/** * Reverb for the OpenAL cross platform audio library * Copyright (C) 2008-2009 by Christopher Fitzgerald. * 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., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "alAuxEffectSlot.h" #include "alEffect.h" #include "alFilter.h" #include "alError.h" /* This is the maximum number of samples processed for each inner loop * iteration. */ #define MAX_UPDATE_SAMPLES 256 typedef struct DelayLine { // The delay lines use sample lengths that are powers of 2 to allow the // use of bit-masking instead of a modulus for wrapping. ALuint Mask; ALfloat *Line; } DelayLine; typedef struct ALreverbState { DERIVE_FROM_TYPE(ALeffectState); ALboolean IsEax; ALuint ExtraChannels; // For HRTF // All delay lines are allocated as a single buffer to reduce memory // fragmentation and management code. ALfloat *SampleBuffer; ALuint TotalSamples; // Master effect filters ALfilterState LpFilter; ALfilterState HpFilter; // EAX only struct { // Modulator delay line. DelayLine Delay; // The vibrato time is tracked with an index over a modulus-wrapped // range (in samples). ALuint Index; ALuint Range; // The depth of frequency change (also in samples) and its filter. ALfloat Depth; ALfloat Coeff; ALfloat Filter; } Mod; // EAX only // Initial effect delay. DelayLine Delay; // The tap points for the initial delay. First tap goes to early // reflections, the last to late reverb. ALuint DelayTap[2]; struct { // Early reflections are done with 4 delay lines. ALfloat Coeff[4]; DelayLine Delay[4]; ALuint Offset[4]; // The gain for each output channel based on 3D panning. // NOTE: With certain output modes, we may be rendering to the dry // buffer and the "real" buffer. The two combined may be using more // than the max output channels, so we need some extra for the real // output too. ALfloat PanGain[4][MAX_OUTPUT_CHANNELS*2]; } Early; // Decorrelator delay line. DelayLine Decorrelator; // There are actually 4 decorrelator taps, but the first occurs at the // initial sample. ALuint DecoTap[3]; struct { // Output gain for late reverb. ALfloat Gain; // Attenuation to compensate for the modal density and decay rate of // the late lines. ALfloat DensityGain; // The feed-back and feed-forward all-pass coefficient. ALfloat ApFeedCoeff; // Mixing matrix coefficient. ALfloat MixCoeff; // Late reverb has 4 parallel all-pass filters. ALfloat ApCoeff[4]; DelayLine ApDelay[4]; ALuint ApOffset[4]; // In addition to 4 cyclical delay lines. ALfloat Coeff[4]; DelayLine Delay[4]; ALuint Offset[4]; // The cyclical delay lines are 1-pole low-pass filtered. ALfloat LpCoeff[4]; ALfloat LpSample[4]; // The gain for each output channel based on 3D panning. // NOTE: Add some extra in case (see note about early pan). ALfloat PanGain[4][MAX_OUTPUT_CHANNELS*2]; } Late; struct { // Attenuation to compensate for the modal density and decay rate of // the echo line. ALfloat DensityGain; // Echo delay and all-pass lines. DelayLine Delay; DelayLine ApDelay; ALfloat Coeff; ALfloat ApFeedCoeff; ALfloat ApCoeff; ALuint Offset; ALuint ApOffset; // The echo line is 1-pole low-pass filtered. ALfloat LpCoeff; ALfloat LpSample; // Echo mixing coefficient. ALfloat MixCoeff; } Echo; // EAX only // The current read offset for all delay lines. ALuint Offset; /* Temporary storage used when processing. */ ALfloat ReverbSamples[MAX_UPDATE_SAMPLES][4]; ALfloat EarlySamples[MAX_UPDATE_SAMPLES][4]; } ALreverbState; static ALvoid ALreverbState_Destruct(ALreverbState *State) { free(State->SampleBuffer); State->SampleBuffer = NULL; ALeffectState_Destruct(STATIC_CAST(ALeffectState,State)); } static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device); static ALvoid ALreverbState_update(ALreverbState *State, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props); static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels); static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels); static ALvoid ALreverbState_process(ALreverbState *State, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALreverbState) DEFINE_ALEFFECTSTATE_VTABLE(ALreverbState); /* This is a user config option for modifying the overall output of the reverb * effect. */ ALfloat ReverbBoost = 1.0f; /* Specifies whether to use a standard reverb effect in place of EAX reverb (no * high-pass, modulation, or echo). */ ALboolean EmulateEAXReverb = AL_FALSE; /* This coefficient is used to define the maximum frequency range controlled * by the modulation depth. The current value of 0.1 will allow it to swing * from 0.9x to 1.1x. This value must be below 1. At 1 it will cause the * sampler to stall on the downswing, and above 1 it will cause it to sample * backwards. */ static const ALfloat MODULATION_DEPTH_COEFF = 0.1f; /* A filter is used to avoid the terrible distortion caused by changing * modulation time and/or depth. To be consistent across different sample * rates, the coefficient must be raised to a constant divided by the sample * rate: coeff^(constant / rate). */ static const ALfloat MODULATION_FILTER_COEFF = 0.048f; static const ALfloat MODULATION_FILTER_CONST = 100000.0f; // When diffusion is above 0, an all-pass filter is used to take the edge off // the echo effect. It uses the following line length (in seconds). static const ALfloat ECHO_ALLPASS_LENGTH = 0.0133f; // Input into the late reverb is decorrelated between four channels. Their // timings are dependent on a fraction and multiplier. See the // UpdateDecorrelator() routine for the calculations involved. static const ALfloat DECO_FRACTION = 0.15f; static const ALfloat DECO_MULTIPLIER = 2.0f; // All delay line lengths are specified in seconds. // The lengths of the early delay lines. static const ALfloat EARLY_LINE_LENGTH[4] = { 0.0015f, 0.0045f, 0.0135f, 0.0405f }; // The lengths of the late all-pass delay lines. static const ALfloat ALLPASS_LINE_LENGTH[4] = { 0.0151f, 0.0167f, 0.0183f, 0.0200f, }; // The lengths of the late cyclical delay lines. static const ALfloat LATE_LINE_LENGTH[4] = { 0.0211f, 0.0311f, 0.0461f, 0.0680f }; // The late cyclical delay lines have a variable length dependent on the // effect's density parameter (inverted for some reason) and this multiplier. static const ALfloat LATE_LINE_MULTIPLIER = 4.0f; #if defined(_WIN32) && !defined (_M_X64) && !defined(_M_ARM) /* HACK: Workaround for a modff bug in 32-bit Windows, which attempts to write * a 64-bit double to the 32-bit float parameter. */ static inline float hack_modff(float x, float *y) { double di; double df = modf((double)x, &di); *y = (float)di; return (float)df; } #define modff hack_modff #endif /************************************** * Device Update * **************************************/ // Given the allocated sample buffer, this function updates each delay line // offset. static inline ALvoid RealizeLineOffset(ALfloat *sampleBuffer, DelayLine *Delay) { Delay->Line = &sampleBuffer[(ptrdiff_t)Delay->Line]; } // Calculate the length of a delay line and store its mask and offset. static ALuint CalcLineLength(ALfloat length, ptrdiff_t offset, ALuint frequency, ALuint extra, DelayLine *Delay) { ALuint samples; // All line lengths are powers of 2, calculated from their lengths, with // an additional sample in case of rounding errors. samples = fastf2u(length*frequency) + extra; samples = NextPowerOf2(samples + 1); // All lines share a single sample buffer. Delay->Mask = samples - 1; Delay->Line = (ALfloat*)offset; // Return the sample count for accumulation. return samples; } /* Calculates the delay line metrics and allocates the shared sample buffer * for all lines given the sample rate (frequency). If an allocation failure * occurs, it returns AL_FALSE. */ static ALboolean AllocLines(ALuint frequency, ALreverbState *State) { ALuint totalSamples, index; ALfloat length; ALfloat *newBuffer = NULL; // All delay line lengths are calculated to accomodate the full range of // lengths given their respective paramters. totalSamples = 0; /* The modulator's line length is calculated from the maximum modulation * time and depth coefficient, and halfed for the low-to-high frequency * swing. An additional sample is added to keep it stable when there is no * modulation. */ length = (AL_EAXREVERB_MAX_MODULATION_TIME*MODULATION_DEPTH_COEFF/2.0f); totalSamples += CalcLineLength(length, totalSamples, frequency, 1, &State->Mod.Delay); // The initial delay is the sum of the reflections and late reverb // delays. This must include space for storing a loop update to feed the // early reflections, decorrelator, and echo. length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY + AL_EAXREVERB_MAX_LATE_REVERB_DELAY; totalSamples += CalcLineLength(length, totalSamples, frequency, MAX_UPDATE_SAMPLES, &State->Delay); // The early reflection lines. for(index = 0;index < 4;index++) totalSamples += CalcLineLength(EARLY_LINE_LENGTH[index], totalSamples, frequency, 0, &State->Early.Delay[index]); // The decorrelator line is calculated from the lowest reverb density (a // parameter value of 1). This must include space for storing a loop update // to feed the late reverb. length = (DECO_FRACTION * DECO_MULTIPLIER * DECO_MULTIPLIER) * LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER); totalSamples += CalcLineLength(length, totalSamples, frequency, MAX_UPDATE_SAMPLES, &State->Decorrelator); // The late all-pass lines. for(index = 0;index < 4;index++) totalSamples += CalcLineLength(ALLPASS_LINE_LENGTH[index], totalSamples, frequency, 0, &State->Late.ApDelay[index]); // The late delay lines are calculated from the lowest reverb density. for(index = 0;index < 4;index++) { length = LATE_LINE_LENGTH[index] * (1.0f + LATE_LINE_MULTIPLIER); totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Late.Delay[index]); } // The echo all-pass and delay lines. totalSamples += CalcLineLength(ECHO_ALLPASS_LENGTH, totalSamples, frequency, 0, &State->Echo.ApDelay); totalSamples += CalcLineLength(AL_EAXREVERB_MAX_ECHO_TIME, totalSamples, frequency, 0, &State->Echo.Delay); if(totalSamples != State->TotalSamples) { TRACE("New reverb buffer length: %u samples (%f sec)\n", totalSamples, totalSamples/(float)frequency); newBuffer = realloc(State->SampleBuffer, sizeof(ALfloat) * totalSamples); if(newBuffer == NULL) return AL_FALSE; State->SampleBuffer = newBuffer; State->TotalSamples = totalSamples; } // Update all delays to reflect the new sample buffer. RealizeLineOffset(State->SampleBuffer, &State->Delay); RealizeLineOffset(State->SampleBuffer, &State->Decorrelator); for(index = 0;index < 4;index++) { RealizeLineOffset(State->SampleBuffer, &State->Early.Delay[index]); RealizeLineOffset(State->SampleBuffer, &State->Late.ApDelay[index]); RealizeLineOffset(State->SampleBuffer, &State->Late.Delay[index]); } RealizeLineOffset(State->SampleBuffer, &State->Mod.Delay); RealizeLineOffset(State->SampleBuffer, &State->Echo.ApDelay); RealizeLineOffset(State->SampleBuffer, &State->Echo.Delay); // Clear the sample buffer. for(index = 0;index < State->TotalSamples;index++) State->SampleBuffer[index] = 0.0f; return AL_TRUE; } static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device) { ALuint frequency = Device->Frequency, index; // Allocate the delay lines. if(!AllocLines(frequency, State)) return AL_FALSE; /* WARNING: This assumes the real output follows the virtual output in the * device's DryBuffer. */ if(Device->Hrtf || Device->Uhj_Encoder) State->ExtraChannels = ChannelsFromDevFmt(Device->FmtChans); else State->ExtraChannels = 0; // Calculate the modulation filter coefficient. Notice that the exponent // is calculated given the current sample rate. This ensures that the // resulting filter response over time is consistent across all sample // rates. State->Mod.Coeff = powf(MODULATION_FILTER_COEFF, MODULATION_FILTER_CONST / frequency); // The early reflection and late all-pass filter line lengths are static, // so their offsets only need to be calculated once. for(index = 0;index < 4;index++) { State->Early.Offset[index] = fastf2u(EARLY_LINE_LENGTH[index] * frequency); State->Late.ApOffset[index] = fastf2u(ALLPASS_LINE_LENGTH[index] * frequency); } // The echo all-pass filter line length is static, so its offset only // needs to be calculated once. State->Echo.ApOffset = fastf2u(ECHO_ALLPASS_LENGTH * frequency); return AL_TRUE; } /************************************** * Effect Update * **************************************/ // Calculate a decay coefficient given the length of each cycle and the time // until the decay reaches -60 dB. static inline ALfloat CalcDecayCoeff(ALfloat length, ALfloat decayTime) { return powf(0.001f/*-60 dB*/, length/decayTime); } // Calculate a decay length from a coefficient and the time until the decay // reaches -60 dB. static inline ALfloat CalcDecayLength(ALfloat coeff, ALfloat decayTime) { return log10f(coeff) * decayTime / log10f(0.001f)/*-60 dB*/; } // Calculate an attenuation to be applied to the input of any echo models to // compensate for modal density and decay time. static inline ALfloat CalcDensityGain(ALfloat a) { /* The energy of a signal can be obtained by finding the area under the * squared signal. This takes the form of Sum(x_n^2), where x is the * amplitude for the sample n. * * Decaying feedback matches exponential decay of the form Sum(a^n), * where a is the attenuation coefficient, and n is the sample. The area * under this decay curve can be calculated as: 1 / (1 - a). * * Modifying the above equation to find the squared area under the curve * (for energy) yields: 1 / (1 - a^2). Input attenuation can then be * calculated by inverting the square root of this approximation, * yielding: 1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2). */ return sqrtf(1.0f - (a * a)); } // Calculate the mixing matrix coefficients given a diffusion factor. static inline ALvoid CalcMatrixCoeffs(ALfloat diffusion, ALfloat *x, ALfloat *y) { ALfloat n, t; // The matrix is of order 4, so n is sqrt (4 - 1). n = sqrtf(3.0f); t = diffusion * atanf(n); // Calculate the first mixing matrix coefficient. *x = cosf(t); // Calculate the second mixing matrix coefficient. *y = sinf(t) / n; } // Calculate the limited HF ratio for use with the late reverb low-pass // filters. static ALfloat CalcLimitedHfRatio(ALfloat hfRatio, ALfloat airAbsorptionGainHF, ALfloat decayTime) { ALfloat limitRatio; /* Find the attenuation due to air absorption in dB (converting delay * time to meters using the speed of sound). Then reversing the decay * equation, solve for HF ratio. The delay length is cancelled out of * the equation, so it can be calculated once for all lines. */ limitRatio = 1.0f / (CalcDecayLength(airAbsorptionGainHF, decayTime) * SPEEDOFSOUNDMETRESPERSEC); /* Using the limit calculated above, apply the upper bound to the HF * ratio. Also need to limit the result to a minimum of 0.1, just like the * HF ratio parameter. */ return clampf(limitRatio, 0.1f, hfRatio); } // Calculate the coefficient for a HF (and eventually LF) decay damping // filter. static inline ALfloat CalcDampingCoeff(ALfloat hfRatio, ALfloat length, ALfloat decayTime, ALfloat decayCoeff, ALfloat cw) { ALfloat coeff, g; // Eventually this should boost the high frequencies when the ratio // exceeds 1. coeff = 0.0f; if (hfRatio < 1.0f) { // Calculate the low-pass coefficient by dividing the HF decay // coefficient by the full decay coefficient. g = CalcDecayCoeff(length, decayTime * hfRatio) / decayCoeff; // Damping is done with a 1-pole filter, so g needs to be squared. g *= g; if(g < 0.9999f) /* 1-epsilon */ { /* Be careful with gains < 0.001, as that causes the coefficient * head towards 1, which will flatten the signal. */ g = maxf(g, 0.001f); coeff = (1 - g*cw - sqrtf(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g); } // Very low decay times will produce minimal output, so apply an // upper bound to the coefficient. coeff = minf(coeff, 0.98f); } return coeff; } // Update the EAX modulation index, range, and depth. Keep in mind that this // kind of vibrato is additive and not multiplicative as one may expect. The // downswing will sound stronger than the upswing. static ALvoid UpdateModulator(ALfloat modTime, ALfloat modDepth, ALuint frequency, ALreverbState *State) { ALuint range; /* Modulation is calculated in two parts. * * The modulation time effects the sinus applied to the change in * frequency. An index out of the current time range (both in samples) * is incremented each sample. The range is bound to a reasonable * minimum (1 sample) and when the timing changes, the index is rescaled * to the new range (to keep the sinus consistent). */ range = maxu(fastf2u(modTime*frequency), 1); State->Mod.Index = (ALuint)(State->Mod.Index * (ALuint64)range / State->Mod.Range); State->Mod.Range = range; /* The modulation depth effects the amount of frequency change over the * range of the sinus. It needs to be scaled by the modulation time so * that a given depth produces a consistent change in frequency over all * ranges of time. Since the depth is applied to a sinus value, it needs * to be halfed once for the sinus range and again for the sinus swing * in time (half of it is spent decreasing the frequency, half is spent * increasing it). */ State->Mod.Depth = modDepth * MODULATION_DEPTH_COEFF * modTime / 2.0f / 2.0f * frequency; } // Update the offsets for the initial effect delay line. static ALvoid UpdateDelayLine(ALfloat earlyDelay, ALfloat lateDelay, ALuint frequency, ALreverbState *State) { // Calculate the initial delay taps. State->DelayTap[0] = fastf2u(earlyDelay * frequency); State->DelayTap[1] = fastf2u((earlyDelay + lateDelay) * frequency); } // Update the early reflections mix and line coefficients. static ALvoid UpdateEarlyLines(ALfloat lateDelay, ALreverbState *State) { ALuint index; // Calculate the gain (coefficient) for each early delay line using the // late delay time. This expands the early reflections to the start of // the late reverb. for(index = 0;index < 4;index++) State->Early.Coeff[index] = CalcDecayCoeff(EARLY_LINE_LENGTH[index], lateDelay); } // Update the offsets for the decorrelator line. static ALvoid UpdateDecorrelator(ALfloat density, ALuint frequency, ALreverbState *State) { ALuint index; ALfloat length; /* The late reverb inputs are decorrelated to smooth the reverb tail and * reduce harsh echos. The first tap occurs immediately, while the * remaining taps are delayed by multiples of a fraction of the smallest * cyclical delay time. * * offset[index] = (FRACTION (MULTIPLIER^index)) smallest_delay */ for(index = 0;index < 3;index++) { length = (DECO_FRACTION * powf(DECO_MULTIPLIER, (ALfloat)index)) * LATE_LINE_LENGTH[0] * (1.0f + (density * LATE_LINE_MULTIPLIER)); State->DecoTap[index] = fastf2u(length * frequency); } } // Update the late reverb mix, line lengths, and line coefficients. static ALvoid UpdateLateLines(ALfloat xMix, ALfloat density, ALfloat decayTime, ALfloat diffusion, ALfloat echoDepth, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALreverbState *State) { ALfloat length; ALuint index; /* Calculate the late reverb gain. Since the output is tapped prior to the * application of the next delay line coefficients, this gain needs to be * attenuated by the 'x' mixing matrix coefficient as well. Also attenuate * the late reverb when echo depth is high and diffusion is low, so the * echo is slightly stronger than the decorrelated echos in the reverb * tail. */ State->Late.Gain = xMix * (1.0f - (echoDepth*0.5f*(1.0f - diffusion))); /* To compensate for changes in modal density and decay time of the late * reverb signal, the input is attenuated based on the maximal energy of * the outgoing signal. This approximation is used to keep the apparent * energy of the signal equal for all ranges of density and decay time. * * The average length of the cyclcical delay lines is used to calculate * the attenuation coefficient. */ length = (LATE_LINE_LENGTH[0] + LATE_LINE_LENGTH[1] + LATE_LINE_LENGTH[2] + LATE_LINE_LENGTH[3]) / 4.0f; length *= 1.0f + (density * LATE_LINE_MULTIPLIER); State->Late.DensityGain = CalcDensityGain( CalcDecayCoeff(length, decayTime) ); // Calculate the all-pass feed-back and feed-forward coefficient. State->Late.ApFeedCoeff = 0.5f * powf(diffusion, 2.0f); for(index = 0;index < 4;index++) { // Calculate the gain (coefficient) for each all-pass line. State->Late.ApCoeff[index] = CalcDecayCoeff( ALLPASS_LINE_LENGTH[index], decayTime ); // Calculate the length (in seconds) of each cyclical delay line. length = LATE_LINE_LENGTH[index] * (1.0f + (density * LATE_LINE_MULTIPLIER)); // Calculate the delay offset for each cyclical delay line. State->Late.Offset[index] = fastf2u(length * frequency); // Calculate the gain (coefficient) for each cyclical line. State->Late.Coeff[index] = CalcDecayCoeff(length, decayTime); // Calculate the damping coefficient for each low-pass filter. State->Late.LpCoeff[index] = CalcDampingCoeff( hfRatio, length, decayTime, State->Late.Coeff[index], cw ); // Attenuate the cyclical line coefficients by the mixing coefficient // (x). State->Late.Coeff[index] *= xMix; } } // Update the echo gain, line offset, line coefficients, and mixing // coefficients. static ALvoid UpdateEchoLine(ALfloat echoTime, ALfloat decayTime, ALfloat diffusion, ALfloat echoDepth, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALreverbState *State) { // Update the offset and coefficient for the echo delay line. State->Echo.Offset = fastf2u(echoTime * frequency); // Calculate the decay coefficient for the echo line. State->Echo.Coeff = CalcDecayCoeff(echoTime, decayTime); // Calculate the energy-based attenuation coefficient for the echo delay // line. State->Echo.DensityGain = CalcDensityGain(State->Echo.Coeff); // Calculate the echo all-pass feed coefficient. State->Echo.ApFeedCoeff = 0.5f * powf(diffusion, 2.0f); // Calculate the echo all-pass attenuation coefficient. State->Echo.ApCoeff = CalcDecayCoeff(ECHO_ALLPASS_LENGTH, decayTime); // Calculate the damping coefficient for each low-pass filter. State->Echo.LpCoeff = CalcDampingCoeff(hfRatio, echoTime, decayTime, State->Echo.Coeff, cw); /* Calculate the echo mixing coefficient. This is applied to the output mix * only, not the feedback. */ State->Echo.MixCoeff = echoDepth; } // Update the early and late 3D panning gains. static ALvoid UpdateMixedPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALfloat EarlyGain, ALfloat LateGain, ALreverbState *State) { ALfloat DirGains[MAX_OUTPUT_CHANNELS]; ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat length; ALuint i; /* With HRTF or UHJ, the normal output provides a panned reverb channel * when a non-0-length vector is specified, while the real stereo output * provides two other "direct" non-panned reverb channels. * * WARNING: This assumes the real output follows the virtual output in the * device's DryBuffer. */ memset(State->Early.PanGain, 0, sizeof(State->Early.PanGain)); length = sqrtf(ReflectionsPan[0]*ReflectionsPan[0] + ReflectionsPan[1]*ReflectionsPan[1] + ReflectionsPan[2]*ReflectionsPan[2]); if(!(length > FLT_EPSILON)) { for(i = 0;i < Device->RealOut.NumChannels;i++) State->Early.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * EarlyGain; } else { /* Note that EAX Reverb's panning vectors are using right-handed * coordinates, rather that the OpenAL's left-handed coordinates. * Negate Z to fix this. */ ALfloat pan[3] = { ReflectionsPan[0] / length, ReflectionsPan[1] / length, -ReflectionsPan[2] / length, }; length = minf(length, 1.0f); CalcDirectionCoeffs(pan, 0.0f, coeffs); ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); for(i = 0;i < Device->Dry.NumChannels;i++) State->Early.PanGain[3][i] = DirGains[i] * EarlyGain * length; for(i = 0;i < Device->RealOut.NumChannels;i++) State->Early.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * EarlyGain * (1.0f-length); } memset(State->Late.PanGain, 0, sizeof(State->Late.PanGain)); length = sqrtf(LateReverbPan[0]*LateReverbPan[0] + LateReverbPan[1]*LateReverbPan[1] + LateReverbPan[2]*LateReverbPan[2]); if(!(length > FLT_EPSILON)) { for(i = 0;i < Device->RealOut.NumChannels;i++) State->Late.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * LateGain; } else { ALfloat pan[3] = { LateReverbPan[0] / length, LateReverbPan[1] / length, -LateReverbPan[2] / length, }; length = minf(length, 1.0f); CalcDirectionCoeffs(pan, 0.0f, coeffs); ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); for(i = 0;i < Device->Dry.NumChannels;i++) State->Late.PanGain[3][i] = DirGains[i] * LateGain * length; for(i = 0;i < Device->RealOut.NumChannels;i++) State->Late.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * LateGain * (1.0f-length); } } static ALvoid UpdateDirectPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALfloat EarlyGain, ALfloat LateGain, ALreverbState *State) { ALfloat AmbientGains[MAX_OUTPUT_CHANNELS]; ALfloat DirGains[MAX_OUTPUT_CHANNELS]; ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat length; ALuint i; /* Apply a boost of about 3dB to better match the expected stereo output volume. */ ComputeAmbientGains(Device->Dry, Gain*1.414213562f, AmbientGains); memset(State->Early.PanGain, 0, sizeof(State->Early.PanGain)); length = sqrtf(ReflectionsPan[0]*ReflectionsPan[0] + ReflectionsPan[1]*ReflectionsPan[1] + ReflectionsPan[2]*ReflectionsPan[2]); if(!(length > FLT_EPSILON)) { for(i = 0;i < Device->Dry.NumChannels;i++) State->Early.PanGain[i&3][i] = AmbientGains[i] * EarlyGain; } else { /* Note that EAX Reverb's panning vectors are using right-handed * coordinates, rather that the OpenAL's left-handed coordinates. * Negate Z to fix this. */ ALfloat pan[3] = { ReflectionsPan[0] / length, ReflectionsPan[1] / length, -ReflectionsPan[2] / length, }; length = minf(length, 1.0f); CalcDirectionCoeffs(pan, 0.0f, coeffs); ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); for(i = 0;i < Device->Dry.NumChannels;i++) State->Early.PanGain[i&3][i] = lerp(AmbientGains[i], DirGains[i], length) * EarlyGain; } memset(State->Late.PanGain, 0, sizeof(State->Late.PanGain)); length = sqrtf(LateReverbPan[0]*LateReverbPan[0] + LateReverbPan[1]*LateReverbPan[1] + LateReverbPan[2]*LateReverbPan[2]); if(!(length > FLT_EPSILON)) { for(i = 0;i < Device->Dry.NumChannels;i++) State->Late.PanGain[i&3][i] = AmbientGains[i] * LateGain; } else { ALfloat pan[3] = { LateReverbPan[0] / length, LateReverbPan[1] / length, -LateReverbPan[2] / length, }; length = minf(length, 1.0f); CalcDirectionCoeffs(pan, 0.0f, coeffs); ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); for(i = 0;i < Device->Dry.NumChannels;i++) State->Late.PanGain[i&3][i] = lerp(AmbientGains[i], DirGains[i], length) * LateGain; } } static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALfloat EarlyGain, ALfloat LateGain, ALreverbState *State) { static const ALfloat PanDirs[4][3] = { { -0.707106781f, 0.0f, -0.707106781f }, /* Front left */ { 0.707106781f, 0.0f, -0.707106781f }, /* Front right */ { 0.707106781f, 0.0f, 0.707106781f }, /* Back right */ { -0.707106781f, 0.0f, 0.707106781f } /* Back left */ }; ALfloat coeffs[MAX_AMBI_COEFFS]; ALfloat gain[4]; ALfloat length; ALuint i; /* sqrt(0.5) would be the gain scaling when the panning vector is 0. This * also equals sqrt(2/4), a nice gain scaling for the four virtual points * producing an "ambient" response. */ gain[0] = gain[1] = gain[2] = gain[3] = 0.707106781f; length = sqrtf(ReflectionsPan[0]*ReflectionsPan[0] + ReflectionsPan[1]*ReflectionsPan[1] + ReflectionsPan[2]*ReflectionsPan[2]); if(length > 1.0f) { ALfloat pan[3] = { ReflectionsPan[0] / length, ReflectionsPan[1] / length, -ReflectionsPan[2] / length, }; for(i = 0;i < 4;i++) { ALfloat dotp = pan[0]*PanDirs[i][0] + pan[1]*PanDirs[i][1] + pan[2]*PanDirs[i][2]; gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); } } else if(length > FLT_EPSILON) { for(i = 0;i < 4;i++) { ALfloat dotp = ReflectionsPan[0]*PanDirs[i][0] + ReflectionsPan[1]*PanDirs[i][1] + -ReflectionsPan[2]*PanDirs[i][2]; gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); } } for(i = 0;i < 4;i++) { CalcDirectionCoeffs(PanDirs[i], 0.0f, coeffs); ComputePanningGains(Device->Dry, coeffs, Gain*EarlyGain*gain[i], State->Early.PanGain[i]); } gain[0] = gain[1] = gain[2] = gain[3] = 0.707106781f; length = sqrtf(LateReverbPan[0]*LateReverbPan[0] + LateReverbPan[1]*LateReverbPan[1] + LateReverbPan[2]*LateReverbPan[2]); if(length > 1.0f) { ALfloat pan[3] = { LateReverbPan[0] / length, LateReverbPan[1] / length, -LateReverbPan[2] / length, }; for(i = 0;i < 4;i++) { ALfloat dotp = pan[0]*PanDirs[i][0] + pan[1]*PanDirs[i][1] + pan[2]*PanDirs[i][2]; gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); } } else if(length > FLT_EPSILON) { for(i = 0;i < 4;i++) { ALfloat dotp = LateReverbPan[0]*PanDirs[i][0] + LateReverbPan[1]*PanDirs[i][1] + -LateReverbPan[2]*PanDirs[i][2]; gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); } } for(i = 0;i < 4;i++) { CalcDirectionCoeffs(PanDirs[i], 0.0f, coeffs); ComputePanningGains(Device->Dry, coeffs, Gain*LateGain*gain[i], State->Late.PanGain[i]); } } static ALvoid ALreverbState_update(ALreverbState *State, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props) { ALuint frequency = Device->Frequency; ALfloat lfscale, hfscale, hfRatio; ALfloat gain, gainlf, gainhf; ALfloat cw, x, y; if(Slot->Params.EffectType == AL_EFFECT_EAXREVERB && !EmulateEAXReverb) State->IsEax = AL_TRUE; else if(Slot->Params.EffectType == AL_EFFECT_REVERB || EmulateEAXReverb) State->IsEax = AL_FALSE; // Calculate the master filters hfscale = props->Reverb.HFReference / frequency; gainhf = maxf(props->Reverb.GainHF, 0.0001f); ALfilterState_setParams(&State->LpFilter, ALfilterType_HighShelf, gainhf, hfscale, calc_rcpQ_from_slope(gainhf, 0.75f)); lfscale = props->Reverb.LFReference / frequency; gainlf = maxf(props->Reverb.GainLF, 0.0001f); ALfilterState_setParams(&State->HpFilter, ALfilterType_LowShelf, gainlf, lfscale, calc_rcpQ_from_slope(gainlf, 0.75f)); // Update the modulator line. UpdateModulator(props->Reverb.ModulationTime, props->Reverb.ModulationDepth, frequency, State); // Update the initial effect delay. UpdateDelayLine(props->Reverb.ReflectionsDelay, props->Reverb.LateReverbDelay, frequency, State); // Update the early lines. UpdateEarlyLines(props->Reverb.LateReverbDelay, State); // Update the decorrelator. UpdateDecorrelator(props->Reverb.Density, frequency, State); // Get the mixing matrix coefficients (x and y). CalcMatrixCoeffs(props->Reverb.Diffusion, &x, &y); // Then divide x into y to simplify the matrix calculation. State->Late.MixCoeff = y / x; // If the HF limit parameter is flagged, calculate an appropriate limit // based on the air absorption parameter. hfRatio = props->Reverb.DecayHFRatio; if(props->Reverb.DecayHFLimit && props->Reverb.AirAbsorptionGainHF < 1.0f) hfRatio = CalcLimitedHfRatio(hfRatio, props->Reverb.AirAbsorptionGainHF, props->Reverb.DecayTime); cw = cosf(F_TAU * hfscale); // Update the late lines. UpdateLateLines(x, props->Reverb.Density, props->Reverb.DecayTime, props->Reverb.Diffusion, props->Reverb.EchoDepth, hfRatio, cw, frequency, State); // Update the echo line. UpdateEchoLine(props->Reverb.EchoTime, props->Reverb.DecayTime, props->Reverb.Diffusion, props->Reverb.EchoDepth, hfRatio, cw, frequency, State); gain = props->Reverb.Gain * Slot->Params.Gain * ReverbBoost; // Update early and late 3D panning. if(Device->Hrtf || Device->Uhj_Encoder) UpdateMixedPanning(Device, props->Reverb.ReflectionsPan, props->Reverb.LateReverbPan, gain, props->Reverb.ReflectionsGain, props->Reverb.LateReverbGain, State); else if(Device->FmtChans == DevFmtBFormat3D || Device->AmbiDecoder) Update3DPanning(Device, props->Reverb.ReflectionsPan, props->Reverb.LateReverbPan, gain, props->Reverb.ReflectionsGain, props->Reverb.LateReverbGain, State); else UpdateDirectPanning(Device, props->Reverb.ReflectionsPan, props->Reverb.LateReverbPan, gain, props->Reverb.ReflectionsGain, props->Reverb.LateReverbGain, State); } /************************************** * Effect Processing * **************************************/ // Basic delay line input/output routines. static inline ALfloat DelayLineOut(DelayLine *Delay, ALuint offset) { return Delay->Line[offset&Delay->Mask]; } static inline ALvoid DelayLineIn(DelayLine *Delay, ALuint offset, ALfloat in) { Delay->Line[offset&Delay->Mask] = in; } // Given an input sample, this function produces modulation for the late // reverb. static inline ALfloat EAXModulation(ALreverbState *State, ALuint offset, ALfloat in) { ALfloat sinus, frac, fdelay; ALfloat out0, out1; ALuint delay; // Calculate the sinus rythm (dependent on modulation time and the // sampling rate). The center of the sinus is moved to reduce the delay // of the effect when the time or depth are low. sinus = 1.0f - cosf(F_TAU * State->Mod.Index / State->Mod.Range); // Step the modulation index forward, keeping it bound to its range. State->Mod.Index = (State->Mod.Index + 1) % State->Mod.Range; // The depth determines the range over which to read the input samples // from, so it must be filtered to reduce the distortion caused by even // small parameter changes. State->Mod.Filter = lerp(State->Mod.Filter, State->Mod.Depth, State->Mod.Coeff); // Calculate the read offset and fraction between it and the next sample. frac = modff(State->Mod.Filter*sinus, &fdelay); delay = fastf2u(fdelay); /* Add the incoming sample to the delay line first, so a 0 delay gets the * incoming sample. */ DelayLineIn(&State->Mod.Delay, offset, in); /* Get the two samples crossed by the offset delay */ out0 = DelayLineOut(&State->Mod.Delay, offset - delay); out1 = DelayLineOut(&State->Mod.Delay, offset - delay - 1); // The output is obtained by linearly interpolating the two samples that // were acquired above. return lerp(out0, out1, frac); } // Given some input sample, this function produces four-channel outputs for the // early reflections. static inline ALvoid EarlyReflection(ALreverbState *State, ALuint todo, ALfloat (*restrict out)[4]) { ALfloat d[4], v, f[4]; ALuint i; for(i = 0;i < todo;i++) { ALuint offset = State->Offset+i; // Obtain the decayed results of each early delay line. d[0] = DelayLineOut(&State->Early.Delay[0], offset-State->Early.Offset[0]) * State->Early.Coeff[0]; d[1] = DelayLineOut(&State->Early.Delay[1], offset-State->Early.Offset[1]) * State->Early.Coeff[1]; d[2] = DelayLineOut(&State->Early.Delay[2], offset-State->Early.Offset[2]) * State->Early.Coeff[2]; d[3] = DelayLineOut(&State->Early.Delay[3], offset-State->Early.Offset[3]) * State->Early.Coeff[3]; /* The following uses a lossless scattering junction from waveguide * theory. It actually amounts to a householder mixing matrix, which * will produce a maximally diffuse response, and means this can * probably be considered a simple feed-back delay network (FDN). * N * --- * \ * v = 2/N / d_i * --- * i=1 */ v = (d[0] + d[1] + d[2] + d[3]) * 0.5f; // The junction is loaded with the input here. v += DelayLineOut(&State->Delay, offset-State->DelayTap[0]); // Calculate the feed values for the delay lines. f[0] = v - d[0]; f[1] = v - d[1]; f[2] = v - d[2]; f[3] = v - d[3]; // Re-feed the delay lines. DelayLineIn(&State->Early.Delay[0], offset, f[0]); DelayLineIn(&State->Early.Delay[1], offset, f[1]); DelayLineIn(&State->Early.Delay[2], offset, f[2]); DelayLineIn(&State->Early.Delay[3], offset, f[3]); /* Output the results of the junction for all four channels with a * constant attenuation of 0.5. */ out[i][0] = f[0] * 0.5f; out[i][1] = f[1] * 0.5f; out[i][2] = f[2] * 0.5f; out[i][3] = f[3] * 0.5f; } } // Basic attenuated all-pass input/output routine. static inline ALfloat AllpassInOut(DelayLine *Delay, ALuint outOffset, ALuint inOffset, ALfloat in, ALfloat feedCoeff, ALfloat coeff) { ALfloat out, feed; out = DelayLineOut(Delay, outOffset); feed = feedCoeff * in; DelayLineIn(Delay, inOffset, (feedCoeff * (out - feed)) + in); // The time-based attenuation is only applied to the delay output to // keep it from affecting the feed-back path (which is already controlled // by the all-pass feed coefficient). return (coeff * out) - feed; } // All-pass input/output routine for late reverb. static inline ALfloat LateAllPassInOut(ALreverbState *State, ALuint offset, ALuint index, ALfloat in) { return AllpassInOut(&State->Late.ApDelay[index], offset - State->Late.ApOffset[index], offset, in, State->Late.ApFeedCoeff, State->Late.ApCoeff[index]); } // Low-pass filter input/output routine for late reverb. static inline ALfloat LateLowPassInOut(ALreverbState *State, ALuint index, ALfloat in) { in = lerp(in, State->Late.LpSample[index], State->Late.LpCoeff[index]); State->Late.LpSample[index] = in; return in; } // Given four decorrelated input samples, this function produces four-channel // output for the late reverb. static inline ALvoid LateReverb(ALreverbState *State, ALuint todo, ALfloat (*restrict out)[4]) { ALfloat d[4], f[4]; ALuint i; // Feed the decorrelator from the energy-attenuated output of the second // delay tap. for(i = 0;i < todo;i++) { ALuint offset = State->Offset+i; ALfloat sample = DelayLineOut(&State->Delay, offset - State->DelayTap[1]) * State->Late.DensityGain; DelayLineIn(&State->Decorrelator, offset, sample); } for(i = 0;i < todo;i++) { ALuint offset = State->Offset+i; /* Obtain four decorrelated input samples. */ f[0] = DelayLineOut(&State->Decorrelator, offset); f[1] = DelayLineOut(&State->Decorrelator, offset-State->DecoTap[0]); f[2] = DelayLineOut(&State->Decorrelator, offset-State->DecoTap[1]); f[3] = DelayLineOut(&State->Decorrelator, offset-State->DecoTap[2]); /* Add the decayed results of the cyclical delay lines, then pass the * results through the low-pass filters. */ f[0] += DelayLineOut(&State->Late.Delay[0], offset-State->Late.Offset[0]) * State->Late.Coeff[0]; f[1] += DelayLineOut(&State->Late.Delay[1], offset-State->Late.Offset[1]) * State->Late.Coeff[1]; f[2] += DelayLineOut(&State->Late.Delay[2], offset-State->Late.Offset[2]) * State->Late.Coeff[2]; f[3] += DelayLineOut(&State->Late.Delay[3], offset-State->Late.Offset[3]) * State->Late.Coeff[3]; // This is where the feed-back cycles from line 0 to 1 to 3 to 2 and // back to 0. d[0] = LateLowPassInOut(State, 2, f[2]); d[1] = LateLowPassInOut(State, 0, f[0]); d[2] = LateLowPassInOut(State, 3, f[3]); d[3] = LateLowPassInOut(State, 1, f[1]); // To help increase diffusion, run each line through an all-pass filter. // When there is no diffusion, the shortest all-pass filter will feed // the shortest delay line. d[0] = LateAllPassInOut(State, offset, 0, d[0]); d[1] = LateAllPassInOut(State, offset, 1, d[1]); d[2] = LateAllPassInOut(State, offset, 2, d[2]); d[3] = LateAllPassInOut(State, offset, 3, d[3]); /* Late reverb is done with a modified feed-back delay network (FDN) * topology. Four input lines are each fed through their own all-pass * filter and then into the mixing matrix. The four outputs of the * mixing matrix are then cycled back to the inputs. Each output feeds * a different input to form a circlular feed cycle. * * The mixing matrix used is a 4D skew-symmetric rotation matrix * derived using a single unitary rotational parameter: * * [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2 * [ -a, d, c, -b ] * [ -b, -c, d, a ] * [ -c, b, -a, d ] * * The rotation is constructed from the effect's diffusion parameter, * yielding: 1 = x^2 + 3 y^2; where a, b, and c are the coefficient y * with differing signs, and d is the coefficient x. The matrix is * thus: * * [ x, y, -y, y ] n = sqrt(matrix_order - 1) * [ -y, x, y, y ] t = diffusion_parameter * atan(n) * [ y, -y, x, y ] x = cos(t) * [ -y, -y, -y, x ] y = sin(t) / n * * To reduce the number of multiplies, the x coefficient is applied * with the cyclical delay line coefficients. Thus only the y * coefficient is applied when mixing, and is modified to be: y / x. */ f[0] = d[0] + (State->Late.MixCoeff * ( d[1] + -d[2] + d[3])); f[1] = d[1] + (State->Late.MixCoeff * (-d[0] + d[2] + d[3])); f[2] = d[2] + (State->Late.MixCoeff * ( d[0] + -d[1] + d[3])); f[3] = d[3] + (State->Late.MixCoeff * (-d[0] + -d[1] + -d[2] )); // Output the results of the matrix for all four channels, attenuated by // the late reverb gain (which is attenuated by the 'x' mix coefficient). out[i][0] = State->Late.Gain * f[0]; out[i][1] = State->Late.Gain * f[1]; out[i][2] = State->Late.Gain * f[2]; out[i][3] = State->Late.Gain * f[3]; // Re-feed the cyclical delay lines. DelayLineIn(&State->Late.Delay[0], offset, f[0]); DelayLineIn(&State->Late.Delay[1], offset, f[1]); DelayLineIn(&State->Late.Delay[2], offset, f[2]); DelayLineIn(&State->Late.Delay[3], offset, f[3]); } } // Given an input sample, this function mixes echo into the four-channel late // reverb. static inline ALvoid EAXEcho(ALreverbState *State, ALuint todo, ALfloat (*restrict late)[4]) { ALfloat out, feed; ALuint i; for(i = 0;i < todo;i++) { ALuint offset = State->Offset+i; // Get the latest attenuated echo sample for output. feed = DelayLineOut(&State->Echo.Delay, offset-State->Echo.Offset) * State->Echo.Coeff; // Mix the output into the late reverb channels. out = State->Echo.MixCoeff * feed; late[i][0] += out; late[i][1] += out; late[i][2] += out; late[i][3] += out; // Mix the energy-attenuated input with the output and pass it through // the echo low-pass filter. feed += DelayLineOut(&State->Delay, offset-State->DelayTap[1]) * State->Echo.DensityGain; feed = lerp(feed, State->Echo.LpSample, State->Echo.LpCoeff); State->Echo.LpSample = feed; // Then the echo all-pass filter. feed = AllpassInOut(&State->Echo.ApDelay, offset-State->Echo.ApOffset, offset, feed, State->Echo.ApFeedCoeff, State->Echo.ApCoeff); // Feed the delay with the mixed and filtered sample. DelayLineIn(&State->Echo.Delay, offset, feed); } } // Perform the non-EAX reverb pass on a given input sample, resulting in // four-channel output. static inline ALvoid VerbPass(ALreverbState *State, ALuint todo, const ALfloat *in, ALfloat (*restrict early)[4], ALfloat (*restrict late)[4]) { ALuint i; // Low-pass filter the incoming samples. for(i = 0;i < todo;i++) DelayLineIn(&State->Delay, State->Offset+i, ALfilterState_processSingle(&State->LpFilter, in[i]) ); // Calculate the early reflection from the first delay tap. EarlyReflection(State, todo, early); // Calculate the late reverb from the decorrelator taps. LateReverb(State, todo, late); // Step all delays forward one sample. State->Offset += todo; } // Perform the EAX reverb pass on a given input sample, resulting in four- // channel output. static inline ALvoid EAXVerbPass(ALreverbState *State, ALuint todo, const ALfloat *input, ALfloat (*restrict early)[4], ALfloat (*restrict late)[4]) { ALuint i; // Band-pass and modulate the incoming samples. for(i = 0;i < todo;i++) { ALfloat sample = input[i]; sample = ALfilterState_processSingle(&State->LpFilter, sample); sample = ALfilterState_processSingle(&State->HpFilter, sample); // Perform any modulation on the input. sample = EAXModulation(State, State->Offset+i, sample); // Feed the initial delay line. DelayLineIn(&State->Delay, State->Offset+i, sample); } // Calculate the early reflection from the first delay tap. EarlyReflection(State, todo, early); // Calculate the late reverb from the decorrelator taps. LateReverb(State, todo, late); // Calculate and mix in any echo. EAXEcho(State, todo, late); // Step all delays forward. State->Offset += todo; } static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALfloat (*restrict early)[4] = State->EarlySamples; ALfloat (*restrict late)[4] = State->ReverbSamples; ALuint index, c, i, l; ALfloat gain; /* Process reverb for these samples. */ for(index = 0;index < SamplesToDo;) { ALuint todo = minu(SamplesToDo-index, MAX_UPDATE_SAMPLES); VerbPass(State, todo, &SamplesIn[index], early, late); for(l = 0;l < 4;l++) { for(c = 0;c < NumChannels;c++) { gain = State->Early.PanGain[l][c]; if(fabsf(gain) > GAIN_SILENCE_THRESHOLD) { for(i = 0;i < todo;i++) SamplesOut[c][index+i] += gain*early[i][l]; } gain = State->Late.PanGain[l][c]; if(fabsf(gain) > GAIN_SILENCE_THRESHOLD) { for(i = 0;i < todo;i++) SamplesOut[c][index+i] += gain*late[i][l]; } } } index += todo; } } static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALfloat (*restrict early)[4] = State->EarlySamples; ALfloat (*restrict late)[4] = State->ReverbSamples; ALuint index, c, i, l; ALfloat gain; /* Process reverb for these samples. */ for(index = 0;index < SamplesToDo;) { ALuint todo = minu(SamplesToDo-index, MAX_UPDATE_SAMPLES); EAXVerbPass(State, todo, &SamplesIn[index], early, late); for(l = 0;l < 4;l++) { for(c = 0;c < NumChannels;c++) { gain = State->Early.PanGain[l][c]; if(fabsf(gain) > GAIN_SILENCE_THRESHOLD) { for(i = 0;i < todo;i++) SamplesOut[c][index+i] += gain*early[i][l]; } gain = State->Late.PanGain[l][c]; if(fabsf(gain) > GAIN_SILENCE_THRESHOLD) { for(i = 0;i < todo;i++) SamplesOut[c][index+i] += gain*late[i][l]; } } } index += todo; } } static ALvoid ALreverbState_process(ALreverbState *State, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { NumChannels += State->ExtraChannels; if(State->IsEax) ALreverbState_processEax(State, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); else ALreverbState_processStandard(State, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); } typedef struct ALreverbStateFactory { DERIVE_FROM_TYPE(ALeffectStateFactory); } ALreverbStateFactory; static ALeffectState *ALreverbStateFactory_create(ALreverbStateFactory* UNUSED(factory)) { ALreverbState *state; ALuint index, l; state = ALreverbState_New(sizeof(*state)); if(!state) return NULL; SET_VTABLE2(ALreverbState, ALeffectState, state); state->IsEax = AL_FALSE; state->ExtraChannels = 0; state->TotalSamples = 0; state->SampleBuffer = NULL; ALfilterState_clear(&state->LpFilter); ALfilterState_clear(&state->HpFilter); state->Mod.Delay.Mask = 0; state->Mod.Delay.Line = NULL; state->Mod.Index = 0; state->Mod.Range = 1; state->Mod.Depth = 0.0f; state->Mod.Coeff = 0.0f; state->Mod.Filter = 0.0f; state->Delay.Mask = 0; state->Delay.Line = NULL; state->DelayTap[0] = 0; state->DelayTap[1] = 0; for(index = 0;index < 4;index++) { state->Early.Coeff[index] = 0.0f; state->Early.Delay[index].Mask = 0; state->Early.Delay[index].Line = NULL; state->Early.Offset[index] = 0; } state->Decorrelator.Mask = 0; state->Decorrelator.Line = NULL; state->DecoTap[0] = 0; state->DecoTap[1] = 0; state->DecoTap[2] = 0; state->Late.Gain = 0.0f; state->Late.DensityGain = 0.0f; state->Late.ApFeedCoeff = 0.0f; state->Late.MixCoeff = 0.0f; for(index = 0;index < 4;index++) { state->Late.ApCoeff[index] = 0.0f; state->Late.ApDelay[index].Mask = 0; state->Late.ApDelay[index].Line = NULL; state->Late.ApOffset[index] = 0; state->Late.Coeff[index] = 0.0f; state->Late.Delay[index].Mask = 0; state->Late.Delay[index].Line = NULL; state->Late.Offset[index] = 0; state->Late.LpCoeff[index] = 0.0f; state->Late.LpSample[index] = 0.0f; } for(l = 0;l < 4;l++) { for(index = 0;index < MAX_OUTPUT_CHANNELS;index++) { state->Early.PanGain[l][index] = 0.0f; state->Late.PanGain[l][index] = 0.0f; } } state->Echo.DensityGain = 0.0f; state->Echo.Delay.Mask = 0; state->Echo.Delay.Line = NULL; state->Echo.ApDelay.Mask = 0; state->Echo.ApDelay.Line = NULL; state->Echo.Coeff = 0.0f; state->Echo.ApFeedCoeff = 0.0f; state->Echo.ApCoeff = 0.0f; state->Echo.Offset = 0; state->Echo.ApOffset = 0; state->Echo.LpCoeff = 0.0f; state->Echo.LpSample = 0.0f; state->Echo.MixCoeff = 0.0f; state->Offset = 0; return STATIC_CAST(ALeffectState, state); } DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALreverbStateFactory); ALeffectStateFactory *ALreverbStateFactory_getFactory(void) { static ALreverbStateFactory ReverbFactory = { { GET_VTABLE2(ALreverbStateFactory, ALeffectStateFactory) } }; return STATIC_CAST(ALeffectStateFactory, &ReverbFactory); } void ALeaxreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DECAY_HFLIMIT: if(!(val >= AL_EAXREVERB_MIN_DECAY_HFLIMIT && val <= AL_EAXREVERB_MAX_DECAY_HFLIMIT)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.DecayHFLimit = val; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALeaxreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALeaxreverb_setParami(effect, context, param, vals[0]); } void ALeaxreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DENSITY: if(!(val >= AL_EAXREVERB_MIN_DENSITY && val <= AL_EAXREVERB_MAX_DENSITY)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.Density = val; break; case AL_EAXREVERB_DIFFUSION: if(!(val >= AL_EAXREVERB_MIN_DIFFUSION && val <= AL_EAXREVERB_MAX_DIFFUSION)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.Diffusion = val; break; case AL_EAXREVERB_GAIN: if(!(val >= AL_EAXREVERB_MIN_GAIN && val <= AL_EAXREVERB_MAX_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.Gain = val; break; case AL_EAXREVERB_GAINHF: if(!(val >= AL_EAXREVERB_MIN_GAINHF && val <= AL_EAXREVERB_MAX_GAINHF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.GainHF = val; break; case AL_EAXREVERB_GAINLF: if(!(val >= AL_EAXREVERB_MIN_GAINLF && val <= AL_EAXREVERB_MAX_GAINLF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.GainLF = val; break; case AL_EAXREVERB_DECAY_TIME: if(!(val >= AL_EAXREVERB_MIN_DECAY_TIME && val <= AL_EAXREVERB_MAX_DECAY_TIME)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.DecayTime = val; break; case AL_EAXREVERB_DECAY_HFRATIO: if(!(val >= AL_EAXREVERB_MIN_DECAY_HFRATIO && val <= AL_EAXREVERB_MAX_DECAY_HFRATIO)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.DecayHFRatio = val; break; case AL_EAXREVERB_DECAY_LFRATIO: if(!(val >= AL_EAXREVERB_MIN_DECAY_LFRATIO && val <= AL_EAXREVERB_MAX_DECAY_LFRATIO)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.DecayLFRatio = val; break; case AL_EAXREVERB_REFLECTIONS_GAIN: if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_GAIN && val <= AL_EAXREVERB_MAX_REFLECTIONS_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.ReflectionsGain = val; break; case AL_EAXREVERB_REFLECTIONS_DELAY: if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_DELAY && val <= AL_EAXREVERB_MAX_REFLECTIONS_DELAY)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.ReflectionsDelay = val; break; case AL_EAXREVERB_LATE_REVERB_GAIN: if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_GAIN && val <= AL_EAXREVERB_MAX_LATE_REVERB_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.LateReverbGain = val; break; case AL_EAXREVERB_LATE_REVERB_DELAY: if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_DELAY && val <= AL_EAXREVERB_MAX_LATE_REVERB_DELAY)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.LateReverbDelay = val; break; case AL_EAXREVERB_AIR_ABSORPTION_GAINHF: if(!(val >= AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.AirAbsorptionGainHF = val; break; case AL_EAXREVERB_ECHO_TIME: if(!(val >= AL_EAXREVERB_MIN_ECHO_TIME && val <= AL_EAXREVERB_MAX_ECHO_TIME)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.EchoTime = val; break; case AL_EAXREVERB_ECHO_DEPTH: if(!(val >= AL_EAXREVERB_MIN_ECHO_DEPTH && val <= AL_EAXREVERB_MAX_ECHO_DEPTH)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.EchoDepth = val; break; case AL_EAXREVERB_MODULATION_TIME: if(!(val >= AL_EAXREVERB_MIN_MODULATION_TIME && val <= AL_EAXREVERB_MAX_MODULATION_TIME)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.ModulationTime = val; break; case AL_EAXREVERB_MODULATION_DEPTH: if(!(val >= AL_EAXREVERB_MIN_MODULATION_DEPTH && val <= AL_EAXREVERB_MAX_MODULATION_DEPTH)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.ModulationDepth = val; break; case AL_EAXREVERB_HFREFERENCE: if(!(val >= AL_EAXREVERB_MIN_HFREFERENCE && val <= AL_EAXREVERB_MAX_HFREFERENCE)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.HFReference = val; break; case AL_EAXREVERB_LFREFERENCE: if(!(val >= AL_EAXREVERB_MIN_LFREFERENCE && val <= AL_EAXREVERB_MAX_LFREFERENCE)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.LFReference = val; break; case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR: if(!(val >= AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.RoomRolloffFactor = val; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALeaxreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_REFLECTIONS_PAN: if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2]))) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.ReflectionsPan[0] = vals[0]; props->Reverb.ReflectionsPan[1] = vals[1]; props->Reverb.ReflectionsPan[2] = vals[2]; break; case AL_EAXREVERB_LATE_REVERB_PAN: if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2]))) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.LateReverbPan[0] = vals[0]; props->Reverb.LateReverbPan[1] = vals[1]; props->Reverb.LateReverbPan[2] = vals[2]; break; default: ALeaxreverb_setParamf(effect, context, param, vals[0]); break; } } void ALeaxreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DECAY_HFLIMIT: *val = props->Reverb.DecayHFLimit; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALeaxreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALeaxreverb_getParami(effect, context, param, vals); } void ALeaxreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DENSITY: *val = props->Reverb.Density; break; case AL_EAXREVERB_DIFFUSION: *val = props->Reverb.Diffusion; break; case AL_EAXREVERB_GAIN: *val = props->Reverb.Gain; break; case AL_EAXREVERB_GAINHF: *val = props->Reverb.GainHF; break; case AL_EAXREVERB_GAINLF: *val = props->Reverb.GainLF; break; case AL_EAXREVERB_DECAY_TIME: *val = props->Reverb.DecayTime; break; case AL_EAXREVERB_DECAY_HFRATIO: *val = props->Reverb.DecayHFRatio; break; case AL_EAXREVERB_DECAY_LFRATIO: *val = props->Reverb.DecayLFRatio; break; case AL_EAXREVERB_REFLECTIONS_GAIN: *val = props->Reverb.ReflectionsGain; break; case AL_EAXREVERB_REFLECTIONS_DELAY: *val = props->Reverb.ReflectionsDelay; break; case AL_EAXREVERB_LATE_REVERB_GAIN: *val = props->Reverb.LateReverbGain; break; case AL_EAXREVERB_LATE_REVERB_DELAY: *val = props->Reverb.LateReverbDelay; break; case AL_EAXREVERB_AIR_ABSORPTION_GAINHF: *val = props->Reverb.AirAbsorptionGainHF; break; case AL_EAXREVERB_ECHO_TIME: *val = props->Reverb.EchoTime; break; case AL_EAXREVERB_ECHO_DEPTH: *val = props->Reverb.EchoDepth; break; case AL_EAXREVERB_MODULATION_TIME: *val = props->Reverb.ModulationTime; break; case AL_EAXREVERB_MODULATION_DEPTH: *val = props->Reverb.ModulationDepth; break; case AL_EAXREVERB_HFREFERENCE: *val = props->Reverb.HFReference; break; case AL_EAXREVERB_LFREFERENCE: *val = props->Reverb.LFReference; break; case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR: *val = props->Reverb.RoomRolloffFactor; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALeaxreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_REFLECTIONS_PAN: vals[0] = props->Reverb.ReflectionsPan[0]; vals[1] = props->Reverb.ReflectionsPan[1]; vals[2] = props->Reverb.ReflectionsPan[2]; break; case AL_EAXREVERB_LATE_REVERB_PAN: vals[0] = props->Reverb.LateReverbPan[0]; vals[1] = props->Reverb.LateReverbPan[1]; vals[2] = props->Reverb.LateReverbPan[2]; break; default: ALeaxreverb_getParamf(effect, context, param, vals); break; } } DEFINE_ALEFFECT_VTABLE(ALeaxreverb); void ALreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DECAY_HFLIMIT: if(!(val >= AL_REVERB_MIN_DECAY_HFLIMIT && val <= AL_REVERB_MAX_DECAY_HFLIMIT)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.DecayHFLimit = val; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALreverb_setParami(effect, context, param, vals[0]); } void ALreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DENSITY: if(!(val >= AL_REVERB_MIN_DENSITY && val <= AL_REVERB_MAX_DENSITY)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.Density = val; break; case AL_REVERB_DIFFUSION: if(!(val >= AL_REVERB_MIN_DIFFUSION && val <= AL_REVERB_MAX_DIFFUSION)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.Diffusion = val; break; case AL_REVERB_GAIN: if(!(val >= AL_REVERB_MIN_GAIN && val <= AL_REVERB_MAX_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.Gain = val; break; case AL_REVERB_GAINHF: if(!(val >= AL_REVERB_MIN_GAINHF && val <= AL_REVERB_MAX_GAINHF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.GainHF = val; break; case AL_REVERB_DECAY_TIME: if(!(val >= AL_REVERB_MIN_DECAY_TIME && val <= AL_REVERB_MAX_DECAY_TIME)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.DecayTime = val; break; case AL_REVERB_DECAY_HFRATIO: if(!(val >= AL_REVERB_MIN_DECAY_HFRATIO && val <= AL_REVERB_MAX_DECAY_HFRATIO)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.DecayHFRatio = val; break; case AL_REVERB_REFLECTIONS_GAIN: if(!(val >= AL_REVERB_MIN_REFLECTIONS_GAIN && val <= AL_REVERB_MAX_REFLECTIONS_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.ReflectionsGain = val; break; case AL_REVERB_REFLECTIONS_DELAY: if(!(val >= AL_REVERB_MIN_REFLECTIONS_DELAY && val <= AL_REVERB_MAX_REFLECTIONS_DELAY)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.ReflectionsDelay = val; break; case AL_REVERB_LATE_REVERB_GAIN: if(!(val >= AL_REVERB_MIN_LATE_REVERB_GAIN && val <= AL_REVERB_MAX_LATE_REVERB_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.LateReverbGain = val; break; case AL_REVERB_LATE_REVERB_DELAY: if(!(val >= AL_REVERB_MIN_LATE_REVERB_DELAY && val <= AL_REVERB_MAX_LATE_REVERB_DELAY)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.LateReverbDelay = val; break; case AL_REVERB_AIR_ABSORPTION_GAINHF: if(!(val >= AL_REVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_REVERB_MAX_AIR_ABSORPTION_GAINHF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.AirAbsorptionGainHF = val; break; case AL_REVERB_ROOM_ROLLOFF_FACTOR: if(!(val >= AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Reverb.RoomRolloffFactor = val; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALreverb_setParamf(effect, context, param, vals[0]); } void ALreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DECAY_HFLIMIT: *val = props->Reverb.DecayHFLimit; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALreverb_getParami(effect, context, param, vals); } void ALreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DENSITY: *val = props->Reverb.Density; break; case AL_REVERB_DIFFUSION: *val = props->Reverb.Diffusion; break; case AL_REVERB_GAIN: *val = props->Reverb.Gain; break; case AL_REVERB_GAINHF: *val = props->Reverb.GainHF; break; case AL_REVERB_DECAY_TIME: *val = props->Reverb.DecayTime; break; case AL_REVERB_DECAY_HFRATIO: *val = props->Reverb.DecayHFRatio; break; case AL_REVERB_REFLECTIONS_GAIN: *val = props->Reverb.ReflectionsGain; break; case AL_REVERB_REFLECTIONS_DELAY: *val = props->Reverb.ReflectionsDelay; break; case AL_REVERB_LATE_REVERB_GAIN: *val = props->Reverb.LateReverbGain; break; case AL_REVERB_LATE_REVERB_DELAY: *val = props->Reverb.LateReverbDelay; break; case AL_REVERB_AIR_ABSORPTION_GAINHF: *val = props->Reverb.AirAbsorptionGainHF; break; case AL_REVERB_ROOM_ROLLOFF_FACTOR: *val = props->Reverb.RoomRolloffFactor; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALreverb_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALreverb);