2021-04-28 12:43:51 +08:00
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/**
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2021-05-14 10:15:42 +08:00
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* This file is part of the OpenAL Soft cross platform audio library
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*
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2021-04-28 12:43:51 +08:00
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* Copyright (C) 2013 by Anis A. Hireche
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*
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2021-05-14 10:15:42 +08:00
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* * Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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*
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* * Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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2021-04-28 12:43:51 +08:00
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*
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2021-05-14 10:15:42 +08:00
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* * Neither the name of Spherical-Harmonic-Transform nor the names of its
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* contributors may be used to endorse or promote products derived from
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* this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE
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* LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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* POSSIBILITY OF SUCH DAMAGE.
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2021-04-28 12:43:51 +08:00
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*/
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#include "config.h"
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#include <array>
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#include <cstdlib>
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#include <iterator>
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#include <utility>
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#include "alc/effects/base.h"
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#include "almalloc.h"
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#include "alnumeric.h"
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#include "alspan.h"
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#include "core/ambidefs.h"
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#include "core/bufferline.h"
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#include "core/devformat.h"
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#include "core/device.h"
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#include "core/effectslot.h"
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#include "core/mixer.h"
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#include "core/mixer/defs.h"
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#include "intrusive_ptr.h"
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struct ContextBase;
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namespace {
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#define AMP_ENVELOPE_MIN 0.5f
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#define AMP_ENVELOPE_MAX 2.0f
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#define ATTACK_TIME 0.1f /* 100ms to rise from min to max */
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#define RELEASE_TIME 0.2f /* 200ms to drop from max to min */
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struct CompressorState final : public EffectState {
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/* Effect gains for each channel */
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float mGain[MaxAmbiChannels][MAX_OUTPUT_CHANNELS]{};
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/* Effect parameters */
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bool mEnabled{true};
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float mAttackMult{1.0f};
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float mReleaseMult{1.0f};
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float mEnvFollower{1.0f};
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void deviceUpdate(const DeviceBase *device, const Buffer &buffer) override;
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void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
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const EffectTarget target) override;
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void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
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const al::span<FloatBufferLine> samplesOut) override;
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DEF_NEWDEL(CompressorState)
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};
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void CompressorState::deviceUpdate(const DeviceBase *device, const Buffer&)
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{
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/* Number of samples to do a full attack and release (non-integer sample
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* counts are okay).
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*/
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const float attackCount{static_cast<float>(device->Frequency) * ATTACK_TIME};
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const float releaseCount{static_cast<float>(device->Frequency) * RELEASE_TIME};
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/* Calculate per-sample multipliers to attack and release at the desired
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* rates.
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*/
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mAttackMult = std::pow(AMP_ENVELOPE_MAX/AMP_ENVELOPE_MIN, 1.0f/attackCount);
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mReleaseMult = std::pow(AMP_ENVELOPE_MIN/AMP_ENVELOPE_MAX, 1.0f/releaseCount);
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}
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void CompressorState::update(const ContextBase*, const EffectSlot *slot,
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const EffectProps *props, const EffectTarget target)
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{
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mEnabled = props->Compressor.OnOff;
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mOutTarget = target.Main->Buffer;
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auto set_gains = [slot,target](auto &gains, al::span<const float,MaxAmbiChannels> coeffs)
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{ ComputePanGains(target.Main, coeffs.data(), slot->Gain, gains); };
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SetAmbiPanIdentity(std::begin(mGain), slot->Wet.Buffer.size(), set_gains);
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}
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void CompressorState::process(const size_t samplesToDo,
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const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
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{
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for(size_t base{0u};base < samplesToDo;)
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{
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float gains[256];
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const size_t td{minz(256, samplesToDo-base)};
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/* Generate the per-sample gains from the signal envelope. */
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float env{mEnvFollower};
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if(mEnabled)
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{
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for(size_t i{0u};i < td;++i)
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{
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/* Clamp the absolute amplitude to the defined envelope limits,
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* then attack or release the envelope to reach it.
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*/
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const float amplitude{clampf(std::fabs(samplesIn[0][base+i]), AMP_ENVELOPE_MIN,
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AMP_ENVELOPE_MAX)};
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if(amplitude > env)
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env = minf(env*mAttackMult, amplitude);
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else if(amplitude < env)
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env = maxf(env*mReleaseMult, amplitude);
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/* Apply the reciprocal of the envelope to normalize the volume
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* (compress the dynamic range).
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*/
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gains[i] = 1.0f / env;
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}
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}
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else
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{
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/* Same as above, except the amplitude is forced to 1. This helps
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* ensure smooth gain changes when the compressor is turned on and
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* off.
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*/
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for(size_t i{0u};i < td;++i)
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{
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const float amplitude{1.0f};
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if(amplitude > env)
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env = minf(env*mAttackMult, amplitude);
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else if(amplitude < env)
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env = maxf(env*mReleaseMult, amplitude);
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gains[i] = 1.0f / env;
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}
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}
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mEnvFollower = env;
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/* Now compress the signal amplitude to output. */
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auto changains = std::addressof(mGain[0]);
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for(const auto &input : samplesIn)
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{
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const float *outgains{*(changains++)};
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for(FloatBufferLine &output : samplesOut)
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{
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const float gain{*(outgains++)};
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if(!(std::fabs(gain) > GainSilenceThreshold))
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continue;
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for(size_t i{0u};i < td;i++)
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output[base+i] += input[base+i] * gains[i] * gain;
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}
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}
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base += td;
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}
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}
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struct CompressorStateFactory final : public EffectStateFactory {
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al::intrusive_ptr<EffectState> create() override
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{ return al::intrusive_ptr<EffectState>{new CompressorState{}}; }
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};
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} // namespace
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EffectStateFactory *CompressorStateFactory_getFactory()
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{
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static CompressorStateFactory CompressorFactory{};
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return &CompressorFactory;
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}
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