Move some temp variables closer to where they're used
[openal-soft.git] / alc / effects / autowah.cpp
blob46cc8fb07aea8b9d0e3f5e5d373230383cbbfd6a
1 /**
2 * OpenAL cross platform audio library
3 * Copyright (C) 2018 by Raul Herraiz.
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc.,
17 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
18 * Or go to http://www.gnu.org/copyleft/lgpl.html
21 #include "config.h"
23 #include <algorithm>
24 #include <array>
25 #include <cstdlib>
26 #include <iterator>
27 #include <utility>
29 #include "alc/effects/base.h"
30 #include "almalloc.h"
31 #include "alnumbers.h"
32 #include "alnumeric.h"
33 #include "alspan.h"
34 #include "core/ambidefs.h"
35 #include "core/bufferline.h"
36 #include "core/context.h"
37 #include "core/devformat.h"
38 #include "core/device.h"
39 #include "core/effectslot.h"
40 #include "core/mixer.h"
41 #include "intrusive_ptr.h"
44 namespace {
46 constexpr float GainScale{31621.0f};
47 constexpr float MinFreq{20.0f};
48 constexpr float MaxFreq{2500.0f};
49 constexpr float QFactor{5.0f};
51 struct AutowahState final : public EffectState {
52 /* Effect parameters */
53 float mAttackRate;
54 float mReleaseRate;
55 float mResonanceGain;
56 float mPeakGain;
57 float mFreqMinNorm;
58 float mBandwidthNorm;
59 float mEnvDelay;
61 /* Filter components derived from the envelope. */
62 struct {
63 float cos_w0;
64 float alpha;
65 } mEnv[BufferLineSize];
67 struct {
68 /* Effect filters' history. */
69 struct {
70 float z1, z2;
71 } Filter;
73 /* Effect gains for each output channel */
74 float CurrentGains[MAX_OUTPUT_CHANNELS];
75 float TargetGains[MAX_OUTPUT_CHANNELS];
76 } mChans[MaxAmbiChannels];
78 /* Effects buffers */
79 alignas(16) float mBufferOut[BufferLineSize];
82 void deviceUpdate(const DeviceBase *device, const Buffer &buffer) override;
83 void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
84 const EffectTarget target) override;
85 void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
86 const al::span<FloatBufferLine> samplesOut) override;
88 DEF_NEWDEL(AutowahState)
91 void AutowahState::deviceUpdate(const DeviceBase*, const Buffer&)
93 /* (Re-)initializing parameters and clear the buffers. */
95 mAttackRate = 1.0f;
96 mReleaseRate = 1.0f;
97 mResonanceGain = 10.0f;
98 mPeakGain = 4.5f;
99 mFreqMinNorm = 4.5e-4f;
100 mBandwidthNorm = 0.05f;
101 mEnvDelay = 0.0f;
103 for(auto &e : mEnv)
105 e.cos_w0 = 0.0f;
106 e.alpha = 0.0f;
109 for(auto &chan : mChans)
111 std::fill(std::begin(chan.CurrentGains), std::end(chan.CurrentGains), 0.0f);
112 chan.Filter.z1 = 0.0f;
113 chan.Filter.z2 = 0.0f;
117 void AutowahState::update(const ContextBase *context, const EffectSlot *slot,
118 const EffectProps *props, const EffectTarget target)
120 const DeviceBase *device{context->mDevice};
121 const auto frequency = static_cast<float>(device->Frequency);
123 const float ReleaseTime{clampf(props->Autowah.ReleaseTime, 0.001f, 1.0f)};
125 mAttackRate = std::exp(-1.0f / (props->Autowah.AttackTime*frequency));
126 mReleaseRate = std::exp(-1.0f / (ReleaseTime*frequency));
127 /* 0-20dB Resonance Peak gain */
128 mResonanceGain = std::sqrt(std::log10(props->Autowah.Resonance)*10.0f / 3.0f);
129 mPeakGain = 1.0f - std::log10(props->Autowah.PeakGain / GainScale);
130 mFreqMinNorm = MinFreq / frequency;
131 mBandwidthNorm = (MaxFreq-MinFreq) / frequency;
133 mOutTarget = target.Main->Buffer;
134 auto set_gains = [slot,target](auto &chan, al::span<const float,MaxAmbiChannels> coeffs)
135 { ComputePanGains(target.Main, coeffs.data(), slot->Gain, chan.TargetGains); };
136 SetAmbiPanIdentity(std::begin(mChans), slot->Wet.Buffer.size(), set_gains);
139 void AutowahState::process(const size_t samplesToDo,
140 const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
142 const float attack_rate{mAttackRate};
143 const float release_rate{mReleaseRate};
144 const float res_gain{mResonanceGain};
145 const float peak_gain{mPeakGain};
146 const float freq_min{mFreqMinNorm};
147 const float bandwidth{mBandwidthNorm};
149 float env_delay{mEnvDelay};
150 for(size_t i{0u};i < samplesToDo;i++)
152 float w0, sample, a;
154 /* Envelope follower described on the book: Audio Effects, Theory,
155 * Implementation and Application.
157 sample = peak_gain * std::fabs(samplesIn[0][i]);
158 a = (sample > env_delay) ? attack_rate : release_rate;
159 env_delay = lerp(sample, env_delay, a);
161 /* Calculate the cos and alpha components for this sample's filter. */
162 w0 = minf((bandwidth*env_delay + freq_min), 0.46f) * (al::numbers::pi_v<float>*2.0f);
163 mEnv[i].cos_w0 = std::cos(w0);
164 mEnv[i].alpha = std::sin(w0)/(2.0f * QFactor);
166 mEnvDelay = env_delay;
168 auto chandata = std::addressof(mChans[0]);
169 for(const auto &insamples : samplesIn)
171 /* This effectively inlines BiquadFilter_setParams for a peaking
172 * filter and BiquadFilter_processC. The alpha and cosine components
173 * for the filter coefficients were previously calculated with the
174 * envelope. Because the filter changes for each sample, the
175 * coefficients are transient and don't need to be held.
177 float z1{chandata->Filter.z1};
178 float z2{chandata->Filter.z2};
180 for(size_t i{0u};i < samplesToDo;i++)
182 const float alpha{mEnv[i].alpha};
183 const float cos_w0{mEnv[i].cos_w0};
184 float input, output;
185 float a[3], b[3];
187 b[0] = 1.0f + alpha*res_gain;
188 b[1] = -2.0f * cos_w0;
189 b[2] = 1.0f - alpha*res_gain;
190 a[0] = 1.0f + alpha/res_gain;
191 a[1] = -2.0f * cos_w0;
192 a[2] = 1.0f - alpha/res_gain;
194 input = insamples[i];
195 output = input*(b[0]/a[0]) + z1;
196 z1 = input*(b[1]/a[0]) - output*(a[1]/a[0]) + z2;
197 z2 = input*(b[2]/a[0]) - output*(a[2]/a[0]);
198 mBufferOut[i] = output;
200 chandata->Filter.z1 = z1;
201 chandata->Filter.z2 = z2;
203 /* Now, mix the processed sound data to the output. */
204 MixSamples({mBufferOut, samplesToDo}, samplesOut, chandata->CurrentGains,
205 chandata->TargetGains, samplesToDo, 0);
206 ++chandata;
211 struct AutowahStateFactory final : public EffectStateFactory {
212 al::intrusive_ptr<EffectState> create() override
213 { return al::intrusive_ptr<EffectState>{new AutowahState{}}; }
216 } // namespace
218 EffectStateFactory *AutowahStateFactory_getFactory()
220 static AutowahStateFactory AutowahFactory{};
221 return &AutowahFactory;