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axmol/thirdparty/openal/alc/effects/reverb.cpp

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/**
* 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 <algorithm>
#include <array>
#include <cstdio>
#include <functional>
#include <iterator>
#include <numeric>
#include <stdint.h>
#include "alc/effects/base.h"
#include "almalloc.h"
#include "alnumbers.h"
#include "alnumeric.h"
#include "alspan.h"
#include "core/ambidefs.h"
#include "core/bufferline.h"
#include "core/context.h"
#include "core/devformat.h"
#include "core/device.h"
#include "core/effectslot.h"
#include "core/filters/biquad.h"
#include "core/filters/splitter.h"
#include "core/mixer.h"
#include "core/mixer/defs.h"
#include "intrusive_ptr.h"
#include "opthelpers.h"
#include "vecmat.h"
#include "vector.h"
/* This is a user config option for modifying the overall output of the reverb
* effect.
*/
float ReverbBoost = 1.0f;
namespace {
using uint = unsigned int;
constexpr float MaxModulationTime{4.0f};
constexpr float DefaultModulationTime{0.25f};
#define MOD_FRACBITS 24
#define MOD_FRACONE (1<<MOD_FRACBITS)
#define MOD_FRACMASK (MOD_FRACONE-1)
struct CubicFilter {
static constexpr size_t sTableBits{8};
static constexpr size_t sTableSteps{1 << sTableBits};
static constexpr size_t sTableMask{sTableSteps - 1};
float mFilter[sTableSteps*2 + 1]{};
constexpr CubicFilter()
{
/* This creates a lookup table for a cubic spline filter, with 256
* steps between samples. Only half the coefficients are needed, since
* Coeff2 is just Coeff1 in reverse and Coeff3 is just Coeff0 in
* reverse.
*/
for(size_t i{0};i < sTableSteps;++i)
{
const double mu{static_cast<double>(i) / double{sTableSteps}};
const double mu2{mu*mu}, mu3{mu2*mu};
const double a0{-0.5*mu3 + mu2 + -0.5*mu};
const double a1{ 1.5*mu3 + -2.5*mu2 + 1.0f};
mFilter[i] = static_cast<float>(a1);
mFilter[sTableSteps+i] = static_cast<float>(a0);
}
}
constexpr float getCoeff0(size_t i) const noexcept { return mFilter[sTableSteps+i]; }
constexpr float getCoeff1(size_t i) const noexcept { return mFilter[i]; }
constexpr float getCoeff2(size_t i) const noexcept { return mFilter[sTableSteps-i]; }
constexpr float getCoeff3(size_t i) const noexcept { return mFilter[sTableSteps*2-i]; }
};
constexpr CubicFilter gCubicTable;
using namespace std::placeholders;
/* Max samples per process iteration. Used to limit the size needed for
* temporary buffers. Must be a multiple of 4 for SIMD alignment.
*/
constexpr size_t MAX_UPDATE_SAMPLES{256};
/* 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 size_t NUM_LINES{4u};
/* This coefficient is used to define the maximum frequency range controlled by
* the modulation depth. The current value of 0.05 will allow it to swing from
* 0.95x to 1.05x. This value must be below 1. At 1 it will cause the sampler
* to stall on the downswing, and above 1 it will cause it to sample backwards.
* The value 0.05 seems be nearest to Creative hardware behavior.
*/
constexpr float MODULATION_DEPTH_COEFF{0.05f};
/* 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 float B2A[NUM_LINES][NUM_LINES]{
{ 0.5f, 0.5f, 0.5f, 0.5f },
{ 0.5f, -0.5f, -0.5f, 0.5f },
{ 0.5f, 0.5f, -0.5f, -0.5f },
{ 0.5f, -0.5f, 0.5f, -0.5f }
};
/* Converts A-Format to B-Format for early reflections. */
alignas(16) constexpr std::array<std::array<float,NUM_LINES>,NUM_LINES> EarlyA2B{{
{{ 0.5f, 0.5f, 0.5f, 0.5f }},
{{ 0.5f, -0.5f, 0.5f, -0.5f }},
{{ 0.5f, -0.5f, -0.5f, 0.5f }},
{{ 0.5f, 0.5f, -0.5f, -0.5f }}
}};
/* Converts A-Format to B-Format for late reverb. */
constexpr auto InvSqrt2 = static_cast<float>(1.0/al::numbers::sqrt2);
alignas(16) constexpr std::array<std::array<float,NUM_LINES>,NUM_LINES> LateA2B{{
{{ 0.5f, 0.5f, 0.5f, 0.5f }},
{{ InvSqrt2, -InvSqrt2, 0.0f, 0.0f }},
{{ 0.0f, 0.0f, InvSqrt2, -InvSqrt2 }},
{{ 0.5f, 0.5f, -0.5f, -0.5f }}
}};
/* 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 float 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<float,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<float,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<float,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<float,NUM_LINES> LATE_ALLPASS_LENGTHS{{
1.6182800e-4f, 2.0389060e-4f, 2.8159360e-4f, 3.2365600e-4f
}};
/* 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<float,NUM_LINES> LATE_LINE_LENGTHS{{
1.9419362e-3f, 2.4466860e-3f, 3.3791220e-3f, 3.8838720e-3f
}};
using ReverbUpdateLine = std::array<float,MAX_UPDATE_SAMPLES>;
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.
*/
size_t Mask{0u};
union {
uintptr_t LineOffset{0u};
std::array<float,NUM_LINES> *Line;
};
/* Given the allocated sample buffer, this function updates each delay line
* offset.
*/
void realizeLineOffset(std::array<float,NUM_LINES> *sampleBuffer) noexcept
{ Line = sampleBuffer + LineOffset; }
/* Calculate the length of a delay line and store its mask and offset. */
uint calcLineLength(const float length, const uintptr_t offset, const float frequency,
const uint extra)
{
/* All line lengths are powers of 2, calculated from their lengths in
* seconds, rounded up.
*/
uint samples{float2uint(std::ceil(length*frequency))};
samples = NextPowerOf2(samples + extra);
/* All lines share a single sample buffer. */
Mask = samples - 1;
LineOffset = offset;
/* Return the sample count for accumulation. */
return samples;
}
void write(size_t offset, const size_t c, const float *RESTRICT in, const size_t count) const noexcept
{
ASSUME(count > 0);
for(size_t i{0u};i < count;)
{
offset &= Mask;
size_t td{minz(Mask+1 - offset, count - i)};
do {
Line[offset++][c] = in[i++];
} while(--td);
}
}
};
struct VecAllpass {
DelayLineI Delay;
float Coeff{0.0f};
size_t Offset[NUM_LINES]{};
void process(const al::span<ReverbUpdateLine,NUM_LINES> samples, size_t offset,
const float xCoeff, const float yCoeff, const size_t todo);
};
struct T60Filter {
/* Two filters are used to adjust the signal. One to control the low
* frequencies, and one to control the high frequencies.
*/
float MidGain{0.0f};
BiquadFilter HFFilter, LFFilter;
void calcCoeffs(const float length, const float lfDecayTime, const float mfDecayTime,
const float hfDecayTime, const float lf0norm, const float hf0norm);
/* Applies the two T60 damping filter sections. */
void process(const al::span<float> samples)
{ DualBiquad{HFFilter, LFFilter}.process(samples, samples.data()); }
void clear() noexcept { HFFilter.clear(); LFFilter.clear(); }
};
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;
size_t Offset[NUM_LINES]{};
float Coeff[NUM_LINES]{};
/* The gain for each output channel based on 3D panning. */
float CurrentGains[NUM_LINES][MaxAmbiChannels]{};
float TargetGains[NUM_LINES][MaxAmbiChannels]{};
void updateLines(const float density_mult, const float diffusion, const float decayTime,
const float frequency);
};
struct Modulation {
/* The vibrato time is tracked with an index over a (MOD_FRACONE)
* normalized range.
*/
uint Index, Step;
/* The depth of frequency change, in samples. */
float Depth;
float ModDelays[MAX_UPDATE_SAMPLES];
void updateModulator(float modTime, float modDepth, float frequency);
void calcDelays(size_t todo);
};
struct LateReverb {
/* A recursive delay line is used fill in the reverb tail. */
DelayLineI Delay;
size_t Offset[NUM_LINES]{};
/* Attenuation to compensate for the modal density and decay rate of the
* late lines.
*/
float DensityGain{0.0f};
/* T60 decay filters are used to simulate absorption. */
T60Filter T60[NUM_LINES];
Modulation Mod;
/* A Gerzon vector all-pass filter is used to simulate diffusion. */
VecAllpass VecAp;
/* The gain for each output channel based on 3D panning. */
float CurrentGains[NUM_LINES][MaxAmbiChannels]{};
float TargetGains[NUM_LINES][MaxAmbiChannels]{};
void updateLines(const float density_mult, const float diffusion, const float lfDecayTime,
const float mfDecayTime, const float hfDecayTime, const float lf0norm,
const float hf0norm, const float frequency);
void clear() noexcept
{
for(auto &filter : T60)
filter.clear();
}
};
struct ReverbPipeline {
/* Master effect filters */
struct {
BiquadFilter Lp;
BiquadFilter Hp;
} mFilter[NUM_LINES];
/* Core delay line (early reflections and late reverb tap from this). */
DelayLineI mEarlyDelayIn;
DelayLineI mLateDelayIn;
/* Tap points for early reflection delay. */
size_t mEarlyDelayTap[NUM_LINES][2]{};
float mEarlyDelayCoeff[NUM_LINES]{};
/* Tap points for late reverb feed and delay. */
size_t mLateDelayTap[NUM_LINES][2]{};
/* Coefficients for the all-pass and line scattering matrices. */
float mMixX{0.0f};
float mMixY{0.0f};
EarlyReflections mEarly;
LateReverb mLate;
std::array<std::array<BandSplitter,NUM_LINES>,2> mAmbiSplitter;
size_t mFadeSampleCount{1};
void updateDelayLine(const float earlyDelay, const float lateDelay, const float density_mult,
const float decayTime, const float frequency);
void update3DPanning(const float *ReflectionsPan, const float *LateReverbPan,
const float earlyGain, const float lateGain, const bool doUpmix, const MixParams *mainMix);
void processEarly(size_t offset, const size_t samplesToDo,
const al::span<ReverbUpdateLine,NUM_LINES> tempSamples,
const al::span<FloatBufferLine,NUM_LINES> outSamples);
void processLate(size_t offset, const size_t samplesToDo,
const al::span<ReverbUpdateLine,NUM_LINES> tempSamples,
const al::span<FloatBufferLine,NUM_LINES> outSamples);
void clear() noexcept
{
for(auto &filter : mFilter)
{
filter.Lp.clear();
filter.Hp.clear();
}
mLate.clear();
for(auto &filters : mAmbiSplitter)
{
for(auto &filter : filters)
filter.clear();
}
}
};
struct ReverbState final : public EffectState {
/* All delay lines are allocated as a single buffer to reduce memory
* fragmentation and management code.
*/
al::vector<std::array<float,NUM_LINES>,16> mSampleBuffer;
struct {
/* Calculated parameters which indicate if cross-fading is needed after
* an update.
*/
float Density{1.0f};
float Diffusion{1.0f};
float DecayTime{1.49f};
float HFDecayTime{0.83f * 1.49f};
float LFDecayTime{1.0f * 1.49f};
float ModulationTime{0.25f};
float ModulationDepth{0.0f};
float HFReference{5000.0f};
float LFReference{250.0f};
} mParams;
enum PipelineState : uint8_t {
DeviceClear,
StartFade,
Fading,
Cleanup,
Normal,
};
PipelineState mPipelineState{DeviceClear};
uint8_t mCurrentPipeline{0};
ReverbPipeline mPipelines[2];
/* The current write offset for all delay lines. */
size_t mOffset{};
/* Temporary storage used when processing. */
union {
alignas(16) FloatBufferLine mTempLine{};
alignas(16) std::array<ReverbUpdateLine,NUM_LINES> mTempSamples;
};
alignas(16) std::array<FloatBufferLine,NUM_LINES> mEarlySamples{};
alignas(16) std::array<FloatBufferLine,NUM_LINES> mLateSamples{};
std::array<float,MaxAmbiOrder+1> mOrderScales{};
bool mUpmixOutput{false};
void MixOutPlain(ReverbPipeline &pipeline, const al::span<FloatBufferLine> samplesOut,
const size_t todo)
{
ASSUME(todo > 0);
/* When not upsampling, the panning gains convert to B-Format and pan
* at the same time.
*/
for(size_t c{0u};c < NUM_LINES;c++)
{
const al::span<float> tmpspan{mEarlySamples[c].data(), todo};
MixSamples(tmpspan, samplesOut, pipeline.mEarly.CurrentGains[c],
pipeline.mEarly.TargetGains[c], todo, 0);
}
for(size_t c{0u};c < NUM_LINES;c++)
{
const al::span<float> tmpspan{mLateSamples[c].data(), todo};
MixSamples(tmpspan, samplesOut, pipeline.mLate.CurrentGains[c],
pipeline.mLate.TargetGains[c], todo, 0);
}
}
void MixOutAmbiUp(ReverbPipeline &pipeline, const al::span<FloatBufferLine> samplesOut,
const size_t todo)
{
ASSUME(todo > 0);
auto DoMixRow = [](const al::span<float> OutBuffer, const al::span<const float,4> Gains,
const float *InSamples, const size_t InStride)
{
std::fill(OutBuffer.begin(), OutBuffer.end(), 0.0f);
for(const float gain : Gains)
{
const float *RESTRICT input{al::assume_aligned<16>(InSamples)};
InSamples += InStride;
if(!(std::fabs(gain) > GainSilenceThreshold))
continue;
auto mix_sample = [gain](const float sample, const float in) noexcept -> float
{ return sample + in*gain; };
std::transform(OutBuffer.begin(), OutBuffer.end(), input, OutBuffer.begin(),
mix_sample);
}
};
/* When upsampling, the B-Format conversion needs to be done separately
* so the proper HF scaling can be applied to each B-Format channel.
* The panning gains then pan and upsample the B-Format channels.
*/
const al::span<float> tmpspan{al::assume_aligned<16>(mTempLine.data()), todo};
for(size_t c{0u};c < NUM_LINES;c++)
{
DoMixRow(tmpspan, EarlyA2B[c], mEarlySamples[0].data(), mEarlySamples[0].size());
/* Apply scaling to the B-Format's HF response to "upsample" it to
* higher-order output.
*/
const float hfscale{(c==0) ? mOrderScales[0] : mOrderScales[1]};
pipeline.mAmbiSplitter[0][c].processHfScale(tmpspan, hfscale);
MixSamples(tmpspan, samplesOut, pipeline.mEarly.CurrentGains[c],
pipeline.mEarly.TargetGains[c], todo, 0);
}
for(size_t c{0u};c < NUM_LINES;c++)
{
DoMixRow(tmpspan, LateA2B[c], mLateSamples[0].data(), mLateSamples[0].size());
const float hfscale{(c==0) ? mOrderScales[0] : mOrderScales[1]};
pipeline.mAmbiSplitter[1][c].processHfScale(tmpspan, hfscale);
MixSamples(tmpspan, samplesOut, pipeline.mLate.CurrentGains[c],
pipeline.mLate.TargetGains[c], todo, 0);
}
}
void mixOut(ReverbPipeline &pipeline, const al::span<FloatBufferLine> samplesOut, const size_t todo)
{
if(mUpmixOutput)
MixOutAmbiUp(pipeline, samplesOut, todo);
else
MixOutPlain(pipeline, samplesOut, todo);
}
void allocLines(const float frequency);
void deviceUpdate(const DeviceBase *device, const BufferStorage *buffer) override;
void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
const EffectTarget target) override;
void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
const al::span<FloatBufferLine> samplesOut) override;
DEF_NEWDEL(ReverbState)
};
/**************************************
* Device Update *
**************************************/
inline float CalcDelayLengthMult(float density)
{ return maxf(5.0f, std::cbrt(density*DENSITY_SCALE)); }
/* Calculates the delay line metrics and allocates the shared sample buffer
* for all lines given the sample rate (frequency).
*/
void ReverbState::allocLines(const float frequency)
{
/* All delay line lengths are calculated to accomodate the full range of
* lengths given their respective paramters.
*/
size_t totalSamples{0u};
/* Multiplier for the maximum density value, i.e. density=1, which is
* actually the least density...
*/
const float multiplier{CalcDelayLengthMult(1.0f)};
/* The modulator's line length is calculated from the maximum modulation
* time and depth coefficient, and halfed for the low-to-high frequency
* swing.
*/
constexpr float max_mod_delay{MaxModulationTime*MODULATION_DEPTH_COEFF / 2.0f};
for(auto &pipeline : mPipelines)
{
/* 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 (BufferLineSize) for block processing.
*/
float length{ReverbMaxReflectionsDelay + EARLY_TAP_LENGTHS.back()*multiplier};
totalSamples += pipeline.mEarlyDelayIn.calcLineLength(length, totalSamples, frequency,
BufferLineSize);
constexpr float LateLineDiffAvg{(LATE_LINE_LENGTHS.back()-LATE_LINE_LENGTHS.front()) /
float{NUM_LINES}};
length = ReverbMaxLateReverbDelay + LateLineDiffAvg*multiplier;
totalSamples += pipeline.mLateDelayIn.calcLineLength(length, totalSamples, frequency,
BufferLineSize);
/* The early vector all-pass line. */
length = EARLY_ALLPASS_LENGTHS.back() * multiplier;
totalSamples += pipeline.mEarly.VecAp.Delay.calcLineLength(length, totalSamples, frequency,
0);
/* The early reflection line. */
length = EARLY_LINE_LENGTHS.back() * multiplier;
totalSamples += pipeline.mEarly.Delay.calcLineLength(length, totalSamples, frequency,
MAX_UPDATE_SAMPLES);
/* The late vector all-pass line. */
length = LATE_ALLPASS_LENGTHS.back() * multiplier;
totalSamples += pipeline.mLate.VecAp.Delay.calcLineLength(length, totalSamples, frequency,
0);
/* The late delay lines are calculated from the largest maximum density
* line length, and the maximum modulation delay. Four additional
* samples are needed for resampling the modulator delay.
*/
length = LATE_LINE_LENGTHS.back()*multiplier + max_mod_delay;
totalSamples += pipeline.mLate.Delay.calcLineLength(length, totalSamples, frequency, 4);
}
if(totalSamples != mSampleBuffer.size())
decltype(mSampleBuffer)(totalSamples).swap(mSampleBuffer);
/* Clear the sample buffer. */
std::fill(mSampleBuffer.begin(), mSampleBuffer.end(), decltype(mSampleBuffer)::value_type{});
/* Update all delays to reflect the new sample buffer. */
for(auto &pipeline : mPipelines)
{
pipeline.mEarlyDelayIn.realizeLineOffset(mSampleBuffer.data());
pipeline.mLateDelayIn.realizeLineOffset(mSampleBuffer.data());
pipeline.mEarly.VecAp.Delay.realizeLineOffset(mSampleBuffer.data());
pipeline.mEarly.Delay.realizeLineOffset(mSampleBuffer.data());
pipeline.mLate.VecAp.Delay.realizeLineOffset(mSampleBuffer.data());
pipeline.mLate.Delay.realizeLineOffset(mSampleBuffer.data());
}
}
void ReverbState::deviceUpdate(const DeviceBase *device, const BufferStorage*)
{
const auto frequency = static_cast<float>(device->Frequency);
/* Allocate the delay lines. */
allocLines(frequency);
for(auto &pipeline : mPipelines)
{
/* Clear filters and gain coefficients since the delay lines were all just
* cleared (if not reallocated).
*/
for(auto &filter : pipeline.mFilter)
{
filter.Lp.clear();
filter.Hp.clear();
}
std::fill(std::begin(pipeline.mEarlyDelayCoeff),std::end(pipeline.mEarlyDelayCoeff), 0.0f);
std::fill(std::begin(pipeline.mEarlyDelayCoeff),std::end(pipeline.mEarlyDelayCoeff), 0.0f);
pipeline.mLate.DensityGain = 0.0f;
for(auto &t60 : pipeline.mLate.T60)
{
t60.MidGain = 0.0f;
t60.HFFilter.clear();
t60.LFFilter.clear();
}
pipeline.mLate.Mod.Index = 0;
pipeline.mLate.Mod.Step = 1;
pipeline.mLate.Mod.Depth = 0.0f;
for(auto &gains : pipeline.mEarly.CurrentGains)
std::fill(std::begin(gains), std::end(gains), 0.0f);
for(auto &gains : pipeline.mEarly.TargetGains)
std::fill(std::begin(gains), std::end(gains), 0.0f);
for(auto &gains : pipeline.mLate.CurrentGains)
std::fill(std::begin(gains), std::end(gains), 0.0f);
for(auto &gains : pipeline.mLate.TargetGains)
std::fill(std::begin(gains), std::end(gains), 0.0f);
}
mPipelineState = DeviceClear;
/* Reset offset base. */
mOffset = 0;
if(device->mAmbiOrder > 1)
{
mUpmixOutput = true;
mOrderScales = AmbiScale::GetHFOrderScales(1, device->mAmbiOrder, device->m2DMixing);
}
else
{
mUpmixOutput = false;
mOrderScales.fill(1.0f);
}
mPipelines[0].mAmbiSplitter[0][0].init(device->mXOverFreq / frequency);
for(auto &pipeline : mPipelines)
{
std::fill(pipeline.mAmbiSplitter[0].begin(), pipeline.mAmbiSplitter[0].end(),
pipeline.mAmbiSplitter[0][0]);
std::fill(pipeline.mAmbiSplitter[1].begin(), pipeline.mAmbiSplitter[1].end(),
pipeline.mAmbiSplitter[0][0]);
}
}
/**************************************
* Effect Update *
**************************************/
/* Calculate a decay coefficient given the length of each cycle and the time
* until the decay reaches -60 dB.
*/
inline float CalcDecayCoeff(const float length, const float decayTime)
{ return std::pow(ReverbDecayGain, length/decayTime); }
/* Calculate a decay length from a coefficient and the time until the decay
* reaches -60 dB.
*/
inline float CalcDecayLength(const float coeff, const float decayTime)
{
constexpr float log10_decaygain{-3.0f/*std::log10(ReverbDecayGain)*/};
return std::log10(coeff) * decayTime / log10_decaygain;
}
/* Calculate an attenuation to be applied to the input of any echo models to
* compensate for modal density and decay time.
*/
inline float CalcDensityGain(const float 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 void CalcMatrixCoeffs(const float diffusion, float *x, float *y)
{
/* The matrix is of order 4, so n is sqrt(4 - 1). */
constexpr float n{al::numbers::sqrt3_v<float>};
const float 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.
*/
float CalcLimitedHfRatio(const float hfRatio, const float airAbsorptionGainHF,
const float decayTime)
{
/* 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.
*/
float limitRatio{1.0f / SpeedOfSoundMetersPerSec /
CalcDecayLength(airAbsorptionGainHF, decayTime)};
/* 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 float length, const float lfDecayTime,
const float mfDecayTime, const float hfDecayTime, const float lf0norm,
const float hf0norm)
{
const float mfGain{CalcDecayCoeff(length, mfDecayTime)};
const float lfGain{CalcDecayCoeff(length, lfDecayTime) / mfGain};
const float hfGain{CalcDecayCoeff(length, hfDecayTime) / mfGain};
MidGain = mfGain;
LFFilter.setParamsFromSlope(BiquadType::LowShelf, lf0norm, lfGain, 1.0f);
HFFilter.setParamsFromSlope(BiquadType::HighShelf, hf0norm, hfGain, 1.0f);
}
/* Update the early reflection line lengths and gain coefficients. */
void EarlyReflections::updateLines(const float density_mult, const float diffusion,
const float decayTime, const float frequency)
{
/* Calculate the all-pass feed-back/forward coefficient. */
VecAp.Coeff = diffusion*diffusion * InvSqrt2;
for(size_t i{0u};i < NUM_LINES;i++)
{
/* Calculate the delay length of each all-pass line. */
float length{EARLY_ALLPASS_LENGTHS[i] * density_mult};
VecAp.Offset[i] = float2uint(length * frequency);
/* Calculate the delay length of each delay line. */
length = EARLY_LINE_LENGTHS[i] * density_mult;
Offset[i] = float2uint(length * frequency);
/* Calculate the gain (coefficient) for each line. */
Coeff[i] = CalcDecayCoeff(length, decayTime);
}
}
/* Update the EAX modulation step 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.
*/
void Modulation::updateModulator(float modTime, float modDepth, float frequency)
{
/* Modulation is calculated in two parts.
*
* The modulation time effects the sinus rate, altering the speed of
* frequency changes. An index is incremented for each sample with an
* appropriate step size to generate an LFO, which will vary the feedback
* delay over time.
*/
Step = maxu(fastf2u(MOD_FRACONE / (frequency * modTime)), 1);
/* 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
* halved 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).
*/
if(modTime >= DefaultModulationTime)
{
/* To cancel the effects of a long period modulation on the late
* reverberation, the amount of pitch should be varied (decreased)
* according to the modulation time. The natural form is varying
* inversely, in fact resulting in an invariant.
*/
Depth = MODULATION_DEPTH_COEFF / 4.0f * DefaultModulationTime * modDepth * frequency;
}
else
Depth = MODULATION_DEPTH_COEFF / 4.0f * modTime * modDepth * frequency;
}
/* Update the late reverb line lengths and T60 coefficients. */
void LateReverb::updateLines(const float density_mult, const float diffusion,
const float lfDecayTime, const float mfDecayTime, const float hfDecayTime,
const float lf0norm, const float hf0norm, const float frequency)
{
/* Scaling factor to convert the normalized reference frequencies from
* representing 0...freq to 0...max_reference.
*/
constexpr float MaxHFReference{20000.0f};
const float norm_weight_factor{frequency / MaxHFReference};
const float late_allpass_avg{
std::accumulate(LATE_ALLPASS_LENGTHS.begin(), LATE_ALLPASS_LENGTHS.end(), 0.0f) /
float{NUM_LINES}};
/* 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.
*/
float length{std::accumulate(LATE_LINE_LENGTHS.begin(), LATE_LINE_LENGTHS.end(), 0.0f) /
float{NUM_LINES} + late_allpass_avg};
length *= density_mult;
/* 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 float decayTimeWeighted{
lf0norm*norm_weight_factor*lfDecayTime +
(hf0norm - lf0norm)*norm_weight_factor*mfDecayTime +
(1.0f - hf0norm*norm_weight_factor)*hfDecayTime};
DensityGain = CalcDensityGain(CalcDecayCoeff(length, decayTimeWeighted));
/* Calculate the all-pass feed-back/forward coefficient. */
VecAp.Coeff = diffusion*diffusion * InvSqrt2;
for(size_t i{0u};i < NUM_LINES;i++)
{
/* Calculate the delay length of each all-pass line. */
length = LATE_ALLPASS_LENGTHS[i] * density_mult;
VecAp.Offset[i] = float2uint(length * frequency);
/* Calculate the delay length of each feedback delay line. A cubic
* resampler is used for modulation on the feedback delay, which
* includes one sample of delay. Reduce by one to compensate.
*/
length = LATE_LINE_LENGTHS[i] * density_mult;
Offset[i] = maxu(float2uint(length*frequency + 0.5f), 1u) - 1u;
/* 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. Also include the average
* modulation delay (depth is half the max delay in samples).
*/
length += lerpf(LATE_ALLPASS_LENGTHS[i], late_allpass_avg, diffusion)*density_mult +
Mod.Depth/frequency;
/* 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 ReverbPipeline::updateDelayLine(const float earlyDelay, const float lateDelay,
const float density_mult, const float decayTime, const float frequency)
{
/* 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(size_t i{0u};i < NUM_LINES;i++)
{
float length{EARLY_TAP_LENGTHS[i]*density_mult};
mEarlyDelayTap[i][1] = float2uint((earlyDelay+length) * frequency);
mEarlyDelayCoeff[i] = CalcDecayCoeff(length, decayTime);
length = (LATE_LINE_LENGTHS[i] - LATE_LINE_LENGTHS.front())/float{NUM_LINES}*density_mult +
lateDelay;
mLateDelayTap[i][1] = float2uint(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.
*/
std::array<std::array<float,4>,4> GetTransformFromVector(const float *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.
*/
float norm[3];
float 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::numbers::sqrt3_v<float>;
norm[1] = vec[1] / mag * al::numbers::sqrt3_v<float>;
norm[2] = vec[2] / mag * al::numbers::sqrt3_v<float>;
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::numbers::sqrt3_v<float>;
norm[1] = vec[1] * al::numbers::sqrt3_v<float>;
norm[2] = vec[2] * al::numbers::sqrt3_v<float>;
}
return std::array<std::array<float,4>,4>{{
{{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 ReverbPipeline::update3DPanning(const float *ReflectionsPan, const float *LateReverbPan,
const float earlyGain, const float lateGain, const bool doUpmix, const MixParams *mainMix)
{
/* Create matrices that transform a B-Format signal according to the
* panning vectors.
*/
const std::array<std::array<float,4>,4> earlymat{GetTransformFromVector(ReflectionsPan)};
const std::array<std::array<float,4>,4> latemat{GetTransformFromVector(LateReverbPan)};
if(doUpmix)
{
/* When upsampling, combine the early and late transforms with the
* first-order upsample matrix. This results in panning gains that
* apply the panning transform to first-order B-Format, which is then
* upsampled.
*/
auto mult_matrix = [](const al::span<const std::array<float,4>,4> mtx1)
{
auto&& mtx2 = AmbiScale::FirstOrderUp;
std::array<std::array<float,MaxAmbiChannels>,NUM_LINES> res{};
for(size_t i{0};i < mtx1[0].size();++i)
{
float *RESTRICT dst{res[i].data()};
for(size_t k{0};k < mtx1.size();++k)
{
const float *RESTRICT src{mtx2[k].data()};
const float a{mtx1[k][i]};
for(size_t j{0};j < mtx2[0].size();++j)
dst[j] += a * src[j];
}
}
return res;
};
auto earlycoeffs = mult_matrix(earlymat);
auto latecoeffs = mult_matrix(latemat);
for(size_t i{0u};i < NUM_LINES;i++)
ComputePanGains(mainMix, earlycoeffs[i].data(), earlyGain, mEarly.TargetGains[i]);
for(size_t i{0u};i < NUM_LINES;i++)
ComputePanGains(mainMix, latecoeffs[i].data(), lateGain, mLate.TargetGains[i]);
}
else
{
/* When not upsampling, combine the early and late A-to-B-Format
* conversions with their respective transform. This results panning
* gains that convert A-Format to B-Format, which is then panned.
*/
auto mult_matrix = [](const al::span<const std::array<float,NUM_LINES>,4> mtx1,
const al::span<const std::array<float,4>,4> mtx2)
{
std::array<std::array<float,MaxAmbiChannels>,NUM_LINES> res{};
for(size_t i{0};i < mtx1[0].size();++i)
{
float *RESTRICT dst{res[i].data()};
for(size_t k{0};k < mtx1.size();++k)
{
const float a{mtx1[k][i]};
for(size_t j{0};j < mtx2.size();++j)
dst[j] += a * mtx2[j][k];
}
}
return res;
};
auto earlycoeffs = mult_matrix(EarlyA2B, earlymat);
auto latecoeffs = mult_matrix(LateA2B, latemat);
for(size_t i{0u};i < NUM_LINES;i++)
ComputePanGains(mainMix, earlycoeffs[i].data(), earlyGain, mEarly.TargetGains[i]);
for(size_t i{0u};i < NUM_LINES;i++)
ComputePanGains(mainMix, latecoeffs[i].data(), lateGain, mLate.TargetGains[i]);
}
}
void ReverbState::update(const ContextBase *Context, const EffectSlot *Slot,
const EffectProps *props, const EffectTarget target)
{
const DeviceBase *Device{Context->mDevice};
const auto frequency = static_cast<float>(Device->Frequency);
/* If the HF limit parameter is flagged, calculate an appropriate limit
* based on the air absorption parameter.
*/
float hfRatio{props->Reverb.DecayHFRatio};
if(props->Reverb.DecayHFLimit && props->Reverb.AirAbsorptionGainHF < 1.0f)
hfRatio = CalcLimitedHfRatio(hfRatio, props->Reverb.AirAbsorptionGainHF,
props->Reverb.DecayTime);
/* Calculate the LF/HF decay times. */
constexpr float MinDecayTime{0.1f}, MaxDecayTime{20.0f};
const float lfDecayTime{clampf(props->Reverb.DecayTime*props->Reverb.DecayLFRatio,
MinDecayTime, MaxDecayTime)};
const float hfDecayTime{clampf(props->Reverb.DecayTime*hfRatio, MinDecayTime, MaxDecayTime)};
/* Determine if a full update is required. */
const bool fullUpdate{mPipelineState == DeviceClear ||
/* Density is essentially a master control for the feedback delays, so
* changes the offsets of many delay lines.
*/
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 ||
/* Modulation time and depth both require fading the modulation delay. */
mParams.ModulationTime != props->Reverb.ModulationTime ||
mParams.ModulationDepth != props->Reverb.ModulationDepth ||
/* HF/LF References control the weighting used to calculate the density
* gain.
*/
mParams.HFReference != props->Reverb.HFReference ||
mParams.LFReference != props->Reverb.LFReference};
if(fullUpdate)
{
mParams.Density = props->Reverb.Density;
mParams.Diffusion = props->Reverb.Diffusion;
mParams.DecayTime = props->Reverb.DecayTime;
mParams.HFDecayTime = hfDecayTime;
mParams.LFDecayTime = lfDecayTime;
mParams.ModulationTime = props->Reverb.ModulationTime;
mParams.ModulationDepth = props->Reverb.ModulationDepth;
mParams.HFReference = props->Reverb.HFReference;
mParams.LFReference = props->Reverb.LFReference;
mPipelineState = (mPipelineState != DeviceClear) ? StartFade : Normal;
mCurrentPipeline ^= 1;
}
auto &pipeline = mPipelines[mCurrentPipeline];
/* Update early and late 3D panning. */
mOutTarget = target.Main->Buffer;
const float gain{props->Reverb.Gain * Slot->Gain * ReverbBoost};
pipeline.update3DPanning(props->Reverb.ReflectionsPan, props->Reverb.LateReverbPan,
props->Reverb.ReflectionsGain*gain, props->Reverb.LateReverbGain*gain, mUpmixOutput,
target.Main);
/* Calculate the master filters */
float hf0norm{minf(props->Reverb.HFReference/frequency, 0.49f)};
pipeline.mFilter[0].Lp.setParamsFromSlope(BiquadType::HighShelf, hf0norm, props->Reverb.GainHF, 1.0f);
float lf0norm{minf(props->Reverb.LFReference/frequency, 0.49f)};
pipeline.mFilter[0].Hp.setParamsFromSlope(BiquadType::LowShelf, lf0norm, props->Reverb.GainLF, 1.0f);
for(size_t i{1u};i < NUM_LINES;i++)
{
pipeline.mFilter[i].Lp.copyParamsFrom(pipeline.mFilter[0].Lp);
pipeline.mFilter[i].Hp.copyParamsFrom(pipeline.mFilter[0].Hp);
}
/* The density-based room size (delay length) multiplier. */
const float density_mult{CalcDelayLengthMult(props->Reverb.Density)};
/* Update the main effect delay and associated taps. */
pipeline.updateDelayLine(props->Reverb.ReflectionsDelay, props->Reverb.LateReverbDelay,
density_mult, props->Reverb.DecayTime, frequency);
if(fullUpdate)
{
/* Update the early lines. */
pipeline.mEarly.updateLines(density_mult, props->Reverb.Diffusion, props->Reverb.DecayTime,
frequency);
/* Get the mixing matrix coefficients. */
CalcMatrixCoeffs(props->Reverb.Diffusion, &pipeline.mMixX, &pipeline.mMixY);
/* Update the modulator rate and depth. */
pipeline.mLate.Mod.updateModulator(props->Reverb.ModulationTime,
props->Reverb.ModulationDepth, frequency);
/* Update the late lines. */
pipeline.mLate.updateLines(density_mult, props->Reverb.Diffusion, lfDecayTime,
props->Reverb.DecayTime, hfDecayTime, lf0norm, hf0norm, frequency);
}
const float decaySamples{(props->Reverb.ReflectionsDelay + props->Reverb.LateReverbDelay
+ props->Reverb.DecayTime) * frequency};
pipeline.mFadeSampleCount = static_cast<size_t>(minf(decaySamples, 1'000'000.0f));
}
/**************************************
* 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 auto VectorPartialScatter(const std::array<float,NUM_LINES> &RESTRICT in,
const float xCoeff, const float yCoeff) -> std::array<float,NUM_LINES>
{
return std::array<float,NUM_LINES>{{
xCoeff*in[0] + yCoeff*( in[1] + -in[2] + in[3]),
xCoeff*in[1] + yCoeff*(-in[0] + in[2] + in[3]),
xCoeff*in[2] + yCoeff*( in[0] + -in[1] + in[3]),
xCoeff*in[3] + yCoeff*(-in[0] + -in[1] + -in[2] )
}};
}
/* Utilizes the above, but reverses the input channels. */
void VectorScatterRevDelayIn(const DelayLineI delay, size_t offset, const float xCoeff,
const float yCoeff, const al::span<const ReverbUpdateLine,NUM_LINES> in, const size_t count)
{
ASSUME(count > 0);
for(size_t i{0u};i < count;)
{
offset &= delay.Mask;
size_t td{minz(delay.Mask+1 - offset, count-i)};
do {
std::array<float,NUM_LINES> f;
for(size_t j{0u};j < NUM_LINES;j++)
f[NUM_LINES-1-j] = in[j][i];
++i;
delay.Line[offset++] = VectorPartialScatter(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::process(const al::span<ReverbUpdateLine,NUM_LINES> samples, size_t offset,
const float xCoeff, const float yCoeff, const size_t todo)
{
const DelayLineI delay{Delay};
const float feedCoeff{Coeff};
ASSUME(todo > 0);
size_t vap_offset[NUM_LINES];
for(size_t j{0u};j < NUM_LINES;j++)
vap_offset[j] = offset - Offset[j];
for(size_t i{0u};i < todo;)
{
for(size_t j{0u};j < NUM_LINES;j++)
vap_offset[j] &= delay.Mask;
offset &= delay.Mask;
size_t maxoff{offset};
for(size_t j{0u};j < NUM_LINES;j++)
maxoff = maxz(maxoff, vap_offset[j]);
size_t td{minz(delay.Mask+1 - maxoff, todo - i)};
do {
std::array<float,NUM_LINES> f;
for(size_t j{0u};j < NUM_LINES;j++)
{
const float input{samples[j][i]};
const float out{delay.Line[vap_offset[j]++][j] - feedCoeff*input};
f[j] = input + feedCoeff*out;
samples[j][i] = out;
}
++i;
delay.Line[offset++] = VectorPartialScatter(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.
*/
void ReverbPipeline::processEarly(size_t offset, const size_t samplesToDo,
const al::span<ReverbUpdateLine, NUM_LINES> tempSamples,
const al::span<FloatBufferLine, NUM_LINES> outSamples)
{
const DelayLineI early_delay{mEarly.Delay};
const DelayLineI in_delay{mEarlyDelayIn};
const float mixX{mMixX};
const float mixY{mMixY};
ASSUME(samplesToDo > 0);
for(size_t base{0};base < samplesToDo;)
{
const size_t todo{minz(samplesToDo-base, MAX_UPDATE_SAMPLES)};
/* First, load decorrelated samples from the main delay line as the
* primary reflections.
*/
const float fadeStep{1.0f / static_cast<float>(todo)};
for(size_t j{0u};j < NUM_LINES;j++)
{
size_t early_delay_tap0{offset - mEarlyDelayTap[j][0]};
size_t early_delay_tap1{offset - mEarlyDelayTap[j][1]};
const float coeff{mEarlyDelayCoeff[j]};
const float coeffStep{early_delay_tap0 != early_delay_tap1 ? coeff*fadeStep : 0.0f};
float fadeCount{0.0f};
for(size_t i{0u};i < todo;)
{
early_delay_tap0 &= in_delay.Mask;
early_delay_tap1 &= in_delay.Mask;
const size_t max_tap{maxz(early_delay_tap0, early_delay_tap1)};
size_t td{minz(in_delay.Mask+1 - max_tap, todo-i)};
do {
const float fade0{coeff - coeffStep*fadeCount};
const float fade1{coeffStep*fadeCount};
fadeCount += 1.0f;
tempSamples[j][i++] = in_delay.Line[early_delay_tap0++][j]*fade0 +
in_delay.Line[early_delay_tap1++][j]*fade1;
} while(--td);
}
mEarlyDelayTap[j][0] = mEarlyDelayTap[j][1];
}
/* Apply a vector all-pass, to help color the initial reflections based
* on the diffusion strength.
*/
mEarly.VecAp.process(tempSamples, 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(size_t j{0u};j < NUM_LINES;j++)
early_delay.write(offset, NUM_LINES-1-j, tempSamples[j].data(), todo);
for(size_t j{0u};j < NUM_LINES;j++)
{
size_t feedb_tap{offset - mEarly.Offset[j]};
const float feedb_coeff{mEarly.Coeff[j]};
float *RESTRICT out{al::assume_aligned<16>(outSamples[j].data() + base)};
for(size_t i{0u};i < todo;)
{
feedb_tap &= early_delay.Mask;
size_t td{minz(early_delay.Mask+1 - feedb_tap, todo - i)};
do {
tempSamples[j][i] += early_delay.Line[feedb_tap++][j]*feedb_coeff;
out[i] = tempSamples[j][i];
++i;
} while(--td);
}
}
/* Finally, write the result to the late delay line input for the late
* reverb stage to pick up at the appropriate time, applying a scatter
* and bounce to improve the initial diffusion in the late reverb.
*/
VectorScatterRevDelayIn(mLateDelayIn, offset, mixX, mixY, tempSamples, todo);
base += todo;
offset += todo;
}
}
void Modulation::calcDelays(size_t todo)
{
constexpr float mod_scale{al::numbers::pi_v<float> * 2.0f / MOD_FRACONE};
uint idx{Index};
const uint step{Step};
const float depth{Depth};
for(size_t i{0};i < todo;++i)
{
idx += step;
const float lfo{std::sin(static_cast<float>(idx&MOD_FRACMASK) * mod_scale)};
ModDelays[i] = (lfo+1.0f) * depth;
}
Index = idx;
}
/* 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
* modulated 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.
*/
void ReverbPipeline::processLate(size_t offset, const size_t samplesToDo,
const al::span<ReverbUpdateLine, NUM_LINES> tempSamples,
const al::span<FloatBufferLine, NUM_LINES> outSamples)
{
const DelayLineI late_delay{mLate.Delay};
const DelayLineI in_delay{mLateDelayIn};
const float mixX{mMixX};
const float mixY{mMixY};
ASSUME(samplesToDo > 0);
for(size_t base{0};base < samplesToDo;)
{
const size_t todo{minz(samplesToDo-base, minz(mLate.Offset[0], MAX_UPDATE_SAMPLES))};
ASSUME(todo > 0);
/* First, calculate the modulated delays for the late feedback. */
mLate.Mod.calcDelays(todo);
/* Next, load decorrelated samples from the main and feedback delay
* lines. Filter the signal to apply its frequency-dependent decay.
*/
const float fadeStep{1.0f / static_cast<float>(todo)};
for(size_t j{0u};j < NUM_LINES;j++)
{
size_t late_delay_tap0{offset - mLateDelayTap[j][0]};
size_t late_delay_tap1{offset - mLateDelayTap[j][1]};
size_t late_feedb_tap{offset - mLate.Offset[j]};
const float midGain{mLate.T60[j].MidGain};
const float densityGain{mLate.DensityGain * midGain};
const float densityStep{late_delay_tap0 != late_delay_tap1 ?
densityGain*fadeStep : 0.0f};
float fadeCount{0.0f};
for(size_t i{0u};i < todo;)
{
late_delay_tap0 &= in_delay.Mask;
late_delay_tap1 &= in_delay.Mask;
size_t td{minz(todo-i, in_delay.Mask+1 - maxz(late_delay_tap0, late_delay_tap1))};
do {
/* Calculate the read offset and offset between it and the
* next sample.
*/
const float fdelay{mLate.Mod.ModDelays[i]};
const size_t idelay{float2uint(fdelay * float{gCubicTable.sTableSteps})};
const size_t delay{late_feedb_tap - (idelay>>gCubicTable.sTableBits)};
const size_t delayoffset{idelay & gCubicTable.sTableMask};
++late_feedb_tap;
/* Get the samples around by the delayed offset. */
const float out0{late_delay.Line[(delay ) & late_delay.Mask][j]};
const float out1{late_delay.Line[(delay-1) & late_delay.Mask][j]};
const float out2{late_delay.Line[(delay-2) & late_delay.Mask][j]};
const float out3{late_delay.Line[(delay-3) & late_delay.Mask][j]};
/* The output is obtained by interpolating the four samples
* that were acquired above, and combined with the main
* delay tap.
*/
const float out{out0*gCubicTable.getCoeff0(delayoffset)
+ out1*gCubicTable.getCoeff1(delayoffset)
+ out2*gCubicTable.getCoeff2(delayoffset)
+ out3*gCubicTable.getCoeff3(delayoffset)};
const float fade0{densityGain - densityStep*fadeCount};
const float fade1{densityStep*fadeCount};
fadeCount += 1.0f;
tempSamples[j][i] = out*midGain +
in_delay.Line[late_delay_tap0++][j]*fade0 +
in_delay.Line[late_delay_tap1++][j]*fade1;
++i;
} while(--td);
}
mLateDelayTap[j][0] = mLateDelayTap[j][1];
mLate.T60[j].process({tempSamples[j].data(), todo});
}
/* Apply a vector all-pass to improve micro-surface diffusion, and
* write out the results for mixing.
*/
mLate.VecAp.process(tempSamples, offset, mixX, mixY, todo);
for(size_t j{0u};j < NUM_LINES;j++)
std::copy_n(tempSamples[j].begin(), todo, outSamples[j].begin()+base);
/* Finally, scatter and bounce the results to refeed the feedback buffer. */
VectorScatterRevDelayIn(late_delay, offset, mixX, mixY, tempSamples, todo);
base += todo;
offset += todo;
}
}
void ReverbState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
const size_t offset{mOffset};
ASSUME(samplesToDo > 0);
auto &oldpipeline = mPipelines[mCurrentPipeline^1];
auto &pipeline = mPipelines[mCurrentPipeline];
if(mPipelineState >= Fading)
{
/* Convert B-Format to A-Format for processing. */
const size_t numInput{minz(samplesIn.size(), NUM_LINES)};
const al::span<float> tmpspan{al::assume_aligned<16>(mTempLine.data()), samplesToDo};
for(size_t c{0u};c < NUM_LINES;c++)
{
std::fill(tmpspan.begin(), tmpspan.end(), 0.0f);
for(size_t i{0};i < numInput;++i)
{
const float gain{B2A[c][i]};
const float *RESTRICT input{al::assume_aligned<16>(samplesIn[i].data())};
auto mix_sample = [gain](const float sample, const float in) noexcept -> float
{ return sample + in*gain; };
std::transform(tmpspan.begin(), tmpspan.end(), input, tmpspan.begin(),
mix_sample);
}
/* Band-pass the incoming samples and feed the initial delay line. */
auto&& filter = DualBiquad{pipeline.mFilter[c].Lp, pipeline.mFilter[c].Hp};
filter.process(tmpspan, tmpspan.data());
pipeline.mEarlyDelayIn.write(offset, c, tmpspan.cbegin(), samplesToDo);
}
if(mPipelineState == Fading)
{
/* Give the old pipeline silence if it's still fading out. */
for(size_t c{0u};c < NUM_LINES;c++)
{
std::fill(tmpspan.begin(), tmpspan.end(), 0.0f);
auto&& filter = DualBiquad{oldpipeline.mFilter[c].Lp, oldpipeline.mFilter[c].Hp};
filter.process(tmpspan, tmpspan.data());
oldpipeline.mEarlyDelayIn.write(offset, c, tmpspan.cbegin(), samplesToDo);
}
}
}
else
{
/* At the start of a fade, fade in input for the current pipeline, and
* fade out input for the old pipeline.
*/
const size_t numInput{minz(samplesIn.size(), NUM_LINES)};
const al::span<float> tmpspan{al::assume_aligned<16>(mTempLine.data()), samplesToDo};
const float fadeStep{1.0f / static_cast<float>(samplesToDo)};
for(size_t c{0u};c < NUM_LINES;c++)
{
std::fill(tmpspan.begin(), tmpspan.end(), 0.0f);
for(size_t i{0};i < numInput;++i)
{
const float gain{B2A[c][i]};
const float *RESTRICT input{al::assume_aligned<16>(samplesIn[i].data())};
auto mix_sample = [gain](const float sample, const float in) noexcept -> float
{ return sample + in*gain; };
std::transform(tmpspan.begin(), tmpspan.end(), input, tmpspan.begin(),
mix_sample);
}
float stepCount{0.0f};
for(float &sample : tmpspan)
{
stepCount += 1.0f;
sample *= stepCount*fadeStep;
}
auto&& filter = DualBiquad{pipeline.mFilter[c].Lp, pipeline.mFilter[c].Hp};
filter.process(tmpspan, tmpspan.data());
pipeline.mEarlyDelayIn.write(offset, c, tmpspan.cbegin(), samplesToDo);
}
for(size_t c{0u};c < NUM_LINES;c++)
{
std::fill(tmpspan.begin(), tmpspan.end(), 0.0f);
for(size_t i{0};i < numInput;++i)
{
const float gain{B2A[c][i]};
const float *RESTRICT input{al::assume_aligned<16>(samplesIn[i].data())};
auto mix_sample = [gain](const float sample, const float in) noexcept -> float
{ return sample + in*gain; };
std::transform(tmpspan.begin(), tmpspan.end(), input, tmpspan.begin(),
mix_sample);
}
float stepCount{0.0f};
for(float &sample : tmpspan)
{
stepCount += 1.0f;
sample *= 1.0f - stepCount*fadeStep;
}
auto&& filter = DualBiquad{oldpipeline.mFilter[c].Lp, oldpipeline.mFilter[c].Hp};
filter.process(tmpspan, tmpspan.data());
oldpipeline.mEarlyDelayIn.write(offset, c, tmpspan.cbegin(), samplesToDo);
}
mPipelineState = Fading;
}
/* Process reverb for these samples. and mix them to the output. */
pipeline.processEarly(offset, samplesToDo, mTempSamples, mEarlySamples);
pipeline.processLate(offset, samplesToDo, mTempSamples, mLateSamples);
mixOut(pipeline, samplesOut, samplesToDo);
if(mPipelineState != Normal)
{
if(mPipelineState == Cleanup)
{
size_t numSamples{mSampleBuffer.size()/2};
size_t pipelineOffset{numSamples * (mCurrentPipeline^1)};
std::fill_n(mSampleBuffer.data()+pipelineOffset, numSamples,
decltype(mSampleBuffer)::value_type{});
oldpipeline.clear();
mPipelineState = Normal;
}
else
{
/* If this is the final mix for this old pipeline, set the target
* gains to 0 to ensure a complete fade out, and set the state to
* Cleanup so the next invocation cleans up the delay buffers and
* filters.
*/
if(samplesToDo >= oldpipeline.mFadeSampleCount)
{
for(auto &gains : oldpipeline.mEarly.TargetGains)
std::fill(std::begin(gains), std::end(gains), 0.0f);
for(auto &gains : oldpipeline.mLate.TargetGains)
std::fill(std::begin(gains), std::end(gains), 0.0f);
oldpipeline.mFadeSampleCount = 0;
mPipelineState = Cleanup;
}
else
oldpipeline.mFadeSampleCount -= samplesToDo;
/* Process the old reverb for these samples. */
oldpipeline.processEarly(offset, samplesToDo, mTempSamples, mEarlySamples);
oldpipeline.processLate(offset, samplesToDo, mTempSamples, mLateSamples);
mixOut(oldpipeline, samplesOut, samplesToDo);
}
}
mOffset = offset + samplesToDo;
}
struct ReverbStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new ReverbState{}}; }
};
struct StdReverbStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new ReverbState{}}; }
};
} // namespace
EffectStateFactory *ReverbStateFactory_getFactory()
{
static ReverbStateFactory ReverbFactory{};
return &ReverbFactory;
}
EffectStateFactory *StdReverbStateFactory_getFactory()
{
static StdReverbStateFactory ReverbFactory{};
return &ReverbFactory;
}
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