kopia lustrzana https://github.com/Aircoookie/WLED
Merge branch 'audioreactive-prototype' into dev
Blaz Kristan
commit
cf51892782
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@ -24,7 +24,7 @@
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// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
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// #define FFT_SAMPLING_LOG // FFT result debugging
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// #define SR_DEBUG // generic SR DEBUG messages
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// #define NO_MIC_LOGGER // exclude MIC_LOGGER from SR_DEBUG
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#ifdef SR_DEBUG
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#define DEBUGSR_PRINT(x) Serial.print(x)
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@ -85,18 +85,25 @@ const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; //
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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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// audioreactive variables shared with FFT task
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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 float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate)
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// peak detection
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static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
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static uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
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static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
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static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
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static unsigned long timeOfPeak = 0; // time of last sample peak detection.
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static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[])
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static void autoResetPeak(void); // peak auto-reset function
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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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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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// lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
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#define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and an a few other speedups
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@ -105,21 +112,22 @@ static float sampleAgc = 0.0f; // Smoothed AGC sample
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#endif
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#include "arduinoFFT.h"
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// FFT Variables
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// FFT Output variables shared with animations
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#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !!
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static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency
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static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency
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static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects
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// FFT Constants
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constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2
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constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
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static float FFT_MajorPeak = 1.0f;
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static float FFT_Magnitude = 0.0f;
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// These are the input and output vectors. Input vectors receive computed results from FFT.
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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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static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
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static float vImag[samplesFFT] = {0.0f}; // imaginary parts
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// the following are observed values, supported by a bit of "educated guessing"
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//#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels
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#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
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#define LOG_256 5.54517744
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@ -128,13 +136,11 @@ static float windowWeighingFactors[samplesFFT] = {0.0f};
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#endif
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// Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256.
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// Oh, and bins 0,1,2 are no good, so we'll zero them out.
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static float fftCalc[16] = {0.0f};
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static uint8_t fftResult[16] = {0}; // Our calculated result table, which we feed to the animations.
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static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f};
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static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON)
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#ifdef SR_DEBUG
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static float fftResultMax[16] = {0.0f}; // A table used for testing to determine how our post-processing is working.
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static float fftResultMax[NUM_GEQ_CHANNELS] = {0.0f}; // A table used for testing to determine how our post-processing is working.
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#endif
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static float fftAvg[16] = {0.0f};
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#ifdef WLED_DEBUG
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static unsigned long fftTime = 0;
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@ -142,7 +148,7 @@ static unsigned long sampleTime = 0;
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#endif
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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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static float fftResultPink[NUM_GEQ_CHANNELS] = { 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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// Create FFT object
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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@ -161,12 +167,12 @@ static float mapf(float x, float in_min, float in_max, float out_min, float out_
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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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result += fftBin[i];
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result += vReal[i];
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}
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return result / float(to - from + 1);
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}
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// FFT main code
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// FFT main task
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void FFTcode(void * parameter)
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{
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DEBUGSR_PRINT("FFT started on core: "); DEBUGSR_PRINTLN(xPortGetCoreID());
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@ -237,9 +243,9 @@ 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; 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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fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max.
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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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} // for()
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// mapping of FFT result bins to frequency channels
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@ -292,14 +298,14 @@ void FFTcode(void * parameter)
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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 - just decay old values
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for (int i=0; i < 16; i++) {
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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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}
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// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
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for (int i=0; i < 16; i++) {
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for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
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if (sampleAvg > 1) { // noise gate open
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// Adjustment for frequency curves.
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@ -378,11 +384,43 @@ void FFTcode(void * parameter)
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fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
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}
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#endif
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// run peak detection
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autoResetPeak();
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detectSamplePeak();
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} // for(;;)
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} // FFTcode()
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} // for(;;)ever
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} // FFTcode() task end
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////////////////////
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// Peak detection //
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////////////////////
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// peak detection is called from FFT task when vReal[] contains valid FFT results
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static void detectSamplePeak(void) {
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// Poor man's beat detection by seeing if sample > Average + some value.
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if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 1) && (vReal[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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}
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}
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static void autoResetPeak(void) {
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uint16_t MinShowDelay = MAX(50, strip.getMinShowDelay()); // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
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if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
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samplePeak = false;
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if (audioSyncEnabled == 0) udpSamplePeak = false; // this is normally reset by transmitAudioData
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}
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}
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////////////////////
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// usermod class //
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////////////////////
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//class name. Use something descriptive and leave the ": public Usermod" part :)
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class AudioReactive : public Usermod {
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@ -453,40 +491,36 @@ class AudioReactive : public Usermod {
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double FFT_MajorPeak; // 08 Bytes
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};
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WiFiUDP fftUdp;
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// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
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bool enabled = false;
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bool initDone = false;
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const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED
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// variables for UDP sound sync
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WiFiUDP fftUdp; // UDP object for sound sync (from WiFi UDP, not Async UDP!)
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bool udpSyncConnected = false;// UDP connection status -> true if connected to multicast group
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unsigned long lastTime = 0; // last time of running UDP Microphone Sync
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const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED
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uint16_t audioSyncPort= 11988;// default port for UDP sound sync
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// used for AGC
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int last_soundAgc = -1; // used to detect AGC mode change (for resetting AGC internal error buffers)
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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 controler.
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float micLev = 0.0f; // 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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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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int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel)
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int16_t rawSampleAgc = 0; // not smoothed AGC sample
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float sampleAgc = 0.0f; // Smoothed AGC sample
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// variables used in effects
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uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
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uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
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bool samplePeak = 0; // Boolean flag for peak. Responding routine must reset this flag
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float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
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int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc
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float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc
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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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double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler.
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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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float expAdjF = 0.0f; // Used for exponential filter.
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bool udpSyncConnected = false;
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uint16_t audioSyncPort = 11988;
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// used for AGC
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uint8_t lastMode = 0; // last known effect mode
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int last_soundAgc = -1;
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double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error
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unsigned long last_update_time = 0;
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unsigned long last_kick_time = 0;
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uint8_t last_user_inputLevel = 0;
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// used to feed "Info" Page
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unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket
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float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds
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@ -503,6 +537,10 @@ class AudioReactive : public Usermod {
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static const char UDP_SYNC_HEADER_v1[];
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// private methods
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////////////////////
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// Debug support //
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////////////////////
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void logAudio()
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{
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#ifdef MIC_LOGGER
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@ -525,7 +563,7 @@ class AudioReactive : public Usermod {
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#ifdef FFT_SAMPLING_LOG
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#if 0
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for(int i=0; i<16; i++) {
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for(int i=0; i<NUM_GEQ_CHANNELS; i++) {
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Serial.print(fftResult[i]);
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Serial.print("\t");
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}
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@ -551,11 +589,11 @@ class AudioReactive : public Usermod {
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int maxVal = minimumMaxVal;
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int minVal = 0;
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for(int i = 0; i < 16; i++) {
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for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
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if(fftResult[i] > maxVal) maxVal = fftResult[i];
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if(fftResult[i] < minVal) minVal = fftResult[i];
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}
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for(int i = 0; i < 16; i++) {
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for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
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Serial.print(i); Serial.print(":");
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Serial.printf("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1));
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}
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@ -574,6 +612,10 @@ class AudioReactive : public Usermod {
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} // logAudio()
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//////////////////////
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// Audio Processing //
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//////////////////////
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/*
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* A "PI controller" multiplier to automatically adjust sound sensitivity.
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*
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@ -668,7 +710,7 @@ class AudioReactive : public Usermod {
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last_soundAgc = soundAgc;
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} // agcAvg()
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// post-processing and filtering of MIC sample (micDataReal) from FFTcode()
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void getSample()
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{
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float sampleAdj; // Gain adjusted sample value
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@ -729,24 +771,6 @@ class AudioReactive : public Usermod {
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if (sampleMax < 0.5f) sampleMax = 0.0f;
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sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
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// Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
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uint16_t MinShowDelay = strip.getMinShowDelay();
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if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
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samplePeak = false;
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udpSamplePeak = false;
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}
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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 ((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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}
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} // getSample()
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@ -781,6 +805,26 @@ class AudioReactive : public Usermod {
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}
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//////////////////////
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// UDP Sound Sync //
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//////////////////////
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// try to establish UDP sound sync connection
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void connectUDPSoundSync(void) {
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// This function tries to establish a UDP sync connection if needed
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// necessary as we also want to transmit in "AP Mode", but the standard "connected()" callback only reacts on STA connection
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static unsigned long last_connection_attempt = 0;
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if ((audioSyncPort <= 0) || ((audioSyncEnabled & 0x03) == 0)) return; // Sound Sync not enabled
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if (udpSyncConnected) return; // already connected
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if (!(apActive || interfacesInited)) return; // neither AP nor other connections availeable
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if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds
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// if we arrive here, we need a UDP connection but don't have one
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last_connection_attempt = millis();
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connected(); // try to start UDP
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}
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void transmitAudioData()
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{
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if (!udpSyncConnected) return;
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@ -795,7 +839,7 @@ class AudioReactive : public Usermod {
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udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it
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transmitData.reserved1 = 0;
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for (int i = 0; i < 16; i++) {
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for (int i = 0; i < NUM_GEQ_CHANNELS; i++) {
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transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254);
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}
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@ -808,12 +852,10 @@ class AudioReactive : public Usermod {
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return;
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} // transmitAudioData()
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|
||||
static bool isValidUdpSyncVersion(const char *header) {
|
||||
return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0;
|
||||
}
|
||||
|
||||
|
||||
bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received.
|
||||
{
|
||||
if (!udpSyncConnected) return false;
|
||||
|
|
@ -839,13 +881,7 @@ class AudioReactive : public Usermod {
|
|||
sampleAgc = volumeSmth;
|
||||
multAgc = 1.0f;
|
||||
|
||||
// auto-reset sample peak. Need to do it here, because getSample() is not running
|
||||
uint16_t MinShowDelay = strip.getMinShowDelay();
|
||||
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
|
||||
samplePeak = false;
|
||||
udpSamplePeak = false;
|
||||
}
|
||||
//if (userVar1 == 0) samplePeak = 0;
|
||||
autoResetPeak();
|
||||
// Only change samplePeak IF it's currently false.
|
||||
// If it's true already, then the animation still needs to respond.
|
||||
if (!samplePeak) {
|
||||
|
|
@ -855,7 +891,7 @@ class AudioReactive : public Usermod {
|
|||
}
|
||||
|
||||
//These values are only available on the ESP32
|
||||
for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i];
|
||||
for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i];
|
||||
|
||||
my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f);
|
||||
FFT_Magnitude = my_magnitude;
|
||||
|
|
@ -869,6 +905,10 @@ class AudioReactive : public Usermod {
|
|||
}
|
||||
|
||||
|
||||
//////////////////////
|
||||
// usermod functions//
|
||||
//////////////////////
|
||||
|
||||
public:
|
||||
//Functions called by WLED or other usermods
|
||||
|
||||
|
|
@ -961,6 +1001,7 @@ class AudioReactive : public Usermod {
|
|||
disableSoundProcessing = true;
|
||||
}
|
||||
|
||||
if (enabled) connectUDPSoundSync();
|
||||
initDone = true;
|
||||
}
|
||||
|
||||
|
|
@ -971,6 +1012,11 @@ class AudioReactive : public Usermod {
|
|||
*/
|
||||
void connected()
|
||||
{
|
||||
if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection
|
||||
udpSyncConnected = false;
|
||||
fftUdp.stop();
|
||||
}
|
||||
|
||||
if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) {
|
||||
#ifndef ESP8266
|
||||
udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort);
|
||||
|
|
@ -1067,9 +1113,13 @@ class AudioReactive : public Usermod {
|
|||
if (soundAgc) my_magnitude *= multAgc;
|
||||
if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
|
||||
|
||||
limitSampleDynamics(); // optional - makes volumeSmth very smooth and fluent
|
||||
}
|
||||
limitSampleDynamics();
|
||||
} // if (!disableSoundProcessing)
|
||||
|
||||
autoResetPeak(); // auto-reset sample peak after strip minShowDelay
|
||||
if (!udpSyncConnected) udpSamplePeak = false; // reset UDP samplePeak while UDP is unconnected
|
||||
|
||||
connectUDPSoundSync(); // ensure we have a connection - if needed
|
||||
|
||||
// UDP Microphone Sync - receive mode
|
||||
if ((audioSyncEnabled & 0x02) && udpSyncConnected) {
|
||||
|
|
@ -1092,7 +1142,7 @@ class AudioReactive : public Usermod {
|
|||
}
|
||||
#endif
|
||||
|
||||
// peak sample from last 5 seconds
|
||||
// Info Page: keep max sample from last 5 seconds
|
||||
if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) {
|
||||
sampleMaxTimer = millis();
|
||||
maxSample5sec = (0.15 * maxSample5sec) + 0.85 *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing
|
||||
|
|
@ -1100,6 +1150,7 @@ class AudioReactive : public Usermod {
|
|||
} else {
|
||||
if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume
|
||||
}
|
||||
|
||||
//UDP Microphone Sync - transmit mode
|
||||
if ((audioSyncEnabled & 0x01) && (millis() - lastTime > 20)) {
|
||||
// Only run the transmit code IF we're in Transmit mode
|
||||
|
|
@ -1137,8 +1188,9 @@ class AudioReactive : public Usermod {
|
|||
memset(fftCalc, 0, sizeof(fftCalc));
|
||||
memset(fftAvg, 0, sizeof(fftAvg));
|
||||
memset(fftResult, 0, sizeof(fftResult));
|
||||
for(int i=(init?0:1); i<16; i+=2) fftResult[i] = 16; // make a tiny pattern
|
||||
for(int i=(init?0:1); i<NUM_GEQ_CHANNELS; i+=2) fftResult[i] = 16; // make a tiny pattern
|
||||
inputLevel = 128; // resset level slider to default
|
||||
autoResetPeak();
|
||||
|
||||
if (init && FFT_Task) {
|
||||
vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash
|
||||
|
|
@ -1186,6 +1238,10 @@ class AudioReactive : public Usermod {
|
|||
}
|
||||
|
||||
|
||||
////////////////////////////
|
||||
// Settings and Info Page //
|
||||
////////////////////////////
|
||||
|
||||
/*
|
||||
* addToJsonInfo() can be used to add custom entries to the /json/info part of the JSON API.
|
||||
* Creating an "u" object allows you to add custom key/value pairs to the Info section of the WLED web UI.
|
||||
|
|
@ -1307,6 +1363,10 @@ class AudioReactive : public Usermod {
|
|||
infoArr.add(fftTime-sampleTime);
|
||||
infoArr.add("ms");
|
||||
#endif
|
||||
|
||||
// add a small horizontal line, for better readability
|
||||
infoArr = user.createNestedArray(F("<hr style=\"height:1px;border-width:0;color:gray;background-color:gray\" />"));
|
||||
infoArr.add(F(" <hr style=\"height:1px;border-width:0;color:gray;background-color:gray\" /> "));
|
||||
}
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -6251,7 +6251,7 @@ uint16_t mode_gravcentric(void) { // Gravcentric. By Andrew
|
|||
|
||||
return FRAMETIME;
|
||||
} // mode_gravcentric()
|
||||
static const char _data_FX_MODE_GRAVCENTRIC[] PROGMEM = "Gravcentric@Rate of fall,Sensitivity;!;!;ix=128,mp12=2,ssim=0,1d,vo"; // Circle, Beatsin
|
||||
static const char _data_FX_MODE_GRAVCENTRIC[] PROGMEM = "Gravcentric@Rate of fall,Sensitivity;!;!;ix=128,mp12=3,ssim=0,1d,vo"; // Corner, Beatsin
|
||||
|
||||
|
||||
///////////////////////
|
||||
|
|
@ -6387,7 +6387,7 @@ uint16_t mode_midnoise(void) { // Midnoise. By Andrew Tuline.
|
|||
|
||||
return FRAMETIME;
|
||||
} // mode_midnoise()
|
||||
static const char _data_FX_MODE_MIDNOISE[] PROGMEM = "Midnoise@Fade rate,Maximum length;,!;!;ix=128,mp12=2,ssim=0,1d,vo"; // Circle, Beatsin
|
||||
static const char _data_FX_MODE_MIDNOISE[] PROGMEM = "Midnoise@Fade rate,Maximum length;,!;!;ix=128,mp12=1,ssim=0,1d,vo"; // Bar, Beatsin
|
||||
|
||||
|
||||
//////////////////////
|
||||
|
|
@ -6512,7 +6512,7 @@ uint16_t mode_plasmoid(void) { // Plasmoid. By Andrew Tuline.
|
|||
}
|
||||
float volumeSmth = *(float*) um_data->u_data[0];
|
||||
|
||||
SEGMENT.fadeToBlackBy(64);
|
||||
SEGMENT.fadeToBlackBy(32);
|
||||
|
||||
plasmoip->thisphase += beatsin8(6,-4,4); // You can change direction and speed individually.
|
||||
plasmoip->thatphase += beatsin8(7,-4,4); // Two phase values to make a complex pattern. By Andrew Tuline.
|
||||
|
|
@ -6664,7 +6664,8 @@ uint16_t mode_blurz(void) { // Blurz. By Andrew Tuline.
|
|||
SEGENV.aux0 = 0;
|
||||
}
|
||||
|
||||
SEGMENT.fade_out(SEGMENT.speed);
|
||||
int fadeoutDelay = (256 - SEGMENT.speed) / 32;
|
||||
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(SEGMENT.speed);
|
||||
|
||||
SEGENV.step += FRAMETIME;
|
||||
if (SEGENV.step > SPEED_FORMULA_L) {
|
||||
|
|
@ -6732,7 +6733,9 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN.
|
|||
float my_magnitude = *(float*) um_data->u_data[5] / 4.0f;
|
||||
if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
|
||||
|
||||
SEGMENT.fade_out(SEGMENT.speed);
|
||||
if (SEGENV.call == 0) SEGMENT.fill(BLACK);
|
||||
int fadeoutDelay = (256 - SEGMENT.speed) / 32;
|
||||
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(SEGMENT.speed);
|
||||
|
||||
int locn = (log10f((float)FFT_MajorPeak) - 1.78f) * (float)SEGLEN/(MAX_FREQ_LOG10 - 1.78f); // log10 frequency range is from 1.78 to 3.71. Let's scale to SEGLEN.
|
||||
if (locn < 1) locn = 0; // avoid underflow
|
||||
|
|
@ -6747,7 +6750,7 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN.
|
|||
|
||||
return FRAMETIME;
|
||||
} // mode_freqmap()
|
||||
static const char _data_FX_MODE_FREQMAP[] PROGMEM = "Freqmap@Fade rate,Starting color;,!;!;mp12=2,ssim=0,1d,fr"; // Circle, Beatsin
|
||||
static const char _data_FX_MODE_FREQMAP[] PROGMEM = "Freqmap@Fade rate,Starting color;,!;!;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin
|
||||
|
||||
|
||||
///////////////////////
|
||||
|
|
@ -6802,7 +6805,7 @@ uint16_t mode_freqmatrix(void) { // Freqmatrix. By Andreas Plesch
|
|||
|
||||
return FRAMETIME;
|
||||
} // mode_freqmatrix()
|
||||
static const char _data_FX_MODE_FREQMATRIX[] PROGMEM = "Freqmatrix@Time delay,Sound effect,Low bin,High bin,Sensivity;;;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin
|
||||
static const char _data_FX_MODE_FREQMATRIX[] PROGMEM = "Freqmatrix@Time delay,Sound effect,Low bin,High bin,Sensivity;;;mp12=3,ssim=0,1d,fr"; // Corner, Beatsin
|
||||
|
||||
|
||||
//////////////////////
|
||||
|
|
@ -6823,7 +6826,10 @@ uint16_t mode_freqpixels(void) { // Freqpixel. By Andrew Tuline.
|
|||
if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
|
||||
|
||||
uint16_t fadeRate = 2*SEGMENT.speed - SEGMENT.speed*SEGMENT.speed/255; // Get to 255 as quick as you can.
|
||||
SEGMENT.fade_out(fadeRate);
|
||||
|
||||
if (SEGENV.call == 0) SEGMENT.fill(BLACK);
|
||||
int fadeoutDelay = (256 - SEGMENT.speed) / 64;
|
||||
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(fadeRate);
|
||||
|
||||
for (int i=0; i < SEGMENT.intensity/32+1; i++) {
|
||||
uint16_t locn = random16(0,SEGLEN);
|
||||
|
|
@ -6955,7 +6961,7 @@ uint16_t mode_gravfreq(void) { // Gravfreq. By Andrew Tuline.
|
|||
|
||||
return FRAMETIME;
|
||||
} // mode_gravfreq()
|
||||
static const char _data_FX_MODE_GRAVFREQ[] PROGMEM = "Gravfreq@Rate of fall,Sensivity;,!;!;ix=128,mp12=2,ssim=0,1d,fr"; // Circle, Beatsin
|
||||
static const char _data_FX_MODE_GRAVFREQ[] PROGMEM = "Gravfreq@Rate of fall,Sensivity;,!;!;ix=128,mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin
|
||||
|
||||
|
||||
//////////////////////
|
||||
|
|
@ -6969,7 +6975,10 @@ uint16_t mode_noisemove(void) { // Noisemove. By: Andrew Tuli
|
|||
}
|
||||
uint8_t *fftResult = (uint8_t*)um_data->u_data[2];
|
||||
|
||||
SEGMENT.fade_out(224); // Just in case something doesn't get faded.
|
||||
if (SEGENV.call == 0) SEGMENT.fill(BLACK);
|
||||
//SEGMENT.fade_out(224); // Just in case something doesn't get faded.
|
||||
int fadeoutDelay = (256 - SEGMENT.speed) / 96;
|
||||
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fadeToBlackBy(4+ SEGMENT.speed/4);
|
||||
|
||||
uint8_t numBins = map(SEGMENT.intensity,0,255,0,16); // Map slider to fftResult bins.
|
||||
for (int i=0; i<numBins; i++) { // How many active bins are we using.
|
||||
|
|
@ -6995,13 +7004,16 @@ uint16_t mode_rocktaves(void) { // Rocktaves. Same note from eac
|
|||
float FFT_MajorPeak = *(float*) um_data->u_data[4];
|
||||
float my_magnitude = *(float*) um_data->u_data[5] / 16.0f;
|
||||
|
||||
SEGMENT.fadeToBlackBy(64); // Just in case something doesn't get faded.
|
||||
if (SEGENV.call == 0) SEGMENT.fill(BLACK);
|
||||
SEGMENT.fadeToBlackBy(16); // Just in case something doesn't get faded.
|
||||
|
||||
float frTemp = FFT_MajorPeak;
|
||||
uint8_t octCount = 0; // Octave counter.
|
||||
uint8_t volTemp = 0;
|
||||
|
||||
if (my_magnitude > 32) volTemp = 255; // We need to squelch out the background noise.
|
||||
volTemp = 32.0f + my_magnitude * 1.5f; // brightness = volume (overflows are handled in next lines)
|
||||
if (my_magnitude < 48) volTemp = 0; // We need to squelch out the background noise.
|
||||
if (my_magnitude > 144) volTemp = 255; // everything above this is full brightness
|
||||
|
||||
while ( frTemp > 249 ) {
|
||||
octCount++; // This should go up to 5.
|
||||
|
|
@ -7017,7 +7029,7 @@ uint16_t mode_rocktaves(void) { // Rocktaves. Same note from eac
|
|||
|
||||
return FRAMETIME;
|
||||
} // mode_rocktaves()
|
||||
static const char _data_FX_MODE_ROCKTAVES[] PROGMEM = "Rocktaves@;,!;!;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin
|
||||
static const char _data_FX_MODE_ROCKTAVES[] PROGMEM = "Rocktaves@;,!;!;mp12=1,ssim=0,1d,fr"; // Bar, Beatsin
|
||||
|
||||
|
||||
///////////////////////
|
||||
|
|
@ -7101,10 +7113,13 @@ uint16_t mode_2DGEQ(void) { // By Will Tatam. Code reduction by Ewoud Wijma.
|
|||
rippleTime = true;
|
||||
}
|
||||
|
||||
SEGMENT.fadeToBlackBy(SEGMENT.speed);
|
||||
if (SEGENV.call == 0) SEGMENT.fill(BLACK);
|
||||
int fadeoutDelay = (256 - SEGMENT.speed) / 64;
|
||||
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fadeToBlackBy(SEGMENT.speed);
|
||||
|
||||
for (int x=0; x < cols; x++) {
|
||||
uint8_t band = map(x, 0, cols-1, 0, NUM_BANDS - 1);
|
||||
if (NUM_BANDS < 16) band = map(band, 0, NUM_BANDS - 1, 0, 15); // always use full range. comment out this line to get the previous behaviour.
|
||||
band = constrain(band, 0, 15);
|
||||
uint16_t colorIndex = band * 17;
|
||||
uint16_t barHeight = map(fftResult[band], 0, 255, 0, rows); // do not subtract -1 from rows here
|
||||
|
|
|
|||
Ładowanie…
Reference in New Issue