removed broken frequency squelch, added frequency scaling options
- removed broken FFTResult "squelch" feature. It was completely broken, and caused flashes in GEQ. - added Frequency scaling options: linear and logarithmic - fixed a few numerical accidents in FX.cpp (bouncing_balls, ripplepeak, freqmap, gravfreq, waterfall)
This commit is contained in:
Frank
parent
d92a93f1d5
commit
3e494cc551
@ -64,6 +64,8 @@ static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1
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static bool limiterOn = true; // bool: enable / disable dynamics limiter
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static uint16_t attackTime = 80; // int: attack time in milliseconds. Default 0.08sec
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static uint16_t decayTime = 1400; // int: decay time in milliseconds. Default 1.40sec
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// user settable options for FFTResult scaling
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static uint8_t FFTScalingMode = 2; // 0 none; 1 optimized logarithmic; 2 optimized linear
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//
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// AGC presets
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@ -88,8 +90,14 @@ static AudioSource *audioSource = nullptr;
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static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks.
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static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point
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static float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC.
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static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
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static int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel)
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static int16_t rawSampleAgc = 0; // not smoothed AGC sample
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static float sampleAvg = 0.0f; // Smoothed Average sampleRaw
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static float sampleAgc = 0.0f; // Smoothed AGC sample
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////////////////////
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// Begin FFT Code //
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////////////////////
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@ -113,6 +121,9 @@ static float vReal[samplesFFT] = {0.0f};
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static float vImag[samplesFFT] = {0.0f};
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static float fftBin[samplesFFT_2] = {0.0f};
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#define FFT_DOWNSCALE 0.65f // downscaling factor for FFT results - "Flat-Top" window
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#define LOG_256 5.54517744
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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static float windowWeighingFactors[samplesFFT] = {0.0f};
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#endif
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@ -131,9 +142,6 @@ static unsigned long fftTime = 0;
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static unsigned long sampleTime = 0;
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#endif
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// Table of linearNoise results to be multiplied by soundSquelch in order to reduce squelch across fftResult bins.
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static uint8_t linearNoise[16] = { 34, 28, 26, 25, 20, 12, 9, 6, 4, 4, 3, 2, 2, 2, 2, 2 };
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// Table of multiplication factors so that we can even out the frequency response.
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static float fftResultPink[16] = { 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f };
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@ -146,6 +154,11 @@ static arduinoFFT FFT = arduinoFFT(vReal, vImag, samplesFFT, SAMPLE_RATE);
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static TaskHandle_t FFT_Task = nullptr;
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// float version of map()
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static float mapf(float x, float in_min, float in_max, float out_min, float out_max){
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return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
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}
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static float fftAddAvg(int from, int to) {
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float result = 0.0f;
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for (int i = from; i <= to; i++) {
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@ -213,7 +226,8 @@ void FFTcode(void * parameter)
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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FFT.dcRemoval(); // remove DC offset
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FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data
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FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy
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//FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection
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FFT.compute( FFTDirection::Forward ); // Compute FFT
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FFT.complexToMagnitude(); // Compute magnitudes
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#else
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@ -253,44 +267,84 @@ void FFTcode(void * parameter)
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* Multiplier = (End frequency/ Start frequency) ^ 1/16
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* Multiplier = 1.320367784
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*/
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if (sampleAvg > 1) { // noise gate open
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// Range
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fftCalc[ 0] = fftAddAvg(3,4); // 60 - 100
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fftCalc[ 1] = fftAddAvg(4,5); // 80 - 120
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fftCalc[ 2] = fftAddAvg(5,7); // 100 - 160
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fftCalc[ 3] = fftAddAvg(7,9); // 140 - 200
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fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
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fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
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fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
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fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
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fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
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fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
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fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
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fftCalc[11] = fftAddAvg(64,84); // 1280 - 1700
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fftCalc[12] = fftAddAvg(84,111); // 1680 - 2240
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fftCalc[13] = fftAddAvg(111,147); // 2220 - 2960
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fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900
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fftCalc[15] = fftAddAvg(194,255); // 3880 - 5120
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fftCalc[ 0] = fftAddAvg(2,4); // 60 - 100
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fftCalc[ 1] = fftAddAvg(4,5); // 80 - 120
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fftCalc[ 2] = fftAddAvg(5,7); // 100 - 160
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fftCalc[ 3] = fftAddAvg(7,9); // 140 - 200
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fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
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fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
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fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
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fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
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fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
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fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
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fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
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fftCalc[11] = fftAddAvg(64,84); // 1280 - 1700
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fftCalc[12] = fftAddAvg(84,111); // 1680 - 2240
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fftCalc[13] = fftAddAvg(111,147); // 2220 - 2960
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fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900
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fftCalc[15] = fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate
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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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if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f;
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}
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}
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for (int i=0; i < 16; i++) {
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// Noise supression of fftCalc bins using soundSquelch adjustment for different input types.
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fftCalc[i] = (fftCalc[i] < ((float)soundSquelch * (float)linearNoise[i] / 4.0f)) ? 0 : fftCalc[i];
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// Adjustment for frequency curves.
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fftCalc[i] *= fftResultPink[i];
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// Manual linear adjustment of gain using sampleGain adjustment for different input types.
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fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //with inputLevel adjustment
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if (sampleAvg > 1) { // noise gate open
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// Adjustment for frequency curves.
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fftCalc[i] *= fftResultPink[i];
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if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
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// Manual linear adjustment of gain using sampleGain adjustment for different input types.
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fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //with inputLevel adjustment
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if(fftCalc[i] < 0) fftCalc[i] = 0;
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}
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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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else // fall slow
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fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i]; // will need approx 5 cycles (150ms) for converging against fftCalc[i]
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//fftAvg[i] = fftCalc[i]*0.05f + 0.95f*fftAvg[i]; // will need approx 10 cycles (250ms) for converging against fftCalc[i]
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fftAvg[i] = fftCalc[i]*0.17f + 0.83f*fftAvg[i]; // will need approx 5 cycles (150ms) for converging against fftCalc[i]
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// constrain internal vars - just to be sure
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fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f);
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fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f);
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float currentResult;
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if(limiterOn == true)
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currentResult = fftAvg[i];
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else
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currentResult = fftCalc[i];
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if (FFTScalingMode > 0) {
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if (FFTScalingMode == 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 = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255]
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} else {
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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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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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}
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} else {
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currentResult -= 4; // just a bit more room for peaks
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}
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// Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely.
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if(limiterOn == true)
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fftResult[i] = constrain((int)fftAvg[i], 0, 254);
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else
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fftResult[i] = constrain((int)fftCalc[i], 0, 254);
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fftResult[i] = constrain((int)currentResult, 0, 255);
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}
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#ifdef WLED_DEBUG
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@ -396,12 +450,7 @@ class AudioReactive : public Usermod {
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bool udpSamplePeak = 0; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
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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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int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel; smoothed over 16 samples)
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double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler.
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float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC.
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float sampleAvg = 0.0f; // Smoothed Average sampleRaw
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float sampleAgc = 0.0f; // Our AGC sample
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int16_t rawSampleAgc = 0; // Our AGC sample - raw
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uint32_t timeOfPeak = 0;
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unsigned long lastTime = 0; // last time of running UDP Microphone Sync
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float micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller
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@ -941,7 +990,7 @@ class AudioReactive : public Usermod {
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return;
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}
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// We cannot wait indefinitely before processing audio data
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if (strip.isUpdating() && (millis() - lastUMRun < 2)) return; // be nice, but not too nice
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if (strip.isUpdating() && (millis() - lastUMRun < 1)) return; // be nice, but not too nice
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// suspend local sound processing when "real time mode" is active (E131, UDP, ADALIGHT, ARTNET)
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if ( (realtimeOverride == REALTIME_OVERRIDE_NONE) // please odd other orrides here if needed
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@ -1115,6 +1164,11 @@ class AudioReactive : public Usermod {
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sampleRaw = 0; rawSampleAgc = 0;
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my_magnitude = 0; FFT_Magnitude = 0; FFT_MajorPeak = 0;
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multAgc = 1;
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// reset FFT data
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memset(fftCalc, 0, sizeof(fftCalc));
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memset(fftAvg, 0, sizeof(fftAvg));
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memset(fftResult, 0, sizeof(fftResult));
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for(int i=(init?0:1); i<16; i+=2) fftResult[i] = 16; // make a tiny pattern
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if (init && FFT_Task) {
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vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash
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@ -1372,6 +1426,9 @@ class AudioReactive : public Usermod {
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dynLim[F("Rise")] = attackTime;
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dynLim[F("Fall")] = decayTime;
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JsonObject freqScale = top.createNestedObject("Frequency");
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freqScale[F("Scale")] = FFTScalingMode;
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JsonObject sync = top.createNestedObject("sync");
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sync[F("port")] = audioSyncPort;
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sync[F("mode")] = audioSyncEnabled;
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@ -1418,6 +1475,8 @@ class AudioReactive : public Usermod {
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configComplete &= getJsonValue(top["dynamics"][F("Rise")], attackTime);
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configComplete &= getJsonValue(top["dynamics"][F("Fall")], decayTime);
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configComplete &= getJsonValue(top["Frequency"][F("Scale")], FFTScalingMode);
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configComplete &= getJsonValue(top["sync"][F("port")], audioSyncPort);
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configComplete &= getJsonValue(top["sync"][F("mode")], audioSyncEnabled);
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@ -1443,11 +1502,14 @@ class AudioReactive : public Usermod {
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oappend(SET_F("dd=addDropdown('AudioReactive','dynamics:Limiter');"));
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oappend(SET_F("addOption(dd,'Off',0);"));
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oappend(SET_F("addOption(dd,'On',1);"));
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oappend(SET_F("addInfo('AudioReactive:dynamics:Limiter',0,' Limiter On ');")); // 0 is field type, 1 is actual field
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//oappend(SET_F("addInfo('AudioReactive:dynamics:Rise',0,'min. ');"));
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oappend(SET_F("addInfo('AudioReactive:dynamics:Rise',1,' ms <br /><i>(volume reactive FX only)</i>');"));
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//oappend(SET_F("addInfo('AudioReactive:dynamics:Fall',0,'min. ');"));
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oappend(SET_F("addInfo('AudioReactive:dynamics:Fall',1,' ms <br /><i>(volume reactive FX only)</i>');"));
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oappend(SET_F("addInfo('AudioReactive:dynamics:Limiter',0,' On ');")); // 0 is field type, 1 is actual field
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oappend(SET_F("addInfo('AudioReactive:dynamics:Rise',1,'ms <i>(♪ effects only)</i>');"));
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oappend(SET_F("addInfo('AudioReactive:dynamics:Fall',1,'ms <i>(♪ effects only)</i>');"));
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oappend(SET_F("dd=addDropdown('AudioReactive','Frequency:Scale');"));
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oappend(SET_F("addOption(dd,'None',0);"));
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oappend(SET_F("addOption(dd,'Linear (Amplitude)',2);"));
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oappend(SET_F("addOption(dd,'Logarithmic (Loudness)',1);"));
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oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');"));
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oappend(SET_F("addOption(dd,'Off',0);"));
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@ -2842,12 +2842,13 @@ uint16_t mode_bouncing_balls(void) {
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for (size_t i = 0; i < numBalls; i++) {
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float timeSinceLastBounce = (time - balls[i].lastBounceTime)/((255-SEGMENT.speed)*8/256 +1);
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balls[i].height = 0.5 * gravity * pow(timeSinceLastBounce/1000 , 2.0) + balls[i].impactVelocity * timeSinceLastBounce/1000;
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float timeSec = timeSinceLastBounce/1000.0f;
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balls[i].height = 0.5 * gravity * (timeSec * timeSec) + balls[i].impactVelocity * timeSec; // avoid use pow(x, 2) - its extremely slow !
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if (balls[i].height < 0) { //start bounce
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balls[i].height = 0;
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//damping for better effect using multiple balls
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float dampening = 0.90 - float(i)/pow(numBalls,2);
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float dampening = 0.90 - float(i)/(float(numBalls) * float(numBalls)); // avoid use pow(x, 2) - its extremely slow !
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balls[i].impactVelocity = dampening * balls[i].impactVelocity;
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balls[i].lastBounceTime = time;
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@ -2863,7 +2864,7 @@ uint16_t mode_bouncing_balls(void) {
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color = SEGCOLOR(i % NUM_COLORS);
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}
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uint16_t pos = round(balls[i].height * (SEGLEN - 1));
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uint16_t pos = roundf(balls[i].height * (SEGLEN - 1));
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SEGMENT.setPixelColor(pos, color);
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}
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@ -6006,7 +6007,9 @@ uint16_t mode_ripplepeak(void) { // * Ripple peak. By Andrew Tuli
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case 255: // Initialize ripple variables.
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ripples[i].pos = random16(SEGLEN);
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#ifdef ESP32
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if (FFT_MajorPeak > 1) // log10(0) is "forbidden" (throws exception)
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ripples[i].color = (int)(log10f(FFT_MajorPeak)*128);
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else ripples[i].color = 0;
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#else
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ripples[i].color = random8();
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#endif
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@ -6716,6 +6719,7 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN.
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}
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float FFT_MajorPeak = *(float*) um_data->u_data[4];
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float my_magnitude = *(float*) um_data->u_data[5] / 4.0f;
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if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
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SEGMENT.fade_out(SEGMENT.speed);
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@ -6806,6 +6810,7 @@ uint16_t mode_freqpixels(void) { // Freqpixel. By Andrew Tuline.
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}
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float FFT_MajorPeak = *(float*) um_data->u_data[4];
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float my_magnitude = *(float*) um_data->u_data[5] / 16.0f;
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if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
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uint16_t fadeRate = 2*SEGMENT.speed - SEGMENT.speed*SEGMENT.speed/255; // Get to 255 as quick as you can.
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SEGMENT.fade_out(fadeRate);
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@ -6908,6 +6913,7 @@ uint16_t mode_gravfreq(void) { // Gravfreq. By Andrew Tuline.
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}
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float FFT_MajorPeak = *(float*) um_data->u_data[4];
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float volumeSmth = *(float*) um_data->u_data[0];
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if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
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SEGMENT.fade_out(240);
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@ -7022,6 +7028,8 @@ uint16_t mode_waterfall(void) { // Waterfall. By: Andrew Tulin
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uint8_t *binNum = (uint8_t*)um_data->u_data[7];
|
||||
float my_magnitude = *(float*) um_data->u_data[5] / 8.0f;
|
||||
|
||||
if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
|
||||
|
||||
if (SEGENV.call == 0) {
|
||||
SEGMENT.setUpLeds();
|
||||
SEGMENT.fill(BLACK);
|
||||
|
||||
Loading…
Reference in New Issue
Block a user