656 changes: 656 additions & 0 deletions Externals/OpenAL/include/al.h
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237 changes: 237 additions & 0 deletions Externals/OpenAL/include/alc.h
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#ifndef AL_ALC_H
#define AL_ALC_H

#if defined(__cplusplus)
extern "C" {
#endif

#ifndef ALC_API
#if defined(AL_LIBTYPE_STATIC)
#define ALC_API
#elif defined(_WIN32)
#define ALC_API __declspec(dllimport)
#else
#define ALC_API extern
#endif
#endif

#if defined(_WIN32)
#define ALC_APIENTRY __cdecl
#else
#define ALC_APIENTRY
#endif


/** Deprecated macro. */
#define ALCAPI ALC_API
#define ALCAPIENTRY ALC_APIENTRY
#define ALC_INVALID 0

/** Supported ALC version? */
#define ALC_VERSION_0_1 1

/** Opaque device handle */
typedef struct ALCdevice_struct ALCdevice;
/** Opaque context handle */
typedef struct ALCcontext_struct ALCcontext;

/** 8-bit boolean */
typedef char ALCboolean;

/** character */
typedef char ALCchar;

/** signed 8-bit 2's complement integer */
typedef signed char ALCbyte;

/** unsigned 8-bit integer */
typedef unsigned char ALCubyte;

/** signed 16-bit 2's complement integer */
typedef short ALCshort;

/** unsigned 16-bit integer */
typedef unsigned short ALCushort;

/** signed 32-bit 2's complement integer */
typedef int ALCint;

/** unsigned 32-bit integer */
typedef unsigned int ALCuint;

/** non-negative 32-bit binary integer size */
typedef int ALCsizei;

/** enumerated 32-bit value */
typedef int ALCenum;

/** 32-bit IEEE754 floating-point */
typedef float ALCfloat;

/** 64-bit IEEE754 floating-point */
typedef double ALCdouble;

/** void type (for opaque pointers only) */
typedef void ALCvoid;


/* Enumerant values begin at column 50. No tabs. */

/** Boolean False. */
#define ALC_FALSE 0

/** Boolean True. */
#define ALC_TRUE 1

/** Context attribute: <int> Hz. */
#define ALC_FREQUENCY 0x1007

/** Context attribute: <int> Hz. */
#define ALC_REFRESH 0x1008

/** Context attribute: AL_TRUE or AL_FALSE. */
#define ALC_SYNC 0x1009

/** Context attribute: <int> requested Mono (3D) Sources. */
#define ALC_MONO_SOURCES 0x1010

/** Context attribute: <int> requested Stereo Sources. */
#define ALC_STEREO_SOURCES 0x1011

/** No error. */
#define ALC_NO_ERROR 0

/** Invalid device handle. */
#define ALC_INVALID_DEVICE 0xA001

/** Invalid context handle. */
#define ALC_INVALID_CONTEXT 0xA002

/** Invalid enum parameter passed to an ALC call. */
#define ALC_INVALID_ENUM 0xA003

/** Invalid value parameter passed to an ALC call. */
#define ALC_INVALID_VALUE 0xA004

/** Out of memory. */
#define ALC_OUT_OF_MEMORY 0xA005


/** Runtime ALC version. */
#define ALC_MAJOR_VERSION 0x1000
#define ALC_MINOR_VERSION 0x1001

/** Context attribute list properties. */
#define ALC_ATTRIBUTES_SIZE 0x1002
#define ALC_ALL_ATTRIBUTES 0x1003

/** String for the default device specifier. */
#define ALC_DEFAULT_DEVICE_SPECIFIER 0x1004
/**
* String for the given device's specifier.
*
* If device handle is NULL, it is instead a null-char separated list of
* strings of known device specifiers (list ends with an empty string).
*/
#define ALC_DEVICE_SPECIFIER 0x1005
/** String for space-separated list of ALC extensions. */
#define ALC_EXTENSIONS 0x1006


/** Capture extension */
#define ALC_EXT_CAPTURE 1
/**
* String for the given capture device's specifier.
*
* If device handle is NULL, it is instead a null-char separated list of
* strings of known capture device specifiers (list ends with an empty string).
*/
#define ALC_CAPTURE_DEVICE_SPECIFIER 0x310
/** String for the default capture device specifier. */
#define ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER 0x311
/** Number of sample frames available for capture. */
#define ALC_CAPTURE_SAMPLES 0x312


/** Enumerate All extension */
#define ALC_ENUMERATE_ALL_EXT 1
/** String for the default extended device specifier. */
#define ALC_DEFAULT_ALL_DEVICES_SPECIFIER 0x1012
/**
* String for the given extended device's specifier.
*
* If device handle is NULL, it is instead a null-char separated list of
* strings of known extended device specifiers (list ends with an empty string).
*/
#define ALC_ALL_DEVICES_SPECIFIER 0x1013


/** Context management. */
ALC_API ALCcontext* ALC_APIENTRY alcCreateContext(ALCdevice *device, const ALCint* attrlist);
ALC_API ALCboolean ALC_APIENTRY alcMakeContextCurrent(ALCcontext *context);
ALC_API void ALC_APIENTRY alcProcessContext(ALCcontext *context);
ALC_API void ALC_APIENTRY alcSuspendContext(ALCcontext *context);
ALC_API void ALC_APIENTRY alcDestroyContext(ALCcontext *context);
ALC_API ALCcontext* ALC_APIENTRY alcGetCurrentContext(void);
ALC_API ALCdevice* ALC_APIENTRY alcGetContextsDevice(ALCcontext *context);

/** Device management. */
ALC_API ALCdevice* ALC_APIENTRY alcOpenDevice(const ALCchar *devicename);
ALC_API ALCboolean ALC_APIENTRY alcCloseDevice(ALCdevice *device);


/**
* Error support.
*
* Obtain the most recent Device error.
*/
ALC_API ALCenum ALC_APIENTRY alcGetError(ALCdevice *device);

/**
* Extension support.
*
* Query for the presence of an extension, and obtain any appropriate
* function pointers and enum values.
*/
ALC_API ALCboolean ALC_APIENTRY alcIsExtensionPresent(ALCdevice *device, const ALCchar *extname);
ALC_API void* ALC_APIENTRY alcGetProcAddress(ALCdevice *device, const ALCchar *funcname);
ALC_API ALCenum ALC_APIENTRY alcGetEnumValue(ALCdevice *device, const ALCchar *enumname);

/** Query function. */
ALC_API const ALCchar* ALC_APIENTRY alcGetString(ALCdevice *device, ALCenum param);
ALC_API void ALC_APIENTRY alcGetIntegerv(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values);

/** Capture function. */
ALC_API ALCdevice* ALC_APIENTRY alcCaptureOpenDevice(const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize);
ALC_API ALCboolean ALC_APIENTRY alcCaptureCloseDevice(ALCdevice *device);
ALC_API void ALC_APIENTRY alcCaptureStart(ALCdevice *device);
ALC_API void ALC_APIENTRY alcCaptureStop(ALCdevice *device);
ALC_API void ALC_APIENTRY alcCaptureSamples(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);

/** Pointer-to-function type, useful for dynamically getting ALC entry points. */
typedef ALCcontext* (ALC_APIENTRY *LPALCCREATECONTEXT)(ALCdevice *device, const ALCint *attrlist);
typedef ALCboolean (ALC_APIENTRY *LPALCMAKECONTEXTCURRENT)(ALCcontext *context);
typedef void (ALC_APIENTRY *LPALCPROCESSCONTEXT)(ALCcontext *context);
typedef void (ALC_APIENTRY *LPALCSUSPENDCONTEXT)(ALCcontext *context);
typedef void (ALC_APIENTRY *LPALCDESTROYCONTEXT)(ALCcontext *context);
typedef ALCcontext* (ALC_APIENTRY *LPALCGETCURRENTCONTEXT)(void);
typedef ALCdevice* (ALC_APIENTRY *LPALCGETCONTEXTSDEVICE)(ALCcontext *context);
typedef ALCdevice* (ALC_APIENTRY *LPALCOPENDEVICE)(const ALCchar *devicename);
typedef ALCboolean (ALC_APIENTRY *LPALCCLOSEDEVICE)(ALCdevice *device);
typedef ALCenum (ALC_APIENTRY *LPALCGETERROR)(ALCdevice *device);
typedef ALCboolean (ALC_APIENTRY *LPALCISEXTENSIONPRESENT)(ALCdevice *device, const ALCchar *extname);
typedef void* (ALC_APIENTRY *LPALCGETPROCADDRESS)(ALCdevice *device, const ALCchar *funcname);
typedef ALCenum (ALC_APIENTRY *LPALCGETENUMVALUE)(ALCdevice *device, const ALCchar *enumname);
typedef const ALCchar* (ALC_APIENTRY *LPALCGETSTRING)(ALCdevice *device, ALCenum param);
typedef void (ALC_APIENTRY *LPALCGETINTEGERV)(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values);
typedef ALCdevice* (ALC_APIENTRY *LPALCCAPTUREOPENDEVICE)(const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize);
typedef ALCboolean (ALC_APIENTRY *LPALCCAPTURECLOSEDEVICE)(ALCdevice *device);
typedef void (ALC_APIENTRY *LPALCCAPTURESTART)(ALCdevice *device);
typedef void (ALC_APIENTRY *LPALCCAPTURESTOP)(ALCdevice *device);
typedef void (ALC_APIENTRY *LPALCCAPTURESAMPLES)(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);

#if defined(__cplusplus)
}
#endif

#endif /* AL_ALC_H */
355 changes: 355 additions & 0 deletions Externals/OpenAL/include/alext.h
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/**
* OpenAL cross platform audio library
* Copyright (C) 2008 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/

#ifndef AL_ALEXT_H
#define AL_ALEXT_H

#include <stddef.h>
/* Define int64_t and uint64_t types */
#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
#include <inttypes.h>
#elif defined(_WIN32) && defined(__GNUC__)
#include <stdint.h>
#elif defined(_WIN32)
typedef __int64 int64_t;
typedef unsigned __int64 uint64_t;
#else
/* Fallback if nothing above works */
#include <inttypes.h>
#endif

#include "alc.h"
#include "al.h"

#ifdef __cplusplus
extern "C" {
#endif

#ifndef AL_LOKI_IMA_ADPCM_format
#define AL_LOKI_IMA_ADPCM_format 1
#define AL_FORMAT_IMA_ADPCM_MONO16_EXT 0x10000
#define AL_FORMAT_IMA_ADPCM_STEREO16_EXT 0x10001
#endif

#ifndef AL_LOKI_WAVE_format
#define AL_LOKI_WAVE_format 1
#define AL_FORMAT_WAVE_EXT 0x10002
#endif

#ifndef AL_EXT_vorbis
#define AL_EXT_vorbis 1
#define AL_FORMAT_VORBIS_EXT 0x10003
#endif

#ifndef AL_LOKI_quadriphonic
#define AL_LOKI_quadriphonic 1
#define AL_FORMAT_QUAD8_LOKI 0x10004
#define AL_FORMAT_QUAD16_LOKI 0x10005
#endif

#ifndef AL_EXT_float32
#define AL_EXT_float32 1
#define AL_FORMAT_MONO_FLOAT32 0x10010
#define AL_FORMAT_STEREO_FLOAT32 0x10011
#endif

#ifndef AL_EXT_double
#define AL_EXT_double 1
#define AL_FORMAT_MONO_DOUBLE_EXT 0x10012
#define AL_FORMAT_STEREO_DOUBLE_EXT 0x10013
#endif

#ifndef AL_EXT_MULAW
#define AL_EXT_MULAW 1
#define AL_FORMAT_MONO_MULAW_EXT 0x10014
#define AL_FORMAT_STEREO_MULAW_EXT 0x10015
#endif

#ifndef AL_EXT_ALAW
#define AL_EXT_ALAW 1
#define AL_FORMAT_MONO_ALAW_EXT 0x10016
#define AL_FORMAT_STEREO_ALAW_EXT 0x10017
#endif

#ifndef ALC_LOKI_audio_channel
#define ALC_LOKI_audio_channel 1
#define ALC_CHAN_MAIN_LOKI 0x500001
#define ALC_CHAN_PCM_LOKI 0x500002
#define ALC_CHAN_CD_LOKI 0x500003
#endif

#ifndef AL_EXT_MCFORMATS
#define AL_EXT_MCFORMATS 1
#define AL_FORMAT_QUAD8 0x1204
#define AL_FORMAT_QUAD16 0x1205
#define AL_FORMAT_QUAD32 0x1206
#define AL_FORMAT_REAR8 0x1207
#define AL_FORMAT_REAR16 0x1208
#define AL_FORMAT_REAR32 0x1209
#define AL_FORMAT_51CHN8 0x120A
#define AL_FORMAT_51CHN16 0x120B
#define AL_FORMAT_51CHN32 0x120C
#define AL_FORMAT_61CHN8 0x120D
#define AL_FORMAT_61CHN16 0x120E
#define AL_FORMAT_61CHN32 0x120F
#define AL_FORMAT_71CHN8 0x1210
#define AL_FORMAT_71CHN16 0x1211
#define AL_FORMAT_71CHN32 0x1212
#endif

#ifndef AL_EXT_MULAW_MCFORMATS
#define AL_EXT_MULAW_MCFORMATS 1
#define AL_FORMAT_MONO_MULAW 0x10014
#define AL_FORMAT_STEREO_MULAW 0x10015
#define AL_FORMAT_QUAD_MULAW 0x10021
#define AL_FORMAT_REAR_MULAW 0x10022
#define AL_FORMAT_51CHN_MULAW 0x10023
#define AL_FORMAT_61CHN_MULAW 0x10024
#define AL_FORMAT_71CHN_MULAW 0x10025
#endif

#ifndef AL_EXT_IMA4
#define AL_EXT_IMA4 1
#define AL_FORMAT_MONO_IMA4 0x1300
#define AL_FORMAT_STEREO_IMA4 0x1301
#endif

#ifndef AL_EXT_STATIC_BUFFER
#define AL_EXT_STATIC_BUFFER 1
typedef ALvoid (AL_APIENTRY*PFNALBUFFERDATASTATICPROC)(const ALint,ALenum,ALvoid*,ALsizei,ALsizei);
#ifdef AL_ALEXT_PROTOTYPES
AL_API ALvoid AL_APIENTRY alBufferDataStatic(const ALint buffer, ALenum format, ALvoid *data, ALsizei len, ALsizei freq);
#endif
#endif

#ifndef ALC_EXT_EFX
#define ALC_EXT_EFX 1
#include "efx.h"
#endif

#ifndef ALC_EXT_disconnect
#define ALC_EXT_disconnect 1
#define ALC_CONNECTED 0x313
#endif

#ifndef ALC_EXT_thread_local_context
#define ALC_EXT_thread_local_context 1
typedef ALCboolean (ALC_APIENTRY*PFNALCSETTHREADCONTEXTPROC)(ALCcontext *context);
typedef ALCcontext* (ALC_APIENTRY*PFNALCGETTHREADCONTEXTPROC)(void);
#ifdef AL_ALEXT_PROTOTYPES
ALC_API ALCboolean ALC_APIENTRY alcSetThreadContext(ALCcontext *context);
ALC_API ALCcontext* ALC_APIENTRY alcGetThreadContext(void);
#endif
#endif

#ifndef AL_EXT_source_distance_model
#define AL_EXT_source_distance_model 1
#define AL_SOURCE_DISTANCE_MODEL 0x200
#endif

#ifndef AL_SOFT_buffer_sub_data
#define AL_SOFT_buffer_sub_data 1
#define AL_BYTE_RW_OFFSETS_SOFT 0x1031
#define AL_SAMPLE_RW_OFFSETS_SOFT 0x1032
typedef ALvoid (AL_APIENTRY*PFNALBUFFERSUBDATASOFTPROC)(ALuint,ALenum,const ALvoid*,ALsizei,ALsizei);
#ifdef AL_ALEXT_PROTOTYPES
AL_API ALvoid AL_APIENTRY alBufferSubDataSOFT(ALuint buffer,ALenum format,const ALvoid *data,ALsizei offset,ALsizei length);
#endif
#endif

#ifndef AL_SOFT_loop_points
#define AL_SOFT_loop_points 1
#define AL_LOOP_POINTS_SOFT 0x2015
#endif

#ifndef AL_EXT_FOLDBACK
#define AL_EXT_FOLDBACK 1
#define AL_EXT_FOLDBACK_NAME "AL_EXT_FOLDBACK"
#define AL_FOLDBACK_EVENT_BLOCK 0x4112
#define AL_FOLDBACK_EVENT_START 0x4111
#define AL_FOLDBACK_EVENT_STOP 0x4113
#define AL_FOLDBACK_MODE_MONO 0x4101
#define AL_FOLDBACK_MODE_STEREO 0x4102
typedef void (AL_APIENTRY*LPALFOLDBACKCALLBACK)(ALenum,ALsizei);
typedef void (AL_APIENTRY*LPALREQUESTFOLDBACKSTART)(ALenum,ALsizei,ALsizei,ALfloat*,LPALFOLDBACKCALLBACK);
typedef void (AL_APIENTRY*LPALREQUESTFOLDBACKSTOP)(void);
#ifdef AL_ALEXT_PROTOTYPES
AL_API void AL_APIENTRY alRequestFoldbackStart(ALenum mode,ALsizei count,ALsizei length,ALfloat *mem,LPALFOLDBACKCALLBACK callback);
AL_API void AL_APIENTRY alRequestFoldbackStop(void);
#endif
#endif

#ifndef ALC_EXT_DEDICATED
#define ALC_EXT_DEDICATED 1
#define AL_DEDICATED_GAIN 0x0001
#define AL_EFFECT_DEDICATED_DIALOGUE 0x9001
#define AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT 0x9000
#endif

#ifndef AL_SOFT_buffer_samples
#define AL_SOFT_buffer_samples 1
/* Channel configurations */
#define AL_MONO_SOFT 0x1500
#define AL_STEREO_SOFT 0x1501
#define AL_REAR_SOFT 0x1502
#define AL_QUAD_SOFT 0x1503
#define AL_5POINT1_SOFT 0x1504
#define AL_6POINT1_SOFT 0x1505
#define AL_7POINT1_SOFT 0x1506

/* Sample types */
#define AL_BYTE_SOFT 0x1400
#define AL_UNSIGNED_BYTE_SOFT 0x1401
#define AL_SHORT_SOFT 0x1402
#define AL_UNSIGNED_SHORT_SOFT 0x1403
#define AL_INT_SOFT 0x1404
#define AL_UNSIGNED_INT_SOFT 0x1405
#define AL_FLOAT_SOFT 0x1406
#define AL_DOUBLE_SOFT 0x1407
#define AL_BYTE3_SOFT 0x1408
#define AL_UNSIGNED_BYTE3_SOFT 0x1409

/* Storage formats */
#define AL_MONO8_SOFT 0x1100
#define AL_MONO16_SOFT 0x1101
#define AL_MONO32F_SOFT 0x10010
#define AL_STEREO8_SOFT 0x1102
#define AL_STEREO16_SOFT 0x1103
#define AL_STEREO32F_SOFT 0x10011
#define AL_QUAD8_SOFT 0x1204
#define AL_QUAD16_SOFT 0x1205
#define AL_QUAD32F_SOFT 0x1206
#define AL_REAR8_SOFT 0x1207
#define AL_REAR16_SOFT 0x1208
#define AL_REAR32F_SOFT 0x1209
#define AL_5POINT1_8_SOFT 0x120A
#define AL_5POINT1_16_SOFT 0x120B
#define AL_5POINT1_32F_SOFT 0x120C
#define AL_6POINT1_8_SOFT 0x120D
#define AL_6POINT1_16_SOFT 0x120E
#define AL_6POINT1_32F_SOFT 0x120F
#define AL_7POINT1_8_SOFT 0x1210
#define AL_7POINT1_16_SOFT 0x1211
#define AL_7POINT1_32F_SOFT 0x1212

/* Buffer attributes */
#define AL_INTERNAL_FORMAT_SOFT 0x2008
#define AL_BYTE_LENGTH_SOFT 0x2009
#define AL_SAMPLE_LENGTH_SOFT 0x200A
#define AL_SEC_LENGTH_SOFT 0x200B

typedef void (AL_APIENTRY*LPALBUFFERSAMPLESSOFT)(ALuint,ALuint,ALenum,ALsizei,ALenum,ALenum,const ALvoid*);
typedef void (AL_APIENTRY*LPALBUFFERSUBSAMPLESSOFT)(ALuint,ALsizei,ALsizei,ALenum,ALenum,const ALvoid*);
typedef void (AL_APIENTRY*LPALGETBUFFERSAMPLESSOFT)(ALuint,ALsizei,ALsizei,ALenum,ALenum,ALvoid*);
typedef ALboolean (AL_APIENTRY*LPALISBUFFERFORMATSUPPORTEDSOFT)(ALenum);
#ifdef AL_ALEXT_PROTOTYPES
AL_API void AL_APIENTRY alBufferSamplesSOFT(ALuint buffer, ALuint samplerate, ALenum internalformat, ALsizei samples, ALenum channels, ALenum type, const ALvoid *data);
AL_API void AL_APIENTRY alBufferSubSamplesSOFT(ALuint buffer, ALsizei offset, ALsizei samples, ALenum channels, ALenum type, const ALvoid *data);
AL_API void AL_APIENTRY alGetBufferSamplesSOFT(ALuint buffer, ALsizei offset, ALsizei samples, ALenum channels, ALenum type, ALvoid *data);
AL_API ALboolean AL_APIENTRY alIsBufferFormatSupportedSOFT(ALenum format);
#endif
#endif

#ifndef AL_SOFT_direct_channels
#define AL_SOFT_direct_channels 1
#define AL_DIRECT_CHANNELS_SOFT 0x1033
#endif

#ifndef ALC_SOFT_loopback
#define ALC_SOFT_loopback 1
#define ALC_FORMAT_CHANNELS_SOFT 0x1990
#define ALC_FORMAT_TYPE_SOFT 0x1991

/* Sample types */
#define ALC_BYTE_SOFT 0x1400
#define ALC_UNSIGNED_BYTE_SOFT 0x1401
#define ALC_SHORT_SOFT 0x1402
#define ALC_UNSIGNED_SHORT_SOFT 0x1403
#define ALC_INT_SOFT 0x1404
#define ALC_UNSIGNED_INT_SOFT 0x1405
#define ALC_FLOAT_SOFT 0x1406

/* Channel configurations */
#define ALC_MONO_SOFT 0x1500
#define ALC_STEREO_SOFT 0x1501
#define ALC_QUAD_SOFT 0x1503
#define ALC_5POINT1_SOFT 0x1504
#define ALC_6POINT1_SOFT 0x1505
#define ALC_7POINT1_SOFT 0x1506

typedef ALCdevice* (ALC_APIENTRY*LPALCLOOPBACKOPENDEVICESOFT)(const ALCchar*);
typedef ALCboolean (ALC_APIENTRY*LPALCISRENDERFORMATSUPPORTEDSOFT)(ALCdevice*,ALCsizei,ALCenum,ALCenum);
typedef void (ALC_APIENTRY*LPALCRENDERSAMPLESSOFT)(ALCdevice*,ALCvoid*,ALCsizei);
#ifdef AL_ALEXT_PROTOTYPES
ALC_API ALCdevice* ALC_APIENTRY alcLoopbackOpenDeviceSOFT(const ALCchar *deviceName);
ALC_API ALCboolean ALC_APIENTRY alcIsRenderFormatSupportedSOFT(ALCdevice *device, ALCsizei freq, ALCenum channels, ALCenum type);
ALC_API void ALC_APIENTRY alcRenderSamplesSOFT(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);
#endif
#endif

#ifndef AL_EXT_STEREO_ANGLES
#define AL_EXT_STEREO_ANGLES 1
#define AL_STEREO_ANGLES 0x1030
#endif

#ifndef AL_EXT_SOURCE_RADIUS
#define AL_EXT_SOURCE_RADIUS 1
#define AL_SOURCE_RADIUS 0x1031
#endif

#ifndef AL_SOFT_source_latency
#define AL_SOFT_source_latency 1
#define AL_SAMPLE_OFFSET_LATENCY_SOFT 0x1200
#define AL_SEC_OFFSET_LATENCY_SOFT 0x1201
typedef int64_t ALint64SOFT;
typedef uint64_t ALuint64SOFT;
typedef void (AL_APIENTRY*LPALSOURCEDSOFT)(ALuint,ALenum,ALdouble);
typedef void (AL_APIENTRY*LPALSOURCE3DSOFT)(ALuint,ALenum,ALdouble,ALdouble,ALdouble);
typedef void (AL_APIENTRY*LPALSOURCEDVSOFT)(ALuint,ALenum,const ALdouble*);
typedef void (AL_APIENTRY*LPALGETSOURCEDSOFT)(ALuint,ALenum,ALdouble*);
typedef void (AL_APIENTRY*LPALGETSOURCE3DSOFT)(ALuint,ALenum,ALdouble*,ALdouble*,ALdouble*);
typedef void (AL_APIENTRY*LPALGETSOURCEDVSOFT)(ALuint,ALenum,ALdouble*);
typedef void (AL_APIENTRY*LPALSOURCEI64SOFT)(ALuint,ALenum,ALint64SOFT);
typedef void (AL_APIENTRY*LPALSOURCE3I64SOFT)(ALuint,ALenum,ALint64SOFT,ALint64SOFT,ALint64SOFT);
typedef void (AL_APIENTRY*LPALSOURCEI64VSOFT)(ALuint,ALenum,const ALint64SOFT*);
typedef void (AL_APIENTRY*LPALGETSOURCEI64SOFT)(ALuint,ALenum,ALint64SOFT*);
typedef void (AL_APIENTRY*LPALGETSOURCE3I64SOFT)(ALuint,ALenum,ALint64SOFT*,ALint64SOFT*,ALint64SOFT*);
typedef void (AL_APIENTRY*LPALGETSOURCEI64VSOFT)(ALuint,ALenum,ALint64SOFT*);
#ifdef AL_ALEXT_PROTOTYPES
AL_API void AL_APIENTRY alSourcedSOFT(ALuint source, ALenum param, ALdouble value);
AL_API void AL_APIENTRY alSource3dSOFT(ALuint source, ALenum param, ALdouble value1, ALdouble value2, ALdouble value3);
AL_API void AL_APIENTRY alSourcedvSOFT(ALuint source, ALenum param, const ALdouble *values);
AL_API void AL_APIENTRY alGetSourcedSOFT(ALuint source, ALenum param, ALdouble *value);
AL_API void AL_APIENTRY alGetSource3dSOFT(ALuint source, ALenum param, ALdouble *value1, ALdouble *value2, ALdouble *value3);
AL_API void AL_APIENTRY alGetSourcedvSOFT(ALuint source, ALenum param, ALdouble *values);
AL_API void AL_APIENTRY alSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT value);
AL_API void AL_APIENTRY alSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT value1, ALint64SOFT value2, ALint64SOFT value3);
AL_API void AL_APIENTRY alSourcei64vSOFT(ALuint source, ALenum param, const ALint64SOFT *values);
AL_API void AL_APIENTRY alGetSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT *value);
AL_API void AL_APIENTRY alGetSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT *value1, ALint64SOFT *value2, ALint64SOFT *value3);
AL_API void AL_APIENTRY alGetSourcei64vSOFT(ALuint source, ALenum param, ALint64SOFT *values);
#endif
#endif

#ifdef __cplusplus
}
#endif

#endif
3 changes: 3 additions & 0 deletions Externals/OpenAL/include/efx-creative.h
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@@ -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. */
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94 changes: 94 additions & 0 deletions Externals/OpenAL/include/xram.h
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#include <al.h>

// X-RAM Function pointer definitions
typedef ALboolean (__cdecl *EAXSetBufferMode)(ALsizei n, ALuint *buffers, ALint value);
typedef ALenum (__cdecl *EAXGetBufferMode)(ALuint buffer, ALint *value);

//////////////////////////////////////////////////////////////////////////////
// Query for X-RAM extension
//
// if (alIsExtensionPresent("EAX-RAM") == AL_TRUE)
// X-RAM Extension found
//
//////////////////////////////////////////////////////////////////////////////


//////////////////////////////////////////////////////////////////////////////
// X-RAM enum names
//
// "AL_EAX_RAM_SIZE"
// "AL_EAX_RAM_FREE"
// "AL_STORAGE_AUTOMATIC"
// "AL_STORAGE_HARDWARE"
// "AL_STORAGE_ACCESSIBLE"
//
// Query enum values using alGetEnumValue, for example
//
// long lRamSizeEnum = alGetEnumValue("AL_EAX_RAM_SIZE")
//
//////////////////////////////////////////////////////////////////////////////


//////////////////////////////////////////////////////////////////////////////
// Query total amount of X-RAM
//
// long lTotalSize = alGetInteger(alGetEnumValue("AL_EAX_RAM_SIZE")
//
//////////////////////////////////////////////////////////////////////////////


//////////////////////////////////////////////////////////////////////////////
// Query free X-RAM available
//
// long lFreeSize = alGetInteger(alGetEnumValue("AL_EAX_RAM_FREE")
//
//////////////////////////////////////////////////////////////////////////////


//////////////////////////////////////////////////////////////////////////////
// Query X-RAM Function pointers
//
// Use typedefs defined above to get the X-RAM function pointers using
// alGetProcAddress
//
// EAXSetBufferMode eaxSetBufferMode;
// EAXGetBufferMode eaxGetBufferMode;
//
// eaxSetBufferMode = (EAXSetBufferMode)alGetProcAddress("EAXSetBufferMode");
// eaxGetBufferMode = (EAXGetBufferMode)alGetProcAddress("EAXGetBufferMode");
//
//////////////////////////////////////////////////////////////////////////////


//////////////////////////////////////////////////////////////////////////////
// Force an Open AL Buffer into X-RAM (good for non-streaming buffers)
//
// ALuint uiBuffer;
// alGenBuffers(1, &uiBuffer);
// eaxSetBufferMode(1, &uiBuffer, alGetEnumValue("AL_STORAGE_HARDWARE"));
// alBufferData(...);
//
//////////////////////////////////////////////////////////////////////////////


//////////////////////////////////////////////////////////////////////////////
// Force an Open AL Buffer into 'accessible' (currently host) RAM (good for streaming buffers)
//
// ALuint uiBuffer;
// alGenBuffers(1, &uiBuffer);
// eaxSetBufferMode(1, &uiBuffer, alGetEnumValue("AL_STORAGE_ACCESSIBLE"));
// alBufferData(...);
//
//////////////////////////////////////////////////////////////////////////////


//////////////////////////////////////////////////////////////////////////////
// Put an Open AL Buffer into X-RAM if memory is available, otherwise use
// host RAM. This is the default mode.
//
// ALuint uiBuffer;
// alGenBuffers(1, &uiBuffer);
// eaxSetBufferMode(1, &uiBuffer, alGetEnumValue("AL_STORAGE_AUTOMATIC"));
// alBufferData(...);
//
//////////////////////////////////////////////////////////////////////////////
184 changes: 184 additions & 0 deletions Externals/soundtouch/AAFilter.cpp
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////////////////////////////////////////////////////////////////////////////////
///
/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
/// MMX optimization.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2009-01-11 13:34:24 +0200 (Sun, 11 Jan 2009) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "AAFilter.h"
#include "FIRFilter.h"

using namespace soundtouch;

#define PI 3.141592655357989
#define TWOPI (2 * PI)

/*****************************************************************************
*
* Implementation of the class 'AAFilter'
*
*****************************************************************************/

AAFilter::AAFilter(uint len)
{
pFIR = FIRFilter::newInstance();
cutoffFreq = 0.5;
setLength(len);
}



AAFilter::~AAFilter()
{
delete pFIR;
}



// Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency.
void AAFilter::setCutoffFreq(double newCutoffFreq)
{
cutoffFreq = newCutoffFreq;
calculateCoeffs();
}



// Sets number of FIR filter taps
void AAFilter::setLength(uint newLength)
{
length = newLength;
calculateCoeffs();
}



// Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs()
{
uint i;
double cntTemp, temp, tempCoeff,h, w;
double fc2, wc;
double scaleCoeff, sum;
double *work;
SAMPLETYPE *coeffs;

assert(length >= 2);
assert(length % 4 == 0);
assert(cutoffFreq >= 0);
assert(cutoffFreq <= 0.5);

work = new double[length];
coeffs = new SAMPLETYPE[length];

fc2 = 2.0 * cutoffFreq;
wc = PI * fc2;
tempCoeff = TWOPI / (double)length;

sum = 0;
for (i = 0; i < length; i ++)
{
cntTemp = (double)i - (double)(length / 2);

temp = cntTemp * wc;
if (temp != 0)
{
h = fc2 * sin(temp) / temp; // sinc function
}
else
{
h = 1.0;
}
w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window

temp = w * h;
work[i] = temp;

// calc net sum of coefficients
sum += temp;
}

// ensure the sum of coefficients is larger than zero
assert(sum > 0);

// ensure we've really designed a lowpass filter...
assert(work[length/2] > 0);
assert(work[length/2 + 1] > -1e-6);
assert(work[length/2 - 1] > -1e-6);

// Calculate a scaling coefficient in such a way that the result can be
// divided by 16384
scaleCoeff = 16384.0f / sum;

for (i = 0; i < length; i ++)
{
// scale & round to nearest integer
temp = work[i] * scaleCoeff;
temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods
assert(temp >= -32768 && temp <= 32767);
coeffs[i] = (SAMPLETYPE)temp;
}

// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
pFIR->setCoefficients(coeffs, length, 14);

delete[] work;
delete[] coeffs;
}


// Applies the filter to the given sequence of samples.
// Note : The amount of outputted samples is by value of 'filter length'
// smaller than the amount of input samples.
uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
return pFIR->evaluate(dest, src, numSamples, numChannels);
}


uint AAFilter::getLength() const
{
return pFIR->getLength();
}
91 changes: 91 additions & 0 deletions Externals/soundtouch/AAFilter.h
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////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#ifndef AAFilter_H
#define AAFilter_H

#include "STTypes.h"

namespace soundtouch
{

class AAFilter
{
protected:
class FIRFilter *pFIR;

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

/// num of filter taps
uint length;

/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
AAFilter(uint length);

~AAFilter();

/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
/// frequency (nyquist frequency = 0.5). The filter will cut off the
/// frequencies than that.
void setCutoffFreq(double newCutoffFreq);

/// Sets number of FIR filter taps, i.e. ~filter complexity
void setLength(uint newLength);

uint getLength() const;

/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
};

}

#endif
370 changes: 370 additions & 0 deletions Externals/soundtouch/BPMDetect.cpp
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////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-08-30 22:45:25 +0300 (Thu, 30 Aug 2012) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.cpp 149 2012-08-30 19:45:25Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#include <math.h>
#include <assert.h>
#include <string.h>
#include <stdio.h>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"

using namespace soundtouch;

#define INPUT_BLOCK_SAMPLES 2048
#define DECIMATED_BLOCK_SAMPLES 256

/// decay constant for calculating RMS volume sliding average approximation
/// (time constant is about 10 sec)
const float avgdecay = 0.99986f;

/// Normalization coefficient for calculating RMS sliding average approximation.
const float avgnorm = (1 - avgdecay);


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

// Enable following define to create bpm analysis file:

// #define _CREATE_BPM_DEBUG_FILE

#ifdef _CREATE_BPM_DEBUG_FILE

#define DEBUGFILE_NAME "c:\\temp\\soundtouch-bpm-debug.txt"

static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
{
FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
int i;

if (fptr)
{
printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
for (i = minpos; i < maxpos; i ++)
{
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
}
fclose(fptr);
}
}
#else
#define _SaveDebugData(a,b,c,d)
#endif

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


BPMDetect::BPMDetect(int numChannels, int aSampleRate)
{
this->sampleRate = aSampleRate;
this->channels = numChannels;

decimateSum = 0;
decimateCount = 0;

envelopeAccu = 0;

// Initialize RMS volume accumulator to RMS level of 1500 (out of 32768) that's
// safe initial RMS signal level value for song data. This value is then adapted
// to the actual level during processing.
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// integer samples
RMSVolumeAccu = (1500 * 1500) / avgnorm;
#else
// float samples, scaled to range [-1..+1[
RMSVolumeAccu = (0.045f * 0.045f) / avgnorm;
#endif

// choose decimation factor so that result is approx. 1000 Hz
decimateBy = sampleRate / 1000;
assert(decimateBy > 0);
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);

// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);

assert(windowLen > windowStart);

// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));

// allocate processing buffer
buffer = new FIFOSampleBuffer();
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
}



BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete buffer;
}



/// convert to mono, low-pass filter & decimate to about 500 Hz.
/// return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;

assert(channels > 0);
assert(decimateBy > 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
int j;

// convert to mono and accumulate
for (j = 0; j < channels; j ++)
{
decimateSum += src[j];
}
src += j;

decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
decimateSum = 0;
decimateCount = 0;
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}



// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;

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

pBuffer = buffer->ptrBegin();
for (offs = windowStart; offs < windowLen; offs ++)
{
LONG_SAMPLETYPE sum;
int i;

sum = 0;
for (i = 0; i < process_samples; i ++)
{
sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
}
// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients
// if it's desired that the system adapts automatically to
// various bpms, e.g. in processing continouos music stream.
// The 'xcorr_decay' should be a value that's smaller than but
// close to one, and should also depend on 'process_samples' value.

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


// Calculates envelope of the sample data
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
{
const static double decay = 0.7f; // decay constant for smoothing the envelope
const static double norm = (1 - decay);

int i;
LONG_SAMPLETYPE out;
double val;

for (i = 0; i < numsamples; i ++)
{
// calc average RMS volume
RMSVolumeAccu *= avgdecay;
val = (float)fabs((float)samples[i]);
RMSVolumeAccu += val * val;

// cut amplitudes that are below cutoff ~2 times RMS volume
// (we're interested in peak values, not the silent moments)
if (val < 0.5 * sqrt(RMSVolumeAccu * avgnorm))
{
val = 0;
}

// smooth amplitude envelope
envelopeAccu *= decay;
envelopeAccu += val;
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);

#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// cut peaks (shouldn't be necessary though)
if (out > 32767) out = 32767;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
samples[i] = (SAMPLETYPE)out;
}
}



void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];

// iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
while (numSamples > 0)
{
int block;
int decSamples;

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

// decimate. note that converts to mono at the same time
decSamples = decimate(decimated, samples, block);
samples += block * channels;
numSamples -= block;

// envelope new samples and add them to buffer
calcEnvelope(decimated, decSamples);
buffer->putSamples(decimated, decSamples);
}

// when the buffer has enought samples for processing...
if ((int)buffer->numSamples() > windowLen)
{
int processLength;

// how many samples are processed
processLength = (int)buffer->numSamples() - windowLen;

// ... calculate autocorrelations for oldest samples...
updateXCorr(processLength);
// ... and remove them from the buffer
buffer->receiveSamples(processLength);
}
}



void BPMDetect::removeBias()
{
int i;
float minval = 1e12f; // arbitrary large number

for (i = windowStart; i < windowLen; i ++)
{
if (xcorr[i] < minval)
{
minval = xcorr[i];
}
}

for (i = windowStart; i < windowLen; i ++)
{
xcorr[i] -= minval;
}
}


float BPMDetect::getBpm()
{
double peakPos;
double coeff;
PeakFinder peakFinder;

coeff = 60.0 * ((double)sampleRate / (double)decimateBy);

// save bpm debug analysis data if debug data enabled
_SaveDebugData(xcorr, windowStart, windowLen, coeff);

// remove bias from xcorr data
removeBias();

// find peak position
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);

assert(decimateBy != 0);
if (peakPos < 1e-9) return 0.0; // detection failed.

// calculate BPM
return (float) (coeff / peakPos);
}
164 changes: 164 additions & 0 deletions Externals/soundtouch/BPMDetect.h
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////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-08-30 22:53:44 +0300 (Thu, 30 Aug 2012) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#ifndef _BPMDetect_H_
#define _BPMDetect_H_

#include "STTypes.h"
#include "FIFOSampleBuffer.h"

namespace soundtouch
{

/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 29

/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM 200


/// Class for calculating BPM rate for audio data.
class BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;

/// Amplitude envelope sliding average approximation level accumulator
double envelopeAccu;

/// RMS volume sliding average approximation level accumulator
double RMSVolumeAccu;

/// Sample average counter.
int decimateCount;

/// Sample average accumulator for FIFO-like decimation.
soundtouch::LONG_SAMPLETYPE decimateSum;

/// Decimate sound by this coefficient to reach approx. 500 Hz.
int decimateBy;

/// Auto-correlation window length
int windowLen;

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

/// sample rate
int sampleRate;

/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
/// the first these many correlation bins.
int windowStart;

/// FIFO-buffer for decimated processing samples.
soundtouch::FIFOSampleBuffer *buffer;

/// Updates auto-correlation function for given number of decimated samples that
/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
/// though).
void updateXCorr(int process_samples /// How many samples are processed.
);

/// Decimates samples to approx. 500 Hz.
///
/// \return Number of output samples.
int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
int numsamples ///< Number of source samples.
);

/// Calculates amplitude envelope for the buffer of samples.
/// Result is output to 'samples'.
void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
int numsamples ///< Number of samples in buffer
);

/// remove constant bias from xcorr data
void removeBias();

public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
int sampleRate ///< Sample rate in Hz.
);

/// Destructor.
virtual ~BPMDetect();

/// Inputs a block of samples for analyzing: Envelopes the samples and then
/// updates the autocorrelation estimation. When whole song data has been input
/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
/// function.
///
/// Notice that data in 'samples' array can be disrupted in processing.
void inputSamples(const soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
int numSamples ///< Number of samples in buffer
);


/// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function.
///
/// \return Beats-per-minute rate, or zero if detection failed.
float getBpm();
};

}

#endif // _BPMDetect_H_
15 changes: 15 additions & 0 deletions Externals/soundtouch/CMakeLists.txt
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set(SRCS
AAFilter.cpp
BPMDetect.cpp
cpu_detect_x86.cpp
FIFOSampleBuffer.cpp
FIRFilter.cpp
mmx_optimized.cpp
PeakFinder.cpp
RateTransposer.cpp
SoundTouch.cpp
sse_optimized.cpp
TDStretch.cpp
)

add_library(SoundTouch STATIC ${SRCS})
274 changes: 274 additions & 0 deletions Externals/soundtouch/FIFOSampleBuffer.cpp
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////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// outputted samples from the buffer, as well as grows the buffer size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#include <stdlib.h>
#include <memory.h>
#include <string.h>
#include <assert.h>

#include "FIFOSampleBuffer.h"

using namespace soundtouch;

// Constructor
FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
{
assert(numChannels > 0);
sizeInBytes = 0; // reasonable initial value
buffer = NULL;
bufferUnaligned = NULL;
samplesInBuffer = 0;
bufferPos = 0;
channels = (uint)numChannels;
ensureCapacity(32); // allocate initial capacity
}


// destructor
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
bufferUnaligned = NULL;
buffer = NULL;
}


// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(int numChannels)
{
uint usedBytes;

assert(numChannels > 0);
usedBytes = channels * samplesInBuffer;
channels = (uint)numChannels;
samplesInBuffer = usedBytes / channels;
}


// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
// zeroes this pointer by copying samples from the 'bufferPos' pointer
// location on to the beginning of the buffer.
void FIFOSampleBuffer::rewind()
{
if (buffer && bufferPos)
{
memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
bufferPos = 0;
}
}


// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// the sample buffer.
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}


// Increases the number of samples in the buffer without copying any actual
// samples.
//
// This function is used to update the number of samples in the sample buffer
// when accessing the buffer directly with 'ptrEnd' function. Please be
// careful though!
void FIFOSampleBuffer::putSamples(uint nSamples)
{
uint req;

req = samplesInBuffer + nSamples;
ensureCapacity(req);
samplesInBuffer += nSamples;
}


// Returns a pointer to the end of the used part of the sample buffer (i.e.
// where the new samples are to be inserted). This function may be used for
// inserting new samples into the sample buffer directly. Please be careful!
//
// Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to
// succesfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement.
//
// When using this function as means for inserting new samples, also remember
// to increase the sample count afterwards, by calling the
// 'putSamples(numSamples)' function.
SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
{
ensureCapacity(samplesInBuffer + slackCapacity);
return buffer + samplesInBuffer * channels;
}


// Returns a pointer to the beginning of the currently non-outputted samples.
// This function is provided for accessing the output samples directly.
// Please be careful!
//
// When using this function to output samples, also remember to 'remove' the
// outputted samples from the buffer by calling the
// 'receiveSamples(numSamples)' function
SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
{
assert(buffer);
return buffer + bufferPos * channels;
}


// Ensures that the buffer has enought capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of
// 4 kilobytes to eliminate the need for frequently growing up the buffer,
// as well as to round the buffer size up to the virtual memory page size.
void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
{
SAMPLETYPE *tempUnaligned, *temp;

if (capacityRequirement > getCapacity())
{
// enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == NULL)
{
ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
}
// Align the buffer to begin at 16byte cache line boundary for optimal performance
temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
if (samplesInBuffer)
{
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
}
delete[] bufferUnaligned;
buffer = temp;
bufferUnaligned = tempUnaligned;
bufferPos = 0;
}
else
{
// simply rewind the buffer (if necessary)
rewind();
}
}


// Returns the current buffer capacity in terms of samples
uint FIFOSampleBuffer::getCapacity() const
{
return sizeInBytes / (channels * sizeof(SAMPLETYPE));
}


// Returns the number of samples currently in the buffer
uint FIFOSampleBuffer::numSamples() const
{
return samplesInBuffer;
}


// Output samples from beginning of the sample buffer. Copies demanded number
// of samples to output and removes them from the sample buffer. If there
// are less than 'numsample' samples in the buffer, returns all available.
//
// Returns number of samples copied.
uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint num;

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

memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
return receiveSamples(num);
}


// Removes samples from the beginning of the sample buffer without copying them
// anywhere. Used to reduce the number of samples in the buffer, when accessing
// the sample buffer with the 'ptrBegin' function.
uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
{
if (maxSamples >= samplesInBuffer)
{
uint temp;

temp = samplesInBuffer;
samplesInBuffer = 0;
return temp;
}

samplesInBuffer -= maxSamples;
bufferPos += maxSamples;

return maxSamples;
}


// Returns nonzero if the sample buffer is empty
int FIFOSampleBuffer::isEmpty() const
{
return (samplesInBuffer == 0) ? 1 : 0;
}


// Clears the sample buffer
void FIFOSampleBuffer::clear()
{
samplesInBuffer = 0;
bufferPos = 0;
}


/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
{
if (numSamples < samplesInBuffer)
{
samplesInBuffer = numSamples;
}
return samplesInBuffer;
}

178 changes: 178 additions & 0 deletions Externals/soundtouch/FIFOSampleBuffer.h
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////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// output samples from the buffer as well as grows the storage size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.h 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#ifndef FIFOSampleBuffer_H
#define FIFOSampleBuffer_H

#include "FIFOSamplePipe.h"

namespace soundtouch
{

/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
/// care of storage size adjustment and data moving during input/output operations.
///
/// Notice that in case of stereo audio, one sample is considered to consist of
/// both channel data.
class FIFOSampleBuffer : public FIFOSamplePipe
{
private:
/// Sample buffer.
SAMPLETYPE *buffer;

// Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
// 16-byte aligned location of this buffer
SAMPLETYPE *bufferUnaligned;

/// Sample buffer size in bytes
uint sizeInBytes;

/// How many samples are currently in buffer.
uint samplesInBuffer;

/// Channels, 1=mono, 2=stereo.
uint channels;

/// Current position pointer to the buffer. This pointer is increased when samples are
/// removed from the pipe so that it's necessary to actually rewind buffer (move data)
/// only new data when is put to the pipe.
uint bufferPos;

/// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
/// beginning of the buffer.
void rewind();

/// Ensures that the buffer has capacity for at least this many samples.
void ensureCapacity(uint capacityRequirement);

/// Returns current capacity.
uint getCapacity() const;

public:

/// Constructor
FIFOSampleBuffer(int numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
///< Default is stereo.
);

/// destructor
~FIFOSampleBuffer();

/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin();

/// Returns a pointer to the end of the used part of the sample buffer (i.e.
/// where the new samples are to be inserted). This function may be used for
/// inserting new samples into the sample buffer directly. Please be careful
/// not corrupt the book-keeping!
///
/// When using this function as means for inserting new samples, also remember
/// to increase the sample count afterwards, by calling the
/// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can succesfully insert the
///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement.
);

/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
);

/// Adjusts the book-keeping to increase number of samples in the buffer without
/// copying any actual samples.
///
/// This function is used to update the number of samples in the sample buffer
/// when accessing the buffer directly with 'ptrEnd' function. Please be
/// careful though!
virtual void putSamples(uint numSamples ///< Number of samples been inserted.
);

/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
);

/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
);

/// Returns number of samples currently available.
virtual uint numSamples() const;

/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(int numChannels);

/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const;

/// Clears all the samples.
virtual void clear();

/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint adjustAmountOfSamples(uint numSamples);
};

}

#endif
234 changes: 234 additions & 0 deletions Externals/soundtouch/FIFOSamplePipe.h
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////////////////////////////////////////////////////////////////////////////////
///
/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
/// samples by operating like a first-in-first-out pipe: New samples are fed
/// into one end of the pipe with the 'putSamples' function, and the processed
/// samples are received from the other end with the 'receiveSamples' function.
///
/// 'FIFOProcessor' : A base class for classes the do signal processing with
/// the samples while operating like a first-in-first-out pipe. When samples
/// are input with the 'putSamples' function, the class processes them
/// and moves the processed samples to the given 'output' pipe object, which
/// may be either another processing stage, or a fifo sample buffer object.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#ifndef FIFOSamplePipe_H
#define FIFOSamplePipe_H

#include <assert.h>
#include <stdlib.h>
#include "STTypes.h"

namespace soundtouch
{

/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
class FIFOSamplePipe
{
public:
// virtual default destructor
virtual ~FIFOSamplePipe() {}


/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() = 0;

/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
) = 0;


// Moves samples from the 'other' pipe instance to this instance.
void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
)
{
int oNumSamples = other.numSamples();

putSamples(other.ptrBegin(), oNumSamples);
other.receiveSamples(oNumSamples);
};

/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
) = 0;

/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) = 0;

/// Returns number of samples currently available.
virtual uint numSamples() const = 0;

// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const = 0;

/// Clears all the samples.
virtual void clear() = 0;

/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples) = 0;

};



/// Base-class for sound processing routines working in FIFO principle. With this base
/// class it's easy to implement sound processing stages that can be chained together,
/// so that samples that are fed into beginning of the pipe automatically go through
/// all the processing stages.
///
/// When samples are input to this class, they're first processed and then put to
/// the FIFO pipe that's defined as output of this class. This output pipe can be
/// either other processing stage or a FIFO sample buffer.
class FIFOProcessor :public FIFOSamplePipe
{
protected:
/// Internal pipe where processed samples are put.
FIFOSamplePipe *output;

/// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput)
{
assert(output == NULL);
assert(pOutput != NULL);
output = pOutput;
}


/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
output = NULL;
}


/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
{
output = pOutput;
}


/// Destructor.
virtual ~FIFOProcessor()
{
}


/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin()
{
return output->ptrBegin();
}

public:

/// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
)
{
return output->receiveSamples(outBuffer, maxSamples);
}


/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
)
{
return output->receiveSamples(maxSamples);
}


/// Returns number of samples currently available.
virtual uint numSamples() const
{
return output->numSamples();
}


/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const
{
return output->isEmpty();
}

/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples)
{
return output->adjustAmountOfSamples(numSamples);
}

};

}

#endif
259 changes: 259 additions & 0 deletions Externals/soundtouch/FIRFilter.cpp
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////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-09-02 21:56:11 +0300 (Fri, 02 Sep 2011) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.cpp 131 2011-09-02 18:56:11Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "FIRFilter.h"
#include "cpu_detect.h"

using namespace soundtouch;

/*****************************************************************************
*
* Implementation of the class 'FIRFilter'
*
*****************************************************************************/

FIRFilter::FIRFilter()
{
resultDivFactor = 0;
resultDivider = 0;
length = 0;
lengthDiv8 = 0;
filterCoeffs = NULL;
}


FIRFilter::~FIRFilter()
{
delete[] filterCoeffs;
}

// Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE suml, sumr;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif

assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);

end = 2 * (numSamples - length);

for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;

suml = sumr = 0;
ptr = src + j;

for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
ptr[2 * i + 2] * filterCoeffs[i + 1] +
ptr[2 * i + 4] * filterCoeffs[i + 2] +
ptr[2 * i + 6] * filterCoeffs[i + 3];
sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
ptr[2 * i + 3] * filterCoeffs[i + 1] +
ptr[2 * i + 5] * filterCoeffs[i + 2] +
ptr[2 * i + 7] * filterCoeffs[i + 3];
}

#ifdef SOUNDTOUCH_INTEGER_SAMPLES
suml >>= resultDivFactor;
sumr >>= resultDivFactor;
// saturate to 16 bit integer limits
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - length;
}




// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
uint i, j, end;
LONG_SAMPLETYPE sum;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif


assert(length != 0);

end = numSamples - length;
for (j = 0; j < end; j ++)
{
sum = 0;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
sum += src[i + 0] * filterCoeffs[i + 0] +
src[i + 1] * filterCoeffs[i + 1] +
src[i + 2] * filterCoeffs[i + 2] +
src[i + 3] * filterCoeffs[i + 3];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
src ++;
}
return end;
}


// Set filter coeffiecients and length.
//
// Throws an exception if filter length isn't divisible by 8
void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
{
assert(newLength > 0);
if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");

lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);

resultDivFactor = uResultDivFactor;
resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);

delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
}


uint FIRFilter::getLength() const
{
return length;
}



// Applies the filter to the given sequence of samples.
//
// Note : The amount of outputted samples is by value of 'filter_length'
// smaller than the amount of input samples.
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
assert(numChannels == 1 || numChannels == 2);

assert(length > 0);
assert(lengthDiv8 * 8 == length);
if (numSamples < length) return 0;
if (numChannels == 2)
{
return evaluateFilterStereo(dest, src, numSamples);
} else {
return evaluateFilterMono(dest, src, numSamples);
}
}



// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX-capable CPU available or not.
void * FIRFilter::operator new(size_t s)
{
// Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");
return newInstance();
}


FIRFilter * FIRFilter::newInstance()
{
uint uExtensions;

uExtensions = detectCPUextensions();

// Check if MMX/SSE instruction set extensions supported by CPU

#ifdef SOUNDTOUCH_ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new FIRFilterMMX;
}
else
#endif // SOUNDTOUCH_ALLOW_MMX

#ifdef SOUNDTOUCH_ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new FIRFilterSSE;
}
else
#endif // SOUNDTOUCH_ALLOW_SSE

{
// ISA optimizations not supported, use plain C version
return ::new FIRFilter;
}
}
145 changes: 145 additions & 0 deletions Externals/soundtouch/FIRFilter.h
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////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-02-13 21:13:57 +0200 (Sun, 13 Feb 2011) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.h 104 2011-02-13 19:13:57Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#ifndef FIRFilter_H
#define FIRFilter_H

#include <stddef.h>
#include "STTypes.h"

namespace soundtouch
{

class FIRFilter
{
protected:
// Number of FIR filter taps
uint length;
// Number of FIR filter taps divided by 8
uint lengthDiv8;

// Result divider factor in 2^k format
uint resultDivFactor;

// Result divider value.
SAMPLETYPE resultDivider;

// Memory for filter coefficients
SAMPLETYPE *filterCoeffs;

virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;

public:
FIRFilter();
virtual ~FIRFilter();

/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we've a MMX-capable CPU available or not.
static void * operator new(size_t s);

static FIRFilter *newInstance();

/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter_length'
/// smaller than the amount of input samples.
///
/// \return Number of samples copied to 'dest'.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;

uint getLength() const;

virtual void setCoefficients(const SAMPLETYPE *coeffs,
uint newLength,
uint uResultDivFactor);
};


// Optional subclasses that implement CPU-specific optimizations:

#ifdef SOUNDTOUCH_ALLOW_MMX

/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
class FIRFilterMMX : public FIRFilter
{
protected:
short *filterCoeffsUnalign;
short *filterCoeffsAlign;

virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const;
public:
FIRFilterMMX();
~FIRFilterMMX();

virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
};

#endif // SOUNDTOUCH_ALLOW_MMX


#ifdef SOUNDTOUCH_ALLOW_SSE
/// Class that implements SSE optimized functions exclusive for floating point samples type.
class FIRFilterSSE : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;

virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const;
public:
FIRFilterSSE();
~FIRFilterSSE();

virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};

#endif // SOUNDTOUCH_ALLOW_SSE

}

#endif // FIRFilter_H
276 changes: 276 additions & 0 deletions Externals/soundtouch/PeakFinder.cpp
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////////////////////////////////////////////////////////////////////////////////
///
/// Peak detection routine.
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-12-28 21:52:47 +0200 (Fri, 28 Dec 2012) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.cpp 164 2012-12-28 19:52:47Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////

#include <math.h>
#include <assert.h>

#include "PeakFinder.h"

using namespace soundtouch;

#define max(x, y) (((x) > (y)) ? (x) : (y))


PeakFinder::PeakFinder()
{
minPos = maxPos = 0;
}


// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
int PeakFinder::findTop(const float *data, int peakpos) const
{
int i;
int start, end;
float refvalue;

refvalue = data[peakpos];

// seek within ±10 points
start = peakpos - 10;
if (start < minPos) start = minPos;
end = peakpos + 10;
if (end > maxPos) end = maxPos;

for (i = start; i <= end; i ++)
{
if (data[i] > refvalue)
{
peakpos = i;
refvalue = data[i];
}
}

// failure if max value is at edges of seek range => it's not peak, it's at slope.
if ((peakpos == start) || (peakpos == end)) return 0;

return peakpos;
}


// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
// to direction defined by 'direction' until next 'hump' after minimum value will
// begin
int PeakFinder::findGround(const float *data, int peakpos, int direction) const
{
int lowpos;
int pos;
int climb_count;
float refvalue;
float delta;

climb_count = 0;
refvalue = data[peakpos];
lowpos = peakpos;

pos = peakpos;

while ((pos > minPos+1) && (pos < maxPos-1))
{
int prevpos;

prevpos = pos;
pos += direction;

// calculate derivate
delta = data[pos] - data[prevpos];
if (delta <= 0)
{
// going downhill, ok
if (climb_count)
{
climb_count --; // decrease climb count
}

// check if new minimum found
if (data[pos] < refvalue)
{
// new minimum found
lowpos = pos;
refvalue = data[pos];
}
}
else
{
// going uphill, increase climbing counter
climb_count ++;
if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit
}
}
return lowpos;
}


// Find offset where the value crosses the given level, when starting from 'peakpos' and
// proceeds to direction defined in 'direction'
int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
{
float peaklevel;
int pos;

peaklevel = data[peakpos];
assert(peaklevel >= level);
pos = peakpos;
while ((pos >= minPos) && (pos < maxPos))
{
if (data[pos + direction] < level) return pos; // crossing found
pos += direction;
}
return -1; // not found
}


// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
{
int i;
float sum;
float wsum;

sum = 0;
wsum = 0;
for (i = firstPos; i <= lastPos; i ++)
{
sum += (float)i * data[i];
wsum += data[i];
}

if (wsum < 1e-6) return 0;
return sum / wsum;
}



/// get exact center of peak near given position by calculating local mass of center
double PeakFinder::getPeakCenter(const float *data, int peakpos) const
{
float peakLevel; // peak level
int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level
float cutLevel; // cutting value
float groundLevel; // ground level of the peak
int gp1, gp2; // bottom positions of the peak 'hump'

// find ground positions.
gp1 = findGround(data, peakpos, -1);
gp2 = findGround(data, peakpos, 1);

groundLevel = 0.5f * (data[gp1] + data[gp2]);
peakLevel = data[peakpos];

// calculate 70%-level of the peak
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
// find mid-level crossings
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);

if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak..

// calculate mass center of the peak surroundings
return calcMassCenter(data, crosspos1, crosspos2);
}



double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
{

int i;
int peakpos; // position of peak level
double highPeak, peak;

this->minPos = aminPos;
this->maxPos = amaxPos;

// find absolute peak
peakpos = minPos;
peak = data[minPos];
for (i = minPos + 1; i < maxPos; i ++)
{
if (data[i] > peak)
{
peak = data[i];
peakpos = i;
}
}

// Calculate exact location of the highest peak mass center
highPeak = getPeakCenter(data, peakpos);
peak = highPeak;

// Now check if the highest peak were in fact harmonic of the true base beat peak
// - sometimes the highest peak can be Nth harmonic of the true base peak yet
// just a slightly higher than the true base

for (i = 3; i < 10; i ++)
{
double peaktmp, harmonic;
int i1,i2;

harmonic = (double)i * 0.5;
peakpos = (int)(highPeak / harmonic + 0.5f);
if (peakpos < minPos) break;
peakpos = findTop(data, peakpos); // seek true local maximum index
if (peakpos == 0) continue; // no local max here

// calculate mass-center of possible harmonic peak
peaktmp = getPeakCenter(data, peakpos);

// accept harmonic peak if
// (a) it is found
// (b) is within ±4% of the expected harmonic interval
// (c) has at least half x-corr value of the max. peak

double diff = harmonic * peaktmp / highPeak;
if ((diff < 0.96) || (diff > 1.04)) continue; // peak too afar from expected

// now compare to highest detected peak
i1 = (int)(highPeak + 0.5);
i2 = (int)(peaktmp + 0.5);
if (data[i2] >= 0.4*data[i1])
{
// The harmonic is at least half as high primary peak,
// thus use the harmonic peak instead
peak = peaktmp;
}
}

return peak;
}