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

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/**
* OpenAL cross platform audio library
* Copyright (C) 2013 by Mike Gorchak
* 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>
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#include <array>
#include <climits>
#include <cstdlib>
#include <iterator>
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#include "alc/effects/base.h"
#include "almalloc.h"
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#include "alnumbers.h"
#include "alnumeric.h"
#include "alspan.h"
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#include "core/bufferline.h"
#include "core/context.h"
#include "core/devformat.h"
#include "core/device.h"
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#include "core/effectslot.h"
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#include "core/mixer.h"
#include "core/mixer/defs.h"
#include "core/resampler_limits.h"
#include "intrusive_ptr.h"
#include "opthelpers.h"
#include "vector.h"
namespace {
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using uint = unsigned int;
#define MAX_UPDATE_SAMPLES 256
struct ChorusState final : public EffectState {
al::vector<float,16> mSampleBuffer;
uint mOffset{0};
uint mLfoOffset{0};
uint mLfoRange{1};
float mLfoScale{0.0f};
uint mLfoDisp{0};
/* Gains for left and right sides */
struct {
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float Current[MaxAmbiChannels]{};
float Target[MaxAmbiChannels]{};
} mGains[2];
/* effect parameters */
ChorusWaveform mWaveform{};
int mDelay{0};
float mDepth{0.0f};
float mFeedback{0.0f};
void getTriangleDelays(uint (*delays)[MAX_UPDATE_SAMPLES], const size_t todo);
void getSinusoidDelays(uint (*delays)[MAX_UPDATE_SAMPLES], const size_t todo);
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void deviceUpdate(const DeviceBase *device, const Buffer &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(ChorusState)
};
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void ChorusState::deviceUpdate(const DeviceBase *Device, const Buffer&)
{
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constexpr float max_delay{maxf(ChorusMaxDelay, FlangerMaxDelay)};
const auto frequency = static_cast<float>(Device->Frequency);
const size_t maxlen{NextPowerOf2(float2uint(max_delay*2.0f*frequency) + 1u)};
if(maxlen != mSampleBuffer.size())
al::vector<float,16>(maxlen).swap(mSampleBuffer);
std::fill(mSampleBuffer.begin(), mSampleBuffer.end(), 0.0f);
for(auto &e : mGains)
{
std::fill(std::begin(e.Current), std::end(e.Current), 0.0f);
std::fill(std::begin(e.Target), std::end(e.Target), 0.0f);
}
}
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void ChorusState::update(const ContextBase *Context, const EffectSlot *Slot,
const EffectProps *props, const EffectTarget target)
{
constexpr int mindelay{(MaxResamplerPadding>>1) << MixerFracBits};
/* The LFO depth is scaled to be relative to the sample delay. Clamp the
* delay and depth to allow enough padding for resampling.
*/
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const DeviceBase *device{Context->mDevice};
const auto frequency = static_cast<float>(device->Frequency);
mWaveform = props->Chorus.Waveform;
mDelay = maxi(float2int(props->Chorus.Delay*frequency*MixerFracOne + 0.5f), mindelay);
mDepth = minf(props->Chorus.Depth * static_cast<float>(mDelay),
static_cast<float>(mDelay - mindelay));
mFeedback = props->Chorus.Feedback;
/* Gains for left and right sides */
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static constexpr auto lcoeffs = CalcDirectionCoeffs({-1.0f, 0.0f, 0.0f});
static constexpr auto rcoeffs = CalcDirectionCoeffs({ 1.0f, 0.0f, 0.0f});
mOutTarget = target.Main->Buffer;
ComputePanGains(target.Main, lcoeffs.data(), Slot->Gain, mGains[0].Target);
ComputePanGains(target.Main, rcoeffs.data(), Slot->Gain, mGains[1].Target);
float rate{props->Chorus.Rate};
if(!(rate > 0.0f))
{
mLfoOffset = 0;
mLfoRange = 1;
mLfoScale = 0.0f;
mLfoDisp = 0;
}
else
{
/* Calculate LFO coefficient (number of samples per cycle). Limit the
* max range to avoid overflow when calculating the displacement.
*/
uint lfo_range{float2uint(minf(frequency/rate + 0.5f, float{INT_MAX/360 - 180}))};
mLfoOffset = mLfoOffset * lfo_range / mLfoRange;
mLfoRange = lfo_range;
switch(mWaveform)
{
case ChorusWaveform::Triangle:
mLfoScale = 4.0f / static_cast<float>(mLfoRange);
break;
case ChorusWaveform::Sinusoid:
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mLfoScale = al::numbers::pi_v<float>*2.0f / static_cast<float>(mLfoRange);
break;
}
/* Calculate lfo phase displacement */
int phase{props->Chorus.Phase};
if(phase < 0) phase = 360 + phase;
mLfoDisp = (mLfoRange*static_cast<uint>(phase) + 180) / 360;
}
}
void ChorusState::getTriangleDelays(uint (*delays)[MAX_UPDATE_SAMPLES], const size_t todo)
{
const uint lfo_range{mLfoRange};
const float lfo_scale{mLfoScale};
const float depth{mDepth};
const int delay{mDelay};
ASSUME(lfo_range > 0);
ASSUME(todo > 0);
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auto gen_lfo = [lfo_scale,depth,delay](const uint offset) -> uint
{
const float offset_norm{static_cast<float>(offset) * lfo_scale};
return static_cast<uint>(fastf2i((1.0f-std::abs(2.0f-offset_norm)) * depth) + delay);
};
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uint offset{mLfoOffset};
for(size_t i{0};i < todo;)
{
size_t rem{minz(todo-i, lfo_range-offset)};
do {
delays[0][i++] = gen_lfo(offset++);
} while(--rem);
if(offset == lfo_range)
offset = 0;
}
offset = (mLfoOffset+mLfoDisp) % lfo_range;
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for(size_t i{0};i < todo;)
{
size_t rem{minz(todo-i, lfo_range-offset)};
do {
delays[1][i++] = gen_lfo(offset++);
} while(--rem);
if(offset == lfo_range)
offset = 0;
}
mLfoOffset = static_cast<uint>(mLfoOffset+todo) % lfo_range;
}
void ChorusState::getSinusoidDelays(uint (*delays)[MAX_UPDATE_SAMPLES], const size_t todo)
{
const uint lfo_range{mLfoRange};
const float lfo_scale{mLfoScale};
const float depth{mDepth};
const int delay{mDelay};
ASSUME(lfo_range > 0);
ASSUME(todo > 0);
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auto gen_lfo = [lfo_scale,depth,delay](const uint offset) -> uint
{
const float offset_norm{static_cast<float>(offset) * lfo_scale};
return static_cast<uint>(fastf2i(std::sin(offset_norm)*depth) + delay);
};
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uint offset{mLfoOffset};
for(size_t i{0};i < todo;)
{
size_t rem{minz(todo-i, lfo_range-offset)};
do {
delays[0][i++] = gen_lfo(offset++);
} while(--rem);
if(offset == lfo_range)
offset = 0;
}
offset = (mLfoOffset+mLfoDisp) % lfo_range;
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for(size_t i{0};i < todo;)
{
size_t rem{minz(todo-i, lfo_range-offset)};
do {
delays[1][i++] = gen_lfo(offset++);
} while(--rem);
if(offset == lfo_range)
offset = 0;
}
mLfoOffset = static_cast<uint>(mLfoOffset+todo) % lfo_range;
}
void ChorusState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
const size_t bufmask{mSampleBuffer.size()-1};
const float feedback{mFeedback};
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const uint avgdelay{(static_cast<uint>(mDelay) + MixerFracHalf) >> MixerFracBits};
float *RESTRICT delaybuf{mSampleBuffer.data()};
uint offset{mOffset};
for(size_t base{0u};base < samplesToDo;)
{
const size_t todo{minz(MAX_UPDATE_SAMPLES, samplesToDo-base)};
uint moddelays[2][MAX_UPDATE_SAMPLES];
if(mWaveform == ChorusWaveform::Sinusoid)
getSinusoidDelays(moddelays, todo);
else /*if(mWaveform == ChorusWaveform::Triangle)*/
getTriangleDelays(moddelays, todo);
alignas(16) float temps[2][MAX_UPDATE_SAMPLES];
for(size_t i{0u};i < todo;++i)
{
// Feed the buffer's input first (necessary for delays < 1).
delaybuf[offset&bufmask] = samplesIn[0][base+i];
// Tap for the left output.
uint delay{offset - (moddelays[0][i]>>MixerFracBits)};
float mu{static_cast<float>(moddelays[0][i]&MixerFracMask) * (1.0f/MixerFracOne)};
temps[0][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay ) & bufmask],
delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask], mu);
// Tap for the right output.
delay = offset - (moddelays[1][i]>>MixerFracBits);
mu = static_cast<float>(moddelays[1][i]&MixerFracMask) * (1.0f/MixerFracOne);
temps[1][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay ) & bufmask],
delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask], mu);
// Accumulate feedback from the average delay of the taps.
delaybuf[offset&bufmask] += delaybuf[(offset-avgdelay) & bufmask] * feedback;
++offset;
}
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for(size_t c{0};c < 2;++c)
MixSamples({temps[c], todo}, samplesOut, mGains[c].Current, mGains[c].Target,
samplesToDo-base, base);
base += todo;
}
mOffset = offset;
}
struct ChorusStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new ChorusState{}}; }
};
/* Flanger is basically a chorus with a really short delay. They can both use
* the same processing functions, so piggyback flanger on the chorus functions.
*/
struct FlangerStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new ChorusState{}}; }
};
} // namespace
EffectStateFactory *ChorusStateFactory_getFactory()
{
static ChorusStateFactory ChorusFactory{};
return &ChorusFactory;
}
EffectStateFactory *FlangerStateFactory_getFactory()
{
static FlangerStateFactory FlangerFactory{};
return &FlangerFactory;
}