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WLED_MM_Infinity/usermods/audioreactive/audio_reactive.h
Frank 523893be02 8266 audioreactive: fix crash during OTA 
* fix crash when starting OTA: `Panic core_esp8266_main.cpp:191 __yield `
* prevent sound sync reconnect during OTA
2023-08-25 16:18:37 +02:00

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#pragma once
#include "wled.h"
#ifdef ARDUINO_ARCH_ESP32
#include <driver/i2s.h>
#include <driver/adc.h>
#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.
* ....
*/
#define FFT_PREFER_EXACT_PEAKS // use different FFT wndowing -> results in "sharper" peaks and less "leaking" into other frequencies
//#define SR_STATS
#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
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
// this applies "pink noise scaling" to FFT results before computing the major peak for effects.
// currently only for ESP32-S3 and classic ESP32, due to increased runtime
#define FFT_MAJORPEAK_HUMAN_EAR
#endif
// high-resolution type for input filters
#if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
#define SR_HIRES_TYPE double // ESP32 and ESP32-S3 (with FPU) are fast enough to use "double"
#else
#define SR_HIRES_TYPE float // prefer faster type on slower boards (-S2, -C3)
#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(x)
#define DEBUGSR_PRINTLN(x) DEBUGOUTLN(x)
#define DEBUGSR_PRINTF(x...) DEBUGOUTF(x)
#else
#define DEBUGSR_PRINT(x)
#define DEBUGSR_PRINTLN(x)
#define DEBUGSR_PRINTF(x...)
#endif
#if defined(SR_DEBUG)
#define ERRORSR_PRINT(x) DEBUGSR_PRINT(x)
#define ERRORSR_PRINTLN(x) DEBUGSR_PRINTLN(x)
#define ERRORSR_PRINTF(x...) DEBUGSR_PRINTF(x)
#else
#if defined(WLED_DEBUG)
#define ERRORSR_PRINT(x) DEBUG_PRINT(x)
#define ERRORSR_PRINTLN(x) DEBUG_PRINTLN(x)
#define ERRORSR_PRINTF(x...) DEBUG_PRINTF(x)
#else
#define ERRORSR_PRINT(x)
#define ERRORSR_PRINTLN(x)
#define ERRORSR_PRINTF(x...)
#endif
#endif
#if defined(MIC_LOGGER) || defined(FFT_SAMPLING_LOG)
#define PLOT_PRINT(x) DEBUGOUT(x)
#define PLOT_PRINTLN(x) DEBUGOUTLN(x)
#define PLOT_PRINTF(x...) DEBUGOUTF(x)
#define PLOT_FLUSH() DEBUGOUTFlush()
#else
#define PLOT_PRINT(x)
#define PLOT_PRINTLN(x)
#define PLOT_PRINTF(x...)
#define PLOT_FLUSH()
#endif
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
#ifdef SR_SQUELCH
static uint8_t soundAgc = 1; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value) - enable AGC if default "squelch" was provided
#else
static uint8_t soundAgc = 0; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value)
#endif
#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()
static bool limiterOn = true; // bool: enable / disable dynamics limiter
static uint16_t attackTime = 50; // int: attack time in milliseconds. Default 0.08sec
static uint16_t decayTime = 300; // int: decay time in milliseconds. New default 300ms. Old default was 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 sqare root
#ifndef SR_FREQ_PROF
static uint8_t pinkIndex = 0; // 0: default; 1: line-in; 2: IMNP441
#else
static uint8_t pinkIndex = SR_FREQ_PROF; // 0: default; 1: line-in; 2: IMNP441
#endif
//
// 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 uint8_t useInputFilter = 0; // enables low-cut filtering. Applies before FFT.
//WLEDMM add experimental settings
static uint8_t micLevelMethod = 0; // 0=old "floating" miclev, 1=new "freeze" mode, 2=fast freeze mode (mode 2 may not work for you)
#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3)
static uint8_t averageByRMS = false; // false: use mean value, true: use RMS (root mean squared). use simpler method on slower MCUs.
#else
static uint8_t averageByRMS = true; // false: use mean value, true: use RMS (root mean squared). use better method on fast MCUs.
#endif
static uint8_t freqDist = 0; // 0=old 1=rightshift mode
// variables used in effects
//static int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc
//static float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc
// shared vars for debugging
#ifdef MIC_LOGGER
static volatile float micReal_min = 0.0f; // MicIn data min from last batch of samples
static volatile float micReal_avg = 0.0f; // MicIn data average (from last batch of samples)
static volatile float micReal_max = 0.0f; // MicIn data max from last batch of samples
#if 0
static volatile float micReal_min2 = 0.0f; // MicIn data min after filtering
static volatile float micReal_max2 = 0.0f; // MicIn data max after filtering
#endif
#endif
////////////////////
// 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.
#define MAX_PINK 10 // 0 = standard, 1= line-in (pink moise only), 2..4 = IMNP441, 5..6 = ICS-43434, ,7=SPM1423, 8..9 = userdef, 10= flat (no pink noise adjustment)
static const float fftResultPink[MAX_PINK+1][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 }, // 0 default from SR WLED
// { 1.30f, 1.32f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.39f, 3.09f, 4.34f }, // - Line-In Generic -> pink noise adjustment only
{ 2.35f, 1.32f, 1.32f, 1.40f, 1.48f, 1.57f, 1.68f, 1.80f, 1.89f, 1.95f, 2.14f, 2.26f, 2.50f, 2.90f, 4.20f, 6.50f }, // 1 Line-In CS5343 + DC blocker
{ 1.82f, 1.72f, 1.70f, 1.50f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.90f, 3.86f, 6.29f}, // 2 IMNP441 datasheet response profile * pink noise
{ 2.80f, 2.20f, 1.30f, 1.15f, 1.55f, 2.45f, 4.20f, 2.80f, 3.20f, 3.60f, 4.20f, 4.90f, 5.70f, 6.05f,10.50f,14.85f}, // 3 IMNP441 - big speaker, strong bass
// next one has not much visual differece compared to default IMNP441 profile
{ 12.0f, 6.60f, 2.60f, 1.15f, 1.35f, 2.05f, 2.85f, 2.50f, 2.85f, 3.30f, 2.25f, 4.35f, 3.80f, 3.75f, 6.50f, 9.00f}, // 4 IMNP441 - voice, or small speaker
{ 2.75f, 1.60f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 1.75f, 2.55f, 3.60f }, // 5 ICS-43434 datasheet response * pink noise
{ 2.90f, 1.25f, 0.75f, 1.08f, 2.35f, 3.55f, 3.60f, 3.40f, 2.75f, 3.45f, 4.40f, 6.35f, 6.80f, 6.80f, 8.50f,10.64f }, // 6 ICS-43434 - big speaker, strong bass
{ 1.65f, 1.00f, 1.05f, 1.30f, 1.48f, 1.30f, 1.80f, 3.00f, 1.50f, 1.65f, 2.56f, 3.00f, 2.60f, 2.30f, 5.00f, 3.00f }, // 7 SPM1423
{ 2.25f, 1.60f, 1.30f, 1.60f, 2.20f, 3.20f, 3.06f, 2.60f, 2.85f, 3.50f, 4.10f, 4.80f, 5.70f, 6.05f,10.50f,14.85f }, // 8 userdef #1 for ewowi (enhance median/high freqs)
{ 4.75f, 3.60f, 2.40f, 2.46f, 3.52f, 1.60f, 1.68f, 3.20f, 2.20f, 2.00f, 2.30f, 2.41f, 2.30f, 1.25f, 4.55f, 6.50f }, // 9 userdef #2 for softhack (mic hidden inside mini-shield)
{ 2.38f, 2.18f, 2.07f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.95f, 1.70f, 2.13f, 2.47f } // 10 almost FLAT (IMNP441 but no PINK noise adjustments)
};
/* how to make your own profile:
* ===============================
* preparation: make sure your microphone has direct line-of-sigh with the speaker, 1-2meter distance is best
* Prepare your HiFi equipment: disable all "Sound enhancements" - like Loudness, Equalizer, Bass Boost. Bass/Trebble controls set to middle.
* Your HiFi equipment should receive its audio input from Line-In, SPDIF, HDMI, or another "undistorted" connection (like CDROM).
* Try not to use Bluetooth or MP3 when playing the "pink noise" audio. BT-audio and MP3 both perform "acoustic adjustments" that we don't want now.
* SR WLED: enable AGC ("standard" or "lazy"), set squelch to a low level, check that LEDs don't reacts in silence.
* SR WLED: select "Generic Line-In" as your Frequency Profile, "Linear" or "Square Root" as Frequency Scale
* SR WLED: Dynamic Limiter On, Dynamics Fall Time around 4200 - makes GEQ hold peaks for much longer
* SR WLED: Select GEQ effect, move all effect slider to max (i.e. right side)
* Measure: play Pink Noise for 2-3 minutes - for examples from youtube https://www.youtube.com/watch?v=ZXtimhT-ff4
* Measure: Take a Photo. Make sure that LEDs for each "bar" are well visible (ou need to count them later)
* Your own profile:
* - Target for each LED bar is 50% to 75% of the max height --> 8(high) x 16(wide) panel means target = 5. 32 x 16 means target = 22.
* - From left to right - count the LEDs in each of the 16 frequency colums (that's why you need the photo). This is the barheight for each channel.
* - math time! Find the multiplier that will bring each bar to to target.
* * in case of square root scale: multiplier = (target * target) / (barheight * barheight)
* * in case of linear scale: multiplier = target / barheight
*
* - replace one of the "userdef" lines with a copy of the parameter line for "Line-In",
* - go through your new "userdef" parameter line, multiply each entry with the mutliplier you found for that column.
* Compile + upload
* Test your new profile (same procedure as above). Iterate the process to improve results.
*/
// globals and FFT Output variables shared with animations
static float FFT_MajPeakSmth = 1.0f; // FFT: (peak) frequency, smooth
#if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS)
static float fftTaskCycle = 0; // avg cycle time for FFT task
static float fftTime = 0; // avg time for single FFT
static float sampleTime = 0; // avg (blocked) time for reading I2S samples
#endif
// FFT Task variables (filtering and post-processing)
static float lastFftCalc[NUM_GEQ_CHANNELS] = {0.0f}; // backup of last FFT channels (before postprocessing)
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)
#if !defined(CONFIG_IDF_TARGET_ESP32C3)
// 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
#ifndef WLEDMM_FASTPATH
#define FFT_MIN_CYCLE 21 // minimum time before FFT task is repeated. Use with 22Khz sampling
#else
#define FFT_MIN_CYCLE 15 // reduce min time, to allow faster catch-up when I2S is lagging
#endif
//#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
#else
// slightly lower the sampling rate for -C3, to improve stability
//constexpr SRate_t SAMPLE_RATE = 20480; // 20Khz; Physical sample time -> 25ms
//#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated.
constexpr SRate_t SAMPLE_RATE = 18000; // 18Khz; Physical sample time -> 28ms
#define FFT_MIN_CYCLE 25 // minimum time before FFT task is repeated.
// try 16Khz in case your device still lags and responds too slowly.
//constexpr SRate_t SAMPLE_RATE = 16000; // 16Khz -> Physical sample time -> 32ms
//#define FFT_MIN_CYCLE 30 // minimum time before FFT task is repeated.
#endif
// 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 FFT_DOWNSCALE 0.40f // downscaling factor for FFT results, RMS averaging
#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
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static float windowWeighingFactors[samplesFFT] = {0.0f};
#endif
#ifdef FFT_MAJORPEAK_HUMAN_EAR
static float pinkFactors[samplesFFT] = {0.0f}; // "pink noise" correction factors
constexpr float pinkcenter = 23.66; // sqrt(560) - center freq for scaling is 560 hz.
constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range of each FFT result bin
#endif
// Create FFT object
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
// lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
// these options actually cause slow-down on -S2 (-S2 doesn't have floating point hardware)
//#define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and an a few other speedups - WLEDMM not faster on ESP32
//#define FFT_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt - WLEDMM slower on ESP32
#endif
#define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 10-50% on ESP32 (as alternative to FFT_SQRT_APPROXIMATION)
#define sqrt_internal sqrtf // see https://github.com/kosme/arduinoFFT/pull/83
#else
// around 50% slower on -S2
// lib_deps += https://github.com/blazoncek/arduinoFFT.git
#endif
#include <arduinoFFT.h>
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors);
#else
static arduinoFFT FFT = arduinoFFT(vReal, vImag, samplesFFT, SAMPLE_RATE);
#endif
// 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 resut bins
// linear average
static float fftAddAvgLin(int from, int to) {
float result = 0.0f;
for (int i = from; i <= to; i++) {
result += vReal[i];
}
return result / float(to - from + 1);
}
// RMS average
static float fftAddAvgRMS(int from, int to) {
double result = 0.0;
for (int i = from; i <= to; i++) {
result += vReal[i] * vReal[i];
}
return sqrtf(result / float(to - from + 1));
}
static float fftAddAvg(int from, int to) {
if (from == to) return vReal[from]; // small optimization
if (averageByRMS) return fftAddAvgRMS(from, to); // use SMS
else return fftAddAvgLin(from, to); // use linear average
}
#if defined(CONFIG_IDF_TARGET_ESP32C3)
constexpr bool skipSecondFFT = true;
#else
constexpr bool skipSecondFFT = false;
#endif
// High-Pass "DC blocker" filter
// see https://www.dsprelated.com/freebooks/filters/DC_Blocker.html
static void runDCBlocker(uint_fast16_t numSamples, float *sampleBuffer) {
constexpr float filterR = 0.990f; // around 40hz
static float xm1 = 0.0f;
static SR_HIRES_TYPE ym1 = 0.0f;
for (unsigned i=0; i < numSamples; i++) {
float value = sampleBuffer[i];
SR_HIRES_TYPE filtered = (SR_HIRES_TYPE)(value-xm1) + filterR*ym1;
xm1 = value;
ym1 = filtered;
sampleBuffer[i] = filtered;
}
}
//
// FFT main task
//
void FFTcode(void * parameter)
{
#ifdef SR_DEBUG
USER_FLUSH();
USER_PRINT("AR: "); USER_PRINT(pcTaskGetTaskName(NULL));
USER_PRINT(" task started on core "); USER_PRINT(xPortGetCoreID()); // causes trouble on -S2
USER_PRINT(" [prio="); USER_PRINT(uxTaskPriorityGet(NULL));
USER_PRINT(", min free stack="); USER_PRINT(uxTaskGetStackHighWaterMark(NULL));
USER_PRINTLN("]"); USER_FLUSH();
#endif
// see https://www.freertos.org/vtaskdelayuntil.html
const TickType_t xFrequency = FFT_MIN_CYCLE * portTICK_PERIOD_MS;
const TickType_t xFrequencyDouble = FFT_MIN_CYCLE * portTICK_PERIOD_MS * 2;
static bool isFirstRun = false;
#ifdef FFT_MAJORPEAK_HUMAN_EAR
// pre-compute pink noise scaling table
for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++) {
float binFreq = binInd * binWidth + binWidth/2.0f;
if (binFreq > (SAMPLE_RATE * 0.42f))
binFreq = (SAMPLE_RATE * 0.42f) - 0.25 * (binFreq - (SAMPLE_RATE * 0.42f)); // supress noise and aliasing
pinkFactors[binInd] = sqrtf(binFreq) / pinkcenter;
}
pinkFactors[0] *= 0.5; // suppress 0-42hz bin
#endif
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)) {
isFirstRun = false;
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
continue;
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS)
uint64_t start = esp_timer_get_time();
bool haveDoneFFT = false; // indicates if second measurement (FFT time) is valid
static uint64_t lastCycleStart = 0;
static uint64_t lastLastTime = 0;
if ((lastCycleStart > 0) && (lastCycleStart < start)) { // filter out overflows
uint64_t taskTimeInMillis = ((start - lastCycleStart) +5ULL) / 10ULL; // "+5" to ensure proper rounding
fftTaskCycle = (((taskTimeInMillis + lastLastTime)/2) *4 + fftTaskCycle*6)/10.0; // smart smooth
lastLastTime = taskTimeInMillis;
}
lastCycleStart = start;
#endif
// get a fresh batch of samples from I2S
if (audioSource) audioSource->getSamples(vReal, samplesFFT);
#if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS)
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)/10.0; // smooth
}
start = esp_timer_get_time(); // start measuring FFT time
#endif
xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay
isFirstRun = !isFirstRun; // toggle throtte
#ifdef MIC_LOGGER
float datMin = 0.0f;
float datMax = 0.0f;
double datAvg = 0.0f;
for (int i=0; i < samplesFFT; i++) {
if (i==0) {
datMin = datMax = vReal[i];
} else {
if (datMin > vReal[i]) datMin = vReal[i];
if (datMax < vReal[i]) datMax = vReal[i];
}
datAvg += vReal[i];
}
#endif
// band pass filter - can reduce noise floor by a factor of 50
// downside: frequencies below 100Hz will be ignored
if ((useInputFilter > 0) && (useInputFilter < 99)) {
switch(useInputFilter) {
case 1: runMicFilter(samplesFFT, vReal); break; // PDM microphone bandpass
case 2: runDCBlocker(samplesFFT, vReal); break; // generic Low-Cut + DC blocker (~40hz cut-off)
}
}
// 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 MIC_LOGGER
micReal_min = datMin;
micReal_max = datMax;
micReal_avg = datAvg / samplesFFT;
#if 0
// compute mix/max again after filering - usefull for filter debugging
for (int i=0; i < samplesFFT; i++) {
if (i==0) {
datMin = datMax = vReal[i];
} else {
if (datMin > vReal[i]) datMin = vReal[i];
if (datMax < vReal[i]) datMax = vReal[i];
}
}
micReal_min2 = datMin;
micReal_max2 = datMax;
#endif
#endif
// run FFT (takes 3-5ms on ESP32)
//if (fabsf(sampleAvg) > 0.25f) { // noise gate open
if (fabsf(volumeSmth) > 0.25f) { // noise gate open
if ((skipSecondFFT == false) || (isFirstRun == true)) {
// run FFT (takes 2-3ms on ESP32, ~12ms on ESP32-S2, ~30ms on -C3)
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
FFT.dcRemoval(); // remove DC offset
#if !defined(FFT_PREFER_EXACT_PEAKS)
FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy
#else
FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection
#endif
FFT.compute( FFTDirection::Forward ); // Compute FFT
FFT.complexToMagnitude(); // Compute magnitudes
#else
FFT.DCRemoval(); // let FFT lib remove DC component, so we don't need to care about this in getSamples()
//FFT.Windowing( FFT_WIN_TYP_HAMMING, FFT_FORWARD ); // Weigh data - standard Hamming window
//FFT.Windowing( FFT_WIN_TYP_BLACKMAN, FFT_FORWARD ); // Blackman window - better side freq rejection
#if !defined(FFT_PREFER_EXACT_PEAKS)
FFT.Windowing( FFT_WIN_TYP_FLT_TOP, FFT_FORWARD ); // Flat Top Window - better amplitude accuracy
#else
FFT.Windowing( FFT_WIN_TYP_BLACKMAN_HARRIS, FFT_FORWARD );// Blackman-Harris - excellent sideband rejection
#endif
FFT.Compute( FFT_FORWARD ); // Compute FFT
FFT.ComplexToMagnitude(); // Compute magnitudes
#endif
#ifdef FFT_MAJORPEAK_HUMAN_EAR
// scale FFT results
for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++)
vReal[binInd] *= pinkFactors[binInd];
#endif
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant
#else
FFT.MajorPeak(&FFT_MajorPeak, &FFT_Magnitude); // let the effects know which freq was most dominant
#endif
#ifdef FFT_MAJORPEAK_HUMAN_EAR
// undo scaling - we want unmodified values for FFTResult[] computations
for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++)
vReal[binInd] *= 1.0f/pinkFactors[binInd];
//fix peak magnitude
if ((FFT_MajorPeak > (binWidth/1.25f)) && (FFT_MajorPeak < (SAMPLE_RATE/2.2f)) && (FFT_Magnitude > 4.0f)) {
unsigned peakBin = constrain((int)((FFT_MajorPeak + binWidth/2.0f) / binWidth), 0, samplesFFT -1);
FFT_Magnitude *= fmaxf(1.0f/pinkFactors[peakBin], 1.0f);
}
#endif
FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
FFT_MajPeakSmth = FFT_MajPeakSmth + 0.42 * (FFT_MajorPeak - FFT_MajPeakSmth); // I like this "swooping peak" look
} else { // skip second run --> clear fft results, keep peaks
memset(vReal, 0, sizeof(vReal));
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS)
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;
}
if ((skipSecondFFT == false) || (isFirstRun == true)) {
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) > 0.25f) { // noise gate open
if (fabsf(volumeSmth) > 0.25f) { // 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 detetermine 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
//WLEDMM: different distributions
if (freqDist == 0) {
/* new mapping, optimized for 22050 Hz by softhack007 --- update: removed overlap */
// bins frequency range
if (useInputFilter==1) {
// skip frequencies below 100hz
fftCalc[ 0] = 0.8f * fftAddAvg(3,3);
fftCalc[ 1] = 0.9f * fftAddAvg(4,4);
fftCalc[ 2] = fftAddAvg(5,5);
fftCalc[ 3] = fftAddAvg(6,6);
// 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(1,1); // 1 43 - 86 sub-bass
fftCalc[ 1] = fftAddAvg(2,2); // 1 86 - 129 bass
fftCalc[ 2] = fftAddAvg(3,4); // 2 129 - 216 bass
fftCalc[ 3] = fftAddAvg(5,6); // 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(7,9); // 3 301 - 430 midrange
fftCalc[ 5] = fftAddAvg(10,12); // 3 430 - 560 midrange
fftCalc[ 6] = fftAddAvg(13,18); // 5 560 - 818 midrange
fftCalc[ 7] = fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center !
fftCalc[ 8] = fftAddAvg(26,32); // 7 1120 - 1421 midrange
fftCalc[ 9] = fftAddAvg(33,43); // 9 1421 - 1895 midrange
fftCalc[10] = fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid
fftCalc[11] = fftAddAvg(56,69); // 14 2412 - 3015 high mid
fftCalc[12] = fftAddAvg(70,85); // 16 3015 - 3704 high mid
fftCalc[13] = fftAddAvg(86,103); // 18 3704 - 4479 high mid
fftCalc[14] = fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
}
else if (freqDist == 1) { //WLEDMM: Rightshft: note ewowi: frequencies in comments are not correct
if (useInputFilter==1) {
// skip frequencies below 100hz
fftCalc[ 0] = 0.8f * fftAddAvg(1,1);
fftCalc[ 1] = 0.9f * fftAddAvg(2,2);
fftCalc[ 2] = fftAddAvg(3,3);
fftCalc[ 3] = fftAddAvg(4,4);
// 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(1,1); // 1 43 - 86 sub-bass
fftCalc[ 1] = fftAddAvg(2,2); // 1 86 - 129 bass
fftCalc[ 2] = fftAddAvg(3,3); // 2 129 - 216 bass
fftCalc[ 3] = fftAddAvg(4,4); // 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(5,6); // 3 301 - 430 midrange
fftCalc[ 5] = fftAddAvg(7,8); // 3 430 - 560 midrange
fftCalc[ 6] = fftAddAvg(9,10); // 5 560 - 818 midrange
fftCalc[ 7] = fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center !
fftCalc[ 8] = fftAddAvg(14,18); // 7 1120 - 1421 midrange
fftCalc[ 9] = fftAddAvg(19,25); // 9 1421 - 1895 midrange
fftCalc[10] = fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid
fftCalc[11] = fftAddAvg(37,45); // 14 2412 - 3015 high mid
fftCalc[12] = fftAddAvg(46,66); // 16 3015 - 3704 high mid
fftCalc[13] = fftAddAvg(67,97); // 18 3704 - 4479 high mid
fftCalc[14] = fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
}
#endif
} else { // noise gate closed - just decay old values
isFirstRun = false;
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;
}
}
memcpy(lastFftCalc, fftCalc, sizeof(lastFftCalc)); // make a backup of last "good" channels
} else { // if second run skipped
memcpy(fftCalc, lastFftCalc, sizeof(fftCalc)); // restore last "good" channels
}
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK;
//postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS);
postProcessFFTResults((fabsf(volumeSmth)>0.25f)? true : false , NUM_GEQ_CHANNELS); // this function modifies fftCalc, fftAvg and fftResult
#if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS)
static uint64_t lastLastFFT = 0;
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 + lastLastFFT)/2) *3 + fftTime*7)/10.0; // smart smooth
lastLastFFT = fftTimeInMillis;
}
#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
{
if ((skipSecondFFT == false) || (fabsf(volumeSmth) < 0.25f)) {
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
} else if (isFirstRun == true) {
vTaskDelayUntil( &xLastWakeTime, xFrequencyDouble); // release CPU after performing FFT in "skip second run" mode
}
}
} // 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) / 2.0;
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]; // spcial 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[pinkIndex][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] *0.78f + 0.22f*fftAvg[i]; // will need approx 1-2 cycles (50ms) for converging against fftCalc[i]
else { // fall slow
if (decayTime < 250) fftAvg[i] = fftCalc[i]*0.4f + 0.6f*fftAvg[i];
else if (decayTime < 500) fftAvg[i] = fftCalc[i]*0.33f + 0.67f*fftAvg[i];
else 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 if (decayTime < 4000) fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i];
else fftAvg[i] = fftCalc[i]*0.05f + 0.95f*fftAvg[i];
}
// 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 *= 0.42; // 42 is the answer ;-)
currentResult -= 8.0; // this skips the lowest row, giving some room for peaks
if (currentResult > 1.0) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
else currentResult = 0.0; // 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 -= 2.0; // 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 *= 0.34f; //experiment
currentResult -= 6.0f;
if (currentResult > 1.0) currentResult = sqrtf(currentResult);
else currentResult = 0.0; // special handling, because sqrt(0) = undefined
currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
//currentResult *= 0.80f + (float(i)/5.6f); //experiment
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 -= 2; // 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;
#if 1
// softhack007: this code continuously triggers while volume in the selected bin is above a certain threshold. So it does not detect peaks - it detects volume 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;
}
#endif
#if 0
// alternate detection, based on FFT_MajorPeak and FFT_Magnitude. Not much better...
if ((binNum > 1) && (maxVol > 8) && (binNum < 10) && (sampleAgc > 127) &&
(FFT_MajorPeak > 50) && (FFT_MajorPeak < 250) && (FFT_Magnitude > (16.0f * (maxVol+42.0)) /*my_magnitude > 136.0f*16.0f*/) &&
(millis() - timeOfPeak > 80)) {
havePeak = true;
}
#endif
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 ES7243_SDAPIN
int8_t sdaPin = -1;
#else
int8_t sdaPin = ES7243_SDAPIN;
#endif
#ifndef ES7243_SCLPIN
int8_t sclPin = -1;
#else
int8_t sclPin = ES7243_SCLPIN;
#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
};
#define UDPSOUND_MAX_PACKET 96 // max packet size for audiosync, with a bit of "headroom"
// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
#ifdef SR_ENABLE_DEFAULT
bool enabled = true; // WLEDMM
#else
bool enabled = false;
#endif
bool initDone = false;
// 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 controler.
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
int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc
float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc
float soundPressure = 0; // Sound Pressure estimation, based on microphone raw readings. 0 ->5db, 255 ->105db
// 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 _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 UDP_SYNC_HEADER[];
static const char UDP_SYNC_HEADER_v1[];
// private methods
////////////////////
// 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("volumeSmth:"); PLOT_PRINT(volumeSmth + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines
//PLOT_PRINT("volumeRaw:"); PLOT_PRINT(volumeRaw + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines
//PLOT_PRINT("samplePeak:"); PLOT_PRINT((samplePeak!=0) ? 128:0); PLOT_PRINT("\t");
#ifdef ARDUINO_ARCH_ESP32
PLOT_PRINT("micMin:"); PLOT_PRINT(0.5f * micReal_min); PLOT_PRINT("\t"); // scaled down to 50%, for better readability
PLOT_PRINT("micMax:"); PLOT_PRINT(0.5f * micReal_max); PLOT_PRINT("\t"); // scaled down to 50%
//PLOT_PRINT("micAvg:"); PLOT_PRINT(0.5f * micReal_avg); PLOT_PRINT("\t"); // scaled down to 50%
//PLOT_PRINT("micDC:"); PLOT_PRINT(0.5f * (micReal_min + micReal_max)/2.0f);PLOT_PRINT("\t"); // scaled down to 50%
PLOT_PRINT("micReal:"); PLOT_PRINT(micDataReal + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines
PLOT_PRINT("DC_Level:"); PLOT_PRINT(micLev + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines
// //PLOT_PRINT("filtmicMin:"); PLOT_PRINT(0.5f * micReal_min2); PLOT_PRINT("\t"); // scaled down to 50%
// //PLOT_PRINT("filtmicMax:"); PLOT_PRINT(0.5f * micReal_max2); PLOT_PRINT("\t"); // scaled down to 50%
//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");
//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("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t");
#endif
PLOT_PRINTLN();
PLOT_FLUSH();
#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 zome (<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 muliplier 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 < 1.0f)) {
// 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 = intgration 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 emergy 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 controler 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 calculatioins.
#ifdef WLEDMM_FASTPATH
constexpr float weighting = 0.35f; // slightly reduced filter strength, to reduce audio latency
constexpr float weighting2 = 0.25f;
#else
const float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
const float weighting2 = 0.073f; // Exponential filter weighting, for rising signal (a bit more robust against spikes)
#endif
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
static bool isFrozen = false;
static bool haveSilence = true;
static unsigned long lastSoundTime = 0; // for delaying un-freeze
static unsigned long startuptime = 0; // "fast freeze" mode: do not interfere during first 12 seconds (filter startup time)
#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
if (startuptime == 0) startuptime = millis(); // fast freeze mode - remember filter startup time
if ((micLevelMethod < 1) || !isFrozen) { // following the input level, UNLESS mic Level was frozen
micLev += (micDataReal-micLev) / 12288.0f;
}
if(micDataReal < (micLev-0.24)) { // MicLev above input signal:
micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // always align MicLev to lowest input signal
if (!haveSilence) isFrozen = true; // freeze mode: freeze micLevel so it cannot rise again
}
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);
if ((micInNoDC > expAdjF) && (expAdjF > soundSquelch)) // MicIn rising, and above squelch threshold?
expAdjF = (weighting2 * micInNoDC + (1.0f-weighting2) * expAdjF); // rise slower
else
expAdjF = (weighting * micInNoDC + (1.0f-weighting) * expAdjF); // fall faster
expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
if ((micLevelMethod == 2) && !haveSilence && (expAdjF >= (1.5f * float(soundSquelch))))
isFrozen = true; // fast freeze mode: freeze micLevel once the volume rises 50% above squelch
//expAdjF = (micInNoDC <= soundSquelch) ? 0: expAdjF; // simple noise gate - experimental
expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
if (expAdjF <= 0.5f)
haveSilence = true;
else {
lastSoundTime = millis();
haveSilence = false;
}
// un-freeze micLev
if (micLevelMethod == 0) isFrozen = false;
if ((micLevelMethod == 1) && isFrozen && haveSilence && ((millis() - lastSoundTime) > 4000)) isFrozen = false; // normal freeze: 4 seconds silence needed
if ((micLevelMethod == 2) && isFrozen && haveSilence && ((millis() - lastSoundTime) > 6000)) isFrozen = false; // fast freeze: 6 seconds silence needed
if ((micLevelMethod == 2) && (millis() - startuptime < 12000)) isFrozen = false; // fast freeze: no freeze in first 12 seconds (filter startup phase)
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
#if 1
// 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;
}
#endif
} 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()
// current sensitivity, based on AGC gain (multAgc)
float getSensitivity()
{
// start with AGC gain factor
float tmpSound = multAgc;
// experimental: this gives you a calculated "real gain"
// if ((sampleAvg> 1.0) && (sampleReal > 0.05)) tmpSound = (float)sampleRaw / sampleReal; // calculate gain from sampleReal
// else tmpSound = ((float)sampleGain/40.0f * (float)inputLevel/128.0f) + 1.0f/16.0f; // silence --> use values from user settings
if (soundAgc == 0)
tmpSound = ((float)sampleGain/40.0f * (float)inputLevel/128.0f) + 1.0f/16.0f; // AGC off -> use non-AGC gain from presets
else
tmpSound /= (float)sampleGain/40.0f + 1.0f/16.0f; // AGC ON -> scale value so 1 = middle value
// scale to 0..255. Actually I'm not absolutely happy with this, but it works
if (tmpSound > 1.0) tmpSound = sqrtf(tmpSound);
if (tmpSound > 1.25) tmpSound = ((tmpSound-1.25f)/3.42f) +1.25f;
// we have a value now that should be between 0 and 4 (representing gain 1/16 ... 16.0)
return fminf(fmaxf(128.0*tmpSound -6.0f, 0), 255.0); // return scaled non-inverted value // "-6" to ignore values below 1/24
}
// estimate sound pressure, based on some assumptions :
// * sample max = 32676 -> Acoustic overload point --> 105db ==> 255
// * sample < squelch -> just above hearing level --> 5db ==> 0
// see https://en.wikipedia.org/wiki/Sound_pressure#Examples_of_sound_pressure
// use with I2S digital microphones. Expect stupid values for analog in, and with Line-In !!
float estimatePressure() {
// some constants
constexpr float logMinSample = 0.8329091229351f; // ln(2.3)
constexpr float sampleMin = 2.3f;
constexpr float logMaxSample = 10.1895683436f; // ln(32767 - 6144)
constexpr float sampleMax = 32767.0f - 6144.0f;
// take the max sample from last I2S batch.
float micSampleMax = fabsf(sampleReal); // from getSample() - nice results, however a bit distorted by MicLev processing
//float micSampleMax = fabsf(micDataReal); // from FFTCode() - better source, but more flickering
if (dmType == 0) micSampleMax *= 2.0f; // correction for ADC analog
//if (dmType == 4) micSampleMax *= 16.0f; // correction for I2S Line-In
if (dmType == 5) micSampleMax *= 2.0f; // correction for PDM
if (dmType == 4) { // I2S Line-In. This is a dirty trick to make sound pressure look interesting for line-in (which doesn't have "sound pressure" as its not a microphone)
micSampleMax /= 11.0f; // reduce to max 128
micSampleMax *= micSampleMax; // blow up --> max 16000
}
// make sure we are in expected ranges
if(micSampleMax <= sampleMin) return 0.0f;
if(micSampleMax >= sampleMax) return 255.0f;
// apply logarithmic scaling
float scaledvalue = logf(micSampleMax);
scaledvalue = (scaledvalue - logMinSample) / (logMaxSample - logMinSample); // 0...1
return fminf(fmaxf(256.0*scaledvalue, 0), 255.0); // scaled value
}
#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 (!(apActive || WLED_CONNECTED || interfacesInited)) {
if (udpSyncConnected) {
udpSyncConnected = false;
fftUdp.stop();
receivedFormat = 0;
DEBUGSR_PRINTLN(F("AR connectUDPSoundSync(): connection lost, UDP closed."));
}
return; // neither AP nor other connections availeable
}
if (udpSyncConnected) return; // already connected
if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds
if (updateIsRunning) return; // don't reconect during OTA
// 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;
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;
for (int i = 0; i < NUM_GEQ_CHANNELS; i++) {
transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254);
}
transmitData.FFT_Magnitude = my_magnitude;
transmitData.FFT_MajorPeak = FFT_MajorPeak;
if (fftUdp.beginMulticastPacket() != 0) { // beginMulticastPacket returns 0 in case of error
fftUdp.write(reinterpret_cast<uint8_t *>(&transmitData), sizeof(transmitData));
fftUdp.endPacket();
}
return;
} // transmitAudioData()
#endif
static bool isValidUdpSyncVersion(const char *header) {
return strncmp_P(header, UDP_SYNC_HEADER, 6) == 0;
}
static bool isValidUdpSyncVersion_v1(const char *header) {
return strncmp_P(header, UDP_SYNC_HEADER_v1, 6) == 0;
}
void decodeAudioData(int packetSize, uint8_t *fftBuff) {
audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff);
// update samples for effects
volumeSmth = fmaxf(receivedPacket->sampleSmth, 0.0f);
volumeRaw = fmaxf(receivedPacket->sampleRaw, 0.0f);
#ifdef ARDUINO_ARCH_ESP32
// update internal samples
sampleRaw = volumeRaw;
sampleAvg = volumeSmth;
rawSampleAgc = volumeRaw;
sampleAgc = volumeSmth;
multAgc = 1.0f;
#endif
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
autoResetPeak();
if (!samplePeak) {
samplePeak = receivedPacket->samplePeak >0 ? true:false;
if (samplePeak) timeOfPeak = millis();
//userVar1 = samplePeak;
}
//These values are only computed by ESP32
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;
FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
}
void decodeAudioData_v1(int packetSize, uint8_t *fftBuff) {
audioSyncPacket_v1 *receivedPacket = reinterpret_cast<audioSyncPacket_v1*>(fftBuff);
// update samples for effects
volumeSmth = fmaxf(receivedPacket->sampleAgc, 0.0f);
volumeRaw = volumeSmth; // V1 format does not have "raw" AGC sample
#ifdef ARDUINO_ARCH_ESP32
// update internal samples
sampleRaw = fmaxf(receivedPacket->sampleRaw, 0.0f);
sampleAvg = fmaxf(receivedPacket->sampleAvg, 0.0f);;
sampleAgc = volumeSmth;
rawSampleAgc = volumeRaw;
multAgc = 1.0f;
#endif
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
autoResetPeak();
if (!samplePeak) {
samplePeak = receivedPacket->samplePeak >0 ? true:false;
if (samplePeak) timeOfPeak = millis();
//userVar1 = samplePeak;
}
//These values are only available on the ESP32
for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i];
my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0);
FFT_Magnitude = my_magnitude;
FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0, 11025.0); // restrict value to range expected by effects
}
bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received.
{
if (!udpSyncConnected) return false;
bool haveFreshData = false;
size_t packetSize = 0;
// WLEDMM use exception handler to catch out-of-memory errors
#if __cpp_exceptions
try{
packetSize = fftUdp.parsePacket();
} catch(...) {
packetSize = 0; // low heap memory -> discard packet.
#ifdef ARDUINO_ARCH_ESP32
fftUdp.flush(); // this does not work on 8266
#endif
DEBUG_PRINTLN(F("receiveAudioData: parsePacket out of memory exception caught!"));
USER_FLUSH();
}
#else
packetSize = fftUdp.parsePacket();
#endif
#ifdef ARDUINO_ARCH_ESP32
if ((packetSize > 0) && ((packetSize < 5) || (packetSize > UDPSOUND_MAX_PACKET))) fftUdp.flush(); // discard invalid packets (too small or too big)
#endif
if ((packetSize > 5) && (packetSize <= UDPSOUND_MAX_PACKET)) {
static uint8_t fftUdpBuffer[UDPSOUND_MAX_PACKET+1] = { 0 }; // static buffer for receiving, to reuse the same memory and avoid heap fragmentation
//DEBUGSR_PRINTLN("Received UDP Sync Packet");
fftUdp.read(fftUdpBuffer, packetSize);
// VERIFY THAT THIS IS A COMPATIBLE PACKET
if (packetSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftUdpBuffer))) {
decodeAudioData(packetSize, fftUdpBuffer);
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet v2");
haveFreshData = true;
receivedFormat = 2;
} else {
if (packetSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftUdpBuffer))) {
decodeAudioData_v1(packetSize, fftUdpBuffer);
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet v1");
haveFreshData = true;
receivedFormat = 1;
} else receivedFormat = 0; // unknown format
}
}
return haveFreshData;
}
//////////////////////
// usermod functions//
//////////////////////
public:
//Functions called by WLED or other usermods
/*
* setup() is called once at boot. WiFi is not yet connected at this point.
* You can use it to initialize variables, sensors or similar.
* It is called *AFTER* readFromConfig()
*/
void setup()
{
disableSoundProcessing = true; // just to be sure
if (!initDone) {
// usermod exchangeable data
// we will assign all usermod exportable data here as pointers to original variables or arrays and allocate memory for pointers
um_data = new um_data_t;
um_data->u_size = 11;
um_data->u_type = new um_types_t[um_data->u_size];
um_data->u_data = new void*[um_data->u_size];
um_data->u_data[0] = &volumeSmth; //*used (New)
um_data->u_type[0] = UMT_FLOAT;
um_data->u_data[1] = &volumeRaw; // used (New)
um_data->u_type[1] = UMT_UINT16;
um_data->u_data[2] = fftResult; //*used (Blurz, DJ Light, Noisemove, GEQ_base, 2D Funky Plank, Akemi)
um_data->u_type[2] = UMT_BYTE_ARR;
um_data->u_data[3] = &samplePeak; //*used (Puddlepeak, Ripplepeak, Waterfall)
um_data->u_type[3] = UMT_BYTE;
um_data->u_data[4] = &FFT_MajorPeak; //*used (Ripplepeak, Freqmap, Freqmatrix, Freqpixels, Freqwave, Gravfreq, Rocktaves, Waterfall)
um_data->u_type[4] = UMT_FLOAT;
um_data->u_data[5] = &my_magnitude; // used (New)
um_data->u_type[5] = UMT_FLOAT;
#ifdef ARDUINO_ARCH_ESP32
um_data->u_data[6] = &maxVol; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall)
um_data->u_type[6] = UMT_BYTE;
um_data->u_data[7] = &binNum; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall)
um_data->u_type[7] = UMT_BYTE;
um_data->u_data[8] = &FFT_MajPeakSmth; // new
um_data->u_type[8] = UMT_FLOAT;
um_data->u_data[9] = &soundPressure; // used (New)
um_data->u_type[9] = UMT_FLOAT;
um_data->u_data[10] = &agcSensitivity; // used (New)
um_data->u_type[10] = UMT_FLOAT;
#else
// ESP8266
// See https://github.com/MoonModules/WLED/pull/60#issuecomment-1666972133 for explaination of these alternative sources of data
um_data->u_data[6] = &maxVol; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall)
um_data->u_type[6] = UMT_BYTE;
um_data->u_data[7] = &binNum; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall)
um_data->u_type[7] = UMT_BYTE;
um_data->u_data[8] = &FFT_MajorPeak; // new - substitute for FFT_MajPeakSmth
um_data->u_type[8] = UMT_FLOAT;
um_data->u_data[9] = &volumeSmth; // used (New) - substitute for soundPressure
um_data->u_type[9] = UMT_FLOAT;
um_data->u_data[10] = &agcSensitivity; // used (New) - dummy value (128 => 50%)
um_data->u_type[10] = UMT_FLOAT;
#endif
}
#ifdef ARDUINO_ARCH_ESP32
// Reset I2S peripheral for good measure
i2s_driver_uninstall(I2S_NUM_0); // E (696) I2S: i2s_driver_uninstall(2006): I2S port 0 has not installed
#if !defined(CONFIG_IDF_TARGET_ESP32C3)
delay(100);
periph_module_reset(PERIPH_I2S0_MODULE); // not possible on -C3
#endif
delay(100); // Give that poor microphone some time to setup.
useInputFilter = 2; // default: DC blocker
switch (dmType) {
#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) || defined(CONFIG_IDF_TARGET_ESP32S3)
// stub cases for not-yet-supported I2S modes on other ESP32 chips
case 0: //ADC analog
#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3)
case 5: //PDM Microphone
case 51: //legacy PDM Microphone
#endif
#endif
case 1:
DEBUGSR_PRINT(F("AR: Generic I2S Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT));
audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE);
delay(100);
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin);
break;
case 2:
DEBUGSR_PRINTLN(F("AR: ES7243 Microphone (right channel only)."));
//useInputFilter = 0; // in case you need to disable low-cut software filtering
audioSource = new ES7243(SAMPLE_RATE, BLOCK_SIZE);
delay(100);
if (audioSource) audioSource->initialize(sdaPin, sclPin, i2swsPin, i2ssdPin, i2sckPin, mclkPin);
break;
case 3:
DEBUGSR_PRINT(F("AR: SPH0645 Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT));
audioSource = new SPH0654(SAMPLE_RATE, BLOCK_SIZE);
delay(100);
audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin);
break;
case 4:
DEBUGSR_PRINT(F("AR: Generic I2S Microphone with Master Clock - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT));
audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f/24.0f);
//audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f/24.0f, false); // I2S SLAVE mode - does not work, unfortunately
delay(100);
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin, mclkPin);
break;
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
case 5:
DEBUGSR_PRINT(F("AR: I2S PDM Microphone - ")); DEBUGSR_PRINTLN(F(I2S_PDM_MIC_CHANNEL_TEXT));
audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f/4.0f);
useInputFilter = 1; // PDM bandpass filter - this reduces the noise floor on SPM1423 from 5% Vpp (~380) down to 0.05% Vpp (~5)
delay(100);
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin);
break;
case 51:
DEBUGSR_PRINT(F("AR: Legacy PDM Microphone - ")); DEBUGSR_PRINTLN(F(I2S_PDM_MIC_CHANNEL_TEXT));
audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f);
useInputFilter = 1; // PDM bandpass filter
delay(100);
if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin);
break;
#endif
case 6:
DEBUGSR_PRINTLN(F("AR: ES8388 Source"));
audioSource = new ES8388Source(SAMPLE_RATE, BLOCK_SIZE, 1.0f);
//useInputFilter = 0; // to disable low-cut software filtering and restore previous behaviour
delay(100);
if (audioSource) audioSource->initialize(sdaPin, sclPin, i2swsPin, i2ssdPin, i2sckPin, mclkPin);
break;
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
// ADC over I2S is only possible on "classic" ESP32
case 0:
default:
DEBUGSR_PRINTLN(F("AR: Analog Microphone (left channel only)."));
useInputFilter = 1; // PDM bandpass filter seems to work well for analog, too
audioSource = new I2SAdcSource(SAMPLE_RATE, BLOCK_SIZE);
delay(100);
if (audioSource) audioSource->initialize(audioPin);
break;
#endif
}
delay(250); // give microphone enough time to initialise
if (!audioSource) enabled = false; // audio failed to initialise
#endif
if (enabled) onUpdateBegin(false); // create FFT task, and initailize network
#ifdef ARDUINO_ARCH_ESP32
if (FFT_Task == nullptr) enabled = false; // FFT task creation failed
if((!audioSource) || (!audioSource->isInitialized())) { // audio source failed to initialize. Still stay "enabled", as there might be input arriving via UDP Sound Sync
#ifdef WLED_DEBUG
DEBUG_PRINTLN(F("AR: Failed to initialize sound input driver. Please check input PIN settings."));
#else
USER_PRINTLN(F("AR: Failed to initialize sound input driver. Please check input PIN settings."));
#endif
disableSoundProcessing = true;
} else {
USER_PRINTLN(F("AR: sound input driver initialized successfully."));
}
#endif
if (enabled) disableSoundProcessing = false; // all good - enable audio processing
// try to start UDP
last_UDPTime = 0;
receivedFormat = 0;
delay(100);
if (enabled) connectUDPSoundSync();
initDone = true;
DEBUGSR_PRINT(F("AR: init done, enabled = "));
DEBUGSR_PRINTLN(enabled ? F("true.") : F("false."));
USER_FLUSH();
}
/*
* connected() is called every time the WiFi is (re)connected
* Use it to initialize network interfaces
*/
void connected()
{
if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection
udpSyncConnected = false;
fftUdp.stop();
receivedFormat = 0;
DEBUGSR_PRINTLN(F("AR connected(): old UDP connection closed."));
}
if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) {
#ifdef ARDUINO_ARCH_ESP32
udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort);
#else
udpSyncConnected = fftUdp.beginMulticast(WiFi.localIP(), IPAddress(239, 0, 0, 1), audioSyncPort);
#endif
receivedFormat = 0;
if (udpSyncConnected) last_UDPTime = millis();
if (apActive && !(WLED_CONNECTED)) {
DEBUGSR_PRINTLN(udpSyncConnected ? F("AR connected(): UDP: connected using AP.") : F("AR connected(): UDP is disconnected (AP)."));
} else {
DEBUGSR_PRINTLN(udpSyncConnected ? F("AR connected(): UDP: connected to WIFI.") : F("AR connected(): UDP is disconnected (Wifi)."));
}
}
}
/*
* loop() is called continuously. Here you can check for events, read sensors, etc.
*
* Tips:
* 1. You can use "if (WLED_CONNECTED)" to check for a successful network connection.
* Additionally, "if (WLED_MQTT_CONNECTED)" is available to check for a connection to an MQTT broker.
*
* 2. Try to avoid using the delay() function. NEVER use delays longer than 10 milliseconds.
* Instead, use a timer check as shown here.
*/
void loop()
{
static unsigned long lastUMRun = millis();
if (!enabled) {
disableSoundProcessing = true; // keep processing suspended (FFT task)
lastUMRun = millis(); // update time keeping
return;
}
// We cannot wait indefinitely before processing audio data
if (strip.isUpdating() && (millis() - lastUMRun < 2)) return; // be nice, but not too nice
// suspend local sound processing when "real time mode" is active (E131, UDP, ADALIGHT, ARTNET)
if ( (realtimeOverride == REALTIME_OVERRIDE_NONE) // please add other overrides here if needed
&&( (realtimeMode == REALTIME_MODE_GENERIC)
||(realtimeMode == REALTIME_MODE_E131)
||(realtimeMode == REALTIME_MODE_UDP)
||(realtimeMode == REALTIME_MODE_ADALIGHT)
||(realtimeMode == REALTIME_MODE_ARTNET) ) ) // please add other modes here if needed
{
#ifdef WLED_DEBUG
if ((disableSoundProcessing == false) && (audioSyncEnabled == 0)) { // we just switched to "disabled"
DEBUG_PRINTLN("[AR userLoop] realtime mode active - audio processing suspended.");
DEBUG_PRINTF( " RealtimeMode = %d; RealtimeOverride = %d\n", int(realtimeMode), int(realtimeOverride));
}
#endif
disableSoundProcessing = true;
} else {
#if defined(ARDUINO_ARCH_ESP32) && defined(WLED_DEBUG)
if ((disableSoundProcessing == true) && (audioSyncEnabled == 0) && audioSource->isInitialized()) { // we just switched to "enabled"
DEBUG_PRINTLN("[AR userLoop] realtime mode ended - audio processing resumed.");
DEBUG_PRINTF( " RealtimeMode = %d; RealtimeOverride = %d\n", int(realtimeMode), int(realtimeOverride));
}
#endif
if ((disableSoundProcessing == true) && (audioSyncEnabled == 0)) lastUMRun = millis(); // just left "realtime mode" - update timekeeping
disableSoundProcessing = false;
}
if (audioSyncEnabled & 0x02) disableSoundProcessing = true; // make sure everything is disabled IF in audio Receive mode
if (audioSyncEnabled & 0x01) disableSoundProcessing = false; // keep running audio IF we're in audio Transmit mode
#ifdef ARDUINO_ARCH_ESP32
if (!audioSource->isInitialized()) disableSoundProcessing = true; // no audio source
// Only run the sampling code IF we're not in Receive mode or realtime mode
if (!(audioSyncEnabled & 0x02) && !disableSoundProcessing) {
if (soundAgc > AGC_NUM_PRESETS) soundAgc = 0; // make sure that AGC preset is valid (to avoid array bounds violation)
unsigned long t_now = millis(); // remember current time
int userloopDelay = int(t_now - lastUMRun);
if (lastUMRun == 0) userloopDelay=0; // startup - don't have valid data from last run.
#if defined(SR_DEBUG)
// complain when audio userloop has been delayed for long time. Currently we need userloop running between 500 and 1500 times per second.
// softhack007 disabled temporarily - avoid serial console spam with MANY leds and low FPS
//if ((userloopDelay > /*23*/ 65) && !disableSoundProcessing && (audioSyncEnabled == 0)) {
//DEBUG_PRINTF("[AR userLoop] hickup detected -> was inactive for last %d millis!\n", userloopDelay);
//}
#endif
// run filters, and repeat in case of loop delays (hick-up compensation)
if (userloopDelay <2) userloopDelay = 0; // minor glitch, no problem
if (userloopDelay >200) userloopDelay = 200; // limit number of filter re-runs
do {
getSample(); // run microphone sampling filters
agcAvg(t_now - userloopDelay); // Calculated the PI adjusted value as sampleAvg
userloopDelay -= 2; // advance "simulated time" by 2ms
} while (userloopDelay > 0);
lastUMRun = t_now; // update time keeping
// update samples for effects (raw, smooth)
volumeSmth = (soundAgc) ? sampleAgc : sampleAvg;
volumeRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
// update FFTMagnitude, taking into account AGC amplification
my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects
if (soundAgc) my_magnitude *= multAgc;
if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
// get AGC sensitivity and sound pressure
static unsigned long lastEstimate = 0;
#ifdef WLEDMM_FASTPATH
if (millis() - lastEstimate > 7) {
#else
if (millis() - lastEstimate > 12) {
#endif
lastEstimate = millis();
agcSensitivity = getSensitivity();
if (limiterOn)
soundPressure = soundPressure + 0.38f * (estimatePressure() - soundPressure); // dynamics limiter on -> some smoothing
else
soundPressure = soundPressure + 0.95f * (estimatePressure() - soundPressure); // dynamics limiter on -> raw value
}
limitSampleDynamics();
} // if (!disableSoundProcessing)
#endif
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) {
// Only run the audio listener code if we're in Receive mode
static float syncVolumeSmth = 0;
bool have_new_sample = false;
if (millis() - lastTime > delayMs) {
have_new_sample = receiveAudioData();
if (have_new_sample) last_UDPTime = millis();
lastTime = millis();
} else {
#ifdef ARDUINO_ARCH_ESP32
fftUdp.flush(); // WLEDMM: Flush this if we haven't read it. Does not work on 8266.
#endif
}
if (have_new_sample) syncVolumeSmth = volumeSmth; // remember received sample
else volumeSmth = syncVolumeSmth; // restore originally received sample for next run of dynamics limiter
limitSampleDynamics(); // run dynamics limiter on received volumeSmth, to hide jumps and hickups
} else {
receivedFormat = 0;
}
if ( (audioSyncEnabled & 0x02) // receive mode
&& udpSyncConnected // connected
&& (receivedFormat > 0) // we actually received something in the past
&& ((millis() - last_UDPTime) > 25000)) { // close connection after 25sec idle
udpSyncConnected = false;
receivedFormat = 0;
fftUdp.stop();
volumeSmth =0.0f;
volumeRaw =0;
my_magnitude = 0.1; FFT_Magnitude = 0.01; FFT_MajorPeak = 2;
#ifdef ARDUINO_ARCH_ESP32
multAgc = 1;
#endif
DEBUGSR_PRINTLN(F("AR loop(): UDP closed due to inactivity."));
}
#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
static unsigned long lastMicLoggerTime = 0;
if (millis()-lastMicLoggerTime > 20) {
lastMicLoggerTime = millis();
logAudio();
}
#endif
// Info Page: keep max sample from last 5 seconds
#ifdef ARDUINO_ARCH_ESP32
if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) {
sampleMaxTimer = millis();
maxSample5sec = (0.15 * maxSample5sec) + 0.85 *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing
if (sampleAvg < 1) maxSample5sec = 0; // noise gate
} else {
if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume
}
#else // similar functionality for 8266 receive only - use VolumeSmth instead of raw sample data
if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) {
sampleMaxTimer = millis();
maxSample5sec = (0.15 * maxSample5sec) + 0.85 * volumeSmth; // reset, and start with some smoothing
if (volumeSmth < 1.0f) maxSample5sec = 0; // noise gate
if (maxSample5sec < 0.0f) maxSample5sec = 0; // avoid negative values
} else {
if (volumeSmth >= 1.0f) maxSample5sec = fmaxf(maxSample5sec, volumeRaw); // follow maximum volume
}
#endif
#ifdef ARDUINO_ARCH_ESP32
//UDP Microphone Sync - transmit mode
if ((audioSyncEnabled & 0x01) && (millis() - lastTime > 20)) {
// Only run the transmit code IF we're in Transmit mode
transmitAudioData();
lastTime = millis();
}
#endif
}
bool getUMData(um_data_t **data)
{
if (!data || !enabled) return false; // no pointer provided by caller or not enabled -> exit
*data = um_data;
return true;
}
#ifdef ARDUINO_ARCH_ESP32
void onUpdateBegin(bool init)
{
#ifdef WLED_DEBUG
fftTime = sampleTime = 0;
#endif
// gracefully suspend FFT task (if running)
disableSoundProcessing = true;
// reset sound data
micDataReal = 0.0f;
volumeRaw = 0; volumeSmth = 0;
sampleAgc = 0; sampleAvg = 0;
sampleRaw = 0; rawSampleAgc = 0;
my_magnitude = 0; FFT_Magnitude = 0; FFT_MajorPeak = 1;
multAgc = 1;
// reset FFT data
memset(fftCalc, 0, sizeof(fftCalc));
memset(fftAvg, 0, sizeof(fftAvg));
memset(fftResult, 0, sizeof(fftResult));
for(int i=(init?0:1); i<NUM_GEQ_CHANNELS; i+=2) fftResult[i] = 16; // make a tiny pattern
inputLevel = 128; // reset level slider to default
autoResetPeak();
if (init && FFT_Task) {
delay(25); // WLEDMM: givesome time for I2S driver to finish sampling
vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash
if (udpSyncConnected) { // close UDP sync connection (if open)
udpSyncConnected = false;
fftUdp.stop();
DEBUGSR_PRINTLN(F("AR onUpdateBegin(true): UDP connection closed."));
receivedFormat = 0;
}
} else {
// update has failed or create task requested
if (FFT_Task) {
vTaskResume(FFT_Task);
connected(); // resume UDP
} else
// xTaskCreatePinnedToCore(
// xTaskCreate( // no need to "pin" this task to core #0
xTaskCreateUniversal(
FFTcode, // Function to implement the task
"FFT", // Name of the task
5000, // Stack size in words
NULL, // Task input parameter
FFTTASK_PRIORITY, // Priority of the task
&FFT_Task // Task handle
, 0 // Core where the task should run
);
}
micDataReal = 0.0f; // just to be sure
if (enabled) disableSoundProcessing = false;
updateIsRunning = init;
}
#else // reduced function for 8266
void onUpdateBegin(bool init)
{
// gracefully suspend audio (if running)
disableSoundProcessing = true;
// reset sound data
volumeRaw = 0; volumeSmth = 0;
for(int i=(init?0:1); i<NUM_GEQ_CHANNELS; i+=2) fftResult[i] = 16; // make a tiny pattern
autoResetPeak();
if (init) {
if (udpSyncConnected) { // close UDP sync connection (if open)
udpSyncConnected = false;
fftUdp.stop();
DEBUGSR_PRINTLN(F("AR onUpdateBegin(true): UDP connection closed."));
receivedFormat = 0;
}
}
if (enabled) disableSoundProcessing = init; // init = true means that OTA is just starting --> don't process audio
updateIsRunning = init;
}
#endif
#ifdef ARDUINO_ARCH_ESP32
/**
* handleButton() can be used to override default button behaviour. Returning true
* will prevent button working in a default way.
*/
bool handleButton(uint8_t b) {
yield();
// crude way of determining if audio input is analog
// better would be for AudioSource to implement getType()
if (enabled
&& dmType == 0 && audioPin>=0
&& (buttonType[b] == BTN_TYPE_ANALOG || buttonType[b] == BTN_TYPE_ANALOG_INVERTED)
) {
return true;
}
return false;
}
#endif
////////////////////////////
// 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.
* Below it is shown how this could be used for e.g. a light sensor
*/
void addToJsonInfo(JsonObject& root)
{
#ifdef ARDUINO_ARCH_ESP32
char myStringBuffer[16]; // buffer for snprintf() - not used yet on 8266
#endif
JsonObject user = root["u"];
if (user.isNull()) user = root.createNestedObject("u");
JsonArray infoArr = user.createNestedArray(FPSTR(_name));
String uiDomString = F("<button class=\"btn btn-xs\" onclick=\"requestJson({");
uiDomString += FPSTR(_name);
uiDomString += F(":{");
uiDomString += FPSTR(_enabled);
uiDomString += enabled ? F(":false}});\">") : F(":true}});\">");
uiDomString += F("<i class=\"icons");
uiDomString += enabled ? F(" on") : F(" off");
uiDomString += F("\">&#xe08f;</i>");
uiDomString += F("</button>");
infoArr.add(uiDomString);
if (enabled) {
#ifdef ARDUINO_ARCH_ESP32
// Input Level Slider
if (disableSoundProcessing == false) { // only show slider when audio processing is running
if (soundAgc > 0) {
infoArr = user.createNestedArray(F("GEQ Input Level")); // if AGC is on, this slider only affects fftResult[] frequencies
// show slider value as a number
float post_gain = (float)inputLevel/128.0f;
if (post_gain < 1.0f) post_gain = ((post_gain -1.0f) * 0.8f) +1.0f;
post_gain = roundf(post_gain * 100.0f);
snprintf_P(myStringBuffer, 15, PSTR("%3.0f %%"), post_gain);
infoArr.add(myStringBuffer);
} else {
infoArr = user.createNestedArray(F("Audio Input Level"));
}
uiDomString = F("<div class=\"slider\"><div class=\"sliderwrap il\"><input class=\"noslide\" onchange=\"requestJson({");
uiDomString += FPSTR(_name);
uiDomString += F(":{");
uiDomString += FPSTR(_inputLvl);
uiDomString += F(":parseInt(this.value)}});\" oninput=\"updateTrail(this);\" max=255 min=0 type=\"range\" value=");
uiDomString += inputLevel;
uiDomString += F(" /><div class=\"sliderdisplay\"></div></div></div>"); //<output class=\"sliderbubble\"></output>
infoArr.add(uiDomString);
}
#endif
// The following can be used for troubleshooting user errors and is so not enclosed in #ifdef WLED_DEBUG
// current Audio input
infoArr = user.createNestedArray(F("Audio Source"));
if (audioSyncEnabled & 0x02) {
// UDP sound sync - receive mode
infoArr.add(F("UDP sound sync"));
if (udpSyncConnected) {
if (millis() - last_UDPTime < 2500)
infoArr.add(F(" - receiving"));
else
infoArr.add(F(" - idle"));
} else {
infoArr.add(F(" - no connection"));
}
#ifndef ARDUINO_ARCH_ESP32 // substitute for 8266
} else {
infoArr.add(F("sound sync Off"));
}
#else // ESP32 only
} else {
// Analog or I2S digital input
if (audioSource && (audioSource->isInitialized())) {
// audio source sucessfully configured
if (audioSource->getType() == AudioSource::Type_I2SAdc) {
infoArr.add(F("ADC analog"));
} else {
if (dmType != 51)
infoArr.add(F("I2S digital"));
else
infoArr.add(F("legacy I2S PDM"));
}
// input level or "silence"
if (maxSample5sec > 1.0) {
float my_usage = 100.0f * (maxSample5sec / 255.0f);
snprintf_P(myStringBuffer, 15, PSTR(" - peak %3d%%"), int(my_usage));
infoArr.add(myStringBuffer);
} else {
infoArr.add(F(" - quiet"));
}
} else {
// error during audio source setup
infoArr.add(F("not initialized"));
infoArr.add(F(" - check pin settings"));
}
}
// Sound processing (FFT and input filters)
infoArr = user.createNestedArray(F("Sound Processing"));
if (audioSource && (disableSoundProcessing == false)) {
infoArr.add(F("running"));
} else {
infoArr.add(F("suspended"));
}
// AGC or manual Gain
if ((soundAgc==0) && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
infoArr = user.createNestedArray(F("Manual Gain"));
float myGain = ((float)sampleGain/40.0f * (float)inputLevel/128.0f) + 1.0f/16.0f; // non-AGC gain from presets
infoArr.add(roundf(myGain*100.0f) / 100.0f);
infoArr.add("x");
}
if (soundAgc && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
infoArr = user.createNestedArray(F("AGC Gain"));
infoArr.add(roundf(multAgc*100.0f) / 100.0f);
infoArr.add("x");
}
#endif
// UDP Sound Sync status
infoArr = user.createNestedArray(F("UDP Sound Sync"));
if (audioSyncEnabled) {
if (audioSyncEnabled & 0x01) {
infoArr.add(F("send mode"));
if ((udpSyncConnected) && (millis() - lastTime < 2500)) infoArr.add(F(" v2"));
} else if (audioSyncEnabled & 0x02) {
infoArr.add(F("receive mode"));
}
} else
infoArr.add("off");
if (audioSyncEnabled && !udpSyncConnected) infoArr.add(" <i>(unconnected)</i>");
if (audioSyncEnabled && udpSyncConnected && (millis() - last_UDPTime < 2500)) {
if (receivedFormat == 1) infoArr.add(F(" v1"));
if (receivedFormat == 2) infoArr.add(F(" v2"));
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS)
#ifdef ARDUINO_ARCH_ESP32
infoArr = user.createNestedArray(F("I2S cycle time"));
infoArr.add(roundf(fftTaskCycle)/100.0f);
infoArr.add(" ms");
infoArr = user.createNestedArray(F("Sampling time"));
infoArr.add(roundf(sampleTime)/100.0f);
infoArr.add(" ms");
infoArr = user.createNestedArray(F("FFT time"));
infoArr.add(roundf(fftTime)/100.0f);
if ((fftTime/100) >= FFT_MIN_CYCLE) // FFT time over budget -> I2S buffer will overflow
infoArr.add("<b style=\"color:red;\">! ms</b>");
else if ((fftTime/80 + sampleTime/80) >= FFT_MIN_CYCLE) // FFT time >75% of budget -> risk of instability
infoArr.add("<b style=\"color:orange;\"> ms!</b>");
else
infoArr.add(" ms");
DEBUGSR_PRINTF("AR I2S cycle time: %5.2f ms\n", roundf(fftTaskCycle)/100.0f);
DEBUGSR_PRINTF("AR Sampling time : %5.2f ms\n", roundf(sampleTime)/100.0f);
DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", roundf(fftTime)/100.0f);
#endif
#endif
}
}
/*
* addToJsonState() can be used to add custom entries to the /json/state part of the JSON API (state object).
* Values in the state object may be modified by connected clients
*/
void addToJsonState(JsonObject& root)
{
if (!initDone) return; // prevent crash on boot applyPreset()
JsonObject usermod = root[FPSTR(_name)];
if (usermod.isNull()) {
usermod = root.createNestedObject(FPSTR(_name));
}
usermod["on"] = enabled;
}
/*
* readFromJsonState() can be used to receive data clients send to the /json/state part of the JSON API (state object).
* Values in the state object may be modified by connected clients
*/
void readFromJsonState(JsonObject& root)
{
if (!initDone) return; // prevent crash on boot applyPreset()
bool prevEnabled = enabled;
JsonObject usermod = root[FPSTR(_name)];
if (!usermod.isNull()) {
if (usermod[FPSTR(_enabled)].is<bool>()) {
enabled = usermod[FPSTR(_enabled)].as<bool>();
if (prevEnabled != enabled) onUpdateBegin(!enabled);
}
#ifdef ARDUINO_ARCH_ESP32
if (usermod[FPSTR(_inputLvl)].is<int>()) {
inputLevel = min(255,max(0,usermod[FPSTR(_inputLvl)].as<int>()));
}
#endif
}
}
/*
* addToConfig() can be used to add custom persistent settings to the cfg.json file in the "um" (usermod) object.
* It will be called by WLED when settings are actually saved (for example, LED settings are saved)
* If you want to force saving the current state, use serializeConfig() in your loop().
*
* CAUTION: serializeConfig() will initiate a filesystem write operation.
* It might cause the LEDs to stutter and will cause flash wear if called too often.
* Use it sparingly and always in the loop, never in network callbacks!
*
* addToConfig() will make your settings editable through the Usermod Settings page automatically.
*
* Usermod Settings Overview:
* - Numeric values are treated as floats in the browser.
* - If the numeric value entered into the browser contains a decimal point, it will be parsed as a C float
* before being returned to the Usermod. The float data type has only 6-7 decimal digits of precision, and
* doubles are not supported, numbers will be rounded to the nearest float value when being parsed.
* The range accepted by the input field is +/- 1.175494351e-38 to +/- 3.402823466e+38.
* - If the numeric value entered into the browser doesn't contain a decimal point, it will be parsed as a
* C int32_t (range: -2147483648 to 2147483647) before being returned to the usermod.
* Overflows or underflows are truncated to the max/min value for an int32_t, and again truncated to the type
* used in the Usermod when reading the value from ArduinoJson.
* - Pin values can be treated differently from an integer value by using the key name "pin"
* - "pin" can contain a single or array of integer values
* - On the Usermod Settings page there is simple checking for pin conflicts and warnings for special pins
* - Red color indicates a conflict. Yellow color indicates a pin with a warning (e.g. an input-only pin)
* - Tip: use int8_t to store the pin value in the Usermod, so a -1 value (pin not set) can be used
*
* See usermod_v2_auto_save.h for an example that saves Flash space by reusing ArduinoJson key name strings
*
* If you need a dedicated settings page with custom layout for your Usermod, that takes a lot more work.
* You will have to add the setting to the HTML, xml.cpp and set.cpp manually.
* See the WLED Soundreactive fork (code and wiki) for reference. https://github.com/atuline/WLED
*
* I highly recommend checking out the basics of ArduinoJson serialization and deserialization in order to use custom settings!
*/
void addToConfig(JsonObject& root)
{
JsonObject top = root.createNestedObject(FPSTR(_name));
top[FPSTR(_enabled)] = enabled;
#ifdef ARDUINO_ARCH_ESP32
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
JsonObject amic = top.createNestedObject(FPSTR(_analogmic));
amic["pin"] = audioPin;
#endif
JsonObject dmic = top.createNestedObject(FPSTR(_digitalmic));
dmic[F("type")] = dmType;
JsonArray pinArray = dmic.createNestedArray("pin");
pinArray.add(i2ssdPin);
pinArray.add(i2swsPin);
pinArray.add(i2sckPin);
pinArray.add(mclkPin);
pinArray.add(sdaPin);
pinArray.add(sclPin);
JsonObject cfg = top.createNestedObject("config");
cfg[F("squelch")] = soundSquelch;
cfg[F("gain")] = sampleGain;
cfg[F("AGC")] = soundAgc;
//WLEDMM: experimental settings
JsonObject poweruser = top.createNestedObject("experiments");
poweruser[F("micLev")] = micLevelMethod;
poweruser[F("freqDist")] = freqDist;
poweruser[F("freqRMS")] = averageByRMS;
JsonObject freqScale = top.createNestedObject("frequency");
freqScale[F("scale")] = FFTScalingMode;
freqScale[F("profile")] = pinkIndex; //WLEDMM
#endif
JsonObject dynLim = top.createNestedObject("dynamics");
dynLim[F("limiter")] = limiterOn;
dynLim[F("rise")] = attackTime;
dynLim[F("fall")] = decayTime;
JsonObject sync = top.createNestedObject("sync");
sync[F("port")] = audioSyncPort;
sync[F("mode")] = audioSyncEnabled;
}
/*
* readFromConfig() can be used to read back the custom settings you added with addToConfig().
* This is called by WLED when settings are loaded (currently this only happens immediately after boot, or after saving on the Usermod Settings page)
*
* readFromConfig() is called BEFORE setup(). This means you can use your persistent values in setup() (e.g. pin assignments, buffer sizes),
* but also that if you want to write persistent values to a dynamic buffer, you'd need to allocate it here instead of in setup.
* If you don't know what that is, don't fret. It most likely doesn't affect your use case :)
*
* Return true in case the config values returned from Usermod Settings were complete, or false if you'd like WLED to save your defaults to disk (so any missing values are editable in Usermod Settings)
*
* getJsonValue() returns false if the value is missing, or copies the value into the variable provided and returns true if the value is present
* The configComplete variable is true only if the "exampleUsermod" object and all values are present. If any values are missing, WLED will know to call addToConfig() to save them
*
* This function is guaranteed to be called on boot, but could also be called every time settings are updated
*/
bool readFromConfig(JsonObject& root)
{
JsonObject top = root[FPSTR(_name)];
bool configComplete = !top.isNull();
configComplete &= getJsonValue(top[FPSTR(_enabled)], enabled);
#ifdef ARDUINO_ARCH_ESP32
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
configComplete &= getJsonValue(top[FPSTR(_analogmic)]["pin"], audioPin);
#else
audioPin = -1; // MCU does not support analog mic
#endif
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["type"], dmType);
#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) || defined(CONFIG_IDF_TARGET_ESP32S3)
if (dmType == 0) dmType = SR_DMTYPE; // MCU does not support analog
#if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3)
if (dmType == 5) dmType = SR_DMTYPE; // MCU does not support PDM
if (dmType == 51) dmType = SR_DMTYPE; // MCU does not support legacy PDM
#endif
#else
if (dmType == 5) useInputFilter = 1; // enable filter for PDM
if (dmType == 51) useInputFilter = 1; // switch on filter for legacy PDM
#endif
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][0], i2ssdPin);
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][1], i2swsPin);
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][2], i2sckPin);
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][3], mclkPin);
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][4], sdaPin);
configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][5], sclPin);
configComplete &= getJsonValue(top["config"][F("squelch")], soundSquelch);
configComplete &= getJsonValue(top["config"][F("gain")], sampleGain);
configComplete &= getJsonValue(top["config"][F("AGC")], soundAgc);
//WLEDMM: experimental settings
configComplete &= getJsonValue(top["experiments"][F("micLev")], micLevelMethod);
configComplete &= getJsonValue(top["experiments"][F("freqDist")], freqDist);
configComplete &= getJsonValue(top["experiments"][F("freqRMS")], averageByRMS);
configComplete &= getJsonValue(top["frequency"][F("scale")], FFTScalingMode);
configComplete &= getJsonValue(top["frequency"][F("profile")], pinkIndex); //WLEDMM
#endif
configComplete &= getJsonValue(top["dynamics"][F("limiter")], limiterOn);
configComplete &= getJsonValue(top["dynamics"][F("rise")], attackTime);
configComplete &= getJsonValue(top["dynamics"][F("fall")], decayTime);
configComplete &= getJsonValue(top["sync"][F("port")], audioSyncPort);
configComplete &= getJsonValue(top["sync"][F("mode")], audioSyncEnabled);
return configComplete;
}
void appendConfigData()
{
oappend(SET_F("addInfo('AudioReactive:help',0,'<button onclick=\"location.href=&quot;https://mm.kno.wled.ge/soundreactive/Sound-Settings&quot;\" type=\"button\">?</button>');"));
#ifdef ARDUINO_ARCH_ESP32
//WLEDMM: add defaults
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) // -S3/-S2/-C3 don't support analog audio
#ifdef AUDIOPIN
oappend(SET_F("xOpt('AudioReactive:analogmic:pin',1,' ⎌',")); oappendi(AUDIOPIN); oappend(");");
#endif
oappend(SET_F("aOpt('AudioReactive:analogmic:pin',1);")); //only analog options
#endif
oappend(SET_F("dd=addDropdown('AudioReactive','digitalmic:type');"));
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
#if SR_DMTYPE==0
oappend(SET_F("addOption(dd,'Generic Analog (⎌)',0);"));
#else
oappend(SET_F("addOption(dd,'Generic Analog',0);"));
#endif
#endif
#if SR_DMTYPE==1
oappend(SET_F("addOption(dd,'Generic I2S (⎌)',1);"));
#else
oappend(SET_F("addOption(dd,'Generic I2S',1);"));
#endif
#if SR_DMTYPE==2
oappend(SET_F("addOption(dd,'ES7243 (⎌)',2);"));
#else
oappend(SET_F("addOption(dd,'ES7243',2);"));
#endif
#if SR_DMTYPE==3
oappend(SET_F("addOption(dd,'SPH0654 (⎌)',3);"));
#else
oappend(SET_F("addOption(dd,'SPH0654',3);"));
#endif
#if SR_DMTYPE==4
oappend(SET_F("addOption(dd,'Generic I2S with Mclk (⎌)',4);"));
#else
oappend(SET_F("addOption(dd,'Generic I2S with Mclk',4);"));
#endif
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
#if SR_DMTYPE==5
oappend(SET_F("addOption(dd,'Generic I2S PDM (⎌)',5);"));
#else
oappend(SET_F("addOption(dd,'Generic I2S PDM',5);"));
#endif
#if SR_DMTYPE==51
oappend(SET_F("addOption(dd,'.Legacy I2S PDM ☾ (⎌)',51);"));
#else
oappend(SET_F("addOption(dd,'.Legacy I2S PDM ☾',51);"));
#endif
#endif
#if SR_DMTYPE==6
oappend(SET_F("addOption(dd,'ES8388 ☾ (⎌)',6);"));
#else
oappend(SET_F("addOption(dd,'ES8388 ☾',6);"));
#endif
#ifdef SR_SQUELCH
oappend(SET_F("addInfo('AudioReactive:config:squelch',1,'<i>&#9100; ")); oappendi(SR_SQUELCH); oappend("</i>');"); // 0 is field type, 1 is actual field
#endif
#ifdef SR_GAIN
oappend(SET_F("addInfo('AudioReactive:config:gain',1,'<i>&#9100; ")); oappendi(SR_GAIN); oappend("</i>');"); // 0 is field type, 1 is actual field
#endif
oappend(SET_F("dd=addDropdown('AudioReactive','config:AGC');"));
oappend(SET_F("addOption(dd,'Off',0);"));
oappend(SET_F("addOption(dd,'Normal',1);"));
oappend(SET_F("addOption(dd,'Vivid',2);"));
oappend(SET_F("addOption(dd,'Lazy',3);"));
//WLEDMM: experimental settings
oappend(SET_F("dd=addDropdown('AudioReactive','experiments:micLev');"));
oappend(SET_F("addOption(dd,'Floating (⎌)',0);"));
oappend(SET_F("addOption(dd,'Freeze',1);"));
oappend(SET_F("addOption(dd,'Fast Freeze',2);"));
oappend(SET_F("addInfo('AudioReactive:experiments:micLev',1,'☾');"));
oappend(SET_F("dd=addDropdown('AudioReactive','experiments:freqDist');"));
oappend(SET_F("addOption(dd,'Normal (⎌)',0);"));
oappend(SET_F("addOption(dd,'RightShift',1);"));
oappend(SET_F("addInfo('AudioReactive:experiments:freqDist',1,'☾');"));
oappend(SET_F("dd=addDropdown('AudioReactive','experiments:freqRMS');"));
oappend(SET_F("addOption(dd,'Off (⎌)',0);"));
oappend(SET_F("addOption(dd,'On',1);"));
oappend(SET_F("addInfo('AudioReactive:experiments:freqRMS',1,'☾');"));
oappend(SET_F("dd=addDropdown('AudioReactive','dynamics:limiter');"));
oappend(SET_F("addOption(dd,'Off',0);"));
oappend(SET_F("addOption(dd,'On',1);"));
oappend(SET_F("addInfo('AudioReactive:dynamics:limiter',0,' On ');")); // 0 is field type, 1 is actual field
oappend(SET_F("addInfo('AudioReactive:dynamics:rise',1,'ms <i>(&#x266A; effects only)</i>');"));
oappend(SET_F("addInfo('AudioReactive:dynamics:fall',1,'ms <i>(&#x266A; effects only)</i>');"));
oappend(SET_F("dd=addDropdown('AudioReactive','frequency:scale');"));
oappend(SET_F("addOption(dd,'None',0);"));
oappend(SET_F("addOption(dd,'Linear (Amplitude)',2);"));
oappend(SET_F("addOption(dd,'Square Root (Energy)',3);"));
oappend(SET_F("addOption(dd,'Logarithmic (Loudness)',1);"));
//WLEDMM add defaults
oappend(SET_F("dd=addDropdown('AudioReactive','frequency:profile');"));
#if SR_FREQ_PROF==0
oappend(SET_F("addOption(dd,'Generic Microphone (⎌)',0);"));
#else
oappend(SET_F("addOption(dd,'Generic Microphone',0);"));
#endif
#if SR_FREQ_PROF==1
oappend(SET_F("addOption(dd,'Generic Line-In (⎌)',1);"));
#else
oappend(SET_F("addOption(dd,'Generic Line-In',1);"));
#endif
#if SR_FREQ_PROF==5
oappend(SET_F("addOption(dd,'ICS-43434 (⎌)',5);"));
#else
oappend(SET_F("addOption(dd,'ICS-43434',5);"));
#endif
#if SR_FREQ_PROF==6
oappend(SET_F("addOption(dd,'ICS-43434 - big speakers (⎌)',6);"));
#else
oappend(SET_F("addOption(dd,'ICS-43434 - big speakers',6);"));
#endif
#if SR_FREQ_PROF==7
oappend(SET_F("addOption(dd,'SPM1423 (⎌)',7);"));
#else
oappend(SET_F("addOption(dd,'SPM1423',7);"));
#endif
#if SR_FREQ_PROF==2
oappend(SET_F("addOption(dd,'IMNP441 (⎌)',2);"));
#else
oappend(SET_F("addOption(dd,'IMNP441',2);"));
#endif
#if SR_FREQ_PROF==3
oappend(SET_F("addOption(dd,'IMNP441 - big speakers (⎌)',3);"));
#else
oappend(SET_F("addOption(dd,'IMNP441 - big speakers',3);"));
#endif
#if SR_FREQ_PROF==4
oappend(SET_F("addOption(dd,'IMNP441 - small speakers (⎌)',4);"));
#else
oappend(SET_F("addOption(dd,'IMNP441 - small speakers',4);"));
#endif
#if SR_FREQ_PROF==10
oappend(SET_F("addOption(dd,'flat - no adjustments (⎌)',10);"));
#else
oappend(SET_F("addOption(dd,'flat - no adjustments',10);"));
#endif
#if SR_FREQ_PROF==8
oappend(SET_F("addOption(dd,'userdefined #1 (⎌)',8);"));
#else
oappend(SET_F("addOption(dd,'userdefined #1',8);"));
#endif
#if SR_FREQ_PROF==9
oappend(SET_F("addOption(dd,'userdefined #2 (⎌)',9);"));
#else
oappend(SET_F("addOption(dd,'userdefined #2',9);"));
#endif
oappend(SET_F("addInfo('AudioReactive:frequency:profile',1,'☾');"));
#endif
oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');"));
oappend(SET_F("addOption(dd,'Off',0);"));
#ifdef ARDUINO_ARCH_ESP32
oappend(SET_F("addOption(dd,'Send',1);"));
#endif
oappend(SET_F("addOption(dd,'Receive',2);"));
oappend(SET_F("addInfo('AudioReactive:sync:mode',1,'<br> Sync audio data with other WLEDs');"));
oappend(SET_F("addInfo('AudioReactive:digitalmic:type',1,'<i>requires reboot!</i>');")); // 0 is field type, 1 is actual field
#ifdef ARDUINO_ARCH_ESP32
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',0,'<i>sd/data/dout</i>','I2S SD');"));
#ifdef I2S_SDPIN
oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',0,' ⎌',")); oappendi(I2S_SDPIN); oappend(");");
#endif
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',1,'<i>ws/clk/lrck</i>','I2S WS');"));
oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',1);")); // disable read only pins
#ifdef I2S_WSPIN
oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',1,' ⎌',")); oappendi(I2S_WSPIN); oappend(");");
#endif
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',2,'<i>sck/bclk</i>','I2S SCK');"));
oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',2);")); // disable read only pins
#ifdef I2S_CKPIN
oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',2,' ⎌',")); oappendi(I2S_CKPIN); oappend(");");
#endif
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',3,'<i>master clock</i>','I2S MCLK');"));
oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',3);")); // disable read only pins
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
oappend(SET_F("dOpt('AudioReactive:digitalmic:pin[]',3,2,2);")); //only use -1, 0, 1 or 3
oappend(SET_F("dOpt('AudioReactive:digitalmic:pin[]',3,4,39);")); //only use -1, 0, 1 or 3
#endif
#ifdef MCLK_PIN
oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',3,' ⎌',")); oappendi(MCLK_PIN); oappend(");");
#endif
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',4,'','I2C SDA');"));
oappend(SET_F("rOpt('AudioReactive:digitalmic:pin[]',4,'use global (")); oappendi(i2c_sda); oappend(")',-1);");
#ifdef ES7243_SDAPIN
oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',4,' ⎌',")); oappendi(ES7243_SDAPIN); oappend(");");
#endif
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',5,'','I2C SCL');"));
oappend(SET_F("rOpt('AudioReactive:digitalmic:pin[]',5,'use global (")); oappendi(i2c_scl); oappend(")',-1);");
#ifdef ES7243_SCLPIN
oappend(SET_F("xOpt('AudioReactive:digitalmic:pin[]',5,' ⎌',")); oappendi(ES7243_SCLPIN); oappend(");");
#endif
oappend(SET_F("dRO('AudioReactive:digitalmic:pin[]',5);")); // disable read only pins
#endif
}
/*
* handleOverlayDraw() is called just before every show() (LED strip update frame) after effects have set the colors.
* Use this to blank out some LEDs or set them to a different color regardless of the set effect mode.
* Commonly used for custom clocks (Cronixie, 7 segment)
*/
//void handleOverlayDraw()
//{
//strip.setPixelColor(0, RGBW32(0,0,0,0)) // set the first pixel to black
//}
/*
* getId() allows you to optionally give your V2 usermod an unique ID (please define it in const.h!).
* This could be used in the future for the system to determine whether your usermod is installed.
*/
uint16_t getId()
{
return USERMOD_ID_AUDIOREACTIVE;
}
};
// strings to reduce flash memory usage (used more than twice)
const char AudioReactive::_name[] PROGMEM = "AudioReactive";
const char AudioReactive::_enabled[] PROGMEM = "enabled";
const char AudioReactive::_inputLvl[] PROGMEM = "inputLevel";
#if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
const char AudioReactive::_analogmic[] PROGMEM = "analogmic";
#endif
const char AudioReactive::_digitalmic[] PROGMEM = "digitalmic";
const char AudioReactive::UDP_SYNC_HEADER[] PROGMEM = "00002"; // new sync header version, as format no longer compatible with previous structure
const char AudioReactive::UDP_SYNC_HEADER_v1[] PROGMEM = "00001"; // old sync header version - need to add backwards-compatibility feature
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