Code sanitation.
Default analog pin -1
This commit is contained in:
Blaz Kristan
parent
d053bc562f
commit
720fae8720
@ -173,7 +173,6 @@ void FFTcode(void * parameter)
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// see https://www.freertos.org/vtaskdelayuntil.html
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const TickType_t xFrequency = FFT_MIN_CYCLE * portTICK_PERIOD_MS;
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//const TickType_t xFrequency_2 = (FFT_MIN_CYCLE * portTICK_PERIOD_MS) / 2;
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for(;;) {
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TickType_t xLastWakeTime = xTaskGetTickCount();
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@ -208,8 +207,8 @@ void FFTcode(void * parameter)
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if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts
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if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]);
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}
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// release sample to volume reactive effects
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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
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// release sample to volume reactive effects (not really necessary as float FFT calculation takes only 2ms)
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micDataReal = maxSample;
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// run FFT
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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@ -228,10 +227,6 @@ void FFTcode(void * parameter)
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FFT.Compute( FFT_FORWARD ); // Compute FFT
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FFT.ComplexToMagnitude(); // Compute magnitudes
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#endif
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//
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// vReal[3 .. 255] contain useful data, each a 20Hz interval (60Hz - 5120Hz).
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// There could be interesting data at bins 0 to 2, but there are too many artifacts.
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//
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant
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@ -240,9 +235,9 @@ void FFTcode(void * parameter)
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#endif
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FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
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for (int i = 0; i < samplesFFT_2; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3.
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for (int i = 0; i < samplesFFT_2; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3.
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float t = fabsf(vReal[i]); // just to be sure - values in fft bins should be positive any way
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fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max.
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fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max.
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} // for()
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@ -294,7 +289,6 @@ void FFTcode(void * parameter)
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fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
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// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
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#endif
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} else { // noise gate closed
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for (int i=0; i < 16; i++) {
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//fftCalc[i] *= 0.82f; // decay to zero
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@ -316,7 +310,7 @@ void FFTcode(void * parameter)
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// smooth results - rise fast, fall slower
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if(fftCalc[i] > fftAvg[i]) // rise fast
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fftAvg[i] = fftCalc[i] *0.75f + 0.25f*fftAvg[i]; // will need approx 2 cycles (50ms) for converging against fftCalc[i]
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fftAvg[i] = fftCalc[i] *0.75f + 0.25f*fftAvg[i]; // will need approx 2 cycles (50ms) for converging against fftCalc[i]
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else { // fall slow
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if (decayTime < 1000) fftAvg[i] = fftCalc[i]*0.22f + 0.78f*fftAvg[i]; // approx 5 cycles (225ms) for falling to zero
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else if (decayTime < 2000) fftAvg[i] = fftCalc[i]*0.17f + 0.83f*fftAvg[i]; // default - approx 9 cycles (225ms) for falling to zero
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@ -336,29 +330,27 @@ void FFTcode(void * parameter)
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switch (FFTScalingMode) {
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case 1:
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// Logarithmic scaling
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currentResult *= 0.42; // 42 is the answer ;-)
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currentResult -= 8.0; // this skips the lowest row, giving some room for peaks
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if (currentResult > 1.0)
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currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
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else currentResult = 0.0; // special handling, because log(1) = 0; log(0) = undefined
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currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies
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currentResult *= 0.42; // 42 is the answer ;-)
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currentResult -= 8.0; // this skips the lowest row, giving some room for peaks
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if (currentResult > 1.0) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
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else currentResult = 0.0; // special handling, because log(1) = 0; log(0) = undefined
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currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies
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currentResult = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255]
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break;
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case 2:
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// Linear scaling
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currentResult *= 0.30f; // needs a bit more damping, get stay below 255
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currentResult -= 4.0; // giving a bit more room for peaks
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currentResult *= 0.30f; // needs a bit more damping, get stay below 255
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currentResult -= 4.0; // giving a bit more room for peaks
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if (currentResult < 1.0f) currentResult = 0.0f;
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currentResult *= 0.85f + (float(i)/1.8f); // extra up-scaling for high frequencies
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currentResult *= 0.85f + (float(i)/1.8f); // extra up-scaling for high frequencies
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break;
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case 3:
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// square root scaling
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currentResult *= 0.38f;
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currentResult -= 6.0f;
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if (currentResult > 1.0)
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currentResult = sqrtf(currentResult);
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else currentResult = 0.0; // special handling, because sqrt(0) = undefined
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currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
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if (currentResult > 1.0) currentResult = sqrtf(currentResult);
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else currentResult = 0.0; // special handling, because sqrt(0) = undefined
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currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
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currentResult = mapf(currentResult, 0.0, 16.0, 0.0, 255.0); // map [sqrt(1) ... sqrt(256)] to [0 ... 255]
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break;
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@ -394,7 +386,7 @@ class AudioReactive : public Usermod {
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private:
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#ifndef AUDIOPIN
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int8_t audioPin = 36;
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int8_t audioPin = -1;
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#else
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int8_t audioPin = AUDIOPIN;
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#endif
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@ -615,7 +607,6 @@ class AudioReactive : public Usermod {
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if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
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// MIC signal is "squelched" - deliver silence
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//multAgcTemp = multAgc; // keep old control value (no change)
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tmpAgc = 0;
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// we need to "spin down" the intgrated error buffer
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if (fabs(control_integrated) < 0.01) control_integrated = 0.0;
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@ -623,27 +614,26 @@ class AudioReactive : public Usermod {
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} else {
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// compute new setpoint
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if (tmpAgc <= agcTarget0Up[AGC_preset])
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multAgcTemp = agcTarget0[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = first setpoint
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multAgcTemp = agcTarget0[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = first setpoint
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else
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multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
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multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
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}
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// limit amplification
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//multAgcTemp = constrain(multAgcTemp, 0.015625f, 32.0f); // 1/64 < multAgcTemp < 32
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if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
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if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
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// compute error terms
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control_error = multAgcTemp - lastMultAgc;
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if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
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&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
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control_integrated += control_error * 0.002 * 0.25; // 2ms = intgration time; 0.25 for damping
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if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
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&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
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control_integrated += control_error * 0.002 * 0.25; // 2ms = intgration time; 0.25 for damping
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else
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control_integrated *= 0.9; // spin down that beasty integrator
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control_integrated *= 0.9; // spin down that beasty integrator
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// apply PI Control
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tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain
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if ((tmpAgc > agcZoneHigh[AGC_preset]) || (tmpAgc < soundSquelch + agcZoneLow[AGC_preset])) { // upper/lower emergy zone
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tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain
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if ((tmpAgc > agcZoneHigh[AGC_preset]) || (tmpAgc < soundSquelch + agcZoneLow[AGC_preset])) { // upper/lower emergy zone
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multAgcTemp = lastMultAgc + agcFollowFast[AGC_preset] * agcControlKp[AGC_preset] * control_error;
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multAgcTemp += agcFollowFast[AGC_preset] * agcControlKi[AGC_preset] * control_integrated;
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} else { // "normal zone"
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@ -673,9 +663,6 @@ class AudioReactive : public Usermod {
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else
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sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
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//userVar0 = sampleAvg * 4;
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//if (userVar0 > 255) userVar0 = 255;
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last_soundAgc = soundAgc;
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} // agcAvg()
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@ -711,7 +698,7 @@ class AudioReactive : public Usermod {
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micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
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if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
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micIn -= micLev; // Let's center it to 0 now
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micIn -= micLev; // Let's center it to 0 now
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// Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
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float micInNoDC = fabs(micDataReal - micLev);
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expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF);
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@ -725,12 +712,12 @@ class AudioReactive : public Usermod {
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sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment
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sampleReal = tmpSample;
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sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
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sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
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sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
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sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
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// keep "peak" sample, but decay value if current sample is below peak
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if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) {
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sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
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sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
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} else {
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if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0))
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sampleMax += 0.5f * (sampleReal - sampleMax); // over AGC Zone - get back quickly
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@ -751,13 +738,12 @@ class AudioReactive : public Usermod {
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//if (userVar1 == 0) samplePeak = 0;
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// Poor man's beat detection by seeing if sample > Average + some value.
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// if (sample > (sampleAvg + maxVol) && millis() > (timeOfPeak + 200)) {
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if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) { // This goes through ALL of the 255 bins - but ignores stupid settings
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// 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.
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if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) {
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// This goes through ALL of the 255 bins - but ignores stupid settings
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// 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.
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samplePeak = true;
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timeOfPeak = millis();
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udpSamplePeak = true;
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//userVar1 = samplePeak;
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}
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} // getSample()
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@ -771,7 +757,7 @@ class AudioReactive : public Usermod {
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static unsigned long last_time = 0;
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static float last_volumeSmth = 0.0f;
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if(limiterOn == false) return;
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if (limiterOn == false) return;
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long delta_time = millis() - last_time;
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delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> sily lil hick-up
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@ -1043,7 +1029,7 @@ class AudioReactive : public Usermod {
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if (audioSyncEnabled & 0x02) disableSoundProcessing = true; // make sure everything is disabled IF in audio Receive mode
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if (audioSyncEnabled & 0x01) disableSoundProcessing = false; // keep running audio IF we're in audio Transmit mode
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if(!audioSource->isInitialized()) disableSoundProcessing = true; // no audio source
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if (!audioSource->isInitialized()) disableSoundProcessing = true; // no audio source
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// Only run the sampling code IF we're not in Receive mode or realtime mode
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@ -1070,7 +1056,7 @@ class AudioReactive : public Usermod {
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agcAvg(t_now - userloopDelay); // Calculated the PI adjusted value as sampleAvg
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userloopDelay -= 2; // advance "simulated time" by 2ms
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} while (userloopDelay > 0);
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lastUMRun = t_now; // update time keeping
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lastUMRun = t_now; // update time keeping
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// update samples for effects (raw, smooth)
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volumeSmth = (soundAgc) ? sampleAgc : sampleAvg;
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@ -1078,7 +1064,7 @@ class AudioReactive : public Usermod {
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// update FFTMagnitude, taking into account AGC amplification
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my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects
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if (soundAgc) my_magnitude *= multAgc;
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if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
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if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
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limitSampleDynamics(); // optional - makes volumeSmth very smooth and fluent
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}
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@ -1239,61 +1225,62 @@ class AudioReactive : public Usermod {
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uiDomString += F(" /><div class=\"sliderdisplay\"></div></div></div>"); //<output class=\"sliderbubble\"></output>
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infoArr.add(uiDomString);
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}
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//else infoArr.add("<br/> <div> </div>"); // no processing - add empty line
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// The following can be used for troubleshooting user errors and is so not enclosed in #ifdef WLED_DEBUG
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// current Audio input
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infoArr = user.createNestedArray(F("Audio Source"));
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if (audioSyncEnabled & 0x02) {
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// UDP sound sync - receive mode
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infoArr.add("UDP sound sync");
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if (udpSyncConnected) {
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if (millis() - last_UDPTime < 2500)
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infoArr.add(" - receiving");
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else
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infoArr.add(" - idle");
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} else {
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infoArr.add(" - no connection");
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}
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// UDP sound sync - receive mode
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infoArr.add(F("UDP sound sync"));
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if (udpSyncConnected) {
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if (millis() - last_UDPTime < 2500)
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infoArr.add(F(" - receiving"));
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else
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infoArr.add(F(" - idle"));
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} else {
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infoArr.add(F(" - no connection"));
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}
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} else {
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// Analog or I2S digital input
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if (audioSource && (audioSource->isInitialized())) {
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// audio source sucessfully configured
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if(audioSource->getType() == AudioSource::Type_I2SAdc) {
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infoArr.add("ADC analog");
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} else {
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infoArr.add("I2S digital");
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}
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// input level or "silence"
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if (maxSample5sec > 1.0) {
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float my_usage = 100.0f * (maxSample5sec / 255.0f);
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snprintf(myStringBuffer, 15, " - peak %3d%%", int(my_usage));
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infoArr.add(myStringBuffer);
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} else {
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infoArr.add(" - quiet");
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}
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// audio source sucessfully configured
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if (audioSource->getType() == AudioSource::Type_I2SAdc) {
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infoArr.add(F("ADC analog"));
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} else {
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infoArr.add(F("I2S digital"));
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}
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// input level or "silence"
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if (maxSample5sec > 1.0) {
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float my_usage = 100.0f * (maxSample5sec / 255.0f);
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snprintf_P(myStringBuffer, 15, PSTR(" - peak %3d%%"), int(my_usage));
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infoArr.add(myStringBuffer);
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} else {
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infoArr.add(F(" - quiet"));
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}
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} else {
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// error during audio source setup
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infoArr.add("not initialized");
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infoArr.add(" - check GPIO config");
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// error during audio source setup
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infoArr.add(F("not initialized"));
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infoArr.add(F(" - check GPIO config"));
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}
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}
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// Sound processing (FFT and input filters)
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infoArr = user.createNestedArray(F("Sound Processing"));
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if (audioSource && (disableSoundProcessing == false)) {
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infoArr.add("running");
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infoArr.add(F("running"));
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} else {
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infoArr.add("suspended");
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infoArr.add(F("suspended"));
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}
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// AGC or manual Gain
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if((soundAgc==0) && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
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if ((soundAgc==0) && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
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infoArr = user.createNestedArray(F("Manual Gain"));
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float myGain = ((float)sampleGain/40.0f * (float)inputLevel/128.0f) + 1.0f/16.0f; // non-AGC gain from presets
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infoArr.add(roundf(myGain*100.0f) / 100.0f);
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infoArr.add("x");
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}
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if(soundAgc && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
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if (soundAgc && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
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infoArr = user.createNestedArray(F("AGC Gain"));
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infoArr.add(roundf(multAgc*100.0f) / 100.0f);
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infoArr.add("x");
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@ -1302,15 +1289,14 @@ class AudioReactive : public Usermod {
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// UDP Sound Sync status
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infoArr = user.createNestedArray(F("UDP Sound Sync"));
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if (audioSyncEnabled) {
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if (audioSyncEnabled & 0x01) {
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infoArr.add("send mode");
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} else if (audioSyncEnabled & 0x02) {
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infoArr.add("receive mode");
|
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}
|
||||
if (audioSyncEnabled & 0x01) {
|
||||
infoArr.add(F("send mode"));
|
||||
} else if (audioSyncEnabled & 0x02) {
|
||||
infoArr.add(F("receive mode"));
|
||||
}
|
||||
} else
|
||||
infoArr.add("off");
|
||||
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"));
|
||||
|
||||
Loading…
Reference in New Issue
Block a user