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Merge remote-tracking branch 'origin/audioreactive-prototype' into audio-reactive

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Frank
2022-10-06 18:22:58 +02:00
parent 5ed0bb8c2b 64441b39ac
commit f5b74a2b79
1 changed files with 9 additions and 6 deletions
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15
usermods/audioreactive/audio_reactive.h
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@@ -278,7 +278,7 @@ 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)
@@ -328,7 +328,7 @@ 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.
*
@@ -387,7 +387,7 @@ void FFTcode(void * parameter)
if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK;
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[pinkIndex][i];
if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
@@ -739,7 +739,7 @@ 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
@@ -797,6 +797,7 @@ 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()
@@ -835,12 +836,13 @@ 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
@@ -862,6 +864,7 @@ 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()