From 64441b39ac4f7ce2f4db812e8c9475da9a76daec Mon Sep 17 00:00:00 2001
From: Frank <91616163+softhack007@users.noreply.github.com>
Date: Thu, 6 Oct 2022 15:39:28 +0200
Subject: [PATCH] critical bugfix

it's possible that volume samples become negative. In this scenario, our simple noise gate does stupid things, and it looks like "effects temporarily lost the sound".
This fix improves the situation, and makes sure that volume samples are always >= 0.
---
 usermods/audioreactive/audio_reactive.h | 15 +++++++++------
 1 file changed, 9 insertions(+), 6 deletions(-)

diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h
index 2a6efbc6..d7fbee7c 100644
--- a/usermods/audioreactive/audio_reactive.h
+++ b/usermods/audioreactive/audio_reactive.h
@@ -229,7 +229,7 @@ void FFTcode(void * parameter)
 #ifdef SR_DEBUG
     if (true) {  // this allows measure FFT runtimes, as it disables the "only when needed" optimization 
 #else
-    if (sampleAvg > 1) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed.
+    if (sampleAvg > 0.5f) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed.
 #endif
 
       // run FFT (takes 3-5ms on ESP32, ~12ms on ESP32-S2)
@@ -273,7 +273,7 @@ void FFTcode(void * parameter)
     } // for()
 
     // mapping of FFT result bins to frequency channels
-    if (sampleAvg > 1) { // noise gate open
+    if (sampleAvg > 0.5f) { // noise gate open
 #if 0
     /* This FFT post processing is a DIY endeavour. What we really need is someone with sound engineering expertise to do a great job here AND most importantly, that the animations look GREAT as a result.
     *
@@ -331,7 +331,7 @@ void FFTcode(void * parameter)
     // post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
     for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
 
-      if (sampleAvg > 1) { // noise gate open
+      if (sampleAvg > 0.5f) { // noise gate open
         // Adjustment for frequency curves.
         fftCalc[i] *= fftResultPink[i];
         if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE;  // adjustment related to FFT windowing function
@@ -678,7 +678,7 @@ class AudioReactive : public Usermod {
       if (time_now - last_time > 2)  {
         last_time = time_now;
 
-        if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
+        if((fabsf(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
           // MIC signal is "squelched" - deliver silence
           tmpAgc = 0;
           // we need to "spin down" the intgrated error buffer
@@ -736,6 +736,7 @@ class AudioReactive : public Usermod {
       else
         sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
 
+      sampleAgc = fabsf(sampleAgc);                                      // // make sure we have a positive value
       last_soundAgc = soundAgc;
     } // agcAvg()
 
@@ -773,12 +774,13 @@ class AudioReactive : public Usermod {
 
       micIn -= micLev;                                  // Let's center it to 0 now
       // Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
-      float micInNoDC = fabs(micDataReal - micLev);
+      float micInNoDC = fabsf(micDataReal - micLev);
       expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF);
+      expAdjF = fabsf(expAdjF);                         // Now (!) take the absolute value
+
       expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
       if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
 
-      expAdjF = fabsf(expAdjF);                         // Now (!) take the absolute value
       tmpSample = expAdjF;
       micIn = abs(micIn);                               // And get the absolute value of each sample
 
@@ -800,6 +802,7 @@ class AudioReactive : public Usermod {
       if (sampleMax < 0.5f) sampleMax = 0.0f;
 
       sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f;   // Smooth it out over the last 16 samples.
+      sampleAvg = fabsf(sampleAvg);                            // make sure we have a positive value
     } // getSample()