update openal-soft

sync point: master-ac5d40e40a0155351fe1be4aab30017b6a13a859

Engine/lib/openal-soft/Alc/effects/autowah.cpp

@@ -0,0 +1,212 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2018 by Raul Herraiz.
+ * 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 <cmath>
+#include <cstdlib>
+
+#include <algorithm>
+
+#include "alcmain.h"
+#include "alcontext.h"
+#include "core/filters/biquad.h"
+#include "effectslot.h"
+#include "vecmat.h"
+
+namespace {
+
+constexpr float GainScale{31621.0f};
+constexpr float MinFreq{20.0f};
+constexpr float MaxFreq{2500.0f};
+constexpr float QFactor{5.0f};
+
+struct AutowahState final : public EffectState {
+    /* Effect parameters */
+    float mAttackRate;
+    float mReleaseRate;
+    float mResonanceGain;
+    float mPeakGain;
+    float mFreqMinNorm;
+    float mBandwidthNorm;
+    float mEnvDelay;
+
+    /* Filter components derived from the envelope. */
+    struct {
+        float cos_w0;
+        float alpha;
+    } mEnv[BufferLineSize];
+
+    struct {
+        /* Effect filters' history. */
+        struct {
+            float z1, z2;
+        } Filter;
+
+        /* Effect gains for each output channel */
+        float CurrentGains[MAX_OUTPUT_CHANNELS];
+        float TargetGains[MAX_OUTPUT_CHANNELS];
+    } mChans[MaxAmbiChannels];
+
+    /* Effects buffers */
+    alignas(16) float mBufferOut[BufferLineSize];
+
+
+    void deviceUpdate(const ALCdevice *device, const Buffer &buffer) override;
+    void update(const ALCcontext *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(AutowahState)
+};
+
+void AutowahState::deviceUpdate(const ALCdevice*, const Buffer&)
+{
+    /* (Re-)initializing parameters and clear the buffers. */
+
+    mAttackRate    = 1.0f;
+    mReleaseRate   = 1.0f;
+    mResonanceGain = 10.0f;
+    mPeakGain      = 4.5f;
+    mFreqMinNorm   = 4.5e-4f;
+    mBandwidthNorm = 0.05f;
+    mEnvDelay      = 0.0f;
+
+    for(auto &e : mEnv)
+    {
+        e.cos_w0 = 0.0f;
+        e.alpha = 0.0f;
+    }
+
+    for(auto &chan : mChans)
+    {
+        std::fill(std::begin(chan.CurrentGains), std::end(chan.CurrentGains), 0.0f);
+        chan.Filter.z1 = 0.0f;
+        chan.Filter.z2 = 0.0f;
+    }
+}
+
+void AutowahState::update(const ALCcontext *context, const EffectSlot *slot,
+    const EffectProps *props, const EffectTarget target)
+{
+    const ALCdevice *device{context->mDevice.get()};
+    const auto frequency = static_cast<float>(device->Frequency);
+
+    const float ReleaseTime{clampf(props->Autowah.ReleaseTime, 0.001f, 1.0f)};
+
+    mAttackRate    = std::exp(-1.0f / (props->Autowah.AttackTime*frequency));
+    mReleaseRate   = std::exp(-1.0f / (ReleaseTime*frequency));
+    /* 0-20dB Resonance Peak gain */
+    mResonanceGain = std::sqrt(std::log10(props->Autowah.Resonance)*10.0f / 3.0f);
+    mPeakGain      = 1.0f - std::log10(props->Autowah.PeakGain / GainScale);
+    mFreqMinNorm   = MinFreq / frequency;
+    mBandwidthNorm = (MaxFreq-MinFreq) / frequency;
+
+    mOutTarget = target.Main->Buffer;
+    auto set_gains = [slot,target](auto &chan, al::span<const float,MaxAmbiChannels> coeffs)
+    { ComputePanGains(target.Main, coeffs.data(), slot->Gain, chan.TargetGains); };
+    SetAmbiPanIdentity(std::begin(mChans), slot->Wet.Buffer.size(), set_gains);
+}
+
+void AutowahState::process(const size_t samplesToDo,
+    const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
+{
+    const float attack_rate{mAttackRate};
+    const float release_rate{mReleaseRate};
+    const float res_gain{mResonanceGain};
+    const float peak_gain{mPeakGain};
+    const float freq_min{mFreqMinNorm};
+    const float bandwidth{mBandwidthNorm};
+
+    float env_delay{mEnvDelay};
+    for(size_t i{0u};i < samplesToDo;i++)
+    {
+        float w0, sample, a;
+
+        /* Envelope follower described on the book: Audio Effects, Theory,
+         * Implementation and Application.
+         */
+        sample = peak_gain * std::fabs(samplesIn[0][i]);
+        a = (sample > env_delay) ? attack_rate : release_rate;
+        env_delay = lerp(sample, env_delay, a);
+
+        /* Calculate the cos and alpha components for this sample's filter. */
+        w0 = minf((bandwidth*env_delay + freq_min), 0.46f) * al::MathDefs<float>::Tau();
+        mEnv[i].cos_w0 = std::cos(w0);
+        mEnv[i].alpha = std::sin(w0)/(2.0f * QFactor);
+    }
+    mEnvDelay = env_delay;
+
+    auto chandata = std::addressof(mChans[0]);
+    for(const auto &insamples : samplesIn)
+    {
+        /* This effectively inlines BiquadFilter_setParams for a peaking
+         * filter and BiquadFilter_processC. The alpha and cosine components
+         * for the filter coefficients were previously calculated with the
+         * envelope. Because the filter changes for each sample, the
+         * coefficients are transient and don't need to be held.
+         */
+        float z1{chandata->Filter.z1};
+        float z2{chandata->Filter.z2};
+
+        for(size_t i{0u};i < samplesToDo;i++)
+        {
+            const float alpha{mEnv[i].alpha};
+            const float cos_w0{mEnv[i].cos_w0};
+            float input, output;
+            float a[3], b[3];
+
+            b[0] =  1.0f + alpha*res_gain;
+            b[1] = -2.0f * cos_w0;
+            b[2] =  1.0f - alpha*res_gain;
+            a[0] =  1.0f + alpha/res_gain;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha/res_gain;
+
+            input = insamples[i];
+            output = input*(b[0]/a[0]) + z1;
+            z1 = input*(b[1]/a[0]) - output*(a[1]/a[0]) + z2;
+            z2 = input*(b[2]/a[0]) - output*(a[2]/a[0]);
+            mBufferOut[i] = output;
+        }
+        chandata->Filter.z1 = z1;
+        chandata->Filter.z2 = z2;
+
+        /* Now, mix the processed sound data to the output. */
+        MixSamples({mBufferOut, samplesToDo}, samplesOut, chandata->CurrentGains,
+            chandata->TargetGains, samplesToDo, 0);
+        ++chandata;
+    }
+}
+
+
+struct AutowahStateFactory final : public EffectStateFactory {
+    al::intrusive_ptr<EffectState> create() override
+    { return al::intrusive_ptr<EffectState>{new AutowahState{}}; }
+};
+
+} // namespace
+
+EffectStateFactory *AutowahStateFactory_getFactory()
+{
+    static AutowahStateFactory AutowahFactory{};
+    return &AutowahFactory;
+}