mirror of
https://github.com/dolphin-emu/dolphin.git
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Merge branch 'master' into wii-network
Conflicts: Source/Core/Common/Src/CommonPaths.h Source/Core/Common/Src/FileUtil.h Source/Core/Core/Src/HLE/HLE.cpp Source/Core/Core/Src/PowerPC/Interpreter/Interpreter.cpp
This commit is contained in:
Matthew Parlane
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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 \
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{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 }
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#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 }
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#define REVERB_PRESET_DRUGGED \
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||||
{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 }
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#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 }
|
||||
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||||
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// 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 }
|
||||
656
Externals/OpenAL/include/al.h
vendored
Normal file
656
Externals/OpenAL/include/al.h
vendored
Normal file
@ -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 */
|
||||
237
Externals/OpenAL/include/alc.h
vendored
Normal file
237
Externals/OpenAL/include/alc.h
vendored
Normal file
@ -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 */
|
||||
355
Externals/OpenAL/include/alext.h
vendored
Normal file
355
Externals/OpenAL/include/alext.h
vendored
Normal file
@ -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
|
||||
3
Externals/OpenAL/include/efx-creative.h
vendored
Normal file
3
Externals/OpenAL/include/efx-creative.h
vendored
Normal file
@ -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. */
|
||||
402
Externals/OpenAL/include/efx-presets.h
vendored
Normal file
402
Externals/OpenAL/include/efx-presets.h
vendored
Normal file
@ -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 */
|
||||
761
Externals/OpenAL/include/efx.h
vendored
Normal file
761
Externals/OpenAL/include/efx.h
vendored
Normal file
@ -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 */
|
||||
94
Externals/OpenAL/include/xram.h
vendored
Normal file
94
Externals/OpenAL/include/xram.h
vendored
Normal file
@ -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(...);
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
184
Externals/soundtouch/AAFilter.cpp
vendored
Normal file
184
Externals/soundtouch/AAFilter.cpp
vendored
Normal file
@ -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();
|
||||
}
|
||||
91
Externals/soundtouch/AAFilter.h
vendored
Normal file
91
Externals/soundtouch/AAFilter.h
vendored
Normal file
@ -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
|
||||
370
Externals/soundtouch/BPMDetect.cpp
vendored
Normal file
370
Externals/soundtouch/BPMDetect.cpp
vendored
Normal file
@ -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);
|
||||
}
|
||||
164
Externals/soundtouch/BPMDetect.h
vendored
Normal file
164
Externals/soundtouch/BPMDetect.h
vendored
Normal file
@ -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_
|
||||
15
Externals/soundtouch/CMakeLists.txt
vendored
Normal file
15
Externals/soundtouch/CMakeLists.txt
vendored
Normal file
@ -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})
|
||||
274
Externals/soundtouch/FIFOSampleBuffer.cpp
vendored
Normal file
274
Externals/soundtouch/FIFOSampleBuffer.cpp
vendored
Normal file
@ -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;
|
||||
}
|
||||
|
||||
178
Externals/soundtouch/FIFOSampleBuffer.h
vendored
Normal file
178
Externals/soundtouch/FIFOSampleBuffer.h
vendored
Normal file
@ -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
|
||||
234
Externals/soundtouch/FIFOSamplePipe.h
vendored
Normal file
234
Externals/soundtouch/FIFOSamplePipe.h
vendored
Normal file
@ -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
|
||||
259
Externals/soundtouch/FIRFilter.cpp
vendored
Normal file
259
Externals/soundtouch/FIRFilter.cpp
vendored
Normal file
@ -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;
|
||||
}
|
||||
}
|
||||
145
Externals/soundtouch/FIRFilter.h
vendored
Normal file
145
Externals/soundtouch/FIRFilter.h
vendored
Normal file
@ -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
|
||||
276
Externals/soundtouch/PeakFinder.cpp
vendored
Normal file
276
Externals/soundtouch/PeakFinder.cpp
vendored
Normal file
@ -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 <20>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 <20>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;
|
||||
}
|
||||
97
Externals/soundtouch/PeakFinder.h
vendored
Normal file
97
Externals/soundtouch/PeakFinder.h
vendored
Normal file
@ -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_
|
||||
626
Externals/soundtouch/RateTransposer.cpp
vendored
Normal file
626
Externals/soundtouch/RateTransposer.cpp
vendored
Normal file
@ -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;
|
||||
}
|
||||
159
Externals/soundtouch/RateTransposer.h
vendored
Normal file
159
Externals/soundtouch/RateTransposer.h
vendored
Normal file
@ -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
|
||||
191
Externals/soundtouch/STTypes.h
vendored
Normal file
191
Externals/soundtouch/STTypes.h
vendored
Normal file
@ -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
|
||||
501
Externals/soundtouch/SoundTouch.cpp
vendored
Normal file
501
Externals/soundtouch/SoundTouch.cpp
vendored
Normal file
@ -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;
|
||||
}
|
||||
277
Externals/soundtouch/SoundTouch.h
vendored
Normal file
277
Externals/soundtouch/SoundTouch.h
vendored
Normal file
@ -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
|
||||
352
Externals/soundtouch/SoundTouch.vcxproj
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352
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||||
<BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
|
||||
<Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
|
||||
<Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
|
||||
<AdditionalIncludeDirectories Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories Condition="'$(Configuration)|$(Platform)'=='Release|x64'">%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<PreprocessorDefinitions Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">%(PreprocessorDefinitions)</PreprocessorDefinitions>
|
||||
<PreprocessorDefinitions Condition="'$(Configuration)|$(Platform)'=='Release|x64'">%(PreprocessorDefinitions)</PreprocessorDefinitions>
|
||||
</ClCompile>
|
||||
<ClCompile Include="sse_optimized.cpp" />
|
||||
<ClCompile Include="TDStretch.cpp">
|
||||
<Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">Disabled</Optimization>
|
||||
<Optimization Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">Disabled</Optimization>
|
||||
<AdditionalIncludeDirectories Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
|
||||
<AdditionalIncludeDirectories Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
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<PreprocessorDefinitions Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">%(PreprocessorDefinitions)</PreprocessorDefinitions>
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||||
<PreprocessorDefinitions Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">%(PreprocessorDefinitions)</PreprocessorDefinitions>
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||||
<BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">EnableFastChecks</BasicRuntimeChecks>
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<BasicRuntimeChecks Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">EnableFastChecks</BasicRuntimeChecks>
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<BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|Win32'">true</BrowseInformation>
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<BrowseInformation Condition="'$(Configuration)|$(Platform)'=='Debug|x64'">true</BrowseInformation>
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<Optimization Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">MaxSpeed</Optimization>
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<Optimization Condition="'$(Configuration)|$(Platform)'=='Release|x64'">MaxSpeed</Optimization>
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<AdditionalIncludeDirectories Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
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<AdditionalIncludeDirectories Condition="'$(Configuration)|$(Platform)'=='Release|x64'">%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
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<PreprocessorDefinitions Condition="'$(Configuration)|$(Platform)'=='Release|Win32'">%(PreprocessorDefinitions)</PreprocessorDefinitions>
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<PreprocessorDefinitions Condition="'$(Configuration)|$(Platform)'=='Release|x64'">%(PreprocessorDefinitions)</PreprocessorDefinitions>
|
||||
</ClCompile>
|
||||
<ClCompile Include="BPMDetect.cpp" />
|
||||
<ClCompile Include="PeakFinder.cpp" />
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<ClInclude Include="AAFilter.h" />
|
||||
<ClInclude Include="BPMDetect.h" />
|
||||
<ClInclude Include="cpu_detect.h" />
|
||||
<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" />
|
||||
</ItemGroup>
|
||||
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
|
||||
<ImportGroup Label="ExtensionTargets">
|
||||
</ImportGroup>
|
||||
</Project>
|
||||
60
Externals/soundtouch/SoundTouch.vcxproj.filters
vendored
Normal file
60
Externals/soundtouch/SoundTouch.vcxproj.filters
vendored
Normal file
@ -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>
|
||||
808
Externals/soundtouch/TDStretch.cpp
vendored
Normal file
808
Externals/soundtouch/TDStretch.cpp
vendored
Normal file
@ -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
|
||||
268
Externals/soundtouch/TDStretch.h
vendored
Normal file
268
Externals/soundtouch/TDStretch.h
vendored
Normal file
@ -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
|
||||
62
Externals/soundtouch/cpu_detect.h
vendored
Normal file
62
Externals/soundtouch/cpu_detect.h
vendored
Normal file
@ -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_
|
||||
137
Externals/soundtouch/cpu_detect_x86.cpp
vendored
Normal file
137
Externals/soundtouch/cpu_detect_x86.cpp
vendored
Normal file
@ -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
|
||||
}
|
||||
317
Externals/soundtouch/mmx_optimized.cpp
vendored
Normal file
317
Externals/soundtouch/mmx_optimized.cpp
vendored
Normal file
@ -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
|
||||
361
Externals/soundtouch/sse_optimized.cpp
vendored
Normal file
361
Externals/soundtouch/sse_optimized.cpp
vendored
Normal file
@ -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
|
||||
Reference in New Issue
Block a user