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1287 lines
47 KiB
C
1287 lines
47 KiB
C
/**
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* Reverb for the OpenAL cross platform audio library
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* Copyright (C) 2008-2009 by Christopher Fitzgerald.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <math.h>
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#include "AL/al.h"
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#include "AL/alc.h"
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#include "alMain.h"
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#include "alAuxEffectSlot.h"
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#include "alEffect.h"
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#include "alError.h"
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#include "alu.h"
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typedef struct DelayLine
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{
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// The delay lines use sample lengths that are powers of 2 to allow the
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// use of bit-masking instead of a modulus for wrapping.
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ALuint Mask;
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ALfloat *Line;
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} DelayLine;
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typedef struct ALverbState {
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// Must be first in all effects!
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ALeffectState state;
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// All delay lines are allocated as a single buffer to reduce memory
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// fragmentation and management code.
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ALfloat *SampleBuffer;
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ALuint TotalSamples;
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// Master effect low-pass filter (2 chained 1-pole filters).
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FILTER LpFilter;
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ALfloat LpHistory[2];
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struct {
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// Modulator delay line.
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DelayLine Delay;
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// The vibrato time is tracked with an index over a modulus-wrapped
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// range (in samples).
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ALuint Index;
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ALuint Range;
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// The depth of frequency change (also in samples) and its filter.
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ALfloat Depth;
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ALfloat Coeff;
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ALfloat Filter;
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} Mod;
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// Initial effect delay.
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DelayLine Delay;
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// The tap points for the initial delay. First tap goes to early
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// reflections, the last to late reverb.
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ALuint DelayTap[2];
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struct {
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// Output gain for early reflections.
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ALfloat Gain;
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// Early reflections are done with 4 delay lines.
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ALfloat Coeff[4];
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DelayLine Delay[4];
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ALuint Offset[4];
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// The gain for each output channel based on 3D panning (only for the
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// EAX path).
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ALfloat PanGain[MAXCHANNELS];
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} Early;
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// Decorrelator delay line.
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DelayLine Decorrelator;
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// There are actually 4 decorrelator taps, but the first occurs at the
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// initial sample.
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ALuint DecoTap[3];
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struct {
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// Output gain for late reverb.
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ALfloat Gain;
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// Attenuation to compensate for the modal density and decay rate of
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// the late lines.
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ALfloat DensityGain;
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// The feed-back and feed-forward all-pass coefficient.
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ALfloat ApFeedCoeff;
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// Mixing matrix coefficient.
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ALfloat MixCoeff;
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// Late reverb has 4 parallel all-pass filters.
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ALfloat ApCoeff[4];
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DelayLine ApDelay[4];
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ALuint ApOffset[4];
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// In addition to 4 cyclical delay lines.
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ALfloat Coeff[4];
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DelayLine Delay[4];
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ALuint Offset[4];
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// The cyclical delay lines are 1-pole low-pass filtered.
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ALfloat LpCoeff[4];
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ALfloat LpSample[4];
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// The gain for each output channel based on 3D panning (only for the
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// EAX path).
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ALfloat PanGain[MAXCHANNELS];
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} Late;
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struct {
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// Attenuation to compensate for the modal density and decay rate of
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// the echo line.
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ALfloat DensityGain;
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// Echo delay and all-pass lines.
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DelayLine Delay;
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DelayLine ApDelay;
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ALfloat Coeff;
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ALfloat ApFeedCoeff;
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ALfloat ApCoeff;
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ALuint Offset;
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ALuint ApOffset;
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// The echo line is 1-pole low-pass filtered.
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ALfloat LpCoeff;
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ALfloat LpSample;
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// Echo mixing coefficients.
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ALfloat MixCoeff[2];
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} Echo;
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// The current read offset for all delay lines.
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ALuint Offset;
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// The gain for each output channel (non-EAX path only; aliased from
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// Late.PanGain)
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ALfloat *Gain;
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} ALverbState;
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/* This is a user config option for modifying the overall output of the reverb
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* effect.
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*/
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ALfloat ReverbBoost = 1.0f;
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/* Specifies whether to use a standard reverb effect in place of EAX reverb */
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ALboolean EmulateEAXReverb = AL_FALSE;
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/* This coefficient is used to define the maximum frequency range controlled
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* by the modulation depth. The current value of 0.1 will allow it to swing
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* from 0.9x to 1.1x. This value must be below 1. At 1 it will cause the
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* sampler to stall on the downswing, and above 1 it will cause it to sample
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* backwards.
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*/
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static const ALfloat MODULATION_DEPTH_COEFF = 0.1f;
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/* A filter is used to avoid the terrible distortion caused by changing
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* modulation time and/or depth. To be consistent across different sample
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* rates, the coefficient must be raised to a constant divided by the sample
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* rate: coeff^(constant / rate).
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*/
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static const ALfloat MODULATION_FILTER_COEFF = 0.048f;
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static const ALfloat MODULATION_FILTER_CONST = 100000.0f;
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// When diffusion is above 0, an all-pass filter is used to take the edge off
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// the echo effect. It uses the following line length (in seconds).
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static const ALfloat ECHO_ALLPASS_LENGTH = 0.0133f;
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// Input into the late reverb is decorrelated between four channels. Their
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// timings are dependent on a fraction and multiplier. See the
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// UpdateDecorrelator() routine for the calculations involved.
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static const ALfloat DECO_FRACTION = 0.15f;
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static const ALfloat DECO_MULTIPLIER = 2.0f;
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// All delay line lengths are specified in seconds.
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// The lengths of the early delay lines.
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static const ALfloat EARLY_LINE_LENGTH[4] =
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{
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0.0015f, 0.0045f, 0.0135f, 0.0405f
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};
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// The lengths of the late all-pass delay lines.
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static const ALfloat ALLPASS_LINE_LENGTH[4] =
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{
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0.0151f, 0.0167f, 0.0183f, 0.0200f,
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};
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// The lengths of the late cyclical delay lines.
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static const ALfloat LATE_LINE_LENGTH[4] =
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{
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0.0211f, 0.0311f, 0.0461f, 0.0680f
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};
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// The late cyclical delay lines have a variable length dependent on the
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// effect's density parameter (inverted for some reason) and this multiplier.
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static const ALfloat LATE_LINE_MULTIPLIER = 4.0f;
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// Basic delay line input/output routines.
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static __inline ALfloat DelayLineOut(DelayLine *Delay, ALuint offset)
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{
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return Delay->Line[offset&Delay->Mask];
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}
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static __inline ALvoid DelayLineIn(DelayLine *Delay, ALuint offset, ALfloat in)
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{
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Delay->Line[offset&Delay->Mask] = in;
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}
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// Attenuated delay line output routine.
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static __inline ALfloat AttenuatedDelayLineOut(DelayLine *Delay, ALuint offset, ALfloat coeff)
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{
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return coeff * Delay->Line[offset&Delay->Mask];
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}
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// Basic attenuated all-pass input/output routine.
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static __inline ALfloat AllpassInOut(DelayLine *Delay, ALuint outOffset, ALuint inOffset, ALfloat in, ALfloat feedCoeff, ALfloat coeff)
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{
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ALfloat out, feed;
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out = DelayLineOut(Delay, outOffset);
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feed = feedCoeff * in;
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DelayLineIn(Delay, inOffset, (feedCoeff * (out - feed)) + in);
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// The time-based attenuation is only applied to the delay output to
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// keep it from affecting the feed-back path (which is already controlled
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// by the all-pass feed coefficient).
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return (coeff * out) - feed;
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}
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// Given an input sample, this function produces modulation for the late
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// reverb.
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static __inline ALfloat EAXModulation(ALverbState *State, ALfloat in)
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{
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ALfloat sinus, frac;
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ALuint offset;
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ALfloat out0, out1;
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// Calculate the sinus rythm (dependent on modulation time and the
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// sampling rate). The center of the sinus is moved to reduce the delay
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// of the effect when the time or depth are low.
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sinus = 1.0f - aluCos(F_PI*2.0f * State->Mod.Index / State->Mod.Range);
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// The depth determines the range over which to read the input samples
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// from, so it must be filtered to reduce the distortion caused by even
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// small parameter changes.
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State->Mod.Filter = lerp(State->Mod.Filter, State->Mod.Depth,
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State->Mod.Coeff);
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// Calculate the read offset and fraction between it and the next sample.
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frac = (1.0f + (State->Mod.Filter * sinus));
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offset = fastf2u(frac);
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frac -= offset;
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// Get the two samples crossed by the offset, and feed the delay line
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// with the next input sample.
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out0 = DelayLineOut(&State->Mod.Delay, State->Offset - offset);
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out1 = DelayLineOut(&State->Mod.Delay, State->Offset - offset - 1);
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DelayLineIn(&State->Mod.Delay, State->Offset, in);
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// Step the modulation index forward, keeping it bound to its range.
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State->Mod.Index = (State->Mod.Index + 1) % State->Mod.Range;
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// The output is obtained by linearly interpolating the two samples that
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// were acquired above.
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return lerp(out0, out1, frac);
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}
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// Delay line output routine for early reflections.
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static __inline ALfloat EarlyDelayLineOut(ALverbState *State, ALuint index)
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{
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return AttenuatedDelayLineOut(&State->Early.Delay[index],
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State->Offset - State->Early.Offset[index],
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State->Early.Coeff[index]);
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}
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// Given an input sample, this function produces four-channel output for the
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// early reflections.
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static __inline ALvoid EarlyReflection(ALverbState *State, ALfloat in, ALfloat *out)
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{
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ALfloat d[4], v, f[4];
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// Obtain the decayed results of each early delay line.
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d[0] = EarlyDelayLineOut(State, 0);
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d[1] = EarlyDelayLineOut(State, 1);
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d[2] = EarlyDelayLineOut(State, 2);
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d[3] = EarlyDelayLineOut(State, 3);
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/* The following uses a lossless scattering junction from waveguide
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* theory. It actually amounts to a householder mixing matrix, which
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* will produce a maximally diffuse response, and means this can probably
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* be considered a simple feed-back delay network (FDN).
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* N
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* ---
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* \
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* v = 2/N / d_i
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* ---
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* i=1
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*/
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v = (d[0] + d[1] + d[2] + d[3]) * 0.5f;
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// The junction is loaded with the input here.
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v += in;
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// Calculate the feed values for the delay lines.
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f[0] = v - d[0];
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f[1] = v - d[1];
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f[2] = v - d[2];
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f[3] = v - d[3];
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// Re-feed the delay lines.
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DelayLineIn(&State->Early.Delay[0], State->Offset, f[0]);
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DelayLineIn(&State->Early.Delay[1], State->Offset, f[1]);
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DelayLineIn(&State->Early.Delay[2], State->Offset, f[2]);
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DelayLineIn(&State->Early.Delay[3], State->Offset, f[3]);
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// Output the results of the junction for all four channels.
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out[0] = State->Early.Gain * f[0];
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out[1] = State->Early.Gain * f[1];
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out[2] = State->Early.Gain * f[2];
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out[3] = State->Early.Gain * f[3];
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}
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// All-pass input/output routine for late reverb.
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static __inline ALfloat LateAllPassInOut(ALverbState *State, ALuint index, ALfloat in)
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{
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return AllpassInOut(&State->Late.ApDelay[index],
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State->Offset - State->Late.ApOffset[index],
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State->Offset, in, State->Late.ApFeedCoeff,
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State->Late.ApCoeff[index]);
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}
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// Delay line output routine for late reverb.
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static __inline ALfloat LateDelayLineOut(ALverbState *State, ALuint index)
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{
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return AttenuatedDelayLineOut(&State->Late.Delay[index],
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State->Offset - State->Late.Offset[index],
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State->Late.Coeff[index]);
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}
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// Low-pass filter input/output routine for late reverb.
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static __inline ALfloat LateLowPassInOut(ALverbState *State, ALuint index, ALfloat in)
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{
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in = lerp(in, State->Late.LpSample[index], State->Late.LpCoeff[index]);
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State->Late.LpSample[index] = in;
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return in;
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}
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// Given four decorrelated input samples, this function produces four-channel
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// output for the late reverb.
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static __inline ALvoid LateReverb(ALverbState *State, ALfloat *in, ALfloat *out)
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{
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ALfloat d[4], f[4];
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// Obtain the decayed results of the cyclical delay lines, and add the
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// corresponding input channels. Then pass the results through the
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// low-pass filters.
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// This is where the feed-back cycles from line 0 to 1 to 3 to 2 and back
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// to 0.
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d[0] = LateLowPassInOut(State, 2, in[2] + LateDelayLineOut(State, 2));
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d[1] = LateLowPassInOut(State, 0, in[0] + LateDelayLineOut(State, 0));
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d[2] = LateLowPassInOut(State, 3, in[3] + LateDelayLineOut(State, 3));
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d[3] = LateLowPassInOut(State, 1, in[1] + LateDelayLineOut(State, 1));
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// To help increase diffusion, run each line through an all-pass filter.
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// When there is no diffusion, the shortest all-pass filter will feed the
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// shortest delay line.
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d[0] = LateAllPassInOut(State, 0, d[0]);
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d[1] = LateAllPassInOut(State, 1, d[1]);
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d[2] = LateAllPassInOut(State, 2, d[2]);
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d[3] = LateAllPassInOut(State, 3, d[3]);
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/* Late reverb is done with a modified feed-back delay network (FDN)
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* topology. Four input lines are each fed through their own all-pass
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* filter and then into the mixing matrix. The four outputs of the
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* mixing matrix are then cycled back to the inputs. Each output feeds
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* a different input to form a circlular feed cycle.
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*
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* The mixing matrix used is a 4D skew-symmetric rotation matrix derived
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* using a single unitary rotational parameter:
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*
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* [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2
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* [ -a, d, c, -b ]
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* [ -b, -c, d, a ]
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* [ -c, b, -a, d ]
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*
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* The rotation is constructed from the effect's diffusion parameter,
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* yielding: 1 = x^2 + 3 y^2; where a, b, and c are the coefficient y
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* with differing signs, and d is the coefficient x. The matrix is thus:
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*
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* [ x, y, -y, y ] n = sqrt(matrix_order - 1)
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* [ -y, x, y, y ] t = diffusion_parameter * atan(n)
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* [ y, -y, x, y ] x = cos(t)
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* [ -y, -y, -y, x ] y = sin(t) / n
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*
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* To reduce the number of multiplies, the x coefficient is applied with
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* the cyclical delay line coefficients. Thus only the y coefficient is
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* applied when mixing, and is modified to be: y / x.
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*/
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f[0] = d[0] + (State->Late.MixCoeff * ( d[1] + -d[2] + d[3]));
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f[1] = d[1] + (State->Late.MixCoeff * (-d[0] + d[2] + d[3]));
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f[2] = d[2] + (State->Late.MixCoeff * ( d[0] + -d[1] + d[3]));
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f[3] = d[3] + (State->Late.MixCoeff * (-d[0] + -d[1] + -d[2] ));
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// Output the results of the matrix for all four channels, attenuated by
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// the late reverb gain (which is attenuated by the 'x' mix coefficient).
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out[0] = State->Late.Gain * f[0];
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out[1] = State->Late.Gain * f[1];
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out[2] = State->Late.Gain * f[2];
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out[3] = State->Late.Gain * f[3];
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// Re-feed the cyclical delay lines.
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DelayLineIn(&State->Late.Delay[0], State->Offset, f[0]);
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DelayLineIn(&State->Late.Delay[1], State->Offset, f[1]);
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DelayLineIn(&State->Late.Delay[2], State->Offset, f[2]);
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DelayLineIn(&State->Late.Delay[3], State->Offset, f[3]);
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}
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// Given an input sample, this function mixes echo into the four-channel late
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// reverb.
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static __inline ALvoid EAXEcho(ALverbState *State, ALfloat in, ALfloat *late)
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{
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ALfloat out, feed;
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// Get the latest attenuated echo sample for output.
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feed = AttenuatedDelayLineOut(&State->Echo.Delay,
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State->Offset - State->Echo.Offset,
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State->Echo.Coeff);
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// Mix the output into the late reverb channels.
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out = State->Echo.MixCoeff[0] * feed;
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late[0] = (State->Echo.MixCoeff[1] * late[0]) + out;
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late[1] = (State->Echo.MixCoeff[1] * late[1]) + out;
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late[2] = (State->Echo.MixCoeff[1] * late[2]) + out;
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late[3] = (State->Echo.MixCoeff[1] * late[3]) + out;
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// Mix the energy-attenuated input with the output and pass it through
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// the echo low-pass filter.
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feed += State->Echo.DensityGain * in;
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feed = lerp(feed, State->Echo.LpSample, State->Echo.LpCoeff);
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State->Echo.LpSample = feed;
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// Then the echo all-pass filter.
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feed = AllpassInOut(&State->Echo.ApDelay,
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State->Offset - State->Echo.ApOffset,
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State->Offset, feed, State->Echo.ApFeedCoeff,
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State->Echo.ApCoeff);
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// Feed the delay with the mixed and filtered sample.
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DelayLineIn(&State->Echo.Delay, State->Offset, feed);
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}
|
|
|
|
// Perform the non-EAX reverb pass on a given input sample, resulting in
|
|
// four-channel output.
|
|
static __inline ALvoid VerbPass(ALverbState *State, ALfloat in, ALfloat *early, ALfloat *late)
|
|
{
|
|
ALfloat feed, taps[4];
|
|
|
|
// Low-pass filter the incoming sample.
|
|
in = lpFilter2P(&State->LpFilter, 0, in);
|
|
|
|
// Feed the initial delay line.
|
|
DelayLineIn(&State->Delay, State->Offset, in);
|
|
|
|
// Calculate the early reflection from the first delay tap.
|
|
in = DelayLineOut(&State->Delay, State->Offset - State->DelayTap[0]);
|
|
EarlyReflection(State, in, early);
|
|
|
|
// Feed the decorrelator from the energy-attenuated output of the second
|
|
// delay tap.
|
|
in = DelayLineOut(&State->Delay, State->Offset - State->DelayTap[1]);
|
|
feed = in * State->Late.DensityGain;
|
|
DelayLineIn(&State->Decorrelator, State->Offset, feed);
|
|
|
|
// Calculate the late reverb from the decorrelator taps.
|
|
taps[0] = feed;
|
|
taps[1] = DelayLineOut(&State->Decorrelator, State->Offset - State->DecoTap[0]);
|
|
taps[2] = DelayLineOut(&State->Decorrelator, State->Offset - State->DecoTap[1]);
|
|
taps[3] = DelayLineOut(&State->Decorrelator, State->Offset - State->DecoTap[2]);
|
|
LateReverb(State, taps, late);
|
|
|
|
// Step all delays forward one sample.
|
|
State->Offset++;
|
|
}
|
|
|
|
// Perform the EAX reverb pass on a given input sample, resulting in four-
|
|
// channel output.
|
|
static __inline ALvoid EAXVerbPass(ALverbState *State, ALfloat in, ALfloat *early, ALfloat *late)
|
|
{
|
|
ALfloat feed, taps[4];
|
|
|
|
// Low-pass filter the incoming sample.
|
|
in = lpFilter2P(&State->LpFilter, 0, in);
|
|
|
|
// Perform any modulation on the input.
|
|
in = EAXModulation(State, in);
|
|
|
|
// Feed the initial delay line.
|
|
DelayLineIn(&State->Delay, State->Offset, in);
|
|
|
|
// Calculate the early reflection from the first delay tap.
|
|
in = DelayLineOut(&State->Delay, State->Offset - State->DelayTap[0]);
|
|
EarlyReflection(State, in, early);
|
|
|
|
// Feed the decorrelator from the energy-attenuated output of the second
|
|
// delay tap.
|
|
in = DelayLineOut(&State->Delay, State->Offset - State->DelayTap[1]);
|
|
feed = in * State->Late.DensityGain;
|
|
DelayLineIn(&State->Decorrelator, State->Offset, feed);
|
|
|
|
// Calculate the late reverb from the decorrelator taps.
|
|
taps[0] = feed;
|
|
taps[1] = DelayLineOut(&State->Decorrelator, State->Offset - State->DecoTap[0]);
|
|
taps[2] = DelayLineOut(&State->Decorrelator, State->Offset - State->DecoTap[1]);
|
|
taps[3] = DelayLineOut(&State->Decorrelator, State->Offset - State->DecoTap[2]);
|
|
LateReverb(State, taps, late);
|
|
|
|
// Calculate and mix in any echo.
|
|
EAXEcho(State, in, late);
|
|
|
|
// Step all delays forward one sample.
|
|
State->Offset++;
|
|
}
|
|
|
|
// This processes the reverb state, given the input samples and an output
|
|
// buffer.
|
|
static ALvoid VerbProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS])
|
|
{
|
|
ALverbState *State = (ALverbState*)effect;
|
|
ALuint index, c;
|
|
ALfloat early[4], late[4], out[4];
|
|
const ALfloat *panGain = State->Gain;
|
|
|
|
for(index = 0;index < SamplesToDo;index++)
|
|
{
|
|
// Process reverb for this sample.
|
|
VerbPass(State, SamplesIn[index], early, late);
|
|
|
|
// Mix early reflections and late reverb.
|
|
out[0] = (early[0] + late[0]);
|
|
out[1] = (early[1] + late[1]);
|
|
out[2] = (early[2] + late[2]);
|
|
out[3] = (early[3] + late[3]);
|
|
|
|
// Output the results.
|
|
for(c = 0;c < MAXCHANNELS;c++)
|
|
SamplesOut[index][c] += panGain[c] * out[c&3];
|
|
}
|
|
}
|
|
|
|
// This processes the EAX reverb state, given the input samples and an output
|
|
// buffer.
|
|
static ALvoid EAXVerbProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS])
|
|
{
|
|
ALverbState *State = (ALverbState*)effect;
|
|
ALuint index, c;
|
|
ALfloat early[4], late[4];
|
|
|
|
for(index = 0;index < SamplesToDo;index++)
|
|
{
|
|
// Process reverb for this sample.
|
|
EAXVerbPass(State, SamplesIn[index], early, late);
|
|
|
|
for(c = 0;c < MAXCHANNELS;c++)
|
|
SamplesOut[index][c] += State->Early.PanGain[c]*early[c&3] +
|
|
State->Late.PanGain[c]*late[c&3];
|
|
}
|
|
}
|
|
|
|
|
|
// Given the allocated sample buffer, this function updates each delay line
|
|
// offset.
|
|
static __inline ALvoid RealizeLineOffset(ALfloat * sampleBuffer, DelayLine *Delay)
|
|
{
|
|
Delay->Line = &sampleBuffer[(ALintptrEXT)Delay->Line];
|
|
}
|
|
|
|
// Calculate the length of a delay line and store its mask and offset.
|
|
static ALuint CalcLineLength(ALfloat length, ALintptrEXT offset, ALuint frequency, 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 = NextPowerOf2(fastf2u(length * frequency) + 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, ALverbState *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) +
|
|
(1.0f / frequency);
|
|
totalSamples += CalcLineLength(length, totalSamples, frequency,
|
|
&State->Mod.Delay);
|
|
|
|
// The initial delay is the sum of the reflections and late reverb
|
|
// delays.
|
|
length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY +
|
|
AL_EAXREVERB_MAX_LATE_REVERB_DELAY;
|
|
totalSamples += CalcLineLength(length, totalSamples, frequency,
|
|
&State->Delay);
|
|
|
|
// The early reflection lines.
|
|
for(index = 0;index < 4;index++)
|
|
totalSamples += CalcLineLength(EARLY_LINE_LENGTH[index], totalSamples,
|
|
frequency, &State->Early.Delay[index]);
|
|
|
|
// The decorrelator line is calculated from the lowest reverb density (a
|
|
// parameter value of 1).
|
|
length = (DECO_FRACTION * DECO_MULTIPLIER * DECO_MULTIPLIER) *
|
|
LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER);
|
|
totalSamples += CalcLineLength(length, totalSamples, frequency,
|
|
&State->Decorrelator);
|
|
|
|
// The late all-pass lines.
|
|
for(index = 0;index < 4;index++)
|
|
totalSamples += CalcLineLength(ALLPASS_LINE_LENGTH[index], totalSamples,
|
|
frequency, &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,
|
|
&State->Late.Delay[index]);
|
|
}
|
|
|
|
// The echo all-pass and delay lines.
|
|
totalSamples += CalcLineLength(ECHO_ALLPASS_LENGTH, totalSamples,
|
|
frequency, &State->Echo.ApDelay);
|
|
totalSamples += CalcLineLength(AL_EAXREVERB_MAX_ECHO_TIME, totalSamples,
|
|
frequency, &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;
|
|
}
|
|
|
|
// This updates the device-dependant EAX reverb state. This is called on
|
|
// initialization and any time the device parameters (eg. playback frequency,
|
|
// format) have been changed.
|
|
static ALboolean ReverbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
|
|
{
|
|
ALverbState *State = (ALverbState*)effect;
|
|
ALuint frequency = Device->Frequency, index;
|
|
|
|
// Allocate the delay lines.
|
|
if(!AllocLines(frequency, State))
|
|
return AL_FALSE;
|
|
|
|
// 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 = aluPow(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;
|
|
}
|
|
|
|
// 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 aluPow(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 aluLog10(coeff) * decayTime / aluLog10(0.001f)/*-60 dB*/;
|
|
}
|
|
|
|
// Calculate the high frequency parameter for the I3DL2 coefficient
|
|
// calculation.
|
|
static __inline ALfloat CalcI3DL2HFreq(ALfloat hfRef, ALuint frequency)
|
|
{
|
|
return aluCos(F_PI*2.0f * hfRef / frequency);
|
|
}
|
|
|
|
// 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 aluSqrt(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 = aluSqrt(3.0f);
|
|
t = diffusion * aluAtan(n);
|
|
|
|
// Calculate the first mixing matrix coefficient.
|
|
*x = aluCos(t);
|
|
// Calculate the second mixing matrix coefficient.
|
|
*y = aluSin(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;
|
|
coeff = lpCoeffCalc(g, cw);
|
|
|
|
// 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, ALverbState *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, ALverbState *State)
|
|
{
|
|
// Calculate the initial delay taps.
|
|
State->DelayTap[0] = fastf2u(earlyDelay * frequency);
|
|
State->DelayTap[1] = fastf2u((earlyDelay + lateDelay) * frequency);
|
|
}
|
|
|
|
// Update the early reflections gain and line coefficients.
|
|
static ALvoid UpdateEarlyLines(ALfloat reverbGain, ALfloat earlyGain, ALfloat lateDelay, ALverbState *State)
|
|
{
|
|
ALuint index;
|
|
|
|
// Calculate the early reflections gain (from the master effect gain, and
|
|
// reflections gain parameters) with a constant attenuation of 0.5.
|
|
State->Early.Gain = 0.5f * reverbGain * earlyGain;
|
|
|
|
// 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, ALverbState *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 * aluPow(DECO_MULTIPLIER, (ALfloat)index)) *
|
|
LATE_LINE_LENGTH[0] * (1.0f + (density * LATE_LINE_MULTIPLIER));
|
|
State->DecoTap[index] = fastf2u(length * frequency);
|
|
}
|
|
}
|
|
|
|
// Update the late reverb gains, line lengths, and line coefficients.
|
|
static ALvoid UpdateLateLines(ALfloat reverbGain, ALfloat lateGain, ALfloat xMix, ALfloat density, ALfloat decayTime, ALfloat diffusion, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALverbState *State)
|
|
{
|
|
ALfloat length;
|
|
ALuint index;
|
|
|
|
/* Calculate the late reverb gain (from the master effect gain, and late
|
|
* reverb gain parameters). 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.
|
|
*/
|
|
State->Late.Gain = reverbGain * lateGain * xMix;
|
|
|
|
/* 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 * aluPow(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 reverbGain, ALfloat lateGain, ALfloat echoTime, ALfloat decayTime, ALfloat diffusion, ALfloat echoDepth, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALverbState *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 * aluPow(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 coefficients. The first is applied to the
|
|
* echo itself. The second is used to 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->Echo.MixCoeff[0] = reverbGain * lateGain * echoDepth;
|
|
State->Echo.MixCoeff[1] = 1.0f - (echoDepth * 0.5f * (1.0f - diffusion));
|
|
}
|
|
|
|
// Update the early and late 3D panning gains.
|
|
static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALverbState *State)
|
|
{
|
|
ALfloat earlyPan[3] = { ReflectionsPan[0], ReflectionsPan[1],
|
|
ReflectionsPan[2] };
|
|
ALfloat latePan[3] = { LateReverbPan[0], LateReverbPan[1],
|
|
LateReverbPan[2] };
|
|
const ALfloat *ChannelGain;
|
|
ALfloat ambientGain;
|
|
ALfloat dirGain;
|
|
ALfloat length;
|
|
ALuint index;
|
|
ALint pos;
|
|
|
|
Gain *= ReverbBoost;
|
|
|
|
// Attenuate non-directional reverb according to the number of channels
|
|
ambientGain = aluSqrt(2.0f/Device->NumChan);
|
|
|
|
// Calculate the 3D-panning gains for the early reflections and late
|
|
// reverb.
|
|
length = earlyPan[0]*earlyPan[0] + earlyPan[1]*earlyPan[1] + earlyPan[2]*earlyPan[2];
|
|
if(length > 1.0f)
|
|
{
|
|
length = 1.0f / aluSqrt(length);
|
|
earlyPan[0] *= length;
|
|
earlyPan[1] *= length;
|
|
earlyPan[2] *= length;
|
|
}
|
|
length = latePan[0]*latePan[0] + latePan[1]*latePan[1] + latePan[2]*latePan[2];
|
|
if(length > 1.0f)
|
|
{
|
|
length = 1.0f / aluSqrt(length);
|
|
latePan[0] *= length;
|
|
latePan[1] *= length;
|
|
latePan[2] *= length;
|
|
}
|
|
|
|
/* This code applies directional reverb just like the mixer applies
|
|
* directional sources. It diffuses the sound toward all speakers as the
|
|
* magnitude of the panning vector drops, which is only a rough
|
|
* approximation of the expansion of sound across the speakers from the
|
|
* panning direction.
|
|
*/
|
|
pos = aluCart2LUTpos(earlyPan[2], earlyPan[0]);
|
|
ChannelGain = Device->PanningLUT[pos];
|
|
dirGain = aluSqrt((earlyPan[0] * earlyPan[0]) + (earlyPan[2] * earlyPan[2]));
|
|
|
|
for(index = 0;index < MAXCHANNELS;index++)
|
|
State->Early.PanGain[index] = 0.0f;
|
|
for(index = 0;index < Device->NumChan;index++)
|
|
{
|
|
enum Channel chan = Device->Speaker2Chan[index];
|
|
State->Early.PanGain[chan] = lerp(ambientGain, ChannelGain[chan], dirGain) * Gain;
|
|
}
|
|
|
|
|
|
pos = aluCart2LUTpos(latePan[2], latePan[0]);
|
|
ChannelGain = Device->PanningLUT[pos];
|
|
dirGain = aluSqrt((latePan[0] * latePan[0]) + (latePan[2] * latePan[2]));
|
|
|
|
for(index = 0;index < MAXCHANNELS;index++)
|
|
State->Late.PanGain[index] = 0.0f;
|
|
for(index = 0;index < Device->NumChan;index++)
|
|
{
|
|
enum Channel chan = Device->Speaker2Chan[index];
|
|
State->Late.PanGain[chan] = lerp(ambientGain, ChannelGain[chan], dirGain) * Gain;
|
|
}
|
|
}
|
|
|
|
// This updates the EAX reverb state. This is called any time the EAX reverb
|
|
// effect is loaded into a slot.
|
|
static ALvoid ReverbUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
|
|
{
|
|
ALverbState *State = (ALverbState*)effect;
|
|
ALuint frequency = Device->Frequency;
|
|
ALboolean isEAX = AL_FALSE;
|
|
ALfloat cw, x, y, hfRatio;
|
|
|
|
if(Slot->effect.type == AL_EFFECT_EAXREVERB && !EmulateEAXReverb)
|
|
{
|
|
State->state.Process = EAXVerbProcess;
|
|
isEAX = AL_TRUE;
|
|
}
|
|
else if(Slot->effect.type == AL_EFFECT_REVERB || EmulateEAXReverb)
|
|
{
|
|
State->state.Process = VerbProcess;
|
|
isEAX = AL_FALSE;
|
|
}
|
|
|
|
// Calculate the master low-pass filter (from the master effect HF gain).
|
|
if(isEAX) cw = CalcI3DL2HFreq(Slot->effect.Reverb.HFReference, frequency);
|
|
else cw = CalcI3DL2HFreq(LOWPASSFREQREF, frequency);
|
|
// This is done with 2 chained 1-pole filters, so no need to square g.
|
|
State->LpFilter.coeff = lpCoeffCalc(Slot->effect.Reverb.GainHF, cw);
|
|
|
|
if(isEAX)
|
|
{
|
|
// Update the modulator line.
|
|
UpdateModulator(Slot->effect.Reverb.ModulationTime,
|
|
Slot->effect.Reverb.ModulationDepth,
|
|
frequency, State);
|
|
}
|
|
|
|
// Update the initial effect delay.
|
|
UpdateDelayLine(Slot->effect.Reverb.ReflectionsDelay,
|
|
Slot->effect.Reverb.LateReverbDelay,
|
|
frequency, State);
|
|
|
|
// Update the early lines.
|
|
UpdateEarlyLines(Slot->effect.Reverb.Gain,
|
|
Slot->effect.Reverb.ReflectionsGain,
|
|
Slot->effect.Reverb.LateReverbDelay, State);
|
|
|
|
// Update the decorrelator.
|
|
UpdateDecorrelator(Slot->effect.Reverb.Density, frequency, State);
|
|
|
|
// Get the mixing matrix coefficients (x and y).
|
|
CalcMatrixCoeffs(Slot->effect.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 = Slot->effect.Reverb.DecayHFRatio;
|
|
if(Slot->effect.Reverb.DecayHFLimit &&
|
|
Slot->effect.Reverb.AirAbsorptionGainHF < 1.0f)
|
|
hfRatio = CalcLimitedHfRatio(hfRatio,
|
|
Slot->effect.Reverb.AirAbsorptionGainHF,
|
|
Slot->effect.Reverb.DecayTime);
|
|
|
|
// Update the late lines.
|
|
UpdateLateLines(Slot->effect.Reverb.Gain, Slot->effect.Reverb.LateReverbGain,
|
|
x, Slot->effect.Reverb.Density, Slot->effect.Reverb.DecayTime,
|
|
Slot->effect.Reverb.Diffusion, hfRatio, cw, frequency, State);
|
|
|
|
if(isEAX)
|
|
{
|
|
// Update the echo line.
|
|
UpdateEchoLine(Slot->effect.Reverb.Gain, Slot->effect.Reverb.LateReverbGain,
|
|
Slot->effect.Reverb.EchoTime, Slot->effect.Reverb.DecayTime,
|
|
Slot->effect.Reverb.Diffusion, Slot->effect.Reverb.EchoDepth,
|
|
hfRatio, cw, frequency, State);
|
|
|
|
// Update early and late 3D panning.
|
|
Update3DPanning(Device, Slot->effect.Reverb.ReflectionsPan,
|
|
Slot->effect.Reverb.LateReverbPan, Slot->Gain, State);
|
|
}
|
|
else
|
|
{
|
|
ALfloat gain = Slot->Gain;
|
|
ALuint index;
|
|
|
|
/* Update channel gains */
|
|
gain *= aluSqrt(2.0f/Device->NumChan) * ReverbBoost;
|
|
for(index = 0;index < MAXCHANNELS;index++)
|
|
State->Gain[index] = 0.0f;
|
|
for(index = 0;index < Device->NumChan;index++)
|
|
{
|
|
enum Channel chan = Device->Speaker2Chan[index];
|
|
State->Gain[chan] = gain;
|
|
}
|
|
}
|
|
}
|
|
|
|
// This destroys the reverb state. It should be called only when the effect
|
|
// slot has a different (or no) effect loaded over the reverb effect.
|
|
static ALvoid ReverbDestroy(ALeffectState *effect)
|
|
{
|
|
ALverbState *State = (ALverbState*)effect;
|
|
if(State)
|
|
{
|
|
free(State->SampleBuffer);
|
|
State->SampleBuffer = NULL;
|
|
free(State);
|
|
}
|
|
}
|
|
|
|
// This creates the reverb state. It should be called only when the reverb
|
|
// effect is loaded into a slot that doesn't already have a reverb effect.
|
|
ALeffectState *ReverbCreate(void)
|
|
{
|
|
ALverbState *State = NULL;
|
|
ALuint index;
|
|
|
|
State = malloc(sizeof(ALverbState));
|
|
if(!State)
|
|
return NULL;
|
|
|
|
State->state.Destroy = ReverbDestroy;
|
|
State->state.DeviceUpdate = ReverbDeviceUpdate;
|
|
State->state.Update = ReverbUpdate;
|
|
State->state.Process = VerbProcess;
|
|
|
|
State->TotalSamples = 0;
|
|
State->SampleBuffer = NULL;
|
|
|
|
State->LpFilter.coeff = 0.0f;
|
|
State->LpFilter.history[0] = 0.0f;
|
|
State->LpFilter.history[1] = 0.0f;
|
|
|
|
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;
|
|
|
|
State->Early.Gain = 0.0f;
|
|
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(index = 0;index < MAXCHANNELS;index++)
|
|
{
|
|
State->Early.PanGain[index] = 0.0f;
|
|
State->Late.PanGain[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] = 0.0f;
|
|
State->Echo.MixCoeff[1] = 0.0f;
|
|
|
|
State->Offset = 0;
|
|
|
|
State->Gain = State->Late.PanGain;
|
|
|
|
return &State->state;
|
|
}
|