AuroraOpenALSoft/OpenAL32/Include/alu.h

495 lines
14 KiB
C

#ifndef _ALU_H_
#define _ALU_H_
#include <limits.h>
#include <math.h>
#ifdef HAVE_FLOAT_H
#include <float.h>
#endif
#ifdef HAVE_IEEEFP_H
#include <ieeefp.h>
#endif
#include "alMain.h"
#include "alBuffer.h"
#include "hrtf.h"
#include "math_defs.h"
#include "filters/defs.h"
#include "filters/nfc.h"
enum class DistanceModel;
#define MAX_PITCH (255)
/* Maximum number of samples to pad on either end of a buffer for resampling.
* Note that both the beginning and end need padding!
*/
#define MAX_RESAMPLE_PADDING 24
#ifdef __cplusplus
extern "C" {
#endif
struct BSincTable;
struct ALsource;
struct ALbufferlistitem;
struct ALvoice;
struct ALeffectslot;
#define DITHER_RNG_SEED 22222
enum SpatializeMode {
SpatializeOff = AL_FALSE,
SpatializeOn = AL_TRUE,
SpatializeAuto = AL_AUTO_SOFT
};
enum Resampler {
PointResampler,
LinearResampler,
FIR4Resampler,
BSinc12Resampler,
BSinc24Resampler,
ResamplerMax = BSinc24Resampler
};
extern enum Resampler ResamplerDefault;
/* The number of distinct scale and phase intervals within the bsinc filter
* table.
*/
#define BSINC_SCALE_BITS 4
#define BSINC_SCALE_COUNT (1<<BSINC_SCALE_BITS)
#define BSINC_PHASE_BITS 4
#define BSINC_PHASE_COUNT (1<<BSINC_PHASE_BITS)
/* Interpolator state. Kind of a misnomer since the interpolator itself is
* stateless. This just keeps it from having to recompute scale-related
* mappings for every sample.
*/
typedef struct BsincState {
ALfloat sf; /* Scale interpolation factor. */
ALsizei m; /* Coefficient count. */
ALsizei l; /* Left coefficient offset. */
/* Filter coefficients, followed by the scale, phase, and scale-phase
* delta coefficients. Starting at phase index 0, each subsequent phase
* index follows contiguously.
*/
const ALfloat *filter;
} BsincState;
typedef union InterpState {
BsincState bsinc;
} InterpState;
typedef const ALfloat* (*ResamplerFunc)(const InterpState *state,
const ALfloat *RESTRICT src, ALsizei frac, ALint increment,
ALfloat *RESTRICT dst, ALsizei dstlen
);
void BsincPrepare(const ALuint increment, BsincState *state, const struct BSincTable *table);
extern const struct BSincTable bsinc12;
extern const struct BSincTable bsinc24;
enum {
AF_None = 0,
AF_LowPass = 1,
AF_HighPass = 2,
AF_BandPass = AF_LowPass | AF_HighPass
};
typedef struct MixHrtfParams {
const ALfloat (*Coeffs)[2];
ALsizei Delay[2];
ALfloat Gain;
ALfloat GainStep;
} MixHrtfParams;
typedef struct DirectParams {
BiquadFilter LowPass;
BiquadFilter HighPass;
NfcFilter NFCtrlFilter;
struct {
HrtfParams Old;
HrtfParams Target;
HrtfState State;
} Hrtf;
struct {
ALfloat Current[MAX_OUTPUT_CHANNELS];
ALfloat Target[MAX_OUTPUT_CHANNELS];
} Gains;
} DirectParams;
typedef struct SendParams {
BiquadFilter LowPass;
BiquadFilter HighPass;
struct {
ALfloat Current[MAX_OUTPUT_CHANNELS];
ALfloat Target[MAX_OUTPUT_CHANNELS];
} Gains;
} SendParams;
struct ALvoiceProps {
ATOMIC(struct ALvoiceProps*) next;
ALfloat Pitch;
ALfloat Gain;
ALfloat OuterGain;
ALfloat MinGain;
ALfloat MaxGain;
ALfloat InnerAngle;
ALfloat OuterAngle;
ALfloat RefDistance;
ALfloat MaxDistance;
ALfloat RolloffFactor;
ALfloat Position[3];
ALfloat Velocity[3];
ALfloat Direction[3];
ALfloat Orientation[2][3];
ALboolean HeadRelative;
DistanceModel mDistanceModel;
enum Resampler Resampler;
ALboolean DirectChannels;
enum SpatializeMode SpatializeMode;
ALboolean DryGainHFAuto;
ALboolean WetGainAuto;
ALboolean WetGainHFAuto;
ALfloat OuterGainHF;
ALfloat AirAbsorptionFactor;
ALfloat RoomRolloffFactor;
ALfloat DopplerFactor;
ALfloat StereoPan[2];
ALfloat Radius;
/** Direct filter and auxiliary send info. */
struct {
ALfloat Gain;
ALfloat GainHF;
ALfloat HFReference;
ALfloat GainLF;
ALfloat LFReference;
} Direct;
struct {
struct ALeffectslot *Slot;
ALfloat Gain;
ALfloat GainHF;
ALfloat HFReference;
ALfloat GainLF;
ALfloat LFReference;
} Send[];
};
#define VOICE_IS_STATIC (1<<0)
#define VOICE_IS_FADING (1<<1) /* Fading sources use gain stepping for smooth transitions. */
#define VOICE_HAS_HRTF (1<<2)
#define VOICE_HAS_NFC (1<<3)
typedef struct ALvoice {
struct ALvoiceProps *Props;
ATOMIC(struct ALvoiceProps*) Update;
ATOMIC(struct ALsource*) Source;
ATOMIC(bool) Playing;
/**
* Source offset in samples, relative to the currently playing buffer, NOT
* the whole queue, and the fractional (fixed-point) offset to the next
* sample.
*/
ATOMIC(ALuint) position;
ATOMIC(ALsizei) position_fraction;
/* Current buffer queue item being played. */
ATOMIC(struct ALbufferlistitem*) current_buffer;
/* Buffer queue item to loop to at end of queue (will be NULL for non-
* looping voices).
*/
ATOMIC(struct ALbufferlistitem*) loop_buffer;
/**
* Number of channels and bytes-per-sample for the attached source's
* buffer(s).
*/
ALsizei NumChannels;
ALsizei SampleSize;
/** Current target parameters used for mixing. */
ALint Step;
ResamplerFunc Resampler;
ALuint Flags;
ALuint Offset; /* Number of output samples mixed since starting. */
alignas(16) ALfloat PrevSamples[MAX_INPUT_CHANNELS][MAX_RESAMPLE_PADDING];
InterpState ResampleState;
struct {
int FilterType;
DirectParams Params[MAX_INPUT_CHANNELS];
ALfloat (*Buffer)[BUFFERSIZE];
ALsizei Channels;
ALsizei ChannelsPerOrder[MAX_AMBI_ORDER+1];
} Direct;
struct {
int FilterType;
SendParams Params[MAX_INPUT_CHANNELS];
ALfloat (*Buffer)[BUFFERSIZE];
ALsizei Channels;
} Send[];
} ALvoice;
void DeinitVoice(ALvoice *voice);
typedef void (*MixerFunc)(const ALfloat *data, ALsizei OutChans,
ALfloat (*RESTRICT OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains,
const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize);
typedef void (*RowMixerFunc)(ALfloat *OutBuffer, const ALfloat *gains,
const ALfloat (*RESTRICT data)[BUFFERSIZE], ALsizei InChans,
ALsizei InPos, ALsizei BufferSize);
typedef void (*HrtfMixerFunc)(ALfloat *RESTRICT LeftOut, ALfloat *RESTRICT RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, MixHrtfParams *hrtfparams,
HrtfState *hrtfstate, ALsizei BufferSize);
typedef void (*HrtfMixerBlendFunc)(ALfloat *RESTRICT LeftOut, ALfloat *RESTRICT RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, const HrtfParams *oldparams,
MixHrtfParams *newparams, HrtfState *hrtfstate,
ALsizei BufferSize);
typedef void (*HrtfDirectMixerFunc)(ALfloat *RESTRICT LeftOut, ALfloat *RESTRICT RightOut,
const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
const ALfloat (*RESTRICT Coeffs)[2],
ALfloat (*RESTRICT Values)[2], ALsizei BufferSize);
#define GAIN_MIX_MAX (16.0f) /* +24dB */
#define GAIN_SILENCE_THRESHOLD (0.00001f) /* -100dB */
#define SPEEDOFSOUNDMETRESPERSEC (343.3f)
#define AIRABSORBGAINHF (0.99426f) /* -0.05dB */
/* Target gain for the reverb decay feedback reaching the decay time. */
#define REVERB_DECAY_GAIN (0.001f) /* -60 dB */
#define FRACTIONBITS (12)
#define FRACTIONONE (1<<FRACTIONBITS)
#define FRACTIONMASK (FRACTIONONE-1)
inline ALfloat minf(ALfloat a, ALfloat b)
{ return ((a > b) ? b : a); }
inline ALfloat maxf(ALfloat a, ALfloat b)
{ return ((a > b) ? a : b); }
inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max)
{ return minf(max, maxf(min, val)); }
inline ALdouble mind(ALdouble a, ALdouble b)
{ return ((a > b) ? b : a); }
inline ALdouble maxd(ALdouble a, ALdouble b)
{ return ((a > b) ? a : b); }
inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max)
{ return mind(max, maxd(min, val)); }
inline ALuint minu(ALuint a, ALuint b)
{ return ((a > b) ? b : a); }
inline ALuint maxu(ALuint a, ALuint b)
{ return ((a > b) ? a : b); }
inline ALuint clampu(ALuint val, ALuint min, ALuint max)
{ return minu(max, maxu(min, val)); }
inline ALint mini(ALint a, ALint b)
{ return ((a > b) ? b : a); }
inline ALint maxi(ALint a, ALint b)
{ return ((a > b) ? a : b); }
inline ALint clampi(ALint val, ALint min, ALint max)
{ return mini(max, maxi(min, val)); }
inline ALint64 mini64(ALint64 a, ALint64 b)
{ return ((a > b) ? b : a); }
inline ALint64 maxi64(ALint64 a, ALint64 b)
{ return ((a > b) ? a : b); }
inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max)
{ return mini64(max, maxi64(min, val)); }
inline ALuint64 minu64(ALuint64 a, ALuint64 b)
{ return ((a > b) ? b : a); }
inline ALuint64 maxu64(ALuint64 a, ALuint64 b)
{ return ((a > b) ? a : b); }
inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max)
{ return minu64(max, maxu64(min, val)); }
inline size_t minz(size_t a, size_t b)
{ return ((a > b) ? b : a); }
inline size_t maxz(size_t a, size_t b)
{ return ((a > b) ? a : b); }
inline size_t clampz(size_t val, size_t min, size_t max)
{ return minz(max, maxz(min, val)); }
inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu)
{
return val1 + (val2-val1)*mu;
}
inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu)
{
ALfloat mu2 = mu*mu, mu3 = mu2*mu;
ALfloat a0 = -0.5f*mu3 + mu2 + -0.5f*mu;
ALfloat a1 = 1.5f*mu3 + -2.5f*mu2 + 1.0f;
ALfloat a2 = -1.5f*mu3 + 2.0f*mu2 + 0.5f*mu;
ALfloat a3 = 0.5f*mu3 + -0.5f*mu2;
return val1*a0 + val2*a1 + val3*a2 + val4*a3;
}
enum HrtfRequestMode {
Hrtf_Default = 0,
Hrtf_Enable = 1,
Hrtf_Disable = 2,
};
void aluInit(void);
void aluInitMixer(void);
ResamplerFunc SelectResampler(enum Resampler resampler);
/* aluInitRenderer
*
* Set up the appropriate panning method and mixing method given the device
* properties.
*/
void aluInitRenderer(ALCdevice *device, ALint hrtf_id, enum HrtfRequestMode hrtf_appreq, enum HrtfRequestMode hrtf_userreq);
void aluInitEffectPanning(struct ALeffectslot *slot);
void aluSelectPostProcess(ALCdevice *device);
/**
* Calculates ambisonic encoder coefficients using the X, Y, and Z direction
* components, which must represent a normalized (unit length) vector, and the
* spread is the angular width of the sound (0...tau).
*
* NOTE: The components use ambisonic coordinates. As a result:
*
* Ambisonic Y = OpenAL -X
* Ambisonic Z = OpenAL Y
* Ambisonic X = OpenAL -Z
*
* The components are ordered such that OpenAL's X, Y, and Z are the first,
* second, and third parameters respectively -- simply negate X and Z.
*/
void CalcAmbiCoeffs(const ALfloat y, const ALfloat z, const ALfloat x, const ALfloat spread,
ALfloat coeffs[MAX_AMBI_COEFFS]);
/**
* CalcDirectionCoeffs
*
* Calculates ambisonic coefficients based on an OpenAL direction vector. The
* vector must be normalized (unit length), and the spread is the angular width
* of the sound (0...tau).
*/
inline void CalcDirectionCoeffs(const ALfloat dir[3], ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS])
{
/* Convert from OpenAL coords to Ambisonics. */
CalcAmbiCoeffs(-dir[0], dir[1], -dir[2], spread, coeffs);
}
/**
* CalcAngleCoeffs
*
* Calculates ambisonic coefficients based on azimuth and elevation. The
* azimuth and elevation parameters are in radians, going right and up
* respectively.
*/
inline void CalcAngleCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS])
{
ALfloat x = -sinf(azimuth) * cosf(elevation);
ALfloat y = sinf(elevation);
ALfloat z = cosf(azimuth) * cosf(elevation);
CalcAmbiCoeffs(x, y, z, spread, coeffs);
}
/**
* ScaleAzimuthFront
*
* Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
* front.
*/
inline float ScaleAzimuthFront(float azimuth, float scale)
{
ALfloat sign = copysignf(1.0f, azimuth);
if(!(fabsf(azimuth) > F_PI_2))
return minf(fabsf(azimuth) * scale, F_PI_2) * sign;
return azimuth;
}
void ComputePanningGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALsizei numcoeffs, const ALfloat*RESTRICT coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
void ComputePanningGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat*RESTRICT coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
/**
* ComputePanGains
*
* Computes panning gains using the given channel decoder coefficients and the
* pre-calculated direction or angle coefficients. For B-Format sources, the
* coeffs are a 'slice' of a transform matrix for the input channel, used to
* scale and orient the sound samples.
*/
inline void ComputePanGains(const MixParams *dry, const ALfloat*RESTRICT coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS])
{
if(dry->CoeffCount > 0)
ComputePanningGainsMC(dry->Ambi.Coeffs, dry->NumChannels, dry->CoeffCount,
coeffs, ingain, gains);
else
ComputePanningGainsBF(dry->Ambi.Map, dry->NumChannels, coeffs, ingain, gains);
}
ALboolean MixSource(struct ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo);
void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples);
/* Caller must lock the device, and the mixer must not be running. */
void aluHandleDisconnect(ALCdevice *device, const char *msg, ...) DECL_FORMAT(printf, 2, 3);
extern MixerFunc MixSamples;
extern RowMixerFunc MixRowSamples;
extern ALfloat ConeScale;
extern ALfloat ZScale;
extern ALboolean OverrideReverbSpeedOfSound;
#ifdef __cplusplus
}
#endif
#endif