mic quality dropdown, cleanup
* mic quality: When set to "low noise" or "perfect", only minimal smoothing is performed on the "Mic Volume" input. Every filter adds a delay, so this option can lead to better on-spot responses from effects. * cleanup: removed a few unused variables and unused code.
This commit is contained in:
Frank
@@ -178,6 +178,7 @@ static bool limiterOn = false; // bool: enable / disable dynamic
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#else
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static bool limiterOn = true;
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#endif
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static uint8_t micQuality = 0; // affects input filtering; 0 normal, 1 minimal filtering, 2 no filtering
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#ifdef FFT_USE_SLIDING_WINDOW
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static uint16_t attackTime = 24; // int: attack time in milliseconds. Default 0.024sec
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static uint16_t decayTime = 250; // int: decay time in milliseconds. New default 250ms.
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@@ -690,7 +691,6 @@ void FFTcode(void * parameter)
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float wc = 1.0; // FFT window correction factor, relative to Blackman_Harris
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// run FFT (takes 3-5ms on ESP32)
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//if (fabsf(sampleAvg) > 0.25f) { // noise gate open
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if (fabsf(volumeSmth) > 0.25f) { // noise gate open
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if ((skipSecondFFT == false) || (isFirstRun == true)) {
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// run FFT (takes 2-3ms on ESP32, ~12ms on ESP32-S2, ~30ms on -C3)
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@@ -776,7 +776,6 @@ void FFTcode(void * parameter)
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}
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if ((skipSecondFFT == false) || (isFirstRun == true)) {
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for (int i = 0; i < samplesFFT; i++) {
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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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vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max.
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@@ -785,102 +784,72 @@ void FFTcode(void * parameter)
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// mapping of FFT result bins to frequency channels
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//if (fabsf(sampleAvg) > 0.25f) { // noise gate open
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if (fabsf(volumeSmth) > 0.25f) { // noise gate open
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#if 0
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/* 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.
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*
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* Andrew's updated mapping of 256 bins down to the 16 result bins with Sample Freq = 10240, samplesFFT = 512 and some overlap.
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* Based on testing, the lowest/Start frequency is 60 Hz (with bin 3) and a highest/End frequency of 5120 Hz in bin 255.
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* Now, Take the 60Hz and multiply by 1.320367784 to get the next frequency and so on until the end. Then determine the bins.
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* End frequency = Start frequency * multiplier ^ 16
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* Multiplier = (End frequency/ Start frequency) ^ 1/16
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* Multiplier = 1.320367784
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*/ // Range
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fftCalc[ 0] = wc * fftAddAvg(2,4); // 60 - 100
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fftCalc[ 1] = wc * fftAddAvg(4,5); // 80 - 120
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fftCalc[ 2] = wc * fftAddAvg(5,7); // 100 - 160
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fftCalc[ 3] = wc * fftAddAvg(7,9); // 140 - 200
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fftCalc[ 4] = wc * fftAddAvg(9,12); // 180 - 260
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fftCalc[ 5] = wc * fftAddAvg(12,16); // 240 - 340
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fftCalc[ 6] = wc * fftAddAvg(16,21); // 320 - 440
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fftCalc[ 7] = wc * fftAddAvg(21,29); // 420 - 600
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fftCalc[ 8] = wc * fftAddAvg(29,37); // 580 - 760
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fftCalc[ 9] = wc * fftAddAvg(37,48); // 740 - 980
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fftCalc[10] = wc * fftAddAvg(48,64); // 960 - 1300
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fftCalc[11] = wc * fftAddAvg(64,84); // 1280 - 1700
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fftCalc[12] = wc * fftAddAvg(84,111); // 1680 - 2240
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fftCalc[13] = wc * fftAddAvg(111,147); // 2220 - 2960
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fftCalc[14] = wc * fftAddAvg(147,194); // 2940 - 3900
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fftCalc[15] = wc * fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate
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#else
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//WLEDMM: different distributions
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if (freqDist == 0) {
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/* new mapping, optimized for 22050 Hz by softhack007 --- update: removed overlap */
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// bins frequency range
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if (useInputFilter==1) {
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// skip frequencies below 100hz
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fftCalc[ 0] = wc * 0.8f * fftAddAvg(3,3);
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fftCalc[ 1] = wc * 0.9f * fftAddAvg(4,4);
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fftCalc[ 2] = wc * fftAddAvg(5,5);
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fftCalc[ 3] = wc * fftAddAvg(6,6);
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// don't use the last bins from 206 to 255.
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fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
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} else {
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fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass
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fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass
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fftCalc[ 2] = wc * fftAddAvg(3,4); // 2 129 - 216 bass
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fftCalc[ 3] = wc * fftAddAvg(5,6); // 2 216 - 301 bass + midrange
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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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fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
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}
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fftCalc[ 4] = wc * fftAddAvg(7,9); // 3 301 - 430 midrange
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fftCalc[ 5] = wc * fftAddAvg(10,12); // 3 430 - 560 midrange
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fftCalc[ 6] = wc * fftAddAvg(13,18); // 5 560 - 818 midrange
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fftCalc[ 7] = wc * fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center !
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fftCalc[ 8] = wc * fftAddAvg(26,32); // 7 1120 - 1421 midrange
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fftCalc[ 9] = wc * fftAddAvg(33,43); // 9 1421 - 1895 midrange
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fftCalc[10] = wc * fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid
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fftCalc[11] = wc * fftAddAvg(56,69); // 14 2412 - 3015 high mid
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fftCalc[12] = wc * fftAddAvg(70,85); // 16 3015 - 3704 high mid
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fftCalc[13] = wc * fftAddAvg(86,103); // 18 3704 - 4479 high mid
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fftCalc[14] = wc * fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
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}
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else if (freqDist == 1) { //WLEDMM: Rightshift: note ewowi: frequencies in comments are not correct
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if (useInputFilter==1) {
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// skip frequencies below 100hz
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fftCalc[ 0] = wc * 0.8f * fftAddAvg(1,1);
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fftCalc[ 1] = wc * 0.9f * fftAddAvg(2,2);
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fftCalc[ 2] = wc * fftAddAvg(3,3);
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fftCalc[ 3] = wc * fftAddAvg(4,4);
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// don't use the last bins from 206 to 255.
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fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
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} else {
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fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass
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fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass
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fftCalc[ 2] = wc * fftAddAvg(3,3); // 2 129 - 216 bass
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fftCalc[ 3] = wc * fftAddAvg(4,4); // 2 216 - 301 bass + midrange
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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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fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
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}
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fftCalc[ 4] = wc * fftAddAvg(5,6); // 3 301 - 430 midrange
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fftCalc[ 5] = wc * fftAddAvg(7,8); // 3 430 - 560 midrange
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fftCalc[ 6] = wc * fftAddAvg(9,10); // 5 560 - 818 midrange
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fftCalc[ 7] = wc * fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center !
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fftCalc[ 8] = wc * fftAddAvg(14,18); // 7 1120 - 1421 midrange
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fftCalc[ 9] = wc * fftAddAvg(19,25); // 9 1421 - 1895 midrange
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fftCalc[10] = wc * fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid
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fftCalc[11] = wc * fftAddAvg(37,45); // 14 2412 - 3015 high mid
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fftCalc[12] = wc * fftAddAvg(46,66); // 16 3015 - 3704 high mid
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fftCalc[13] = wc * fftAddAvg(67,97); // 18 3704 - 4479 high mid
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fftCalc[14] = wc * fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
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}
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#endif
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} else { // noise gate closed - just decay old values
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isFirstRun = false;
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for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
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fftCalc[i] *= 0.85f; // decay to zero
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if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f;
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//WLEDMM: different distributions
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if (freqDist == 0) {
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/* new mapping, optimized for 22050 Hz by softhack007 --- update: removed overlap */
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// bins frequency range
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if (useInputFilter==1) {
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// skip frequencies below 100hz
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fftCalc[ 0] = wc * 0.8f * fftAddAvg(3,3);
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fftCalc[ 1] = wc * 0.9f * fftAddAvg(4,4);
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fftCalc[ 2] = wc * fftAddAvg(5,5);
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fftCalc[ 3] = wc * fftAddAvg(6,6);
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// don't use the last bins from 206 to 255.
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fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
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} else {
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fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass
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fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass
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fftCalc[ 2] = wc * fftAddAvg(3,4); // 2 129 - 216 bass
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fftCalc[ 3] = wc * fftAddAvg(5,6); // 2 216 - 301 bass + midrange
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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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fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
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}
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fftCalc[ 4] = wc * fftAddAvg(7,9); // 3 301 - 430 midrange
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fftCalc[ 5] = wc * fftAddAvg(10,12); // 3 430 - 560 midrange
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fftCalc[ 6] = wc * fftAddAvg(13,18); // 5 560 - 818 midrange
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fftCalc[ 7] = wc * fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center !
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fftCalc[ 8] = wc * fftAddAvg(26,32); // 7 1120 - 1421 midrange
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fftCalc[ 9] = wc * fftAddAvg(33,43); // 9 1421 - 1895 midrange
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fftCalc[10] = wc * fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid
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fftCalc[11] = wc * fftAddAvg(56,69); // 14 2412 - 3015 high mid
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fftCalc[12] = wc * fftAddAvg(70,85); // 16 3015 - 3704 high mid
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fftCalc[13] = wc * fftAddAvg(86,103); // 18 3704 - 4479 high mid
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fftCalc[14] = wc * fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
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} else if (freqDist == 1) { //WLEDMM: Rightshift: note ewowi: frequencies in comments are not correct
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if (useInputFilter==1) {
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// skip frequencies below 100hz
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fftCalc[ 0] = wc * 0.8f * fftAddAvg(1,1);
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fftCalc[ 1] = wc * 0.9f * fftAddAvg(2,2);
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fftCalc[ 2] = wc * fftAddAvg(3,3);
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fftCalc[ 3] = wc * fftAddAvg(4,4);
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// don't use the last bins from 206 to 255.
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fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
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} else {
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fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass
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fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass
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fftCalc[ 2] = wc * fftAddAvg(3,3); // 2 129 - 216 bass
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fftCalc[ 3] = wc * fftAddAvg(4,4); // 2 216 - 301 bass + midrange
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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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fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
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}
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fftCalc[ 4] = wc * fftAddAvg(5,6); // 3 301 - 430 midrange
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fftCalc[ 5] = wc * fftAddAvg(7,8); // 3 430 - 560 midrange
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fftCalc[ 6] = wc * fftAddAvg(9,10); // 5 560 - 818 midrange
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fftCalc[ 7] = wc * fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center !
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fftCalc[ 8] = wc * fftAddAvg(14,18); // 7 1120 - 1421 midrange
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fftCalc[ 9] = wc * fftAddAvg(19,25); // 9 1421 - 1895 midrange
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fftCalc[10] = wc * fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid
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fftCalc[11] = wc * fftAddAvg(37,45); // 14 2412 - 3015 high mid
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fftCalc[12] = wc * fftAddAvg(46,66); // 16 3015 - 3704 high mid
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fftCalc[13] = wc * fftAddAvg(67,97); // 18 3704 - 4479 high mid
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fftCalc[14] = wc * fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
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}
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} else { // noise gate closed - just decay old values
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isFirstRun = false;
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for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
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fftCalc[i] *= 0.85f; // decay to zero
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if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f;
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} }
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memcpy(lastFftCalc, fftCalc, sizeof(lastFftCalc)); // make a backup of last "good" channels
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@@ -1211,7 +1180,6 @@ class AudioReactive : public Usermod {
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double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error
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// variables used by getSample() and agcAvg()
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int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed
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double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controller.
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double micLev = 0.0; // Used to convert returned value to have '0' as minimum. A leveller
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float expAdjF = 0.0f; // Used for exponential filter.
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@@ -1268,7 +1236,6 @@ class AudioReactive : public Usermod {
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//PLOT_PRINT("sampleAgc:"); PLOT_PRINT(sampleAgc); PLOT_PRINT("\t");
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//PLOT_PRINT("sampleAvg:"); PLOT_PRINT(sampleAvg); PLOT_PRINT("\t");
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//PLOT_PRINT("sampleReal:"); PLOT_PRINT(sampleReal); PLOT_PRINT("\t");
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//PLOT_PRINT("micIn:"); PLOT_PRINT(micIn); PLOT_PRINT("\t");
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//PLOT_PRINT("sample:"); PLOT_PRINT(sample); PLOT_PRINT("\t");
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//PLOT_PRINT("sampleMax:"); PLOT_PRINT(sampleMax); PLOT_PRINT("\t");
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//PLOT_PRINT("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t");
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@@ -1418,6 +1385,15 @@ class AudioReactive : public Usermod {
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// update global vars ONCE - multAgc, sampleAGC, rawSampleAgc
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multAgc = multAgcTemp;
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if (micQuality > 0) {
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if (micQuality > 1) {
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rawSampleAgc = 0.95f * tmpAgc + 0.05f * (float)rawSampleAgc; // raw path
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sampleAgc += 0.95f * (tmpAgc - sampleAgc); // smooth path
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} else {
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rawSampleAgc = 0.70f * tmpAgc + 0.30f * (float)rawSampleAgc; // min filtering path
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sampleAgc += 0.70f * (tmpAgc - sampleAgc);
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}
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} else {
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#if defined(WLEDMM_FASTPATH)
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rawSampleAgc = 0.65f * tmpAgc + 0.35f * (float)rawSampleAgc;
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#else
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@@ -1428,7 +1404,7 @@ class AudioReactive : public Usermod {
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sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero
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else
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sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
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}
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sampleAgc = fabsf(sampleAgc); // // make sure we have a positive value
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last_soundAgc = soundAgc;
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} // agcAvg()
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@@ -1446,27 +1422,6 @@ class AudioReactive : public Usermod {
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static unsigned long lastSoundTime = 0; // for delaying un-freeze
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static unsigned long startuptime = 0; // "fast freeze" mode: do not interfere during first 12 seconds (filter startup time)
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#ifdef WLED_DISABLE_SOUND
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micIn = inoise8(millis(), millis()); // Simulated analog read
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micDataReal = micIn;
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#else
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#ifdef ARDUINO_ARCH_ESP32
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micIn = int(micDataReal); // micDataSm = ((micData * 3) + micData)/4;
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#else
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// this is the minimal code for reading analog mic input on 8266.
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// warning!! Absolutely experimental code. Audio on 8266 is still not working. Expects a million follow-on problems.
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static unsigned long lastAnalogTime = 0;
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static float lastAnalogValue = 0.0f;
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if (millis() - lastAnalogTime > 20) {
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micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only.
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lastAnalogTime = millis();
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lastAnalogValue = micDataReal;
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yield();
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} else micDataReal = lastAnalogValue;
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micIn = int(micDataReal);
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#endif
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#endif
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if (startuptime == 0) startuptime = millis(); // fast freeze mode - remember filter startup time
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if ((micLevelMethod < 1) || !isFrozen) { // following the input level, UNLESS mic Level was frozen
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micLev += (micDataReal-micLev) / 12288.0f;
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@@ -1477,7 +1432,6 @@ class AudioReactive : public Usermod {
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if (!haveSilence) isFrozen = true; // freeze mode: freeze micLevel so it cannot rise again
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}
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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 = fabsf(micDataReal - micLev);
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@@ -1511,10 +1465,15 @@ class AudioReactive : public Usermod {
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if ((micLevelMethod == 2) && (millis() - startuptime < 12000)) isFrozen = false; // fast freeze: no freeze in first 12 seconds (filter startup phase)
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tmpSample = expAdjF;
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micIn = abs(micIn); // And get the absolute value of each sample
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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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// Adjust the gain. with inputLevel adjustment.
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if (micQuality > 0) {
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sampleAdj = micInNoDC * sampleGain / 40.0f * inputLevel/128.0f + micInNoDC / 16.0f; // ... using unfiltered sample
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sampleReal = micInNoDC;
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} else {
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sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // ... using pre-filtered sample
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sampleReal = tmpSample;
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}
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|
||||
sampleAdj = fmax(fmin(sampleAdj, 255.0f), 0.0f); // Question: why are we limiting the value to 8 bits ???
|
||||
sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
|
||||
@@ -1538,11 +1497,16 @@ class AudioReactive : public Usermod {
|
||||
}
|
||||
if (sampleMax < 0.5f) sampleMax = 0.0f;
|
||||
|
||||
if (micQuality > 0) {
|
||||
if (micQuality > 1) sampleAvg += 0.95f * (sampleAdj - sampleAvg);
|
||||
else sampleAvg += 0.70f * (sampleAdj - sampleAvg);
|
||||
} else {
|
||||
#if defined(WLEDMM_FASTPATH)
|
||||
sampleAvg = ((sampleAvg * 11.0f) + sampleAdj) / 12.0f; // make reactions a bit more "crisp" in fastpath mode
|
||||
#else
|
||||
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
|
||||
#endif
|
||||
}
|
||||
sampleAvg = fabsf(sampleAvg); // make sure we have a positive value
|
||||
} // getSample()
|
||||
|
||||
@@ -2784,6 +2748,7 @@ class AudioReactive : public Usermod {
|
||||
//WLEDMM: experimental settings
|
||||
JsonObject poweruser = top.createNestedObject("experiments");
|
||||
poweruser[F("micLev")] = micLevelMethod;
|
||||
poweruser[F("Mic_Quality")] = micQuality;
|
||||
poweruser[F("freqDist")] = freqDist;
|
||||
//poweruser[F("freqRMS")] = averageByRMS;
|
||||
poweruser[F("FFT_Window")] = fftWindow;
|
||||
@@ -2859,6 +2824,7 @@ class AudioReactive : public Usermod {
|
||||
|
||||
//WLEDMM: experimental settings
|
||||
configComplete &= getJsonValue(top["experiments"][F("micLev")], micLevelMethod);
|
||||
configComplete &= getJsonValue(top["experiments"][F("Mic_Quality")], micQuality);
|
||||
configComplete &= getJsonValue(top["experiments"][F("freqDist")], freqDist);
|
||||
//configComplete &= getJsonValue(top["experiments"][F("freqRMS")], averageByRMS);
|
||||
configComplete &= getJsonValue(top["experiments"][F("FFT_Window")], fftWindow);
|
||||
@@ -2959,23 +2925,29 @@ class AudioReactive : public Usermod {
|
||||
oappend(SET_F("addOption(dd,'Lazy',3);"));
|
||||
|
||||
//WLEDMM: experimental settings
|
||||
oappend(SET_F("dd=addDropdown(ux,'experiments:micLev');"));
|
||||
oappend(SET_F("xx='experiments';")); // shortcut
|
||||
oappend(SET_F("dd=addDropdown(ux,xx+':micLev');"));
|
||||
oappend(SET_F("addOption(dd,'Floating (⎌)',0);"));
|
||||
oappend(SET_F("addOption(dd,'Freeze',1);"));
|
||||
oappend(SET_F("addOption(dd,'Fast Freeze',2);"));
|
||||
oappend(SET_F("addInfo(ux+':experiments:micLev',1,'☾');"));
|
||||
oappend(SET_F("addInfo(ux+':'+xx+':micLev',1,'☾');"));
|
||||
|
||||
oappend(SET_F("dd=addDropdown(ux,'experiments:freqDist');"));
|
||||
oappend(SET_F("dd=addDropdown(ux,xx+':Mic_Quality');"));
|
||||
oappend(SET_F("addOption(dd,'average (standard)',0);"));
|
||||
oappend(SET_F("addOption(dd,'low noise',1);"));
|
||||
oappend(SET_F("addOption(dd,'perfect',2);"));
|
||||
|
||||
oappend(SET_F("dd=addDropdown(ux,xx+':freqDist');"));
|
||||
oappend(SET_F("addOption(dd,'Normal (⎌)',0);"));
|
||||
oappend(SET_F("addOption(dd,'RightShift',1);"));
|
||||
oappend(SET_F("addInfo(ux+':experiments:freqDist',1,'☾');"));
|
||||
oappend(SET_F("addInfo(ux+':'+xx+':freqDist',1,'☾');"));
|
||||
|
||||
//oappend(SET_F("dd=addDropdown(ux,'experiments:freqRMS');"));
|
||||
//oappend(SET_F("dd=addDropdown(ux,xx+':freqRMS');"));
|
||||
//oappend(SET_F("addOption(dd,'Off (⎌)',0);"));
|
||||
//oappend(SET_F("addOption(dd,'On',1);"));
|
||||
//oappend(SET_F("addInfo(ux+':experiments:freqRMS',1,'☾');"));
|
||||
|
||||
oappend(SET_F("dd=addDropdown(ux,'experiments:FFT_Window');"));
|
||||
oappend(SET_F("dd=addDropdown(ux,xx+':FFT_Window');"));
|
||||
oappend(SET_F("addOption(dd,'Blackman-Harris (MM standard)',0);"));
|
||||
oappend(SET_F("addOption(dd,'Hann (balanced)',1);"));
|
||||
oappend(SET_F("addOption(dd,'Nuttall (more accurate)',2);"));
|
||||
@@ -2984,10 +2956,10 @@ class AudioReactive : public Usermod {
|
||||
oappend(SET_F("addOption(dd,'Flat-Top (AC WLED, inaccurate)',4);"));
|
||||
|
||||
#ifdef FFT_USE_SLIDING_WINDOW
|
||||
oappend(SET_F("dd=addDropdown(ux,'experiments:I2S_FastPath');"));
|
||||
oappend(SET_F("dd=addDropdown(ux,xx+':I2S_FastPath');"));
|
||||
oappend(SET_F("addOption(dd,'Off',0);"));
|
||||
oappend(SET_F("addOption(dd,'On (⎌)',1);"));
|
||||
oappend(SET_F("addInfo(ux+':experiments:I2S_FastPath',1,'☾');"));
|
||||
oappend(SET_F("addInfo(ux+':'+xx+':I2S_FastPath',1,'☾');"));
|
||||
#endif
|
||||
|
||||
oappend(SET_F("dd=addDropdown(ux,'dynamics:limiter');"));
|
||||
|
||||
Reference in New Issue
Block a user