1 files changed, 25 insertions, 24 deletions

diff --git a/alc/effects/autowah.cpp b/alc/effects/autowah.cpp
index 4f874ef2..6d66f99f 100644
--- a/alc/effects/autowah.cpp
+++ b/alc/effects/autowah.cpp
@@ -59,12 +59,13 @@ struct AutowahState final : public EffectState {
     float mEnvDelay;
 
     /* Filter components derived from the envelope. */
-    struct {
+    struct FilterParam {
         float cos_w0;
         float alpha;
-    } mEnv[BufferLineSize];
+    };
+    std::array<FilterParam,BufferLineSize> mEnv;
 
-    struct {
+    struct ChannelData {
         uint mTargetChannel{InvalidChannelIndex};
 
         /* Effect filters' history. */
@@ -75,10 +76,11 @@ struct AutowahState final : public EffectState {
         /* Effect gains for each output channel */
         float mCurrentGain;
         float mTargetGain;
-    } mChans[MaxAmbiChannels];
+    };
+    std::array<ChannelData,MaxAmbiChannels> mChans;
 
     /* Effects buffers */
-    alignas(16) float mBufferOut[BufferLineSize];
+    alignas(16) FloatBufferLine mBufferOut;
 
 
     void deviceUpdate(const DeviceBase *device, const BufferStorage *buffer) override;
@@ -155,17 +157,16 @@ void AutowahState::process(const size_t samplesToDo,
     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;
+        const float sample{peak_gain * std::fabs(samplesIn[0][i])};
+        const float a{(sample > env_delay) ? attack_rate : release_rate};
         env_delay = lerpf(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::numbers::pi_v<float>*2.0f);
+        const float w0{minf((bandwidth*env_delay + freq_min), 0.46f) *
+            (al::numbers::pi_v<float>*2.0f)};
         mEnv[i].cos_w0 = std::cos(w0);
         mEnv[i].alpha = std::sin(w0)/(2.0f * QFactor);
     }
@@ -194,18 +195,18 @@ void AutowahState::process(const size_t samplesToDo,
         {
             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;
+
+            const std::array b{
+                1.0f + alpha*res_gain,
+                -2.0f * cos_w0,
+                1.0f - alpha*res_gain};
+            const std::array a{
+                1.0f + alpha/res_gain,
+                -2.0f * cos_w0,
+                1.0f - alpha/res_gain};
+
+            const float input{insamples[i]};
+            const float 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;
@@ -214,8 +215,8 @@ void AutowahState::process(const size_t samplesToDo,
         chandata->mFilter.z2 = z2;
 
         /* Now, mix the processed sound data to the output. */
-        MixSamples({mBufferOut, samplesToDo}, samplesOut[outidx].data(), chandata->mCurrentGain,
-            chandata->mTargetGain, samplesToDo);
+        MixSamples({mBufferOut.data(), samplesToDo}, samplesOut[outidx].data(),
+            chandata->mCurrentGain, chandata->mTargetGain, samplesToDo);
         ++chandata;
     }
 }