1.0000f, 0.0500f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PADDEDCELL \
|
||||
{ 0.1715f, 1.0000f, 0.3162f, 0.0010f, 1.0000f, 0.1700f, 0.1000f, 1.0000f, 0.2500f, 0.0010f, { 0.0000f, 0.0000f, 0.0000f }, 1.2691f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ROOM \
|
||||
{ 0.4287f, 1.0000f, 0.3162f, 0.5929f, 1.0000f, 0.4000f, 0.8300f, 1.0000f, 0.1503f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 1.0629f, 0.0030f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_BATHROOM \
|
||||
{ 0.1715f, 1.0000f, 0.3162f, 0.2512f, 1.0000f, 1.4900f, 0.5400f, 1.0000f, 0.6531f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 3.2734f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_LIVINGROOM \
|
||||
{ 0.9766f, 1.0000f, 0.3162f, 0.0010f, 1.0000f, 0.5000f, 0.1000f, 1.0000f, 0.2051f, 0.0030f, { 0.0000f, 0.0000f, 0.0000f }, 0.2805f, 0.0040f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_STONEROOM \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.7079f, 1.0000f, 2.3100f, 0.6400f, 1.0000f, 0.4411f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1003f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_AUDITORIUM \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.5781f, 1.0000f, 4.3200f, 0.5900f, 1.0000f, 0.4032f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.7170f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CONCERTHALL \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.5623f, 1.0000f, 3.9200f, 0.7000f, 1.0000f, 0.2427f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.9977f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CAVE \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 1.0000f, 1.0000f, 2.9100f, 1.3000f, 1.0000f, 0.5000f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.7063f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ARENA \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.4477f, 1.0000f, 7.2400f, 0.3300f, 1.0000f, 0.2612f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.0186f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_HANGAR \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.3162f, 1.0000f, 10.0500f, 0.2300f, 1.0000f, 0.5000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.2560f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CARPETEDHALLWAY \
|
||||
{ 0.4287f, 1.0000f, 0.3162f, 0.0100f, 1.0000f, 0.3000f, 0.1000f, 1.0000f, 0.1215f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 0.1531f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_HALLWAY \
|
||||
{ 0.3645f, 1.0000f, 0.3162f, 0.7079f, 1.0000f, 1.4900f, 0.5900f, 1.0000f, 0.2458f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.6615f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_STONECORRIDOR \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.7612f, 1.0000f, 2.7000f, 0.7900f, 1.0000f, 0.2472f, 0.0130f, { 0.0000f, 0.0000f, 0.0000f }, 1.5758f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ALLEY \
|
||||
{ 1.0000f, 0.3000f, 0.3162f, 0.7328f, 1.0000f, 1.4900f, 0.8600f, 1.0000f, 0.2500f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.9954f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 0.9500f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FOREST \
|
||||
{ 1.0000f, 0.3000f, 0.3162f, 0.0224f, 1.0000f, 1.4900f, 0.5400f, 1.0000f, 0.0525f, 0.1620f, { 0.0000f, 0.0000f, 0.0000f }, 0.7682f, 0.0880f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CITY \
|
||||
{ 1.0000f, 0.5000f, 0.3162f, 0.3981f, 1.0000f, 1.4900f, 0.6700f, 1.0000f, 0.0730f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.1427f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_MOUNTAINS \
|
||||
{ 1.0000f, 0.2700f, 0.3162f, 0.0562f, 1.0000f, 1.4900f, 0.2100f, 1.0000f, 0.0407f, 0.3000f, { 0.0000f, 0.0000f, 0.0000f }, 0.1919f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_QUARRY \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.3162f, 1.0000f, 1.4900f, 0.8300f, 1.0000f, 0.0000f, 0.0610f, { 0.0000f, 0.0000f, 0.0000f }, 1.7783f, 0.0250f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 0.7000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PLAIN \
|
||||
{ 1.0000f, 0.2100f, 0.3162f, 0.1000f, 1.0000f, 1.4900f, 0.5000f, 1.0000f, 0.0585f, 0.1790f, { 0.0000f, 0.0000f, 0.0000f }, 0.1089f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PARKINGLOT \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 1.0000f, 1.0000f, 1.6500f, 1.5000f, 1.0000f, 0.2082f, 0.0080f, { 0.0000f, 0.0000f, 0.0000f }, 0.2652f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SEWERPIPE \
|
||||
{ 0.3071f, 0.8000f, 0.3162f, 0.3162f, 1.0000f, 2.8100f, 0.1400f, 1.0000f, 1.6387f, 0.0140f, { 0.0000f, 0.0000f, 0.0000f }, 3.2471f, 0.0210f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_UNDERWATER \
|
||||
{ 0.3645f, 1.0000f, 0.3162f, 0.0100f, 1.0000f, 1.4900f, 0.1000f, 1.0000f, 0.5963f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 7.0795f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 1.1800f, 0.3480f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRUGGED \
|
||||
{ 0.4287f, 0.5000f, 0.3162f, 1.0000f, 1.0000f, 8.3900f, 1.3900f, 1.0000f, 0.8760f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 3.1081f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 1.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DIZZY \
|
||||
{ 0.3645f, 0.6000f, 0.3162f, 0.6310f, 1.0000f, 17.2300f, 0.5600f, 1.0000f, 0.1392f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.4937f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.8100f, 0.3100f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PSYCHOTIC \
|
||||
{ 0.0625f, 0.5000f, 0.3162f, 0.8404f, 1.0000f, 7.5600f, 0.9100f, 1.0000f, 0.4864f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 2.4378f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 4.0000f, 1.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
/* Castle Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_SMALLROOM \
|
||||
{ 1.0000f, 0.8900f, 0.3162f, 0.3981f, 0.1000f, 1.2200f, 0.8300f, 0.3100f, 0.8913f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_SHORTPASSAGE \
|
||||
{ 1.0000f, 0.8900f, 0.3162f, 0.3162f, 0.1000f, 2.3200f, 0.8300f, 0.3100f, 0.8913f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_MEDIUMROOM \
|
||||
{ 1.0000f, 0.9300f, 0.3162f, 0.2818f, 0.1000f, 2.0400f, 0.8300f, 0.4600f, 0.6310f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 1.5849f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1550f, 0.0300f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_LARGEROOM \
|
||||
{ 1.0000f, 0.8200f, 0.3162f, 0.2818f, 0.1259f, 2.5300f, 0.8300f, 0.5000f, 0.4467f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.1850f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_LONGPASSAGE \
|
||||
{ 1.0000f, 0.8900f, 0.3162f, 0.3981f, 0.1000f, 3.4200f, 0.8300f, 0.3100f, 0.8913f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_HALL \
|
||||
{ 1.0000f, 0.8100f, 0.3162f, 0.2818f, 0.1778f, 3.1400f, 0.7900f, 0.6200f, 0.1778f, 0.0560f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_CUPBOARD \
|
||||
{ 1.0000f, 0.8900f, 0.3162f, 0.2818f, 0.1000f, 0.6700f, 0.8700f, 0.3100f, 1.4125f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 3.5481f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_COURTYARD \
|
||||
{ 1.0000f, 0.4200f, 0.3162f, 0.4467f, 0.1995f, 2.1300f, 0.6100f, 0.2300f, 0.2239f, 0.1600f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0360f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.3700f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CASTLE_ALCOVE \
|
||||
{ 1.0000f, 0.8900f, 0.3162f, 0.5012f, 0.1000f, 1.6400f, 0.8700f, 0.3100f, 1.0000f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
|
||||
|
||||
/* Factory Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_SMALLROOM \
|
||||
{ 0.3645f, 0.8200f, 0.3162f, 0.7943f, 0.5012f, 1.7200f, 0.6500f, 1.3100f, 0.7079f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.7783f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.1190f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_SHORTPASSAGE \
|
||||
{ 0.3645f, 0.6400f, 0.2512f, 0.7943f, 0.5012f, 2.5300f, 0.6500f, 1.3100f, 1.0000f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.1350f, 0.2300f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_MEDIUMROOM \
|
||||
{ 0.4287f, 0.8200f, 0.2512f, 0.7943f, 0.5012f, 2.7600f, 0.6500f, 1.3100f, 0.2818f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1740f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_LARGEROOM \
|
||||
{ 0.4287f, 0.7500f, 0.2512f, 0.7079f, 0.6310f, 4.2400f, 0.5100f, 1.3100f, 0.1778f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.2310f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_LONGPASSAGE \
|
||||
{ 0.3645f, 0.6400f, 0.2512f, 0.7943f, 0.5012f, 4.0600f, 0.6500f, 1.3100f, 1.0000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0370f, { 0.0000f, 0.0000f, 0.0000f }, 0.1350f, 0.2300f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_HALL \
|
||||
{ 0.4287f, 0.7500f, 0.3162f, 0.7079f, 0.6310f, 7.4300f, 0.5100f, 1.3100f, 0.0631f, 0.0730f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0270f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_CUPBOARD \
|
||||
{ 0.3071f, 0.6300f, 0.2512f, 0.7943f, 0.5012f, 0.4900f, 0.6500f, 1.3100f, 1.2589f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.1070f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_COURTYARD \
|
||||
{ 0.3071f, 0.5700f, 0.3162f, 0.3162f, 0.6310f, 2.3200f, 0.2900f, 0.5600f, 0.2239f, 0.1400f, { 0.0000f, 0.0000f, 0.0000f }, 0.3981f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2900f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_FACTORY_ALCOVE \
|
||||
{ 0.3645f, 0.5900f, 0.2512f, 0.7943f, 0.5012f, 3.1400f, 0.6500f, 1.3100f, 1.4125f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.1140f, 0.1000f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
|
||||
|
||||
/* Ice Palace Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_SMALLROOM \
|
||||
{ 1.0000f, 0.8400f, 0.3162f, 0.5623f, 0.2818f, 1.5100f, 1.5300f, 0.2700f, 0.8913f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1640f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_SHORTPASSAGE \
|
||||
{ 1.0000f, 0.7500f, 0.3162f, 0.5623f, 0.2818f, 1.7900f, 1.4600f, 0.2800f, 0.5012f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0190f, { 0.0000f, 0.0000f, 0.0000f }, 0.1770f, 0.0900f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_MEDIUMROOM \
|
||||
{ 1.0000f, 0.8700f, 0.3162f, 0.5623f, 0.4467f, 2.2200f, 1.5300f, 0.3200f, 0.3981f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0270f, { 0.0000f, 0.0000f, 0.0000f }, 0.1860f, 0.1200f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_LARGEROOM \
|
||||
{ 1.0000f, 0.8100f, 0.3162f, 0.5623f, 0.4467f, 3.1400f, 1.5300f, 0.3200f, 0.2512f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0270f, { 0.0000f, 0.0000f, 0.0000f }, 0.2140f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_LONGPASSAGE \
|
||||
{ 1.0000f, 0.7700f, 0.3162f, 0.5623f, 0.3981f, 3.0100f, 1.4600f, 0.2800f, 0.7943f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0250f, { 0.0000f, 0.0000f, 0.0000f }, 0.1860f, 0.0400f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_HALL \
|
||||
{ 1.0000f, 0.7600f, 0.3162f, 0.4467f, 0.5623f, 5.4900f, 1.5300f, 0.3800f, 0.1122f, 0.0540f, { 0.0000f, 0.0000f, 0.0000f }, 0.6310f, 0.0520f, { 0.0000f, 0.0000f, 0.0000f }, 0.2260f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_CUPBOARD \
|
||||
{ 1.0000f, 0.8300f, 0.3162f, 0.5012f, 0.2239f, 0.7600f, 1.5300f, 0.2600f, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.1430f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_COURTYARD \
|
||||
{ 1.0000f, 0.5900f, 0.3162f, 0.2818f, 0.3162f, 2.0400f, 1.2000f, 0.3800f, 0.3162f, 0.1730f, { 0.0000f, 0.0000f, 0.0000f }, 0.3162f, 0.0430f, { 0.0000f, 0.0000f, 0.0000f }, 0.2350f, 0.4800f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_ICEPALACE_ALCOVE \
|
||||
{ 1.0000f, 0.8400f, 0.3162f, 0.5623f, 0.2818f, 2.7600f, 1.4600f, 0.2800f, 1.1220f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1610f, 0.0900f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
/* Space Station Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_SMALLROOM \
|
||||
{ 0.2109f, 0.7000f, 0.3162f, 0.7079f, 0.8913f, 1.7200f, 0.8200f, 0.5500f, 0.7943f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0130f, { 0.0000f, 0.0000f, 0.0000f }, 0.1880f, 0.2600f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_SHORTPASSAGE \
|
||||
{ 0.2109f, 0.8700f, 0.3162f, 0.6310f, 0.8913f, 3.5700f, 0.5000f, 0.5500f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.1720f, 0.2000f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_MEDIUMROOM \
|
||||
{ 0.2109f, 0.7500f, 0.3162f, 0.6310f, 0.8913f, 3.0100f, 0.5000f, 0.5500f, 0.3981f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0350f, { 0.0000f, 0.0000f, 0.0000f }, 0.2090f, 0.3100f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_LARGEROOM \
|
||||
{ 0.3645f, 0.8100f, 0.3162f, 0.6310f, 0.8913f, 3.8900f, 0.3800f, 0.6100f, 0.3162f, 0.0560f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0350f, { 0.0000f, 0.0000f, 0.0000f }, 0.2330f, 0.2800f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_LONGPASSAGE \
|
||||
{ 0.4287f, 0.8200f, 0.3162f, 0.6310f, 0.8913f, 4.6200f, 0.6200f, 0.5500f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0310f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2300f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_HALL \
|
||||
{ 0.4287f, 0.8700f, 0.3162f, 0.6310f, 0.8913f, 7.1100f, 0.3800f, 0.6100f, 0.1778f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.6310f, 0.0470f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2500f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_CUPBOARD \
|
||||
{ 0.1715f, 0.5600f, 0.3162f, 0.7079f, 0.8913f, 0.7900f, 0.8100f, 0.5500f, 1.4125f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.7783f, 0.0180f, { 0.0000f, 0.0000f, 0.0000f }, 0.1810f, 0.3100f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPACESTATION_ALCOVE \
|
||||
{ 0.2109f, 0.7800f, 0.3162f, 0.7079f, 0.8913f, 1.1600f, 0.8100f, 0.5500f, 1.4125f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0180f, { 0.0000f, 0.0000f, 0.0000f }, 0.1920f, 0.2100f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
|
||||
|
||||
/* Wooden Galleon Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_SMALLROOM \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.1122f, 0.3162f, 0.7900f, 0.3200f, 0.8700f, 1.0000f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_SHORTPASSAGE \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.1259f, 0.3162f, 1.7500f, 0.5000f, 0.8700f, 0.8913f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.6310f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_MEDIUMROOM \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.1000f, 0.2818f, 1.4700f, 0.4200f, 0.8200f, 0.8913f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_LARGEROOM \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.0891f, 0.2818f, 2.6500f, 0.3300f, 0.8200f, 0.8913f, 0.0660f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_LONGPASSAGE \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.1000f, 0.3162f, 1.9900f, 0.4000f, 0.7900f, 1.0000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.4467f, 0.0360f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_HALL \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.0794f, 0.2818f, 3.4500f, 0.3000f, 0.8200f, 0.8913f, 0.0880f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0630f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_CUPBOARD \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.1413f, 0.3162f, 0.5600f, 0.4600f, 0.9100f, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0280f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_COURTYARD \
|
||||
{ 1.0000f, 0.6500f, 0.3162f, 0.0794f, 0.3162f, 1.7900f, 0.3500f, 0.7900f, 0.5623f, 0.1230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1000f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_WOODEN_ALCOVE \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.1259f, 0.3162f, 1.2200f, 0.6200f, 0.9100f, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
|
||||
|
||||
/* Sports Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_SPORT_EMPTYSTADIUM \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.4467f, 0.7943f, 6.2600f, 0.5100f, 1.1000f, 0.0631f, 0.1830f, { 0.0000f, 0.0000f, 0.0000f }, 0.3981f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPORT_SQUASHCOURT \
|
||||
{ 1.0000f, 0.7500f, 0.3162f, 0.3162f, 0.7943f, 2.2200f, 0.9100f, 1.1600f, 0.4467f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1260f, 0.1900f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPORT_SMALLSWIMMINGPOOL \
|
||||
{ 1.0000f, 0.7000f, 0.3162f, 0.7943f, 0.8913f, 2.7600f, 1.2500f, 1.1400f, 0.6310f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1790f, 0.1500f, 0.8950f, 0.1900f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPORT_LARGESWIMMINGPOOL \
|
||||
{ 1.0000f, 0.8200f, 0.3162f, 0.7943f, 1.0000f, 5.4900f, 1.3100f, 1.1400f, 0.4467f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 0.5012f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2220f, 0.5500f, 1.1590f, 0.2100f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPORT_GYMNASIUM \
|
||||
{ 1.0000f, 0.8100f, 0.3162f, 0.4467f, 0.8913f, 3.1400f, 1.0600f, 1.3500f, 0.3981f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.5623f, 0.0450f, { 0.0000f, 0.0000f, 0.0000f }, 0.1460f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPORT_FULLSTADIUM \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.0708f, 0.7943f, 5.2500f, 0.1700f, 0.8000f, 0.1000f, 0.1880f, { 0.0000f, 0.0000f, 0.0000f }, 0.2818f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SPORT_STADIUMTANNOY \
|
||||
{ 1.0000f, 0.7800f, 0.3162f, 0.5623f, 0.5012f, 2.5300f, 0.8800f, 0.6800f, 0.2818f, 0.2300f, { 0.0000f, 0.0000f, 0.0000f }, 0.5012f, 0.0630f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
/* Prefab Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_PREFAB_WORKSHOP \
|
||||
{ 0.4287f, 1.0000f, 0.3162f, 0.1413f, 0.3981f, 0.7600f, 1.0000f, 1.0000f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PREFAB_SCHOOLROOM \
|
||||
{ 0.4022f, 0.6900f, 0.3162f, 0.6310f, 0.5012f, 0.9800f, 0.4500f, 0.1800f, 1.4125f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.0950f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PREFAB_PRACTISEROOM \
|
||||
{ 0.4022f, 0.8700f, 0.3162f, 0.3981f, 0.5012f, 1.1200f, 0.5600f, 0.1800f, 1.2589f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.0950f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PREFAB_OUTHOUSE \
|
||||
{ 1.0000f, 0.8200f, 0.3162f, 0.1122f, 0.1585f, 1.3800f, 0.3800f, 0.3500f, 0.8913f, 0.0240f, { 0.0000f, 0.0000f, -0.0000f }, 0.6310f, 0.0440f, { 0.0000f, 0.0000f, 0.0000f }, 0.1210f, 0.1700f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PREFAB_CARAVAN \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.0891f, 0.1259f, 0.4300f, 1.5000f, 1.0000f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
/* Dome and Pipe Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_DOME_TOMB \
|
||||
{ 1.0000f, 0.7900f, 0.3162f, 0.3548f, 0.2239f, 4.1800f, 0.2100f, 0.1000f, 0.3868f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 1.6788f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.1770f, 0.1900f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PIPE_SMALL \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.3548f, 0.2239f, 5.0400f, 0.1000f, 0.1000f, 0.5012f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 2.5119f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DOME_SAINTPAULS \
|
||||
{ 1.0000f, 0.8700f, 0.3162f, 0.3548f, 0.2239f, 10.4800f, 0.1900f, 0.1000f, 0.1778f, 0.0900f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0420f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1200f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PIPE_LONGTHIN \
|
||||
{ 0.2560f, 0.9100f, 0.3162f, 0.4467f, 0.2818f, 9.2100f, 0.1800f, 0.1000f, 0.7079f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PIPE_LARGE \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.3548f, 0.2239f, 8.4500f, 0.1000f, 0.1000f, 0.3981f, 0.0460f, { 0.0000f, 0.0000f, 0.0000f }, 1.5849f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_PIPE_RESONANT \
|
||||
{ 0.1373f, 0.9100f, 0.3162f, 0.4467f, 0.2818f, 6.8100f, 0.1800f, 0.1000f, 0.7079f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x0 }
|
||||
|
||||
/* Outdoors Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_OUTDOORS_BACKYARD \
|
||||
{ 1.0000f, 0.4500f, 0.3162f, 0.2512f, 0.5012f, 1.1200f, 0.3400f, 0.4600f, 0.4467f, 0.0690f, { 0.0000f, 0.0000f, -0.0000f }, 0.7079f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.2180f, 0.3400f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_OUTDOORS_ROLLINGPLAINS \
|
||||
{ 1.0000f, 0.0000f, 0.3162f, 0.0112f, 0.6310f, 2.1300f, 0.2100f, 0.4600f, 0.1778f, 0.3000f, { 0.0000f, 0.0000f, -0.0000f }, 0.4467f, 0.0190f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_OUTDOORS_DEEPCANYON \
|
||||
{ 1.0000f, 0.7400f, 0.3162f, 0.1778f, 0.6310f, 3.8900f, 0.2100f, 0.4600f, 0.3162f, 0.2230f, { 0.0000f, 0.0000f, -0.0000f }, 0.3548f, 0.0190f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_OUTDOORS_CREEK \
|
||||
{ 1.0000f, 0.3500f, 0.3162f, 0.1778f, 0.5012f, 2.1300f, 0.2100f, 0.4600f, 0.3981f, 0.1150f, { 0.0000f, 0.0000f, -0.0000f }, 0.1995f, 0.0310f, { 0.0000f, 0.0000f, 0.0000f }, 0.2180f, 0.3400f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_OUTDOORS_VALLEY \
|
||||
{ 1.0000f, 0.2800f, 0.3162f, 0.0282f, 0.1585f, 2.8800f, 0.2600f, 0.3500f, 0.1413f, 0.2630f, { 0.0000f, 0.0000f, -0.0000f }, 0.3981f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.3400f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
|
||||
|
||||
/* Mood Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_MOOD_HEAVEN \
|
||||
{ 1.0000f, 0.9400f, 0.3162f, 0.7943f, 0.4467f, 5.0400f, 1.1200f, 0.5600f, 0.2427f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0800f, 2.7420f, 0.0500f, 0.9977f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_MOOD_HELL \
|
||||
{ 1.0000f, 0.5700f, 0.3162f, 0.3548f, 0.4467f, 3.5700f, 0.4900f, 2.0000f, 0.0000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1100f, 0.0400f, 2.1090f, 0.5200f, 0.9943f, 5000.0000f, 139.5000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_MOOD_MEMORY \
|
||||
{ 1.0000f, 0.8500f, 0.3162f, 0.6310f, 0.3548f, 4.0600f, 0.8200f, 0.5600f, 0.0398f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.4740f, 0.4500f, 0.9886f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
/* Driving Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_COMMENTATOR \
|
||||
{ 1.0000f, 0.0000f, 3.1623f, 0.5623f, 0.5012f, 2.4200f, 0.8800f, 0.6800f, 0.1995f, 0.0930f, { 0.0000f, 0.0000f, 0.0000f }, 0.2512f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9886f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_PITGARAGE \
|
||||
{ 0.4287f, 0.5900f, 0.3162f, 0.7079f, 0.5623f, 1.7200f, 0.9300f, 0.8700f, 0.5623f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_INCAR_RACER \
|
||||
{ 0.0832f, 0.8000f, 0.3162f, 1.0000f, 0.7943f, 0.1700f, 2.0000f, 0.4100f, 1.7783f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10268.2002f, 251.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_INCAR_SPORTS \
|
||||
{ 0.0832f, 0.8000f, 0.3162f, 0.6310f, 1.0000f, 0.1700f, 0.7500f, 0.4100f, 1.0000f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.5623f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10268.2002f, 251.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_INCAR_LUXURY \
|
||||
{ 0.2560f, 1.0000f, 0.3162f, 0.1000f, 0.5012f, 0.1300f, 0.4100f, 0.4600f, 0.7943f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.5849f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10268.2002f, 251.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_FULLGRANDSTAND \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 0.2818f, 0.6310f, 3.0100f, 1.3700f, 1.2800f, 0.3548f, 0.0900f, { 0.0000f, 0.0000f, 0.0000f }, 0.1778f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10420.2002f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_EMPTYGRANDSTAND \
|
||||
{ 1.0000f, 1.0000f, 0.3162f, 1.0000f, 0.7943f, 4.6200f, 1.7500f, 1.4000f, 0.2082f, 0.0900f, { 0.0000f, 0.0000f, 0.0000f }, 0.2512f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10420.2002f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_DRIVING_TUNNEL \
|
||||
{ 1.0000f, 0.8100f, 0.3162f, 0.3981f, 0.8913f, 3.4200f, 0.9400f, 1.3100f, 0.7079f, 0.0510f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0470f, { 0.0000f, 0.0000f, 0.0000f }, 0.2140f, 0.0500f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 155.3000f, 0.0000f, 0x1 }
|
||||
|
||||
/* City Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_CITY_STREETS \
|
||||
{ 1.0000f, 0.7800f, 0.3162f, 0.7079f, 0.8913f, 1.7900f, 1.1200f, 0.9100f, 0.2818f, 0.0460f, { 0.0000f, 0.0000f, 0.0000f }, 0.1995f, 0.0280f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CITY_SUBWAY \
|
||||
{ 1.0000f, 0.7400f, 0.3162f, 0.7079f, 0.8913f, 3.0100f, 1.2300f, 0.9100f, 0.7079f, 0.0460f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0280f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 0.2100f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CITY_MUSEUM \
|
||||
{ 1.0000f, 0.8200f, 0.3162f, 0.1778f, 0.1778f, 3.2800f, 1.4000f, 0.5700f, 0.2512f, 0.0390f, { 0.0000f, 0.0000f, -0.0000f }, 0.8913f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 0.1300f, 0.1700f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CITY_LIBRARY \
|
||||
{ 1.0000f, 0.8200f, 0.3162f, 0.2818f, 0.0891f, 2.7600f, 0.8900f, 0.4100f, 0.3548f, 0.0290f, { 0.0000f, 0.0000f, -0.0000f }, 0.8913f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.1300f, 0.1700f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CITY_UNDERPASS \
|
||||
{ 1.0000f, 0.8200f, 0.3162f, 0.4467f, 0.8913f, 3.5700f, 1.1200f, 0.9100f, 0.3981f, 0.0590f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0370f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1400f, 0.2500f, 0.0000f, 0.9920f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CITY_ABANDONED \
|
||||
{ 1.0000f, 0.6900f, 0.3162f, 0.7943f, 0.8913f, 3.2800f, 1.1700f, 0.9100f, 0.4467f, 0.0440f, { 0.0000f, 0.0000f, 0.0000f }, 0.2818f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2000f, 0.2500f, 0.0000f, 0.9966f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
/* Misc. Presets */
|
||||
|
||||
#define EFX_REVERB_PRESET_DUSTYROOM \
|
||||
{ 0.3645f, 0.5600f, 0.3162f, 0.7943f, 0.7079f, 1.7900f, 0.3800f, 0.2100f, 0.5012f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0060f, { 0.0000f, 0.0000f, 0.0000f }, 0.2020f, 0.0500f, 0.2500f, 0.0000f, 0.9886f, 13046.0000f, 163.3000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_CHAPEL \
|
||||
{ 1.0000f, 0.8400f, 0.3162f, 0.5623f, 1.0000f, 4.6200f, 0.6400f, 1.2300f, 0.4467f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.1100f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
|
||||
|
||||
#define EFX_REVERB_PRESET_SMALLWATERROOM \
|
||||
{ 1.0000f, 0.7000f, 0.3162f, 0.4477f, 1.0000f, 1.5100f, 1.2500f, 1.1400f, 0.8913f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1790f, 0.1500f, 0.8950f, 0.1900f, 0.9920f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
|
||||
|
||||
#endif /* EFX_PRESETS_H */
|
||||
|
|
@ -0,0 +1,761 @@
|
|||
#ifndef AL_EFX_H
|
||||
#define AL_EFX_H
|
||||
|
||||
|
||||
#include "alc.h"
|
||||
#include "al.h"
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
#define ALC_EXT_EFX_NAME "ALC_EXT_EFX"
|
||||
|
||||
#define ALC_EFX_MAJOR_VERSION 0x20001
|
||||
#define ALC_EFX_MINOR_VERSION 0x20002
|
||||
#define ALC_MAX_AUXILIARY_SENDS 0x20003
|
||||
|
||||
|
||||
/* Listener properties. */
|
||||
#define AL_METERS_PER_UNIT 0x20004
|
||||
|
||||
/* Source properties. */
|
||||
#define AL_DIRECT_FILTER 0x20005
|
||||
#define AL_AUXILIARY_SEND_FILTER 0x20006
|
||||
#define AL_AIR_ABSORPTION_FACTOR 0x20007
|
||||
#define AL_ROOM_ROLLOFF_FACTOR 0x20008
|
||||
#define AL_CONE_OUTER_GAINHF 0x20009
|
||||
#define AL_DIRECT_FILTER_GAINHF_AUTO 0x2000A
|
||||
#define AL_AUXILIARY_SEND_FILTER_GAIN_AUTO 0x2000B
|
||||
#define AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO 0x2000C
|
||||
|
||||
|
||||
/* Effect properties. */
|
||||
|
||||
/* Reverb effect parameters */
|
||||
#define AL_REVERB_DENSITY 0x0001
|
||||
#define AL_REVERB_DIFFUSION 0x0002
|
||||
#define AL_REVERB_GAIN 0x0003
|
||||
#define AL_REVERB_GAINHF 0x0004
|
||||
#define AL_REVERB_DECAY_TIME 0x0005
|
||||
#define AL_REVERB_DECAY_HFRATIO 0x0006
|
||||
#define AL_REVERB_REFLECTIONS_GAIN 0x0007
|
||||
#define AL_REVERB_REFLECTIONS_DELAY 0x0008
|
||||
#define AL_REVERB_LATE_REVERB_GAIN 0x0009
|
||||
#define AL_REVERB_LATE_REVERB_DELAY 0x000A
|
||||
#define AL_REVERB_AIR_ABSORPTION_GAINHF 0x000B
|
||||
#define AL_REVERB_ROOM_ROLLOFF_FACTOR 0x000C
|
||||
#define AL_REVERB_DECAY_HFLIMIT 0x000D
|
||||
|
||||
/* EAX Reverb effect parameters */
|
||||
#define AL_EAXREVERB_DENSITY 0x0001
|
||||
#define AL_EAXREVERB_DIFFUSION 0x0002
|
||||
#define AL_EAXREVERB_GAIN 0x0003
|
||||
#define AL_EAXREVERB_GAINHF 0x0004
|
||||
#define AL_EAXREVERB_GAINLF 0x0005
|
||||
#define AL_EAXREVERB_DECAY_TIME 0x0006
|
||||
#define AL_EAXREVERB_DECAY_HFRATIO 0x0007
|
||||
#define AL_EAXREVERB_DECAY_LFRATIO 0x0008
|
||||
#define AL_EAXREVERB_REFLECTIONS_GAIN 0x0009
|
||||
#define AL_EAXREVERB_REFLECTIONS_DELAY 0x000A
|
||||
#define AL_EAXREVERB_REFLECTIONS_PAN 0x000B
|
||||
#define AL_EAXREVERB_LATE_REVERB_GAIN 0x000C
|
||||
#define AL_EAXREVERB_LATE_REVERB_DELAY 0x000D
|
||||
#define AL_EAXREVERB_LATE_REVERB_PAN 0x000E
|
||||
#define AL_EAXREVERB_ECHO_TIME 0x000F
|
||||
#define AL_EAXREVERB_ECHO_DEPTH 0x0010
|
||||
#define AL_EAXREVERB_MODULATION_TIME 0x0011
|
||||
#define AL_EAXREVERB_MODULATION_DEPTH 0x0012
|
||||
#define AL_EAXREVERB_AIR_ABSORPTION_GAINHF 0x0013
|
||||
#define AL_EAXREVERB_HFREFERENCE 0x0014
|
||||
#define AL_EAXREVERB_LFREFERENCE 0x0015
|
||||
#define AL_EAXREVERB_ROOM_ROLLOFF_FACTOR 0x0016
|
||||
#define AL_EAXREVERB_DECAY_HFLIMIT 0x0017
|
||||
|
||||
/* Chorus effect parameters */
|
||||
#define AL_CHORUS_WAVEFORM 0x0001
|
||||
#define AL_CHORUS_PHASE 0x0002
|
||||
#define AL_CHORUS_RATE 0x0003
|
||||
#define AL_CHORUS_DEPTH 0x0004
|
||||
#define AL_CHORUS_FEEDBACK 0x0005
|
||||
#define AL_CHORUS_DELAY 0x0006
|
||||
|
||||
/* Distortion effect parameters */
|
||||
#define AL_DISTORTION_EDGE 0x0001
|
||||
#define AL_DISTORTION_GAIN 0x0002
|
||||
#define AL_DISTORTION_LOWPASS_CUTOFF 0x0003
|
||||
#define AL_DISTORTION_EQCENTER 0x0004
|
||||
#define AL_DISTORTION_EQBANDWIDTH 0x0005
|
||||
|
||||
/* Echo effect parameters */
|
||||
#define AL_ECHO_DELAY 0x0001
|
||||
#define AL_ECHO_LRDELAY 0x0002
|
||||
#define AL_ECHO_DAMPING 0x0003
|
||||
#define AL_ECHO_FEEDBACK 0x0004
|
||||
#define AL_ECHO_SPREAD 0x0005
|
||||
|
||||
/* Flanger effect parameters */
|
||||
#define AL_FLANGER_WAVEFORM 0x0001
|
||||
#define AL_FLANGER_PHASE 0x0002
|
||||
#define AL_FLANGER_RATE 0x0003
|
||||
#define AL_FLANGER_DEPTH 0x0004
|
||||
#define AL_FLANGER_FEEDBACK 0x0005
|
||||
#define AL_FLANGER_DELAY 0x0006
|
||||
|
||||
/* Frequency shifter effect parameters */
|
||||
#define AL_FREQUENCY_SHIFTER_FREQUENCY 0x0001
|
||||
#define AL_FREQUENCY_SHIFTER_LEFT_DIRECTION 0x0002
|
||||
#define AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION 0x0003
|
||||
|
||||
/* Vocal morpher effect parameters */
|
||||
#define AL_VOCAL_MORPHER_PHONEMEA 0x0001
|
||||
#define AL_VOCAL_MORPHER_PHONEMEA_COARSE_TUNING 0x0002
|
||||
#define AL_VOCAL_MORPHER_PHONEMEB 0x0003
|
||||
#define AL_VOCAL_MORPHER_PHONEMEB_COARSE_TUNING 0x0004
|
||||
#define AL_VOCAL_MORPHER_WAVEFORM 0x0005
|
||||
#define AL_VOCAL_MORPHER_RATE 0x0006
|
||||
|
||||
/* Pitchshifter effect parameters */
|
||||
#define AL_PITCH_SHIFTER_COARSE_TUNE 0x0001
|
||||
#define AL_PITCH_SHIFTER_FINE_TUNE 0x0002
|
||||
|
||||
/* Ringmodulator effect parameters */
|
||||
#define AL_RING_MODULATOR_FREQUENCY 0x0001
|
||||
#define AL_RING_MODULATOR_HIGHPASS_CUTOFF 0x0002
|
||||
#define AL_RING_MODULATOR_WAVEFORM 0x0003
|
||||
|
||||
/* Autowah effect parameters */
|
||||
#define AL_AUTOWAH_ATTACK_TIME 0x0001
|
||||
#define AL_AUTOWAH_RELEASE_TIME 0x0002
|
||||
#define AL_AUTOWAH_RESONANCE 0x0003
|
||||
#define AL_AUTOWAH_PEAK_GAIN 0x0004
|
||||
|
||||
/* Compressor effect parameters */
|
||||
#define AL_COMPRESSOR_ONOFF 0x0001
|
||||
|
||||
/* Equalizer effect parameters */
|
||||
#define AL_EQUALIZER_LOW_GAIN 0x0001
|
||||
#define AL_EQUALIZER_LOW_CUTOFF 0x0002
|
||||
#define AL_EQUALIZER_MID1_GAIN 0x0003
|
||||
#define AL_EQUALIZER_MID1_CENTER 0x0004
|
||||
#define AL_EQUALIZER_MID1_WIDTH 0x0005
|
||||
#define AL_EQUALIZER_MID2_GAIN 0x0006
|
||||
#define AL_EQUALIZER_MID2_CENTER 0x0007
|
||||
#define AL_EQUALIZER_MID2_WIDTH 0x0008
|
||||
#define AL_EQUALIZER_HIGH_GAIN 0x0009
|
||||
#define AL_EQUALIZER_HIGH_CUTOFF 0x000A
|
||||
|
||||
/* Effect type */
|
||||
#define AL_EFFECT_FIRST_PARAMETER 0x0000
|
||||
#define AL_EFFECT_LAST_PARAMETER 0x8000
|
||||
#define AL_EFFECT_TYPE 0x8001
|
||||
|
||||
/* Effect types, used with the AL_EFFECT_TYPE property */
|
||||
#define AL_EFFECT_NULL 0x0000
|
||||
#define AL_EFFECT_REVERB 0x0001
|
||||
#define AL_EFFECT_CHORUS 0x0002
|
||||
#define AL_EFFECT_DISTORTION 0x0003
|
||||
#define AL_EFFECT_ECHO 0x0004
|
||||
#define AL_EFFECT_FLANGER 0x0005
|
||||
#define AL_EFFECT_FREQUENCY_SHIFTER 0x0006
|
||||
#define AL_EFFECT_VOCAL_MORPHER 0x0007
|
||||
#define AL_EFFECT_PITCH_SHIFTER 0x0008
|
||||
#define AL_EFFECT_RING_MODULATOR 0x0009
|
||||
#define AL_EFFECT_AUTOWAH 0x000A
|
||||
#define AL_EFFECT_COMPRESSOR 0x000B
|
||||
#define AL_EFFECT_EQUALIZER 0x000C
|
||||
#define AL_EFFECT_EAXREVERB 0x8000
|
||||
|
||||
/* Auxiliary Effect Slot properties. */
|
||||
#define AL_EFFECTSLOT_EFFECT 0x0001
|
||||
#define AL_EFFECTSLOT_GAIN 0x0002
|
||||
#define AL_EFFECTSLOT_AUXILIARY_SEND_AUTO 0x0003
|
||||
|
||||
/* NULL Auxiliary Slot ID to disable a source send. */
|
||||
#define AL_EFFECTSLOT_NULL 0x0000
|
||||
|
||||
|
||||
/* Filter properties. */
|
||||
|
||||
/* Lowpass filter parameters */
|
||||
#define AL_LOWPASS_GAIN 0x0001
|
||||
#define AL_LOWPASS_GAINHF 0x0002
|
||||
|
||||
/* Highpass filter parameters */
|
||||
#define AL_HIGHPASS_GAIN 0x0001
|
||||
#define AL_HIGHPASS_GAINLF 0x0002
|
||||
|
||||
/* Bandpass filter parameters */
|
||||
#define AL_BANDPASS_GAIN 0x0001
|
||||
#define AL_BANDPASS_GAINLF 0x0002
|
||||
#define AL_BANDPASS_GAINHF 0x0003
|
||||
|
||||
/* Filter type */
|
||||
#define AL_FILTER_FIRST_PARAMETER 0x0000
|
||||
#define AL_FILTER_LAST_PARAMETER 0x8000
|
||||
#define AL_FILTER_TYPE 0x8001
|
||||
|
||||
/* Filter types, used with the AL_FILTER_TYPE property */
|
||||
#define AL_FILTER_NULL 0x0000
|
||||
#define AL_FILTER_LOWPASS 0x0001
|
||||
#define AL_FILTER_HIGHPASS 0x0002
|
||||
#define AL_FILTER_BANDPASS 0x0003
|
||||
|
||||
|
||||
/* Effect object function types. */
|
||||
typedef void (AL_APIENTRY *LPALGENEFFECTS)(ALsizei, ALuint*);
|
||||
typedef void (AL_APIENTRY *LPALDELETEEFFECTS)(ALsizei, const ALuint*);
|
||||
typedef ALboolean (AL_APIENTRY *LPALISEFFECT)(ALuint);
|
||||
typedef void (AL_APIENTRY *LPALEFFECTI)(ALuint, ALenum, ALint);
|
||||
typedef void (AL_APIENTRY *LPALEFFECTIV)(ALuint, ALenum, const ALint*);
|
||||
typedef void (AL_APIENTRY *LPALEFFECTF)(ALuint, ALenum, ALfloat);
|
||||
typedef void (AL_APIENTRY *LPALEFFECTFV)(ALuint, ALenum, const ALfloat*);
|
||||
typedef void (AL_APIENTRY *LPALGETEFFECTI)(ALuint, ALenum, ALint*);
|
||||
typedef void (AL_APIENTRY *LPALGETEFFECTIV)(ALuint, ALenum, ALint*);
|
||||
typedef void (AL_APIENTRY *LPALGETEFFECTF)(ALuint, ALenum, ALfloat*);
|
||||
typedef void (AL_APIENTRY *LPALGETEFFECTFV)(ALuint, ALenum, ALfloat*);
|
||||
|
||||
/* Filter object function types. */
|
||||
typedef void (AL_APIENTRY *LPALGENFILTERS)(ALsizei, ALuint*);
|
||||
typedef void (AL_APIENTRY *LPALDELETEFILTERS)(ALsizei, const ALuint*);
|
||||
typedef ALboolean (AL_APIENTRY *LPALISFILTER)(ALuint);
|
||||
typedef void (AL_APIENTRY *LPALFILTERI)(ALuint, ALenum, ALint);
|
||||
typedef void (AL_APIENTRY *LPALFILTERIV)(ALuint, ALenum, const ALint*);
|
||||
typedef void (AL_APIENTRY *LPALFILTERF)(ALuint, ALenum, ALfloat);
|
||||
typedef void (AL_APIENTRY *LPALFILTERFV)(ALuint, ALenum, const ALfloat*);
|
||||
typedef void (AL_APIENTRY *LPALGETFILTERI)(ALuint, ALenum, ALint*);
|
||||
typedef void (AL_APIENTRY *LPALGETFILTERIV)(ALuint, ALenum, ALint*);
|
||||
typedef void (AL_APIENTRY *LPALGETFILTERF)(ALuint, ALenum, ALfloat*);
|
||||
typedef void (AL_APIENTRY *LPALGETFILTERFV)(ALuint, ALenum, ALfloat*);
|
||||
|
||||
/* Auxiliary Effect Slot object function types. */
|
||||
typedef void (AL_APIENTRY *LPALGENAUXILIARYEFFECTSLOTS)(ALsizei, ALuint*);
|
||||
typedef void (AL_APIENTRY *LPALDELETEAUXILIARYEFFECTSLOTS)(ALsizei, const ALuint*);
|
||||
typedef ALboolean (AL_APIENTRY *LPALISAUXILIARYEFFECTSLOT)(ALuint);
|
||||
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTI)(ALuint, ALenum, ALint);
|
||||
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTIV)(ALuint, ALenum, const ALint*);
|
||||
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTF)(ALuint, ALenum, ALfloat);
|
||||
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTFV)(ALuint, ALenum, const ALfloat*);
|
||||
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTI)(ALuint, ALenum, ALint*);
|
||||
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTIV)(ALuint, ALenum, ALint*);
|
||||
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTF)(ALuint, ALenum, ALfloat*);
|
||||
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTFV)(ALuint, ALenum, ALfloat*);
|
||||
|
||||
#ifdef AL_ALEXT_PROTOTYPES
|
||||
AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects);
|
||||
AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects);
|
||||
AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect);
|
||||
AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint iValue);
|
||||
AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *piValues);
|
||||
AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat flValue);
|
||||
AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *pflValues);
|
||||
AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *piValue);
|
||||
AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *piValues);
|
||||
AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *pflValue);
|
||||
AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *pflValues);
|
||||
|
||||
AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters);
|
||||
AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters);
|
||||
AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter);
|
||||
AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint iValue);
|
||||
AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *piValues);
|
||||
AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat flValue);
|
||||
AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *pflValues);
|
||||
AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *piValue);
|
||||
AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *piValues);
|
||||
AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *pflValue);
|
||||
AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *pflValues);
|
||||
|
||||
AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots);
|
||||
AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots);
|
||||
AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot);
|
||||
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint iValue);
|
||||
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *piValues);
|
||||
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat flValue);
|
||||
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *pflValues);
|
||||
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *piValue);
|
||||
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *piValues);
|
||||
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *pflValue);
|
||||
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *pflValues);
|
||||
#endif
|
||||
|
||||
/* Filter ranges and defaults. */
|
||||
|
||||
/* Lowpass filter */
|
||||
#define AL_LOWPASS_MIN_GAIN (0.0f)
|
||||
#define AL_LOWPASS_MAX_GAIN (1.0f)
|
||||
#define AL_LOWPASS_DEFAULT_GAIN (1.0f)
|
||||
|
||||
#define AL_LOWPASS_MIN_GAINHF (0.0f)
|
||||
#define AL_LOWPASS_MAX_GAINHF (1.0f)
|
||||
#define AL_LOWPASS_DEFAULT_GAINHF (1.0f)
|
||||
|
||||
/* Highpass filter */
|
||||
#define AL_HIGHPASS_MIN_GAIN (0.0f)
|
||||
#define AL_HIGHPASS_MAX_GAIN (1.0f)
|
||||
#define AL_HIGHPASS_DEFAULT_GAIN (1.0f)
|
||||
|
||||
#define AL_HIGHPASS_MIN_GAINLF (0.0f)
|
||||
#define AL_HIGHPASS_MAX_GAINLF (1.0f)
|
||||
#define AL_HIGHPASS_DEFAULT_GAINLF (1.0f)
|
||||
|
||||
/* Bandpass filter */
|
||||
#define AL_BANDPASS_MIN_GAIN (0.0f)
|
||||
#define AL_BANDPASS_MAX_GAIN (1.0f)
|
||||
#define AL_BANDPASS_DEFAULT_GAIN (1.0f)
|
||||
|
||||
#define AL_BANDPASS_MIN_GAINHF (0.0f)
|
||||
#define AL_BANDPASS_MAX_GAINHF (1.0f)
|
||||
#define AL_BANDPASS_DEFAULT_GAINHF (1.0f)
|
||||
|
||||
#define AL_BANDPASS_MIN_GAINLF (0.0f)
|
||||
#define AL_BANDPASS_MAX_GAINLF (1.0f)
|
||||
#define AL_BANDPASS_DEFAULT_GAINLF (1.0f)
|
||||
|
||||
|
||||
/* Effect parameter ranges and defaults. */
|
||||
|
||||
/* Standard reverb effect */
|
||||
#define AL_REVERB_MIN_DENSITY (0.0f)
|
||||
#define AL_REVERB_MAX_DENSITY (1.0f)
|
||||
#define AL_REVERB_DEFAULT_DENSITY (1.0f)
|
||||
|
||||
#define AL_REVERB_MIN_DIFFUSION (0.0f)
|
||||
#define AL_REVERB_MAX_DIFFUSION (1.0f)
|
||||
#define AL_REVERB_DEFAULT_DIFFUSION (1.0f)
|
||||
|
||||
#define AL_REVERB_MIN_GAIN (0.0f)
|
||||
#define AL_REVERB_MAX_GAIN (1.0f)
|
||||
#define AL_REVERB_DEFAULT_GAIN (0.32f)
|
||||
|
||||
#define AL_REVERB_MIN_GAINHF (0.0f)
|
||||
#define AL_REVERB_MAX_GAINHF (1.0f)
|
||||
#define AL_REVERB_DEFAULT_GAINHF (0.89f)
|
||||
|
||||
#define AL_REVERB_MIN_DECAY_TIME (0.1f)
|
||||
#define AL_REVERB_MAX_DECAY_TIME (20.0f)
|
||||
#define AL_REVERB_DEFAULT_DECAY_TIME (1.49f)
|
||||
|
||||
#define AL_REVERB_MIN_DECAY_HFRATIO (0.1f)
|
||||
#define AL_REVERB_MAX_DECAY_HFRATIO (2.0f)
|
||||
#define AL_REVERB_DEFAULT_DECAY_HFRATIO (0.83f)
|
||||
|
||||
#define AL_REVERB_MIN_REFLECTIONS_GAIN (0.0f)
|
||||
#define AL_REVERB_MAX_REFLECTIONS_GAIN (3.16f)
|
||||
#define AL_REVERB_DEFAULT_REFLECTIONS_GAIN (0.05f)
|
||||
|
||||
#define AL_REVERB_MIN_REFLECTIONS_DELAY (0.0f)
|
||||
#define AL_REVERB_MAX_REFLECTIONS_DELAY (0.3f)
|
||||
#define AL_REVERB_DEFAULT_REFLECTIONS_DELAY (0.007f)
|
||||
|
||||
#define AL_REVERB_MIN_LATE_REVERB_GAIN (0.0f)
|
||||
#define AL_REVERB_MAX_LATE_REVERB_GAIN (10.0f)
|
||||
#define AL_REVERB_DEFAULT_LATE_REVERB_GAIN (1.26f)
|
||||
|
||||
#define AL_REVERB_MIN_LATE_REVERB_DELAY (0.0f)
|
||||
#define AL_REVERB_MAX_LATE_REVERB_DELAY (0.1f)
|
||||
#define AL_REVERB_DEFAULT_LATE_REVERB_DELAY (0.011f)
|
||||
|
||||
#define AL_REVERB_MIN_AIR_ABSORPTION_GAINHF (0.892f)
|
||||
#define AL_REVERB_MAX_AIR_ABSORPTION_GAINHF (1.0f)
|
||||
#define AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF (0.994f)
|
||||
|
||||
#define AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR (0.0f)
|
||||
#define AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR (10.0f)
|
||||
#define AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f)
|
||||
|
||||
#define AL_REVERB_MIN_DECAY_HFLIMIT AL_FALSE
|
||||
#define AL_REVERB_MAX_DECAY_HFLIMIT AL_TRUE
|
||||
#define AL_REVERB_DEFAULT_DECAY_HFLIMIT AL_TRUE
|
||||
|
||||
/* EAX reverb effect */
|
||||
#define AL_EAXREVERB_MIN_DENSITY (0.0f)
|
||||
#define AL_EAXREVERB_MAX_DENSITY (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_DENSITY (1.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_DIFFUSION (0.0f)
|
||||
#define AL_EAXREVERB_MAX_DIFFUSION (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_DIFFUSION (1.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_GAIN (0.0f)
|
||||
#define AL_EAXREVERB_MAX_GAIN (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_GAIN (0.32f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_GAINHF (0.0f)
|
||||
#define AL_EAXREVERB_MAX_GAINHF (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_GAINHF (0.89f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_GAINLF (0.0f)
|
||||
#define AL_EAXREVERB_MAX_GAINLF (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_GAINLF (1.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_DECAY_TIME (0.1f)
|
||||
#define AL_EAXREVERB_MAX_DECAY_TIME (20.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_DECAY_TIME (1.49f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_DECAY_HFRATIO (0.1f)
|
||||
#define AL_EAXREVERB_MAX_DECAY_HFRATIO (2.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_DECAY_HFRATIO (0.83f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_DECAY_LFRATIO (0.1f)
|
||||
#define AL_EAXREVERB_MAX_DECAY_LFRATIO (2.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_DECAY_LFRATIO (1.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_REFLECTIONS_GAIN (0.0f)
|
||||
#define AL_EAXREVERB_MAX_REFLECTIONS_GAIN (3.16f)
|
||||
#define AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN (0.05f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_REFLECTIONS_DELAY (0.0f)
|
||||
#define AL_EAXREVERB_MAX_REFLECTIONS_DELAY (0.3f)
|
||||
#define AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY (0.007f)
|
||||
|
||||
#define AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ (0.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_LATE_REVERB_GAIN (0.0f)
|
||||
#define AL_EAXREVERB_MAX_LATE_REVERB_GAIN (10.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN (1.26f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_LATE_REVERB_DELAY (0.0f)
|
||||
#define AL_EAXREVERB_MAX_LATE_REVERB_DELAY (0.1f)
|
||||
#define AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY (0.011f)
|
||||
|
||||
#define AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ (0.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_ECHO_TIME (0.075f)
|
||||
#define AL_EAXREVERB_MAX_ECHO_TIME (0.25f)
|
||||
#define AL_EAXREVERB_DEFAULT_ECHO_TIME (0.25f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_ECHO_DEPTH (0.0f)
|
||||
#define AL_EAXREVERB_MAX_ECHO_DEPTH (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_ECHO_DEPTH (0.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_MODULATION_TIME (0.04f)
|
||||
#define AL_EAXREVERB_MAX_MODULATION_TIME (4.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_MODULATION_TIME (0.25f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_MODULATION_DEPTH (0.0f)
|
||||
#define AL_EAXREVERB_MAX_MODULATION_DEPTH (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_MODULATION_DEPTH (0.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF (0.892f)
|
||||
#define AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF (1.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF (0.994f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_HFREFERENCE (1000.0f)
|
||||
#define AL_EAXREVERB_MAX_HFREFERENCE (20000.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_HFREFERENCE (5000.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_LFREFERENCE (20.0f)
|
||||
#define AL_EAXREVERB_MAX_LFREFERENCE (1000.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_LFREFERENCE (250.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR (0.0f)
|
||||
#define AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR (10.0f)
|
||||
#define AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f)
|
||||
|
||||
#define AL_EAXREVERB_MIN_DECAY_HFLIMIT AL_FALSE
|
||||
#define AL_EAXREVERB_MAX_DECAY_HFLIMIT AL_TRUE
|
||||
#define AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT AL_TRUE
|
||||
|
||||
/* Chorus effect */
|
||||
#define AL_CHORUS_WAVEFORM_SINUSOID (0)
|
||||
#define AL_CHORUS_WAVEFORM_TRIANGLE (1)
|
||||
|
||||
#define AL_CHORUS_MIN_WAVEFORM (0)
|
||||
#define AL_CHORUS_MAX_WAVEFORM (1)
|
||||
#define AL_CHORUS_DEFAULT_WAVEFORM (1)
|
||||
|
||||
#define AL_CHORUS_MIN_PHASE (-180)
|
||||
#define AL_CHORUS_MAX_PHASE (180)
|
||||
#define AL_CHORUS_DEFAULT_PHASE (90)
|
||||
|
||||
#define AL_CHORUS_MIN_RATE (0.0f)
|
||||
#define AL_CHORUS_MAX_RATE (10.0f)
|
||||
#define AL_CHORUS_DEFAULT_RATE (1.1f)
|
||||
|
||||
#define AL_CHORUS_MIN_DEPTH (0.0f)
|
||||
#define AL_CHORUS_MAX_DEPTH (1.0f)
|
||||
#define AL_CHORUS_DEFAULT_DEPTH (0.1f)
|
||||
|
||||
#define AL_CHORUS_MIN_FEEDBACK (-1.0f)
|
||||
#define AL_CHORUS_MAX_FEEDBACK (1.0f)
|
||||
#define AL_CHORUS_DEFAULT_FEEDBACK (0.25f)
|
||||
|
||||
#define AL_CHORUS_MIN_DELAY (0.0f)
|
||||
#define AL_CHORUS_MAX_DELAY (0.016f)
|
||||
#define AL_CHORUS_DEFAULT_DELAY (0.016f)
|
||||
|
||||
/* Distortion effect */
|
||||
#define AL_DISTORTION_MIN_EDGE (0.0f)
|
||||
#define AL_DISTORTION_MAX_EDGE (1.0f)
|
||||
#define AL_DISTORTION_DEFAULT_EDGE (0.2f)
|
||||
|
||||
#define AL_DISTORTION_MIN_GAIN (0.01f)
|
||||
#define AL_DISTORTION_MAX_GAIN (1.0f)
|
||||
#define AL_DISTORTION_DEFAULT_GAIN (0.05f)
|
||||
|
||||
#define AL_DISTORTION_MIN_LOWPASS_CUTOFF (80.0f)
|
||||
#define AL_DISTORTION_MAX_LOWPASS_CUTOFF (24000.0f)
|
||||
#define AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF (8000.0f)
|
||||
|
||||
#define AL_DISTORTION_MIN_EQCENTER (80.0f)
|
||||
#define AL_DISTORTION_MAX_EQCENTER (24000.0f)
|
||||
#define AL_DISTORTION_DEFAULT_EQCENTER (3600.0f)
|
||||
|
||||
#define AL_DISTORTION_MIN_EQBANDWIDTH (80.0f)
|
||||
#define AL_DISTORTION_MAX_EQBANDWIDTH (24000.0f)
|
||||
#define AL_DISTORTION_DEFAULT_EQBANDWIDTH (3600.0f)
|
||||
|
||||
/* Echo effect */
|
||||
#define AL_ECHO_MIN_DELAY (0.0f)
|
||||
#define AL_ECHO_MAX_DELAY (0.207f)
|
||||
#define AL_ECHO_DEFAULT_DELAY (0.1f)
|
||||
|
||||
#define AL_ECHO_MIN_LRDELAY (0.0f)
|
||||
#define AL_ECHO_MAX_LRDELAY (0.404f)
|
||||
#define AL_ECHO_DEFAULT_LRDELAY (0.1f)
|
||||
|
||||
#define AL_ECHO_MIN_DAMPING (0.0f)
|
||||
#define AL_ECHO_MAX_DAMPING (0.99f)
|
||||
#define AL_ECHO_DEFAULT_DAMPING (0.5f)
|
||||
|
||||
#define AL_ECHO_MIN_FEEDBACK (0.0f)
|
||||
#define AL_ECHO_MAX_FEEDBACK (1.0f)
|
||||
#define AL_ECHO_DEFAULT_FEEDBACK (0.5f)
|
||||
|
||||
#define AL_ECHO_MIN_SPREAD (-1.0f)
|
||||
#define AL_ECHO_MAX_SPREAD (1.0f)
|
||||
#define AL_ECHO_DEFAULT_SPREAD (-1.0f)
|
||||
|
||||
/* Flanger effect */
|
||||
#define AL_FLANGER_WAVEFORM_SINUSOID (0)
|
||||
#define AL_FLANGER_WAVEFORM_TRIANGLE (1)
|
||||
|
||||
#define AL_FLANGER_MIN_WAVEFORM (0)
|
||||
#define AL_FLANGER_MAX_WAVEFORM (1)
|
||||
#define AL_FLANGER_DEFAULT_WAVEFORM (1)
|
||||
|
||||
#define AL_FLANGER_MIN_PHASE (-180)
|
||||
#define AL_FLANGER_MAX_PHASE (180)
|
||||
#define AL_FLANGER_DEFAULT_PHASE (0)
|
||||
|
||||
#define AL_FLANGER_MIN_RATE (0.0f)
|
||||
#define AL_FLANGER_MAX_RATE (10.0f)
|
||||
#define AL_FLANGER_DEFAULT_RATE (0.27f)
|
||||
|
||||
#define AL_FLANGER_MIN_DEPTH (0.0f)
|
||||
#define AL_FLANGER_MAX_DEPTH (1.0f)
|
||||
#define AL_FLANGER_DEFAULT_DEPTH (1.0f)
|
||||
|
||||
#define AL_FLANGER_MIN_FEEDBACK (-1.0f)
|
||||
#define AL_FLANGER_MAX_FEEDBACK (1.0f)
|
||||
#define AL_FLANGER_DEFAULT_FEEDBACK (-0.5f)
|
||||
|
||||
#define AL_FLANGER_MIN_DELAY (0.0f)
|
||||
#define AL_FLANGER_MAX_DELAY (0.004f)
|
||||
#define AL_FLANGER_DEFAULT_DELAY (0.002f)
|
||||
|
||||
/* Frequency shifter effect */
|
||||
#define AL_FREQUENCY_SHIFTER_MIN_FREQUENCY (0.0f)
|
||||
#define AL_FREQUENCY_SHIFTER_MAX_FREQUENCY (24000.0f)
|
||||
#define AL_FREQUENCY_SHIFTER_DEFAULT_FREQUENCY (0.0f)
|
||||
|
||||
#define AL_FREQUENCY_SHIFTER_MIN_LEFT_DIRECTION (0)
|
||||
#define AL_FREQUENCY_SHIFTER_MAX_LEFT_DIRECTION (2)
|
||||
#define AL_FREQUENCY_SHIFTER_DEFAULT_LEFT_DIRECTION (0)
|
||||
|
||||
#define AL_FREQUENCY_SHIFTER_DIRECTION_DOWN (0)
|
||||
#define AL_FREQUENCY_SHIFTER_DIRECTION_UP (1)
|
||||
#define AL_FREQUENCY_SHIFTER_DIRECTION_OFF (2)
|
||||
|
||||
#define AL_FREQUENCY_SHIFTER_MIN_RIGHT_DIRECTION (0)
|
||||
#define AL_FREQUENCY_SHIFTER_MAX_RIGHT_DIRECTION (2)
|
||||
#define AL_FREQUENCY_SHIFTER_DEFAULT_RIGHT_DIRECTION (0)
|
||||
|
||||
/* Vocal morpher effect */
|
||||
#define AL_VOCAL_MORPHER_MIN_PHONEMEA (0)
|
||||
#define AL_VOCAL_MORPHER_MAX_PHONEMEA (29)
|
||||
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEA (0)
|
||||
|
||||
#define AL_VOCAL_MORPHER_MIN_PHONEMEA_COARSE_TUNING (-24)
|
||||
#define AL_VOCAL_MORPHER_MAX_PHONEMEA_COARSE_TUNING (24)
|
||||
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEA_COARSE_TUNING (0)
|
||||
|
||||
#define AL_VOCAL_MORPHER_MIN_PHONEMEB (0)
|
||||
#define AL_VOCAL_MORPHER_MAX_PHONEMEB (29)
|
||||
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEB (10)
|
||||
|
||||
#define AL_VOCAL_MORPHER_MIN_PHONEMEB_COARSE_TUNING (-24)
|
||||
#define AL_VOCAL_MORPHER_MAX_PHONEMEB_COARSE_TUNING (24)
|
||||
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEB_COARSE_TUNING (0)
|
||||
|
||||
#define AL_VOCAL_MORPHER_PHONEME_A (0)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_E (1)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_I (2)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_O (3)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_U (4)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_AA (5)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_AE (6)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_AH (7)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_AO (8)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_EH (9)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_ER (10)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_IH (11)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_IY (12)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_UH (13)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_UW (14)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_B (15)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_D (16)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_F (17)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_G (18)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_J (19)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_K (20)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_L (21)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_M (22)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_N (23)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_P (24)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_R (25)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_S (26)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_T (27)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_V (28)
|
||||
#define AL_VOCAL_MORPHER_PHONEME_Z (29)
|
||||
|
||||
#define AL_VOCAL_MORPHER_WAVEFORM_SINUSOID (0)
|
||||
#define AL_VOCAL_MORPHER_WAVEFORM_TRIANGLE (1)
|
||||
#define AL_VOCAL_MORPHER_WAVEFORM_SAWTOOTH (2)
|
||||
|
||||
#define AL_VOCAL_MORPHER_MIN_WAVEFORM (0)
|
||||
#define AL_VOCAL_MORPHER_MAX_WAVEFORM (2)
|
||||
#define AL_VOCAL_MORPHER_DEFAULT_WAVEFORM (0)
|
||||
|
||||
#define AL_VOCAL_MORPHER_MIN_RATE (0.0f)
|
||||
#define AL_VOCAL_MORPHER_MAX_RATE (10.0f)
|
||||
#define AL_VOCAL_MORPHER_DEFAULT_RATE (1.41f)
|
||||
|
||||
/* Pitch shifter effect */
|
||||
#define AL_PITCH_SHIFTER_MIN_COARSE_TUNE (-12)
|
||||
#define AL_PITCH_SHIFTER_MAX_COARSE_TUNE (12)
|
||||
#define AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE (12)
|
||||
|
||||
#define AL_PITCH_SHIFTER_MIN_FINE_TUNE (-50)
|
||||
#define AL_PITCH_SHIFTER_MAX_FINE_TUNE (50)
|
||||
#define AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE (0)
|
||||
|
||||
/* Ring modulator effect */
|
||||
#define AL_RING_MODULATOR_MIN_FREQUENCY (0.0f)
|
||||
#define AL_RING_MODULATOR_MAX_FREQUENCY (8000.0f)
|
||||
#define AL_RING_MODULATOR_DEFAULT_FREQUENCY (440.0f)
|
||||
|
||||
#define AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF (0.0f)
|
||||
#define AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF (24000.0f)
|
||||
#define AL_RING_MODULATOR_DEFAULT_HIGHPASS_CUTOFF (800.0f)
|
||||
|
||||
#define AL_RING_MODULATOR_SINUSOID (0)
|
||||
#define AL_RING_MODULATOR_SAWTOOTH (1)
|
||||
#define AL_RING_MODULATOR_SQUARE (2)
|
||||
|
||||
#define AL_RING_MODULATOR_MIN_WAVEFORM (0)
|
||||
#define AL_RING_MODULATOR_MAX_WAVEFORM (2)
|
||||
#define AL_RING_MODULATOR_DEFAULT_WAVEFORM (0)
|
||||
|
||||
/* Autowah effect */
|
||||
#define AL_AUTOWAH_MIN_ATTACK_TIME (0.0001f)
|
||||
#define AL_AUTOWAH_MAX_ATTACK_TIME (1.0f)
|
||||
#define AL_AUTOWAH_DEFAULT_ATTACK_TIME (0.06f)
|
||||
|
||||
#define AL_AUTOWAH_MIN_RELEASE_TIME (0.0001f)
|
||||
#define AL_AUTOWAH_MAX_RELEASE_TIME (1.0f)
|
||||
#define AL_AUTOWAH_DEFAULT_RELEASE_TIME (0.06f)
|
||||
|
||||
#define AL_AUTOWAH_MIN_RESONANCE (2.0f)
|
||||
#define AL_AUTOWAH_MAX_RESONANCE (1000.0f)
|
||||
#define AL_AUTOWAH_DEFAULT_RESONANCE (1000.0f)
|
||||
|
||||
#define AL_AUTOWAH_MIN_PEAK_GAIN (0.00003f)
|
||||
#define AL_AUTOWAH_MAX_PEAK_GAIN (31621.0f)
|
||||
#define AL_AUTOWAH_DEFAULT_PEAK_GAIN (11.22f)
|
||||
|
||||
/* Compressor effect */
|
||||
#define AL_COMPRESSOR_MIN_ONOFF (0)
|
||||
#define AL_COMPRESSOR_MAX_ONOFF (1)
|
||||
#define AL_COMPRESSOR_DEFAULT_ONOFF (1)
|
||||
|
||||
/* Equalizer effect */
|
||||
#define AL_EQUALIZER_MIN_LOW_GAIN (0.126f)
|
||||
#define AL_EQUALIZER_MAX_LOW_GAIN (7.943f)
|
||||
#define AL_EQUALIZER_DEFAULT_LOW_GAIN (1.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_LOW_CUTOFF (50.0f)
|
||||
#define AL_EQUALIZER_MAX_LOW_CUTOFF (800.0f)
|
||||
#define AL_EQUALIZER_DEFAULT_LOW_CUTOFF (200.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_MID1_GAIN (0.126f)
|
||||
#define AL_EQUALIZER_MAX_MID1_GAIN (7.943f)
|
||||
#define AL_EQUALIZER_DEFAULT_MID1_GAIN (1.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_MID1_CENTER (200.0f)
|
||||
#define AL_EQUALIZER_MAX_MID1_CENTER (3000.0f)
|
||||
#define AL_EQUALIZER_DEFAULT_MID1_CENTER (500.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_MID1_WIDTH (0.01f)
|
||||
#define AL_EQUALIZER_MAX_MID1_WIDTH (1.0f)
|
||||
#define AL_EQUALIZER_DEFAULT_MID1_WIDTH (1.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_MID2_GAIN (0.126f)
|
||||
#define AL_EQUALIZER_MAX_MID2_GAIN (7.943f)
|
||||
#define AL_EQUALIZER_DEFAULT_MID2_GAIN (1.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_MID2_CENTER (1000.0f)
|
||||
#define AL_EQUALIZER_MAX_MID2_CENTER (8000.0f)
|
||||
#define AL_EQUALIZER_DEFAULT_MID2_CENTER (3000.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_MID2_WIDTH (0.01f)
|
||||
#define AL_EQUALIZER_MAX_MID2_WIDTH (1.0f)
|
||||
#define AL_EQUALIZER_DEFAULT_MID2_WIDTH (1.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_HIGH_GAIN (0.126f)
|
||||
#define AL_EQUALIZER_MAX_HIGH_GAIN (7.943f)
|
||||
#define AL_EQUALIZER_DEFAULT_HIGH_GAIN (1.0f)
|
||||
|
||||
#define AL_EQUALIZER_MIN_HIGH_CUTOFF (4000.0f)
|
||||
#define AL_EQUALIZER_MAX_HIGH_CUTOFF (16000.0f)
|
||||
#define AL_EQUALIZER_DEFAULT_HIGH_CUTOFF (6000.0f)
|
||||
|
||||
|
||||
/* Source parameter value ranges and defaults. */
|
||||
#define AL_MIN_AIR_ABSORPTION_FACTOR (0.0f)
|
||||
#define AL_MAX_AIR_ABSORPTION_FACTOR (10.0f)
|
||||
#define AL_DEFAULT_AIR_ABSORPTION_FACTOR (0.0f)
|
||||
|
||||
#define AL_MIN_ROOM_ROLLOFF_FACTOR (0.0f)
|
||||
#define AL_MAX_ROOM_ROLLOFF_FACTOR (10.0f)
|
||||
#define AL_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f)
|
||||
|
||||
#define AL_MIN_CONE_OUTER_GAINHF (0.0f)
|
||||
#define AL_MAX_CONE_OUTER_GAINHF (1.0f)
|
||||
#define AL_DEFAULT_CONE_OUTER_GAINHF (1.0f)
|
||||
|
||||
#define AL_MIN_DIRECT_FILTER_GAINHF_AUTO AL_FALSE
|
||||
#define AL_MAX_DIRECT_FILTER_GAINHF_AUTO AL_TRUE
|
||||
#define AL_DEFAULT_DIRECT_FILTER_GAINHF_AUTO AL_TRUE
|
||||
|
||||
#define AL_MIN_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_FALSE
|
||||
#define AL_MAX_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_TRUE
|
||||
#define AL_DEFAULT_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_TRUE
|
||||
|
||||
#define AL_MIN_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_FALSE
|
||||
#define AL_MAX_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_TRUE
|
||||
#define AL_DEFAULT_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_TRUE
|
||||
|
||||
|
||||
/* Listener parameter value ranges and defaults. */
|
||||
#define AL_MIN_METERS_PER_UNIT FLT_MIN
|
||||
#define AL_MAX_METERS_PER_UNIT FLT_MAX
|
||||
#define AL_DEFAULT_METERS_PER_UNIT (1.0f)
|
||||
|
||||
|
||||
#ifdef __cplusplus
|
||||
} /* extern "C" */
|
||||
#endif
|
||||
|
||||
#endif /* AL_EFX_H */
|
||||
|
|
@ -0,0 +1,94 @@
|
|||
#include <al.h>
|
||||
|
||||
// X-RAM Function pointer definitions
|
||||
typedef ALboolean (__cdecl *EAXSetBufferMode)(ALsizei n, ALuint *buffers, ALint value);
|
||||
typedef ALenum (__cdecl *EAXGetBufferMode)(ALuint buffer, ALint *value);
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// Query for X-RAM extension
|
||||
//
|
||||
// if (alIsExtensionPresent("EAX-RAM") == AL_TRUE)
|
||||
// X-RAM Extension found
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// X-RAM enum names
|
||||
//
|
||||
// "AL_EAX_RAM_SIZE"
|
||||
// "AL_EAX_RAM_FREE"
|
||||
// "AL_STORAGE_AUTOMATIC"
|
||||
// "AL_STORAGE_HARDWARE"
|
||||
// "AL_STORAGE_ACCESSIBLE"
|
||||
//
|
||||
// Query enum values using alGetEnumValue, for example
|
||||
//
|
||||
// long lRamSizeEnum = alGetEnumValue("AL_EAX_RAM_SIZE")
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// Query total amount of X-RAM
|
||||
//
|
||||
// long lTotalSize = alGetInteger(alGetEnumValue("AL_EAX_RAM_SIZE")
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// Query free X-RAM available
|
||||
//
|
||||
// long lFreeSize = alGetInteger(alGetEnumValue("AL_EAX_RAM_FREE")
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// Query X-RAM Function pointers
|
||||
//
|
||||
// Use typedefs defined above to get the X-RAM function pointers using
|
||||
// alGetProcAddress
|
||||
//
|
||||
// EAXSetBufferMode eaxSetBufferMode;
|
||||
// EAXGetBufferMode eaxGetBufferMode;
|
||||
//
|
||||
// eaxSetBufferMode = (EAXSetBufferMode)alGetProcAddress("EAXSetBufferMode");
|
||||
// eaxGetBufferMode = (EAXGetBufferMode)alGetProcAddress("EAXGetBufferMode");
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// Force an Open AL Buffer into X-RAM (good for non-streaming buffers)
|
||||
//
|
||||
// ALuint uiBuffer;
|
||||
// alGenBuffers(1, &uiBuffer);
|
||||
// eaxSetBufferMode(1, &uiBuffer, alGetEnumValue("AL_STORAGE_HARDWARE"));
|
||||
// alBufferData(...);
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// Force an Open AL Buffer into 'accessible' (currently host) RAM (good for streaming buffers)
|
||||
//
|
||||
// ALuint uiBuffer;
|
||||
// alGenBuffers(1, &uiBuffer);
|
||||
// eaxSetBufferMode(1, &uiBuffer, alGetEnumValue("AL_STORAGE_ACCESSIBLE"));
|
||||
// alBufferData(...);
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
// Put an Open AL Buffer into X-RAM if memory is available, otherwise use
|
||||
// host RAM. This is the default mode.
|
||||
//
|
||||
// ALuint uiBuffer;
|
||||
// alGenBuffers(1, &uiBuffer);
|
||||
// eaxSetBufferMode(1, &uiBuffer, alGetEnumValue("AL_STORAGE_AUTOMATIC"));
|
||||
// alBufferData(...);
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
|
@ -0,0 +1,184 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2009-01-11 13:34:24 +0200 (Sun, 11 Jan 2009) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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();
|
||||
}
|
||||
|
|
@ -0,0 +1,91 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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
|
||||
|
|
@ -0,0 +1,370 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-08-30 22:45:25 +0300 (Thu, 30 Aug 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: BPMDetect.cpp 149 2012-08-30 19:45:25Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 <stdio.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);
|
||||
|
||||
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// Enable following define to create bpm analysis file:
|
||||
|
||||
// #define _CREATE_BPM_DEBUG_FILE
|
||||
|
||||
#ifdef _CREATE_BPM_DEBUG_FILE
|
||||
|
||||
#define DEBUGFILE_NAME "c:\\temp\\soundtouch-bpm-debug.txt"
|
||||
|
||||
static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
|
||||
{
|
||||
FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
|
||||
int i;
|
||||
|
||||
if (fptr)
|
||||
{
|
||||
printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
|
||||
for (i = minpos; i < maxpos; i ++)
|
||||
{
|
||||
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
|
||||
}
|
||||
fclose(fptr);
|
||||
}
|
||||
}
|
||||
#else
|
||||
#define _SaveDebugData(a,b,c,d)
|
||||
#endif
|
||||
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
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 1500 (out of 32768) that's
|
||||
// safe initial 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 = (1500 * 1500) / avgnorm;
|
||||
#else
|
||||
// float samples, scaled to range [-1..+1[
|
||||
RMSVolumeAccu = (0.045f * 0.045f) / avgnorm;
|
||||
#endif
|
||||
|
||||
// choose decimation factor so that result is approx. 1000 Hz
|
||||
decimateBy = sampleRate / 1000;
|
||||
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)
|
||||
if (val < 0.5 * sqrt(RMSVolumeAccu * avgnorm))
|
||||
{
|
||||
val = 0;
|
||||
}
|
||||
|
||||
// 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;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
|
||||
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);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
|
||||
void BPMDetect::removeBias()
|
||||
{
|
||||
int i;
|
||||
float minval = 1e12f; // arbitrary large number
|
||||
|
||||
for (i = windowStart; i < windowLen; i ++)
|
||||
{
|
||||
if (xcorr[i] < minval)
|
||||
{
|
||||
minval = xcorr[i];
|
||||
}
|
||||
}
|
||||
|
||||
for (i = windowStart; i < windowLen; i ++)
|
||||
{
|
||||
xcorr[i] -= minval;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
float BPMDetect::getBpm()
|
||||
{
|
||||
double peakPos;
|
||||
double coeff;
|
||||
PeakFinder peakFinder;
|
||||
|
||||
coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
|
||||
|
||||
// save bpm debug analysis data if debug data enabled
|
||||
_SaveDebugData(xcorr, windowStart, windowLen, coeff);
|
||||
|
||||
// remove bias from xcorr data
|
||||
removeBias();
|
||||
|
||||
// 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) (coeff / peakPos);
|
||||
}
|
||||
|
|
@ -0,0 +1,164 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-08-30 22:53:44 +0300 (Thu, 30 Aug 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 200
|
||||
|
||||
|
||||
/// 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;
|
||||
|
||||
/// 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
|
||||
);
|
||||
|
||||
/// remove constant bias from xcorr data
|
||||
void removeBias();
|
||||
|
||||
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_
|
||||
|
|
@ -0,0 +1,15 @@
|
|||
set(SRCS
|
||||
AAFilter.cpp
|
||||
BPMDetect.cpp
|
||||
cpu_detect_x86.cpp
|
||||
FIFOSampleBuffer.cpp
|
||||
FIRFilter.cpp
|
||||
mmx_optimized.cpp
|
||||
PeakFinder.cpp
|
||||
RateTransposer.cpp
|
||||
SoundTouch.cpp
|
||||
sse_optimized.cpp
|
||||
TDStretch.cpp
|
||||
)
|
||||
|
||||
add_library(SoundTouch STATIC ${SRCS})
|
||||
|
|
@ -0,0 +1,274 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 "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)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
|
||||
}
|
||||
// Align the buffer to begin at 16byte cache line boundary for optimal performance
|
||||
temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
|
||||
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;
|
||||
}
|
||||
|
||||
|
||||
/// allow trimming (downwards) amount of samples in pipeline.
|
||||
/// Returns adjusted amount of samples
|
||||
uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
|
||||
{
|
||||
if (numSamples < samplesInBuffer)
|
||||
{
|
||||
samplesInBuffer = numSamples;
|
||||
}
|
||||
return samplesInBuffer;
|
||||
}
|
||||
|
||||
|
|
@ -0,0 +1,178 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIFOSampleBuffer.h 143 2012-06-13 19:29:53Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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();
|
||||
|
||||
/// allow trimming (downwards) amount of samples in pipeline.
|
||||
/// Returns adjusted amount of samples
|
||||
uint adjustAmountOfSamples(uint numSamples);
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif
|
||||
|
|
@ -0,0 +1,234 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// '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: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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;
|
||||
|
||||
/// allow trimming (downwards) amount of samples in pipeline.
|
||||
/// Returns adjusted amount of samples
|
||||
virtual uint adjustAmountOfSamples(uint numSamples) = 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();
|
||||
}
|
||||
|
||||
/// allow trimming (downwards) amount of samples in pipeline.
|
||||
/// Returns adjusted amount of samples
|
||||
virtual uint adjustAmountOfSamples(uint numSamples)
|
||||
{
|
||||
return output->adjustAmountOfSamples(numSamples);
|
||||
}
|
||||
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
#endif
|
||||
|
|
@ -0,0 +1,259 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2011-09-02 21:56:11 +0300 (Fri, 02 Sep 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIRFilter.cpp 131 2011-09-02 18:56:11Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 "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) ST_THROW_RT_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!
|
||||
ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
|
||||
return newInstance();
|
||||
}
|
||||
|
||||
|
||||
FIRFilter * FIRFilter::newInstance()
|
||||
{
|
||||
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
|
||||
|
||||
{
|
||||
// ISA optimizations not supported, use plain C version
|
||||
return ::new FIRFilter;
|
||||
}
|
||||
}
|
||||
|
|
@ -0,0 +1,145 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2011-02-13 21:13:57 +0200 (Sun, 13 Feb 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: FIRFilter.h 104 2011-02-13 19:13:57Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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
|
||||
|
|
@ -0,0 +1,276 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-12-28 21:52:47 +0200 (Fri, 28 Dec 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: PeakFinder.cpp 164 2012-12-28 19:52:47Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
|
||||
int PeakFinder::findTop(const float *data, int peakpos) const
|
||||
{
|
||||
int i;
|
||||
int start, end;
|
||||
float refvalue;
|
||||
|
||||
refvalue = data[peakpos];
|
||||
|
||||
// seek within ±10 points
|
||||
start = peakpos - 10;
|
||||
if (start < minPos) start = minPos;
|
||||
end = peakpos + 10;
|
||||
if (end > maxPos) end = maxPos;
|
||||
|
||||
for (i = start; i <= end; i ++)
|
||||
{
|
||||
if (data[i] > refvalue)
|
||||
{
|
||||
peakpos = i;
|
||||
refvalue = data[i];
|
||||
}
|
||||
}
|
||||
|
||||
// failure if max value is at edges of seek range => it's not peak, it's at slope.
|
||||
if ((peakpos == start) || (peakpos == end)) return 0;
|
||||
|
||||
return peakpos;
|
||||
}
|
||||
|
||||
|
||||
// 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
|
||||
{
|
||||
int lowpos;
|
||||
int pos;
|
||||
int climb_count;
|
||||
float refvalue;
|
||||
float delta;
|
||||
|
||||
climb_count = 0;
|
||||
refvalue = data[peakpos];
|
||||
lowpos = peakpos;
|
||||
|
||||
pos = peakpos;
|
||||
|
||||
while ((pos > minPos+1) && (pos < maxPos-1))
|
||||
{
|
||||
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 = 0.5f * (data[gp1] + data[gp2]);
|
||||
peakLevel = data[peakpos];
|
||||
|
||||
// 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 = 3; i < 10; i ++)
|
||||
{
|
||||
double peaktmp, harmonic;
|
||||
int i1,i2;
|
||||
|
||||
harmonic = (double)i * 0.5;
|
||||
peakpos = (int)(highPeak / harmonic + 0.5f);
|
||||
if (peakpos < minPos) break;
|
||||
peakpos = findTop(data, peakpos); // seek true local maximum index
|
||||
if (peakpos == 0) continue; // no local max here
|
||||
|
||||
// calculate mass-center of possible harmonic peak
|
||||
peaktmp = getPeakCenter(data, peakpos);
|
||||
|
||||
// accept harmonic peak if
|
||||
// (a) it is found
|
||||
// (b) is within ±4% of the expected harmonic interval
|
||||
// (c) has at least half x-corr value of the max. peak
|
||||
|
||||
double diff = harmonic * peaktmp / highPeak;
|
||||
if ((diff < 0.96) || (diff > 1.04)) continue; // peak too afar from expected
|
||||
|
||||
// now compare to highest detected peak
|
||||
i1 = (int)(highPeak + 0.5);
|
||||
i2 = (int)(peaktmp + 0.5);
|
||||
if (data[i2] >= 0.4*data[i1])
|
||||
{
|
||||
// The harmonic is at least half as high primary peak,
|
||||
// thus use the harmonic peak instead
|
||||
peak = peaktmp;
|
||||
}
|
||||
}
|
||||
|
||||
return peak;
|
||||
}
|
||||
|
|
@ -0,0 +1,97 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2011-12-30 22:33:46 +0200 (Fri, 30 Dec 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: PeakFinder.h 132 2011-12-30 20:33:46Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
|
||||
int findTop(const float *data, int peakpos) 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_
|
||||
|
|
@ -0,0 +1,626 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2011-09-02 21:56:11 +0300 (Fri, 02 Sep 2011) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: RateTransposer.cpp 131 2011-09-02 18:56:11Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 "RateTransposer.h"
|
||||
#include "AAFilter.h"
|
||||
|
||||
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)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
|
||||
return newInstance();
|
||||
}
|
||||
|
||||
|
||||
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;
|
||||
}
|
||||
|
|
@ -0,0 +1,159 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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
|
||||
|
|
@ -0,0 +1,191 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-12-28 16:53:56 +0200 (Fri, 28 Dec 2012) $
|
||||
// File revision : $Revision: 3 $
|
||||
//
|
||||
// $Id: STTypes.h 162 2012-12-28 14:53:56Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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;
|
||||
|
||||
// Patch for MinGW: on Win64 long is 32-bit
|
||||
#ifdef _WIN64
|
||||
typedef unsigned long long ulongptr;
|
||||
#else
|
||||
typedef ulong ulongptr;
|
||||
#endif
|
||||
|
||||
|
||||
// Helper macro for aligning pointer up to next 16-byte boundary
|
||||
#define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
|
||||
|
||||
|
||||
#if (defined(__GNUC__) && !defined(ANDROID))
|
||||
// In GCC, include soundtouch_config.h made by config scritps.
|
||||
// Skip this in Android compilation that uses GCC but without configure scripts.
|
||||
//#include "soundtouch_config.h"
|
||||
#endif
|
||||
|
||||
#ifndef _WINDEF_
|
||||
// if these aren't defined already by Windows headers, define now
|
||||
#if defined(__APPLE__)
|
||||
typedef signed char BOOL;
|
||||
#else
|
||||
typedef int BOOL;
|
||||
#endif
|
||||
#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 (defined(__SOFTFP__))
|
||||
// For Android compilation: Force use of Integer samples in case that
|
||||
// compilation uses soft-floating point emulation - soft-fp is way too slow
|
||||
#undef SOUNDTOUCH_FLOAT_SAMPLES
|
||||
#define SOUNDTOUCH_INTEGER_SAMPLES 1
|
||||
#endif
|
||||
|
||||
#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
|
||||
|
||||
#if (_M_IX86 || __i386__ || __x86_64__ || _M_X64)
|
||||
/// 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
|
||||
|
||||
/// In GNU environment, allow the user to override this setting by
|
||||
/// giving the following switch to the configure script:
|
||||
/// ./configure --disable-x86-optimizations
|
||||
/// ./configure --enable-x86-optimizations=no
|
||||
#ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
|
||||
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
|
||||
#endif
|
||||
#else
|
||||
/// Always disable optimizations when not using a x86 systems.
|
||||
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
|
||||
|
||||
#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
|
||||
#ifndef _M_X64
|
||||
#define SOUNDTOUCH_ALLOW_MMX 1
|
||||
#endif
|
||||
#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
|
||||
|
||||
};
|
||||
|
||||
// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
|
||||
#define ST_NO_EXCEPTION_HANDLING 1
|
||||
#ifdef ST_NO_EXCEPTION_HANDLING
|
||||
// Exceptions disabled. Throw asserts instead if enabled.
|
||||
#include <assert.h>
|
||||
#define ST_THROW_RT_ERROR(x) {assert((const char *)x);}
|
||||
#else
|
||||
// use c++ standard exceptions
|
||||
#include <stdexcept>
|
||||
#define ST_THROW_RT_ERROR(x) {throw std::runtime_error(x);}
|
||||
#endif
|
||||
|
||||
// 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
|
||||
|
|
@ -0,0 +1,501 @@
|
|||
//////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: SoundTouch.cpp 143 2012-06-13 19:29:53Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 <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)
|
||||
{
|
||||
ST_THROW_RT_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)
|
||||
{
|
||||
ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
|
||||
}
|
||||
else if (channels == 0)
|
||||
{
|
||||
ST_THROW_RT_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;
|
||||
int nUnprocessed;
|
||||
int nOut;
|
||||
SAMPLETYPE buff[64*2]; // note: allocate 2*64 to cater 64 sample frames of stereo sound
|
||||
|
||||
// check how many samples still await processing, and scale
|
||||
// that by tempo & rate to get expected output sample count
|
||||
nUnprocessed = numUnprocessedSamples();
|
||||
nUnprocessed = (int)((double)nUnprocessed / (tempo * rate) + 0.5);
|
||||
|
||||
nOut = numSamples(); // ready samples currently in buffer ...
|
||||
nOut += nUnprocessed; // ... and how many we expect there to be in the end
|
||||
|
||||
memset(buff, 0, 64 * channels * 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 ((int)numSamples() >= nOut)
|
||||
{
|
||||
// Enough new samples have appeared into the output!
|
||||
// As samples come from processing with bigger chunks, now truncate it
|
||||
// back to maximum "nOut" samples to improve duration accuracy
|
||||
adjustAmountOfSamples(nOut);
|
||||
|
||||
// finish
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
// 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;
|
||||
}
|
||||
|
|
@ -0,0 +1,277 @@
|
|||
//////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-12-28 21:32:59 +0200 (Fri, 28 Dec 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: SoundTouch.h 163 2012-12-28 19:32:59Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 "1.7.1"
|
||||
|
||||
/// SoundTouch library version id
|
||||
#define SOUNDTOUCH_VERSION_ID (10701)
|
||||
|
||||
//
|
||||
// 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
|
||||
|
|
@ -0,0 +1,352 @@
|
|||
<?xml version="1.0" encoding="utf-8"?>
|
||||
<Project DefaultTargets="Build" ToolsVersion="4.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
|
||||
<ItemGroup Label="ProjectConfigurations">
|
||||
<ProjectConfiguration Include="Debug|Win32">
|
||||
<Configuration>Debug</Configuration>
|
||||
<Platform>Win32</Platform>
|
||||
</ProjectConfiguration>
|
||||
<ProjectConfiguration Include="Debug|x64">
|
||||
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<ClInclude Include="SoundTouch.h" />
|
||||
<ClInclude Include="STTypes.h" />
|
||||
<ClInclude Include="TDStretch.h" />
|
||||
</ItemGroup>
|
||||
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
|
||||
<ImportGroup Label="ExtensionTargets">
|
||||
</ImportGroup>
|
||||
</Project>
|
||||
|
|
@ -0,0 +1,60 @@
|
|||
<?xml version="1.0" encoding="utf-8"?>
|
||||
<Project ToolsVersion="4.0" xmlns="http://schemas.microsoft.com/developer/msbuild/2003">
|
||||
<ItemGroup>
|
||||
<Filter Include="Source Files">
|
||||
<UniqueIdentifier>{b7786182-6345-4203-8b48-39eec5ec85dc}</UniqueIdentifier>
|
||||
<Extensions>cpp;c;cxx;rc;def;r;odl;idl;hpj;bat</Extensions>
|
||||
</Filter>
|
||||
<Filter Include="Source Files\bpm">
|
||||
<UniqueIdentifier>{75380bb9-1e58-4186-a9cd-ec7cd284e6a5}</UniqueIdentifier>
|
||||
</Filter>
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClCompile Include="AAFilter.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="cpu_detect_x86.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="FIFOSampleBuffer.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="FIRFilter.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="mmx_optimized.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="RateTransposer.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="SoundTouch.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="sse_optimized.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="TDStretch.cpp">
|
||||
<Filter>Source Files</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="BPMDetect.cpp">
|
||||
<Filter>Source Files\bpm</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="PeakFinder.cpp">
|
||||
<Filter>Source Files\bpm</Filter>
|
||||
</ClCompile>
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClInclude Include="FIFOSampleBuffer.h" />
|
||||
<ClInclude Include="FIFOSamplePipe.h" />
|
||||
<ClInclude Include="FIRFilter.h" />
|
||||
<ClInclude Include="PeakFinder.h" />
|
||||
<ClInclude Include="RateTransposer.h" />
|
||||
<ClInclude Include="SoundTouch.h" />
|
||||
<ClInclude Include="STTypes.h" />
|
||||
<ClInclude Include="TDStretch.h" />
|
||||
<ClInclude Include="AAFilter.h" />
|
||||
<ClInclude Include="BPMDetect.h" />
|
||||
<ClInclude Include="cpu_detect.h" />
|
||||
</ItemGroup>
|
||||
</Project>
|
||||
|
|
@ -0,0 +1,808 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
|
||||
// File revision : $Revision: 1.12 $
|
||||
//
|
||||
// $Id: TDStretch.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 "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;
|
||||
pMidBufferUnaligned = 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[] pMidBufferUnaligned;
|
||||
}
|
||||
|
||||
|
||||
|
||||
// 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;
|
||||
SAMPLETYPE m1, m2;
|
||||
|
||||
m1 = (SAMPLETYPE)0;
|
||||
m2 = (SAMPLETYPE)overlapLength;
|
||||
|
||||
for (i = 0; i < overlapLength ; i ++)
|
||||
{
|
||||
pOutput[i] = (pInput[i] * m1 + pMidBuffer[i] * m2 ) / overlapLength;
|
||||
m1 += 1;
|
||||
m2 -= 1;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
|
||||
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 (bQuickSeek)
|
||||
{
|
||||
return seekBestOverlapPositionQuick(refPos);
|
||||
}
|
||||
else
|
||||
{
|
||||
return seekBestOverlapPositionFull(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::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
|
||||
{
|
||||
int bestOffs;
|
||||
double bestCorr, corr;
|
||||
int i;
|
||||
|
||||
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 = calcCrossCorr(refPos + channels * i, pMidBuffer);
|
||||
// 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::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
|
||||
{
|
||||
int j;
|
||||
int bestOffs;
|
||||
double bestCorr, corr;
|
||||
int scanCount, corrOffset, tempOffset;
|
||||
|
||||
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)calcCrossCorr(refPos + channels * tempOffset, pMidBuffer);
|
||||
// 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[] pMidBufferUnaligned;
|
||||
|
||||
pMidBufferUnaligned = new SAMPLETYPE[overlapLength * 2 + 16 / sizeof(SAMPLETYPE)];
|
||||
// ensure that 'pMidBuffer' is aligned to 16 byte boundary for efficiency
|
||||
pMidBuffer = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(pMidBufferUnaligned);
|
||||
|
||||
clearMidBuffer();
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// 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!
|
||||
ST_THROW_RT_ERROR("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");
|
||||
return newInstance();
|
||||
}
|
||||
|
||||
|
||||
TDStretch * TDStretch::newInstance()
|
||||
{
|
||||
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
|
||||
|
||||
{
|
||||
// ISA optimizations not supported, use plain C version
|
||||
return ::new TDStretch;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Integer arithmetics specific algorithm implementations.
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
||||
|
||||
// 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;
|
||||
}
|
||||
|
||||
|
||||
double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare) const
|
||||
{
|
||||
long corr;
|
||||
long norm;
|
||||
int i;
|
||||
|
||||
corr = norm = 0;
|
||||
// Same routine for stereo and mono. For stereo, unroll loop for better
|
||||
// efficiency and gives slightly better resolution against rounding.
|
||||
// For mono it same routine, just unrolls loop by factor of 4
|
||||
for (i = 0; i < channels * overlapLength; i += 4)
|
||||
{
|
||||
corr += (mixingPos[i] * compare[i] +
|
||||
mixingPos[i + 1] * compare[i + 1] +
|
||||
mixingPos[i + 2] * compare[i + 2] +
|
||||
mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBits;
|
||||
norm += (mixingPos[i] * mixingPos[i] +
|
||||
mixingPos[i + 1] * mixingPos[i + 1] +
|
||||
mixingPos[i + 2] * mixingPos[i + 2] +
|
||||
mixingPos[i + 3] * mixingPos[i + 3]) >> 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 (double)corr / sqrt((double)norm);
|
||||
}
|
||||
|
||||
#endif // SOUNDTOUCH_INTEGER_SAMPLES
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Floating point arithmetics specific algorithm implementations.
|
||||
//
|
||||
|
||||
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
|
||||
|
||||
// Overlaps samples in 'midBuffer' with the samples in 'pInput'
|
||||
void TDStretch::overlapStereo(float *pOutput, const float *pInput) const
|
||||
{
|
||||
int i;
|
||||
float fScale;
|
||||
float f1;
|
||||
float f2;
|
||||
|
||||
fScale = 1.0f / (float)overlapLength;
|
||||
|
||||
f1 = 0;
|
||||
f2 = 1.0f;
|
||||
|
||||
for (i = 0; i < 2 * (int)overlapLength ; i += 2)
|
||||
{
|
||||
pOutput[i + 0] = pInput[i + 0] * f1 + pMidBuffer[i + 0] * f2;
|
||||
pOutput[i + 1] = pInput[i + 1] * f1 + pMidBuffer[i + 1] * f2;
|
||||
|
||||
f1 += fScale;
|
||||
f2 -= fScale;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
/// 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::calcCrossCorr(const float *mixingPos, const float *compare) const
|
||||
{
|
||||
double corr;
|
||||
double norm;
|
||||
int i;
|
||||
|
||||
corr = norm = 0;
|
||||
// Same routine for stereo and mono. For Stereo, unroll by factor of 2.
|
||||
// For mono it's same routine yet unrollsd by factor of 4.
|
||||
for (i = 0; i < channels * overlapLength; i += 4)
|
||||
{
|
||||
corr += mixingPos[i] * compare[i] +
|
||||
mixingPos[i + 1] * compare[i + 1];
|
||||
|
||||
norm += mixingPos[i] * mixingPos[i] +
|
||||
mixingPos[i + 1] * mixingPos[i + 1];
|
||||
|
||||
// unroll the loop for better CPU efficiency:
|
||||
corr += mixingPos[i + 2] * compare[i + 2] +
|
||||
mixingPos[i + 3] * compare[i + 3];
|
||||
|
||||
norm += mixingPos[i + 2] * mixingPos[i + 2] +
|
||||
mixingPos[i + 3] * mixingPos[i + 3];
|
||||
}
|
||||
|
||||
if (norm < 1e-9) norm = 1.0; // to avoid div by zero
|
||||
return corr / sqrt(norm);
|
||||
}
|
||||
|
||||
#endif // SOUNDTOUCH_FLOAT_SAMPLES
|
||||
|
|
@ -0,0 +1,268 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-04-01 22:49:30 +0300 (Sun, 01 Apr 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: TDStretch.h 137 2012-04-01 19:49:30Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 *pMidBufferUnaligned;
|
||||
int overlapLength;
|
||||
int seekLength;
|
||||
int seekWindowLength;
|
||||
int overlapDividerBits;
|
||||
int slopingDivider;
|
||||
float nominalSkip;
|
||||
float skipFract;
|
||||
FIFOSampleBuffer outputBuffer;
|
||||
FIFOSampleBuffer inputBuffer;
|
||||
BOOL bQuickSeek;
|
||||
|
||||
int sampleRate;
|
||||
int sequenceMs;
|
||||
int seekWindowMs;
|
||||
int overlapMs;
|
||||
BOOL bAutoSeqSetting;
|
||||
BOOL bAutoSeekSetting;
|
||||
|
||||
void acceptNewOverlapLength(int newOverlapLength);
|
||||
|
||||
virtual void clearCrossCorrState();
|
||||
void calculateOverlapLength(int overlapMs);
|
||||
|
||||
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare) const;
|
||||
|
||||
virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
|
||||
virtual int seekBestOverlapPositionQuick(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 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:
|
||||
double calcCrossCorr(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 calcCrossCorr(const float *mixingPos, const float *compare) const;
|
||||
};
|
||||
|
||||
#endif /// SOUNDTOUCH_ALLOW_SSE
|
||||
|
||||
}
|
||||
#endif /// TDStretch_H
|
||||
|
|
@ -0,0 +1,62 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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_
|
||||
|
|
@ -0,0 +1,137 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// Generic version of the x86 CPU extension detection routine.
|
||||
///
|
||||
/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
|
||||
/// for the Microsoft compiler version.
|
||||
///
|
||||
/// Author : Copyright (c) Olli Parviainen
|
||||
/// Author e-mail : oparviai 'at' iki.fi
|
||||
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// Last changed : $Date: 2012-11-08 20:44:37 +0200 (Thu, 08 Nov 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: cpu_detect_x86.cpp 159 2012-11-08 18:44:37Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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"
|
||||
|
||||
#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
|
||||
|
||||
#if defined(__GNUC__) && defined(__i386__)
|
||||
// gcc
|
||||
#include "cpuid.h"
|
||||
#elif defined(_M_IX86)
|
||||
// windows non-gcc
|
||||
#include <intrin.h>
|
||||
#endif
|
||||
|
||||
#define bit_MMX (1 << 23)
|
||||
#define bit_SSE (1 << 25)
|
||||
#define bit_SSE2 (1 << 26)
|
||||
#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)
|
||||
{
|
||||
/// If building for a 64bit system (no Itanium) and the user wants optimizations.
|
||||
/// Return the OR of SUPPORT_{MMX,SSE,SSE2}. 11001 or 0x19.
|
||||
/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
|
||||
#if ((defined(__GNUC__) && defined(__x86_64__)) \
|
||||
|| defined(_M_X64)) \
|
||||
&& defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
|
||||
return 0x19 & ~_dwDisabledISA;
|
||||
|
||||
/// If building for a 32bit system and the user wants optimizations.
|
||||
/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
|
||||
#elif ((defined(__GNUC__) && defined(__i386__)) \
|
||||
|| defined(_M_IX86)) \
|
||||
&& defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
|
||||
|
||||
if (_dwDisabledISA == 0xffffffff) return 0;
|
||||
|
||||
uint res = 0;
|
||||
|
||||
#if defined(__GNUC__)
|
||||
// GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
|
||||
uint eax, ebx, ecx, edx; // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
|
||||
|
||||
// Check if no cpuid support.
|
||||
if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
|
||||
|
||||
if (edx & bit_MMX) res = res | SUPPORT_MMX;
|
||||
if (edx & bit_SSE) res = res | SUPPORT_SSE;
|
||||
if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
|
||||
|
||||
#else
|
||||
// Window / VS version of cpuid. Notice that Visual Studio 2005 or later required
|
||||
// for __cpuid intrinsic support.
|
||||
int reg[4] = {-1};
|
||||
|
||||
// Check if no cpuid support.
|
||||
__cpuid(reg,0);
|
||||
if ((unsigned int)reg[0] == 0) return 0; // always disable extensions.
|
||||
|
||||
__cpuid(reg,1);
|
||||
if ((unsigned int)reg[3] & bit_MMX) res = res | SUPPORT_MMX;
|
||||
if ((unsigned int)reg[3] & bit_SSE) res = res | SUPPORT_SSE;
|
||||
if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
|
||||
|
||||
#endif
|
||||
|
||||
return res & ~_dwDisabledISA;
|
||||
|
||||
#else
|
||||
|
||||
/// One of these is true:
|
||||
/// 1) We don't want optimizations.
|
||||
/// 2) Using an unsupported compiler.
|
||||
/// 3) Running on a non-x86 platform.
|
||||
return 0;
|
||||
|
||||
#endif
|
||||
}
|
||||
|
|
@ -0,0 +1,317 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: mmx_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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
|
||||
|
||||
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
|
||||
double TDStretchMMX::calcCrossCorr(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 or 4 * mono samples
|
||||
// during each round for improved CPU-level parallellization.
|
||||
for (i = 0; i < channels * overlapLength / 16; 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 (double)corr / 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 *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
|
||||
|
||||
// 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
|
||||
|
|
@ -0,0 +1,361 @@
|
|||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
|
||||
// File revision : $Revision: 4 $
|
||||
//
|
||||
// $Id: sse_optimized.cpp 160 2012-11-08 18:53:01Z oparviai $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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::calcCrossCorr(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 (((ulongptr)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. Use same routine for
|
||||
// stereo & mono, for mono it just means twice the amount of unrolling.
|
||||
for (i = 0; i < channels * overlapLength / 16; 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 < channels * overlapLength / 16; 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);
|
||||
*/
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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 *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
|
||||
|
||||
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(((ulongptr)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;
|
||||
}
|
||||
*/
|
||||
}
|
||||
|
||||
#endif // SOUNDTOUCH_ALLOW_SSE
|
||||
|
|
@ -109,78 +109,90 @@
|
|||
<PropertyGroup Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'" />
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">
|
||||
<ClCompile>
|
||||
<AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
<Link>
|
||||
<GenerateDebugInformation>true</GenerateDebugInformation>
|
||||
</Link>
|
||||
<Lib />
|
||||
<Lib>
|
||||
<AdditionalDependencies>SoundTouchD.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
|
||||
<AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
|
||||
</Lib>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">
|
||||
<ClCompile>
|
||||
<AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
<Link>
|
||||
<GenerateDebugInformation>true</GenerateDebugInformation>
|
||||
</Link>
|
||||
<Lib />
|
||||
<Lib>
|
||||
<AdditionalDependencies>SoundTouchD.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
|
||||
<AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
|
||||
</Lib>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">
|
||||
<ClCompile>
|
||||
<AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
<Link>
|
||||
<GenerateDebugInformation>true</GenerateDebugInformation>
|
||||
<EnableCOMDATFolding>true</EnableCOMDATFolding>
|
||||
<OptimizeReferences>true</OptimizeReferences>
|
||||
</Link>
|
||||
<Lib />
|
||||
<Lib>
|
||||
<AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
|
||||
<AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
|
||||
</Lib>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='DebugFast|Win32'">
|
||||
<ClCompile>
|
||||
<AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
<Link>
|
||||
<GenerateDebugInformation>true</GenerateDebugInformation>
|
||||
<EnableCOMDATFolding>true</EnableCOMDATFolding>
|
||||
<OptimizeReferences>true</OptimizeReferences>
|
||||
</Link>
|
||||
<Lib />
|
||||
<Lib>
|
||||
<AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
|
||||
<AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
|
||||
</Lib>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='Release|x64'">
|
||||
<ClCompile>
|
||||
<AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
<Link>
|
||||
<GenerateDebugInformation>true</GenerateDebugInformation>
|
||||
<EnableCOMDATFolding>true</EnableCOMDATFolding>
|
||||
<OptimizeReferences>true</OptimizeReferences>
|
||||
</Link>
|
||||
<Lib />
|
||||
<Lib>
|
||||
<AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
|
||||
<AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
|
||||
</Lib>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemDefinitionGroup Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'">
|
||||
<ClCompile>
|
||||
<AdditionalIncludeDirectories>..\Common\Src;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories>..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
</ClCompile>
|
||||
<Link>
|
||||
<GenerateDebugInformation>true</GenerateDebugInformation>
|
||||
<EnableCOMDATFolding>true</EnableCOMDATFolding>
|
||||
<OptimizeReferences>true</OptimizeReferences>
|
||||
</Link>
|
||||
<Lib />
|
||||
<Lib>
|
||||
<AdditionalDependencies>SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib</AdditionalDependencies>
|
||||
<AdditionalLibraryDirectories>..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories)</AdditionalLibraryDirectories>
|
||||
</Lib>
|
||||
</ItemDefinitionGroup>
|
||||
<ItemGroup>
|
||||
<ClCompile Include="Src\aldlist.cpp">
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|Win32'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|x64'">true</ExcludedFromBuild>
|
||||
</ClCompile>
|
||||
<ClCompile Include="Src\aldlist.cpp" />
|
||||
<ClCompile Include="Src\AOSoundStream.cpp" />
|
||||
<ClCompile Include="Src\AudioCommon.cpp" />
|
||||
<ClCompile Include="Src\AudioCommonConfig.cpp" />
|
||||
<ClCompile Include="Src\DPL2Decoder.cpp" />
|
||||
<ClCompile Include="Src\DSoundStream.cpp" />
|
||||
<ClCompile Include="Src\Mixer.cpp" />
|
||||
<ClCompile Include="Src\NullSoundStream.cpp" />
|
||||
|
|
@ -189,17 +201,11 @@
|
|||
<ClCompile Include="Src\XAudio2Stream.cpp" />
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClInclude Include="Src\aldlist.h">
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|Win32'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='DebugFast|x64'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</ExcludedFromBuild>
|
||||
<ExcludedFromBuild Condition="'$(Configuration)|$(Platform)'=='Release|x64'">true</ExcludedFromBuild>
|
||||
</ClInclude>
|
||||
<ClInclude Include="Src\aldlist.h" />
|
||||
<ClInclude Include="Src\AOSoundStream.h" />
|
||||
<ClInclude Include="Src\AudioCommon.h" />
|
||||
<ClInclude Include="Src\AudioCommonConfig.h" />
|
||||
<ClInclude Include="Src\DPL2Decoder.h" />
|
||||
<ClInclude Include="Src\DSoundStream.h" />
|
||||
<ClInclude Include="Src\Mixer.h" />
|
||||
<ClInclude Include="Src\NullSoundStream.h" />
|
||||
|
|
@ -212,6 +218,14 @@
|
|||
<None Include="CMakeLists.txt" />
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ProjectReference Include="..\..\..\Externals\SoundTouch\SoundTouch.vcxproj">
|
||||
<Project>{68a5dd20-7057-448b-8fe0-b6ac8d205509}</Project>
|
||||
<Private>true</Private>
|
||||
<ReferenceOutputAssembly>true</ReferenceOutputAssembly>
|
||||
<CopyLocalSatelliteAssemblies>false</CopyLocalSatelliteAssemblies>
|
||||
<LinkLibraryDependencies>true</LinkLibraryDependencies>
|
||||
<UseLibraryDependencyInputs>false</UseLibraryDependencyInputs>
|
||||
</ProjectReference>
|
||||
<ProjectReference Include="..\Common\Common.vcxproj">
|
||||
<Project>{c87a4178-44f6-49b2-b7aa-c79af1b8c534}</Project>
|
||||
<Private>true</Private>
|
||||
|
|
|
|||
|
|
@ -21,6 +21,7 @@
|
|||
<ClCompile Include="Src\NullSoundStream.cpp">
|
||||
<Filter>SoundStreams</Filter>
|
||||
</ClCompile>
|
||||
<ClCompile Include="Src\DPL2Decoder.cpp" />
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClInclude Include="Src\aldlist.h" />
|
||||
|
|
@ -44,6 +45,7 @@
|
|||
<ClInclude Include="Src\XAudio2Stream.h">
|
||||
<Filter>SoundStreams</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="Src\DPL2Decoder.h" />
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<None Include="CMakeLists.txt" />
|
||||
|
|
|
|||
|
|
@ -1,5 +1,6 @@
|
|||
set(SRCS Src/AudioCommon.cpp
|
||||
Src/AudioCommonConfig.cpp
|
||||
Src/DPL2Decoder.cpp
|
||||
Src/Mixer.cpp
|
||||
Src/WaveFile.cpp
|
||||
Src/NullSoundStream.cpp)
|
||||
|
|
@ -18,7 +19,7 @@ endif(AO_FOUND)
|
|||
|
||||
if(OPENAL_FOUND)
|
||||
set(SRCS ${SRCS} Src/OpenALStream.cpp Src/aldlist.cpp)
|
||||
set(LIBS ${LIBS} ${OPENAL_LIBRARY})
|
||||
set(LIBS ${LIBS} ${OPENAL_LIBRARY} SoundTouch )
|
||||
endif(OPENAL_FOUND)
|
||||
|
||||
if(PULSEAUDIO_FOUND)
|
||||
|
|
|
|||
|
|
@ -42,7 +42,6 @@ void AudioCommonConfig::Load()
|
|||
#else
|
||||
file.Get("Config", "Backend", &sBackend, BACKEND_NULLSOUND);
|
||||
#endif
|
||||
file.Get("Config", "Frequency", &iFrequency, 48000);
|
||||
file.Get("Config", "Volume", &m_Volume, 100);
|
||||
}
|
||||
|
||||
|
|
@ -55,7 +54,6 @@ void AudioCommonConfig::SaveSettings()
|
|||
file.Set("Config", "EnableJIT", m_EnableJIT);
|
||||
file.Set("Config", "DumpAudio", m_DumpAudio);
|
||||
file.Set("Config", "Backend", sBackend);
|
||||
file.Set("Config", "Frequency", iFrequency);
|
||||
file.Set("Config", "Volume", m_Volume);
|
||||
|
||||
file.Save(File::GetUserPath(F_DSPCONFIG_IDX));
|
||||
|
|
|
|||
|
|
@ -37,7 +37,6 @@ struct AudioCommonConfig
|
|||
bool m_DumpAudio;
|
||||
int m_Volume;
|
||||
std::string sBackend;
|
||||
int iFrequency;
|
||||
|
||||
// Load from given file
|
||||
void Load();
|
||||
|
|
|
|||
|
|
@ -0,0 +1,400 @@
|
|||
// Copyright (C) 2003 Dolphin Project.
|
||||
|
||||
// This program is free software: you can redistribute it and/or modify
|
||||
// it under the terms of the GNU General Public License as published by
|
||||
// the Free Software Foundation, version 2.0.
|
||||
|
||||
// This program is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
||||
// GNU General Public License 2.0 for more details.
|
||||
|
||||
// A copy of the GPL 2.0 should have been included with the program.
|
||||
// If not, see http://www.gnu.org/licenses/
|
||||
|
||||
// Official SVN repository and contact information can be found at
|
||||
// http://code.google.com/p/dolphin-emu/
|
||||
|
||||
// Dolby Pro Logic 2 decoder from ffdshow-tryout
|
||||
// * Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
|
||||
// * Copyright (c) 2004-2006 Milan Cutka
|
||||
// * based on mplayer HRTF plugin by ylai
|
||||
|
||||
#include <functional>
|
||||
#include <vector>
|
||||
#include <math.h>
|
||||
#include <stdlib.h>
|
||||
#include <string.h>
|
||||
#include "DPL2Decoder.h"
|
||||
|
||||
#define M_PI 3.14159265358979323846
|
||||
#define M_SQRT1_2 0.70710678118654752440
|
||||
|
||||
int olddelay = -1;
|
||||
unsigned int oldfreq = 0;
|
||||
unsigned int dlbuflen;
|
||||
int cyc_pos;
|
||||
float l_fwr, r_fwr, lpr_fwr, lmr_fwr;
|
||||
std::vector<float> fwrbuf_l, fwrbuf_r;
|
||||
float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain;
|
||||
std::vector<float> lf, rf, lr, rr, cf, cr;
|
||||
float LFE_buf[256];
|
||||
unsigned int lfe_pos;
|
||||
float *filter_coefs_lfe;
|
||||
unsigned int len125;
|
||||
|
||||
template<class T,class _ftype_t> static _ftype_t dotproduct(int count,const T *buf,const _ftype_t *coefficients)
|
||||
{
|
||||
float sum0=0,sum1=0,sum2=0,sum3=0;
|
||||
for (;count>=4;buf+=4,coefficients+=4,count-=4)
|
||||
{
|
||||
sum0+=buf[0]*coefficients[0];
|
||||
sum1+=buf[1]*coefficients[1];
|
||||
sum2+=buf[2]*coefficients[2];
|
||||
sum3+=buf[3]*coefficients[3];
|
||||
}
|
||||
while (count--) sum0+= *buf++ * *coefficients++;
|
||||
return sum0+sum1+sum2+sum3;
|
||||
}
|
||||
|
||||
template<class T> static T firfilter(const T *buf, int pos, int len, int count, const float *coefficients)
|
||||
{
|
||||
int count1, count2;
|
||||
|
||||
if (pos >= count)
|
||||
{
|
||||
pos -= count;
|
||||
count1 = count; count2 = 0;
|
||||
}
|
||||
else
|
||||
{
|
||||
count2 = pos;
|
||||
count1 = count - pos;
|
||||
pos = len - count1;
|
||||
}
|
||||
|
||||
// high part of window
|
||||
const T *ptr = &buf[pos];
|
||||
|
||||
float r1=dotproduct(count1,ptr,coefficients);coefficients+=count1;
|
||||
float r2=dotproduct(count2,buf,coefficients);
|
||||
return T(r1+r2);
|
||||
}
|
||||
|
||||
template<class T> inline const T& limit(const T& val, const T& min, const T& max)
|
||||
{
|
||||
if (val < min) {
|
||||
return min;
|
||||
} else if (val > max) {
|
||||
return max;
|
||||
} else {
|
||||
return val;
|
||||
}
|
||||
}
|
||||
|
||||
/*
|
||||
// Hamming
|
||||
// 2*pi*k
|
||||
// w(k) = 0.54 - 0.46*cos(------), where 0 <= k < N
|
||||
// N-1
|
||||
//
|
||||
// n window length
|
||||
// w buffer for the window parameters
|
||||
*/
|
||||
void hamming(int n, float* w)
|
||||
{
|
||||
int i;
|
||||
float k = float(2*M_PI/((float)(n-1))); // 2*pi/(N-1)
|
||||
|
||||
// Calculate window coefficients
|
||||
for (i=0; i<n; i++)
|
||||
*w++ = float(0.54 - 0.46*cos(k*(float)i));
|
||||
}
|
||||
|
||||
/******************************************************************************
|
||||
* FIR filter design
|
||||
******************************************************************************/
|
||||
|
||||
/* Design FIR filter using the Window method
|
||||
|
||||
n filter length must be odd for HP and BS filters
|
||||
w buffer for the filter taps (must be n long)
|
||||
fc cutoff frequencies (1 for LP and HP, 2 for BP and BS)
|
||||
0 < fc < 1 where 1 <=> Fs/2
|
||||
flags window and filter type as defined in filter.h
|
||||
variables are ored together: i.e. LP|HAMMING will give a
|
||||
low pass filter designed using a hamming window
|
||||
opt beta constant used only when designing using kaiser windows
|
||||
|
||||
returns 0 if OK, -1 if fail
|
||||
*/
|
||||
float* design_fir(unsigned int *n, float* fc, float opt)
|
||||
{
|
||||
unsigned int o = *n & 1; // Indicator for odd filter length
|
||||
unsigned int end = ((*n + 1) >> 1) - o; // Loop end
|
||||
unsigned int i; // Loop index
|
||||
|
||||
float k1 = 2 * float(M_PI); // 2*pi*fc1
|
||||
float k2 = 0.5f * (float)(1 - o);// Constant used if the filter has even length
|
||||
float g = 0.0f; // Gain
|
||||
float t1; // Temporary variables
|
||||
float fc1; // Cutoff frequencies
|
||||
|
||||
// Sanity check
|
||||
if(*n==0) return NULL;
|
||||
fc[0]=limit(fc[0],float(0.001),float(1));
|
||||
|
||||
float *w=(float*)calloc(sizeof(float),*n);
|
||||
|
||||
// Get window coefficients
|
||||
hamming(*n,w);
|
||||
|
||||
fc1=*fc;
|
||||
// Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2
|
||||
fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25f;
|
||||
k1 *= fc1;
|
||||
|
||||
// Low pass filter
|
||||
|
||||
// If the filter length is odd, there is one point which is exactly
|
||||
// in the middle. The value at this point is 2*fCutoff*sin(x)/x,
|
||||
// where x is zero. To make sure nothing strange happens, we set this
|
||||
// value separately.
|
||||
if (o)
|
||||
{
|
||||
w[end] = fc1 * w[end] * 2.0f;
|
||||
g=w[end];
|
||||
}
|
||||
|
||||
// Create filter
|
||||
for (i=0 ; i<end ; i++)
|
||||
{
|
||||
t1 = (float)(i+1) - k2;
|
||||
w[end-i-1] = w[*n-end+i] = float(w[end-i-1] * sin(k1 * t1)/(M_PI * t1)); // Sinc
|
||||
g += 2*w[end-i-1]; // Total gain in filter
|
||||
}
|
||||
|
||||
|
||||
// Normalize gain
|
||||
g=1/g;
|
||||
for (i=0; i<*n; i++)
|
||||
w[i] *= g;
|
||||
|
||||
return w;
|
||||
}
|
||||
|
||||
void onSeek(void)
|
||||
{
|
||||
l_fwr = r_fwr = lpr_fwr = lmr_fwr = 0;
|
||||
std::fill(fwrbuf_l.begin(), fwrbuf_l.end(), 0.0f);
|
||||
std::fill(fwrbuf_r.begin(), fwrbuf_r.end(), 0.0f);
|
||||
adapt_l_gain = adapt_r_gain = adapt_lpr_gain = adapt_lmr_gain = 0;
|
||||
std::fill(lf.begin(), lf.end(), 0.0f);
|
||||
std::fill(rf.begin(), rf.end(), 0.0f);
|
||||
std::fill(lr.begin(), lr.end(), 0.0f);
|
||||
std::fill(rr.begin(), rr.end(), 0.0f);
|
||||
std::fill(cf.begin(), cf.end(), 0.0f);
|
||||
std::fill(cr.begin(), cr.end(), 0.0f);
|
||||
lfe_pos = 0;
|
||||
memset(LFE_buf, 0, sizeof(LFE_buf));
|
||||
}
|
||||
|
||||
void done(void)
|
||||
{
|
||||
onSeek();
|
||||
if (filter_coefs_lfe)
|
||||
{
|
||||
free(filter_coefs_lfe);
|
||||
}
|
||||
filter_coefs_lfe = NULL;
|
||||
}
|
||||
|
||||
float* calc_coefficients_125Hz_lowpass(int rate)
|
||||
{
|
||||
len125 = 256;
|
||||
float f = 125.0f / (rate / 2);
|
||||
float *coeffs = design_fir(&len125, &f, 0);
|
||||
static const float M3_01DB = 0.7071067812f;
|
||||
for (unsigned int i = 0; i < len125; i++)
|
||||
{
|
||||
coeffs[i] *= M3_01DB;
|
||||
}
|
||||
return coeffs;
|
||||
}
|
||||
|
||||
float passive_lock(float x)
|
||||
{
|
||||
static const float MATAGCLOCK = 0.2f; /* AGC range (around 1) where the matrix behaves passively */
|
||||
const float x1 = x - 1;
|
||||
const float ax1s = fabs(x - 1) * (1.0f / MATAGCLOCK);
|
||||
return x1 - x1 / (1 + ax1s * ax1s) + 1;
|
||||
}
|
||||
|
||||
void matrix_decode(const float *in, const int k, const int il,
|
||||
const int ir, bool decode_rear,
|
||||
const int dlbuflen,
|
||||
float l_fwr, float r_fwr,
|
||||
float lpr_fwr, float lmr_fwr,
|
||||
float *adapt_l_gain, float *adapt_r_gain,
|
||||
float *adapt_lpr_gain, float *adapt_lmr_gain,
|
||||
float *lf, float *rf, float *lr,
|
||||
float *rr, float *cf)
|
||||
{
|
||||
static const float M9_03DB = 0.3535533906f;
|
||||
static const float MATAGCTRIG = 8.0f; /* (Fuzzy) AGC trigger */
|
||||
static const float MATAGCDECAY = 1.0f; /* AGC baseline decay rate (1/samp.) */
|
||||
static const float MATCOMPGAIN = 0.37f; /* Cross talk compensation gain, 0.50 - 0.55 is full cancellation. */
|
||||
|
||||
const int kr = (k + olddelay) % dlbuflen;
|
||||
float l_gain = (l_fwr + r_fwr) / (1 + l_fwr + l_fwr);
|
||||
float r_gain = (l_fwr + r_fwr) / (1 + r_fwr + r_fwr);
|
||||
/* The 2nd axis has strong gain fluctuations, and therefore require
|
||||
limits. The factor corresponds to the 1 / amplification of (Lt
|
||||
- Rt) when (Lt, Rt) is strongly correlated. (e.g. during
|
||||
dialogues). It should be bigger than -12 dB to prevent
|
||||
distortion. */
|
||||
float lmr_lim_fwr = lmr_fwr > M9_03DB * lpr_fwr ? lmr_fwr : M9_03DB * lpr_fwr;
|
||||
float lpr_gain = (lpr_fwr + lmr_lim_fwr) / (1 + lpr_fwr + lpr_fwr);
|
||||
float lmr_gain = (lpr_fwr + lmr_lim_fwr) / (1 + lmr_lim_fwr + lmr_lim_fwr);
|
||||
float lmr_unlim_gain = (lpr_fwr + lmr_fwr) / (1 + lmr_fwr + lmr_fwr);
|
||||
float lpr, lmr;
|
||||
float l_agc, r_agc, lpr_agc, lmr_agc;
|
||||
float f, d_gain, c_gain, c_agc_cfk;
|
||||
|
||||
/*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/
|
||||
/* AGC adaption */
|
||||
d_gain = (fabs(l_gain - *adapt_l_gain) + fabs(r_gain - *adapt_r_gain)) * 0.5f;
|
||||
f = d_gain * (1.0f / MATAGCTRIG);
|
||||
f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
|
||||
*adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain;
|
||||
*adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain;
|
||||
/* Matrix */
|
||||
l_agc = in[il] * passive_lock(*adapt_l_gain);
|
||||
r_agc = in[ir] * passive_lock(*adapt_r_gain);
|
||||
cf[k] = (l_agc + r_agc) * (float)M_SQRT1_2;
|
||||
if (decode_rear)
|
||||
{
|
||||
lr[kr] = rr[kr] = (l_agc - r_agc) * (float)M_SQRT1_2;
|
||||
/* Stereo rear channel is steered with the same AGC steering as
|
||||
the decoding matrix. Note this requires a fast updating AGC
|
||||
at the order of 20 ms (which is the case here). */
|
||||
lr[kr] *= (l_fwr + l_fwr) / (1 + l_fwr + r_fwr);
|
||||
rr[kr] *= (r_fwr + r_fwr) / (1 + l_fwr + r_fwr);
|
||||
}
|
||||
|
||||
/*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/
|
||||
lpr = (in[il] + in[ir]) * (float)M_SQRT1_2;
|
||||
lmr = (in[il] - in[ir]) * (float)M_SQRT1_2;
|
||||
/* AGC adaption */
|
||||
d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain);
|
||||
f = d_gain * (1.0f / MATAGCTRIG);
|
||||
f = MATAGCDECAY - MATAGCDECAY / (1 + f * f);
|
||||
*adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain;
|
||||
*adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain;
|
||||
/* Matrix */
|
||||
lpr_agc = lpr * passive_lock(*adapt_lpr_gain);
|
||||
lmr_agc = lmr * passive_lock(*adapt_lmr_gain);
|
||||
lf[k] = (lpr_agc + lmr_agc) * (float)M_SQRT1_2;
|
||||
rf[k] = (lpr_agc - lmr_agc) * (float)M_SQRT1_2;
|
||||
|
||||
/*** CENTER FRONT CANCELLATION ***/
|
||||
/* A heuristic approach exploits that Lt + Rt gain contains the
|
||||
information about Lt, Rt correlation. This effectively reshapes
|
||||
the front and rear "cones" to concentrate Lt + Rt to C and
|
||||
introduce Lt - Rt in L, R. */
|
||||
/* 0.67677 is the empirical lower bound for lpr_gain. */
|
||||
c_gain = 8 * (*adapt_lpr_gain - 0.67677f);
|
||||
c_gain = c_gain > 0 ? c_gain : 0;
|
||||
/* c_gain should not be too high, not even reaching full
|
||||
cancellation (~ 0.50 - 0.55 at current AGC implementation), or
|
||||
the center will sound too narrow. */
|
||||
c_gain = MATCOMPGAIN / (1 + c_gain * c_gain);
|
||||
c_agc_cfk = c_gain * cf[k];
|
||||
lf[k] -= c_agc_cfk;
|
||||
rf[k] -= c_agc_cfk;
|
||||
cf[k] += c_agc_cfk + c_agc_cfk;
|
||||
}
|
||||
|
||||
void dpl2decode(float *samples, int numsamples, float *out)
|
||||
{
|
||||
static const unsigned int FWRDURATION = 240; /* FWR average duration (samples) */
|
||||
static const int cfg_delay = 0;
|
||||
static const unsigned int fmt_freq = 48000;
|
||||
static const unsigned int fmt_nchannels = 2; // input channels
|
||||
|
||||
int cur = 0;
|
||||
|
||||
if (olddelay != cfg_delay || oldfreq != fmt_freq)
|
||||
{
|
||||
done();
|
||||
olddelay = cfg_delay;
|
||||
oldfreq = fmt_freq;
|
||||
dlbuflen = std::max(FWRDURATION, (fmt_freq * cfg_delay / 1000)); //+(len7000-1);
|
||||
cyc_pos = dlbuflen - 1;
|
||||
fwrbuf_l.resize(dlbuflen);
|
||||
fwrbuf_r.resize(dlbuflen);
|
||||
lf.resize(dlbuflen);
|
||||
rf.resize(dlbuflen);
|
||||
lr.resize(dlbuflen);
|
||||
rr.resize(dlbuflen);
|
||||
cf.resize(dlbuflen);
|
||||
cr.resize(dlbuflen);
|
||||
filter_coefs_lfe = calc_coefficients_125Hz_lowpass(fmt_freq);
|
||||
lfe_pos = 0;
|
||||
memset(LFE_buf, 0, sizeof(LFE_buf));
|
||||
}
|
||||
|
||||
float *in = samples; // Input audio data
|
||||
float *end = in + numsamples * fmt_nchannels; // Loop end
|
||||
|
||||
while (in < end)
|
||||
{
|
||||
const int k = cyc_pos;
|
||||
|
||||
const int fwr_pos = (k + FWRDURATION) % dlbuflen;
|
||||
/* Update the full wave rectified total amplitude */
|
||||
/* Input matrix decoder */
|
||||
l_fwr += fabs(in[0]) - fabs(fwrbuf_l[fwr_pos]);
|
||||
r_fwr += fabs(in[1]) - fabs(fwrbuf_r[fwr_pos]);
|
||||
lpr_fwr += fabs(in[0] + in[1]) - fabs(fwrbuf_l[fwr_pos] + fwrbuf_r[fwr_pos]);
|
||||
lmr_fwr += fabs(in[0] - in[1]) - fabs(fwrbuf_l[fwr_pos] - fwrbuf_r[fwr_pos]);
|
||||
|
||||
/* Matrix encoded 2 channel sources */
|
||||
fwrbuf_l[k] = in[0];
|
||||
fwrbuf_r[k] = in[1];
|
||||
matrix_decode(in, k, 0, 1, true, dlbuflen,
|
||||
l_fwr, r_fwr,
|
||||
lpr_fwr, lmr_fwr,
|
||||
&adapt_l_gain, &adapt_r_gain,
|
||||
&adapt_lpr_gain, &adapt_lmr_gain,
|
||||
&lf[0], &rf[0], &lr[0], &rr[0], &cf[0]);
|
||||
|
||||
out[cur + 0] = lf[k];
|
||||
out[cur + 1] = rf[k];
|
||||
out[cur + 2] = cf[k];
|
||||
LFE_buf[lfe_pos] = (out[0] + out[1]) / 2;
|
||||
out[cur + 3] = firfilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe);
|
||||
lfe_pos++;
|
||||
if (lfe_pos == len125)
|
||||
{
|
||||
lfe_pos = 0;
|
||||
}
|
||||
out[cur + 4] = lr[k];
|
||||
out[cur + 5] = rr[k];
|
||||
// Next sample...
|
||||
in += 2;
|
||||
cur += 6;
|
||||
cyc_pos--;
|
||||
if (cyc_pos < 0)
|
||||
{
|
||||
cyc_pos += dlbuflen;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void dpl2reset()
|
||||
{
|
||||
olddelay = -1;
|
||||
oldfreq = 0;
|
||||
filter_coefs_lfe = NULL;
|
||||
}
|
||||
|
|
@ -0,0 +1,24 @@
|
|||
// Copyright (C) 2003 Dolphin Project.
|
||||
|
||||
// This program is free software: you can redistribute it and/or modify
|
||||
// it under the terms of the GNU General Public License as published by
|
||||
// the Free Software Foundation, version 2.0.
|
||||
|
||||
// This program is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
||||
// GNU General Public License 2.0 for more details.
|
||||
|
||||
// A copy of the GPL 2.0 should have been included with the program.
|
||||
// If not, see http://www.gnu.org/licenses/
|
||||
|
||||
// Official SVN repository and contact information can be found at
|
||||
// http://code.google.com/p/dolphin-emu/
|
||||
|
||||
#ifndef _DPL2DECODER_H_
|
||||
#define _DPL2DECODER_H_
|
||||
|
||||
void dpl2decode(float *samples, int numsamples, float *out);
|
||||
void dpl2reset();
|
||||
|
||||
#endif // _DPL2DECODER_H_
|
||||
|
|
@ -92,6 +92,9 @@ public:
|
|||
|
||||
std::mutex& MixerCritical() { return m_csMixing; }
|
||||
|
||||
volatile float GetCurrentSpeed() const { return m_speed; }
|
||||
void UpdateSpeed(volatile float val) { m_speed = val; }
|
||||
|
||||
protected:
|
||||
unsigned int m_sampleRate;
|
||||
unsigned int m_aiSampleRate;
|
||||
|
|
@ -113,6 +116,8 @@ protected:
|
|||
|
||||
bool m_AIplaying;
|
||||
std::mutex m_csMixing;
|
||||
|
||||
volatile float m_speed; // Current rate of the emulation (1.0 = 100% speed)
|
||||
private:
|
||||
|
||||
};
|
||||
|
|
|
|||
|
|
@ -15,13 +15,14 @@
|
|||
// Official SVN repository and contact information can be found at
|
||||
// http://code.google.com/p/dolphin-emu/
|
||||
|
||||
#include <functional>
|
||||
|
||||
#include "aldlist.h"
|
||||
#include "OpenALStream.h"
|
||||
#include "DPL2Decoder.h"
|
||||
|
||||
#if defined HAVE_OPENAL && HAVE_OPENAL
|
||||
|
||||
soundtouch::SoundTouch soundTouch;
|
||||
|
||||
//
|
||||
// AyuanX: Spec says OpenAL1.1 is thread safe already
|
||||
//
|
||||
|
|
@ -35,14 +36,21 @@ bool OpenALStream::Start()
|
|||
pDeviceList = new ALDeviceList();
|
||||
if ((pDeviceList) && (pDeviceList->GetNumDevices()))
|
||||
{
|
||||
char *defDevName = pDeviceList-> \
|
||||
GetDeviceName(pDeviceList->GetDefaultDevice());
|
||||
char *defDevName = pDeviceList->GetDeviceName(pDeviceList->GetDefaultDevice());
|
||||
|
||||
WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);
|
||||
|
||||
pDevice = alcOpenDevice(defDevName);
|
||||
if (pDevice)
|
||||
{
|
||||
pContext = alcCreateContext(pDevice, NULL);
|
||||
if (pContext)
|
||||
{
|
||||
// Used to determine an appropriate period size (2x period = total buffer size)
|
||||
//ALCint refresh;
|
||||
//alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
|
||||
//period_size_in_millisec = 1000 / refresh;
|
||||
|
||||
alcMakeContextCurrent(pContext);
|
||||
thread = std::thread(std::mem_fun(&OpenALStream::SoundLoop), this);
|
||||
bReturn = true;
|
||||
|
|
@ -50,8 +58,7 @@ bool OpenALStream::Start()
|
|||
else
|
||||
{
|
||||
alcCloseDevice(pDevice);
|
||||
PanicAlertT("OpenAL: can't create context "
|
||||
"for device %s", defDevName);
|
||||
PanicAlertT("OpenAL: can't create context for device %s", defDevName);
|
||||
}
|
||||
}
|
||||
else
|
||||
|
|
@ -65,6 +72,10 @@ bool OpenALStream::Start()
|
|||
PanicAlertT("OpenAL: can't find sound devices");
|
||||
}
|
||||
|
||||
// Initialise DPL2 parameters
|
||||
dpl2reset();
|
||||
|
||||
soundTouch.clear();
|
||||
return bReturn;
|
||||
}
|
||||
|
||||
|
|
@ -74,6 +85,8 @@ void OpenALStream::Stop()
|
|||
// kick the thread if it's waiting
|
||||
soundSyncEvent.Set();
|
||||
|
||||
soundTouch.clear();
|
||||
|
||||
thread.join();
|
||||
|
||||
alSourceStop(uiSource);
|
||||
|
|
@ -82,7 +95,7 @@ void OpenALStream::Stop()
|
|||
// Clean up buffers and sources
|
||||
alDeleteSources(1, &uiSource);
|
||||
uiSource = 0;
|
||||
alDeleteBuffers(OAL_NUM_BUFFERS, uiBuffers);
|
||||
alDeleteBuffers(numBuffers, uiBuffers);
|
||||
|
||||
ALCcontext *pContext = alcGetCurrentContext();
|
||||
ALCdevice *pDevice = alcGetContextsDevice(pContext);
|
||||
|
|
@ -111,6 +124,7 @@ void OpenALStream::Clear(bool mute)
|
|||
|
||||
if(m_muted)
|
||||
{
|
||||
soundTouch.clear();
|
||||
alSourceStop(uiSource);
|
||||
}
|
||||
else
|
||||
|
|
@ -124,20 +138,29 @@ void OpenALStream::SoundLoop()
|
|||
Common::SetCurrentThreadName("Audio thread - openal");
|
||||
|
||||
u32 ulFrequency = m_mixer->GetSampleRate();
|
||||
numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
|
||||
|
||||
memset(uiBuffers, 0, OAL_NUM_BUFFERS * sizeof(ALuint));
|
||||
memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
|
||||
uiSource = 0;
|
||||
|
||||
// Generate some AL Buffers for streaming
|
||||
alGenBuffers(OAL_NUM_BUFFERS, (ALuint *)uiBuffers);
|
||||
alGenBuffers(numBuffers, (ALuint *)uiBuffers);
|
||||
// Generate a Source to playback the Buffers
|
||||
alGenSources(1, &uiSource);
|
||||
|
||||
// Short Silence
|
||||
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
|
||||
memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
|
||||
for (int i = 0; i < OAL_NUM_BUFFERS; i++)
|
||||
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, OAL_MAX_SAMPLES, ulFrequency);
|
||||
alSourceQueueBuffers(uiSource, OAL_NUM_BUFFERS, uiBuffers);
|
||||
for (int i = 0; i < numBuffers; i++)
|
||||
{
|
||||
#if !defined(__APPLE__)
|
||||
if (Core::g_CoreStartupParameter.bDPL2Decoder)
|
||||
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
|
||||
else
|
||||
#endif
|
||||
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
|
||||
}
|
||||
alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
|
||||
alSourcePlay(uiSource);
|
||||
|
||||
// Set the default sound volume as saved in the config file.
|
||||
|
|
@ -148,41 +171,154 @@ void OpenALStream::SoundLoop()
|
|||
|
||||
ALint iBuffersFilled = 0;
|
||||
ALint iBuffersProcessed = 0;
|
||||
ALuint uiBufferTemp[OAL_NUM_BUFFERS] = {0};
|
||||
ALint iState = 0;
|
||||
ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
|
||||
|
||||
soundTouch.setChannels(2);
|
||||
soundTouch.setSampleRate(ulFrequency);
|
||||
soundTouch.setTempo(1.0);
|
||||
soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
|
||||
soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
|
||||
soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
|
||||
soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
|
||||
soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
|
||||
|
||||
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
|
||||
|
||||
while (!threadData)
|
||||
{
|
||||
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
|
||||
const u32 stereo_16_bit_size = 4;
|
||||
const u32 dma_length = 32;
|
||||
const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
|
||||
u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
|
||||
u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
|
||||
|
||||
unsigned int numSamples = (unsigned int)num_samples_to_render;
|
||||
|
||||
numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
|
||||
numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
|
||||
|
||||
// Convert the samples from short to float
|
||||
float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
|
||||
for (u32 i = 0; i < numSamples; ++i)
|
||||
{
|
||||
dest[i * 2 + 0] = (float)realtimeBuffer[i * 2 + 0] / (1 << 16);
|
||||
dest[i * 2 + 1] = (float)realtimeBuffer[i * 2 + 1] / (1 << 16);
|
||||
}
|
||||
|
||||
soundTouch.putSamples(dest, numSamples);
|
||||
|
||||
if (iBuffersProcessed == iBuffersFilled)
|
||||
{
|
||||
alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
|
||||
iBuffersFilled = 0;
|
||||
}
|
||||
|
||||
unsigned int numSamples = m_mixer->GetNumSamples();
|
||||
|
||||
if (iBuffersProcessed && (numSamples >= OAL_THRESHOLD))
|
||||
if (iBuffersProcessed)
|
||||
{
|
||||
numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
|
||||
// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued Buffer)
|
||||
if (iBuffersFilled == 0)
|
||||
alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
|
||||
float rate = m_mixer->GetCurrentSpeed();
|
||||
if (rate <= 0)
|
||||
{
|
||||
Core::RequestRefreshInfo();
|
||||
rate = m_mixer->GetCurrentSpeed();
|
||||
}
|
||||
|
||||
m_mixer->Mix(realtimeBuffer, numSamples);
|
||||
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, realtimeBuffer, numSamples * 4, ulFrequency);
|
||||
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
|
||||
iBuffersFilled++;
|
||||
// Place a lower limit of 10% speed. When a game boots up, there will be
|
||||
// many silence samples. These do not need to be timestretched.
|
||||
if (rate > 0.10)
|
||||
{
|
||||
// Adjust SETTING_SEQUENCE_MS to balance between lag vs hollow audio
|
||||
soundTouch.setSetting(SETTING_SEQUENCE_MS, (int)(1 / (rate * rate)));
|
||||
soundTouch.setTempo(rate);
|
||||
}
|
||||
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS);
|
||||
if (nSamples > 0)
|
||||
{
|
||||
// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued Buffer)
|
||||
if (iBuffersFilled == 0)
|
||||
{
|
||||
alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
|
||||
ALenum err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
|
||||
}
|
||||
}
|
||||
#if defined(__APPLE__)
|
||||
// OSX does not have the alext AL_FORMAT_51CHN32 yet.
|
||||
surround_capable = false;
|
||||
#else
|
||||
if (surround_capable)
|
||||
{
|
||||
float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
|
||||
dpl2decode(sampleBuffer, nSamples, dpl2);
|
||||
|
||||
if (iBuffersFilled == OAL_NUM_BUFFERS)
|
||||
alSourcePlay(uiSource);
|
||||
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
|
||||
ALenum err = alGetError();
|
||||
if (err == AL_INVALID_ENUM)
|
||||
{
|
||||
// 5.1 is not supported by the host, fallback to stereo
|
||||
WARN_LOG(AUDIO, "Unable to set 5.1 surround mode. Updating OpenAL Soft might fix this issue.");
|
||||
surround_capable = false;
|
||||
}
|
||||
else if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
|
||||
}
|
||||
}
|
||||
#endif
|
||||
if (!surround_capable)
|
||||
{
|
||||
#if defined(__APPLE__)
|
||||
// Convert the samples from float to short
|
||||
short stereo[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
|
||||
for (u32 i = 0; i < nSamples; ++i)
|
||||
{
|
||||
stereo[i * 2 + 0] = (short)((float)sampleBuffer[i * 2 + 0] * (1 << 16));
|
||||
stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16));
|
||||
}
|
||||
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, ulFrequency);
|
||||
#else
|
||||
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, ulFrequency);
|
||||
#endif
|
||||
}
|
||||
|
||||
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
|
||||
ALenum err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
|
||||
}
|
||||
iBuffersFilled++;
|
||||
|
||||
if (iBuffersFilled == numBuffers)
|
||||
{
|
||||
alSourcePlay(uiSource);
|
||||
ALenum err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
|
||||
}
|
||||
}
|
||||
|
||||
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
|
||||
if (iState != AL_PLAYING)
|
||||
{
|
||||
// Buffer underrun occurred, resume playback
|
||||
alSourcePlay(uiSource);
|
||||
ALenum err = alGetError();
|
||||
if (err != 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
else if (numSamples >= OAL_THRESHOLD)
|
||||
else
|
||||
{
|
||||
ALint state = 0;
|
||||
alGetSourcei(uiSource, AL_SOURCE_STATE, &state);
|
||||
if (state == AL_STOPPED)
|
||||
alSourcePlay(uiSource);
|
||||
soundSyncEvent.Wait();
|
||||
}
|
||||
soundSyncEvent.Wait();
|
||||
}
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -24,21 +24,30 @@
|
|||
|
||||
#if defined HAVE_OPENAL && HAVE_OPENAL
|
||||
#ifdef _WIN32
|
||||
#include "../../../../Externals/OpenAL/include/al.h"
|
||||
#include "../../../../Externals/OpenAL/include/alc.h"
|
||||
#include <OpenAL/include/al.h>
|
||||
#include <OpenAL/include/alc.h>
|
||||
#include <OpenAL/include/alext.h>
|
||||
#elif defined __APPLE__
|
||||
#include <OpenAL/al.h>
|
||||
#include <OpenAL/alc.h>
|
||||
#else
|
||||
#include <AL/al.h>
|
||||
#include <AL/alc.h>
|
||||
#include <AL/alext.h>
|
||||
#endif
|
||||
|
||||
#include "Core.h"
|
||||
#include "HW/SystemTimers.h"
|
||||
#include "HW/AudioInterface.h"
|
||||
#include <soundtouch/SoundTouch.h>
|
||||
#include <soundtouch/STTypes.h>
|
||||
|
||||
// 16 bit Stereo
|
||||
#define SFX_MAX_SOURCE 1
|
||||
#define OAL_NUM_BUFFERS 16
|
||||
#define OAL_MAX_SAMPLES 512 // AyuanX: Don't make it too large, as larger buffer means longer delay
|
||||
#define OAL_THRESHOLD 128 // Some games are quite sensitive to delay
|
||||
#define OAL_MAX_BUFFERS 32
|
||||
#define OAL_MAX_SAMPLES 256
|
||||
#define SURROUND_CHANNELS 6 // number of channels in surround mode
|
||||
#define SIZE_FLOAT 4 // size of a float in bytes
|
||||
#endif
|
||||
|
||||
class OpenALStream: public SoundStream
|
||||
|
|
@ -66,9 +75,12 @@ private:
|
|||
Common::Event soundSyncEvent;
|
||||
|
||||
short realtimeBuffer[OAL_MAX_SAMPLES * 2];
|
||||
ALuint uiBuffers[OAL_NUM_BUFFERS];
|
||||
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
|
||||
ALuint uiBuffers[OAL_MAX_BUFFERS];
|
||||
ALuint uiSource;
|
||||
ALfloat fVolume;
|
||||
|
||||
u8 numBuffers;
|
||||
#else
|
||||
public:
|
||||
OpenALStream(CMixer *mixer, void *hWnd = NULL): SoundStream(mixer) {}
|
||||
|
|
|
|||
|
|
@ -81,8 +81,9 @@ private:
|
|||
#define GC_ALIGNED16_DECL(x) __declspec(align(16)) x
|
||||
#define GC_ALIGNED64_DECL(x) __declspec(align(64)) x
|
||||
|
||||
// Since it is always around on windows
|
||||
// Since they are always around on windows
|
||||
#define HAVE_WX 1
|
||||
#define HAVE_OPENAL 1
|
||||
|
||||
#define HAVE_PORTAUDIO 1
|
||||
|
||||
|
|
|
|||
|
|
@ -226,6 +226,8 @@ void SConfig::SaveSettings()
|
|||
ini.Set("Core", "Apploader", m_LocalCoreStartupParameter.m_strApploader);
|
||||
ini.Set("Core", "EnableCheats", m_LocalCoreStartupParameter.bEnableCheats);
|
||||
ini.Set("Core", "SelectedLanguage", m_LocalCoreStartupParameter.SelectedLanguage);
|
||||
ini.Set("Core", "DPL2Decoder", m_LocalCoreStartupParameter.bDPL2Decoder);
|
||||
ini.Set("Core", "Latency", m_LocalCoreStartupParameter.iLatency);
|
||||
ini.Set("Core", "MemcardA", m_strMemoryCardA);
|
||||
ini.Set("Core", "MemcardB", m_strMemoryCardB);
|
||||
ini.Set("Core", "SlotA", m_EXIDevice[0]);
|
||||
|
|
@ -365,6 +367,8 @@ void SConfig::LoadSettings()
|
|||
ini.Get("Core", "Apploader", &m_LocalCoreStartupParameter.m_strApploader);
|
||||
ini.Get("Core", "EnableCheats", &m_LocalCoreStartupParameter.bEnableCheats, false);
|
||||
ini.Get("Core", "SelectedLanguage", &m_LocalCoreStartupParameter.SelectedLanguage, 0);
|
||||
ini.Get("Core", "DPL2Decoder", &m_LocalCoreStartupParameter.bDPL2Decoder, false);
|
||||
ini.Get("Core", "Latency", &m_LocalCoreStartupParameter.iLatency, 14);
|
||||
ini.Get("Core", "MemcardA", &m_strMemoryCardA);
|
||||
ini.Get("Core", "MemcardB", &m_strMemoryCardB);
|
||||
ini.Get("Core", "SlotA", (int*)&m_EXIDevice[0], EXIDEVICE_MEMORYCARD);
|
||||
|
|
|
|||
|
|
@ -682,6 +682,13 @@ void VideoThrottle()
|
|||
// Show message
|
||||
g_video_backend->UpdateFPSDisplay(SMessage.c_str());
|
||||
|
||||
// Update the audio timestretcher with the current speed
|
||||
if (soundStream)
|
||||
{
|
||||
CMixer* pMixer = soundStream->GetMixer();
|
||||
pMixer->UpdateSpeed((float)Speed / 100);
|
||||
}
|
||||
|
||||
if (_CoreParameter.bRenderToMain &&
|
||||
SConfig::GetInstance().m_InterfaceStatusbar) {
|
||||
Host_UpdateStatusBar(SMessage.c_str());
|
||||
|
|
|
|||
|
|
@ -48,6 +48,7 @@ SCoreStartupParameter::SCoreStartupParameter()
|
|||
bHLE_BS2(true), bEnableCheats(false),
|
||||
bMergeBlocks(false),
|
||||
bRunCompareServer(false), bRunCompareClient(false),
|
||||
bDPL2Decoder(false), iLatency(14),
|
||||
bMMU(false), bMMUBAT(false), iTLBHack(0), bVBeam(false),
|
||||
bFastDiscSpeed(false),
|
||||
SelectedLanguage(0), bWii(false), bDisableWiimoteSpeaker(false),
|
||||
|
|
@ -82,6 +83,8 @@ void SCoreStartupParameter::LoadDefaults()
|
|||
bMergeBlocks = false;
|
||||
SelectedLanguage = 0;
|
||||
bWii = false;
|
||||
bDPL2Decoder = false;
|
||||
iLatency = 14;
|
||||
|
||||
iPosX = 100;
|
||||
iPosY = 100;
|
||||
|
|
|
|||
|
|
@ -106,6 +106,9 @@ struct SCoreStartupParameter
|
|||
bool bEnableCheats;
|
||||
bool bMergeBlocks;
|
||||
|
||||
bool bDPL2Decoder;
|
||||
int iLatency;
|
||||
|
||||
bool bRunCompareServer;
|
||||
bool bRunCompareClient;
|
||||
|
||||
|
|
|
|||
|
|
@ -251,7 +251,7 @@ void DSPHLE::InitMixer()
|
|||
unsigned int AISampleRate, DACSampleRate;
|
||||
AudioInterface::Callback_GetSampleRate(AISampleRate, DACSampleRate);
|
||||
delete soundStream;
|
||||
soundStream = AudioCommon::InitSoundStream(new HLEMixer(this, AISampleRate, DACSampleRate, ac_Config.iFrequency), m_hWnd);
|
||||
soundStream = AudioCommon::InitSoundStream(new HLEMixer(this, AISampleRate, DACSampleRate, 48000), m_hWnd);
|
||||
if(!soundStream) PanicAlert("Error starting up sound stream");
|
||||
// Mixer is initialized
|
||||
m_InitMixer = true;
|
||||
|
|
|
|||
|
|
@ -203,7 +203,7 @@ void DSPLLE::InitMixer()
|
|||
unsigned int AISampleRate, DACSampleRate;
|
||||
AudioInterface::Callback_GetSampleRate(AISampleRate, DACSampleRate);
|
||||
delete soundStream;
|
||||
soundStream = AudioCommon::InitSoundStream(new CMixer(AISampleRate, DACSampleRate, ac_Config.iFrequency), m_hWnd);
|
||||
soundStream = AudioCommon::InitSoundStream(new CMixer(AISampleRate, DACSampleRate, 48000), m_hWnd);
|
||||
if(!soundStream) PanicAlert("Error starting up sound stream");
|
||||
// Mixer is initialized
|
||||
m_InitMixer = true;
|
||||
|
|
|
|||
|
|
@ -119,7 +119,8 @@ EVT_RADIOBOX(ID_DSPENGINE, CConfigMain::AudioSettingsChanged)
|
|||
EVT_CHECKBOX(ID_DSPTHREAD, CConfigMain::AudioSettingsChanged)
|
||||
EVT_CHECKBOX(ID_ENABLE_THROTTLE, CConfigMain::AudioSettingsChanged)
|
||||
EVT_CHECKBOX(ID_DUMP_AUDIO, CConfigMain::AudioSettingsChanged)
|
||||
EVT_CHOICE(ID_FREQUENCY, CConfigMain::AudioSettingsChanged)
|
||||
EVT_CHECKBOX(ID_DPL2DECODER, CConfigMain::AudioSettingsChanged)
|
||||
EVT_SLIDER(ID_LATENCY, CConfigMain::AudioSettingsChanged)
|
||||
EVT_CHOICE(ID_BACKEND, CConfigMain::AudioSettingsChanged)
|
||||
EVT_SLIDER(ID_VOLUME, CConfigMain::AudioSettingsChanged)
|
||||
|
||||
|
|
@ -215,6 +216,8 @@ void CConfigMain::UpdateGUI()
|
|||
// Disable stuff on AudioPage
|
||||
DSPEngine->Disable();
|
||||
DSPThread->Disable();
|
||||
DPL2Decoder->Disable();
|
||||
LatencySlider->Disable();
|
||||
|
||||
// Disable stuff on GamecubePage
|
||||
GCSystemLang->Disable();
|
||||
|
|
@ -360,8 +363,11 @@ void CConfigMain::InitializeGUIValues()
|
|||
VolumeText->SetLabel(wxString::Format(wxT("%d %%"), ac_Config.m_Volume));
|
||||
DSPThread->SetValue(startup_params.bDSPThread);
|
||||
DumpAudio->SetValue(ac_Config.m_DumpAudio ? true : false);
|
||||
FrequencySelection->SetSelection(
|
||||
FrequencySelection->FindString(wxString::Format(_("%d Hz"), ac_Config.iFrequency)));
|
||||
DPL2Decoder->Enable(std::string(ac_Config.sBackend) == BACKEND_OPENAL);
|
||||
DPL2Decoder->SetValue(startup_params.bDPL2Decoder);
|
||||
LatencySlider->Enable(std::string(ac_Config.sBackend) == BACKEND_OPENAL);
|
||||
LatencySlider->SetValue(startup_params.iLatency);
|
||||
LatencyText->SetLabel(wxString::Format(wxT("%d"), startup_params.iLatency));
|
||||
// add backends to the list
|
||||
AddAudioBackends();
|
||||
|
||||
|
|
@ -503,7 +509,6 @@ void CConfigMain::InitializeGUITooltips()
|
|||
|
||||
// Audio tooltips
|
||||
DSPThread->SetToolTip(_("Run DSP LLE on a dedicated thread (not recommended)."));
|
||||
FrequencySelection->SetToolTip(_("Changing this will have no effect while the emulator is running!"));
|
||||
BackendSelection->SetToolTip(_("Changing this will have no effect while the emulator is running!"));
|
||||
|
||||
// Gamecube - Devices
|
||||
|
|
@ -511,6 +516,16 @@ void CConfigMain::InitializeGUITooltips()
|
|||
|
||||
// Wii - Devices
|
||||
WiiKeyboard->SetToolTip(_("This could cause slow down in Wii Menu and some games."));
|
||||
|
||||
#if defined(__APPLE__)
|
||||
DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. Not available on OSX."));
|
||||
#elif defined(__linux__)
|
||||
DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. OpenAL backend only."));
|
||||
#elif defined(_WIN32)
|
||||
DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. OpenAL backend only. May need to rename soft_oal.dll to OpenAL32.dll to make it work."));
|
||||
#endif
|
||||
|
||||
LatencySlider->SetToolTip(_("Sets the latency (in ms). Higher values may reduce audio crackling. OpenAL backend only."));
|
||||
}
|
||||
|
||||
void CConfigMain::CreateGUIControls()
|
||||
|
|
@ -608,20 +623,23 @@ void CConfigMain::CreateGUIControls()
|
|||
DSPThread = new wxCheckBox(AudioPage, ID_DSPTHREAD, _("DSP LLE on Thread"));
|
||||
DumpAudio = new wxCheckBox(AudioPage, ID_DUMP_AUDIO, _("Dump Audio"),
|
||||
wxDefaultPosition, wxDefaultSize, 0, wxDefaultValidator);
|
||||
DPL2Decoder = new wxCheckBox(AudioPage, ID_DPL2DECODER, _("Dolby Pro Logic II decoder"));
|
||||
VolumeSlider = new wxSlider(AudioPage, ID_VOLUME, 0, 1, 100,
|
||||
wxDefaultPosition, wxDefaultSize, wxSL_VERTICAL|wxSL_INVERSE);
|
||||
VolumeText = new wxStaticText(AudioPage, wxID_ANY, wxT(""),
|
||||
wxDefaultPosition, wxDefaultSize, 0);
|
||||
BackendSelection = new wxChoice(AudioPage, ID_BACKEND, wxDefaultPosition,
|
||||
wxDefaultSize, wxArrayBackends, 0, wxDefaultValidator, wxEmptyString);
|
||||
FrequencySelection = new wxChoice(AudioPage, ID_FREQUENCY);
|
||||
FrequencySelection->Append(wxString::Format(_("%d Hz"), 48000));
|
||||
FrequencySelection->Append(wxString::Format(_("%d Hz"), 32000));
|
||||
LatencySlider = new wxSlider(AudioPage, ID_LATENCY, 0, 0, 30,
|
||||
wxDefaultPosition, wxDefaultSize, wxSL_HORIZONTAL);
|
||||
LatencyText = new wxStaticText(AudioPage, wxID_ANY, wxT(""),
|
||||
wxDefaultPosition, wxDefaultSize, 0);
|
||||
|
||||
if (Core::GetState() != Core::CORE_UNINITIALIZED)
|
||||
{
|
||||
FrequencySelection->Disable();
|
||||
LatencySlider->Disable();
|
||||
BackendSelection->Disable();
|
||||
DPL2Decoder->Disable();
|
||||
}
|
||||
|
||||
// Create sizer and add items to dialog
|
||||
|
|
@ -629,6 +647,7 @@ void CConfigMain::CreateGUIControls()
|
|||
sbAudioSettings->Add(DSPEngine, 0, wxALL | wxEXPAND, 5);
|
||||
sbAudioSettings->Add(DSPThread, 0, wxALL, 5);
|
||||
sbAudioSettings->Add(DumpAudio, 0, wxALL, 5);
|
||||
sbAudioSettings->Add(DPL2Decoder, 0, wxALL, 5);
|
||||
|
||||
wxStaticBoxSizer *sbVolume = new wxStaticBoxSizer(wxVERTICAL, AudioPage, _("Volume"));
|
||||
sbVolume->Add(VolumeSlider, 1, wxLEFT|wxRIGHT, 13);
|
||||
|
|
@ -637,8 +656,9 @@ void CConfigMain::CreateGUIControls()
|
|||
wxGridBagSizer *sBackend = new wxGridBagSizer();
|
||||
sBackend->Add(TEXT_BOX(AudioPage, _("Audio Backend:")), wxGBPosition(0, 0), wxDefaultSpan, wxALIGN_CENTER_VERTICAL|wxALL, 5);
|
||||
sBackend->Add(BackendSelection, wxGBPosition(0, 1), wxDefaultSpan, wxALL, 5);
|
||||
sBackend->Add(TEXT_BOX(AudioPage, _("Sample Rate:")), wxGBPosition(1, 0), wxDefaultSpan, wxALIGN_CENTER_VERTICAL|wxALL, 5);
|
||||
sBackend->Add(FrequencySelection, wxGBPosition(1, 1), wxDefaultSpan, wxALL, 5);
|
||||
sBackend->Add(TEXT_BOX(AudioPage, _("Latency:")), wxGBPosition(1, 0), wxDefaultSpan, wxALIGN_CENTER_VERTICAL|wxALL, 5);
|
||||
sBackend->Add(LatencySlider, wxGBPosition(1, 1), wxDefaultSpan, wxALL, 5);
|
||||
sBackend->Add(LatencyText, wxGBPosition(1, 2), wxDefaultSpan, wxALL, 5);
|
||||
wxStaticBoxSizer *sbBackend = new wxStaticBoxSizer(wxHORIZONTAL, AudioPage, _("Backend Settings"));
|
||||
sbBackend->Add(sBackend, 0, wxEXPAND);
|
||||
|
||||
|
|
@ -919,19 +939,25 @@ void CConfigMain::AudioSettingsChanged(wxCommandEvent& event)
|
|||
SConfig::GetInstance().m_LocalCoreStartupParameter.bDSPThread = DSPThread->IsChecked();
|
||||
break;
|
||||
|
||||
case ID_DPL2DECODER:
|
||||
SConfig::GetInstance().m_LocalCoreStartupParameter.bDPL2Decoder = DPL2Decoder->IsChecked();
|
||||
break;
|
||||
|
||||
case ID_BACKEND:
|
||||
VolumeSlider->Enable(SupportsVolumeChanges(std::string(BackendSelection->GetStringSelection().mb_str())));
|
||||
LatencySlider->Enable(std::string(BackendSelection->GetStringSelection().mb_str()) == BACKEND_OPENAL);
|
||||
DPL2Decoder->Enable(std::string(BackendSelection->GetStringSelection().mb_str()) == BACKEND_OPENAL);
|
||||
ac_Config.sBackend = BackendSelection->GetStringSelection().mb_str();
|
||||
ac_Config.Update();
|
||||
break;
|
||||
|
||||
case ID_LATENCY:
|
||||
SConfig::GetInstance().m_LocalCoreStartupParameter.iLatency = LatencySlider->GetValue();
|
||||
LatencyText->SetLabel(wxString::Format(wxT("%d"), LatencySlider->GetValue()));
|
||||
break;
|
||||
|
||||
default:
|
||||
ac_Config.m_DumpAudio = DumpAudio->GetValue();
|
||||
|
||||
long int frequency;
|
||||
FrequencySelection->GetStringSelection().ToLong(&frequency);
|
||||
ac_Config.iFrequency = frequency;
|
||||
ac_Config.Update();
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -78,7 +78,8 @@ private:
|
|||
ID_ENABLE_HLE_AUDIO,
|
||||
ID_ENABLE_THROTTLE,
|
||||
ID_DUMP_AUDIO,
|
||||
ID_FREQUENCY,
|
||||
ID_DPL2DECODER,
|
||||
ID_LATENCY,
|
||||
ID_BACKEND,
|
||||
ID_VOLUME,
|
||||
|
||||
|
|
@ -155,9 +156,11 @@ private:
|
|||
wxSlider* VolumeSlider;
|
||||
wxStaticText* VolumeText;
|
||||
wxCheckBox* DumpAudio;
|
||||
wxCheckBox* DPL2Decoder;
|
||||
wxArrayString wxArrayBackends;
|
||||
wxChoice* BackendSelection;
|
||||
wxChoice* FrequencySelection;
|
||||
wxSlider* LatencySlider;
|
||||
wxStaticText* LatencyText;
|
||||
|
||||
// Interface
|
||||
wxCheckBox* ConfirmStop;
|
||||
|
|
|
|||
|
|
@ -108,6 +108,8 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "png", "..\Externals\libpng\
|
|||
EndProject
|
||||
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SCMRevGen", "Core\Common\SVNRevGen.vcxproj", "{69F00340-5C3D-449F-9A80-958435C6CF06}"
|
||||
EndProject
|
||||
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\Externals\SoundTouch\SoundTouch.vcxproj", "{68A5DD20-7057-448B-8FE0-B6AC8D205509}"
|
||||
EndProject
|
||||
Global
|
||||
GlobalSection(SolutionConfigurationPlatforms) = preSolution
|
||||
Debug|Win32 = Debug|Win32
|
||||
|
|
@ -382,6 +384,18 @@ Global
|
|||
{69F00340-5C3D-449F-9A80-958435C6CF06}.Release|Win32.Build.0 = Release|x64
|
||||
{69F00340-5C3D-449F-9A80-958435C6CF06}.Release|x64.ActiveCfg = Release|x64
|
||||
{69F00340-5C3D-449F-9A80-958435C6CF06}.Release|x64.Build.0 = Release|x64
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|Win32.ActiveCfg = Debug|Win32
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|Win32.Build.0 = Debug|Win32
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|x64.ActiveCfg = Debug|x64
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|x64.Build.0 = Debug|x64
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|Win32.ActiveCfg = Debug|Win32
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|Win32.Build.0 = Debug|Win32
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|x64.ActiveCfg = Release|x64
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|x64.Build.0 = Release|x64
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.ActiveCfg = Release|Win32
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.Build.0 = Release|Win32
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.ActiveCfg = Release|x64
|
||||
{68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.Build.0 = Release|x64
|
||||
EndGlobalSection
|
||||
GlobalSection(SolutionProperties) = preSolution
|
||||
HideSolutionNode = FALSE
|
||||
|
|
|
|||
Loading…
Reference in New Issue