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Diffstat (limited to 'alc/effects/reverb.cpp')
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diff --git a/alc/effects/reverb.cpp b/alc/effects/reverb.cpp new file mode 100644 index 00000000..ac996b3f --- /dev/null +++ b/alc/effects/reverb.cpp @@ -0,0 +1,2102 @@ +/** + * Ambisonic reverb engine for the OpenAL cross platform audio library + * Copyright (C) 2008-2017 by Chris Robinson and 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 <cstdio> +#include <cstdlib> +#include <cmath> + +#include <array> +#include <numeric> +#include <algorithm> +#include <functional> + +#include "alcmain.h" +#include "alcontext.h" +#include "alu.h" +#include "alAuxEffectSlot.h" +#include "alListener.h" +#include "alError.h" +#include "bformatdec.h" +#include "filters/biquad.h" +#include "vector.h" +#include "vecmat.h" + +/* This is a user config option for modifying the overall output of the reverb + * effect. + */ +ALfloat ReverbBoost = 1.0f; + +namespace { + +using namespace std::placeholders; + +/* The number of samples used for cross-faded delay lines. This can be used + * to balance the compensation for abrupt line changes and attenuation due to + * minimally lengthed recursive lines. Try to keep this below the device + * update size. + */ +constexpr int FADE_SAMPLES{128}; + +/* The number of spatialized lines or channels to process. Four channels allows + * for a 3D A-Format response. NOTE: This can't be changed without taking care + * of the conversion matrices, and a few places where the length arrays are + * assumed to have 4 elements. + */ +constexpr int NUM_LINES{4}; + + +/* The B-Format to A-Format conversion matrix. The arrangement of rows is + * deliberately chosen to align the resulting lines to their spatial opposites + * (0:above front left <-> 3:above back right, 1:below front right <-> 2:below + * back left). It's not quite opposite, since the A-Format results in a + * tetrahedron, but it's close enough. Should the model be extended to 8-lines + * in the future, true opposites can be used. + */ +alignas(16) constexpr ALfloat B2A[NUM_LINES][MAX_AMBI_CHANNELS]{ + { 0.288675134595f, 0.288675134595f, 0.288675134595f, 0.288675134595f }, + { 0.288675134595f, -0.288675134595f, -0.288675134595f, 0.288675134595f }, + { 0.288675134595f, 0.288675134595f, -0.288675134595f, -0.288675134595f }, + { 0.288675134595f, -0.288675134595f, 0.288675134595f, -0.288675134595f } +}; + +/* Converts A-Format to B-Format. */ +alignas(16) constexpr ALfloat A2B[NUM_LINES][NUM_LINES]{ + { 0.866025403785f, 0.866025403785f, 0.866025403785f, 0.866025403785f }, + { 0.866025403785f, -0.866025403785f, 0.866025403785f, -0.866025403785f }, + { 0.866025403785f, -0.866025403785f, -0.866025403785f, 0.866025403785f }, + { 0.866025403785f, 0.866025403785f, -0.866025403785f, -0.866025403785f } +}; + + +constexpr ALfloat FadeStep{1.0f / FADE_SAMPLES}; + +/* The all-pass and delay lines have a variable length dependent on the + * effect's density parameter, which helps alter the perceived environment + * size. The size-to-density conversion is a cubed scale: + * + * density = min(1.0, pow(size, 3.0) / DENSITY_SCALE); + * + * The line lengths scale linearly with room size, so the inverse density + * conversion is needed, taking the cube root of the re-scaled density to + * calculate the line length multiplier: + * + * length_mult = max(5.0, cbrt(density*DENSITY_SCALE)); + * + * The density scale below will result in a max line multiplier of 50, for an + * effective size range of 5m to 50m. + */ +constexpr ALfloat DENSITY_SCALE{125000.0f}; + +/* All delay line lengths are specified in seconds. + * + * To approximate early reflections, we break them up into primary (those + * arriving from the same direction as the source) and secondary (those + * arriving from the opposite direction). + * + * The early taps decorrelate the 4-channel signal to approximate an average + * room response for the primary reflections after the initial early delay. + * + * Given an average room dimension (d_a) and the speed of sound (c) we can + * calculate the average reflection delay (r_a) regardless of listener and + * source positions as: + * + * r_a = d_a / c + * c = 343.3 + * + * This can extended to finding the average difference (r_d) between the + * maximum (r_1) and minimum (r_0) reflection delays: + * + * r_0 = 2 / 3 r_a + * = r_a - r_d / 2 + * = r_d + * r_1 = 4 / 3 r_a + * = r_a + r_d / 2 + * = 2 r_d + * r_d = 2 / 3 r_a + * = r_1 - r_0 + * + * As can be determined by integrating the 1D model with a source (s) and + * listener (l) positioned across the dimension of length (d_a): + * + * r_d = int_(l=0)^d_a (int_(s=0)^d_a |2 d_a - 2 (l + s)| ds) dl / c + * + * The initial taps (T_(i=0)^N) are then specified by taking a power series + * that ranges between r_0 and half of r_1 less r_0: + * + * R_i = 2^(i / (2 N - 1)) r_d + * = r_0 + (2^(i / (2 N - 1)) - 1) r_d + * = r_0 + T_i + * T_i = R_i - r_0 + * = (2^(i / (2 N - 1)) - 1) r_d + * + * Assuming an average of 1m, we get the following taps: + */ +constexpr std::array<ALfloat,NUM_LINES> EARLY_TAP_LENGTHS{{ + 0.0000000e+0f, 2.0213520e-4f, 4.2531060e-4f, 6.7171600e-4f +}}; + +/* The early all-pass filter lengths are based on the early tap lengths: + * + * A_i = R_i / a + * + * Where a is the approximate maximum all-pass cycle limit (20). + */ +constexpr std::array<ALfloat,NUM_LINES> EARLY_ALLPASS_LENGTHS{{ + 9.7096800e-5f, 1.0720356e-4f, 1.1836234e-4f, 1.3068260e-4f +}}; + +/* The early delay lines are used to transform the primary reflections into + * the secondary reflections. The A-format is arranged in such a way that + * the channels/lines are spatially opposite: + * + * C_i is opposite C_(N-i-1) + * + * The delays of the two opposing reflections (R_i and O_i) from a source + * anywhere along a particular dimension always sum to twice its full delay: + * + * 2 r_a = R_i + O_i + * + * With that in mind we can determine the delay between the two reflections + * and thus specify our early line lengths (L_(i=0)^N) using: + * + * O_i = 2 r_a - R_(N-i-1) + * L_i = O_i - R_(N-i-1) + * = 2 (r_a - R_(N-i-1)) + * = 2 (r_a - T_(N-i-1) - r_0) + * = 2 r_a (1 - (2 / 3) 2^((N - i - 1) / (2 N - 1))) + * + * Using an average dimension of 1m, we get: + */ +constexpr std::array<ALfloat,NUM_LINES> EARLY_LINE_LENGTHS{{ + 5.9850400e-4f, 1.0913150e-3f, 1.5376658e-3f, 1.9419362e-3f +}}; + +/* The late all-pass filter lengths are based on the late line lengths: + * + * A_i = (5 / 3) L_i / r_1 + */ +constexpr std::array<ALfloat,NUM_LINES> LATE_ALLPASS_LENGTHS{{ + 1.6182800e-4f, 2.0389060e-4f, 2.8159360e-4f, 3.2365600e-4f +}}; +constexpr auto LATE_ALLPASS_LENGTHS_size = LATE_ALLPASS_LENGTHS.size(); + +/* The late lines are used to approximate the decaying cycle of recursive + * late reflections. + * + * Splitting the lines in half, we start with the shortest reflection paths + * (L_(i=0)^(N/2)): + * + * L_i = 2^(i / (N - 1)) r_d + * + * Then for the opposite (longest) reflection paths (L_(i=N/2)^N): + * + * L_i = 2 r_a - L_(i-N/2) + * = 2 r_a - 2^((i - N / 2) / (N - 1)) r_d + * + * For our 1m average room, we get: + */ +constexpr std::array<ALfloat,NUM_LINES> LATE_LINE_LENGTHS{{ + 1.9419362e-3f, 2.4466860e-3f, 3.3791220e-3f, 3.8838720e-3f +}}; +constexpr auto LATE_LINE_LENGTHS_size = LATE_LINE_LENGTHS.size(); + + +struct DelayLineI { + /* The delay lines use interleaved samples, with the lengths being powers + * of 2 to allow the use of bit-masking instead of a modulus for wrapping. + */ + ALsizei Mask{0}; + ALfloat (*Line)[NUM_LINES]{nullptr}; + + + void write(ALsizei offset, const ALsizei c, const ALfloat *RESTRICT in, const ALsizei count) const noexcept + { + ASSUME(count > 0); + for(ALsizei i{0};i < count;) + { + offset &= Mask; + ALsizei td{mini(Mask+1 - offset, count - i)}; + do { + Line[offset++][c] = in[i++]; + } while(--td); + } + } +}; + +struct VecAllpass { + DelayLineI Delay; + ALfloat Coeff{0.0f}; + ALsizei Offset[NUM_LINES][2]{}; + + void processFaded(const al::span<FloatBufferLine,NUM_LINES> samples, ALsizei offset, + const ALfloat xCoeff, const ALfloat yCoeff, ALfloat fade, const ALsizei todo); + void processUnfaded(const al::span<FloatBufferLine,NUM_LINES> samples, ALsizei offset, + const ALfloat xCoeff, const ALfloat yCoeff, const ALsizei todo); +}; + +struct T60Filter { + /* Two filters are used to adjust the signal. One to control the low + * frequencies, and one to control the high frequencies. + */ + ALfloat MidGain[2]{0.0f, 0.0f}; + BiquadFilter HFFilter, LFFilter; + + void calcCoeffs(const ALfloat length, const ALfloat lfDecayTime, const ALfloat mfDecayTime, + const ALfloat hfDecayTime, const ALfloat lf0norm, const ALfloat hf0norm); + + /* Applies the two T60 damping filter sections. */ + void process(ALfloat *samples, const ALsizei todo) + { + HFFilter.process(samples, samples, todo); + LFFilter.process(samples, samples, todo); + } +}; + +struct EarlyReflections { + /* A Gerzon vector all-pass filter is used to simulate initial diffusion. + * The spread from this filter also helps smooth out the reverb tail. + */ + VecAllpass VecAp; + + /* An echo line is used to complete the second half of the early + * reflections. + */ + DelayLineI Delay; + ALsizei Offset[NUM_LINES][2]{}; + ALfloat Coeff[NUM_LINES][2]{}; + + /* The gain for each output channel based on 3D panning. */ + ALfloat CurrentGain[NUM_LINES][MAX_OUTPUT_CHANNELS]{}; + ALfloat PanGain[NUM_LINES][MAX_OUTPUT_CHANNELS]{}; + + void updateLines(const ALfloat density, const ALfloat diffusion, const ALfloat decayTime, + const ALfloat frequency); +}; + +struct LateReverb { + /* A recursive delay line is used fill in the reverb tail. */ + DelayLineI Delay; + ALsizei Offset[NUM_LINES][2]{}; + + /* Attenuation to compensate for the modal density and decay rate of the + * late lines. + */ + ALfloat DensityGain[2]{0.0f, 0.0f}; + + /* T60 decay filters are used to simulate absorption. */ + T60Filter T60[NUM_LINES]; + + /* A Gerzon vector all-pass filter is used to simulate diffusion. */ + VecAllpass VecAp; + + /* The gain for each output channel based on 3D panning. */ + ALfloat CurrentGain[NUM_LINES][MAX_OUTPUT_CHANNELS]{}; + ALfloat PanGain[NUM_LINES][MAX_OUTPUT_CHANNELS]{}; + + void updateLines(const ALfloat density, const ALfloat diffusion, const ALfloat lfDecayTime, + const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lf0norm, + const ALfloat hf0norm, const ALfloat frequency); +}; + +struct ReverbState final : public EffectState { + /* All delay lines are allocated as a single buffer to reduce memory + * fragmentation and management code. + */ + al::vector<ALfloat,16> mSampleBuffer; + + struct { + /* Calculated parameters which indicate if cross-fading is needed after + * an update. + */ + ALfloat Density{AL_EAXREVERB_DEFAULT_DENSITY}; + ALfloat Diffusion{AL_EAXREVERB_DEFAULT_DIFFUSION}; + ALfloat DecayTime{AL_EAXREVERB_DEFAULT_DECAY_TIME}; + ALfloat HFDecayTime{AL_EAXREVERB_DEFAULT_DECAY_HFRATIO * AL_EAXREVERB_DEFAULT_DECAY_TIME}; + ALfloat LFDecayTime{AL_EAXREVERB_DEFAULT_DECAY_LFRATIO * AL_EAXREVERB_DEFAULT_DECAY_TIME}; + ALfloat HFReference{AL_EAXREVERB_DEFAULT_HFREFERENCE}; + ALfloat LFReference{AL_EAXREVERB_DEFAULT_LFREFERENCE}; + } mParams; + + /* Master effect filters */ + struct { + BiquadFilter Lp; + BiquadFilter Hp; + } mFilter[NUM_LINES]; + + /* Core delay line (early reflections and late reverb tap from this). */ + DelayLineI mDelay; + + /* Tap points for early reflection delay. */ + ALsizei mEarlyDelayTap[NUM_LINES][2]{}; + ALfloat mEarlyDelayCoeff[NUM_LINES][2]{}; + + /* Tap points for late reverb feed and delay. */ + ALsizei mLateFeedTap{}; + ALsizei mLateDelayTap[NUM_LINES][2]{}; + + /* Coefficients for the all-pass and line scattering matrices. */ + ALfloat mMixX{0.0f}; + ALfloat mMixY{0.0f}; + + EarlyReflections mEarly; + + LateReverb mLate; + + /* Indicates the cross-fade point for delay line reads [0,FADE_SAMPLES]. */ + ALsizei mFadeCount{0}; + + /* Maximum number of samples to process at once. */ + ALsizei mMaxUpdate[2]{BUFFERSIZE, BUFFERSIZE}; + + /* The current write offset for all delay lines. */ + ALsizei mOffset{0}; + + /* Temporary storage used when processing. */ + alignas(16) std::array<FloatBufferLine,NUM_LINES> mTempSamples{}; + alignas(16) std::array<FloatBufferLine,NUM_LINES> mEarlyBuffer{}; + alignas(16) std::array<FloatBufferLine,NUM_LINES> mLateBuffer{}; + + using MixOutT = void (ReverbState::*)(const al::span<FloatBufferLine> samplesOut, + const ALsizei todo); + + MixOutT mMixOut{&ReverbState::MixOutPlain}; + std::array<ALfloat,MAX_AMBI_ORDER+1> mOrderScales{}; + std::array<std::array<BandSplitter,NUM_LINES>,2> mAmbiSplitter; + + + void MixOutPlain(const al::span<FloatBufferLine> samplesOut, const ALsizei todo) + { + ASSUME(todo > 0); + + /* Convert back to B-Format, and mix the results to output. */ + for(ALsizei c{0};c < NUM_LINES;c++) + { + std::fill_n(mTempSamples[0].begin(), todo, 0.0f); + MixRowSamples(mTempSamples[0], A2B[c], mEarlyBuffer, 0, todo); + MixSamples(mTempSamples[0].data(), samplesOut, mEarly.CurrentGain[c], + mEarly.PanGain[c], todo, 0, todo); + } + + for(ALsizei c{0};c < NUM_LINES;c++) + { + std::fill_n(mTempSamples[0].begin(), todo, 0.0f); + MixRowSamples(mTempSamples[0], A2B[c], mLateBuffer, 0, todo); + MixSamples(mTempSamples[0].data(), samplesOut, mLate.CurrentGain[c], mLate.PanGain[c], + todo, 0, todo); + } + } + + void MixOutAmbiUp(const al::span<FloatBufferLine> samplesOut, const ALsizei todo) + { + ASSUME(todo > 0); + + for(ALsizei c{0};c < NUM_LINES;c++) + { + std::fill_n(mTempSamples[0].begin(), todo, 0.0f); + MixRowSamples(mTempSamples[0], A2B[c], mEarlyBuffer, 0, todo); + + /* Apply scaling to the B-Format's HF response to "upsample" it to + * higher-order output. + */ + const ALfloat hfscale{(c==0) ? mOrderScales[0] : mOrderScales[1]}; + mAmbiSplitter[0][c].applyHfScale(mTempSamples[0].data(), hfscale, todo); + + MixSamples(mTempSamples[0].data(), samplesOut, mEarly.CurrentGain[c], + mEarly.PanGain[c], todo, 0, todo); + } + + for(ALsizei c{0};c < NUM_LINES;c++) + { + std::fill_n(mTempSamples[0].begin(), todo, 0.0f); + MixRowSamples(mTempSamples[0], A2B[c], mLateBuffer, 0, todo); + + const ALfloat hfscale{(c==0) ? mOrderScales[0] : mOrderScales[1]}; + mAmbiSplitter[1][c].applyHfScale(mTempSamples[0].data(), hfscale, todo); + + MixSamples(mTempSamples[0].data(), samplesOut, mLate.CurrentGain[c], mLate.PanGain[c], + todo, 0, todo); + } + } + + bool allocLines(const ALfloat frequency); + + void updateDelayLine(const ALfloat earlyDelay, const ALfloat lateDelay, const ALfloat density, + const ALfloat decayTime, const ALfloat frequency); + void update3DPanning(const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, + const ALfloat earlyGain, const ALfloat lateGain, const EffectTarget &target); + + ALboolean deviceUpdate(const ALCdevice *device) override; + void update(const ALCcontext *context, const ALeffectslot *slot, const EffectProps *props, const EffectTarget target) override; + void process(const ALsizei samplesToDo, const FloatBufferLine *RESTRICT samplesIn, const ALsizei numInput, const al::span<FloatBufferLine> samplesOut) override; + + DEF_NEWDEL(ReverbState) +}; + +/************************************** + * Device Update * + **************************************/ + +inline ALfloat CalcDelayLengthMult(ALfloat density) +{ return maxf(5.0f, std::cbrt(density*DENSITY_SCALE)); } + +/* Given the allocated sample buffer, this function updates each delay line + * offset. + */ +inline ALvoid RealizeLineOffset(ALfloat *sampleBuffer, DelayLineI *Delay) +{ + union { + ALfloat *f; + ALfloat (*f4)[NUM_LINES]; + } u; + u.f = &sampleBuffer[reinterpret_cast<ptrdiff_t>(Delay->Line) * NUM_LINES]; + Delay->Line = u.f4; +} + +/* Calculate the length of a delay line and store its mask and offset. */ +ALuint CalcLineLength(const ALfloat length, const ptrdiff_t offset, const ALfloat frequency, + const ALuint extra, DelayLineI *Delay) +{ + /* All line lengths are powers of 2, calculated from their lengths in + * seconds, rounded up. + */ + auto samples = static_cast<ALuint>(float2int(std::ceil(length*frequency))); + samples = NextPowerOf2(samples + extra); + + /* All lines share a single sample buffer. */ + Delay->Mask = samples - 1; + Delay->Line = reinterpret_cast<ALfloat(*)[NUM_LINES]>(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. + */ +bool ReverbState::allocLines(const ALfloat frequency) +{ + /* All delay line lengths are calculated to accomodate the full range of + * lengths given their respective paramters. + */ + ALuint totalSamples{0u}; + + /* Multiplier for the maximum density value, i.e. density=1, which is + * actually the least density... + */ + ALfloat multiplier{CalcDelayLengthMult(AL_EAXREVERB_MAX_DENSITY)}; + + /* The main delay length includes the maximum early reflection delay, the + * largest early tap width, the maximum late reverb delay, and the + * largest late tap width. Finally, it must also be extended by the + * update size (BUFFERSIZE) for block processing. + */ + ALfloat length{AL_EAXREVERB_MAX_REFLECTIONS_DELAY + EARLY_TAP_LENGTHS.back()*multiplier + + AL_EAXREVERB_MAX_LATE_REVERB_DELAY + + (LATE_LINE_LENGTHS.back() - LATE_LINE_LENGTHS.front())/float{LATE_LINE_LENGTHS_size}*multiplier}; + totalSamples += CalcLineLength(length, totalSamples, frequency, BUFFERSIZE, &mDelay); + + /* The early vector all-pass line. */ + length = EARLY_ALLPASS_LENGTHS.back() * multiplier; + totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &mEarly.VecAp.Delay); + + /* The early reflection line. */ + length = EARLY_LINE_LENGTHS.back() * multiplier; + totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &mEarly.Delay); + + /* The late vector all-pass line. */ + length = LATE_ALLPASS_LENGTHS.back() * multiplier; + totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &mLate.VecAp.Delay); + + /* The late delay lines are calculated from the largest maximum density + * line length. + */ + length = LATE_LINE_LENGTHS.back() * multiplier; + totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &mLate.Delay); + + totalSamples *= NUM_LINES; + if(totalSamples != mSampleBuffer.size()) + { + mSampleBuffer.resize(totalSamples); + mSampleBuffer.shrink_to_fit(); + } + + /* Clear the sample buffer. */ + std::fill(mSampleBuffer.begin(), mSampleBuffer.end(), 0.0f); + + /* Update all delays to reflect the new sample buffer. */ + RealizeLineOffset(mSampleBuffer.data(), &mDelay); + RealizeLineOffset(mSampleBuffer.data(), &mEarly.VecAp.Delay); + RealizeLineOffset(mSampleBuffer.data(), &mEarly.Delay); + RealizeLineOffset(mSampleBuffer.data(), &mLate.VecAp.Delay); + RealizeLineOffset(mSampleBuffer.data(), &mLate.Delay); + + return true; +} + +ALboolean ReverbState::deviceUpdate(const ALCdevice *device) +{ + const auto frequency = static_cast<ALfloat>(device->Frequency); + + /* Allocate the delay lines. */ + if(!allocLines(frequency)) + return AL_FALSE; + + const ALfloat multiplier{CalcDelayLengthMult(AL_EAXREVERB_MAX_DENSITY)}; + + /* The late feed taps are set a fixed position past the latest delay tap. */ + mLateFeedTap = float2int( + (AL_EAXREVERB_MAX_REFLECTIONS_DELAY + EARLY_TAP_LENGTHS.back()*multiplier) * frequency); + + /* Clear filters and gain coefficients since the delay lines were all just + * cleared (if not reallocated). + */ + for(auto &filter : mFilter) + { + filter.Lp.clear(); + filter.Hp.clear(); + } + + for(auto &coeff : mEarlyDelayCoeff) + std::fill(std::begin(coeff), std::end(coeff), 0.0f); + for(auto &coeff : mEarly.Coeff) + std::fill(std::begin(coeff), std::end(coeff), 0.0f); + + mLate.DensityGain[0] = 0.0f; + mLate.DensityGain[1] = 0.0f; + for(auto &t60 : mLate.T60) + { + t60.MidGain[0] = 0.0f; + t60.MidGain[1] = 0.0f; + t60.HFFilter.clear(); + t60.LFFilter.clear(); + } + + for(auto &gains : mEarly.CurrentGain) + std::fill(std::begin(gains), std::end(gains), 0.0f); + for(auto &gains : mEarly.PanGain) + std::fill(std::begin(gains), std::end(gains), 0.0f); + for(auto &gains : mLate.CurrentGain) + std::fill(std::begin(gains), std::end(gains), 0.0f); + for(auto &gains : mLate.PanGain) + std::fill(std::begin(gains), std::end(gains), 0.0f); + + /* Reset counters and offset base. */ + mFadeCount = 0; + std::fill(std::begin(mMaxUpdate), std::end(mMaxUpdate), BUFFERSIZE); + mOffset = 0; + + if(device->mAmbiOrder > 1) + { + mMixOut = &ReverbState::MixOutAmbiUp; + mOrderScales = BFormatDec::GetHFOrderScales(1, device->mAmbiOrder); + } + else + { + mMixOut = &ReverbState::MixOutPlain; + mOrderScales.fill(1.0f); + } + mAmbiSplitter[0][0].init(400.0f / frequency); + std::fill(mAmbiSplitter[0].begin()+1, mAmbiSplitter[0].end(), mAmbiSplitter[0][0]); + std::fill(mAmbiSplitter[1].begin(), mAmbiSplitter[1].end(), mAmbiSplitter[0][0]); + + return AL_TRUE; +} + +/************************************** + * Effect Update * + **************************************/ + +/* Calculate a decay coefficient given the length of each cycle and the time + * until the decay reaches -60 dB. + */ +inline ALfloat CalcDecayCoeff(const ALfloat length, const ALfloat decayTime) +{ return std::pow(REVERB_DECAY_GAIN, length/decayTime); } + +/* Calculate a decay length from a coefficient and the time until the decay + * reaches -60 dB. + */ +inline ALfloat CalcDecayLength(const ALfloat coeff, const ALfloat decayTime) +{ return std::log10(coeff) * decayTime / std::log10(REVERB_DECAY_GAIN); } + +/* Calculate an attenuation to be applied to the input of any echo models to + * compensate for modal density and decay time. + */ +inline ALfloat CalcDensityGain(const 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 area under the squared 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 std::sqrt(1.0f - a*a); +} + +/* Calculate the scattering matrix coefficients given a diffusion factor. */ +inline ALvoid CalcMatrixCoeffs(const ALfloat diffusion, ALfloat *x, ALfloat *y) +{ + /* The matrix is of order 4, so n is sqrt(4 - 1). */ + ALfloat n{std::sqrt(3.0f)}; + ALfloat t{diffusion * std::atan(n)}; + + /* Calculate the first mixing matrix coefficient. */ + *x = std::cos(t); + /* Calculate the second mixing matrix coefficient. */ + *y = std::sin(t) / n; +} + +/* Calculate the limited HF ratio for use with the late reverb low-pass + * filters. + */ +ALfloat CalcLimitedHfRatio(const ALfloat hfRatio, const ALfloat airAbsorptionGainHF, + const ALfloat decayTime, const ALfloat SpeedOfSound) +{ + /* 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. + */ + ALfloat limitRatio{1.0f / (CalcDecayLength(airAbsorptionGainHF, decayTime) * SpeedOfSound)}; + + /* Using the limit calculated above, apply the upper bound to the HF ratio. + */ + return minf(limitRatio, hfRatio); +} + + +/* Calculates the 3-band T60 damping coefficients for a particular delay line + * of specified length, using a combination of two shelf filter sections given + * decay times for each band split at two reference frequencies. + */ +void T60Filter::calcCoeffs(const ALfloat length, const ALfloat lfDecayTime, + const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lf0norm, + const ALfloat hf0norm) +{ + const ALfloat mfGain{CalcDecayCoeff(length, mfDecayTime)}; + const ALfloat lfGain{maxf(CalcDecayCoeff(length, lfDecayTime)/mfGain, 0.001f)}; + const ALfloat hfGain{maxf(CalcDecayCoeff(length, hfDecayTime)/mfGain, 0.001f)}; + + MidGain[1] = mfGain; + LFFilter.setParams(BiquadType::LowShelf, lfGain, lf0norm, + LFFilter.rcpQFromSlope(lfGain, 1.0f)); + HFFilter.setParams(BiquadType::HighShelf, hfGain, hf0norm, + HFFilter.rcpQFromSlope(hfGain, 1.0f)); +} + +/* Update the early reflection line lengths and gain coefficients. */ +void EarlyReflections::updateLines(const ALfloat density, const ALfloat diffusion, + const ALfloat decayTime, const ALfloat frequency) +{ + const ALfloat multiplier{CalcDelayLengthMult(density)}; + + /* Calculate the all-pass feed-back/forward coefficient. */ + VecAp.Coeff = std::sqrt(0.5f) * std::pow(diffusion, 2.0f); + + for(ALsizei i{0};i < NUM_LINES;i++) + { + /* Calculate the length (in seconds) of each all-pass line. */ + ALfloat length{EARLY_ALLPASS_LENGTHS[i] * multiplier}; + + /* Calculate the delay offset for each all-pass line. */ + VecAp.Offset[i][1] = float2int(length * frequency); + + /* Calculate the length (in seconds) of each delay line. */ + length = EARLY_LINE_LENGTHS[i] * multiplier; + + /* Calculate the delay offset for each delay line. */ + Offset[i][1] = float2int(length * frequency); + + /* Calculate the gain (coefficient) for each line. */ + Coeff[i][1] = CalcDecayCoeff(length, decayTime); + } +} + +/* Update the late reverb line lengths and T60 coefficients. */ +void LateReverb::updateLines(const ALfloat density, const ALfloat diffusion, + const ALfloat lfDecayTime, const ALfloat mfDecayTime, const ALfloat hfDecayTime, + const ALfloat lf0norm, const ALfloat hf0norm, const ALfloat frequency) +{ + /* Scaling factor to convert the normalized reference frequencies from + * representing 0...freq to 0...max_reference. + */ + const ALfloat norm_weight_factor{frequency / AL_EAXREVERB_MAX_HFREFERENCE}; + + const ALfloat late_allpass_avg{ + std::accumulate(LATE_ALLPASS_LENGTHS.begin(), LATE_ALLPASS_LENGTHS.end(), 0.0f) / + float{LATE_ALLPASS_LENGTHS_size}}; + + /* 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 delay lines is used to calculate the + * attenuation coefficient. + */ + const ALfloat multiplier{CalcDelayLengthMult(density)}; + ALfloat length{std::accumulate(LATE_LINE_LENGTHS.begin(), LATE_LINE_LENGTHS.end(), 0.0f) / + float{LATE_LINE_LENGTHS_size} * multiplier}; + length += late_allpass_avg * multiplier; + /* The density gain calculation uses an average decay time weighted by + * approximate bandwidth. This attempts to compensate for losses of energy + * that reduce decay time due to scattering into highly attenuated bands. + */ + const ALfloat bandWeights[3]{ + lf0norm*norm_weight_factor, + hf0norm*norm_weight_factor - lf0norm*norm_weight_factor, + 1.0f - hf0norm*norm_weight_factor}; + DensityGain[1] = CalcDensityGain( + CalcDecayCoeff(length, + bandWeights[0]*lfDecayTime + bandWeights[1]*mfDecayTime + bandWeights[2]*hfDecayTime + ) + ); + + /* Calculate the all-pass feed-back/forward coefficient. */ + VecAp.Coeff = std::sqrt(0.5f) * std::pow(diffusion, 2.0f); + + for(ALsizei i{0};i < NUM_LINES;i++) + { + /* Calculate the length (in seconds) of each all-pass line. */ + length = LATE_ALLPASS_LENGTHS[i] * multiplier; + + /* Calculate the delay offset for each all-pass line. */ + VecAp.Offset[i][1] = float2int(length * frequency); + + /* Calculate the length (in seconds) of each delay line. */ + length = LATE_LINE_LENGTHS[i] * multiplier; + + /* Calculate the delay offset for each delay line. */ + Offset[i][1] = float2int(length*frequency + 0.5f); + + /* Approximate the absorption that the vector all-pass would exhibit + * given the current diffusion so we don't have to process a full T60 + * filter for each of its four lines. + */ + length += lerp(LATE_ALLPASS_LENGTHS[i], late_allpass_avg, diffusion) * multiplier; + + /* Calculate the T60 damping coefficients for each line. */ + T60[i].calcCoeffs(length, lfDecayTime, mfDecayTime, hfDecayTime, lf0norm, hf0norm); + } +} + + +/* Update the offsets for the main effect delay line. */ +void ReverbState::updateDelayLine(const ALfloat earlyDelay, const ALfloat lateDelay, + const ALfloat density, const ALfloat decayTime, const ALfloat frequency) +{ + const ALfloat multiplier{CalcDelayLengthMult(density)}; + + /* Early reflection taps are decorrelated by means of an average room + * reflection approximation described above the definition of the taps. + * This approximation is linear and so the above density multiplier can + * be applied to adjust the width of the taps. A single-band decay + * coefficient is applied to simulate initial attenuation and absorption. + * + * Late reverb taps are based on the late line lengths to allow a zero- + * delay path and offsets that would continue the propagation naturally + * into the late lines. + */ + for(ALsizei i{0};i < NUM_LINES;i++) + { + ALfloat length{earlyDelay + EARLY_TAP_LENGTHS[i]*multiplier}; + mEarlyDelayTap[i][1] = float2int(length * frequency); + + length = EARLY_TAP_LENGTHS[i]*multiplier; + mEarlyDelayCoeff[i][1] = CalcDecayCoeff(length, decayTime); + + length = lateDelay + (LATE_LINE_LENGTHS[i] - LATE_LINE_LENGTHS.front()) / + float{LATE_LINE_LENGTHS_size} * multiplier; + mLateDelayTap[i][1] = mLateFeedTap + float2int(length * frequency); + } +} + +/* Creates a transform matrix given a reverb vector. The vector pans the reverb + * reflections toward the given direction, using its magnitude (up to 1) as a + * focal strength. This function results in a B-Format transformation matrix + * that spatially focuses the signal in the desired direction. + */ +alu::Matrix GetTransformFromVector(const ALfloat *vec) +{ + /* Normalize the panning vector according to the N3D scale, which has an + * extra sqrt(3) term on the directional components. Converting from OpenAL + * to B-Format also requires negating X (ACN 1) and Z (ACN 3). Note however + * that the reverb panning vectors use left-handed coordinates, unlike the + * rest of OpenAL which use right-handed. This is fixed by negating Z, + * which cancels out with the B-Format Z negation. + */ + ALfloat norm[3]; + ALfloat mag{std::sqrt(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2])}; + if(mag > 1.0f) + { + norm[0] = vec[0] / mag * -al::MathDefs<float>::Sqrt3(); + norm[1] = vec[1] / mag * al::MathDefs<float>::Sqrt3(); + norm[2] = vec[2] / mag * al::MathDefs<float>::Sqrt3(); + mag = 1.0f; + } + else + { + /* If the magnitude is less than or equal to 1, just apply the sqrt(3) + * term. There's no need to renormalize the magnitude since it would + * just be reapplied in the matrix. + */ + norm[0] = vec[0] * -al::MathDefs<float>::Sqrt3(); + norm[1] = vec[1] * al::MathDefs<float>::Sqrt3(); + norm[2] = vec[2] * al::MathDefs<float>::Sqrt3(); + } + + return alu::Matrix{ + 1.0f, 0.0f, 0.0f, 0.0f, + norm[0], 1.0f-mag, 0.0f, 0.0f, + norm[1], 0.0f, 1.0f-mag, 0.0f, + norm[2], 0.0f, 0.0f, 1.0f-mag + }; +} + +/* Update the early and late 3D panning gains. */ +void ReverbState::update3DPanning(const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, + const ALfloat earlyGain, const ALfloat lateGain, const EffectTarget &target) +{ + /* Create matrices that transform a B-Format signal according to the + * panning vectors. + */ + const alu::Matrix earlymat{GetTransformFromVector(ReflectionsPan)}; + const alu::Matrix latemat{GetTransformFromVector(LateReverbPan)}; + + mOutTarget = target.Main->Buffer; + for(ALsizei i{0};i < NUM_LINES;i++) + { + const ALfloat coeffs[MAX_AMBI_CHANNELS]{earlymat[0][i], earlymat[1][i], earlymat[2][i], + earlymat[3][i]}; + ComputePanGains(target.Main, coeffs, earlyGain, mEarly.PanGain[i]); + } + for(ALsizei i{0};i < NUM_LINES;i++) + { + const ALfloat coeffs[MAX_AMBI_CHANNELS]{latemat[0][i], latemat[1][i], latemat[2][i], + latemat[3][i]}; + ComputePanGains(target.Main, coeffs, lateGain, mLate.PanGain[i]); + } +} + +void ReverbState::update(const ALCcontext *Context, const ALeffectslot *Slot, const EffectProps *props, const EffectTarget target) +{ + const ALCdevice *Device{Context->Device}; + const ALlistener &Listener = Context->Listener; + const auto frequency = static_cast<ALfloat>(Device->Frequency); + + /* Calculate the master filters */ + ALfloat hf0norm{minf(props->Reverb.HFReference / frequency, 0.49f)}; + /* Restrict the filter gains from going below -60dB to keep the filter from + * killing most of the signal. + */ + ALfloat gainhf{maxf(props->Reverb.GainHF, 0.001f)}; + mFilter[0].Lp.setParams(BiquadType::HighShelf, gainhf, hf0norm, + mFilter[0].Lp.rcpQFromSlope(gainhf, 1.0f)); + ALfloat lf0norm{minf(props->Reverb.LFReference / frequency, 0.49f)}; + ALfloat gainlf{maxf(props->Reverb.GainLF, 0.001f)}; + mFilter[0].Hp.setParams(BiquadType::LowShelf, gainlf, lf0norm, + mFilter[0].Hp.rcpQFromSlope(gainlf, 1.0f)); + for(ALsizei i{1};i < NUM_LINES;i++) + { + mFilter[i].Lp.copyParamsFrom(mFilter[0].Lp); + mFilter[i].Hp.copyParamsFrom(mFilter[0].Hp); + } + + /* Update the main effect delay and associated taps. */ + updateDelayLine(props->Reverb.ReflectionsDelay, props->Reverb.LateReverbDelay, + props->Reverb.Density, props->Reverb.DecayTime, frequency); + + /* Update the early lines. */ + mEarly.updateLines(props->Reverb.Density, props->Reverb.Diffusion, props->Reverb.DecayTime, + frequency); + + /* Get the mixing matrix coefficients. */ + CalcMatrixCoeffs(props->Reverb.Diffusion, &mMixX, &mMixY); + + /* If the HF limit parameter is flagged, calculate an appropriate limit + * based on the air absorption parameter. + */ + ALfloat hfRatio{props->Reverb.DecayHFRatio}; + if(props->Reverb.DecayHFLimit && props->Reverb.AirAbsorptionGainHF < 1.0f) + hfRatio = CalcLimitedHfRatio(hfRatio, props->Reverb.AirAbsorptionGainHF, + props->Reverb.DecayTime, Listener.Params.ReverbSpeedOfSound + ); + + /* Calculate the LF/HF decay times. */ + const ALfloat lfDecayTime{clampf(props->Reverb.DecayTime * props->Reverb.DecayLFRatio, + AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME)}; + const ALfloat hfDecayTime{clampf(props->Reverb.DecayTime * hfRatio, + AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME)}; + + /* Update the late lines. */ + mLate.updateLines(props->Reverb.Density, props->Reverb.Diffusion, lfDecayTime, + props->Reverb.DecayTime, hfDecayTime, lf0norm, hf0norm, frequency); + + /* Update early and late 3D panning. */ + const ALfloat gain{props->Reverb.Gain * Slot->Params.Gain * ReverbBoost}; + update3DPanning(props->Reverb.ReflectionsPan, props->Reverb.LateReverbPan, + props->Reverb.ReflectionsGain*gain, props->Reverb.LateReverbGain*gain, target); + + /* Calculate the max update size from the smallest relevant delay. */ + mMaxUpdate[1] = mini(BUFFERSIZE, mini(mEarly.Offset[0][1], mLate.Offset[0][1])); + + /* Determine if delay-line cross-fading is required. Density is essentially + * a master control for the feedback delays, so changes the offsets of many + * delay lines. + */ + if(mParams.Density != props->Reverb.Density || + /* Diffusion and decay times influences the decay rate (gain) of the + * late reverb T60 filter. + */ + mParams.Diffusion != props->Reverb.Diffusion || + mParams.DecayTime != props->Reverb.DecayTime || + mParams.HFDecayTime != hfDecayTime || + mParams.LFDecayTime != lfDecayTime || + /* HF/LF References control the weighting used to calculate the density + * gain. + */ + mParams.HFReference != props->Reverb.HFReference || + mParams.LFReference != props->Reverb.LFReference) + mFadeCount = 0; + mParams.Density = props->Reverb.Density; + mParams.Diffusion = props->Reverb.Diffusion; + mParams.DecayTime = props->Reverb.DecayTime; + mParams.HFDecayTime = hfDecayTime; + mParams.LFDecayTime = lfDecayTime; + mParams.HFReference = props->Reverb.HFReference; + mParams.LFReference = props->Reverb.LFReference; +} + + +/************************************** + * Effect Processing * + **************************************/ + +/* Applies a scattering matrix to the 4-line (vector) input. This is used + * for both the below vector all-pass model and to perform modal feed-back + * delay network (FDN) mixing. + * + * The matrix is derived from a skew-symmetric matrix to form a 4D rotation + * matrix with 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 final 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 + * + * Any square orthogonal matrix with an order that is a power of two will + * work (where ^T is transpose, ^-1 is inverse): + * + * M^T = M^-1 + * + * Using that knowledge, finding an appropriate matrix can be accomplished + * naively by searching all combinations of: + * + * M = D + S - S^T + * + * Where D is a diagonal matrix (of x), and S is a triangular matrix (of y) + * whose combination of signs are being iterated. + */ +inline void VectorPartialScatter(ALfloat *RESTRICT out, const ALfloat *RESTRICT in, + const ALfloat xCoeff, const ALfloat yCoeff) +{ + out[0] = xCoeff*in[0] + yCoeff*( in[1] + -in[2] + in[3]); + out[1] = xCoeff*in[1] + yCoeff*(-in[0] + in[2] + in[3]); + out[2] = xCoeff*in[2] + yCoeff*( in[0] + -in[1] + in[3]); + out[3] = xCoeff*in[3] + yCoeff*(-in[0] + -in[1] + -in[2] ); +} + +/* Utilizes the above, but reverses the input channels. */ +void VectorScatterRevDelayIn(const DelayLineI delay, ALint offset, const ALfloat xCoeff, + const ALfloat yCoeff, const ALsizei base, const al::span<const FloatBufferLine,NUM_LINES> in, + const ALsizei count) +{ + ASSUME(base >= 0); + ASSUME(count > 0); + + for(ALsizei i{0};i < count;) + { + offset &= delay.Mask; + ALsizei td{mini(delay.Mask+1 - offset, count-i)}; + do { + ALfloat f[NUM_LINES]; + for(ALsizei j{0};j < NUM_LINES;j++) + f[NUM_LINES-1-j] = in[j][base+i]; + ++i; + + VectorPartialScatter(delay.Line[offset++], f, xCoeff, yCoeff); + } while(--td); + } +} + +/* This applies a Gerzon multiple-in/multiple-out (MIMO) vector all-pass + * filter to the 4-line input. + * + * It works by vectorizing a regular all-pass filter and replacing the delay + * element with a scattering matrix (like the one above) and a diagonal + * matrix of delay elements. + * + * Two static specializations are used for transitional (cross-faded) delay + * line processing and non-transitional processing. + */ +void VecAllpass::processUnfaded(const al::span<FloatBufferLine,NUM_LINES> samples, ALsizei offset, + const ALfloat xCoeff, const ALfloat yCoeff, const ALsizei todo) +{ + const DelayLineI delay{Delay}; + const ALfloat feedCoeff{Coeff}; + + ASSUME(todo > 0); + + ALsizei vap_offset[NUM_LINES]; + for(ALsizei j{0};j < NUM_LINES;j++) + vap_offset[j] = offset - Offset[j][0]; + for(ALsizei i{0};i < todo;) + { + for(ALsizei j{0};j < NUM_LINES;j++) + vap_offset[j] &= delay.Mask; + offset &= delay.Mask; + + ALsizei maxoff{offset}; + for(ALsizei j{0};j < NUM_LINES;j++) + maxoff = maxi(maxoff, vap_offset[j]); + ALsizei td{mini(delay.Mask+1 - maxoff, todo - i)}; + + do { + ALfloat f[NUM_LINES]; + for(ALsizei j{0};j < NUM_LINES;j++) + { + const ALfloat input{samples[j][i]}; + const ALfloat out{delay.Line[vap_offset[j]++][j] - feedCoeff*input}; + f[j] = input + feedCoeff*out; + + samples[j][i] = out; + } + ++i; + + VectorPartialScatter(delay.Line[offset++], f, xCoeff, yCoeff); + } while(--td); + } +} +void VecAllpass::processFaded(const al::span<FloatBufferLine,NUM_LINES> samples, ALsizei offset, + const ALfloat xCoeff, const ALfloat yCoeff, ALfloat fade, const ALsizei todo) +{ + const DelayLineI delay{Delay}; + const ALfloat feedCoeff{Coeff}; + + ASSUME(todo > 0); + + fade *= 1.0f/FADE_SAMPLES; + ALsizei vap_offset[NUM_LINES][2]; + for(ALsizei j{0};j < NUM_LINES;j++) + { + vap_offset[j][0] = offset - Offset[j][0]; + vap_offset[j][1] = offset - Offset[j][1]; + } + for(ALsizei i{0};i < todo;) + { + for(ALsizei j{0};j < NUM_LINES;j++) + { + vap_offset[j][0] &= delay.Mask; + vap_offset[j][1] &= delay.Mask; + } + offset &= delay.Mask; + + ALsizei maxoff{offset}; + for(ALsizei j{0};j < NUM_LINES;j++) + maxoff = maxi(maxoff, maxi(vap_offset[j][0], vap_offset[j][1])); + ALsizei td{mini(delay.Mask+1 - maxoff, todo - i)}; + + do { + fade += FadeStep; + ALfloat f[NUM_LINES]; + for(ALsizei j{0};j < NUM_LINES;j++) + f[j] = delay.Line[vap_offset[j][0]++][j]*(1.0f-fade) + + delay.Line[vap_offset[j][1]++][j]*fade; + + for(ALsizei j{0};j < NUM_LINES;j++) + { + const ALfloat input{samples[j][i]}; + const ALfloat out{f[j] - feedCoeff*input}; + f[j] = input + feedCoeff*out; + + samples[j][i] = out; + } + ++i; + + VectorPartialScatter(delay.Line[offset++], f, xCoeff, yCoeff); + } while(--td); + } +} + +/* This generates early reflections. + * + * This is done by obtaining the primary reflections (those arriving from the + * same direction as the source) from the main delay line. These are + * attenuated and all-pass filtered (based on the diffusion parameter). + * + * The early lines are then fed in reverse (according to the approximately + * opposite spatial location of the A-Format lines) to create the secondary + * reflections (those arriving from the opposite direction as the source). + * + * The early response is then completed by combining the primary reflections + * with the delayed and attenuated output from the early lines. + * + * Finally, the early response is reversed, scattered (based on diffusion), + * and fed into the late reverb section of the main delay line. + * + * Two static specializations are used for transitional (cross-faded) delay + * line processing and non-transitional processing. + */ +void EarlyReflection_Unfaded(ReverbState *State, const ALsizei offset, const ALsizei todo, + const ALsizei base, const al::span<FloatBufferLine,NUM_LINES> out) +{ + const al::span<FloatBufferLine,NUM_LINES> temps{State->mTempSamples}; + const DelayLineI early_delay{State->mEarly.Delay}; + const DelayLineI main_delay{State->mDelay}; + const ALfloat mixX{State->mMixX}; + const ALfloat mixY{State->mMixY}; + + ASSUME(todo > 0); + + /* First, load decorrelated samples from the main delay line as the primary + * reflections. + */ + for(ALsizei j{0};j < NUM_LINES;j++) + { + ALsizei early_delay_tap{offset - State->mEarlyDelayTap[j][0]}; + const ALfloat coeff{State->mEarlyDelayCoeff[j][0]}; + for(ALsizei i{0};i < todo;) + { + early_delay_tap &= main_delay.Mask; + ALsizei td{mini(main_delay.Mask+1 - early_delay_tap, todo - i)}; + do { + temps[j][i++] = main_delay.Line[early_delay_tap++][j] * coeff; + } while(--td); + } + } + + /* Apply a vector all-pass, to help color the initial reflections based on + * the diffusion strength. + */ + State->mEarly.VecAp.processUnfaded(temps, offset, mixX, mixY, todo); + + /* Apply a delay and bounce to generate secondary reflections, combine with + * the primary reflections and write out the result for mixing. + */ + for(ALsizei j{0};j < NUM_LINES;j++) + { + ALint feedb_tap{offset - State->mEarly.Offset[j][0]}; + const ALfloat feedb_coeff{State->mEarly.Coeff[j][0]}; + + ASSUME(base >= 0); + for(ALsizei i{0};i < todo;) + { + feedb_tap &= early_delay.Mask; + ALsizei td{mini(early_delay.Mask+1 - feedb_tap, todo - i)}; + do { + out[j][base+i] = temps[j][i] + early_delay.Line[feedb_tap++][j]*feedb_coeff; + ++i; + } while(--td); + } + } + for(ALsizei j{0};j < NUM_LINES;j++) + early_delay.write(offset, NUM_LINES-1-j, temps[j].data(), todo); + + /* Also write the result back to the main delay line for the late reverb + * stage to pick up at the appropriate time, appplying a scatter and + * bounce to improve the initial diffusion in the late reverb. + */ + const ALsizei late_feed_tap{offset - State->mLateFeedTap}; + VectorScatterRevDelayIn(main_delay, late_feed_tap, mixX, mixY, base, + {out.cbegin(), out.cend()}, todo); +} +void EarlyReflection_Faded(ReverbState *State, const ALsizei offset, const ALsizei todo, + const ALfloat fade, const ALsizei base, const al::span<FloatBufferLine,NUM_LINES> out) +{ + const al::span<FloatBufferLine,NUM_LINES> temps{State->mTempSamples}; + const DelayLineI early_delay{State->mEarly.Delay}; + const DelayLineI main_delay{State->mDelay}; + const ALfloat mixX{State->mMixX}; + const ALfloat mixY{State->mMixY}; + + ASSUME(todo > 0); + + for(ALsizei j{0};j < NUM_LINES;j++) + { + ALsizei early_delay_tap0{offset - State->mEarlyDelayTap[j][0]}; + ALsizei early_delay_tap1{offset - State->mEarlyDelayTap[j][1]}; + const ALfloat oldCoeff{State->mEarlyDelayCoeff[j][0]}; + const ALfloat oldCoeffStep{-oldCoeff / FADE_SAMPLES}; + const ALfloat newCoeffStep{State->mEarlyDelayCoeff[j][1] / FADE_SAMPLES}; + ALfloat fadeCount{fade}; + + for(ALsizei i{0};i < todo;) + { + early_delay_tap0 &= main_delay.Mask; + early_delay_tap1 &= main_delay.Mask; + ALsizei td{mini(main_delay.Mask+1 - maxi(early_delay_tap0, early_delay_tap1), todo-i)}; + do { + fadeCount += 1.0f; + const ALfloat fade0{oldCoeff + oldCoeffStep*fadeCount}; + const ALfloat fade1{newCoeffStep*fadeCount}; + temps[j][i++] = + main_delay.Line[early_delay_tap0++][j]*fade0 + + main_delay.Line[early_delay_tap1++][j]*fade1; + } while(--td); + } + } + + State->mEarly.VecAp.processFaded(temps, offset, mixX, mixY, fade, todo); + + for(ALsizei j{0};j < NUM_LINES;j++) + { + ALint feedb_tap0{offset - State->mEarly.Offset[j][0]}; + ALint feedb_tap1{offset - State->mEarly.Offset[j][1]}; + const ALfloat feedb_oldCoeff{State->mEarly.Coeff[j][0]}; + const ALfloat feedb_oldCoeffStep{-feedb_oldCoeff / FADE_SAMPLES}; + const ALfloat feedb_newCoeffStep{State->mEarly.Coeff[j][1] / FADE_SAMPLES}; + ALfloat fadeCount{fade}; + + ASSUME(base >= 0); + for(ALsizei i{0};i < todo;) + { + feedb_tap0 &= early_delay.Mask; + feedb_tap1 &= early_delay.Mask; + ALsizei td{mini(early_delay.Mask+1 - maxi(feedb_tap0, feedb_tap1), todo - i)}; + + do { + fadeCount += 1.0f; + const ALfloat fade0{feedb_oldCoeff + feedb_oldCoeffStep*fadeCount}; + const ALfloat fade1{feedb_newCoeffStep*fadeCount}; + out[j][base+i] = temps[j][i] + + early_delay.Line[feedb_tap0++][j]*fade0 + + early_delay.Line[feedb_tap1++][j]*fade1; + ++i; + } while(--td); + } + } + for(ALsizei j{0};j < NUM_LINES;j++) + early_delay.write(offset, NUM_LINES-1-j, temps[j].data(), todo); + + const ALsizei late_feed_tap{offset - State->mLateFeedTap}; + VectorScatterRevDelayIn(main_delay, late_feed_tap, mixX, mixY, base, + {out.cbegin(), out.cend()}, todo); +} + +/* This generates the reverb tail using a modified feed-back delay network + * (FDN). + * + * Results from the early reflections are mixed with the output from the late + * delay lines. + * + * The late response is then completed by T60 and all-pass filtering the mix. + * + * Finally, the lines are reversed (so they feed their opposite directions) + * and scattered with the FDN matrix before re-feeding the delay lines. + * + * Two variations are made, one for for transitional (cross-faded) delay line + * processing and one for non-transitional processing. + */ +void LateReverb_Unfaded(ReverbState *State, const ALsizei offset, const ALsizei todo, + const ALsizei base, const al::span<FloatBufferLine,NUM_LINES> out) +{ + const al::span<FloatBufferLine,NUM_LINES> temps{State->mTempSamples}; + const DelayLineI late_delay{State->mLate.Delay}; + const DelayLineI main_delay{State->mDelay}; + const ALfloat mixX{State->mMixX}; + const ALfloat mixY{State->mMixY}; + + ASSUME(todo > 0); + + /* First, load decorrelated samples from the main and feedback delay lines. + * Filter the signal to apply its frequency-dependent decay. + */ + for(ALsizei j{0};j < NUM_LINES;j++) + { + ALsizei late_delay_tap{offset - State->mLateDelayTap[j][0]}; + ALsizei late_feedb_tap{offset - State->mLate.Offset[j][0]}; + const ALfloat midGain{State->mLate.T60[j].MidGain[0]}; + const ALfloat densityGain{State->mLate.DensityGain[0] * midGain}; + for(ALsizei i{0};i < todo;) + { + late_delay_tap &= main_delay.Mask; + late_feedb_tap &= late_delay.Mask; + ALsizei td{mini( + mini(main_delay.Mask+1 - late_delay_tap, late_delay.Mask+1 - late_feedb_tap), + todo - i)}; + do { + temps[j][i++] = + main_delay.Line[late_delay_tap++][j]*densityGain + + late_delay.Line[late_feedb_tap++][j]*midGain; + } while(--td); + } + State->mLate.T60[j].process(temps[j].data(), todo); + } + + /* Apply a vector all-pass to improve micro-surface diffusion, and write + * out the results for mixing. + */ + State->mLate.VecAp.processUnfaded(temps, offset, mixX, mixY, todo); + + for(ALsizei j{0};j < NUM_LINES;j++) + std::copy_n(temps[j].begin(), todo, out[j].begin()+base); + + /* Finally, scatter and bounce the results to refeed the feedback buffer. */ + VectorScatterRevDelayIn(late_delay, offset, mixX, mixY, base, + {out.cbegin(), out.cend()}, todo); +} +void LateReverb_Faded(ReverbState *State, const ALsizei offset, const ALsizei todo, + const ALfloat fade, const ALsizei base, const al::span<FloatBufferLine,NUM_LINES> out) +{ + const al::span<FloatBufferLine,NUM_LINES> temps{State->mTempSamples}; + const DelayLineI late_delay{State->mLate.Delay}; + const DelayLineI main_delay{State->mDelay}; + const ALfloat mixX{State->mMixX}; + const ALfloat mixY{State->mMixY}; + + ASSUME(todo > 0); + + for(ALsizei j{0};j < NUM_LINES;j++) + { + const ALfloat oldMidGain{State->mLate.T60[j].MidGain[0]}; + const ALfloat midGain{State->mLate.T60[j].MidGain[1]}; + const ALfloat oldMidStep{-oldMidGain / FADE_SAMPLES}; + const ALfloat midStep{midGain / FADE_SAMPLES}; + const ALfloat oldDensityGain{State->mLate.DensityGain[0] * oldMidGain}; + const ALfloat densityGain{State->mLate.DensityGain[1] * midGain}; + const ALfloat oldDensityStep{-oldDensityGain / FADE_SAMPLES}; + const ALfloat densityStep{densityGain / FADE_SAMPLES}; + ALsizei late_delay_tap0{offset - State->mLateDelayTap[j][0]}; + ALsizei late_delay_tap1{offset - State->mLateDelayTap[j][1]}; + ALsizei late_feedb_tap0{offset - State->mLate.Offset[j][0]}; + ALsizei late_feedb_tap1{offset - State->mLate.Offset[j][1]}; + ALfloat fadeCount{fade}; + + for(ALsizei i{0};i < todo;) + { + late_delay_tap0 &= main_delay.Mask; + late_delay_tap1 &= main_delay.Mask; + late_feedb_tap0 &= late_delay.Mask; + late_feedb_tap1 &= late_delay.Mask; + ALsizei td{mini( + mini(main_delay.Mask+1 - maxi(late_delay_tap0, late_delay_tap1), + late_delay.Mask+1 - maxi(late_feedb_tap0, late_feedb_tap1)), + todo - i)}; + do { + fadeCount += 1.0f; + const ALfloat fade0{oldDensityGain + oldDensityStep*fadeCount}; + const ALfloat fade1{densityStep*fadeCount}; + const ALfloat gfade0{oldMidGain + oldMidStep*fadeCount}; + const ALfloat gfade1{midStep*fadeCount}; + temps[j][i++] = + main_delay.Line[late_delay_tap0++][j]*fade0 + + main_delay.Line[late_delay_tap1++][j]*fade1 + + late_delay.Line[late_feedb_tap0++][j]*gfade0 + + late_delay.Line[late_feedb_tap1++][j]*gfade1; + } while(--td); + } + State->mLate.T60[j].process(temps[j].data(), todo); + } + + State->mLate.VecAp.processFaded(temps, offset, mixX, mixY, fade, todo); + + for(ALsizei j{0};j < NUM_LINES;j++) + std::copy_n(temps[j].begin(), todo, out[j].begin()+base); + + VectorScatterRevDelayIn(late_delay, offset, mixX, mixY, base, + {out.cbegin(), out.cend()}, todo); +} + +void ReverbState::process(const ALsizei samplesToDo, const FloatBufferLine *RESTRICT samplesIn, const ALsizei numInput, const al::span<FloatBufferLine> samplesOut) +{ + ALsizei fadeCount{mFadeCount}; + + ASSUME(samplesToDo > 0); + + /* Convert B-Format to A-Format for processing. */ + const al::span<FloatBufferLine,NUM_LINES> afmt{mTempSamples}; + for(ALsizei c{0};c < NUM_LINES;c++) + { + std::fill_n(afmt[c].begin(), samplesToDo, 0.0f); + MixRowSamples(afmt[c], B2A[c], {samplesIn, samplesIn+numInput}, 0, samplesToDo); + + /* Band-pass the incoming samples. */ + mFilter[c].Lp.process(afmt[c].data(), afmt[c].data(), samplesToDo); + mFilter[c].Hp.process(afmt[c].data(), afmt[c].data(), samplesToDo); + } + + /* Process reverb for these samples. */ + for(ALsizei base{0};base < samplesToDo;) + { + ALsizei todo{samplesToDo - base}; + /* If cross-fading, don't do more samples than there are to fade. */ + if(FADE_SAMPLES-fadeCount > 0) + { + todo = mini(todo, FADE_SAMPLES-fadeCount); + todo = mini(todo, mMaxUpdate[0]); + } + todo = mini(todo, mMaxUpdate[1]); + ASSUME(todo > 0 && todo <= BUFFERSIZE); + + const ALsizei offset{mOffset + base}; + ASSUME(offset >= 0); + + /* Feed the initial delay line. */ + for(ALsizei c{0};c < NUM_LINES;c++) + mDelay.write(offset, c, afmt[c].data()+base, todo); + + /* Process the samples for reverb. */ + if(UNLIKELY(fadeCount < FADE_SAMPLES)) + { + auto fade = static_cast<ALfloat>(fadeCount); + + /* Generate early reflections and late reverb. */ + EarlyReflection_Faded(this, offset, todo, fade, base, mEarlyBuffer); + + LateReverb_Faded(this, offset, todo, fade, base, mLateBuffer); + + /* Step fading forward. */ + fadeCount += todo; + if(fadeCount >= FADE_SAMPLES) + { + /* Update the cross-fading delay line taps. */ + fadeCount = FADE_SAMPLES; + for(ALsizei c{0};c < NUM_LINES;c++) + { + mEarlyDelayTap[c][0] = mEarlyDelayTap[c][1]; + mEarlyDelayCoeff[c][0] = mEarlyDelayCoeff[c][1]; + mEarly.VecAp.Offset[c][0] = mEarly.VecAp.Offset[c][1]; + mEarly.Offset[c][0] = mEarly.Offset[c][1]; + mEarly.Coeff[c][0] = mEarly.Coeff[c][1]; + mLateDelayTap[c][0] = mLateDelayTap[c][1]; + mLate.VecAp.Offset[c][0] = mLate.VecAp.Offset[c][1]; + mLate.Offset[c][0] = mLate.Offset[c][1]; + mLate.T60[c].MidGain[0] = mLate.T60[c].MidGain[1]; + } + mLate.DensityGain[0] = mLate.DensityGain[1]; + mMaxUpdate[0] = mMaxUpdate[1]; + } + } + else + { + /* Generate early reflections and late reverb. */ + EarlyReflection_Unfaded(this, offset, todo, base, mEarlyBuffer); + + LateReverb_Unfaded(this, offset, todo, base, mLateBuffer); + } + + base += todo; + } + mOffset = (mOffset+samplesToDo) & 0x3fffffff; + mFadeCount = fadeCount; + + /* Finally, mix early reflections and late reverb. */ + (this->*mMixOut)(samplesOut, samplesToDo); +} + + +void EAXReverb_setParami(EffectProps *props, ALCcontext *context, ALenum param, ALint val) +{ + switch(param) + { + case AL_EAXREVERB_DECAY_HFLIMIT: + if(!(val >= AL_EAXREVERB_MIN_DECAY_HFLIMIT && val <= AL_EAXREVERB_MAX_DECAY_HFLIMIT)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay hflimit out of range"); + props->Reverb.DecayHFLimit = val; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb integer property 0x%04x", + param); + } +} +void EAXReverb_setParamiv(EffectProps *props, ALCcontext *context, ALenum param, const ALint *vals) +{ EAXReverb_setParami(props, context, param, vals[0]); } +void EAXReverb_setParamf(EffectProps *props, ALCcontext *context, ALenum param, ALfloat val) +{ + switch(param) + { + case AL_EAXREVERB_DENSITY: + if(!(val >= AL_EAXREVERB_MIN_DENSITY && val <= AL_EAXREVERB_MAX_DENSITY)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb density out of range"); + props->Reverb.Density = val; + break; + + case AL_EAXREVERB_DIFFUSION: + if(!(val >= AL_EAXREVERB_MIN_DIFFUSION && val <= AL_EAXREVERB_MAX_DIFFUSION)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb diffusion out of range"); + props->Reverb.Diffusion = val; + break; + + case AL_EAXREVERB_GAIN: + if(!(val >= AL_EAXREVERB_MIN_GAIN && val <= AL_EAXREVERB_MAX_GAIN)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gain out of range"); + props->Reverb.Gain = val; + break; + + case AL_EAXREVERB_GAINHF: + if(!(val >= AL_EAXREVERB_MIN_GAINHF && val <= AL_EAXREVERB_MAX_GAINHF)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gainhf out of range"); + props->Reverb.GainHF = val; + break; + + case AL_EAXREVERB_GAINLF: + if(!(val >= AL_EAXREVERB_MIN_GAINLF && val <= AL_EAXREVERB_MAX_GAINLF)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gainlf out of range"); + props->Reverb.GainLF = val; + break; + + case AL_EAXREVERB_DECAY_TIME: + if(!(val >= AL_EAXREVERB_MIN_DECAY_TIME && val <= AL_EAXREVERB_MAX_DECAY_TIME)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay time out of range"); + props->Reverb.DecayTime = val; + break; + + case AL_EAXREVERB_DECAY_HFRATIO: + if(!(val >= AL_EAXREVERB_MIN_DECAY_HFRATIO && val <= AL_EAXREVERB_MAX_DECAY_HFRATIO)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay hfratio out of range"); + props->Reverb.DecayHFRatio = val; + break; + + case AL_EAXREVERB_DECAY_LFRATIO: + if(!(val >= AL_EAXREVERB_MIN_DECAY_LFRATIO && val <= AL_EAXREVERB_MAX_DECAY_LFRATIO)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay lfratio out of range"); + props->Reverb.DecayLFRatio = val; + break; + + case AL_EAXREVERB_REFLECTIONS_GAIN: + if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_GAIN && val <= AL_EAXREVERB_MAX_REFLECTIONS_GAIN)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections gain out of range"); + props->Reverb.ReflectionsGain = val; + break; + + case AL_EAXREVERB_REFLECTIONS_DELAY: + if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_DELAY && val <= AL_EAXREVERB_MAX_REFLECTIONS_DELAY)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections delay out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb gain out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb delay out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb air absorption gainhf out of range"); + props->Reverb.AirAbsorptionGainHF = val; + break; + + case AL_EAXREVERB_ECHO_TIME: + if(!(val >= AL_EAXREVERB_MIN_ECHO_TIME && val <= AL_EAXREVERB_MAX_ECHO_TIME)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb echo time out of range"); + props->Reverb.EchoTime = val; + break; + + case AL_EAXREVERB_ECHO_DEPTH: + if(!(val >= AL_EAXREVERB_MIN_ECHO_DEPTH && val <= AL_EAXREVERB_MAX_ECHO_DEPTH)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb echo depth out of range"); + props->Reverb.EchoDepth = val; + break; + + case AL_EAXREVERB_MODULATION_TIME: + if(!(val >= AL_EAXREVERB_MIN_MODULATION_TIME && val <= AL_EAXREVERB_MAX_MODULATION_TIME)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb modulation time out of range"); + props->Reverb.ModulationTime = val; + break; + + case AL_EAXREVERB_MODULATION_DEPTH: + if(!(val >= AL_EAXREVERB_MIN_MODULATION_DEPTH && val <= AL_EAXREVERB_MAX_MODULATION_DEPTH)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb modulation depth out of range"); + props->Reverb.ModulationDepth = val; + break; + + case AL_EAXREVERB_HFREFERENCE: + if(!(val >= AL_EAXREVERB_MIN_HFREFERENCE && val <= AL_EAXREVERB_MAX_HFREFERENCE)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb hfreference out of range"); + props->Reverb.HFReference = val; + break; + + case AL_EAXREVERB_LFREFERENCE: + if(!(val >= AL_EAXREVERB_MIN_LFREFERENCE && val <= AL_EAXREVERB_MAX_LFREFERENCE)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb lfreference out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb room rolloff factor out of range"); + props->Reverb.RoomRolloffFactor = val; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb float property 0x%04x", + param); + } +} +void EAXReverb_setParamfv(EffectProps *props, ALCcontext *context, ALenum param, const ALfloat *vals) +{ + switch(param) + { + case AL_EAXREVERB_REFLECTIONS_PAN: + if(!(std::isfinite(vals[0]) && std::isfinite(vals[1]) && std::isfinite(vals[2]))) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections pan out of range"); + 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(!(std::isfinite(vals[0]) && std::isfinite(vals[1]) && std::isfinite(vals[2]))) + SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb pan out of range"); + props->Reverb.LateReverbPan[0] = vals[0]; + props->Reverb.LateReverbPan[1] = vals[1]; + props->Reverb.LateReverbPan[2] = vals[2]; + break; + + default: + EAXReverb_setParamf(props, context, param, vals[0]); + break; + } +} + +void EAXReverb_getParami(const EffectProps *props, ALCcontext *context, ALenum param, ALint *val) +{ + switch(param) + { + case AL_EAXREVERB_DECAY_HFLIMIT: + *val = props->Reverb.DecayHFLimit; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb integer property 0x%04x", + param); + } +} +void EAXReverb_getParamiv(const EffectProps *props, ALCcontext *context, ALenum param, ALint *vals) +{ EAXReverb_getParami(props, context, param, vals); } +void EAXReverb_getParamf(const EffectProps *props, ALCcontext *context, ALenum param, ALfloat *val) +{ + 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: + alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb float property 0x%04x", + param); + } +} +void EAXReverb_getParamfv(const EffectProps *props, ALCcontext *context, ALenum param, ALfloat *vals) +{ + 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: + EAXReverb_getParamf(props, context, param, vals); + break; + } +} + +DEFINE_ALEFFECT_VTABLE(EAXReverb); + + +struct ReverbStateFactory final : public EffectStateFactory { + EffectState *create() override { return new ReverbState{}; } + EffectProps getDefaultProps() const noexcept override; + const EffectVtable *getEffectVtable() const noexcept override { return &EAXReverb_vtable; } +}; + +EffectProps ReverbStateFactory::getDefaultProps() const noexcept +{ + EffectProps props{}; + props.Reverb.Density = AL_EAXREVERB_DEFAULT_DENSITY; + props.Reverb.Diffusion = AL_EAXREVERB_DEFAULT_DIFFUSION; + props.Reverb.Gain = AL_EAXREVERB_DEFAULT_GAIN; + props.Reverb.GainHF = AL_EAXREVERB_DEFAULT_GAINHF; + props.Reverb.GainLF = AL_EAXREVERB_DEFAULT_GAINLF; + props.Reverb.DecayTime = AL_EAXREVERB_DEFAULT_DECAY_TIME; + props.Reverb.DecayHFRatio = AL_EAXREVERB_DEFAULT_DECAY_HFRATIO; + props.Reverb.DecayLFRatio = AL_EAXREVERB_DEFAULT_DECAY_LFRATIO; + props.Reverb.ReflectionsGain = AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN; + props.Reverb.ReflectionsDelay = AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY; + props.Reverb.ReflectionsPan[0] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ; + props.Reverb.ReflectionsPan[1] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ; + props.Reverb.ReflectionsPan[2] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ; + props.Reverb.LateReverbGain = AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN; + props.Reverb.LateReverbDelay = AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY; + props.Reverb.LateReverbPan[0] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ; + props.Reverb.LateReverbPan[1] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ; + props.Reverb.LateReverbPan[2] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ; + props.Reverb.EchoTime = AL_EAXREVERB_DEFAULT_ECHO_TIME; + props.Reverb.EchoDepth = AL_EAXREVERB_DEFAULT_ECHO_DEPTH; + props.Reverb.ModulationTime = AL_EAXREVERB_DEFAULT_MODULATION_TIME; + props.Reverb.ModulationDepth = AL_EAXREVERB_DEFAULT_MODULATION_DEPTH; + props.Reverb.AirAbsorptionGainHF = AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF; + props.Reverb.HFReference = AL_EAXREVERB_DEFAULT_HFREFERENCE; + props.Reverb.LFReference = AL_EAXREVERB_DEFAULT_LFREFERENCE; + props.Reverb.RoomRolloffFactor = AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR; + props.Reverb.DecayHFLimit = AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT; + return props; +} + + +void StdReverb_setParami(EffectProps *props, ALCcontext *context, ALenum param, ALint val) +{ + switch(param) + { + case AL_REVERB_DECAY_HFLIMIT: + if(!(val >= AL_REVERB_MIN_DECAY_HFLIMIT && val <= AL_REVERB_MAX_DECAY_HFLIMIT)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay hflimit out of range"); + props->Reverb.DecayHFLimit = val; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid reverb integer property 0x%04x", param); + } +} +void StdReverb_setParamiv(EffectProps *props, ALCcontext *context, ALenum param, const ALint *vals) +{ StdReverb_setParami(props, context, param, vals[0]); } +void StdReverb_setParamf(EffectProps *props, ALCcontext *context, ALenum param, ALfloat val) +{ + switch(param) + { + case AL_REVERB_DENSITY: + if(!(val >= AL_REVERB_MIN_DENSITY && val <= AL_REVERB_MAX_DENSITY)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb density out of range"); + props->Reverb.Density = val; + break; + + case AL_REVERB_DIFFUSION: + if(!(val >= AL_REVERB_MIN_DIFFUSION && val <= AL_REVERB_MAX_DIFFUSION)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb diffusion out of range"); + props->Reverb.Diffusion = val; + break; + + case AL_REVERB_GAIN: + if(!(val >= AL_REVERB_MIN_GAIN && val <= AL_REVERB_MAX_GAIN)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb gain out of range"); + props->Reverb.Gain = val; + break; + + case AL_REVERB_GAINHF: + if(!(val >= AL_REVERB_MIN_GAINHF && val <= AL_REVERB_MAX_GAINHF)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb gainhf out of range"); + props->Reverb.GainHF = val; + break; + + case AL_REVERB_DECAY_TIME: + if(!(val >= AL_REVERB_MIN_DECAY_TIME && val <= AL_REVERB_MAX_DECAY_TIME)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay time out of range"); + props->Reverb.DecayTime = val; + break; + + case AL_REVERB_DECAY_HFRATIO: + if(!(val >= AL_REVERB_MIN_DECAY_HFRATIO && val <= AL_REVERB_MAX_DECAY_HFRATIO)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay hfratio out of range"); + props->Reverb.DecayHFRatio = val; + break; + + case AL_REVERB_REFLECTIONS_GAIN: + if(!(val >= AL_REVERB_MIN_REFLECTIONS_GAIN && val <= AL_REVERB_MAX_REFLECTIONS_GAIN)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb reflections gain out of range"); + props->Reverb.ReflectionsGain = val; + break; + + case AL_REVERB_REFLECTIONS_DELAY: + if(!(val >= AL_REVERB_MIN_REFLECTIONS_DELAY && val <= AL_REVERB_MAX_REFLECTIONS_DELAY)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb reflections delay out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb late reverb gain out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb late reverb delay out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb air absorption gainhf out of range"); + 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)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb room rolloff factor out of range"); + props->Reverb.RoomRolloffFactor = val; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid reverb float property 0x%04x", param); + } +} +void StdReverb_setParamfv(EffectProps *props, ALCcontext *context, ALenum param, const ALfloat *vals) +{ StdReverb_setParamf(props, context, param, vals[0]); } + +void StdReverb_getParami(const EffectProps *props, ALCcontext *context, ALenum param, ALint *val) +{ + switch(param) + { + case AL_REVERB_DECAY_HFLIMIT: + *val = props->Reverb.DecayHFLimit; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid reverb integer property 0x%04x", param); + } +} +void StdReverb_getParamiv(const EffectProps *props, ALCcontext *context, ALenum param, ALint *vals) +{ StdReverb_getParami(props, context, param, vals); } +void StdReverb_getParamf(const EffectProps *props, ALCcontext *context, ALenum param, ALfloat *val) +{ + 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: + alSetError(context, AL_INVALID_ENUM, "Invalid reverb float property 0x%04x", param); + } +} +void StdReverb_getParamfv(const EffectProps *props, ALCcontext *context, ALenum param, ALfloat *vals) +{ StdReverb_getParamf(props, context, param, vals); } + +DEFINE_ALEFFECT_VTABLE(StdReverb); + + +struct StdReverbStateFactory final : public EffectStateFactory { + EffectState *create() override { return new ReverbState{}; } + EffectProps getDefaultProps() const noexcept override; + const EffectVtable *getEffectVtable() const noexcept override { return &StdReverb_vtable; } +}; + +EffectProps StdReverbStateFactory::getDefaultProps() const noexcept +{ + EffectProps props{}; + props.Reverb.Density = AL_REVERB_DEFAULT_DENSITY; + props.Reverb.Diffusion = AL_REVERB_DEFAULT_DIFFUSION; + props.Reverb.Gain = AL_REVERB_DEFAULT_GAIN; + props.Reverb.GainHF = AL_REVERB_DEFAULT_GAINHF; + props.Reverb.GainLF = 1.0f; + props.Reverb.DecayTime = AL_REVERB_DEFAULT_DECAY_TIME; + props.Reverb.DecayHFRatio = AL_REVERB_DEFAULT_DECAY_HFRATIO; + props.Reverb.DecayLFRatio = 1.0f; + props.Reverb.ReflectionsGain = AL_REVERB_DEFAULT_REFLECTIONS_GAIN; + props.Reverb.ReflectionsDelay = AL_REVERB_DEFAULT_REFLECTIONS_DELAY; + props.Reverb.ReflectionsPan[0] = 0.0f; + props.Reverb.ReflectionsPan[1] = 0.0f; + props.Reverb.ReflectionsPan[2] = 0.0f; + props.Reverb.LateReverbGain = AL_REVERB_DEFAULT_LATE_REVERB_GAIN; + props.Reverb.LateReverbDelay = AL_REVERB_DEFAULT_LATE_REVERB_DELAY; + props.Reverb.LateReverbPan[0] = 0.0f; + props.Reverb.LateReverbPan[1] = 0.0f; + props.Reverb.LateReverbPan[2] = 0.0f; + props.Reverb.EchoTime = 0.25f; + props.Reverb.EchoDepth = 0.0f; + props.Reverb.ModulationTime = 0.25f; + props.Reverb.ModulationDepth = 0.0f; + props.Reverb.AirAbsorptionGainHF = AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF; + props.Reverb.HFReference = 5000.0f; + props.Reverb.LFReference = 250.0f; + props.Reverb.RoomRolloffFactor = AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR; + props.Reverb.DecayHFLimit = AL_REVERB_DEFAULT_DECAY_HFLIMIT; + return props; +} + +} // namespace + +EffectStateFactory *ReverbStateFactory_getFactory() +{ + static ReverbStateFactory ReverbFactory{}; + return &ReverbFactory; +} + +EffectStateFactory *StdReverbStateFactory_getFactory() +{ + static StdReverbStateFactory ReverbFactory{}; + return &ReverbFactory; +} |