diff --git a/CMakeLists.txt b/CMakeLists.txt index 4b0d5c42dc..255e3c95eb 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -405,6 +405,19 @@ else() set(LZO lzo2) endif() +if(OPENAL_FOUND) + if(NOT ${CMAKE_SYSTEM_NAME} MATCHES "Darwin") + check_lib(SOUNDTOUCH SoundTouch soundtouch/soundtouch.h QUIET) + endif() + if (SOUNDTOUCH_FOUND) + message("Using shared soundtouch") + else() + message("Using static soundtouch from Externals") + add_subdirectory(Externals/soundtouch) + include_directories(Externals) + endif() +endif() + if(NOT ${CMAKE_SYSTEM_NAME} MATCHES "Darwin") include(FindSDL2 OPTIONAL) endif() diff --git a/Externals/OpenAL/Win32/EFX-Util.lib b/Externals/OpenAL/Win32/EFX-Util.lib new file mode 100644 index 0000000000..94ab9c817a Binary files /dev/null and b/Externals/OpenAL/Win32/EFX-Util.lib differ diff --git a/Externals/OpenAL/Win32/OpenAL32.dll b/Externals/OpenAL/Win32/OpenAL32.dll new file mode 100644 index 0000000000..b63bb2f301 Binary files /dev/null and b/Externals/OpenAL/Win32/OpenAL32.dll differ diff --git a/Externals/OpenAL/Win32/OpenAL32.lib b/Externals/OpenAL/Win32/OpenAL32.lib new file mode 100644 index 0000000000..d635de97ef Binary files /dev/null and b/Externals/OpenAL/Win32/OpenAL32.lib differ diff --git a/Externals/OpenAL/Win32/soft_oal.dll b/Externals/OpenAL/Win32/soft_oal.dll new file mode 100644 index 0000000000..71ced6a21f Binary files /dev/null and b/Externals/OpenAL/Win32/soft_oal.dll differ diff --git a/Externals/OpenAL/Win32/wrap_oal.dll b/Externals/OpenAL/Win32/wrap_oal.dll new file mode 100644 index 0000000000..9752892aab Binary files /dev/null and b/Externals/OpenAL/Win32/wrap_oal.dll differ diff --git a/Externals/OpenAL/Win64/EFX-Util.lib b/Externals/OpenAL/Win64/EFX-Util.lib new file mode 100644 index 0000000000..5c5e98177a Binary files /dev/null and b/Externals/OpenAL/Win64/EFX-Util.lib differ diff --git a/Externals/OpenAL/Win64/OpenAL32.dll b/Externals/OpenAL/Win64/OpenAL32.dll new file mode 100644 index 0000000000..f36943df40 Binary files /dev/null and b/Externals/OpenAL/Win64/OpenAL32.dll differ diff --git a/Externals/OpenAL/Win64/OpenAL32.lib b/Externals/OpenAL/Win64/OpenAL32.lib new file mode 100644 index 0000000000..17663e0a38 Binary files /dev/null and b/Externals/OpenAL/Win64/OpenAL32.lib differ diff --git a/Externals/OpenAL/Win64/soft_oal.dll b/Externals/OpenAL/Win64/soft_oal.dll new file mode 100644 index 0000000000..b47ff952b7 Binary files /dev/null and b/Externals/OpenAL/Win64/soft_oal.dll differ diff --git a/Externals/OpenAL/Win64/wrap_oal.dll b/Externals/OpenAL/Win64/wrap_oal.dll new file mode 100644 index 0000000000..1ab0365711 Binary files /dev/null and b/Externals/OpenAL/Win64/wrap_oal.dll differ diff --git a/Externals/OpenAL/include/EFX-Util.h b/Externals/OpenAL/include/EFX-Util.h new file mode 100644 index 0000000000..b4a6b4e2d8 --- /dev/null +++ b/Externals/OpenAL/include/EFX-Util.h @@ -0,0 +1,422 @@ +/*******************************************************************\ +* * +* EFX-UTIL.H - EFX Utilities functions and Reverb Presets * +* * +* File revision 1.0 * +* * +\*******************************************************************/ + +#ifndef EAXVECTOR_DEFINED +#define EAXVECTOR_DEFINED +typedef struct _EAXVECTOR { + float x; + float y; + float z; +} EAXVECTOR; +#endif + +#ifndef EAXREVERBPROPERTIES_DEFINED +#define EAXREVERBPROPERTIES_DEFINED +typedef struct _EAXREVERBPROPERTIES +{ + unsigned long ulEnvironment; + float flEnvironmentSize; + float flEnvironmentDiffusion; + long lRoom; + long lRoomHF; + long lRoomLF; + float flDecayTime; + float flDecayHFRatio; + float flDecayLFRatio; + long lReflections; + float flReflectionsDelay; + EAXVECTOR vReflectionsPan; + long lReverb; + float flReverbDelay; + EAXVECTOR vReverbPan; + float flEchoTime; + float flEchoDepth; + float flModulationTime; + float flModulationDepth; + float flAirAbsorptionHF; + float flHFReference; + float flLFReference; + float flRoomRolloffFactor; + unsigned long ulFlags; +} EAXREVERBPROPERTIES, *LPEAXREVERBPROPERTIES; +#endif + +#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 + +#ifndef EAXOBSTRUCTIONPROPERTIES_DEFINED +#define EAXOBSTRUCTIONPROPERTIES_DEFINED +typedef struct _EAXOBSTRUCTIONPROPERTIES +{ + long lObstruction; + float flObstructionLFRatio; +} EAXOBSTRUCTIONPROPERTIES, *LPEAXOBSTRUCTIONPROPERTIES; +#endif + +#ifndef EAXOCCLUSIONPROPERTIES_DEFINED +#define EAXOCCLUSIONPROPERTIES_DEFINED +typedef struct _EAXOCCLUSIONPROPERTIES +{ + long lOcclusion; + float flOcclusionLFRatio; + float flOcclusionRoomRatio; + float flOcclusionDirectRatio; +} EAXOCCLUSIONPROPERTIES, *LPEAXOCCLUSIONPROPERTIES; +#endif + +#ifndef EAXEXCLUSIONPROPERTIES_DEFINED +#define EAXEXCLUSIONPROPERTIES_DEFINED +typedef struct _EAXEXCLUSIONPROPERTIES +{ + long lExclusion; + float flExclusionLFRatio; +} EAXEXCLUSIONPROPERTIES, *LPEAXEXCLUSIONPROPERTIES; +#endif + +#ifndef EFXLOWPASSFILTER_DEFINED +#define EFXLOWPASSFILTER_DEFINED +typedef struct _EFXLOWPASSFILTER +{ + float flGain; + float flGainHF; +} EFXLOWPASSFILTER, *LPEFXLOWPASSFILTER; +#endif + +void ConvertReverbParameters(EAXREVERBPROPERTIES *pEAXProp, EFXEAXREVERBPROPERTIES *pEFXEAXReverb); +void ConvertObstructionParameters(EAXOBSTRUCTIONPROPERTIES *pObProp, EFXLOWPASSFILTER *pDirectLowPassFilter); +void ConvertExclusionParameters(EAXEXCLUSIONPROPERTIES *pExProp, EFXLOWPASSFILTER *pSendLowPassFilter); +void ConvertOcclusionParameters(EAXOCCLUSIONPROPERTIES *pOcProp, EFXLOWPASSFILTER *pDirectLowPassFilter, EFXLOWPASSFILTER *pSendLowPassFilter); + + +/***********************************************************************************************\ +* +* EAX Reverb Presets in legacy format - use ConvertReverbParameters() to convert to +* EFX EAX Reverb Presets for use with the OpenAL Effects Extension. +* +************************************************************************************************/ + +// 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_GENERIC \ + {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 } +#define REVERB_PRESET_PADDEDCELL \ + {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 } +#define REVERB_PRESET_ROOM \ + {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 } +#define REVERB_PRESET_BATHROOM \ + {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 } +#define REVERB_PRESET_LIVINGROOM \ + {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 } +#define REVERB_PRESET_STONEROOM \ + {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 } +#define REVERB_PRESET_AUDITORIUM \ + {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 } +#define REVERB_PRESET_CONCERTHALL \ + {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 } +#define REVERB_PRESET_CAVE \ + {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 } +#define REVERB_PRESET_ARENA \ + {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 } +#define REVERB_PRESET_HANGAR \ + {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 } +#define REVERB_PRESET_CARPETTEDHALLWAY \ + {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 } +#define REVERB_PRESET_HALLWAY \ + {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 } +#define REVERB_PRESET_STONECORRIDOR \ + {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 } +#define REVERB_PRESET_ALLEY \ + {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 } +#define REVERB_PRESET_FOREST \ + {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 } +#define REVERB_PRESET_CITY \ + {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 } +#define REVERB_PRESET_MOUNTAINS \ + {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 } +#define REVERB_PRESET_QUARRY \ + {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 } +#define REVERB_PRESET_PLAIN \ + {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 } +#define REVERB_PRESET_PARKINGLOT \ + {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 } +#define REVERB_PRESET_SEWERPIPE \ + {21, 1.7f, 0.800f, -1000, -1000, 0, 2.81f, 0.14f, 1.00f, 429, 0.014f, 0.00f,0.00f,0.00f, 1023, 0.021f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f } +#define REVERB_PRESET_UNDERWATER \ + {22, 1.8f, 1.000f, -1000, -4000, 0, 1.49f, 0.10f, 1.00f, -449, 0.007f, 0.00f,0.00f,0.00f, 1700, 0.011f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 1.180f, 0.348f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f } +#define REVERB_PRESET_DRUGGED \ + {23, 1.9f, 0.500f, -1000, 0, 0, 8.39f, 1.39f, 1.00f, -115, 0.002f, 0.00f,0.00f,0.00f, 985, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 1.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f } +#define REVERB_PRESET_DIZZY \ + {24, 1.8f, 0.600f, -1000, -400, 0, 17.23f, 0.56f, 1.00f, -1713, 0.020f, 0.00f,0.00f,0.00f, -613, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.810f, 0.310f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f } +#define REVERB_PRESET_PSYCHOTIC \ + {25, 1.0f, 0.500f, -1000, -151, 0, 7.56f, 0.91f, 1.00f, -626, 0.020f, 0.00f,0.00f,0.00f, 774, 0.030f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 4.000f, 1.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f } + + +// CASTLE PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_CASTLE_SMALLROOM \ + { 26, 8.3f, 0.890f, -1000, -800, -2000, 1.22f, 0.83f, 0.31f, -100, 0.022f, 0.00f,0.00f,0.00f, 600, 0.011f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } +#define REVERB_PRESET_CASTLE_SHORTPASSAGE \ + { 26, 8.3f, 0.890f, -1000, -1000, -2000, 2.32f, 0.83f, 0.31f, -100, 0.007f, 0.00f,0.00f,0.00f, 200, 0.023f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } +#define REVERB_PRESET_CASTLE_MEDIUMROOM \ + { 26, 8.3f, 0.930f, -1000, -1100, -2000, 2.04f, 0.83f, 0.46f, -400, 0.022f, 0.00f,0.00f,0.00f, 400, 0.011f, 0.00f,0.00f,0.00f, 0.155f, 0.030f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } +#define REVERB_PRESET_CASTLE_LONGPASSAGE \ + { 26, 8.3f, 0.890f, -1000, -800, -2000, 3.42f, 0.83f, 0.31f, -100, 0.007f, 0.00f,0.00f,0.00f, 300, 0.023f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } +#define REVERB_PRESET_CASTLE_LARGEROOM \ + { 26, 8.3f, 0.820f, -1000, -1100, -1800, 2.53f, 0.83f, 0.50f, -700, 0.034f, 0.00f,0.00f,0.00f, 200, 0.016f, 0.00f,0.00f,0.00f, 0.185f, 0.070f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } +#define REVERB_PRESET_CASTLE_HALL \ + { 26, 8.3f, 0.810f, -1000, -1100, -1500, 3.14f, 0.79f, 0.62f, -1500, 0.056f, 0.00f,0.00f,0.00f, 100, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } +#define REVERB_PRESET_CASTLE_CUPBOARD \ + { 26, 8.3f, 0.890f, -1000, -1100, -2000, 0.67f, 0.87f, 0.31f, 300, 0.010f, 0.00f,0.00f,0.00f, 1100, 0.007f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } +#define REVERB_PRESET_CASTLE_COURTYARD \ + { 26, 8.3f, 0.420f, -1000, -700, -1400, 2.13f, 0.61f, 0.23f, -1300, 0.160f, 0.00f,0.00f,0.00f, -300, 0.036f, 0.00f,0.00f,0.00f, 0.250f, 0.370f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f } +#define REVERB_PRESET_CASTLE_ALCOVE \ + { 26, 8.3f, 0.890f, -1000, -600, -2000, 1.64f, 0.87f, 0.31f, 00, 0.007f, 0.00f,0.00f,0.00f, 300, 0.034f, 0.00f,0.00f,0.00f, 0.138f, 0.080f, 0.250f, 0.000f, -5.0f, 5168.6f, 139.5f, 0.00f, 0x20 } + + +// FACTORY PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_FACTORY_ALCOVE \ + { 26, 1.8f, 0.590f, -1200, -200, -600, 3.14f, 0.65f, 1.31f, 300, 0.010f, 0.00f,0.00f,0.00f, 000, 0.038f, 0.00f,0.00f,0.00f, 0.114f, 0.100f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_SHORTPASSAGE \ + { 26, 1.8f, 0.640f, -1200, -200, -600, 2.53f, 0.65f, 1.31f, 0, 0.010f, 0.00f,0.00f,0.00f, 200, 0.038f, 0.00f,0.00f,0.00f, 0.135f, 0.230f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_MEDIUMROOM \ + { 26, 1.9f, 0.820f, -1200, -200, -600, 2.76f, 0.65f, 1.31f, -1100, 0.022f, 0.00f,0.00f,0.00f, 300, 0.023f, 0.00f,0.00f,0.00f, 0.174f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_LONGPASSAGE \ + { 26, 1.8f, 0.640f, -1200, -200, -600, 4.06f, 0.65f, 1.31f, 0, 0.020f, 0.00f,0.00f,0.00f, 200, 0.037f, 0.00f,0.00f,0.00f, 0.135f, 0.230f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_LARGEROOM \ + { 26, 1.9f, 0.750f, -1200, -300, -400, 4.24f, 0.51f, 1.31f, -1500, 0.039f, 0.00f,0.00f,0.00f, 100, 0.023f, 0.00f,0.00f,0.00f, 0.231f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_HALL \ + { 26, 1.9f, 0.750f, -1000, -300, -400, 7.43f, 0.51f, 1.31f, -2400, 0.073f, 0.00f,0.00f,0.00f, -100, 0.027f, 0.00f,0.00f,0.00f, 0.250f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_CUPBOARD \ + { 26, 1.7f, 0.630f, -1200, -200, -600, 0.49f, 0.65f, 1.31f, 200, 0.010f, 0.00f,0.00f,0.00f, 600, 0.032f, 0.00f,0.00f,0.00f, 0.107f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_COURTYARD \ + { 26, 1.7f, 0.570f, -1000, -1000, -400, 2.32f, 0.29f, 0.56f, -1300, 0.140f, 0.00f,0.00f,0.00f, -800, 0.039f, 0.00f,0.00f,0.00f, 0.250f, 0.290f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } +#define REVERB_PRESET_FACTORY_SMALLROOM \ + { 26, 1.8f, 0.820f, -1000, -200, -600, 1.72f, 0.65f, 1.31f, -300, 0.010f, 0.00f,0.00f,0.00f, 500, 0.024f, 0.00f,0.00f,0.00f, 0.119f, 0.070f, 0.250f, 0.000f, -5.0f, 3762.6f, 362.5f, 0.00f, 0x20 } + + +// ICE PALACE PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_ICEPALACE_ALCOVE \ + { 26, 2.7f, 0.840f, -1000, -500, -1100, 2.76f, 1.46f, 0.28f, 100, 0.010f, 0.00f,0.00f,0.00f, -100, 0.030f, 0.00f,0.00f,0.00f, 0.161f, 0.090f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_SHORTPASSAGE \ + { 26, 2.7f, 0.750f, -1000, -500, -1100, 1.79f, 1.46f, 0.28f, -600, 0.010f, 0.00f,0.00f,0.00f, 100, 0.019f, 0.00f,0.00f,0.00f, 0.177f, 0.090f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_MEDIUMROOM \ + { 26, 2.7f, 0.870f, -1000, -500, -700, 2.22f, 1.53f, 0.32f, -800, 0.039f, 0.00f,0.00f,0.00f, 100, 0.027f, 0.00f,0.00f,0.00f, 0.186f, 0.120f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_LONGPASSAGE \ + { 26, 2.7f, 0.770f, -1000, -500, -800, 3.01f, 1.46f, 0.28f, -200, 0.012f, 0.00f,0.00f,0.00f, 200, 0.025f, 0.00f,0.00f,0.00f, 0.186f, 0.040f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_LARGEROOM \ + { 26, 2.9f, 0.810f, -1000, -500, -700, 3.14f, 1.53f, 0.32f, -1200, 0.039f, 0.00f,0.00f,0.00f, 000, 0.027f, 0.00f,0.00f,0.00f, 0.214f, 0.110f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_HALL \ + { 26, 2.9f, 0.760f, -1000, -700, -500, 5.49f, 1.53f, 0.38f, -1900, 0.054f, 0.00f,0.00f,0.00f, -400, 0.052f, 0.00f,0.00f,0.00f, 0.226f, 0.110f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_CUPBOARD \ + { 26, 2.7f, 0.830f, -1000, -600, -1300, 0.76f, 1.53f, 0.26f, 100, 0.012f, 0.00f,0.00f,0.00f, 600, 0.016f, 0.00f,0.00f,0.00f, 0.143f, 0.080f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_COURTYARD \ + { 26, 2.9f, 0.590f, -1000, -1100, -1000, 2.04f, 1.20f, 0.38f, -1000, 0.173f, 0.00f,0.00f,0.00f, -1000, 0.043f, 0.00f,0.00f,0.00f, 0.235f, 0.480f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } +#define REVERB_PRESET_ICEPALACE_SMALLROOM \ + { 26, 2.7f, 0.840f, -1000, -500, -1100, 1.51f, 1.53f, 0.27f, -100, 0.010f, 0.00f,0.00f,0.00f, 300, 0.011f, 0.00f,0.00f,0.00f, 0.164f, 0.140f, 0.250f, 0.000f, -5.0f, 12428.5f, 99.6f, 0.00f, 0x20 } + + +// SPACE STATION PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_SPACESTATION_ALCOVE \ + { 26, 1.5f, 0.780f, -1000, -300, -100, 1.16f, 0.81f, 0.55f, 300, 0.007f, 0.00f,0.00f,0.00f, 000, 0.018f, 0.00f,0.00f,0.00f, 0.192f, 0.210f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPACESTATION_MEDIUMROOM \ + { 26, 1.5f, 0.750f, -1000, -400, -100, 3.01f, 0.50f, 0.55f, -800, 0.034f, 0.00f,0.00f,0.00f, 100, 0.035f, 0.00f,0.00f,0.00f, 0.209f, 0.310f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPACESTATION_SHORTPASSAGE \ + { 26, 1.5f, 0.870f, -1000, -400, -100, 3.57f, 0.50f, 0.55f, 0, 0.012f, 0.00f,0.00f,0.00f, 100, 0.016f, 0.00f,0.00f,0.00f, 0.172f, 0.200f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPACESTATION_LONGPASSAGE \ + { 26, 1.9f, 0.820f, -1000, -400, -100, 4.62f, 0.62f, 0.55f, 0, 0.012f, 0.00f,0.00f,0.00f, 200, 0.031f, 0.00f,0.00f,0.00f, 0.250f, 0.230f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPACESTATION_LARGEROOM \ + { 26, 1.8f, 0.810f, -1000, -400, -100, 3.89f, 0.38f, 0.61f, -1000, 0.056f, 0.00f,0.00f,0.00f, -100, 0.035f, 0.00f,0.00f,0.00f, 0.233f, 0.280f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPACESTATION_HALL \ + { 26, 1.9f, 0.870f, -1000, -400, -100, 7.11f, 0.38f, 0.61f, -1500, 0.100f, 0.00f,0.00f,0.00f, -400, 0.047f, 0.00f,0.00f,0.00f, 0.250f, 0.250f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPACESTATION_CUPBOARD \ + { 26, 1.4f, 0.560f, -1000, -300, -100, 0.79f, 0.81f, 0.55f, 300, 0.007f, 0.00f,0.00f,0.00f, 500, 0.018f, 0.00f,0.00f,0.00f, 0.181f, 0.310f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPACESTATION_SMALLROOM \ + { 26, 1.5f, 0.700f, -1000, -300, -100, 1.72f, 0.82f, 0.55f, -200, 0.007f, 0.00f,0.00f,0.00f, 300, 0.013f, 0.00f,0.00f,0.00f, 0.188f, 0.260f, 0.250f, 0.000f, -5.0f, 3316.1f, 458.2f, 0.00f, 0x20 } + + +// WOODEN GALLEON PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_WOODEN_ALCOVE \ + { 26, 7.5f, 1.000f, -1000, -1800, -1000, 1.22f, 0.62f, 0.91f, 100, 0.012f, 0.00f,0.00f,0.00f, -300, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_SHORTPASSAGE \ + { 26, 7.5f, 1.000f, -1000, -1800, -1000, 1.75f, 0.50f, 0.87f, -100, 0.012f, 0.00f,0.00f,0.00f, -400, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_MEDIUMROOM \ + { 26, 7.5f, 1.000f, -1000, -2000, -1100, 1.47f, 0.42f, 0.82f, -100, 0.049f, 0.00f,0.00f,0.00f, -100, 0.029f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_LONGPASSAGE \ + { 26, 7.5f, 1.000f, -1000, -2000, -1000, 1.99f, 0.40f, 0.79f, 000, 0.020f, 0.00f,0.00f,0.00f, -700, 0.036f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_LARGEROOM \ + { 26, 7.5f, 1.000f, -1000, -2100, -1100, 2.65f, 0.33f, 0.82f, -100, 0.066f, 0.00f,0.00f,0.00f, -200, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_HALL \ + { 26, 7.5f, 1.000f, -1000, -2200, -1100, 3.45f, 0.30f, 0.82f, -100, 0.088f, 0.00f,0.00f,0.00f, -200, 0.063f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_CUPBOARD \ + { 26, 7.5f, 1.000f, -1000, -1700, -1000, 0.56f, 0.46f, 0.91f, 100, 0.012f, 0.00f,0.00f,0.00f, 100, 0.028f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_SMALLROOM \ + { 26, 7.5f, 1.000f, -1000, -1900, -1000, 0.79f, 0.32f, 0.87f, 00, 0.032f, 0.00f,0.00f,0.00f, -100, 0.029f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } +#define REVERB_PRESET_WOODEN_COURTYARD \ + { 26, 7.5f, 0.650f, -1000, -2200, -1000, 1.79f, 0.35f, 0.79f, -500, 0.123f, 0.00f,0.00f,0.00f, -2000, 0.032f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 4705.0f, 99.6f, 0.00f, 0x3f } + + +// SPORTS PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_SPORT_EMPTYSTADIUM \ + { 26, 7.2f, 1.000f, -1000, -700, -200, 6.26f, 0.51f, 1.10f, -2400, 0.183f, 0.00f,0.00f,0.00f, -800, 0.038f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 } +#define REVERB_PRESET_SPORT_SQUASHCOURT \ + { 26, 7.5f, 0.750f, -1000, -1000, -200, 2.22f, 0.91f, 1.16f, -700, 0.007f, 0.00f,0.00f,0.00f, -200, 0.011f, 0.00f,0.00f,0.00f, 0.126f, 0.190f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPORT_SMALLSWIMMINGPOOL \ + { 26, 36.2f, 0.700f, -1000, -200, -100, 2.76f, 1.25f, 1.14f, -400, 0.020f, 0.00f,0.00f,0.00f, -200, 0.030f, 0.00f,0.00f,0.00f, 0.179f, 0.150f, 0.895f, 0.190f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 } +#define REVERB_PRESET_SPORT_LARGESWIMMINGPOOL\ + { 26, 36.2f, 0.820f, -1000, -200, 0, 5.49f, 1.31f, 1.14f, -700, 0.039f, 0.00f,0.00f,0.00f, -600, 0.049f, 0.00f,0.00f,0.00f, 0.222f, 0.550f, 1.159f, 0.210f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 } +#define REVERB_PRESET_SPORT_GYMNASIUM \ + { 26, 7.5f, 0.810f, -1000, -700, -100, 3.14f, 1.06f, 1.35f, -800, 0.029f, 0.00f,0.00f,0.00f, -500, 0.045f, 0.00f,0.00f,0.00f, 0.146f, 0.140f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 } +#define REVERB_PRESET_SPORT_FULLSTADIUM \ + { 26, 7.2f, 1.000f, -1000, -2300, -200, 5.25f, 0.17f, 0.80f, -2000, 0.188f, 0.00f,0.00f,0.00f, -1100, 0.038f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 } +#define REVERB_PRESET_SPORT_STADIUMTANNOY \ + { 26, 3.0f, 0.780f, -1000, -500, -600, 2.53f, 0.88f, 0.68f, -1100, 0.230f, 0.00f,0.00f,0.00f, -600, 0.063f, 0.00f,0.00f,0.00f, 0.250f, 0.200f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 } + + +// PREFAB PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_PREFAB_WORKSHOP \ + { 26, 1.9f, 1.000f, -1000, -1700, -800, 0.76f, 1.00f, 1.00f, 0, 0.012f, 0.00f,0.00f,0.00f, 100, 0.012f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 } +#define REVERB_PRESET_PREFAB_SCHOOLROOM \ + { 26, 1.86f, 0.690f, -1000, -400, -600, 0.98f, 0.45f, 0.18f, 300, 0.017f, 0.00f,0.00f,0.00f, 300, 0.015f, 0.00f,0.00f,0.00f, 0.095f, 0.140f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 } +#define REVERB_PRESET_PREFAB_PRACTISEROOM \ + { 26, 1.86f, 0.870f, -1000, -800, -600, 1.12f, 0.56f, 0.18f, 200, 0.010f, 0.00f,0.00f,0.00f, 300, 0.011f, 0.00f,0.00f,0.00f, 0.095f, 0.140f, 0.250f, 0.000f, -5.0f, 7176.9f, 211.2f, 0.00f, 0x20 } +#define REVERB_PRESET_PREFAB_OUTHOUSE \ + { 26, 80.3f, 0.820f, -1000, -1900, -1600, 1.38f, 0.38f, 0.35f, -100, 0.024f, 0.00f,0.00f,-0.00f, -400, 0.044f, 0.00f,0.00f,0.00f, 0.121f, 0.170f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 } +#define REVERB_PRESET_PREFAB_CARAVAN \ + { 26, 8.3f, 1.000f, -1000, -2100, -1800, 0.43f, 1.50f, 1.00f, 0, 0.012f, 0.00f,0.00f,0.00f, 600, 0.012f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x1f } + // for US developers, a caravan is the same as a trailer =o) + + +// DOME AND PIPE PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_DOME_TOMB \ + { 26, 51.8f, 0.790f, -1000, -900, -1300, 4.18f, 0.21f, 0.10f, -825, 0.030f, 0.00f,0.00f,0.00f, 450, 0.022f, 0.00f,0.00f,0.00f, 0.177f, 0.190f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x0 } +#define REVERB_PRESET_PIPE_SMALL \ + { 26, 50.3f, 1.000f, -1000, -900, -1300, 5.04f, 0.10f, 0.10f, -600, 0.032f, 0.00f,0.00f,0.00f, 800, 0.015f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x3f } +#define REVERB_PRESET_DOME_SAINTPAULS \ + { 26, 50.3f, 0.870f, -1000, -900, -1300, 10.48f, 0.19f, 0.10f, -1500, 0.090f, 0.00f,0.00f,0.00f, 200, 0.042f, 0.00f,0.00f,0.00f, 0.250f, 0.120f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x3f } +#define REVERB_PRESET_PIPE_LONGTHIN \ + { 26, 1.6f, 0.910f, -1000, -700, -1100, 9.21f, 0.18f, 0.10f, -300, 0.010f, 0.00f,0.00f,0.00f, -300, 0.022f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x0 } +#define REVERB_PRESET_PIPE_LARGE \ + { 26, 50.3f, 1.000f, -1000, -900, -1300, 8.45f, 0.10f, 0.10f, -800, 0.046f, 0.00f,0.00f,0.00f, 400, 0.032f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x3f } +#define REVERB_PRESET_PIPE_RESONANT \ + { 26, 1.3f, 0.910f, -1000, -700, -1100, 6.81f, 0.18f, 0.10f, -300, 0.010f, 0.00f,0.00f,0.00f, 00, 0.022f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 2854.4f, 20.0f, 0.00f, 0x0 } + + +// OUTDOORS PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_OUTDOORS_BACKYARD \ + { 26, 80.3f, 0.450f, -1000, -1200, -600, 1.12f, 0.34f, 0.46f, -700, 0.069f, 0.00f,0.00f,-0.00f, -300, 0.023f, 0.00f,0.00f,0.00f, 0.218f, 0.340f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 } +#define REVERB_PRESET_OUTDOORS_ROLLINGPLAINS \ + { 26, 80.3f, 0.000f, -1000, -3900, -400, 2.13f, 0.21f, 0.46f, -1500, 0.300f, 0.00f,0.00f,-0.00f, -700, 0.019f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 } +#define REVERB_PRESET_OUTDOORS_DEEPCANYON \ + { 26, 80.3f, 0.740f, -1000, -1500, -400, 3.89f, 0.21f, 0.46f, -1000, 0.223f, 0.00f,0.00f,-0.00f, -900, 0.019f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 } +#define REVERB_PRESET_OUTDOORS_CREEK \ + { 26, 80.3f, 0.350f, -1000, -1500, -600, 2.13f, 0.21f, 0.46f, -800, 0.115f, 0.00f,0.00f,-0.00f, -1400, 0.031f, 0.00f,0.00f,0.00f, 0.218f, 0.340f, 0.250f, 0.000f, -5.0f, 4399.1f, 242.9f, 0.00f, 0x0 } +#define REVERB_PRESET_OUTDOORS_VALLEY \ + { 26, 80.3f, 0.280f, -1000, -3100, -1600, 2.88f, 0.26f, 0.35f, -1700, 0.263f, 0.00f,0.00f,-0.00f, -800, 0.100f, 0.00f,0.00f,0.00f, 0.250f, 0.340f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 } + + +// MOOD PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_MOOD_HEAVEN \ + { 26, 19.6f, 0.940f, -1000, -200, -700, 5.04f, 1.12f, 0.56f, -1230, 0.020f, 0.00f,0.00f,0.00f, 200, 0.029f, 0.00f,0.00f,0.00f, 0.250f, 0.080f, 2.742f, 0.050f, -2.0f, 5000.0f, 250.0f, 0.00f, 0x3f } +#define REVERB_PRESET_MOOD_HELL \ + { 26, 100.0f, 0.570f, -1000, -900, -700, 3.57f, 0.49f, 2.00f, -10000, 0.020f, 0.00f,0.00f,0.00f, 300, 0.030f, 0.00f,0.00f,0.00f, 0.110f, 0.040f, 2.109f, 0.520f, -5.0f, 5000.0f, 139.5f, 0.00f, 0x40 } +#define REVERB_PRESET_MOOD_MEMORY \ + { 26, 8.0f, 0.850f, -1000, -400, -900, 4.06f, 0.82f, 0.56f, -2800, 0.000f, 0.00f,0.00f,0.00f, 100, 0.000f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.474f, 0.450f, -10.0f, 5000.0f, 250.0f, 0.00f, 0x0 } + + +// DRIVING SIMULATION PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_DRIVING_COMMENTATOR \ + { 26, 3.0f, 0.000f, 1000, -500, -600, 2.42f, 0.88f, 0.68f, -1400, 0.093f, 0.00f,0.00f,0.00f, -1200, 0.017f, 0.00f,0.00f,0.00f, 0.250f, 1.000f, 0.250f, 0.000f, -10.0f, 5000.0f, 250.0f, 0.00f, 0x20 } +#define REVERB_PRESET_DRIVING_PITGARAGE \ + { 26, 1.9f, 0.590f, -1000, -300, -500, 1.72f, 0.93f, 0.87f, -500, 0.000f, 0.00f,0.00f,0.00f, 200, 0.016f, 0.00f,0.00f,0.00f, 0.250f, 0.110f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x0 } +#define REVERB_PRESET_DRIVING_INCAR_RACER \ + { 26, 1.1f, 0.800f, -1000, 0, -200, 0.17f, 2.00f, 0.41f, 500, 0.007f, 0.00f,0.00f,0.00f, -300, 0.015f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10268.2f, 251.0f, 0.00f, 0x20 } +#define REVERB_PRESET_DRIVING_INCAR_SPORTS \ + { 26, 1.1f, 0.800f, -1000, -400, 0, 0.17f, 0.75f, 0.41f, 0, 0.010f, 0.00f,0.00f,0.00f, -500, 0.000f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10268.2f, 251.0f, 0.00f, 0x20 } +#define REVERB_PRESET_DRIVING_INCAR_LUXURY \ + { 26, 1.6f, 1.000f, -1000, -2000, -600, 0.13f, 0.41f, 0.46f, -200, 0.010f, 0.00f,0.00f,0.00f, 400, 0.010f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10268.2f, 251.0f, 0.00f, 0x20 } +#define REVERB_PRESET_DRIVING_FULLGRANDSTAND \ + { 26, 8.3f, 1.000f, -1000, -1100, -400, 3.01f, 1.37f, 1.28f, -900, 0.090f, 0.00f,0.00f,0.00f, -1500, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10420.2f, 250.0f, 0.00f, 0x1f } +#define REVERB_PRESET_DRIVING_EMPTYGRANDSTAND \ + { 26, 8.3f, 1.000f, -1000, 0, -200, 4.62f, 1.75f, 1.40f, -1363, 0.090f, 0.00f,0.00f,0.00f, -1200, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.000f, -5.0f, 10420.2f, 250.0f, 0.00f, 0x1f } +#define REVERB_PRESET_DRIVING_TUNNEL \ + { 26, 3.1f, 0.810f, -1000, -800, -100, 3.42f, 0.94f, 1.31f, -300, 0.051f, 0.00f,0.00f,0.00f, -300, 0.047f, 0.00f,0.00f,0.00f, 0.214f, 0.050f, 0.250f, 0.000f, -5.0f, 5000.0f, 155.3f, 0.00f, 0x20 } + + +// CITY PRESETS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_CITY_STREETS \ + { 26, 3.0f, 0.780f, -1000, -300, -100, 1.79f, 1.12f, 0.91f, -1100, 0.046f, 0.00f,0.00f,0.00f, -1400, 0.028f, 0.00f,0.00f,0.00f, 0.250f, 0.200f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 } +#define REVERB_PRESET_CITY_SUBWAY \ + { 26, 3.0f, 0.740f, -1000, -300, -100, 3.01f, 1.23f, 0.91f, -300, 0.046f, 0.00f,0.00f,0.00f, 200, 0.028f, 0.00f,0.00f,0.00f, 0.125f, 0.210f, 0.250f, 0.000f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x20 } +#define REVERB_PRESET_CITY_MUSEUM \ + { 26, 80.3f, 0.820f, -1000, -1500, -1500, 3.28f, 1.40f, 0.57f, -1200, 0.039f, 0.00f,0.00f,-0.00f, -100, 0.034f, 0.00f,0.00f,0.00f, 0.130f, 0.170f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 } +#define REVERB_PRESET_CITY_LIBRARY \ + { 26, 80.3f, 0.820f, -1000, -1100, -2100, 2.76f, 0.89f, 0.41f, -900, 0.029f, 0.00f,0.00f,-0.00f, -100, 0.020f, 0.00f,0.00f,0.00f, 0.130f, 0.170f, 0.250f, 0.000f, -5.0f, 2854.4f, 107.5f, 0.00f, 0x0 } +#define REVERB_PRESET_CITY_UNDERPASS \ + { 26, 3.0f, 0.820f, -1000, -700, -100, 3.57f, 1.12f, 0.91f, -800, 0.059f, 0.00f,0.00f,0.00f, -100, 0.037f, 0.00f,0.00f,0.00f, 0.250f, 0.140f, 0.250f, 0.000f, -7.0f, 5000.0f, 250.0f, 0.00f, 0x20 } +#define REVERB_PRESET_CITY_ABANDONED \ + { 26, 3.0f, 0.690f, -1000, -200, -100, 3.28f, 1.17f, 0.91f, -700, 0.044f, 0.00f,0.00f,0.00f, -1100, 0.024f, 0.00f,0.00f,0.00f, 0.250f, 0.200f, 0.250f, 0.000f, -3.0f, 5000.0f, 250.0f, 0.00f, 0x20 } + + +// MISC ROOMS + +// Env Size Diffus Room RoomHF RoomLF DecTm DcHF DcLF Refl RefDel Ref Pan Revb RevDel Rev Pan EchTm EchDp ModTm ModDp AirAbs HFRef LFRef RRlOff FLAGS +#define REVERB_PRESET_DUSTYROOM \ + { 26, 1.8f, 0.560f, -1000, -200, -300, 1.79f, 0.38f, 0.21f, -600, 0.002f, 0.00f,0.00f,0.00f, 200, 0.006f, 0.00f,0.00f,0.00f, 0.202f, 0.050f, 0.250f, 0.000f, -10.0f, 13046.0f, 163.3f, 0.00f, 0x20 } +#define REVERB_PRESET_CHAPEL \ + { 26, 19.6f, 0.840f, -1000, -500, 0, 4.62f, 0.64f, 1.23f, -700, 0.032f, 0.00f,0.00f,0.00f, -200, 0.049f, 0.00f,0.00f,0.00f, 0.250f, 0.000f, 0.250f, 0.110f, -5.0f, 5000.0f, 250.0f, 0.00f, 0x3f } +#define REVERB_PRESET_SMALLWATERROOM \ + { 26, 36.2f, 0.700f, -1000, -698, 0, 1.51f, 1.25f, 1.14f, -100, 0.020f, 0.00f,0.00f,0.00f, 300, 0.030f, 0.00f,0.00f,0.00f, 0.179f, 0.150f, 0.895f, 0.190f, -7.0f, 5000.0f, 250.0f, 0.00f, 0x0 } diff --git a/Externals/OpenAL/include/al.h b/Externals/OpenAL/include/al.h new file mode 100644 index 0000000000..413b38331e --- /dev/null +++ b/Externals/OpenAL/include/al.h @@ -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 */ diff --git a/Externals/OpenAL/include/alc.h b/Externals/OpenAL/include/alc.h new file mode 100644 index 0000000000..294e8b33c6 --- /dev/null +++ b/Externals/OpenAL/include/alc.h @@ -0,0 +1,237 @@ +#ifndef AL_ALC_H +#define AL_ALC_H + +#if defined(__cplusplus) +extern "C" { +#endif + +#ifndef ALC_API + #if defined(AL_LIBTYPE_STATIC) + #define ALC_API + #elif defined(_WIN32) + #define ALC_API __declspec(dllimport) + #else + #define ALC_API extern + #endif +#endif + +#if defined(_WIN32) + #define ALC_APIENTRY __cdecl +#else + #define ALC_APIENTRY +#endif + + +/** Deprecated macro. */ +#define ALCAPI ALC_API +#define ALCAPIENTRY ALC_APIENTRY +#define ALC_INVALID 0 + +/** Supported ALC version? */ +#define ALC_VERSION_0_1 1 + +/** Opaque device handle */ +typedef struct ALCdevice_struct ALCdevice; +/** Opaque context handle */ +typedef struct ALCcontext_struct ALCcontext; + +/** 8-bit boolean */ +typedef char ALCboolean; + +/** character */ +typedef char ALCchar; + +/** signed 8-bit 2's complement integer */ +typedef signed char ALCbyte; + +/** unsigned 8-bit integer */ +typedef unsigned char ALCubyte; + +/** signed 16-bit 2's complement integer */ +typedef short ALCshort; + +/** unsigned 16-bit integer */ +typedef unsigned short ALCushort; + +/** signed 32-bit 2's complement integer */ +typedef int ALCint; + +/** unsigned 32-bit integer */ +typedef unsigned int ALCuint; + +/** non-negative 32-bit binary integer size */ +typedef int ALCsizei; + +/** enumerated 32-bit value */ +typedef int ALCenum; + +/** 32-bit IEEE754 floating-point */ +typedef float ALCfloat; + +/** 64-bit IEEE754 floating-point */ +typedef double ALCdouble; + +/** void type (for opaque pointers only) */ +typedef void ALCvoid; + + +/* Enumerant values begin at column 50. No tabs. */ + +/** Boolean False. */ +#define ALC_FALSE 0 + +/** Boolean True. */ +#define ALC_TRUE 1 + +/** Context attribute: Hz. */ +#define ALC_FREQUENCY 0x1007 + +/** Context attribute: Hz. */ +#define ALC_REFRESH 0x1008 + +/** Context attribute: AL_TRUE or AL_FALSE. */ +#define ALC_SYNC 0x1009 + +/** Context attribute: requested Mono (3D) Sources. */ +#define ALC_MONO_SOURCES 0x1010 + +/** Context attribute: 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 */ diff --git a/Externals/OpenAL/include/alext.h b/Externals/OpenAL/include/alext.h new file mode 100644 index 0000000000..0447f2bb45 --- /dev/null +++ b/Externals/OpenAL/include/alext.h @@ -0,0 +1,355 @@ +/** + * OpenAL cross platform audio library + * Copyright (C) 2008 by authors. + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + * Or go to http://www.gnu.org/copyleft/lgpl.html + */ + +#ifndef AL_ALEXT_H +#define AL_ALEXT_H + +#include +/* Define int64_t and uint64_t types */ +#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L +#include +#elif defined(_WIN32) && defined(__GNUC__) +#include +#elif defined(_WIN32) +typedef __int64 int64_t; +typedef unsigned __int64 uint64_t; +#else +/* Fallback if nothing above works */ +#include +#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 diff --git a/Externals/OpenAL/include/efx-creative.h b/Externals/OpenAL/include/efx-creative.h new file mode 100644 index 0000000000..0a04c982e1 --- /dev/null +++ b/Externals/OpenAL/include/efx-creative.h @@ -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. */ diff --git a/Externals/OpenAL/include/efx-presets.h b/Externals/OpenAL/include/efx-presets.h new file mode 100644 index 0000000000..86dcbda2f3 --- /dev/null +++ b/Externals/OpenAL/include/efx-presets.h @@ -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 */ diff --git a/Externals/OpenAL/include/efx.h b/Externals/OpenAL/include/efx.h new file mode 100644 index 0000000000..57766983f6 --- /dev/null +++ b/Externals/OpenAL/include/efx.h @@ -0,0 +1,761 @@ +#ifndef AL_EFX_H +#define AL_EFX_H + + +#include "alc.h" +#include "al.h" + +#ifdef __cplusplus +extern "C" { +#endif + +#define ALC_EXT_EFX_NAME "ALC_EXT_EFX" + +#define ALC_EFX_MAJOR_VERSION 0x20001 +#define ALC_EFX_MINOR_VERSION 0x20002 +#define ALC_MAX_AUXILIARY_SENDS 0x20003 + + +/* Listener properties. */ +#define AL_METERS_PER_UNIT 0x20004 + +/* Source properties. */ +#define AL_DIRECT_FILTER 0x20005 +#define AL_AUXILIARY_SEND_FILTER 0x20006 +#define AL_AIR_ABSORPTION_FACTOR 0x20007 +#define AL_ROOM_ROLLOFF_FACTOR 0x20008 +#define AL_CONE_OUTER_GAINHF 0x20009 +#define AL_DIRECT_FILTER_GAINHF_AUTO 0x2000A +#define AL_AUXILIARY_SEND_FILTER_GAIN_AUTO 0x2000B +#define AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO 0x2000C + + +/* Effect properties. */ + +/* Reverb effect parameters */ +#define AL_REVERB_DENSITY 0x0001 +#define AL_REVERB_DIFFUSION 0x0002 +#define AL_REVERB_GAIN 0x0003 +#define AL_REVERB_GAINHF 0x0004 +#define AL_REVERB_DECAY_TIME 0x0005 +#define AL_REVERB_DECAY_HFRATIO 0x0006 +#define AL_REVERB_REFLECTIONS_GAIN 0x0007 +#define AL_REVERB_REFLECTIONS_DELAY 0x0008 +#define AL_REVERB_LATE_REVERB_GAIN 0x0009 +#define AL_REVERB_LATE_REVERB_DELAY 0x000A +#define AL_REVERB_AIR_ABSORPTION_GAINHF 0x000B +#define AL_REVERB_ROOM_ROLLOFF_FACTOR 0x000C +#define AL_REVERB_DECAY_HFLIMIT 0x000D + +/* EAX Reverb effect parameters */ +#define AL_EAXREVERB_DENSITY 0x0001 +#define AL_EAXREVERB_DIFFUSION 0x0002 +#define AL_EAXREVERB_GAIN 0x0003 +#define AL_EAXREVERB_GAINHF 0x0004 +#define AL_EAXREVERB_GAINLF 0x0005 +#define AL_EAXREVERB_DECAY_TIME 0x0006 +#define AL_EAXREVERB_DECAY_HFRATIO 0x0007 +#define AL_EAXREVERB_DECAY_LFRATIO 0x0008 +#define AL_EAXREVERB_REFLECTIONS_GAIN 0x0009 +#define AL_EAXREVERB_REFLECTIONS_DELAY 0x000A +#define AL_EAXREVERB_REFLECTIONS_PAN 0x000B +#define AL_EAXREVERB_LATE_REVERB_GAIN 0x000C +#define AL_EAXREVERB_LATE_REVERB_DELAY 0x000D +#define AL_EAXREVERB_LATE_REVERB_PAN 0x000E +#define AL_EAXREVERB_ECHO_TIME 0x000F +#define AL_EAXREVERB_ECHO_DEPTH 0x0010 +#define AL_EAXREVERB_MODULATION_TIME 0x0011 +#define AL_EAXREVERB_MODULATION_DEPTH 0x0012 +#define AL_EAXREVERB_AIR_ABSORPTION_GAINHF 0x0013 +#define AL_EAXREVERB_HFREFERENCE 0x0014 +#define AL_EAXREVERB_LFREFERENCE 0x0015 +#define AL_EAXREVERB_ROOM_ROLLOFF_FACTOR 0x0016 +#define AL_EAXREVERB_DECAY_HFLIMIT 0x0017 + +/* Chorus effect parameters */ +#define AL_CHORUS_WAVEFORM 0x0001 +#define AL_CHORUS_PHASE 0x0002 +#define AL_CHORUS_RATE 0x0003 +#define AL_CHORUS_DEPTH 0x0004 +#define AL_CHORUS_FEEDBACK 0x0005 +#define AL_CHORUS_DELAY 0x0006 + +/* Distortion effect parameters */ +#define AL_DISTORTION_EDGE 0x0001 +#define AL_DISTORTION_GAIN 0x0002 +#define AL_DISTORTION_LOWPASS_CUTOFF 0x0003 +#define AL_DISTORTION_EQCENTER 0x0004 +#define AL_DISTORTION_EQBANDWIDTH 0x0005 + +/* Echo effect parameters */ +#define AL_ECHO_DELAY 0x0001 +#define AL_ECHO_LRDELAY 0x0002 +#define AL_ECHO_DAMPING 0x0003 +#define AL_ECHO_FEEDBACK 0x0004 +#define AL_ECHO_SPREAD 0x0005 + +/* Flanger effect parameters */ +#define AL_FLANGER_WAVEFORM 0x0001 +#define AL_FLANGER_PHASE 0x0002 +#define AL_FLANGER_RATE 0x0003 +#define AL_FLANGER_DEPTH 0x0004 +#define AL_FLANGER_FEEDBACK 0x0005 +#define AL_FLANGER_DELAY 0x0006 + +/* Frequency shifter effect parameters */ +#define AL_FREQUENCY_SHIFTER_FREQUENCY 0x0001 +#define AL_FREQUENCY_SHIFTER_LEFT_DIRECTION 0x0002 +#define AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION 0x0003 + +/* Vocal morpher effect parameters */ +#define AL_VOCAL_MORPHER_PHONEMEA 0x0001 +#define AL_VOCAL_MORPHER_PHONEMEA_COARSE_TUNING 0x0002 +#define AL_VOCAL_MORPHER_PHONEMEB 0x0003 +#define AL_VOCAL_MORPHER_PHONEMEB_COARSE_TUNING 0x0004 +#define AL_VOCAL_MORPHER_WAVEFORM 0x0005 +#define AL_VOCAL_MORPHER_RATE 0x0006 + +/* Pitchshifter effect parameters */ +#define AL_PITCH_SHIFTER_COARSE_TUNE 0x0001 +#define AL_PITCH_SHIFTER_FINE_TUNE 0x0002 + +/* Ringmodulator effect parameters */ +#define AL_RING_MODULATOR_FREQUENCY 0x0001 +#define AL_RING_MODULATOR_HIGHPASS_CUTOFF 0x0002 +#define AL_RING_MODULATOR_WAVEFORM 0x0003 + +/* Autowah effect parameters */ +#define AL_AUTOWAH_ATTACK_TIME 0x0001 +#define AL_AUTOWAH_RELEASE_TIME 0x0002 +#define AL_AUTOWAH_RESONANCE 0x0003 +#define AL_AUTOWAH_PEAK_GAIN 0x0004 + +/* Compressor effect parameters */ +#define AL_COMPRESSOR_ONOFF 0x0001 + +/* Equalizer effect parameters */ +#define AL_EQUALIZER_LOW_GAIN 0x0001 +#define AL_EQUALIZER_LOW_CUTOFF 0x0002 +#define AL_EQUALIZER_MID1_GAIN 0x0003 +#define AL_EQUALIZER_MID1_CENTER 0x0004 +#define AL_EQUALIZER_MID1_WIDTH 0x0005 +#define AL_EQUALIZER_MID2_GAIN 0x0006 +#define AL_EQUALIZER_MID2_CENTER 0x0007 +#define AL_EQUALIZER_MID2_WIDTH 0x0008 +#define AL_EQUALIZER_HIGH_GAIN 0x0009 +#define AL_EQUALIZER_HIGH_CUTOFF 0x000A + +/* Effect type */ +#define AL_EFFECT_FIRST_PARAMETER 0x0000 +#define AL_EFFECT_LAST_PARAMETER 0x8000 +#define AL_EFFECT_TYPE 0x8001 + +/* Effect types, used with the AL_EFFECT_TYPE property */ +#define AL_EFFECT_NULL 0x0000 +#define AL_EFFECT_REVERB 0x0001 +#define AL_EFFECT_CHORUS 0x0002 +#define AL_EFFECT_DISTORTION 0x0003 +#define AL_EFFECT_ECHO 0x0004 +#define AL_EFFECT_FLANGER 0x0005 +#define AL_EFFECT_FREQUENCY_SHIFTER 0x0006 +#define AL_EFFECT_VOCAL_MORPHER 0x0007 +#define AL_EFFECT_PITCH_SHIFTER 0x0008 +#define AL_EFFECT_RING_MODULATOR 0x0009 +#define AL_EFFECT_AUTOWAH 0x000A +#define AL_EFFECT_COMPRESSOR 0x000B +#define AL_EFFECT_EQUALIZER 0x000C +#define AL_EFFECT_EAXREVERB 0x8000 + +/* Auxiliary Effect Slot properties. */ +#define AL_EFFECTSLOT_EFFECT 0x0001 +#define AL_EFFECTSLOT_GAIN 0x0002 +#define AL_EFFECTSLOT_AUXILIARY_SEND_AUTO 0x0003 + +/* NULL Auxiliary Slot ID to disable a source send. */ +#define AL_EFFECTSLOT_NULL 0x0000 + + +/* Filter properties. */ + +/* Lowpass filter parameters */ +#define AL_LOWPASS_GAIN 0x0001 +#define AL_LOWPASS_GAINHF 0x0002 + +/* Highpass filter parameters */ +#define AL_HIGHPASS_GAIN 0x0001 +#define AL_HIGHPASS_GAINLF 0x0002 + +/* Bandpass filter parameters */ +#define AL_BANDPASS_GAIN 0x0001 +#define AL_BANDPASS_GAINLF 0x0002 +#define AL_BANDPASS_GAINHF 0x0003 + +/* Filter type */ +#define AL_FILTER_FIRST_PARAMETER 0x0000 +#define AL_FILTER_LAST_PARAMETER 0x8000 +#define AL_FILTER_TYPE 0x8001 + +/* Filter types, used with the AL_FILTER_TYPE property */ +#define AL_FILTER_NULL 0x0000 +#define AL_FILTER_LOWPASS 0x0001 +#define AL_FILTER_HIGHPASS 0x0002 +#define AL_FILTER_BANDPASS 0x0003 + + +/* Effect object function types. */ +typedef void (AL_APIENTRY *LPALGENEFFECTS)(ALsizei, ALuint*); +typedef void (AL_APIENTRY *LPALDELETEEFFECTS)(ALsizei, const ALuint*); +typedef ALboolean (AL_APIENTRY *LPALISEFFECT)(ALuint); +typedef void (AL_APIENTRY *LPALEFFECTI)(ALuint, ALenum, ALint); +typedef void (AL_APIENTRY *LPALEFFECTIV)(ALuint, ALenum, const ALint*); +typedef void (AL_APIENTRY *LPALEFFECTF)(ALuint, ALenum, ALfloat); +typedef void (AL_APIENTRY *LPALEFFECTFV)(ALuint, ALenum, const ALfloat*); +typedef void (AL_APIENTRY *LPALGETEFFECTI)(ALuint, ALenum, ALint*); +typedef void (AL_APIENTRY *LPALGETEFFECTIV)(ALuint, ALenum, ALint*); +typedef void (AL_APIENTRY *LPALGETEFFECTF)(ALuint, ALenum, ALfloat*); +typedef void (AL_APIENTRY *LPALGETEFFECTFV)(ALuint, ALenum, ALfloat*); + +/* Filter object function types. */ +typedef void (AL_APIENTRY *LPALGENFILTERS)(ALsizei, ALuint*); +typedef void (AL_APIENTRY *LPALDELETEFILTERS)(ALsizei, const ALuint*); +typedef ALboolean (AL_APIENTRY *LPALISFILTER)(ALuint); +typedef void (AL_APIENTRY *LPALFILTERI)(ALuint, ALenum, ALint); +typedef void (AL_APIENTRY *LPALFILTERIV)(ALuint, ALenum, const ALint*); +typedef void (AL_APIENTRY *LPALFILTERF)(ALuint, ALenum, ALfloat); +typedef void (AL_APIENTRY *LPALFILTERFV)(ALuint, ALenum, const ALfloat*); +typedef void (AL_APIENTRY *LPALGETFILTERI)(ALuint, ALenum, ALint*); +typedef void (AL_APIENTRY *LPALGETFILTERIV)(ALuint, ALenum, ALint*); +typedef void (AL_APIENTRY *LPALGETFILTERF)(ALuint, ALenum, ALfloat*); +typedef void (AL_APIENTRY *LPALGETFILTERFV)(ALuint, ALenum, ALfloat*); + +/* Auxiliary Effect Slot object function types. */ +typedef void (AL_APIENTRY *LPALGENAUXILIARYEFFECTSLOTS)(ALsizei, ALuint*); +typedef void (AL_APIENTRY *LPALDELETEAUXILIARYEFFECTSLOTS)(ALsizei, const ALuint*); +typedef ALboolean (AL_APIENTRY *LPALISAUXILIARYEFFECTSLOT)(ALuint); +typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTI)(ALuint, ALenum, ALint); +typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTIV)(ALuint, ALenum, const ALint*); +typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTF)(ALuint, ALenum, ALfloat); +typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTFV)(ALuint, ALenum, const ALfloat*); +typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTI)(ALuint, ALenum, ALint*); +typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTIV)(ALuint, ALenum, ALint*); +typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTF)(ALuint, ALenum, ALfloat*); +typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTFV)(ALuint, ALenum, ALfloat*); + +#ifdef AL_ALEXT_PROTOTYPES +AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects); +AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects); +AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect); +AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint iValue); +AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *piValues); +AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat flValue); +AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *pflValues); +AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *piValue); +AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *piValues); +AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *pflValue); +AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *pflValues); + +AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters); +AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters); +AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter); +AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint iValue); +AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *piValues); +AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat flValue); +AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *pflValues); +AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *piValue); +AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *piValues); +AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *pflValue); +AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *pflValues); + +AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots); +AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots); +AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot); +AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint iValue); +AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *piValues); +AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat flValue); +AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *pflValues); +AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *piValue); +AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *piValues); +AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *pflValue); +AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *pflValues); +#endif + +/* Filter ranges and defaults. */ + +/* Lowpass filter */ +#define AL_LOWPASS_MIN_GAIN (0.0f) +#define AL_LOWPASS_MAX_GAIN (1.0f) +#define AL_LOWPASS_DEFAULT_GAIN (1.0f) + +#define AL_LOWPASS_MIN_GAINHF (0.0f) +#define AL_LOWPASS_MAX_GAINHF (1.0f) +#define AL_LOWPASS_DEFAULT_GAINHF (1.0f) + +/* Highpass filter */ +#define AL_HIGHPASS_MIN_GAIN (0.0f) +#define AL_HIGHPASS_MAX_GAIN (1.0f) +#define AL_HIGHPASS_DEFAULT_GAIN (1.0f) + +#define AL_HIGHPASS_MIN_GAINLF (0.0f) +#define AL_HIGHPASS_MAX_GAINLF (1.0f) +#define AL_HIGHPASS_DEFAULT_GAINLF (1.0f) + +/* Bandpass filter */ +#define AL_BANDPASS_MIN_GAIN (0.0f) +#define AL_BANDPASS_MAX_GAIN (1.0f) +#define AL_BANDPASS_DEFAULT_GAIN (1.0f) + +#define AL_BANDPASS_MIN_GAINHF (0.0f) +#define AL_BANDPASS_MAX_GAINHF (1.0f) +#define AL_BANDPASS_DEFAULT_GAINHF (1.0f) + +#define AL_BANDPASS_MIN_GAINLF (0.0f) +#define AL_BANDPASS_MAX_GAINLF (1.0f) +#define AL_BANDPASS_DEFAULT_GAINLF (1.0f) + + +/* Effect parameter ranges and defaults. */ + +/* Standard reverb effect */ +#define AL_REVERB_MIN_DENSITY (0.0f) +#define AL_REVERB_MAX_DENSITY (1.0f) +#define AL_REVERB_DEFAULT_DENSITY (1.0f) + +#define AL_REVERB_MIN_DIFFUSION (0.0f) +#define AL_REVERB_MAX_DIFFUSION (1.0f) +#define AL_REVERB_DEFAULT_DIFFUSION (1.0f) + +#define AL_REVERB_MIN_GAIN (0.0f) +#define AL_REVERB_MAX_GAIN (1.0f) +#define AL_REVERB_DEFAULT_GAIN (0.32f) + +#define AL_REVERB_MIN_GAINHF (0.0f) +#define AL_REVERB_MAX_GAINHF (1.0f) +#define AL_REVERB_DEFAULT_GAINHF (0.89f) + +#define AL_REVERB_MIN_DECAY_TIME (0.1f) +#define AL_REVERB_MAX_DECAY_TIME (20.0f) +#define AL_REVERB_DEFAULT_DECAY_TIME (1.49f) + +#define AL_REVERB_MIN_DECAY_HFRATIO (0.1f) +#define AL_REVERB_MAX_DECAY_HFRATIO (2.0f) +#define AL_REVERB_DEFAULT_DECAY_HFRATIO (0.83f) + +#define AL_REVERB_MIN_REFLECTIONS_GAIN (0.0f) +#define AL_REVERB_MAX_REFLECTIONS_GAIN (3.16f) +#define AL_REVERB_DEFAULT_REFLECTIONS_GAIN (0.05f) + +#define AL_REVERB_MIN_REFLECTIONS_DELAY (0.0f) +#define AL_REVERB_MAX_REFLECTIONS_DELAY (0.3f) +#define AL_REVERB_DEFAULT_REFLECTIONS_DELAY (0.007f) + +#define AL_REVERB_MIN_LATE_REVERB_GAIN (0.0f) +#define AL_REVERB_MAX_LATE_REVERB_GAIN (10.0f) +#define AL_REVERB_DEFAULT_LATE_REVERB_GAIN (1.26f) + +#define AL_REVERB_MIN_LATE_REVERB_DELAY (0.0f) +#define AL_REVERB_MAX_LATE_REVERB_DELAY (0.1f) +#define AL_REVERB_DEFAULT_LATE_REVERB_DELAY (0.011f) + +#define AL_REVERB_MIN_AIR_ABSORPTION_GAINHF (0.892f) +#define AL_REVERB_MAX_AIR_ABSORPTION_GAINHF (1.0f) +#define AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF (0.994f) + +#define AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR (0.0f) +#define AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR (10.0f) +#define AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f) + +#define AL_REVERB_MIN_DECAY_HFLIMIT AL_FALSE +#define AL_REVERB_MAX_DECAY_HFLIMIT AL_TRUE +#define AL_REVERB_DEFAULT_DECAY_HFLIMIT AL_TRUE + +/* EAX reverb effect */ +#define AL_EAXREVERB_MIN_DENSITY (0.0f) +#define AL_EAXREVERB_MAX_DENSITY (1.0f) +#define AL_EAXREVERB_DEFAULT_DENSITY (1.0f) + +#define AL_EAXREVERB_MIN_DIFFUSION (0.0f) +#define AL_EAXREVERB_MAX_DIFFUSION (1.0f) +#define AL_EAXREVERB_DEFAULT_DIFFUSION (1.0f) + +#define AL_EAXREVERB_MIN_GAIN (0.0f) +#define AL_EAXREVERB_MAX_GAIN (1.0f) +#define AL_EAXREVERB_DEFAULT_GAIN (0.32f) + +#define AL_EAXREVERB_MIN_GAINHF (0.0f) +#define AL_EAXREVERB_MAX_GAINHF (1.0f) +#define AL_EAXREVERB_DEFAULT_GAINHF (0.89f) + +#define AL_EAXREVERB_MIN_GAINLF (0.0f) +#define AL_EAXREVERB_MAX_GAINLF (1.0f) +#define AL_EAXREVERB_DEFAULT_GAINLF (1.0f) + +#define AL_EAXREVERB_MIN_DECAY_TIME (0.1f) +#define AL_EAXREVERB_MAX_DECAY_TIME (20.0f) +#define AL_EAXREVERB_DEFAULT_DECAY_TIME (1.49f) + +#define AL_EAXREVERB_MIN_DECAY_HFRATIO (0.1f) +#define AL_EAXREVERB_MAX_DECAY_HFRATIO (2.0f) +#define AL_EAXREVERB_DEFAULT_DECAY_HFRATIO (0.83f) + +#define AL_EAXREVERB_MIN_DECAY_LFRATIO (0.1f) +#define AL_EAXREVERB_MAX_DECAY_LFRATIO (2.0f) +#define AL_EAXREVERB_DEFAULT_DECAY_LFRATIO (1.0f) + +#define AL_EAXREVERB_MIN_REFLECTIONS_GAIN (0.0f) +#define AL_EAXREVERB_MAX_REFLECTIONS_GAIN (3.16f) +#define AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN (0.05f) + +#define AL_EAXREVERB_MIN_REFLECTIONS_DELAY (0.0f) +#define AL_EAXREVERB_MAX_REFLECTIONS_DELAY (0.3f) +#define AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY (0.007f) + +#define AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ (0.0f) + +#define AL_EAXREVERB_MIN_LATE_REVERB_GAIN (0.0f) +#define AL_EAXREVERB_MAX_LATE_REVERB_GAIN (10.0f) +#define AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN (1.26f) + +#define AL_EAXREVERB_MIN_LATE_REVERB_DELAY (0.0f) +#define AL_EAXREVERB_MAX_LATE_REVERB_DELAY (0.1f) +#define AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY (0.011f) + +#define AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ (0.0f) + +#define AL_EAXREVERB_MIN_ECHO_TIME (0.075f) +#define AL_EAXREVERB_MAX_ECHO_TIME (0.25f) +#define AL_EAXREVERB_DEFAULT_ECHO_TIME (0.25f) + +#define AL_EAXREVERB_MIN_ECHO_DEPTH (0.0f) +#define AL_EAXREVERB_MAX_ECHO_DEPTH (1.0f) +#define AL_EAXREVERB_DEFAULT_ECHO_DEPTH (0.0f) + +#define AL_EAXREVERB_MIN_MODULATION_TIME (0.04f) +#define AL_EAXREVERB_MAX_MODULATION_TIME (4.0f) +#define AL_EAXREVERB_DEFAULT_MODULATION_TIME (0.25f) + +#define AL_EAXREVERB_MIN_MODULATION_DEPTH (0.0f) +#define AL_EAXREVERB_MAX_MODULATION_DEPTH (1.0f) +#define AL_EAXREVERB_DEFAULT_MODULATION_DEPTH (0.0f) + +#define AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF (0.892f) +#define AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF (1.0f) +#define AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF (0.994f) + +#define AL_EAXREVERB_MIN_HFREFERENCE (1000.0f) +#define AL_EAXREVERB_MAX_HFREFERENCE (20000.0f) +#define AL_EAXREVERB_DEFAULT_HFREFERENCE (5000.0f) + +#define AL_EAXREVERB_MIN_LFREFERENCE (20.0f) +#define AL_EAXREVERB_MAX_LFREFERENCE (1000.0f) +#define AL_EAXREVERB_DEFAULT_LFREFERENCE (250.0f) + +#define AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR (0.0f) +#define AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR (10.0f) +#define AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f) + +#define AL_EAXREVERB_MIN_DECAY_HFLIMIT AL_FALSE +#define AL_EAXREVERB_MAX_DECAY_HFLIMIT AL_TRUE +#define AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT AL_TRUE + +/* Chorus effect */ +#define AL_CHORUS_WAVEFORM_SINUSOID (0) +#define AL_CHORUS_WAVEFORM_TRIANGLE (1) + +#define AL_CHORUS_MIN_WAVEFORM (0) +#define AL_CHORUS_MAX_WAVEFORM (1) +#define AL_CHORUS_DEFAULT_WAVEFORM (1) + +#define AL_CHORUS_MIN_PHASE (-180) +#define AL_CHORUS_MAX_PHASE (180) +#define AL_CHORUS_DEFAULT_PHASE (90) + +#define AL_CHORUS_MIN_RATE (0.0f) +#define AL_CHORUS_MAX_RATE (10.0f) +#define AL_CHORUS_DEFAULT_RATE (1.1f) + +#define AL_CHORUS_MIN_DEPTH (0.0f) +#define AL_CHORUS_MAX_DEPTH (1.0f) +#define AL_CHORUS_DEFAULT_DEPTH (0.1f) + +#define AL_CHORUS_MIN_FEEDBACK (-1.0f) +#define AL_CHORUS_MAX_FEEDBACK (1.0f) +#define AL_CHORUS_DEFAULT_FEEDBACK (0.25f) + +#define AL_CHORUS_MIN_DELAY (0.0f) +#define AL_CHORUS_MAX_DELAY (0.016f) +#define AL_CHORUS_DEFAULT_DELAY (0.016f) + +/* Distortion effect */ +#define AL_DISTORTION_MIN_EDGE (0.0f) +#define AL_DISTORTION_MAX_EDGE (1.0f) +#define AL_DISTORTION_DEFAULT_EDGE (0.2f) + +#define AL_DISTORTION_MIN_GAIN (0.01f) +#define AL_DISTORTION_MAX_GAIN (1.0f) +#define AL_DISTORTION_DEFAULT_GAIN (0.05f) + +#define AL_DISTORTION_MIN_LOWPASS_CUTOFF (80.0f) +#define AL_DISTORTION_MAX_LOWPASS_CUTOFF (24000.0f) +#define AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF (8000.0f) + +#define AL_DISTORTION_MIN_EQCENTER (80.0f) +#define AL_DISTORTION_MAX_EQCENTER (24000.0f) +#define AL_DISTORTION_DEFAULT_EQCENTER (3600.0f) + +#define AL_DISTORTION_MIN_EQBANDWIDTH (80.0f) +#define AL_DISTORTION_MAX_EQBANDWIDTH (24000.0f) +#define AL_DISTORTION_DEFAULT_EQBANDWIDTH (3600.0f) + +/* Echo effect */ +#define AL_ECHO_MIN_DELAY (0.0f) +#define AL_ECHO_MAX_DELAY (0.207f) +#define AL_ECHO_DEFAULT_DELAY (0.1f) + +#define AL_ECHO_MIN_LRDELAY (0.0f) +#define AL_ECHO_MAX_LRDELAY (0.404f) +#define AL_ECHO_DEFAULT_LRDELAY (0.1f) + +#define AL_ECHO_MIN_DAMPING (0.0f) +#define AL_ECHO_MAX_DAMPING (0.99f) +#define AL_ECHO_DEFAULT_DAMPING (0.5f) + +#define AL_ECHO_MIN_FEEDBACK (0.0f) +#define AL_ECHO_MAX_FEEDBACK (1.0f) +#define AL_ECHO_DEFAULT_FEEDBACK (0.5f) + +#define AL_ECHO_MIN_SPREAD (-1.0f) +#define AL_ECHO_MAX_SPREAD (1.0f) +#define AL_ECHO_DEFAULT_SPREAD (-1.0f) + +/* Flanger effect */ +#define AL_FLANGER_WAVEFORM_SINUSOID (0) +#define AL_FLANGER_WAVEFORM_TRIANGLE (1) + +#define AL_FLANGER_MIN_WAVEFORM (0) +#define AL_FLANGER_MAX_WAVEFORM (1) +#define AL_FLANGER_DEFAULT_WAVEFORM (1) + +#define AL_FLANGER_MIN_PHASE (-180) +#define AL_FLANGER_MAX_PHASE (180) +#define AL_FLANGER_DEFAULT_PHASE (0) + +#define AL_FLANGER_MIN_RATE (0.0f) +#define AL_FLANGER_MAX_RATE (10.0f) +#define AL_FLANGER_DEFAULT_RATE (0.27f) + +#define AL_FLANGER_MIN_DEPTH (0.0f) +#define AL_FLANGER_MAX_DEPTH (1.0f) +#define AL_FLANGER_DEFAULT_DEPTH (1.0f) + +#define AL_FLANGER_MIN_FEEDBACK (-1.0f) +#define AL_FLANGER_MAX_FEEDBACK (1.0f) +#define AL_FLANGER_DEFAULT_FEEDBACK (-0.5f) + +#define AL_FLANGER_MIN_DELAY (0.0f) +#define AL_FLANGER_MAX_DELAY (0.004f) +#define AL_FLANGER_DEFAULT_DELAY (0.002f) + +/* Frequency shifter effect */ +#define AL_FREQUENCY_SHIFTER_MIN_FREQUENCY (0.0f) +#define AL_FREQUENCY_SHIFTER_MAX_FREQUENCY (24000.0f) +#define AL_FREQUENCY_SHIFTER_DEFAULT_FREQUENCY (0.0f) + +#define AL_FREQUENCY_SHIFTER_MIN_LEFT_DIRECTION (0) +#define AL_FREQUENCY_SHIFTER_MAX_LEFT_DIRECTION (2) +#define AL_FREQUENCY_SHIFTER_DEFAULT_LEFT_DIRECTION (0) + +#define AL_FREQUENCY_SHIFTER_DIRECTION_DOWN (0) +#define AL_FREQUENCY_SHIFTER_DIRECTION_UP (1) +#define AL_FREQUENCY_SHIFTER_DIRECTION_OFF (2) + +#define AL_FREQUENCY_SHIFTER_MIN_RIGHT_DIRECTION (0) +#define AL_FREQUENCY_SHIFTER_MAX_RIGHT_DIRECTION (2) +#define AL_FREQUENCY_SHIFTER_DEFAULT_RIGHT_DIRECTION (0) + +/* Vocal morpher effect */ +#define AL_VOCAL_MORPHER_MIN_PHONEMEA (0) +#define AL_VOCAL_MORPHER_MAX_PHONEMEA (29) +#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEA (0) + +#define AL_VOCAL_MORPHER_MIN_PHONEMEA_COARSE_TUNING (-24) +#define AL_VOCAL_MORPHER_MAX_PHONEMEA_COARSE_TUNING (24) +#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEA_COARSE_TUNING (0) + +#define AL_VOCAL_MORPHER_MIN_PHONEMEB (0) +#define AL_VOCAL_MORPHER_MAX_PHONEMEB (29) +#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEB (10) + +#define AL_VOCAL_MORPHER_MIN_PHONEMEB_COARSE_TUNING (-24) +#define AL_VOCAL_MORPHER_MAX_PHONEMEB_COARSE_TUNING (24) +#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEB_COARSE_TUNING (0) + +#define AL_VOCAL_MORPHER_PHONEME_A (0) +#define AL_VOCAL_MORPHER_PHONEME_E (1) +#define AL_VOCAL_MORPHER_PHONEME_I (2) +#define AL_VOCAL_MORPHER_PHONEME_O (3) +#define AL_VOCAL_MORPHER_PHONEME_U (4) +#define AL_VOCAL_MORPHER_PHONEME_AA (5) +#define AL_VOCAL_MORPHER_PHONEME_AE (6) +#define AL_VOCAL_MORPHER_PHONEME_AH (7) +#define AL_VOCAL_MORPHER_PHONEME_AO (8) +#define AL_VOCAL_MORPHER_PHONEME_EH (9) +#define AL_VOCAL_MORPHER_PHONEME_ER (10) +#define AL_VOCAL_MORPHER_PHONEME_IH (11) +#define AL_VOCAL_MORPHER_PHONEME_IY (12) +#define AL_VOCAL_MORPHER_PHONEME_UH (13) +#define AL_VOCAL_MORPHER_PHONEME_UW (14) +#define AL_VOCAL_MORPHER_PHONEME_B (15) +#define AL_VOCAL_MORPHER_PHONEME_D (16) +#define AL_VOCAL_MORPHER_PHONEME_F (17) +#define AL_VOCAL_MORPHER_PHONEME_G (18) +#define AL_VOCAL_MORPHER_PHONEME_J (19) +#define AL_VOCAL_MORPHER_PHONEME_K (20) +#define AL_VOCAL_MORPHER_PHONEME_L (21) +#define AL_VOCAL_MORPHER_PHONEME_M (22) +#define AL_VOCAL_MORPHER_PHONEME_N (23) +#define AL_VOCAL_MORPHER_PHONEME_P (24) +#define AL_VOCAL_MORPHER_PHONEME_R (25) +#define AL_VOCAL_MORPHER_PHONEME_S (26) +#define AL_VOCAL_MORPHER_PHONEME_T (27) +#define AL_VOCAL_MORPHER_PHONEME_V (28) +#define AL_VOCAL_MORPHER_PHONEME_Z (29) + +#define AL_VOCAL_MORPHER_WAVEFORM_SINUSOID (0) +#define AL_VOCAL_MORPHER_WAVEFORM_TRIANGLE (1) +#define AL_VOCAL_MORPHER_WAVEFORM_SAWTOOTH (2) + +#define AL_VOCAL_MORPHER_MIN_WAVEFORM (0) +#define AL_VOCAL_MORPHER_MAX_WAVEFORM (2) +#define AL_VOCAL_MORPHER_DEFAULT_WAVEFORM (0) + +#define AL_VOCAL_MORPHER_MIN_RATE (0.0f) +#define AL_VOCAL_MORPHER_MAX_RATE (10.0f) +#define AL_VOCAL_MORPHER_DEFAULT_RATE (1.41f) + +/* Pitch shifter effect */ +#define AL_PITCH_SHIFTER_MIN_COARSE_TUNE (-12) +#define AL_PITCH_SHIFTER_MAX_COARSE_TUNE (12) +#define AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE (12) + +#define AL_PITCH_SHIFTER_MIN_FINE_TUNE (-50) +#define AL_PITCH_SHIFTER_MAX_FINE_TUNE (50) +#define AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE (0) + +/* Ring modulator effect */ +#define AL_RING_MODULATOR_MIN_FREQUENCY (0.0f) +#define AL_RING_MODULATOR_MAX_FREQUENCY (8000.0f) +#define AL_RING_MODULATOR_DEFAULT_FREQUENCY (440.0f) + +#define AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF (0.0f) +#define AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF (24000.0f) +#define AL_RING_MODULATOR_DEFAULT_HIGHPASS_CUTOFF (800.0f) + +#define AL_RING_MODULATOR_SINUSOID (0) +#define AL_RING_MODULATOR_SAWTOOTH (1) +#define AL_RING_MODULATOR_SQUARE (2) + +#define AL_RING_MODULATOR_MIN_WAVEFORM (0) +#define AL_RING_MODULATOR_MAX_WAVEFORM (2) +#define AL_RING_MODULATOR_DEFAULT_WAVEFORM (0) + +/* Autowah effect */ +#define AL_AUTOWAH_MIN_ATTACK_TIME (0.0001f) +#define AL_AUTOWAH_MAX_ATTACK_TIME (1.0f) +#define AL_AUTOWAH_DEFAULT_ATTACK_TIME (0.06f) + +#define AL_AUTOWAH_MIN_RELEASE_TIME (0.0001f) +#define AL_AUTOWAH_MAX_RELEASE_TIME (1.0f) +#define AL_AUTOWAH_DEFAULT_RELEASE_TIME (0.06f) + +#define AL_AUTOWAH_MIN_RESONANCE (2.0f) +#define AL_AUTOWAH_MAX_RESONANCE (1000.0f) +#define AL_AUTOWAH_DEFAULT_RESONANCE (1000.0f) + +#define AL_AUTOWAH_MIN_PEAK_GAIN (0.00003f) +#define AL_AUTOWAH_MAX_PEAK_GAIN (31621.0f) +#define AL_AUTOWAH_DEFAULT_PEAK_GAIN (11.22f) + +/* Compressor effect */ +#define AL_COMPRESSOR_MIN_ONOFF (0) +#define AL_COMPRESSOR_MAX_ONOFF (1) +#define AL_COMPRESSOR_DEFAULT_ONOFF (1) + +/* Equalizer effect */ +#define AL_EQUALIZER_MIN_LOW_GAIN (0.126f) +#define AL_EQUALIZER_MAX_LOW_GAIN (7.943f) +#define AL_EQUALIZER_DEFAULT_LOW_GAIN (1.0f) + +#define AL_EQUALIZER_MIN_LOW_CUTOFF (50.0f) +#define AL_EQUALIZER_MAX_LOW_CUTOFF (800.0f) +#define AL_EQUALIZER_DEFAULT_LOW_CUTOFF (200.0f) + +#define AL_EQUALIZER_MIN_MID1_GAIN (0.126f) +#define AL_EQUALIZER_MAX_MID1_GAIN (7.943f) +#define AL_EQUALIZER_DEFAULT_MID1_GAIN (1.0f) + +#define AL_EQUALIZER_MIN_MID1_CENTER (200.0f) +#define AL_EQUALIZER_MAX_MID1_CENTER (3000.0f) +#define AL_EQUALIZER_DEFAULT_MID1_CENTER (500.0f) + +#define AL_EQUALIZER_MIN_MID1_WIDTH (0.01f) +#define AL_EQUALIZER_MAX_MID1_WIDTH (1.0f) +#define AL_EQUALIZER_DEFAULT_MID1_WIDTH (1.0f) + +#define AL_EQUALIZER_MIN_MID2_GAIN (0.126f) +#define AL_EQUALIZER_MAX_MID2_GAIN (7.943f) +#define AL_EQUALIZER_DEFAULT_MID2_GAIN (1.0f) + +#define AL_EQUALIZER_MIN_MID2_CENTER (1000.0f) +#define AL_EQUALIZER_MAX_MID2_CENTER (8000.0f) +#define AL_EQUALIZER_DEFAULT_MID2_CENTER (3000.0f) + +#define AL_EQUALIZER_MIN_MID2_WIDTH (0.01f) +#define AL_EQUALIZER_MAX_MID2_WIDTH (1.0f) +#define AL_EQUALIZER_DEFAULT_MID2_WIDTH (1.0f) + +#define AL_EQUALIZER_MIN_HIGH_GAIN (0.126f) +#define AL_EQUALIZER_MAX_HIGH_GAIN (7.943f) +#define AL_EQUALIZER_DEFAULT_HIGH_GAIN (1.0f) + +#define AL_EQUALIZER_MIN_HIGH_CUTOFF (4000.0f) +#define AL_EQUALIZER_MAX_HIGH_CUTOFF (16000.0f) +#define AL_EQUALIZER_DEFAULT_HIGH_CUTOFF (6000.0f) + + +/* Source parameter value ranges and defaults. */ +#define AL_MIN_AIR_ABSORPTION_FACTOR (0.0f) +#define AL_MAX_AIR_ABSORPTION_FACTOR (10.0f) +#define AL_DEFAULT_AIR_ABSORPTION_FACTOR (0.0f) + +#define AL_MIN_ROOM_ROLLOFF_FACTOR (0.0f) +#define AL_MAX_ROOM_ROLLOFF_FACTOR (10.0f) +#define AL_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f) + +#define AL_MIN_CONE_OUTER_GAINHF (0.0f) +#define AL_MAX_CONE_OUTER_GAINHF (1.0f) +#define AL_DEFAULT_CONE_OUTER_GAINHF (1.0f) + +#define AL_MIN_DIRECT_FILTER_GAINHF_AUTO AL_FALSE +#define AL_MAX_DIRECT_FILTER_GAINHF_AUTO AL_TRUE +#define AL_DEFAULT_DIRECT_FILTER_GAINHF_AUTO AL_TRUE + +#define AL_MIN_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_FALSE +#define AL_MAX_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_TRUE +#define AL_DEFAULT_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_TRUE + +#define AL_MIN_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_FALSE +#define AL_MAX_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_TRUE +#define AL_DEFAULT_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_TRUE + + +/* Listener parameter value ranges and defaults. */ +#define AL_MIN_METERS_PER_UNIT FLT_MIN +#define AL_MAX_METERS_PER_UNIT FLT_MAX +#define AL_DEFAULT_METERS_PER_UNIT (1.0f) + + +#ifdef __cplusplus +} /* extern "C" */ +#endif + +#endif /* AL_EFX_H */ diff --git a/Externals/OpenAL/include/xram.h b/Externals/OpenAL/include/xram.h new file mode 100644 index 0000000000..cfff054329 --- /dev/null +++ b/Externals/OpenAL/include/xram.h @@ -0,0 +1,94 @@ +#include + +// 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(...); +// +////////////////////////////////////////////////////////////////////////////// diff --git a/Externals/soundtouch/AAFilter.cpp b/Externals/soundtouch/AAFilter.cpp new file mode 100644 index 0000000000..f099bced93 --- /dev/null +++ b/Externals/soundtouch/AAFilter.cpp @@ -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 +#include +#include +#include +#include "AAFilter.h" +#include "FIRFilter.h" + +using namespace soundtouch; + +#define PI 3.141592655357989 +#define TWOPI (2 * PI) + +/***************************************************************************** + * + * Implementation of the class 'AAFilter' + * + *****************************************************************************/ + +AAFilter::AAFilter(uint len) +{ + pFIR = FIRFilter::newInstance(); + cutoffFreq = 0.5; + setLength(len); +} + + + +AAFilter::~AAFilter() +{ + delete pFIR; +} + + + +// Sets new anti-alias filter cut-off edge frequency, scaled to +// sampling frequency (nyquist frequency = 0.5). +// The filter will cut frequencies higher than the given frequency. +void AAFilter::setCutoffFreq(double newCutoffFreq) +{ + cutoffFreq = newCutoffFreq; + calculateCoeffs(); +} + + + +// Sets number of FIR filter taps +void AAFilter::setLength(uint newLength) +{ + length = newLength; + calculateCoeffs(); +} + + + +// Calculates coefficients for a low-pass FIR filter using Hamming window +void AAFilter::calculateCoeffs() +{ + uint i; + double cntTemp, temp, tempCoeff,h, w; + double fc2, wc; + double scaleCoeff, sum; + double *work; + SAMPLETYPE *coeffs; + + assert(length >= 2); + assert(length % 4 == 0); + assert(cutoffFreq >= 0); + assert(cutoffFreq <= 0.5); + + work = new double[length]; + coeffs = new SAMPLETYPE[length]; + + fc2 = 2.0 * cutoffFreq; + wc = PI * fc2; + tempCoeff = TWOPI / (double)length; + + sum = 0; + for (i = 0; i < length; i ++) + { + cntTemp = (double)i - (double)(length / 2); + + temp = cntTemp * wc; + if (temp != 0) + { + h = fc2 * sin(temp) / temp; // sinc function + } + else + { + h = 1.0; + } + w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window + + temp = w * h; + work[i] = temp; + + // calc net sum of coefficients + sum += temp; + } + + // ensure the sum of coefficients is larger than zero + assert(sum > 0); + + // ensure we've really designed a lowpass filter... + assert(work[length/2] > 0); + assert(work[length/2 + 1] > -1e-6); + assert(work[length/2 - 1] > -1e-6); + + // Calculate a scaling coefficient in such a way that the result can be + // divided by 16384 + scaleCoeff = 16384.0f / sum; + + for (i = 0; i < length; i ++) + { + // scale & round to nearest integer + temp = work[i] * scaleCoeff; + temp += (temp >= 0) ? 0.5 : -0.5; + // ensure no overfloods + assert(temp >= -32768 && temp <= 32767); + coeffs[i] = (SAMPLETYPE)temp; + } + + // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384 + pFIR->setCoefficients(coeffs, length, 14); + + delete[] work; + delete[] coeffs; +} + + +// Applies the filter to the given sequence of samples. +// Note : The amount of outputted samples is by value of 'filter length' +// smaller than the amount of input samples. +uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const +{ + return pFIR->evaluate(dest, src, numSamples, numChannels); +} + + +uint AAFilter::getLength() const +{ + return pFIR->getLength(); +} diff --git a/Externals/soundtouch/AAFilter.h b/Externals/soundtouch/AAFilter.h new file mode 100644 index 0000000000..d0997570d3 --- /dev/null +++ b/Externals/soundtouch/AAFilter.h @@ -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 diff --git a/Externals/soundtouch/BPMDetect.cpp b/Externals/soundtouch/BPMDetect.cpp new file mode 100644 index 0000000000..a48cbd97c3 --- /dev/null +++ b/Externals/soundtouch/BPMDetect.cpp @@ -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 +#include +#include +#include +#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); +} diff --git a/Externals/soundtouch/BPMDetect.h b/Externals/soundtouch/BPMDetect.h new file mode 100644 index 0000000000..72489894bd --- /dev/null +++ b/Externals/soundtouch/BPMDetect.h @@ -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_ diff --git a/Externals/soundtouch/CMakeLists.txt b/Externals/soundtouch/CMakeLists.txt new file mode 100644 index 0000000000..a09aac1946 --- /dev/null +++ b/Externals/soundtouch/CMakeLists.txt @@ -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}) diff --git a/Externals/soundtouch/FIFOSampleBuffer.cpp b/Externals/soundtouch/FIFOSampleBuffer.cpp new file mode 100644 index 0000000000..7f088b80be --- /dev/null +++ b/Externals/soundtouch/FIFOSampleBuffer.cpp @@ -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 +#include +#include +#include + +#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; +} + diff --git a/Externals/soundtouch/FIFOSampleBuffer.h b/Externals/soundtouch/FIFOSampleBuffer.h new file mode 100644 index 0000000000..3789b4d307 --- /dev/null +++ b/Externals/soundtouch/FIFOSampleBuffer.h @@ -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 diff --git a/Externals/soundtouch/FIFOSamplePipe.h b/Externals/soundtouch/FIFOSamplePipe.h new file mode 100644 index 0000000000..f26c57b0b2 --- /dev/null +++ b/Externals/soundtouch/FIFOSamplePipe.h @@ -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 +#include +#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 diff --git a/Externals/soundtouch/FIRFilter.cpp b/Externals/soundtouch/FIRFilter.cpp new file mode 100644 index 0000000000..1570516b77 --- /dev/null +++ b/Externals/soundtouch/FIRFilter.cpp @@ -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 +#include +#include +#include +#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; + } +} diff --git a/Externals/soundtouch/FIRFilter.h b/Externals/soundtouch/FIRFilter.h new file mode 100644 index 0000000000..e1563094af --- /dev/null +++ b/Externals/soundtouch/FIRFilter.h @@ -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 +#include "STTypes.h" + +namespace soundtouch +{ + +class FIRFilter +{ +protected: + // Number of FIR filter taps + uint length; + // Number of FIR filter taps divided by 8 + uint lengthDiv8; + + // Result divider factor in 2^k format + uint resultDivFactor; + + // Result divider value. + SAMPLETYPE resultDivider; + + // Memory for filter coefficients + SAMPLETYPE *filterCoeffs; + + virtual uint evaluateFilterStereo(SAMPLETYPE *dest, + const SAMPLETYPE *src, + uint numSamples) const; + virtual uint evaluateFilterMono(SAMPLETYPE *dest, + const SAMPLETYPE *src, + uint numSamples) const; + +public: + FIRFilter(); + virtual ~FIRFilter(); + + /// Operator 'new' is overloaded so that it automatically creates a suitable instance + /// depending on if we've a MMX-capable CPU available or not. + static void * operator new(size_t s); + + static FIRFilter *newInstance(); + + /// Applies the filter to the given sequence of samples. + /// Note : The amount of outputted samples is by value of 'filter_length' + /// smaller than the amount of input samples. + /// + /// \return Number of samples copied to 'dest'. + uint evaluate(SAMPLETYPE *dest, + const SAMPLETYPE *src, + uint numSamples, + uint numChannels) const; + + uint getLength() const; + + virtual void setCoefficients(const SAMPLETYPE *coeffs, + uint newLength, + uint uResultDivFactor); +}; + + +// Optional subclasses that implement CPU-specific optimizations: + +#ifdef SOUNDTOUCH_ALLOW_MMX + +/// Class that implements MMX optimized functions exclusive for 16bit integer samples type. + class FIRFilterMMX : public FIRFilter + { + protected: + short *filterCoeffsUnalign; + short *filterCoeffsAlign; + + virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const; + public: + FIRFilterMMX(); + ~FIRFilterMMX(); + + virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor); + }; + +#endif // SOUNDTOUCH_ALLOW_MMX + + +#ifdef SOUNDTOUCH_ALLOW_SSE + /// Class that implements SSE optimized functions exclusive for floating point samples type. + class FIRFilterSSE : public FIRFilter + { + protected: + float *filterCoeffsUnalign; + float *filterCoeffsAlign; + + virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const; + public: + FIRFilterSSE(); + ~FIRFilterSSE(); + + virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor); + }; + +#endif // SOUNDTOUCH_ALLOW_SSE + +} + +#endif // FIRFilter_H diff --git a/Externals/soundtouch/PeakFinder.cpp b/Externals/soundtouch/PeakFinder.cpp new file mode 100644 index 0000000000..47ed27cb50 --- /dev/null +++ b/Externals/soundtouch/PeakFinder.cpp @@ -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 +#include + +#include "PeakFinder.h" + +using namespace soundtouch; + +#define max(x, y) (((x) > (y)) ? (x) : (y)) + + +PeakFinder::PeakFinder() +{ + minPos = maxPos = 0; +} + + +// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'. +int PeakFinder::findTop(const float *data, int peakpos) const +{ + int i; + int start, end; + float refvalue; + + refvalue = data[peakpos]; + + // seek within ±10 points + start = peakpos - 10; + if (start < minPos) start = minPos; + end = peakpos + 10; + if (end > maxPos) end = maxPos; + + for (i = start; i <= end; i ++) + { + if (data[i] > refvalue) + { + peakpos = i; + refvalue = data[i]; + } + } + + // failure if max value is at edges of seek range => it's not peak, it's at slope. + if ((peakpos == start) || (peakpos == end)) return 0; + + return peakpos; +} + + +// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding +// to direction defined by 'direction' until next 'hump' after minimum value will +// begin +int PeakFinder::findGround(const float *data, int peakpos, int direction) const +{ + int lowpos; + int pos; + int climb_count; + float refvalue; + float delta; + + climb_count = 0; + refvalue = data[peakpos]; + lowpos = peakpos; + + pos = peakpos; + + while ((pos > minPos+1) && (pos < maxPos-1)) + { + int prevpos; + + prevpos = pos; + pos += direction; + + // calculate derivate + delta = data[pos] - data[prevpos]; + if (delta <= 0) + { + // going downhill, ok + if (climb_count) + { + climb_count --; // decrease climb count + } + + // check if new minimum found + if (data[pos] < refvalue) + { + // new minimum found + lowpos = pos; + refvalue = data[pos]; + } + } + else + { + // going uphill, increase climbing counter + climb_count ++; + if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit + } + } + return lowpos; +} + + +// Find offset where the value crosses the given level, when starting from 'peakpos' and +// proceeds to direction defined in 'direction' +int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const +{ + float peaklevel; + int pos; + + peaklevel = data[peakpos]; + assert(peaklevel >= level); + pos = peakpos; + while ((pos >= minPos) && (pos < maxPos)) + { + if (data[pos + direction] < level) return pos; // crossing found + pos += direction; + } + return -1; // not found +} + + +// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos' +double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const +{ + int i; + float sum; + float wsum; + + sum = 0; + wsum = 0; + for (i = firstPos; i <= lastPos; i ++) + { + sum += (float)i * data[i]; + wsum += data[i]; + } + + if (wsum < 1e-6) return 0; + return sum / wsum; +} + + + +/// get exact center of peak near given position by calculating local mass of center +double PeakFinder::getPeakCenter(const float *data, int peakpos) const +{ + float peakLevel; // peak level + int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level + float cutLevel; // cutting value + float groundLevel; // ground level of the peak + int gp1, gp2; // bottom positions of the peak 'hump' + + // find ground positions. + gp1 = findGround(data, peakpos, -1); + gp2 = findGround(data, peakpos, 1); + + groundLevel = 0.5f * (data[gp1] + data[gp2]); + peakLevel = data[peakpos]; + + // calculate 70%-level of the peak + cutLevel = 0.70f * peakLevel + 0.30f * groundLevel; + // find mid-level crossings + crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1); + crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1); + + if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak.. + + // calculate mass center of the peak surroundings + return calcMassCenter(data, crosspos1, crosspos2); +} + + + +double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos) +{ + + int i; + int peakpos; // position of peak level + double highPeak, peak; + + this->minPos = aminPos; + this->maxPos = amaxPos; + + // find absolute peak + peakpos = minPos; + peak = data[minPos]; + for (i = minPos + 1; i < maxPos; i ++) + { + if (data[i] > peak) + { + peak = data[i]; + peakpos = i; + } + } + + // Calculate exact location of the highest peak mass center + highPeak = getPeakCenter(data, peakpos); + peak = highPeak; + + // Now check if the highest peak were in fact harmonic of the true base beat peak + // - sometimes the highest peak can be Nth harmonic of the true base peak yet + // just a slightly higher than the true base + + for (i = 3; i < 10; i ++) + { + double peaktmp, harmonic; + int i1,i2; + + harmonic = (double)i * 0.5; + peakpos = (int)(highPeak / harmonic + 0.5f); + if (peakpos < minPos) break; + peakpos = findTop(data, peakpos); // seek true local maximum index + if (peakpos == 0) continue; // no local max here + + // calculate mass-center of possible harmonic peak + peaktmp = getPeakCenter(data, peakpos); + + // accept harmonic peak if + // (a) it is found + // (b) is within ±4% of the expected harmonic interval + // (c) has at least half x-corr value of the max. peak + + double diff = harmonic * peaktmp / highPeak; + if ((diff < 0.96) || (diff > 1.04)) continue; // peak too afar from expected + + // now compare to highest detected peak + i1 = (int)(highPeak + 0.5); + i2 = (int)(peaktmp + 0.5); + if (data[i2] >= 0.4*data[i1]) + { + // The harmonic is at least half as high primary peak, + // thus use the harmonic peak instead + peak = peaktmp; + } + } + + return peak; +} diff --git a/Externals/soundtouch/PeakFinder.h b/Externals/soundtouch/PeakFinder.h new file mode 100644 index 0000000000..d170b1c58b --- /dev/null +++ b/Externals/soundtouch/PeakFinder.h @@ -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_ diff --git a/Externals/soundtouch/RateTransposer.cpp b/Externals/soundtouch/RateTransposer.cpp new file mode 100644 index 0000000000..3ad8d93e5c --- /dev/null +++ b/Externals/soundtouch/RateTransposer.cpp @@ -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 +#include +#include +#include +#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; +} diff --git a/Externals/soundtouch/RateTransposer.h b/Externals/soundtouch/RateTransposer.h new file mode 100644 index 0000000000..48f7bed5c2 --- /dev/null +++ b/Externals/soundtouch/RateTransposer.h @@ -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 +#include "AAFilter.h" +#include "FIFOSamplePipe.h" +#include "FIFOSampleBuffer.h" + +#include "STTypes.h" + +namespace soundtouch +{ + +/// A common linear samplerate transposer class. +/// +/// Note: Use function "RateTransposer::newInstance()" to create a new class +/// instance instead of the "new" operator; that function automatically +/// chooses a correct implementation depending on if integer or floating +/// arithmetics are to be used. +class RateTransposer : public FIFOProcessor +{ +protected: + /// Anti-alias filter object + AAFilter *pAAFilter; + + float fRate; + + int numChannels; + + /// Buffer for collecting samples to feed the anti-alias filter between + /// two batches + FIFOSampleBuffer storeBuffer; + + /// Buffer for keeping samples between transposing & anti-alias filter + FIFOSampleBuffer tempBuffer; + + /// Output sample buffer + FIFOSampleBuffer outputBuffer; + + BOOL bUseAAFilter; + + virtual void resetRegisters() = 0; + + virtual uint transposeStereo(SAMPLETYPE *dest, + const SAMPLETYPE *src, + uint numSamples) = 0; + virtual uint transposeMono(SAMPLETYPE *dest, + const SAMPLETYPE *src, + uint numSamples) = 0; + inline uint transpose(SAMPLETYPE *dest, + const SAMPLETYPE *src, + uint numSamples); + + void downsample(const SAMPLETYPE *src, + uint numSamples); + void upsample(const SAMPLETYPE *src, + uint numSamples); + + /// Transposes sample rate by applying anti-alias filter to prevent folding. + /// Returns amount of samples returned in the "dest" buffer. + /// The maximum amount of samples that can be returned at a time is set by + /// the 'set_returnBuffer_size' function. + void processSamples(const SAMPLETYPE *src, + uint numSamples); + + +public: + RateTransposer(); + virtual ~RateTransposer(); + + /// Operator 'new' is overloaded so that it automatically creates a suitable instance + /// depending on if we're to use integer or floating point arithmetics. + static void *operator new(size_t s); + + /// Use this function instead of "new" operator to create a new instance of this class. + /// This function automatically chooses a correct implementation, depending on if + /// integer ot floating point arithmetics are to be used. + static RateTransposer *newInstance(); + + /// Returns the output buffer object + FIFOSamplePipe *getOutput() { return &outputBuffer; }; + + /// Returns the store buffer object + FIFOSamplePipe *getStore() { return &storeBuffer; }; + + /// Return anti-alias filter object + AAFilter *getAAFilter(); + + /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable + void enableAAFilter(BOOL newMode); + + /// Returns nonzero if anti-alias filter is enabled. + BOOL isAAFilterEnabled() const; + + /// Sets new target rate. Normal rate = 1.0, smaller values represent slower + /// rate, larger faster rates. + virtual void setRate(float newRate); + + /// Sets the number of channels, 1 = mono, 2 = stereo + void setChannels(int channels); + + /// Adds 'numSamples' pcs of samples from the 'samples' memory position into + /// the input of the object. + void putSamples(const SAMPLETYPE *samples, uint numSamples); + + /// Clears all the samples in the object + void clear(); + + /// Returns nonzero if there aren't any samples available for outputting. + int isEmpty() const; +}; + +} + +#endif diff --git a/Externals/soundtouch/STTypes.h b/Externals/soundtouch/STTypes.h new file mode 100644 index 0000000000..9855939119 --- /dev/null +++ b/Externals/soundtouch/STTypes.h @@ -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 + #define ST_THROW_RT_ERROR(x) {assert((const char *)x);} +#else + // use c++ standard exceptions + #include + #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 diff --git a/Externals/soundtouch/SoundTouch.cpp b/Externals/soundtouch/SoundTouch.cpp new file mode 100644 index 0000000000..f9160ed56d --- /dev/null +++ b/Externals/soundtouch/SoundTouch.cpp @@ -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 +#include +#include +#include +#include + +#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; +} diff --git a/Externals/soundtouch/SoundTouch.h b/Externals/soundtouch/SoundTouch.h new file mode 100644 index 0000000000..c6af895811 --- /dev/null +++ b/Externals/soundtouch/SoundTouch.h @@ -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 diff --git a/Externals/soundtouch/SoundTouch.vcxproj b/Externals/soundtouch/SoundTouch.vcxproj new file mode 100644 index 0000000000..8f5c475687 --- /dev/null +++ b/Externals/soundtouch/SoundTouch.vcxproj @@ -0,0 +1,352 @@ + + + + + Debug + Win32 + + + Debug + x64 + + + Release + Win32 + + + Release + x64 + + + + {68A5DD20-7057-448B-8FE0-B6AC8D205509} + + + + + + StaticLibrary + false + MultiByte + + + StaticLibrary + false + MultiByte + + + StaticLibrary + false + MultiByte + + + StaticLibrary + false + MultiByte + + + + + + + + + + + + + + + + + + + <_ProjectFileVersion>10.0.40219.1 + Release\ + Release\ + Release\ + Release\ + Debug\ + Debug\ + Debug\ + Debug\ + AllRules.ruleset + AllRules.ruleset + + + + + AllRules.ruleset + AllRules.ruleset + + + + + + + + Full + AnySuitable + true + %(AdditionalIncludeDirectories) + WIN32;NDEBUG;_LIB;%(PreprocessorDefinitions) + true + MultiThreaded + true + + + Release/SoundTouch.pch + Release/ + Release/ + Release/ + Level3 + true + + + Default + + + Win32\SoundTouch.lib + true + + + + + + + NDEBUG;%(PreprocessorDefinitions) + 0x040b + + + + + Full + AnySuitable + true + %(AdditionalIncludeDirectories) + WIN32;NDEBUG;_LIB;%(PreprocessorDefinitions) + true + MultiThreaded + true + + + Release/SoundTouch.pch + Release/ + Release/ + Release/ + Level3 + true + + + Default + StreamingSIMDExtensions2 + + + Win64\SoundTouch.lib + true + + + + + + + NDEBUG;%(PreprocessorDefinitions) + 0x040b + + + + + Disabled + %(AdditionalIncludeDirectories) + WIN32;_DEBUG;_LIB;%(PreprocessorDefinitions) + EnableFastChecks + MultiThreadedDebug + + + Debug/SoundTouch.pch + Debug/ + Debug/ + Debug/ + true + Level3 + true + EditAndContinue + Default + + + Win32\SoundTouch.lib + true + + + + + + + _DEBUG;%(PreprocessorDefinitions) + 0x040b + + + + + Disabled + %(AdditionalIncludeDirectories) + WIN32;_DEBUG;_LIB;%(PreprocessorDefinitions) + EnableFastChecks + MultiThreadedDebug + + + Debug/SoundTouch.pch + Debug/ + Debug/ + Debug/ + true + Level3 + true + ProgramDatabase + Default + + + Win64\SoundTouchD.lib + true + + + + + + + _DEBUG;%(PreprocessorDefinitions) + 0x040b + + + + + Disabled + Disabled + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + EnableFastChecks + EnableFastChecks + true + true + MaxSpeed + MaxSpeed + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + + + + Disabled + Disabled + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + EnableFastChecks + EnableFastChecks + true + true + MaxSpeed + MaxSpeed + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + + + Disabled + Disabled + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + EnableFastChecks + EnableFastChecks + true + true + MaxSpeed + MaxSpeed + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + + + + Disabled + Disabled + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + EnableFastChecks + EnableFastChecks + true + true + MaxSpeed + MaxSpeed + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + + + Disabled + Disabled + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + EnableFastChecks + EnableFastChecks + true + true + MaxSpeed + MaxSpeed + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + + + + Disabled + Disabled + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + EnableFastChecks + EnableFastChecks + true + true + MaxSpeed + MaxSpeed + %(AdditionalIncludeDirectories) + %(AdditionalIncludeDirectories) + %(PreprocessorDefinitions) + %(PreprocessorDefinitions) + + + + + + + + + + + + + + + + + + + + + \ No newline at end of file diff --git a/Externals/soundtouch/SoundTouch.vcxproj.filters b/Externals/soundtouch/SoundTouch.vcxproj.filters new file mode 100644 index 0000000000..1dee5b2a6e --- /dev/null +++ b/Externals/soundtouch/SoundTouch.vcxproj.filters @@ -0,0 +1,60 @@ + + + + + {b7786182-6345-4203-8b48-39eec5ec85dc} + cpp;c;cxx;rc;def;r;odl;idl;hpj;bat + + + {75380bb9-1e58-4186-a9cd-ec7cd284e6a5} + + + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files + + + Source Files\bpm + + + Source Files\bpm + + + + + + + + + + + + + + + + \ No newline at end of file diff --git a/Externals/soundtouch/TDStretch.cpp b/Externals/soundtouch/TDStretch.cpp new file mode 100644 index 0000000000..779cf7aa65 --- /dev/null +++ b/Externals/soundtouch/TDStretch.cpp @@ -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 +#include +#include +#include +#include + +#include "STTypes.h" +#include "cpu_detect.h" +#include "TDStretch.h" + +#include + +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 + +// 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 diff --git a/Externals/soundtouch/TDStretch.h b/Externals/soundtouch/TDStretch.h new file mode 100644 index 0000000000..6d6e7359ff --- /dev/null +++ b/Externals/soundtouch/TDStretch.h @@ -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 +#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 diff --git a/Externals/soundtouch/cpu_detect.h b/Externals/soundtouch/cpu_detect.h new file mode 100644 index 0000000000..7859ffb55d --- /dev/null +++ b/Externals/soundtouch/cpu_detect.h @@ -0,0 +1,62 @@ +//////////////////////////////////////////////////////////////////////////////// +/// +/// A header file for detecting the Intel MMX instructions set extension. +/// +/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the +/// routine implementations for x86 Windows, x86 gnu version and non-x86 +/// platforms, respectively. +/// +/// Author : Copyright (c) Olli Parviainen +/// Author e-mail : oparviai 'at' iki.fi +/// SoundTouch WWW: http://www.surina.net/soundtouch +/// +//////////////////////////////////////////////////////////////////////////////// +// +// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $ +// File revision : $Revision: 4 $ +// +// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $ +// +//////////////////////////////////////////////////////////////////////////////// +// +// License : +// +// SoundTouch audio processing library +// Copyright (c) Olli Parviainen +// +// This library is free software; you can redistribute it and/or +// modify it under the terms of the GNU Lesser General Public +// License as published by the Free Software Foundation; either +// version 2.1 of the License, or (at your option) any later version. +// +// This library is distributed in the hope that it will be useful, +// but WITHOUT ANY WARRANTY; without even the implied warranty of +// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +// Lesser General Public License for more details. +// +// You should have received a copy of the GNU Lesser General Public +// License along with this library; if not, write to the Free Software +// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +// +//////////////////////////////////////////////////////////////////////////////// + +#ifndef _CPU_DETECT_H_ +#define _CPU_DETECT_H_ + +#include "STTypes.h" + +#define SUPPORT_MMX 0x0001 +#define SUPPORT_3DNOW 0x0002 +#define SUPPORT_ALTIVEC 0x0004 +#define SUPPORT_SSE 0x0008 +#define SUPPORT_SSE2 0x0010 + +/// Checks which instruction set extensions are supported by the CPU. +/// +/// \return A bitmask of supported extensions, see SUPPORT_... defines. +uint detectCPUextensions(void); + +/// Disables given set of instruction extensions. See SUPPORT_... defines. +void disableExtensions(uint wDisableMask); + +#endif // _CPU_DETECT_H_ diff --git a/Externals/soundtouch/cpu_detect_x86.cpp b/Externals/soundtouch/cpu_detect_x86.cpp new file mode 100644 index 0000000000..1103adab57 --- /dev/null +++ b/Externals/soundtouch/cpu_detect_x86.cpp @@ -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 + #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 +} diff --git a/Externals/soundtouch/mmx_optimized.cpp b/Externals/soundtouch/mmx_optimized.cpp new file mode 100644 index 0000000000..0952fd4f84 --- /dev/null +++ b/Externals/soundtouch/mmx_optimized.cpp @@ -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 +#include +#include + + +// 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 diff --git a/Externals/soundtouch/sse_optimized.cpp b/Externals/soundtouch/sse_optimized.cpp new file mode 100644 index 0000000000..ffb6706288 --- /dev/null +++ b/Externals/soundtouch/sse_optimized.cpp @@ -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 +#include + +// 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 diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj b/Source/Core/AudioCommon/AudioCommon.vcxproj index 0c5f256c19..a983005318 100644 --- a/Source/Core/AudioCommon/AudioCommon.vcxproj +++ b/Source/Core/AudioCommon/AudioCommon.vcxproj @@ -109,78 +109,90 @@ - ..\Common\Src;%(AdditionalIncludeDirectories) + ..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories) true - + + SoundTouchD.lib;OpenAL32.lib;dsound.lib;dxerr.lib + ..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories) + - ..\Common\Src;%(AdditionalIncludeDirectories) + ..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories) true - + + SoundTouchD.lib;OpenAL32.lib;dsound.lib;dxerr.lib + ..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories) + - ..\Common\Src;%(AdditionalIncludeDirectories) + ..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories) true true true - + + SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib + ..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories) + - ..\Common\Src;%(AdditionalIncludeDirectories) + ..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories) true true true - + + SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib + ..\..\..\Externals\OpenAL\Win32;..\..\..\Externals\SoundTouch\Win32;%(AdditionalLibraryDirectories) + - ..\Common\Src;%(AdditionalIncludeDirectories) + ..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories) true true true - + + SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib + ..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories) + - ..\Common\Src;%(AdditionalIncludeDirectories) + ..\Core\Src;..\Common\Src;..\..\..\Externals;%(AdditionalIncludeDirectories) true true true - + + SoundTouch.lib;OpenAL32.lib;dsound.lib;dxerr.lib + ..\..\..\Externals\OpenAL\Win64;..\..\..\Externals\SoundTouch\Win64;%(AdditionalLibraryDirectories) + - - true - true - true - true - true - true - + + @@ -189,17 +201,11 @@ - - true - true - true - true - true - true - + + @@ -212,6 +218,14 @@ + + {68a5dd20-7057-448b-8fe0-b6ac8d205509} + true + true + false + true + false + {c87a4178-44f6-49b2-b7aa-c79af1b8c534} true diff --git a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters index 7e098ecb85..af5fe12220 100644 --- a/Source/Core/AudioCommon/AudioCommon.vcxproj.filters +++ b/Source/Core/AudioCommon/AudioCommon.vcxproj.filters @@ -21,6 +21,7 @@ SoundStreams + @@ -44,6 +45,7 @@ SoundStreams + diff --git a/Source/Core/AudioCommon/CMakeLists.txt b/Source/Core/AudioCommon/CMakeLists.txt index be8c58d14a..0e610bd847 100644 --- a/Source/Core/AudioCommon/CMakeLists.txt +++ b/Source/Core/AudioCommon/CMakeLists.txt @@ -1,5 +1,6 @@ set(SRCS Src/AudioCommon.cpp Src/AudioCommonConfig.cpp + Src/DPL2Decoder.cpp Src/Mixer.cpp Src/WaveFile.cpp Src/NullSoundStream.cpp) @@ -18,7 +19,7 @@ endif(AO_FOUND) if(OPENAL_FOUND) set(SRCS ${SRCS} Src/OpenALStream.cpp Src/aldlist.cpp) - set(LIBS ${LIBS} ${OPENAL_LIBRARY}) + set(LIBS ${LIBS} ${OPENAL_LIBRARY} SoundTouch ) endif(OPENAL_FOUND) if(PULSEAUDIO_FOUND) diff --git a/Source/Core/AudioCommon/Src/AudioCommonConfig.cpp b/Source/Core/AudioCommon/Src/AudioCommonConfig.cpp index 08902e1c3f..49f50b6880 100644 --- a/Source/Core/AudioCommon/Src/AudioCommonConfig.cpp +++ b/Source/Core/AudioCommon/Src/AudioCommonConfig.cpp @@ -42,7 +42,6 @@ void AudioCommonConfig::Load() #else file.Get("Config", "Backend", &sBackend, BACKEND_NULLSOUND); #endif - file.Get("Config", "Frequency", &iFrequency, 48000); file.Get("Config", "Volume", &m_Volume, 100); } @@ -55,7 +54,6 @@ void AudioCommonConfig::SaveSettings() file.Set("Config", "EnableJIT", m_EnableJIT); file.Set("Config", "DumpAudio", m_DumpAudio); file.Set("Config", "Backend", sBackend); - file.Set("Config", "Frequency", iFrequency); file.Set("Config", "Volume", m_Volume); file.Save(File::GetUserPath(F_DSPCONFIG_IDX)); diff --git a/Source/Core/AudioCommon/Src/AudioCommonConfig.h b/Source/Core/AudioCommon/Src/AudioCommonConfig.h index 76c50c6408..48807b3ffd 100644 --- a/Source/Core/AudioCommon/Src/AudioCommonConfig.h +++ b/Source/Core/AudioCommon/Src/AudioCommonConfig.h @@ -37,7 +37,6 @@ struct AudioCommonConfig bool m_DumpAudio; int m_Volume; std::string sBackend; - int iFrequency; // Load from given file void Load(); diff --git a/Source/Core/AudioCommon/Src/DPL2Decoder.cpp b/Source/Core/AudioCommon/Src/DPL2Decoder.cpp new file mode 100644 index 0000000000..a5d2720878 --- /dev/null +++ b/Source/Core/AudioCommon/Src/DPL2Decoder.cpp @@ -0,0 +1,400 @@ +// Copyright (C) 2003 Dolphin Project. + +// This program is free software: you can redistribute it and/or modify +// it under the terms of the GNU General Public License as published by +// the Free Software Foundation, version 2.0. + +// This program is distributed in the hope that it will be useful, +// but WITHOUT ANY WARRANTY; without even the implied warranty of +// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +// GNU General Public License 2.0 for more details. + +// A copy of the GPL 2.0 should have been included with the program. +// If not, see http://www.gnu.org/licenses/ + +// Official SVN repository and contact information can be found at +// http://code.google.com/p/dolphin-emu/ + +// Dolby Pro Logic 2 decoder from ffdshow-tryout +// * Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au +// * Copyright (c) 2004-2006 Milan Cutka +// * based on mplayer HRTF plugin by ylai + +#include +#include +#include +#include +#include +#include "DPL2Decoder.h" + +#define M_PI 3.14159265358979323846 +#define M_SQRT1_2 0.70710678118654752440 + +int olddelay = -1; +unsigned int oldfreq = 0; +unsigned int dlbuflen; +int cyc_pos; +float l_fwr, r_fwr, lpr_fwr, lmr_fwr; +std::vector fwrbuf_l, fwrbuf_r; +float adapt_l_gain, adapt_r_gain, adapt_lpr_gain, adapt_lmr_gain; +std::vector lf, rf, lr, rr, cf, cr; +float LFE_buf[256]; +unsigned int lfe_pos; +float *filter_coefs_lfe; +unsigned int len125; + +template static _ftype_t dotproduct(int count,const T *buf,const _ftype_t *coefficients) +{ + float sum0=0,sum1=0,sum2=0,sum3=0; + for (;count>=4;buf+=4,coefficients+=4,count-=4) + { + sum0+=buf[0]*coefficients[0]; + sum1+=buf[1]*coefficients[1]; + sum2+=buf[2]*coefficients[2]; + sum3+=buf[3]*coefficients[3]; + } + while (count--) sum0+= *buf++ * *coefficients++; + return sum0+sum1+sum2+sum3; +} + +template static T firfilter(const T *buf, int pos, int len, int count, const float *coefficients) +{ + int count1, count2; + + if (pos >= count) + { + pos -= count; + count1 = count; count2 = 0; + } + else + { + count2 = pos; + count1 = count - pos; + pos = len - count1; + } + + // high part of window + const T *ptr = &buf[pos]; + + float r1=dotproduct(count1,ptr,coefficients);coefficients+=count1; + float r2=dotproduct(count2,buf,coefficients); + return T(r1+r2); +} + +template inline const T& limit(const T& val, const T& min, const T& max) +{ + if (val < min) { + return min; + } else if (val > max) { + return max; + } else { + return val; + } +} + +/* +// Hamming +// 2*pi*k +// w(k) = 0.54 - 0.46*cos(------), where 0 <= k < N +// N-1 +// +// n window length +// w buffer for the window parameters +*/ +void hamming(int n, float* w) +{ + int i; + float k = float(2*M_PI/((float)(n-1))); // 2*pi/(N-1) + + // Calculate window coefficients + for (i=0; i Fs/2 +flags window and filter type as defined in filter.h +variables are ored together: i.e. LP|HAMMING will give a +low pass filter designed using a hamming window +opt beta constant used only when designing using kaiser windows + +returns 0 if OK, -1 if fail +*/ +float* design_fir(unsigned int *n, float* fc, float opt) +{ + unsigned int o = *n & 1; // Indicator for odd filter length + unsigned int end = ((*n + 1) >> 1) - o; // Loop end + unsigned int i; // Loop index + + float k1 = 2 * float(M_PI); // 2*pi*fc1 + float k2 = 0.5f * (float)(1 - o);// Constant used if the filter has even length + float g = 0.0f; // Gain + float t1; // Temporary variables + float fc1; // Cutoff frequencies + + // Sanity check + if(*n==0) return NULL; + fc[0]=limit(fc[0],float(0.001),float(1)); + + float *w=(float*)calloc(sizeof(float),*n); + + // Get window coefficients + hamming(*n,w); + + fc1=*fc; + // Cutoff frequency must be < 0.5 where 0.5 <=> Fs/2 + fc1 = ((fc1 <= 1.0) && (fc1 > 0.0)) ? fc1/2 : 0.25f; + k1 *= fc1; + + // Low pass filter + + // If the filter length is odd, there is one point which is exactly + // in the middle. The value at this point is 2*fCutoff*sin(x)/x, + // where x is zero. To make sure nothing strange happens, we set this + // value separately. + if (o) + { + w[end] = fc1 * w[end] * 2.0f; + g=w[end]; + } + + // Create filter + for (i=0 ; i M9_03DB * lpr_fwr ? lmr_fwr : M9_03DB * lpr_fwr; + float lpr_gain = (lpr_fwr + lmr_lim_fwr) / (1 + lpr_fwr + lpr_fwr); + float lmr_gain = (lpr_fwr + lmr_lim_fwr) / (1 + lmr_lim_fwr + lmr_lim_fwr); + float lmr_unlim_gain = (lpr_fwr + lmr_fwr) / (1 + lmr_fwr + lmr_fwr); + float lpr, lmr; + float l_agc, r_agc, lpr_agc, lmr_agc; + float f, d_gain, c_gain, c_agc_cfk; + + /*** AXIS NO. 1: (Lt, Rt) -> (C, Ls, Rs) ***/ + /* AGC adaption */ + d_gain = (fabs(l_gain - *adapt_l_gain) + fabs(r_gain - *adapt_r_gain)) * 0.5f; + f = d_gain * (1.0f / MATAGCTRIG); + f = MATAGCDECAY - MATAGCDECAY / (1 + f * f); + *adapt_l_gain = (1 - f) * *adapt_l_gain + f * l_gain; + *adapt_r_gain = (1 - f) * *adapt_r_gain + f * r_gain; + /* Matrix */ + l_agc = in[il] * passive_lock(*adapt_l_gain); + r_agc = in[ir] * passive_lock(*adapt_r_gain); + cf[k] = (l_agc + r_agc) * (float)M_SQRT1_2; + if (decode_rear) + { + lr[kr] = rr[kr] = (l_agc - r_agc) * (float)M_SQRT1_2; + /* Stereo rear channel is steered with the same AGC steering as + the decoding matrix. Note this requires a fast updating AGC + at the order of 20 ms (which is the case here). */ + lr[kr] *= (l_fwr + l_fwr) / (1 + l_fwr + r_fwr); + rr[kr] *= (r_fwr + r_fwr) / (1 + l_fwr + r_fwr); + } + + /*** AXIS NO. 2: (Lt + Rt, Lt - Rt) -> (L, R) ***/ + lpr = (in[il] + in[ir]) * (float)M_SQRT1_2; + lmr = (in[il] - in[ir]) * (float)M_SQRT1_2; + /* AGC adaption */ + d_gain = fabs(lmr_unlim_gain - *adapt_lmr_gain); + f = d_gain * (1.0f / MATAGCTRIG); + f = MATAGCDECAY - MATAGCDECAY / (1 + f * f); + *adapt_lpr_gain = (1 - f) * *adapt_lpr_gain + f * lpr_gain; + *adapt_lmr_gain = (1 - f) * *adapt_lmr_gain + f * lmr_gain; + /* Matrix */ + lpr_agc = lpr * passive_lock(*adapt_lpr_gain); + lmr_agc = lmr * passive_lock(*adapt_lmr_gain); + lf[k] = (lpr_agc + lmr_agc) * (float)M_SQRT1_2; + rf[k] = (lpr_agc - lmr_agc) * (float)M_SQRT1_2; + + /*** CENTER FRONT CANCELLATION ***/ + /* A heuristic approach exploits that Lt + Rt gain contains the + information about Lt, Rt correlation. This effectively reshapes + the front and rear "cones" to concentrate Lt + Rt to C and + introduce Lt - Rt in L, R. */ + /* 0.67677 is the empirical lower bound for lpr_gain. */ + c_gain = 8 * (*adapt_lpr_gain - 0.67677f); + c_gain = c_gain > 0 ? c_gain : 0; + /* c_gain should not be too high, not even reaching full + cancellation (~ 0.50 - 0.55 at current AGC implementation), or + the center will sound too narrow. */ + c_gain = MATCOMPGAIN / (1 + c_gain * c_gain); + c_agc_cfk = c_gain * cf[k]; + lf[k] -= c_agc_cfk; + rf[k] -= c_agc_cfk; + cf[k] += c_agc_cfk + c_agc_cfk; +} + +void dpl2decode(float *samples, int numsamples, float *out) +{ + static const unsigned int FWRDURATION = 240; /* FWR average duration (samples) */ + static const int cfg_delay = 0; + static const unsigned int fmt_freq = 48000; + static const unsigned int fmt_nchannels = 2; // input channels + + int cur = 0; + + if (olddelay != cfg_delay || oldfreq != fmt_freq) + { + done(); + olddelay = cfg_delay; + oldfreq = fmt_freq; + dlbuflen = std::max(FWRDURATION, (fmt_freq * cfg_delay / 1000)); //+(len7000-1); + cyc_pos = dlbuflen - 1; + fwrbuf_l.resize(dlbuflen); + fwrbuf_r.resize(dlbuflen); + lf.resize(dlbuflen); + rf.resize(dlbuflen); + lr.resize(dlbuflen); + rr.resize(dlbuflen); + cf.resize(dlbuflen); + cr.resize(dlbuflen); + filter_coefs_lfe = calc_coefficients_125Hz_lowpass(fmt_freq); + lfe_pos = 0; + memset(LFE_buf, 0, sizeof(LFE_buf)); + } + + float *in = samples; // Input audio data + float *end = in + numsamples * fmt_nchannels; // Loop end + + while (in < end) + { + const int k = cyc_pos; + + const int fwr_pos = (k + FWRDURATION) % dlbuflen; + /* Update the full wave rectified total amplitude */ + /* Input matrix decoder */ + l_fwr += fabs(in[0]) - fabs(fwrbuf_l[fwr_pos]); + r_fwr += fabs(in[1]) - fabs(fwrbuf_r[fwr_pos]); + lpr_fwr += fabs(in[0] + in[1]) - fabs(fwrbuf_l[fwr_pos] + fwrbuf_r[fwr_pos]); + lmr_fwr += fabs(in[0] - in[1]) - fabs(fwrbuf_l[fwr_pos] - fwrbuf_r[fwr_pos]); + + /* Matrix encoded 2 channel sources */ + fwrbuf_l[k] = in[0]; + fwrbuf_r[k] = in[1]; + matrix_decode(in, k, 0, 1, true, dlbuflen, + l_fwr, r_fwr, + lpr_fwr, lmr_fwr, + &adapt_l_gain, &adapt_r_gain, + &adapt_lpr_gain, &adapt_lmr_gain, + &lf[0], &rf[0], &lr[0], &rr[0], &cf[0]); + + out[cur + 0] = lf[k]; + out[cur + 1] = rf[k]; + out[cur + 2] = cf[k]; + LFE_buf[lfe_pos] = (out[0] + out[1]) / 2; + out[cur + 3] = firfilter(LFE_buf, lfe_pos, len125, len125, filter_coefs_lfe); + lfe_pos++; + if (lfe_pos == len125) + { + lfe_pos = 0; + } + out[cur + 4] = lr[k]; + out[cur + 5] = rr[k]; + // Next sample... + in += 2; + cur += 6; + cyc_pos--; + if (cyc_pos < 0) + { + cyc_pos += dlbuflen; + } + } +} + +void dpl2reset() +{ + olddelay = -1; + oldfreq = 0; + filter_coefs_lfe = NULL; +} diff --git a/Source/Core/AudioCommon/Src/DPL2Decoder.h b/Source/Core/AudioCommon/Src/DPL2Decoder.h new file mode 100644 index 0000000000..eee8fbf15e --- /dev/null +++ b/Source/Core/AudioCommon/Src/DPL2Decoder.h @@ -0,0 +1,24 @@ +// Copyright (C) 2003 Dolphin Project. + +// This program is free software: you can redistribute it and/or modify +// it under the terms of the GNU General Public License as published by +// the Free Software Foundation, version 2.0. + +// This program is distributed in the hope that it will be useful, +// but WITHOUT ANY WARRANTY; without even the implied warranty of +// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +// GNU General Public License 2.0 for more details. + +// A copy of the GPL 2.0 should have been included with the program. +// If not, see http://www.gnu.org/licenses/ + +// Official SVN repository and contact information can be found at +// http://code.google.com/p/dolphin-emu/ + +#ifndef _DPL2DECODER_H_ +#define _DPL2DECODER_H_ + +void dpl2decode(float *samples, int numsamples, float *out); +void dpl2reset(); + +#endif // _DPL2DECODER_H_ diff --git a/Source/Core/AudioCommon/Src/Mixer.h b/Source/Core/AudioCommon/Src/Mixer.h index da456d5bf7..c1a387cf49 100644 --- a/Source/Core/AudioCommon/Src/Mixer.h +++ b/Source/Core/AudioCommon/Src/Mixer.h @@ -92,6 +92,9 @@ public: std::mutex& MixerCritical() { return m_csMixing; } + volatile float GetCurrentSpeed() const { return m_speed; } + void UpdateSpeed(volatile float val) { m_speed = val; } + protected: unsigned int m_sampleRate; unsigned int m_aiSampleRate; @@ -113,6 +116,8 @@ protected: bool m_AIplaying; std::mutex m_csMixing; + + volatile float m_speed; // Current rate of the emulation (1.0 = 100% speed) private: }; diff --git a/Source/Core/AudioCommon/Src/OpenALStream.cpp b/Source/Core/AudioCommon/Src/OpenALStream.cpp index 2abd357782..489954e351 100644 --- a/Source/Core/AudioCommon/Src/OpenALStream.cpp +++ b/Source/Core/AudioCommon/Src/OpenALStream.cpp @@ -15,13 +15,14 @@ // Official SVN repository and contact information can be found at // http://code.google.com/p/dolphin-emu/ -#include - #include "aldlist.h" #include "OpenALStream.h" +#include "DPL2Decoder.h" #if defined HAVE_OPENAL && HAVE_OPENAL +soundtouch::SoundTouch soundTouch; + // // AyuanX: Spec says OpenAL1.1 is thread safe already // @@ -35,14 +36,21 @@ bool OpenALStream::Start() pDeviceList = new ALDeviceList(); if ((pDeviceList) && (pDeviceList->GetNumDevices())) { - char *defDevName = pDeviceList-> \ - GetDeviceName(pDeviceList->GetDefaultDevice()); + char *defDevName = pDeviceList->GetDeviceName(pDeviceList->GetDefaultDevice()); + + WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName); + pDevice = alcOpenDevice(defDevName); if (pDevice) { pContext = alcCreateContext(pDevice, NULL); if (pContext) { + // Used to determine an appropriate period size (2x period = total buffer size) + //ALCint refresh; + //alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh); + //period_size_in_millisec = 1000 / refresh; + alcMakeContextCurrent(pContext); thread = std::thread(std::mem_fun(&OpenALStream::SoundLoop), this); bReturn = true; @@ -50,8 +58,7 @@ bool OpenALStream::Start() else { alcCloseDevice(pDevice); - PanicAlertT("OpenAL: can't create context " - "for device %s", defDevName); + PanicAlertT("OpenAL: can't create context for device %s", defDevName); } } else @@ -65,6 +72,10 @@ bool OpenALStream::Start() PanicAlertT("OpenAL: can't find sound devices"); } + // Initialise DPL2 parameters + dpl2reset(); + + soundTouch.clear(); return bReturn; } @@ -74,6 +85,8 @@ void OpenALStream::Stop() // kick the thread if it's waiting soundSyncEvent.Set(); + soundTouch.clear(); + thread.join(); alSourceStop(uiSource); @@ -82,7 +95,7 @@ void OpenALStream::Stop() // Clean up buffers and sources alDeleteSources(1, &uiSource); uiSource = 0; - alDeleteBuffers(OAL_NUM_BUFFERS, uiBuffers); + alDeleteBuffers(numBuffers, uiBuffers); ALCcontext *pContext = alcGetCurrentContext(); ALCdevice *pDevice = alcGetContextsDevice(pContext); @@ -111,6 +124,7 @@ void OpenALStream::Clear(bool mute) if(m_muted) { + soundTouch.clear(); alSourceStop(uiSource); } else @@ -124,20 +138,29 @@ void OpenALStream::SoundLoop() Common::SetCurrentThreadName("Audio thread - openal"); u32 ulFrequency = m_mixer->GetSampleRate(); + numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers - memset(uiBuffers, 0, OAL_NUM_BUFFERS * sizeof(ALuint)); + memset(uiBuffers, 0, numBuffers * sizeof(ALuint)); uiSource = 0; // Generate some AL Buffers for streaming - alGenBuffers(OAL_NUM_BUFFERS, (ALuint *)uiBuffers); + alGenBuffers(numBuffers, (ALuint *)uiBuffers); // Generate a Source to playback the Buffers alGenSources(1, &uiSource); // Short Silence + memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers); memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4); - for (int i = 0; i < OAL_NUM_BUFFERS; i++) - alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, OAL_MAX_SAMPLES, ulFrequency); - alSourceQueueBuffers(uiSource, OAL_NUM_BUFFERS, uiBuffers); + for (int i = 0; i < numBuffers; i++) + { +#if !defined(__APPLE__) + if (Core::g_CoreStartupParameter.bDPL2Decoder) + alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency); + else +#endif + alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency); + } + alSourceQueueBuffers(uiSource, numBuffers, uiBuffers); alSourcePlay(uiSource); // Set the default sound volume as saved in the config file. @@ -148,41 +171,154 @@ void OpenALStream::SoundLoop() ALint iBuffersFilled = 0; ALint iBuffersProcessed = 0; - ALuint uiBufferTemp[OAL_NUM_BUFFERS] = {0}; + ALint iState = 0; + ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0}; + + soundTouch.setChannels(2); + soundTouch.setSampleRate(ulFrequency); + soundTouch.setTempo(1.0); + soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0); + soundTouch.setSetting(SETTING_USE_AA_FILTER, 0); + soundTouch.setSetting(SETTING_SEQUENCE_MS, 1); + soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28); + soundTouch.setSetting(SETTING_OVERLAP_MS, 12); + + bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder; while (!threadData) { + // num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD. + const u32 stereo_16_bit_size = 4; + const u32 dma_length = 32; + const u64 ais_samples_per_second = 48000 * stereo_16_bit_size; + u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length); + u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond(); + + unsigned int numSamples = (unsigned int)num_samples_to_render; + + numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples; + numSamples = m_mixer->Mix(realtimeBuffer, numSamples); + + // Convert the samples from short to float + float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS]; + for (u32 i = 0; i < numSamples; ++i) + { + dest[i * 2 + 0] = (float)realtimeBuffer[i * 2 + 0] / (1 << 16); + dest[i * 2 + 1] = (float)realtimeBuffer[i * 2 + 1] / (1 << 16); + } + + soundTouch.putSamples(dest, numSamples); + if (iBuffersProcessed == iBuffersFilled) { alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed); iBuffersFilled = 0; } - unsigned int numSamples = m_mixer->GetNumSamples(); - - if (iBuffersProcessed && (numSamples >= OAL_THRESHOLD)) + if (iBuffersProcessed) { - numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples; - // Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued Buffer) - if (iBuffersFilled == 0) - alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp); + float rate = m_mixer->GetCurrentSpeed(); + if (rate <= 0) + { + Core::RequestRefreshInfo(); + rate = m_mixer->GetCurrentSpeed(); + } - m_mixer->Mix(realtimeBuffer, numSamples); - alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, realtimeBuffer, numSamples * 4, ulFrequency); - alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]); - iBuffersFilled++; + // Place a lower limit of 10% speed. When a game boots up, there will be + // many silence samples. These do not need to be timestretched. + if (rate > 0.10) + { + // Adjust SETTING_SEQUENCE_MS to balance between lag vs hollow audio + soundTouch.setSetting(SETTING_SEQUENCE_MS, (int)(1 / (rate * rate))); + soundTouch.setTempo(rate); + } + unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS); + if (nSamples > 0) + { + // Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued Buffer) + if (iBuffersFilled == 0) + { + alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp); + ALenum err = alGetError(); + if (err != 0) + { + ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err); + } + } +#if defined(__APPLE__) + // OSX does not have the alext AL_FORMAT_51CHN32 yet. + surround_capable = false; +#else + if (surround_capable) + { + float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS]; + dpl2decode(sampleBuffer, nSamples, dpl2); - if (iBuffersFilled == OAL_NUM_BUFFERS) - alSourcePlay(uiSource); + alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency); + ALenum err = alGetError(); + if (err == AL_INVALID_ENUM) + { + // 5.1 is not supported by the host, fallback to stereo + WARN_LOG(AUDIO, "Unable to set 5.1 surround mode. Updating OpenAL Soft might fix this issue."); + surround_capable = false; + } + else if (err != 0) + { + ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err); + } + } +#endif + if (!surround_capable) + { +#if defined(__APPLE__) + // Convert the samples from float to short + short stereo[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS]; + for (u32 i = 0; i < nSamples; ++i) + { + stereo[i * 2 + 0] = (short)((float)sampleBuffer[i * 2 + 0] * (1 << 16)); + stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16)); + } + alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, ulFrequency); +#else + alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, ulFrequency); +#endif + } + + alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]); + ALenum err = alGetError(); + if (err != 0) + { + ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err); + } + iBuffersFilled++; + + if (iBuffersFilled == numBuffers) + { + alSourcePlay(uiSource); + ALenum err = alGetError(); + if (err != 0) + { + ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err); + } + } + + alGetSourcei(uiSource, AL_SOURCE_STATE, &iState); + if (iState != AL_PLAYING) + { + // Buffer underrun occurred, resume playback + alSourcePlay(uiSource); + ALenum err = alGetError(); + if (err != 0) + { + ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err); + } + } + } } - else if (numSamples >= OAL_THRESHOLD) + else { - ALint state = 0; - alGetSourcei(uiSource, AL_SOURCE_STATE, &state); - if (state == AL_STOPPED) - alSourcePlay(uiSource); + soundSyncEvent.Wait(); } - soundSyncEvent.Wait(); } } diff --git a/Source/Core/AudioCommon/Src/OpenALStream.h b/Source/Core/AudioCommon/Src/OpenALStream.h index 2c0d75b510..a808fcefc9 100644 --- a/Source/Core/AudioCommon/Src/OpenALStream.h +++ b/Source/Core/AudioCommon/Src/OpenALStream.h @@ -24,21 +24,30 @@ #if defined HAVE_OPENAL && HAVE_OPENAL #ifdef _WIN32 -#include "../../../../Externals/OpenAL/include/al.h" -#include "../../../../Externals/OpenAL/include/alc.h" +#include +#include +#include #elif defined __APPLE__ #include #include #else #include #include +#include #endif +#include "Core.h" +#include "HW/SystemTimers.h" +#include "HW/AudioInterface.h" +#include +#include + // 16 bit Stereo #define SFX_MAX_SOURCE 1 -#define OAL_NUM_BUFFERS 16 -#define OAL_MAX_SAMPLES 512 // AyuanX: Don't make it too large, as larger buffer means longer delay -#define OAL_THRESHOLD 128 // Some games are quite sensitive to delay +#define OAL_MAX_BUFFERS 32 +#define OAL_MAX_SAMPLES 256 +#define SURROUND_CHANNELS 6 // number of channels in surround mode +#define SIZE_FLOAT 4 // size of a float in bytes #endif class OpenALStream: public SoundStream @@ -64,11 +73,14 @@ public: private: std::thread thread; Common::Event soundSyncEvent; - + short realtimeBuffer[OAL_MAX_SAMPLES * 2]; - ALuint uiBuffers[OAL_NUM_BUFFERS]; + soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS]; + ALuint uiBuffers[OAL_MAX_BUFFERS]; ALuint uiSource; ALfloat fVolume; + + u8 numBuffers; #else public: OpenALStream(CMixer *mixer, void *hWnd = NULL): SoundStream(mixer) {} diff --git a/Source/Core/Common/Src/Common.h b/Source/Core/Common/Src/Common.h index 884a1e6c16..c3cab37dd8 100644 --- a/Source/Core/Common/Src/Common.h +++ b/Source/Core/Common/Src/Common.h @@ -81,8 +81,9 @@ private: #define GC_ALIGNED16_DECL(x) __declspec(align(16)) x #define GC_ALIGNED64_DECL(x) __declspec(align(64)) x -// Since it is always around on windows +// Since they are always around on windows #define HAVE_WX 1 + #define HAVE_OPENAL 1 #define HAVE_PORTAUDIO 1 diff --git a/Source/Core/Core/Src/ConfigManager.cpp b/Source/Core/Core/Src/ConfigManager.cpp index 3417b0eb55..0d145daf6f 100644 --- a/Source/Core/Core/Src/ConfigManager.cpp +++ b/Source/Core/Core/Src/ConfigManager.cpp @@ -226,6 +226,8 @@ void SConfig::SaveSettings() ini.Set("Core", "Apploader", m_LocalCoreStartupParameter.m_strApploader); ini.Set("Core", "EnableCheats", m_LocalCoreStartupParameter.bEnableCheats); ini.Set("Core", "SelectedLanguage", m_LocalCoreStartupParameter.SelectedLanguage); + ini.Set("Core", "DPL2Decoder", m_LocalCoreStartupParameter.bDPL2Decoder); + ini.Set("Core", "Latency", m_LocalCoreStartupParameter.iLatency); ini.Set("Core", "MemcardA", m_strMemoryCardA); ini.Set("Core", "MemcardB", m_strMemoryCardB); ini.Set("Core", "SlotA", m_EXIDevice[0]); @@ -365,6 +367,8 @@ void SConfig::LoadSettings() ini.Get("Core", "Apploader", &m_LocalCoreStartupParameter.m_strApploader); ini.Get("Core", "EnableCheats", &m_LocalCoreStartupParameter.bEnableCheats, false); ini.Get("Core", "SelectedLanguage", &m_LocalCoreStartupParameter.SelectedLanguage, 0); + ini.Get("Core", "DPL2Decoder", &m_LocalCoreStartupParameter.bDPL2Decoder, false); + ini.Get("Core", "Latency", &m_LocalCoreStartupParameter.iLatency, 14); ini.Get("Core", "MemcardA", &m_strMemoryCardA); ini.Get("Core", "MemcardB", &m_strMemoryCardB); ini.Get("Core", "SlotA", (int*)&m_EXIDevice[0], EXIDEVICE_MEMORYCARD); diff --git a/Source/Core/Core/Src/Core.cpp b/Source/Core/Core/Src/Core.cpp index 141071d70a..b1269ab7c7 100644 --- a/Source/Core/Core/Src/Core.cpp +++ b/Source/Core/Core/Src/Core.cpp @@ -680,7 +680,14 @@ void VideoThrottle() SMessage; // Show message - g_video_backend->UpdateFPSDisplay(SMessage.c_str()); + g_video_backend->UpdateFPSDisplay(SMessage.c_str()); + + // Update the audio timestretcher with the current speed + if (soundStream) + { + CMixer* pMixer = soundStream->GetMixer(); + pMixer->UpdateSpeed((float)Speed / 100); + } if (_CoreParameter.bRenderToMain && SConfig::GetInstance().m_InterfaceStatusbar) { diff --git a/Source/Core/Core/Src/CoreParameter.cpp b/Source/Core/Core/Src/CoreParameter.cpp index b38c49ef7a..35f5afa57a 100644 --- a/Source/Core/Core/Src/CoreParameter.cpp +++ b/Source/Core/Core/Src/CoreParameter.cpp @@ -48,6 +48,7 @@ SCoreStartupParameter::SCoreStartupParameter() bHLE_BS2(true), bEnableCheats(false), bMergeBlocks(false), bRunCompareServer(false), bRunCompareClient(false), + bDPL2Decoder(false), iLatency(14), bMMU(false), bMMUBAT(false), iTLBHack(0), bVBeam(false), bFastDiscSpeed(false), SelectedLanguage(0), bWii(false), bDisableWiimoteSpeaker(false), @@ -82,6 +83,8 @@ void SCoreStartupParameter::LoadDefaults() bMergeBlocks = false; SelectedLanguage = 0; bWii = false; + bDPL2Decoder = false; + iLatency = 14; iPosX = 100; iPosY = 100; diff --git a/Source/Core/Core/Src/CoreParameter.h b/Source/Core/Core/Src/CoreParameter.h index 8da55e7d8a..6f5852675a 100644 --- a/Source/Core/Core/Src/CoreParameter.h +++ b/Source/Core/Core/Src/CoreParameter.h @@ -106,6 +106,9 @@ struct SCoreStartupParameter bool bEnableCheats; bool bMergeBlocks; + bool bDPL2Decoder; + int iLatency; + bool bRunCompareServer; bool bRunCompareClient; diff --git a/Source/Core/Core/Src/HW/DSPHLE/DSPHLE.cpp b/Source/Core/Core/Src/HW/DSPHLE/DSPHLE.cpp index 817610ea0f..7981c677a8 100644 --- a/Source/Core/Core/Src/HW/DSPHLE/DSPHLE.cpp +++ b/Source/Core/Core/Src/HW/DSPHLE/DSPHLE.cpp @@ -251,7 +251,7 @@ void DSPHLE::InitMixer() unsigned int AISampleRate, DACSampleRate; AudioInterface::Callback_GetSampleRate(AISampleRate, DACSampleRate); delete soundStream; - soundStream = AudioCommon::InitSoundStream(new HLEMixer(this, AISampleRate, DACSampleRate, ac_Config.iFrequency), m_hWnd); + soundStream = AudioCommon::InitSoundStream(new HLEMixer(this, AISampleRate, DACSampleRate, 48000), m_hWnd); if(!soundStream) PanicAlert("Error starting up sound stream"); // Mixer is initialized m_InitMixer = true; diff --git a/Source/Core/Core/Src/HW/DSPLLE/DSPLLE.cpp b/Source/Core/Core/Src/HW/DSPLLE/DSPLLE.cpp index 54709a9cce..0d2b375268 100644 --- a/Source/Core/Core/Src/HW/DSPLLE/DSPLLE.cpp +++ b/Source/Core/Core/Src/HW/DSPLLE/DSPLLE.cpp @@ -203,7 +203,7 @@ void DSPLLE::InitMixer() unsigned int AISampleRate, DACSampleRate; AudioInterface::Callback_GetSampleRate(AISampleRate, DACSampleRate); delete soundStream; - soundStream = AudioCommon::InitSoundStream(new CMixer(AISampleRate, DACSampleRate, ac_Config.iFrequency), m_hWnd); + soundStream = AudioCommon::InitSoundStream(new CMixer(AISampleRate, DACSampleRate, 48000), m_hWnd); if(!soundStream) PanicAlert("Error starting up sound stream"); // Mixer is initialized m_InitMixer = true; diff --git a/Source/Core/DolphinWX/Src/ConfigMain.cpp b/Source/Core/DolphinWX/Src/ConfigMain.cpp index 33a639db1c..03f3a7af11 100644 --- a/Source/Core/DolphinWX/Src/ConfigMain.cpp +++ b/Source/Core/DolphinWX/Src/ConfigMain.cpp @@ -119,7 +119,8 @@ EVT_RADIOBOX(ID_DSPENGINE, CConfigMain::AudioSettingsChanged) EVT_CHECKBOX(ID_DSPTHREAD, CConfigMain::AudioSettingsChanged) EVT_CHECKBOX(ID_ENABLE_THROTTLE, CConfigMain::AudioSettingsChanged) EVT_CHECKBOX(ID_DUMP_AUDIO, CConfigMain::AudioSettingsChanged) -EVT_CHOICE(ID_FREQUENCY, CConfigMain::AudioSettingsChanged) +EVT_CHECKBOX(ID_DPL2DECODER, CConfigMain::AudioSettingsChanged) +EVT_SLIDER(ID_LATENCY, CConfigMain::AudioSettingsChanged) EVT_CHOICE(ID_BACKEND, CConfigMain::AudioSettingsChanged) EVT_SLIDER(ID_VOLUME, CConfigMain::AudioSettingsChanged) @@ -215,6 +216,8 @@ void CConfigMain::UpdateGUI() // Disable stuff on AudioPage DSPEngine->Disable(); DSPThread->Disable(); + DPL2Decoder->Disable(); + LatencySlider->Disable(); // Disable stuff on GamecubePage GCSystemLang->Disable(); @@ -360,8 +363,11 @@ void CConfigMain::InitializeGUIValues() VolumeText->SetLabel(wxString::Format(wxT("%d %%"), ac_Config.m_Volume)); DSPThread->SetValue(startup_params.bDSPThread); DumpAudio->SetValue(ac_Config.m_DumpAudio ? true : false); - FrequencySelection->SetSelection( - FrequencySelection->FindString(wxString::Format(_("%d Hz"), ac_Config.iFrequency))); + DPL2Decoder->Enable(std::string(ac_Config.sBackend) == BACKEND_OPENAL); + DPL2Decoder->SetValue(startup_params.bDPL2Decoder); + LatencySlider->Enable(std::string(ac_Config.sBackend) == BACKEND_OPENAL); + LatencySlider->SetValue(startup_params.iLatency); + LatencyText->SetLabel(wxString::Format(wxT("%d"), startup_params.iLatency)); // add backends to the list AddAudioBackends(); @@ -503,7 +509,6 @@ void CConfigMain::InitializeGUITooltips() // Audio tooltips DSPThread->SetToolTip(_("Run DSP LLE on a dedicated thread (not recommended).")); - FrequencySelection->SetToolTip(_("Changing this will have no effect while the emulator is running!")); BackendSelection->SetToolTip(_("Changing this will have no effect while the emulator is running!")); // Gamecube - Devices @@ -511,6 +516,16 @@ void CConfigMain::InitializeGUITooltips() // Wii - Devices WiiKeyboard->SetToolTip(_("This could cause slow down in Wii Menu and some games.")); + +#if defined(__APPLE__) + DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. Not available on OSX.")); +#elif defined(__linux__) + DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. OpenAL backend only.")); +#elif defined(_WIN32) + DPL2Decoder->SetToolTip(_("Enables Dolby Pro Logic II emulation using 5.1 surround. OpenAL backend only. May need to rename soft_oal.dll to OpenAL32.dll to make it work.")); +#endif + + LatencySlider->SetToolTip(_("Sets the latency (in ms). Higher values may reduce audio crackling. OpenAL backend only.")); } void CConfigMain::CreateGUIControls() @@ -608,20 +623,23 @@ void CConfigMain::CreateGUIControls() DSPThread = new wxCheckBox(AudioPage, ID_DSPTHREAD, _("DSP LLE on Thread")); DumpAudio = new wxCheckBox(AudioPage, ID_DUMP_AUDIO, _("Dump Audio"), wxDefaultPosition, wxDefaultSize, 0, wxDefaultValidator); + DPL2Decoder = new wxCheckBox(AudioPage, ID_DPL2DECODER, _("Dolby Pro Logic II decoder")); VolumeSlider = new wxSlider(AudioPage, ID_VOLUME, 0, 1, 100, wxDefaultPosition, wxDefaultSize, wxSL_VERTICAL|wxSL_INVERSE); VolumeText = new wxStaticText(AudioPage, wxID_ANY, wxT(""), wxDefaultPosition, wxDefaultSize, 0); BackendSelection = new wxChoice(AudioPage, ID_BACKEND, wxDefaultPosition, wxDefaultSize, wxArrayBackends, 0, wxDefaultValidator, wxEmptyString); - FrequencySelection = new wxChoice(AudioPage, ID_FREQUENCY); - FrequencySelection->Append(wxString::Format(_("%d Hz"), 48000)); - FrequencySelection->Append(wxString::Format(_("%d Hz"), 32000)); + LatencySlider = new wxSlider(AudioPage, ID_LATENCY, 0, 0, 30, + wxDefaultPosition, wxDefaultSize, wxSL_HORIZONTAL); + LatencyText = new wxStaticText(AudioPage, wxID_ANY, wxT(""), + wxDefaultPosition, wxDefaultSize, 0); if (Core::GetState() != Core::CORE_UNINITIALIZED) { - FrequencySelection->Disable(); + LatencySlider->Disable(); BackendSelection->Disable(); + DPL2Decoder->Disable(); } // Create sizer and add items to dialog @@ -629,6 +647,7 @@ void CConfigMain::CreateGUIControls() sbAudioSettings->Add(DSPEngine, 0, wxALL | wxEXPAND, 5); sbAudioSettings->Add(DSPThread, 0, wxALL, 5); sbAudioSettings->Add(DumpAudio, 0, wxALL, 5); + sbAudioSettings->Add(DPL2Decoder, 0, wxALL, 5); wxStaticBoxSizer *sbVolume = new wxStaticBoxSizer(wxVERTICAL, AudioPage, _("Volume")); sbVolume->Add(VolumeSlider, 1, wxLEFT|wxRIGHT, 13); @@ -637,8 +656,9 @@ void CConfigMain::CreateGUIControls() wxGridBagSizer *sBackend = new wxGridBagSizer(); sBackend->Add(TEXT_BOX(AudioPage, _("Audio Backend:")), wxGBPosition(0, 0), wxDefaultSpan, wxALIGN_CENTER_VERTICAL|wxALL, 5); sBackend->Add(BackendSelection, wxGBPosition(0, 1), wxDefaultSpan, wxALL, 5); - sBackend->Add(TEXT_BOX(AudioPage, _("Sample Rate:")), wxGBPosition(1, 0), wxDefaultSpan, wxALIGN_CENTER_VERTICAL|wxALL, 5); - sBackend->Add(FrequencySelection, wxGBPosition(1, 1), wxDefaultSpan, wxALL, 5); + sBackend->Add(TEXT_BOX(AudioPage, _("Latency:")), wxGBPosition(1, 0), wxDefaultSpan, wxALIGN_CENTER_VERTICAL|wxALL, 5); + sBackend->Add(LatencySlider, wxGBPosition(1, 1), wxDefaultSpan, wxALL, 5); + sBackend->Add(LatencyText, wxGBPosition(1, 2), wxDefaultSpan, wxALL, 5); wxStaticBoxSizer *sbBackend = new wxStaticBoxSizer(wxHORIZONTAL, AudioPage, _("Backend Settings")); sbBackend->Add(sBackend, 0, wxEXPAND); @@ -919,19 +939,25 @@ void CConfigMain::AudioSettingsChanged(wxCommandEvent& event) SConfig::GetInstance().m_LocalCoreStartupParameter.bDSPThread = DSPThread->IsChecked(); break; + case ID_DPL2DECODER: + SConfig::GetInstance().m_LocalCoreStartupParameter.bDPL2Decoder = DPL2Decoder->IsChecked(); + break; + case ID_BACKEND: VolumeSlider->Enable(SupportsVolumeChanges(std::string(BackendSelection->GetStringSelection().mb_str()))); + LatencySlider->Enable(std::string(BackendSelection->GetStringSelection().mb_str()) == BACKEND_OPENAL); + DPL2Decoder->Enable(std::string(BackendSelection->GetStringSelection().mb_str()) == BACKEND_OPENAL); ac_Config.sBackend = BackendSelection->GetStringSelection().mb_str(); ac_Config.Update(); break; + case ID_LATENCY: + SConfig::GetInstance().m_LocalCoreStartupParameter.iLatency = LatencySlider->GetValue(); + LatencyText->SetLabel(wxString::Format(wxT("%d"), LatencySlider->GetValue())); + break; + default: ac_Config.m_DumpAudio = DumpAudio->GetValue(); - - long int frequency; - FrequencySelection->GetStringSelection().ToLong(&frequency); - ac_Config.iFrequency = frequency; - ac_Config.Update(); break; } } diff --git a/Source/Core/DolphinWX/Src/ConfigMain.h b/Source/Core/DolphinWX/Src/ConfigMain.h index d8a362d205..6f7a5b83b8 100644 --- a/Source/Core/DolphinWX/Src/ConfigMain.h +++ b/Source/Core/DolphinWX/Src/ConfigMain.h @@ -78,7 +78,8 @@ private: ID_ENABLE_HLE_AUDIO, ID_ENABLE_THROTTLE, ID_DUMP_AUDIO, - ID_FREQUENCY, + ID_DPL2DECODER, + ID_LATENCY, ID_BACKEND, ID_VOLUME, @@ -153,11 +154,13 @@ private: wxBoxSizer* sAudioPage; // GC settings wxRadioBox* DSPEngine; wxSlider* VolumeSlider; - wxStaticText* VolumeText; + wxStaticText* VolumeText; wxCheckBox* DumpAudio; + wxCheckBox* DPL2Decoder; wxArrayString wxArrayBackends; wxChoice* BackendSelection; - wxChoice* FrequencySelection; + wxSlider* LatencySlider; + wxStaticText* LatencyText; // Interface wxCheckBox* ConfirmStop; diff --git a/Source/Dolphin_2010.sln b/Source/Dolphin_2010.sln index 041f6c95bb..a65ba2e894 100644 --- a/Source/Dolphin_2010.sln +++ b/Source/Dolphin_2010.sln @@ -108,6 +108,8 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "png", "..\Externals\libpng\ EndProject Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SCMRevGen", "Core\Common\SVNRevGen.vcxproj", "{69F00340-5C3D-449F-9A80-958435C6CF06}" EndProject +Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\Externals\SoundTouch\SoundTouch.vcxproj", "{68A5DD20-7057-448B-8FE0-B6AC8D205509}" +EndProject Global GlobalSection(SolutionConfigurationPlatforms) = preSolution Debug|Win32 = Debug|Win32 @@ -382,6 +384,18 @@ Global {69F00340-5C3D-449F-9A80-958435C6CF06}.Release|Win32.Build.0 = Release|x64 {69F00340-5C3D-449F-9A80-958435C6CF06}.Release|x64.ActiveCfg = Release|x64 {69F00340-5C3D-449F-9A80-958435C6CF06}.Release|x64.Build.0 = Release|x64 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|Win32.ActiveCfg = Debug|Win32 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|Win32.Build.0 = Debug|Win32 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|x64.ActiveCfg = Debug|x64 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Debug|x64.Build.0 = Debug|x64 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|Win32.ActiveCfg = Debug|Win32 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|Win32.Build.0 = Debug|Win32 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|x64.ActiveCfg = Release|x64 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.DebugFast|x64.Build.0 = Release|x64 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.ActiveCfg = Release|Win32 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|Win32.Build.0 = Release|Win32 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.ActiveCfg = Release|x64 + {68A5DD20-7057-448B-8FE0-B6AC8D205509}.Release|x64.Build.0 = Release|x64 EndGlobalSection GlobalSection(SolutionProperties) = preSolution HideSolutionNode = FALSE