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audio_reactive.h
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audio_reactive.h
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#pragma once
#include "wled.h"
#ifdef ARDUINO_ARCH_ESP32
#include <driver/i2s.h>
#include <driver/adc.h>
#ifdef WLED_ENABLE_DMX
#error This audio reactive usermod is not compatible with DMX Out.
#endif
#endif
#if defined(ARDUINO_ARCH_ESP32) && (defined(WLED_DEBUG) || defined(SR_DEBUG))
#include <esp_timer.h>
#endif
/*
* Usermods allow you to add own functionality to WLED more easily
* See: https://github.com/Aircoookie/WLED/wiki/Add-own-functionality
*
* This is an audioreactive v2 usermod.
* ....
*/
#if !defined(FFTTASK_PRIORITY)
#define FFTTASK_PRIORITY 1 // standard: looptask prio
//#define FFTTASK_PRIORITY 2 // above looptask, below asyc_tcp
//#define FFTTASK_PRIORITY 4 // above asyc_tcp
#endif
// Comment/Uncomment to toggle usb serial debugging
// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
// #define FFT_SAMPLING_LOG // FFT result debugging
// #define SR_DEBUG // generic SR DEBUG messages
#ifdef SR_DEBUG
#define DEBUGSR_PRINT(x) DEBUGOUT.print(x)
#define DEBUGSR_PRINTLN(x) DEBUGOUT.println(x)
#define DEBUGSR_PRINTF(x...) DEBUGOUT.printf(x)
#else
#define DEBUGSR_PRINT(x)
#define DEBUGSR_PRINTLN(x)
#define DEBUGSR_PRINTF(x...)
#endif
#if defined(MIC_LOGGER) || defined(FFT_SAMPLING_LOG)
#define PLOT_PRINT(x) DEBUGOUT.print(x)
#define PLOT_PRINTLN(x) DEBUGOUT.println(x)
#define PLOT_PRINTF(x...) DEBUGOUT.printf(x)
#else
#define PLOT_PRINT(x)
#define PLOT_PRINTLN(x)
#define PLOT_PRINTF(x...)
#endif
#define MAX_PALETTES 3
static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks.
static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value)
static bool udpSyncConnected = false; // UDP connection status -> true if connected to multicast group
#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !!
// audioreactive variables
#ifdef ARDUINO_ARCH_ESP32
static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point
static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
static float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate)
static float sampleAgc = 0.0f; // Smoothed AGC sample
static uint8_t soundAgc = 0; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value)
#endif
//static float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency
static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency
static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
static unsigned long timeOfPeak = 0; // time of last sample peak detection.
static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects
// TODO: probably best not used by receive nodes
//static float agcSensitivity = 128; // AGC sensitivity estimation, based on agc gain (multAgc). calculated by getSensitivity(). range 0..255
// user settable parameters for limitSoundDynamics()
#ifdef UM_AUDIOREACTIVE_DYNAMICS_LIMITER_OFF
static bool limiterOn = false; // bool: enable / disable dynamics limiter
#else
static bool limiterOn = true;
#endif
static uint16_t attackTime = 80; // int: attack time in milliseconds. Default 0.08sec
static uint16_t decayTime = 1400; // int: decay time in milliseconds. Default 1.40sec
// peak detection
#ifdef ARDUINO_ARCH_ESP32
static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[]) - no used for 8266 receive-only mode
#endif
static void autoResetPeak(void); // peak auto-reset function
static uint8_t maxVol = 31; // (was 10) Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
#ifdef ARDUINO_ARCH_ESP32
// use audio source class (ESP32 specific)
#include "audio_source.h"
constexpr i2s_port_t I2S_PORT = I2S_NUM_0; // I2S port to use (do not change !)
constexpr int BLOCK_SIZE = 128; // I2S buffer size (samples)
// globals
static uint8_t inputLevel = 128; // UI slider value
#ifndef SR_SQUELCH
uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value)
#else
uint8_t soundSquelch = SR_SQUELCH; // squelch value for volume reactive routines (config value)
#endif
#ifndef SR_GAIN
uint8_t sampleGain = 60; // sample gain (config value)
#else
uint8_t sampleGain = SR_GAIN; // sample gain (config value)
#endif
// user settable options for FFTResult scaling
static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized square root
//
// AGC presets
// Note: in C++, "const" implies "static" - no need to explicitly declare everything as "static const"
//
#define AGC_NUM_PRESETS 3 // AGC presets: normal, vivid, lazy
const double agcSampleDecay[AGC_NUM_PRESETS] = { 0.9994f, 0.9985f, 0.9997f}; // decay factor for sampleMax, in case the current sample is below sampleMax
const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone
const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone
const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level
const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65%
const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang)
const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85%
const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // quickly follow setpoint - ~0.15 sec
const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs
const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter
const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter
const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value)
// AGC presets end
static AudioSource *audioSource = nullptr;
static bool useBandPassFilter = false; // if true, enables a bandpass filter 80Hz-16Khz to remove noise. Applies before FFT.
////////////////////
// Begin FFT Code //
////////////////////
// some prototypes, to ensure consistent interfaces
static float mapf(float x, float in_min, float in_max, float out_min, float out_max); // map function for float
static float fftAddAvg(int from, int to); // average of several FFT result bins
void FFTcode(void * parameter); // audio processing task: read samples, run FFT, fill GEQ channels from FFT results
static void runMicFilter(uint16_t numSamples, float *sampleBuffer); // pre-filtering of raw samples (band-pass)
static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels); // post-processing and post-amp of GEQ channels
static TaskHandle_t FFT_Task = nullptr;
// Table of multiplication factors so that we can even out the frequency response.
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 };
// globals and FFT Output variables shared with animations
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
static uint64_t fftTime = 0;
static uint64_t sampleTime = 0;
#endif
// FFT Task variables (filtering and post-processing)
static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f}; // Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256.
static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON)
#ifdef SR_DEBUG
static float fftResultMax[NUM_GEQ_CHANNELS] = {0.0f}; // A table used for testing to determine how our post-processing is working.
#endif
// audio source parameters and constant
constexpr SRate_t SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz is a standard rate. Physical sample time -> 23ms
//constexpr SRate_t SAMPLE_RATE = 16000; // 16kHz - use if FFTtask takes more than 20ms. Physical sample time -> 32ms
//constexpr SRate_t SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental. Physical sample time -> 25ms
//constexpr SRate_t SAMPLE_RATE = 10240; // Base sample rate in Hz - previous default. Physical sample time -> 50ms
#define FFT_MIN_CYCLE 21 // minimum time before FFT task is repeated. Use with 22Khz sampling
//#define FFT_MIN_CYCLE 30 // Use with 16Khz sampling
//#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated. Use with 20Khz sampling
//#define FFT_MIN_CYCLE 46 // minimum time before FFT task is repeated. Use with 10Khz sampling
// FFT Constants
constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2
constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
// the following are observed values, supported by a bit of "educated guessing"
//#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels
#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
#define LOG_256 5.54517744f // log(256)
// These are the input and output vectors. Input vectors receive computed results from FFT.
static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
static float vImag[samplesFFT] = {0.0f}; // imaginary parts
// Create FFT object
// lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
// these options actually cause slow-downs on all esp32 processors, don't use them.
// #define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc) - not faster on ESP32
// #define FFT_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt - slower on ESP32
// Below options are forcing ArduinoFFT to use sqrtf() instead of sqrt()
#define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 10-50% on ESP32
#define sqrt_internal sqrtf // see https://github.com/kosme/arduinoFFT/pull/83
#include <arduinoFFT.h>
/* Create FFT object with weighing factor storage */
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, true);
// Helper functions
// float version of map()
static float mapf(float x, float in_min, float in_max, float out_min, float out_max){
return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
}
// compute average of several FFT result bins
static float fftAddAvg(int from, int to) {
float result = 0.0f;
for (int i = from; i <= to; i++) {
result += vReal[i];
}
return result / float(to - from + 1);
}
//
// FFT main task
//
void FFTcode(void * parameter)
{
DEBUGSR_PRINT("FFT started on core: "); DEBUGSR_PRINTLN(xPortGetCoreID());
// see https://www.freertos.org/vtaskdelayuntil.html
const TickType_t xFrequency = FFT_MIN_CYCLE * portTICK_PERIOD_MS;
TickType_t xLastWakeTime = xTaskGetTickCount();
for(;;) {
delay(1); // DO NOT DELETE THIS LINE! It is needed to give the IDLE(0) task enough time and to keep the watchdog happy.
// taskYIELD(), yield(), vTaskDelay() and esp_task_wdt_feed() didn't seem to work.
// Don't run FFT computing code if we're in Receive mode or in realtime mode
if (disableSoundProcessing || (audioSyncEnabled & 0x02)) {
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
continue;
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
uint64_t start = esp_timer_get_time();
bool haveDoneFFT = false; // indicates if second measurement (FFT time) is valid
#endif
// get a fresh batch of samples from I2S
if (audioSource) audioSource->getSamples(vReal, samplesFFT);
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
if (start < esp_timer_get_time()) { // filter out overflows
uint64_t sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding
sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10; // smooth
}
start = esp_timer_get_time(); // start measuring FFT time
#endif
xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay
// band pass filter - can reduce noise floor by a factor of 50
// downside: frequencies below 100Hz will be ignored
if (useBandPassFilter) runMicFilter(samplesFFT, vReal);
// find highest sample in the batch
float maxSample = 0.0f; // max sample from FFT batch
for (int i=0; i < samplesFFT; i++) {
// set imaginary parts to 0
vImag[i] = 0;
// pick our our current mic sample - we take the max value from all samples that go into FFT
if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts
if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]);
}
// release highest sample to volume reactive effects early - not strictly necessary here - could also be done at the end of the function
// early release allows the filters (getSample() and agcAvg()) to work with fresh values - we will have matching gain and noise gate values when we want to process the FFT results.
micDataReal = maxSample;
#ifdef SR_DEBUG
if (true) { // this allows measure FFT runtimes, as it disables the "only when needed" optimization
#else
if (sampleAvg > 0.25f) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed.
#endif
// run FFT (takes 3-5ms on ESP32, ~12ms on ESP32-S2)
FFT.dcRemoval(); // remove DC offset
FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy
//FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection
FFT.compute( FFTDirection::Forward ); // Compute FFT
FFT.complexToMagnitude(); // Compute magnitudes
vReal[0] = 0; // The remaining DC offset on the signal produces a strong spike on position 0 that should be eliminated to avoid issues.
FFT.majorPeak(&FFT_MajorPeak, &FFT_Magnitude); // let the effects know which freq was most dominant
FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
haveDoneFFT = true;
#endif
} else { // noise gate closed - only clear results as FFT was skipped. MIC samples are still valid when we do this.
memset(vReal, 0, sizeof(vReal));
FFT_MajorPeak = 1;
FFT_Magnitude = 0.001;
}
for (int i = 0; i < samplesFFT; i++) {
float t = fabsf(vReal[i]); // just to be sure - values in fft bins should be positive any way
vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max.
} // for()
// mapping of FFT result bins to frequency channels
if (fabsf(sampleAvg) > 0.5f) { // noise gate open
#if 0
/* 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.
*
* Andrew's updated mapping of 256 bins down to the 16 result bins with Sample Freq = 10240, samplesFFT = 512 and some overlap.
* Based on testing, the lowest/Start frequency is 60 Hz (with bin 3) and a highest/End frequency of 5120 Hz in bin 255.
* Now, Take the 60Hz and multiply by 1.320367784 to get the next frequency and so on until the end. Then determine the bins.
* End frequency = Start frequency * multiplier ^ 16
* Multiplier = (End frequency/ Start frequency) ^ 1/16
* Multiplier = 1.320367784
*/ // Range
fftCalc[ 0] = fftAddAvg(2,4); // 60 - 100
fftCalc[ 1] = fftAddAvg(4,5); // 80 - 120
fftCalc[ 2] = fftAddAvg(5,7); // 100 - 160
fftCalc[ 3] = fftAddAvg(7,9); // 140 - 200
fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
fftCalc[11] = fftAddAvg(64,84); // 1280 - 1700
fftCalc[12] = fftAddAvg(84,111); // 1680 - 2240
fftCalc[13] = fftAddAvg(111,147); // 2220 - 2960
fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900
fftCalc[15] = fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate
#else
/* new mapping, optimized for 22050 Hz by softhack007 */
// bins frequency range
if (useBandPassFilter) {
// skip frequencies below 100hz
fftCalc[ 0] = 0.8f * fftAddAvg(3,4);
fftCalc[ 1] = 0.9f * fftAddAvg(4,5);
fftCalc[ 2] = fftAddAvg(5,6);
fftCalc[ 3] = fftAddAvg(6,7);
// don't use the last bins from 206 to 255.
fftCalc[15] = fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
} else {
fftCalc[ 0] = fftAddAvg(1,2); // 1 43 - 86 sub-bass
fftCalc[ 1] = fftAddAvg(2,3); // 1 86 - 129 bass
fftCalc[ 2] = fftAddAvg(3,5); // 2 129 - 216 bass
fftCalc[ 3] = fftAddAvg(5,7); // 2 216 - 301 bass + midrange
// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
}
fftCalc[ 4] = fftAddAvg(7,10); // 3 301 - 430 midrange
fftCalc[ 5] = fftAddAvg(10,13); // 3 430 - 560 midrange
fftCalc[ 6] = fftAddAvg(13,19); // 5 560 - 818 midrange
fftCalc[ 7] = fftAddAvg(19,26); // 7 818 - 1120 midrange -- 1Khz should always be the center !
fftCalc[ 8] = fftAddAvg(26,33); // 7 1120 - 1421 midrange
fftCalc[ 9] = fftAddAvg(33,44); // 9 1421 - 1895 midrange
fftCalc[10] = fftAddAvg(44,56); // 12 1895 - 2412 midrange + high mid
fftCalc[11] = fftAddAvg(56,70); // 14 2412 - 3015 high mid
fftCalc[12] = fftAddAvg(70,86); // 16 3015 - 3704 high mid
fftCalc[13] = fftAddAvg(86,104); // 18 3704 - 4479 high mid
fftCalc[14] = fftAddAvg(104,165) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
#endif
} else { // noise gate closed - just decay old values
for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
fftCalc[i] *= 0.85f; // decay to zero
if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f;
}
}
// post-processing of frequency channels (pink noise adjustment, AGC, smoothing, scaling)
postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS);
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows
uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
}
#endif
// run peak detection
autoResetPeak();
detectSamplePeak();
#if !defined(I2S_GRAB_ADC1_COMPLETELY)
if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC
#endif
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
} // for(;;)ever
} // FFTcode() task end
///////////////////////////
// Pre / Postprocessing //
///////////////////////////
static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // pre-filtering of raw samples (band-pass)
{
// low frequency cutoff parameter - see https://dsp.stackexchange.com/questions/40462/exponential-moving-average-cut-off-frequency
//constexpr float alpha = 0.04f; // 150Hz
//constexpr float alpha = 0.03f; // 110Hz
constexpr float alpha = 0.0225f; // 80hz
//constexpr float alpha = 0.01693f;// 60hz
// high frequency cutoff parameter
//constexpr float beta1 = 0.75f; // 11Khz
//constexpr float beta1 = 0.82f; // 15Khz
//constexpr float beta1 = 0.8285f; // 18Khz
constexpr float beta1 = 0.85f; // 20Khz
constexpr float beta2 = (1.0f - beta1) / 2.0f;
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
static float lowfilt = 0.0f; // IIR low frequency cutoff filter
for (int i=0; i < numSamples; i++) {
// FIR lowpass, to remove high frequency noise
float highFilteredSample;
if (i < (numSamples-1)) highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*sampleBuffer[i+1]; // smooth out spikes
else highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*last_vals[1]; // special handling for last sample in array
last_vals[1] = last_vals[0];
last_vals[0] = sampleBuffer[i];
sampleBuffer[i] = highFilteredSample;
// IIR highpass, to remove low frequency noise
lowfilt += alpha * (sampleBuffer[i] - lowfilt);
sampleBuffer[i] = sampleBuffer[i] - lowfilt;
}
}
static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // post-processing and post-amp of GEQ channels
{
for (int i=0; i < numberOfChannels; i++) {
if (noiseGateOpen) { // noise gate open
// Adjustment for frequency curves.
fftCalc[i] *= fftResultPink[i];
if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
// Manual linear adjustment of gain using sampleGain adjustment for different input types.
fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //apply gain, with inputLevel adjustment
if(fftCalc[i] < 0) fftCalc[i] = 0;
}
// smooth results - rise fast, fall slower
if(fftCalc[i] > fftAvg[i]) // rise fast
fftAvg[i] = fftCalc[i] *0.75f + 0.25f*fftAvg[i]; // will need approx 2 cycles (50ms) for converging against fftCalc[i]
else { // fall slow
if (decayTime < 1000) fftAvg[i] = fftCalc[i]*0.22f + 0.78f*fftAvg[i]; // approx 5 cycles (225ms) for falling to zero
else if (decayTime < 2000) fftAvg[i] = fftCalc[i]*0.17f + 0.83f*fftAvg[i]; // default - approx 9 cycles (225ms) for falling to zero
else if (decayTime < 3000) fftAvg[i] = fftCalc[i]*0.14f + 0.86f*fftAvg[i]; // approx 14 cycles (350ms) for falling to zero
else fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i]; // approx 20 cycles (500ms) for falling to zero
}
// constrain internal vars - just to be sure
fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f);
fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f);
float currentResult;
if(limiterOn == true)
currentResult = fftAvg[i];
else
currentResult = fftCalc[i];
switch (FFTScalingMode) {
case 1:
// Logarithmic scaling
currentResult *= 0.42f; // 42 is the answer ;-)
currentResult -= 8.0f; // this skips the lowest row, giving some room for peaks
if (currentResult > 1.0f) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
else currentResult = 0.0f; // special handling, because log(1) = 0; log(0) = undefined
currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies
currentResult = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255]
break;
case 2:
// Linear scaling
currentResult *= 0.30f; // needs a bit more damping, get stay below 255
currentResult -= 4.0f; // giving a bit more room for peaks
if (currentResult < 1.0f) currentResult = 0.0f;
currentResult *= 0.85f + (float(i)/1.8f); // extra up-scaling for high frequencies
break;
case 3:
// square root scaling
currentResult *= 0.38f;
currentResult -= 6.0f;
if (currentResult > 1.0f) currentResult = sqrtf(currentResult);
else currentResult = 0.0f; // special handling, because sqrt(0) = undefined
currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
currentResult = mapf(currentResult, 0.0, 16.0, 0.0, 255.0); // map [sqrt(1) ... sqrt(256)] to [0 ... 255]
break;
case 0:
default:
// no scaling - leave freq bins as-is
currentResult -= 4; // just a bit more room for peaks
break;
}
// Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely.
if (soundAgc > 0) { // apply extra "GEQ Gain" if set by user
float post_gain = (float)inputLevel/128.0f;
if (post_gain < 1.0f) post_gain = ((post_gain -1.0f) * 0.8f) +1.0f;
currentResult *= post_gain;
}
fftResult[i] = constrain((int)currentResult, 0, 255);
}
}
////////////////////
// Peak detection //
////////////////////
// peak detection is called from FFT task when vReal[] contains valid FFT results
static void detectSamplePeak(void) {
bool havePeak = false;
// softhack007: this code continuously triggers while amplitude in the selected bin is above a certain threshold. So it does not detect peaks - it detects high activity in a frequency bin.
// Poor man's beat detection by seeing if sample > Average + some value.
// This goes through ALL of the 255 bins - but ignores stupid settings
// Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync.
if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 4) && (vReal[binNum] > maxVol) && ((millis() - timeOfPeak) > 100)) {
havePeak = true;
}
if (havePeak) {
samplePeak = true;
timeOfPeak = millis();
udpSamplePeak = true;
}
}
#endif
static void autoResetPeak(void) {
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
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
if (audioSyncEnabled == 0) udpSamplePeak = false; // this is normally reset by transmitAudioData
}
}
////////////////////
// usermod class //
////////////////////
//class name. Use something descriptive and leave the ": public Usermod" part :)
class AudioReactive : public Usermod {
private:
#ifdef ARDUINO_ARCH_ESP32
#ifndef AUDIOPIN
int8_t audioPin = -1;
#else
int8_t audioPin = AUDIOPIN;
#endif
#ifndef SR_DMTYPE // I2S mic type
uint8_t dmType = 1; // 0=none/disabled/analog; 1=generic I2S
#define SR_DMTYPE 1 // default type = I2S
#else
uint8_t dmType = SR_DMTYPE;
#endif
#ifndef I2S_SDPIN // aka DOUT
int8_t i2ssdPin = 32;
#else
int8_t i2ssdPin = I2S_SDPIN;
#endif
#ifndef I2S_WSPIN // aka LRCL
int8_t i2swsPin = 15;
#else
int8_t i2swsPin = I2S_WSPIN;
#endif
#ifndef I2S_CKPIN // aka BCLK
int8_t i2sckPin = 14; /*PDM: set to I2S_PIN_NO_CHANGE*/
#else
int8_t i2sckPin = I2S_CKPIN;
#endif
#ifndef MCLK_PIN
int8_t mclkPin = I2S_PIN_NO_CHANGE; /* ESP32: only -1, 0, 1, 3 allowed*/
#else
int8_t mclkPin = MCLK_PIN;
#endif
#endif
// new "V2" audiosync struct - 40 Bytes
struct audioSyncPacket {
char header[6]; // 06 Bytes
float sampleRaw; // 04 Bytes - either "sampleRaw" or "rawSampleAgc" depending on soundAgc setting
float sampleSmth; // 04 Bytes - either "sampleAvg" or "sampleAgc" depending on soundAgc setting
uint8_t samplePeak; // 01 Bytes - 0 no peak; >=1 peak detected. In future, this will also provide peak Magnitude
uint8_t reserved1; // 01 Bytes - for future extensions - not used yet
uint8_t fftResult[16]; // 16 Bytes
float FFT_Magnitude; // 04 Bytes
float FFT_MajorPeak; // 04 Bytes
};
// old "V1" audiosync struct - 83 Bytes - for backwards compatibility
struct audioSyncPacket_v1 {
char header[6]; // 06 Bytes
uint8_t myVals[32]; // 32 Bytes
int sampleAgc; // 04 Bytes
int sampleRaw; // 04 Bytes
float sampleAvg; // 04 Bytes
bool samplePeak; // 01 Bytes
uint8_t fftResult[16]; // 16 Bytes
double FFT_Magnitude; // 08 Bytes
double FFT_MajorPeak; // 08 Bytes
};
// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
#ifdef UM_AUDIOREACTIVE_ENABLE
bool enabled = true;
#else
bool enabled = false;
#endif
bool initDone = false;
bool addPalettes = false;
int8_t palettes = 0;
// variables for UDP sound sync
WiFiUDP fftUdp; // UDP object for sound sync (from WiFi UDP, not Async UDP!)
unsigned long lastTime = 0; // last time of running UDP Microphone Sync
const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED
uint16_t audioSyncPort= 11988;// default port for UDP sound sync
bool updateIsRunning = false; // true during OTA.
#ifdef ARDUINO_ARCH_ESP32
// used for AGC
int last_soundAgc = -1; // used to detect AGC mode change (for resetting AGC internal error buffers)
double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error
// variables used by getSample() and agcAvg()
int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed
double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controller.
double micLev = 0.0; // Used to convert returned value to have '0' as minimum. A leveller
float expAdjF = 0.0f; // Used for exponential filter.
float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC.
int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel)
int16_t rawSampleAgc = 0; // not smoothed AGC sample
#endif
// variables used in effects
float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc
float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc
// used to feed "Info" Page
unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket
int receivedFormat = 0; // last received UDP sound sync format - 0=none, 1=v1 (0.13.x), 2=v2 (0.14.x)
float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds
unsigned long sampleMaxTimer = 0; // last time maxSample5sec was reset
#define CYCLE_SAMPLEMAX 3500 // time window for merasuring
// strings to reduce flash memory usage (used more than twice)
static const char _name[];
static const char _enabled[];
static const char _config[];
static const char _dynamics[];
static const char _frequency[];
static const char _inputLvl[];
#if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
static const char _analogmic[];
#endif
static const char _digitalmic[];
static const char _addPalettes[];
static const char UDP_SYNC_HEADER[];
static const char UDP_SYNC_HEADER_v1[];
// private methods
void removeAudioPalettes(void);
void createAudioPalettes(void);
CRGB getCRGBForBand(int x, int pal);
void fillAudioPalettes(void);
////////////////////
// Debug support //
////////////////////
void logAudio()
{
if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & 0x02) == 0))) return; // no audio availeable
#ifdef MIC_LOGGER
// Debugging functions for audio input and sound processing. Comment out the values you want to see
PLOT_PRINT("micReal:"); PLOT_PRINT(micDataReal); PLOT_PRINT("\t");
PLOT_PRINT("volumeSmth:"); PLOT_PRINT(volumeSmth); PLOT_PRINT("\t");
//PLOT_PRINT("volumeRaw:"); PLOT_PRINT(volumeRaw); PLOT_PRINT("\t");
PLOT_PRINT("DC_Level:"); PLOT_PRINT(micLev); PLOT_PRINT("\t");
//PLOT_PRINT("sampleAgc:"); PLOT_PRINT(sampleAgc); PLOT_PRINT("\t");
//PLOT_PRINT("sampleAvg:"); PLOT_PRINT(sampleAvg); PLOT_PRINT("\t");
//PLOT_PRINT("sampleReal:"); PLOT_PRINT(sampleReal); PLOT_PRINT("\t");
#ifdef ARDUINO_ARCH_ESP32
//PLOT_PRINT("micIn:"); PLOT_PRINT(micIn); PLOT_PRINT("\t");
//PLOT_PRINT("sample:"); PLOT_PRINT(sample); PLOT_PRINT("\t");
//PLOT_PRINT("sampleMax:"); PLOT_PRINT(sampleMax); PLOT_PRINT("\t");
//PLOT_PRINT("samplePeak:"); PLOT_PRINT((samplePeak!=0) ? 128:0); PLOT_PRINT("\t");
//PLOT_PRINT("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t");
#endif
PLOT_PRINTLN();
#endif
#ifdef FFT_SAMPLING_LOG
#if 0
for(int i=0; i<NUM_GEQ_CHANNELS; i++) {
PLOT_PRINT(fftResult[i]);
PLOT_PRINT("\t");
}
PLOT_PRINTLN();
#endif
// OPTIONS are in the following format: Description \n Option
//
// Set true if wanting to see all the bands in their own vertical space on the Serial Plotter, false if wanting to see values in Serial Monitor
const bool mapValuesToPlotterSpace = false;
// Set true to apply an auto-gain like setting to to the data (this hasn't been tested recently)
const bool scaleValuesFromCurrentMaxVal = false;
// prints the max value seen in the current data
const bool printMaxVal = false;
// prints the min value seen in the current data
const bool printMinVal = false;
// if !scaleValuesFromCurrentMaxVal, we scale values from [0..defaultScalingFromHighValue] to [0..scalingToHighValue], lower this if you want to see smaller values easier
const int defaultScalingFromHighValue = 256;
// Print values to terminal in range of [0..scalingToHighValue] if !mapValuesToPlotterSpace, or [(i)*scalingToHighValue..(i+1)*scalingToHighValue] if mapValuesToPlotterSpace
const int scalingToHighValue = 256;
// set higher if using scaleValuesFromCurrentMaxVal and you want a small value that's also the current maxVal to look small on the plotter (can't be 0 to avoid divide by zero error)
const int minimumMaxVal = 1;
int maxVal = minimumMaxVal;
int minVal = 0;
for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
if(fftResult[i] > maxVal) maxVal = fftResult[i];
if(fftResult[i] < minVal) minVal = fftResult[i];
}
for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
PLOT_PRINT(i); PLOT_PRINT(":");
PLOT_PRINTF("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1));
}
if(printMaxVal) {
PLOT_PRINTF("maxVal:%04d ", maxVal + (mapValuesToPlotterSpace ? 16*256 : 0));
}
if(printMinVal) {
PLOT_PRINTF("%04d:minVal ", minVal); // printed with value first, then label, so negative values can be seen in Serial Monitor but don't throw off y axis in Serial Plotter
}
if(mapValuesToPlotterSpace)
PLOT_PRINTF("max:%04d ", (printMaxVal ? 17 : 16)*256); // print line above the maximum value we expect to see on the plotter to avoid autoscaling y axis
else {
PLOT_PRINTF("max:%04d ", 256);
}
PLOT_PRINTLN();
#endif // FFT_SAMPLING_LOG
} // logAudio()
#ifdef ARDUINO_ARCH_ESP32
//////////////////////
// Audio Processing //
//////////////////////
/*
* A "PI controller" multiplier to automatically adjust sound sensitivity.
*
* A few tricks are implemented so that sampleAgc does't only utilize 0% and 100%:
* 0. don't amplify anything below squelch (but keep previous gain)
* 1. gain input = maximum signal observed in the last 5-10 seconds
* 2. we use two setpoints, one at ~60%, and one at ~80% of the maximum signal
* 3. the amplification depends on signal level:
* a) normal zone - very slow adjustment
* b) emergency zone (<10% or >90%) - very fast adjustment
*/
void agcAvg(unsigned long the_time)
{
const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function
float lastMultAgc = multAgc; // last multiplier used
float multAgcTemp = multAgc; // new multiplier
float tmpAgc = sampleReal * multAgc; // what-if amplified signal
float control_error; // "control error" input for PI control
if (last_soundAgc != soundAgc)
control_integrated = 0.0; // new preset - reset integrator
// For PI controller, we need to have a constant "frequency"
// so let's make sure that the control loop is not running at insane speed
static unsigned long last_time = 0;
unsigned long time_now = millis();
if ((the_time > 0) && (the_time < time_now)) time_now = the_time; // allow caller to override my clock
if (time_now - last_time > 2) {
last_time = time_now;
if((fabsf(sampleReal) < 2.0f) || (sampleMax < 1.0)) {
// MIC signal is "squelched" - deliver silence
tmpAgc = 0;
// we need to "spin down" the intgrated error buffer
if (fabs(control_integrated) < 0.01) control_integrated = 0.0;
else control_integrated *= 0.91;
} else {
// compute new setpoint
if (tmpAgc <= agcTarget0Up[AGC_preset])
multAgcTemp = agcTarget0[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = first setpoint
else
multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
}
// limit amplification
if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
// compute error terms
control_error = multAgcTemp - lastMultAgc;
if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
control_integrated += control_error * 0.002 * 0.25; // 2ms = integration time; 0.25 for damping
else
control_integrated *= 0.9; // spin down that beasty integrator
// apply PI Control
tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain
if ((tmpAgc > agcZoneHigh[AGC_preset]) || (tmpAgc < soundSquelch + agcZoneLow[AGC_preset])) { // upper/lower energy zone
multAgcTemp = lastMultAgc + agcFollowFast[AGC_preset] * agcControlKp[AGC_preset] * control_error;
multAgcTemp += agcFollowFast[AGC_preset] * agcControlKi[AGC_preset] * control_integrated;
} else { // "normal zone"
multAgcTemp = lastMultAgc + agcFollowSlow[AGC_preset] * agcControlKp[AGC_preset] * control_error;
multAgcTemp += agcFollowSlow[AGC_preset] * agcControlKi[AGC_preset] * control_integrated;
}
// limit amplification again - PI controller sometimes "overshoots"
//multAgcTemp = constrain(multAgcTemp, 0.015625f, 32.0f); // 1/64 < multAgcTemp < 32
if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
}
// NOW finally amplify the signal
tmpAgc = sampleReal * multAgcTemp; // apply gain to signal
if (fabsf(sampleReal) < 2.0f) tmpAgc = 0.0f; // apply squelch threshold
//tmpAgc = constrain(tmpAgc, 0, 255);
if (tmpAgc > 255) tmpAgc = 255.0f; // limit to 8bit
if (tmpAgc < 1) tmpAgc = 0.0f; // just to be sure
// update global vars ONCE - multAgc, sampleAGC, rawSampleAgc
multAgc = multAgcTemp;
rawSampleAgc = 0.8f * tmpAgc + 0.2f * (float)rawSampleAgc;
// update smoothed AGC sample
if (fabsf(tmpAgc) < 1.0f)
sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero
else
sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
sampleAgc = fabsf(sampleAgc); // // make sure we have a positive value
last_soundAgc = soundAgc;
} // agcAvg()
// post-processing and filtering of MIC sample (micDataReal) from FFTcode()
void getSample()
{
float sampleAdj; // Gain adjusted sample value
float tmpSample; // An interim sample variable used for calculations.
const float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function
#ifdef WLED_DISABLE_SOUND
micIn = inoise8(millis(), millis()); // Simulated analog read
micDataReal = micIn;
#else
#ifdef ARDUINO_ARCH_ESP32
micIn = int(micDataReal); // micDataSm = ((micData * 3) + micData)/4;
#else
// this is the minimal code for reading analog mic input on 8266.
// warning!! Absolutely experimental code. Audio on 8266 is still not working. Expects a million follow-on problems.
static unsigned long lastAnalogTime = 0;
static float lastAnalogValue = 0.0f;
if (millis() - lastAnalogTime > 20) {
micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only.
lastAnalogTime = millis();
lastAnalogValue = micDataReal;
yield();
} else micDataReal = lastAnalogValue;
micIn = int(micDataReal);
#endif
#endif
micLev += (micDataReal-micLev) / 12288.0f;
if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
micIn -= micLev; // Let's center it to 0 now
// Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
float micInNoDC = fabsf(micDataReal - micLev);
expAdjF = (weighting * micInNoDC + (1.0f-weighting) * expAdjF);
expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
tmpSample = expAdjF;
micIn = abs(micIn); // And get the absolute value of each sample
sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment
sampleReal = tmpSample;
sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
// keep "peak" sample, but decay value if current sample is below peak
if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) {
sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
// another simple way to detect samplePeak - cannot detect beats, but reacts on peak volume
if (((binNum < 12) || ((maxVol < 1))) && (millis() - timeOfPeak > 80) && (sampleAvg > 1)) {
samplePeak = true;
timeOfPeak = millis();
udpSamplePeak = true;
}
} else {
if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0))
sampleMax += 0.5f * (sampleReal - sampleMax); // over AGC Zone - get back quickly
else
sampleMax *= agcSampleDecay[AGC_preset]; // signal to zero --> 5-8sec
}
if (sampleMax < 0.5f) sampleMax = 0.0f;
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
sampleAvg = fabsf(sampleAvg); // make sure we have a positive value
} // getSample()
#endif
/* Limits the dynamics of volumeSmth (= sampleAvg or sampleAgc).
* does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc)
*/
// effects: Gravimeter, Gravcenter, Gravcentric, Noisefire, Plasmoid, Freqpixels, Freqwave, Gravfreq, (2D Swirl, 2D Waverly)
void limitSampleDynamics(void) {
const float bigChange = 196; // just a representative number - a large, expected sample value
static unsigned long last_time = 0;
static float last_volumeSmth = 0.0f;
if (limiterOn == false) return;
long delta_time = millis() - last_time;
delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> sily lil hick-up
float deltaSample = volumeSmth - last_volumeSmth;
if (attackTime > 0) { // user has defined attack time > 0
float maxAttack = bigChange * float(delta_time) / float(attackTime);
if (deltaSample > maxAttack) deltaSample = maxAttack;
}
if (decayTime > 0) { // user has defined decay time > 0
float maxDecay = - bigChange * float(delta_time) / float(decayTime);
if (deltaSample < maxDecay) deltaSample = maxDecay;
}
volumeSmth = last_volumeSmth + deltaSample;
last_volumeSmth = volumeSmth;
last_time = millis();
}
//////////////////////
// UDP Sound Sync //
//////////////////////
// try to establish UDP sound sync connection
void connectUDPSoundSync(void) {
// This function tries to establish a UDP sync connection if needed
// necessary as we also want to transmit in "AP Mode", but the standard "connected()" callback only reacts on STA connection
static unsigned long last_connection_attempt = 0;
if ((audioSyncPort <= 0) || ((audioSyncEnabled & 0x03) == 0)) return; // Sound Sync not enabled
if (udpSyncConnected) return; // already connected
if (!(apActive || interfacesInited)) return; // neither AP nor other connections availeable
if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds
if (updateIsRunning) return;
// if we arrive here, we need a UDP connection but don't have one
last_connection_attempt = millis();
connected(); // try to start UDP
}
#ifdef ARDUINO_ARCH_ESP32
void transmitAudioData()
{
if (!udpSyncConnected) return;
//DEBUGSR_PRINTLN("Transmitting UDP Mic Packet");
audioSyncPacket transmitData;
memset(reinterpret_cast<void *>(&transmitData), 0, sizeof(transmitData)); // make sure that the packet - including "invisible" padding bytes added by the compiler - is fully initialized
strncpy_P(transmitData.header, PSTR(UDP_SYNC_HEADER), 6);
// transmit samples that were not modified by limitSampleDynamics()
transmitData.sampleRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
transmitData.sampleSmth = (soundAgc) ? sampleAgc : sampleAvg;
transmitData.samplePeak = udpSamplePeak ? 1:0;
udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it
transmitData.reserved1 = 0;