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Code sanitation.

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Default analog pin -1
This commit is contained in:
Blaz Kristan 2022-08-21 19:15:42 +02:00
parent d053bc562f
commit 720fae8720
1 changed files with 75 additions and 89 deletions
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60
usermods/audioreactive/audio_reactive.h
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@ -173,7 +173,6 @@ void FFTcode(void * parameter)
// see https://www.freertos.org/vtaskdelayuntil.html
const TickType_t xFrequency = FFT_MIN_CYCLE * portTICK_PERIOD_MS;
//const TickType_t xFrequency_2 = (FFT_MIN_CYCLE * portTICK_PERIOD_MS) / 2;
for(;;) {
TickType_t xLastWakeTime = xTaskGetTickCount();
@ -208,8 +207,8 @@ void FFTcode(void * parameter)
if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts
if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]);
}
// release sample to volume reactive effects
micDataReal = maxSample; // doing this early allows filters (getSample() and agcAvg()) to run on latest values - we'll have up-to-date gain and noise gate values when FFT is done
// release sample to volume reactive effects (not really necessary as float FFT calculation takes only 2ms)
micDataReal = maxSample;
// run FFT
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
@ -228,10 +227,6 @@ void FFTcode(void * parameter)
FFT.Compute( FFT_FORWARD ); // Compute FFT
FFT.ComplexToMagnitude(); // Compute magnitudes
#endif
//
// vReal[3 .. 255] contain useful data, each a 20Hz interval (60Hz - 5120Hz).
// There could be interesting data at bins 0 to 2, but there are too many artifacts.
//
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant
@ -294,7 +289,6 @@ void FFTcode(void * parameter)
fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
#endif
} else { // noise gate closed
for (int i=0; i < 16; i++) {
//fftCalc[i] *= 0.82f; // decay to zero
@ -338,8 +332,7 @@ void FFTcode(void * parameter)
// Logarithmic scaling
currentResult *= 0.42; // 42 is the answer ;-)
currentResult -= 8.0; // this skips the lowest row, giving some room for peaks
if (currentResult > 1.0)
currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
if (currentResult > 1.0) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
else currentResult = 0.0; // special handling, because log(1) = 0; log(0) = undefined
currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies
currentResult = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255]
@ -355,8 +348,7 @@ void FFTcode(void * parameter)
// square root scaling
currentResult *= 0.38f;
currentResult -= 6.0f;
if (currentResult > 1.0)
currentResult = sqrtf(currentResult);
if (currentResult > 1.0) currentResult = sqrtf(currentResult);
else currentResult = 0.0; // special handling, because sqrt(0) = undefined
currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
currentResult = mapf(currentResult, 0.0, 16.0, 0.0, 255.0); // map [sqrt(1) ... sqrt(256)] to [0 ... 255]
@ -394,7 +386,7 @@ class AudioReactive : public Usermod {
private:
#ifndef AUDIOPIN
int8_t audioPin = 36;
int8_t audioPin = -1;
#else
int8_t audioPin = AUDIOPIN;
#endif
@ -615,7 +607,6 @@ class AudioReactive : public Usermod {
if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
// MIC signal is "squelched" - deliver silence
//multAgcTemp = multAgc; // keep old control value (no change)
tmpAgc = 0;
// we need to "spin down" the intgrated error buffer
if (fabs(control_integrated) < 0.01) control_integrated = 0.0;
@ -628,7 +619,6 @@ class AudioReactive : public Usermod {
multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
}
// limit amplification
//multAgcTemp = constrain(multAgcTemp, 0.015625f, 32.0f); // 1/64 < multAgcTemp < 32
if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
@ -673,9 +663,6 @@ class AudioReactive : public Usermod {
else
sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
//userVar0 = sampleAvg * 4;
//if (userVar0 > 255) userVar0 = 255;
last_soundAgc = soundAgc;
} // agcAvg()
@ -751,13 +738,12 @@ class AudioReactive : public Usermod {
//if (userVar1 == 0) samplePeak = 0;
// Poor man's beat detection by seeing if sample > Average + some value.
// if (sample > (sampleAvg + maxVol) && millis() > (timeOfPeak + 200)) {
if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) { // This goes through ALL of the 255 bins - but ignores stupid settings
if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) {
// This goes through ALL of the 255 bins - but ignores stupid settings
// Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync.
samplePeak = true;
timeOfPeak = millis();
udpSamplePeak = true;
//userVar1 = samplePeak;
}
} // getSample()
@ -1239,51 +1225,52 @@ class AudioReactive : public Usermod {
uiDomString += F(" /><div class=\"sliderdisplay\"></div></div></div>"); //<output class=\"sliderbubble\"></output>
infoArr.add(uiDomString);
}
//else infoArr.add("<br/> <div>&nbsp;</div>"); // no processing - add empty line
// The following can be used for troubleshooting user errors and is so not enclosed in #ifdef WLED_DEBUG
// current Audio input
infoArr = user.createNestedArray(F("Audio Source"));
if (audioSyncEnabled & 0x02) {
// UDP sound sync - receive mode
infoArr.add("UDP sound sync");
infoArr.add(F("UDP sound sync"));
if (udpSyncConnected) {
if (millis() - last_UDPTime < 2500)
infoArr.add(" - receiving");
infoArr.add(F(" - receiving"));
else
infoArr.add(" - idle");
infoArr.add(F(" - idle"));
} else {
infoArr.add(" - no connection");
infoArr.add(F(" - no connection"));
}
} else {
// Analog or I2S digital input
if (audioSource && (audioSource->isInitialized())) {
// audio source sucessfully configured
if (audioSource->getType() == AudioSource::Type_I2SAdc) {
infoArr.add("ADC analog");
infoArr.add(F("ADC analog"));
} else {
infoArr.add("I2S digital");
infoArr.add(F("I2S digital"));
}
// input level or "silence"
if (maxSample5sec > 1.0) {
float my_usage = 100.0f * (maxSample5sec / 255.0f);
snprintf(myStringBuffer, 15, " - peak %3d%%", int(my_usage));
snprintf_P(myStringBuffer, 15, PSTR(" - peak %3d%%"), int(my_usage));
infoArr.add(myStringBuffer);
} else {
infoArr.add(" - quiet");
infoArr.add(F(" - quiet"));
}
} else {
// error during audio source setup
infoArr.add("not initialized");
infoArr.add(" - check GPIO config");
infoArr.add(F("not initialized"));
infoArr.add(F(" - check GPIO config"));
}
}
// Sound processing (FFT and input filters)
infoArr = user.createNestedArray(F("Sound Processing"));
if (audioSource && (disableSoundProcessing == false)) {
infoArr.add("running");
infoArr.add(F("running"));
} else {
infoArr.add("suspended");
infoArr.add(F("suspended"));
}
// AGC or manual Gain
@ -1303,14 +1290,13 @@ class AudioReactive : public Usermod {
infoArr = user.createNestedArray(F("UDP Sound Sync"));
if (audioSyncEnabled) {
if (audioSyncEnabled & 0x01) {
infoArr.add("send mode");
infoArr.add(F("send mode"));
} else if (audioSyncEnabled & 0x02) {
infoArr.add("receive mode");
infoArr.add(F("receive mode"));
}
} else
infoArr.add("off");
if (audioSyncEnabled && !udpSyncConnected) infoArr.add(" <i>(unconnected)</i>");
//if (!udpSyncConnected) infoArr.add(" <i>(unconnected)</i>");
#ifdef WLED_DEBUG
infoArr = user.createNestedArray(F("Sampling time"));