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critical bugfix

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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.
This commit is contained in:
Frank 2022-10-06 15:39:28 +02:00
parent 1ae0dd574d
commit 64441b39ac
1 changed files with 9 additions and 6 deletions
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15
usermods/audioreactive/audio_reactive.h
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@ -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()