* BugFix: Correct convexDecomp compilation by setting the LINUX flag when necessary.

* BugFix: Update OpenAL to correct a compilation error on Linux.

Engine/lib/openal-soft/alc/effects/compressor.cpp

@@ -0,0 +1,192 @@
+/**
+ * This file is part of the OpenAL Soft cross platform audio library
+ *
+ * Copyright (C) 2013 by Anis A. Hireche
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * * Redistributions of source code must retain the above copyright notice,
+ *   this list of conditions and the following disclaimer.
+ *
+ * * Redistributions in binary form must reproduce the above copyright notice,
+ *   this list of conditions and the following disclaimer in the documentation
+ *   and/or other materials provided with the distribution.
+ *
+ * * Neither the name of Spherical-Harmonic-Transform nor the names of its
+ *   contributors may be used to endorse or promote products derived from
+ *   this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE
+ * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+ * POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+
+#include <array>
+#include <cstdlib>
+#include <iterator>
+#include <utility>
+
+#include "alc/effects/base.h"
+#include "almalloc.h"
+#include "alnumeric.h"
+#include "alspan.h"
+#include "core/ambidefs.h"
+#include "core/bufferline.h"
+#include "core/devformat.h"
+#include "core/device.h"
+#include "core/effectslot.h"
+#include "core/mixer.h"
+#include "core/mixer/defs.h"
+#include "intrusive_ptr.h"
+
+struct ContextBase;
+
+
+namespace {
+
+#define AMP_ENVELOPE_MIN  0.5f
+#define AMP_ENVELOPE_MAX  2.0f
+
+#define ATTACK_TIME  0.1f /* 100ms to rise from min to max */
+#define RELEASE_TIME 0.2f /* 200ms to drop from max to min */
+
+
+struct CompressorState final : public EffectState {
+    /* Effect gains for each channel */
+    float mGain[MaxAmbiChannels][MAX_OUTPUT_CHANNELS]{};
+
+    /* Effect parameters */
+    bool mEnabled{true};
+    float mAttackMult{1.0f};
+    float mReleaseMult{1.0f};
+    float mEnvFollower{1.0f};
+
+
+    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(CompressorState)
+};
+
+void CompressorState::deviceUpdate(const DeviceBase *device, const Buffer&)
+{
+    /* Number of samples to do a full attack and release (non-integer sample
+     * counts are okay).
+     */
+    const float attackCount{static_cast<float>(device->Frequency) * ATTACK_TIME};
+    const float releaseCount{static_cast<float>(device->Frequency) * RELEASE_TIME};
+
+    /* Calculate per-sample multipliers to attack and release at the desired
+     * rates.
+     */
+    mAttackMult  = std::pow(AMP_ENVELOPE_MAX/AMP_ENVELOPE_MIN, 1.0f/attackCount);
+    mReleaseMult = std::pow(AMP_ENVELOPE_MIN/AMP_ENVELOPE_MAX, 1.0f/releaseCount);
+}
+
+void CompressorState::update(const ContextBase*, const EffectSlot *slot,
+    const EffectProps *props, const EffectTarget target)
+{
+    mEnabled = props->Compressor.OnOff;
+
+    mOutTarget = target.Main->Buffer;
+    auto set_gains = [slot,target](auto &gains, al::span<const float,MaxAmbiChannels> coeffs)
+    { ComputePanGains(target.Main, coeffs.data(), slot->Gain, gains); };
+    SetAmbiPanIdentity(std::begin(mGain), slot->Wet.Buffer.size(), set_gains);
+}
+
+void CompressorState::process(const size_t samplesToDo,
+    const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
+{
+    for(size_t base{0u};base < samplesToDo;)
+    {
+        float gains[256];
+        const size_t td{minz(256, samplesToDo-base)};
+
+        /* Generate the per-sample gains from the signal envelope. */
+        float env{mEnvFollower};
+        if(mEnabled)
+        {
+            for(size_t i{0u};i < td;++i)
+            {
+                /* Clamp the absolute amplitude to the defined envelope limits,
+                 * then attack or release the envelope to reach it.
+                 */
+                const float amplitude{clampf(std::fabs(samplesIn[0][base+i]), AMP_ENVELOPE_MIN,
+                    AMP_ENVELOPE_MAX)};
+                if(amplitude > env)
+                    env = minf(env*mAttackMult, amplitude);
+                else if(amplitude < env)
+                    env = maxf(env*mReleaseMult, amplitude);
+
+                /* Apply the reciprocal of the envelope to normalize the volume
+                 * (compress the dynamic range).
+                 */
+                gains[i] = 1.0f / env;
+            }
+        }
+        else
+        {
+            /* Same as above, except the amplitude is forced to 1. This helps
+             * ensure smooth gain changes when the compressor is turned on and
+             * off.
+             */
+            for(size_t i{0u};i < td;++i)
+            {
+                const float amplitude{1.0f};
+                if(amplitude > env)
+                    env = minf(env*mAttackMult, amplitude);
+                else if(amplitude < env)
+                    env = maxf(env*mReleaseMult, amplitude);
+
+                gains[i] = 1.0f / env;
+            }
+        }
+        mEnvFollower = env;
+
+        /* Now compress the signal amplitude to output. */
+        auto changains = std::addressof(mGain[0]);
+        for(const auto &input : samplesIn)
+        {
+            const float *outgains{*(changains++)};
+            for(FloatBufferLine &output : samplesOut)
+            {
+                const float gain{*(outgains++)};
+                if(!(std::fabs(gain) > GainSilenceThreshold))
+                    continue;
+
+                for(size_t i{0u};i < td;i++)
+                    output[base+i] += input[base+i] * gains[i] * gain;
+            }
+        }
+
+        base += td;
+    }
+}
+
+
+struct CompressorStateFactory final : public EffectStateFactory {
+    al::intrusive_ptr<EffectState> create() override
+    { return al::intrusive_ptr<EffectState>{new CompressorState{}}; }
+};
+
+} // namespace
+
+EffectStateFactory *CompressorStateFactory_getFactory()
+{
+    static CompressorStateFactory CompressorFactory{};
+    return &CompressorFactory;
+}