#ifndef _ALU_H_ #define _ALU_H_ #include #include #ifdef HAVE_FLOAT_H #include #endif #ifdef HAVE_IEEEFP_H #include #endif #include #include #include "alMain.h" #include "alBuffer.h" #include "hrtf.h" #include "math_defs.h" #include "filters/biquad.h" #include "filters/nfc.h" #include "almalloc.h" enum class DistanceModel; #define MAX_PITCH 255 #define MAX_SENDS 16 /* 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 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 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< next{nullptr}; DEF_NEWDEL(ALvoiceProps) }; #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) struct ALvoice { std::atomic Update{nullptr}; std::atomic SourceID{0u}; std::atomic Playing{false}; ALvoicePropsBase Props; /** * Source offset in samples, relative to the currently playing buffer, NOT * the whole queue, and the fractional (fixed-point) offset to the next * sample. */ std::atomic position; std::atomic position_fraction; /* Current buffer queue item being played. */ std::atomic current_buffer; /* Buffer queue item to loop to at end of queue (will be NULL for non- * looping voices). */ std::atomic 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) std::array,MAX_INPUT_CHANNELS> PrevSamples; InterpState ResampleState; struct { int FilterType; DirectParams Params[MAX_INPUT_CHANNELS]; ALfloat (*Buffer)[BUFFERSIZE]; ALsizei Channels; ALsizei ChannelsPerOrder[MAX_AMBI_ORDER+1]; } Direct; struct SendData { int FilterType; SendParams Params[MAX_INPUT_CHANNELS]; ALfloat (*Buffer)[BUFFERSIZE]; ALsizei Channels; } Send[]; }; void DeinitVoice(ALvoice *voice) noexcept; using MixerFunc = void(*)(const ALfloat *data, ALsizei OutChans, ALfloat (*RESTRICT OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos, ALsizei BufferSize); using RowMixerFunc = void(*)(ALfloat *OutBuffer, const ALfloat *gains, const ALfloat (*RESTRICT data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize); using HrtfMixerFunc = void(*)(ALfloat *RESTRICT LeftOut, ALfloat *RESTRICT RightOut, const ALfloat *data, ALsizei Offset, ALsizei OutPos, const ALsizei IrSize, MixHrtfParams *hrtfparams, HrtfState *hrtfstate, ALsizei BufferSize); using HrtfMixerBlendFunc = void(*)(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); using HrtfDirectMixerFunc = void(*)(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 (1000.0f) /* +60dB */ #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< b) ? b : a); } inline ALfloat maxf(ALfloat a, ALfloat b) noexcept { return ((a > b) ? a : b); } inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max) noexcept { return minf(max, maxf(min, val)); } inline ALdouble mind(ALdouble a, ALdouble b) noexcept { return ((a > b) ? b : a); } inline ALdouble maxd(ALdouble a, ALdouble b) noexcept { return ((a > b) ? a : b); } inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max) noexcept { return mind(max, maxd(min, val)); } inline ALuint minu(ALuint a, ALuint b) noexcept { return ((a > b) ? b : a); } inline ALuint maxu(ALuint a, ALuint b) noexcept { return ((a > b) ? a : b); } inline ALuint clampu(ALuint val, ALuint min, ALuint max) noexcept { return minu(max, maxu(min, val)); } inline ALint mini(ALint a, ALint b) noexcept { return ((a > b) ? b : a); } inline ALint maxi(ALint a, ALint b) noexcept { return ((a > b) ? a : b); } inline ALint clampi(ALint val, ALint min, ALint max) noexcept { return mini(max, maxi(min, val)); } inline ALint64 mini64(ALint64 a, ALint64 b) noexcept { return ((a > b) ? b : a); } inline ALint64 maxi64(ALint64 a, ALint64 b) noexcept { return ((a > b) ? a : b); } inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max) noexcept { return mini64(max, maxi64(min, val)); } inline ALuint64 minu64(ALuint64 a, ALuint64 b) noexcept { return ((a > b) ? b : a); } inline ALuint64 maxu64(ALuint64 a, ALuint64 b) noexcept { return ((a > b) ? a : b); } inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max) noexcept { return minu64(max, maxu64(min, val)); } inline size_t minz(size_t a, size_t b) noexcept { return ((a > b) ? b : a); } inline size_t maxz(size_t a, size_t b) noexcept { return ((a > b) ? a : b); } inline size_t clampz(size_t val, size_t min, size_t max) noexcept { return minz(max, maxz(min, val)); } inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu) noexcept { return val1 + (val2-val1)*mu; } inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu) noexcept { 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(Resampler resampler); /* aluInitRenderer * * Set up the appropriate panning method and mixing method given the device * properties. */ void aluInitRenderer(ALCdevice *device, ALint hrtf_id, HrtfRequestMode hrtf_appreq, HrtfRequestMode hrtf_userreq); void aluInitEffectPanning(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 = -std::sin(azimuth) * std::cos(elevation); ALfloat y = std::sin(elevation); ALfloat z = std::cos(azimuth) * std::cos(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 = std::copysign(1.0f, azimuth); if(!(std::fabs(azimuth) > F_PI_2)) return minf(std::fabs(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); } void ComputePanGains(const ALeffectslot *slot, const ALfloat*RESTRICT coeffs, ALfloat ingain, ALfloat (&gains)[MAX_OUTPUT_CHANNELS]); ALboolean MixSource(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 const ALfloat ConeScale; extern const ALfloat ZScale; extern const ALboolean OverrideReverbSpeedOfSound; #endif