4245767357
audioSyncPacket contains four "invisible" padding bytes added by the compiler. These need to be initialized to zero, as future versions of the protocol will make use of these fields.
1835 lines
90 KiB
C++
1835 lines
90 KiB
C++
#pragma once
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#include "wled.h"
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#include <driver/i2s.h>
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#include <driver/adc.h>
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#ifndef ARDUINO_ARCH_ESP32
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#error This audio reactive usermod does not support the ESP8266.
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#endif
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#if defined(WLED_DEBUG) || defined(SR_DEBUG)
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#include <esp_timer.h>
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#endif
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/*
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* Usermods allow you to add own functionality to WLED more easily
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* See: https://github.com/Aircoookie/WLED/wiki/Add-own-functionality
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*
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* This is an audioreactive v2 usermod.
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* ....
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*/
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#if !defined(FFTTASK_PRIORITY)
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#define FFTTASK_PRIORITY 1 // standard: looptask prio
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//#define FFTTASK_PRIORITY 2 // above looptask, below asyc_tcp
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//#define FFTTASK_PRIORITY 4 // above asyc_tcp
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#endif
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// Comment/Uncomment to toggle usb serial debugging
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// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
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// #define FFT_SAMPLING_LOG // FFT result debugging
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// #define SR_DEBUG // generic SR DEBUG messages
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#ifdef SR_DEBUG
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#define DEBUGSR_PRINT(x) DEBUGOUT.print(x)
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#define DEBUGSR_PRINTLN(x) DEBUGOUT.println(x)
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#define DEBUGSR_PRINTF(x...) DEBUGOUT.printf(x)
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#else
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#define DEBUGSR_PRINT(x)
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#define DEBUGSR_PRINTLN(x)
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#define DEBUGSR_PRINTF(x...)
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#endif
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#if defined(MIC_LOGGER) || defined(FFT_SAMPLING_LOG)
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#define PLOT_PRINT(x) DEBUGOUT.print(x)
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#define PLOT_PRINTLN(x) DEBUGOUT.println(x)
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#define PLOT_PRINTF(x...) DEBUGOUT.printf(x)
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#else
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#define PLOT_PRINT(x)
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#define PLOT_PRINTLN(x)
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#define PLOT_PRINTF(x...)
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#endif
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// use audio source class (ESP32 specific)
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#include "audio_source.h"
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constexpr i2s_port_t I2S_PORT = I2S_NUM_0; // I2S port to use (do not change !)
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constexpr int BLOCK_SIZE = 128; // I2S buffer size (samples)
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// globals
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static uint8_t inputLevel = 128; // UI slider value
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#ifndef SR_SQUELCH
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uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value)
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#else
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uint8_t soundSquelch = SR_SQUELCH; // squelch value for volume reactive routines (config value)
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#endif
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#ifndef SR_GAIN
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uint8_t sampleGain = 60; // sample gain (config value)
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#else
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uint8_t sampleGain = SR_GAIN; // sample gain (config value)
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#endif
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static uint8_t soundAgc = 1; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value)
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static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value)
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static bool udpSyncConnected = false; // UDP connection status -> true if connected to multicast group
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// user settable parameters for limitSoundDynamics()
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static bool limiterOn = true; // bool: enable / disable dynamics limiter
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static uint16_t attackTime = 80; // int: attack time in milliseconds. Default 0.08sec
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static uint16_t decayTime = 1400; // int: decay time in milliseconds. Default 1.40sec
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// user settable options for FFTResult scaling
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static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized sqare root
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//
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// AGC presets
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// Note: in C++, "const" implies "static" - no need to explicitly declare everything as "static const"
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//
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#define AGC_NUM_PRESETS 3 // AGC presets: normal, vivid, lazy
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const double agcSampleDecay[AGC_NUM_PRESETS] = { 0.9994f, 0.9985f, 0.9997f}; // decay factor for sampleMax, in case the current sample is below sampleMax
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const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone
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const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone
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const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level
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const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65%
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const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang)
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const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85%
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const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // quickly follow setpoint - ~0.15 sec
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const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs
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const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter
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const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter
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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)
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// AGC presets end
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static AudioSource *audioSource = nullptr;
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static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks.
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static bool useBandPassFilter = false; // if true, enables a bandpass filter 80Hz-16Khz to remove noise. Applies before FFT.
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// audioreactive variables shared with FFT task
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static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point
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static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
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static float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate)
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static float sampleAgc = 0.0f; // Smoothed AGC sample
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// peak detection
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static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
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static uint8_t maxVol = 31; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
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static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
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static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
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static unsigned long timeOfPeak = 0; // time of last sample peak detection.
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static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[])
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static void autoResetPeak(void); // peak auto-reset function
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////////////////////
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// Begin FFT Code //
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////////////////////
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// some prototypes, to ensure consistent interfaces
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static float mapf(float x, float in_min, float in_max, float out_min, float out_max); // map function for float
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static float fftAddAvg(int from, int to); // average of several FFT result bins
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void FFTcode(void * parameter); // audio processing task: read samples, run FFT, fill GEQ channels from FFT results
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static void runMicFilter(uint16_t numSamples, float *sampleBuffer); // pre-filtering of raw samples (band-pass)
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static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels); // post-processing and post-amp of GEQ channels
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#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !!
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static TaskHandle_t FFT_Task = nullptr;
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// Table of multiplication factors so that we can even out the frequency response.
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static float fftResultPink[NUM_GEQ_CHANNELS] = { 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f };
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// globals and FFT Output variables shared with animations
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static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency
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static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency
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static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects
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#if defined(WLED_DEBUG) || defined(SR_DEBUG)
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static uint64_t fftTime = 0;
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static uint64_t sampleTime = 0;
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#endif
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// FFT Task variables (filtering and post-processing)
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static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f}; // Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256.
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static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON)
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#ifdef SR_DEBUG
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static float fftResultMax[NUM_GEQ_CHANNELS] = {0.0f}; // A table used for testing to determine how our post-processing is working.
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#endif
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// audio source parameters and constant
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constexpr SRate_t SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz is a standard rate. Physical sample time -> 23ms
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//constexpr SRate_t SAMPLE_RATE = 16000; // 16kHz - use if FFTtask takes more than 20ms. Physical sample time -> 32ms
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//constexpr SRate_t SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental. Physical sample time -> 25ms
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//constexpr SRate_t SAMPLE_RATE = 10240; // Base sample rate in Hz - previous default. Physical sample time -> 50ms
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#define FFT_MIN_CYCLE 21 // minimum time before FFT task is repeated. Use with 22Khz sampling
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//#define FFT_MIN_CYCLE 30 // Use with 16Khz sampling
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//#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated. Use with 20Khz sampling
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//#define FFT_MIN_CYCLE 46 // minimum time before FFT task is repeated. Use with 10Khz sampling
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// FFT Constants
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constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2
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constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
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// the following are observed values, supported by a bit of "educated guessing"
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//#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels
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#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
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#define LOG_256 5.54517744f // log(256)
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// These are the input and output vectors. Input vectors receive computed results from FFT.
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static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
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static float vImag[samplesFFT] = {0.0f}; // imaginary parts
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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static float windowWeighingFactors[samplesFFT] = {0.0f};
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#endif
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// Create FFT object
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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// lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
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// these options actually cause slow-downs on all esp32 processors, don't use them.
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// #define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc) - not faster on ESP32
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// #define FFT_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt - slower on ESP32
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// Below options are forcing ArduinoFFT to use sqrtf() instead of sqrt()
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#define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 10-50% on ESP32
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#define sqrt_internal sqrtf // see https://github.com/kosme/arduinoFFT/pull/83
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#else
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// around 40% slower on -S2
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// lib_deps += https://github.com/blazoncek/arduinoFFT.git
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#endif
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#include <arduinoFFT.h>
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors);
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#else
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static arduinoFFT FFT = arduinoFFT(vReal, vImag, samplesFFT, SAMPLE_RATE);
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#endif
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// Helper functions
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// float version of map()
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static float mapf(float x, float in_min, float in_max, float out_min, float out_max){
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return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
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}
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// compute average of several FFT resut bins
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static float fftAddAvg(int from, int to) {
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float result = 0.0f;
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for (int i = from; i <= to; i++) {
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result += vReal[i];
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}
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return result / float(to - from + 1);
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}
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//
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// FFT main task
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//
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void FFTcode(void * parameter)
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{
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DEBUGSR_PRINT("FFT started on core: "); DEBUGSR_PRINTLN(xPortGetCoreID());
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// see https://www.freertos.org/vtaskdelayuntil.html
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const TickType_t xFrequency = FFT_MIN_CYCLE * portTICK_PERIOD_MS;
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TickType_t xLastWakeTime = xTaskGetTickCount();
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for(;;) {
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delay(1); // DO NOT DELETE THIS LINE! It is needed to give the IDLE(0) task enough time and to keep the watchdog happy.
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// taskYIELD(), yield(), vTaskDelay() and esp_task_wdt_feed() didn't seem to work.
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// Don't run FFT computing code if we're in Receive mode or in realtime mode
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if (disableSoundProcessing || (audioSyncEnabled & 0x02)) {
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vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
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continue;
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}
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#if defined(WLED_DEBUG) || defined(SR_DEBUG)
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uint64_t start = esp_timer_get_time();
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bool haveDoneFFT = false; // indicates if second measurement (FFT time) is valid
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#endif
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// get a fresh batch of samples from I2S
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if (audioSource) audioSource->getSamples(vReal, samplesFFT);
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#if defined(WLED_DEBUG) || defined(SR_DEBUG)
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if (start < esp_timer_get_time()) { // filter out overflows
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uint64_t sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding
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sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10; // smooth
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}
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start = esp_timer_get_time(); // start measuring FFT time
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#endif
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xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay
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// band pass filter - can reduce noise floor by a factor of 50
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// downside: frequencies below 100Hz will be ignored
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if (useBandPassFilter) runMicFilter(samplesFFT, vReal);
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// find highest sample in the batch
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float maxSample = 0.0f; // max sample from FFT batch
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for (int i=0; i < samplesFFT; i++) {
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// set imaginary parts to 0
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vImag[i] = 0;
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// pick our our current mic sample - we take the max value from all samples that go into FFT
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if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) //skip extreme values - normally these are artefacts
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if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]);
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}
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// release highest sample to volume reactive effects early - not strictly necessary here - could also be done at the end of the function
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// 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.
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micDataReal = maxSample;
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#ifdef SR_DEBUG
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if (true) { // this allows measure FFT runtimes, as it disables the "only when needed" optimization
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#else
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if (sampleAvg > 0.25f) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed.
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#endif
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// run FFT (takes 3-5ms on ESP32, ~12ms on ESP32-S2)
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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FFT.dcRemoval(); // remove DC offset
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FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy
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//FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection
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FFT.compute( FFTDirection::Forward ); // Compute FFT
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FFT.complexToMagnitude(); // Compute magnitudes
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#else
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FFT.DCRemoval(); // let FFT lib remove DC component, so we don't need to care about this in getSamples()
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//FFT.Windowing( FFT_WIN_TYP_HAMMING, FFT_FORWARD ); // Weigh data - standard Hamming window
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//FFT.Windowing( FFT_WIN_TYP_BLACKMAN, FFT_FORWARD ); // Blackman window - better side freq rejection
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//FFT.Windowing( FFT_WIN_TYP_BLACKMAN_HARRIS, FFT_FORWARD );// Blackman-Harris - excellent sideband rejection
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FFT.Windowing( FFT_WIN_TYP_FLT_TOP, FFT_FORWARD ); // Flat Top Window - better amplitude accuracy
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FFT.Compute( FFT_FORWARD ); // Compute FFT
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FFT.ComplexToMagnitude(); // Compute magnitudes
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#endif
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#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
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FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant
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#else
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FFT.MajorPeak(&FFT_MajorPeak, &FFT_Magnitude); // let the effects know which freq was most dominant
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#endif
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FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
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#if defined(WLED_DEBUG) || defined(SR_DEBUG)
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haveDoneFFT = true;
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#endif
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} else { // noise gate closed - only clear results as FFT was skipped. MIC samples are still valid when we do this.
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memset(vReal, 0, sizeof(vReal));
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FFT_MajorPeak = 1;
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FFT_Magnitude = 0.001;
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}
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for (int i = 0; i < samplesFFT; i++) {
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float t = fabsf(vReal[i]); // just to be sure - values in fft bins should be positive any way
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vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max.
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} // for()
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// mapping of FFT result bins to frequency channels
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if (fabsf(sampleAvg) > 0.5f) { // noise gate open
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#if 0
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/* This FFT post processing is a DIY endeavour. What we really need is someone with sound engineering expertise to do a great job here AND most importantly, that the animations look GREAT as a result.
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*
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* Andrew's updated mapping of 256 bins down to the 16 result bins with Sample Freq = 10240, samplesFFT = 512 and some overlap.
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* Based on testing, the lowest/Start frequency is 60 Hz (with bin 3) and a highest/End frequency of 5120 Hz in bin 255.
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* Now, Take the 60Hz and multiply by 1.320367784 to get the next frequency and so on until the end. Then detetermine the bins.
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* End frequency = Start frequency * multiplier ^ 16
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* Multiplier = (End frequency/ Start frequency) ^ 1/16
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* Multiplier = 1.320367784
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*/ // Range
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fftCalc[ 0] = fftAddAvg(2,4); // 60 - 100
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fftCalc[ 1] = fftAddAvg(4,5); // 80 - 120
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fftCalc[ 2] = fftAddAvg(5,7); // 100 - 160
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fftCalc[ 3] = fftAddAvg(7,9); // 140 - 200
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fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
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fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
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fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
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fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
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fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
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fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
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fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
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fftCalc[11] = fftAddAvg(64,84); // 1280 - 1700
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fftCalc[12] = fftAddAvg(84,111); // 1680 - 2240
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fftCalc[13] = fftAddAvg(111,147); // 2220 - 2960
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fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900
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fftCalc[15] = fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate
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#else
|
|
/* new mapping, optimized for 22050 Hz by softhack007 */
|
|
// bins frequency range
|
|
if (useBandPassFilter) {
|
|
// skip frequencies below 100hz
|
|
fftCalc[ 0] = 0.8f * fftAddAvg(3,4);
|
|
fftCalc[ 1] = 0.9f * fftAddAvg(4,5);
|
|
fftCalc[ 2] = fftAddAvg(5,6);
|
|
fftCalc[ 3] = fftAddAvg(6,7);
|
|
// don't use the last bins from 206 to 255.
|
|
fftCalc[15] = fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping
|
|
} else {
|
|
fftCalc[ 0] = fftAddAvg(1,2); // 1 43 - 86 sub-bass
|
|
fftCalc[ 1] = fftAddAvg(2,3); // 1 86 - 129 bass
|
|
fftCalc[ 2] = fftAddAvg(3,5); // 2 129 - 216 bass
|
|
fftCalc[ 3] = fftAddAvg(5,7); // 2 216 - 301 bass + midrange
|
|
// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
|
|
fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
|
|
}
|
|
fftCalc[ 4] = fftAddAvg(7,10); // 3 301 - 430 midrange
|
|
fftCalc[ 5] = fftAddAvg(10,13); // 3 430 - 560 midrange
|
|
fftCalc[ 6] = fftAddAvg(13,19); // 5 560 - 818 midrange
|
|
fftCalc[ 7] = fftAddAvg(19,26); // 7 818 - 1120 midrange -- 1Khz should always be the center !
|
|
fftCalc[ 8] = fftAddAvg(26,33); // 7 1120 - 1421 midrange
|
|
fftCalc[ 9] = fftAddAvg(33,44); // 9 1421 - 1895 midrange
|
|
fftCalc[10] = fftAddAvg(44,56); // 12 1895 - 2412 midrange + high mid
|
|
fftCalc[11] = fftAddAvg(56,70); // 14 2412 - 3015 high mid
|
|
fftCalc[12] = fftAddAvg(70,86); // 16 3015 - 3704 high mid
|
|
fftCalc[13] = fftAddAvg(86,104); // 18 3704 - 4479 high mid
|
|
fftCalc[14] = fftAddAvg(104,165) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
|
|
#endif
|
|
} else { // noise gate closed - just decay old values
|
|
for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
|
|
fftCalc[i] *= 0.85f; // decay to zero
|
|
if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f;
|
|
}
|
|
}
|
|
|
|
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
|
|
postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS);
|
|
|
|
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
|
|
if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows
|
|
uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
|
|
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
|
|
}
|
|
#endif
|
|
// run peak detection
|
|
autoResetPeak();
|
|
detectSamplePeak();
|
|
|
|
#if !defined(I2S_GRAB_ADC1_COMPLETELY)
|
|
if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC
|
|
#endif
|
|
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
|
|
|
|
} // for(;;)ever
|
|
} // FFTcode() task end
|
|
|
|
|
|
///////////////////////////
|
|
// Pre / Postprocessing //
|
|
///////////////////////////
|
|
|
|
static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // pre-filtering of raw samples (band-pass)
|
|
{
|
|
// low frequency cutoff parameter - see https://dsp.stackexchange.com/questions/40462/exponential-moving-average-cut-off-frequency
|
|
//constexpr float alpha = 0.04f; // 150Hz
|
|
//constexpr float alpha = 0.03f; // 110Hz
|
|
constexpr float alpha = 0.0225f; // 80hz
|
|
//constexpr float alpha = 0.01693f;// 60hz
|
|
// high frequency cutoff parameter
|
|
//constexpr float beta1 = 0.75f; // 11Khz
|
|
//constexpr float beta1 = 0.82f; // 15Khz
|
|
//constexpr float beta1 = 0.8285f; // 18Khz
|
|
constexpr float beta1 = 0.85f; // 20Khz
|
|
|
|
constexpr float beta2 = (1.0f - beta1) / 2.0f;
|
|
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
|
|
static float lowfilt = 0.0f; // IIR low frequency cutoff filter
|
|
|
|
for (int i=0; i < numSamples; i++) {
|
|
// FIR lowpass, to remove high frequency noise
|
|
float highFilteredSample;
|
|
if (i < (numSamples-1)) highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*sampleBuffer[i+1]; // smooth out spikes
|
|
else highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*last_vals[1]; // 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[i];
|
|
if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
|
|
// Manual linear adjustment of gain using sampleGain adjustment for different input types.
|
|
fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //apply gain, with inputLevel adjustment
|
|
if(fftCalc[i] < 0) fftCalc[i] = 0;
|
|
}
|
|
|
|
// smooth results - rise fast, fall slower
|
|
if(fftCalc[i] > fftAvg[i]) // rise fast
|
|
fftAvg[i] = fftCalc[i] *0.75f + 0.25f*fftAvg[i]; // will need approx 2 cycles (50ms) for converging against fftCalc[i]
|
|
else { // fall slow
|
|
if (decayTime < 1000) fftAvg[i] = fftCalc[i]*0.22f + 0.78f*fftAvg[i]; // approx 5 cycles (225ms) for falling to zero
|
|
else if (decayTime < 2000) fftAvg[i] = fftCalc[i]*0.17f + 0.83f*fftAvg[i]; // default - approx 9 cycles (225ms) for falling to zero
|
|
else if (decayTime < 3000) fftAvg[i] = fftCalc[i]*0.14f + 0.86f*fftAvg[i]; // approx 14 cycles (350ms) for falling to zero
|
|
else fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i]; // approx 20 cycles (500ms) for falling to zero
|
|
}
|
|
// constrain internal vars - just to be sure
|
|
fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f);
|
|
fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f);
|
|
|
|
float currentResult;
|
|
if(limiterOn == true)
|
|
currentResult = fftAvg[i];
|
|
else
|
|
currentResult = fftCalc[i];
|
|
|
|
switch (FFTScalingMode) {
|
|
case 1:
|
|
// Logarithmic scaling
|
|
currentResult *= 0.42f; // 42 is the answer ;-)
|
|
currentResult -= 8.0f; // this skips the lowest row, giving some room for peaks
|
|
if (currentResult > 1.0f) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
|
|
else currentResult = 0.0f; // special handling, because log(1) = 0; log(0) = undefined
|
|
currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies
|
|
currentResult = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255]
|
|
break;
|
|
case 2:
|
|
// Linear scaling
|
|
currentResult *= 0.30f; // needs a bit more damping, get stay below 255
|
|
currentResult -= 4.0f; // giving a bit more room for peaks
|
|
if (currentResult < 1.0f) currentResult = 0.0f;
|
|
currentResult *= 0.85f + (float(i)/1.8f); // extra up-scaling for high frequencies
|
|
break;
|
|
case 3:
|
|
// square root scaling
|
|
currentResult *= 0.38f;
|
|
currentResult -= 6.0f;
|
|
if (currentResult > 1.0f) currentResult = sqrtf(currentResult);
|
|
else currentResult = 0.0f; // special handling, because sqrt(0) = undefined
|
|
currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
|
|
currentResult = mapf(currentResult, 0.0, 16.0, 0.0, 255.0); // map [sqrt(1) ... sqrt(256)] to [0 ... 255]
|
|
break;
|
|
|
|
case 0:
|
|
default:
|
|
// no scaling - leave freq bins as-is
|
|
currentResult -= 4; // just a bit more room for peaks
|
|
break;
|
|
}
|
|
|
|
// Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely.
|
|
if (soundAgc > 0) { // apply extra "GEQ Gain" if set by user
|
|
float post_gain = (float)inputLevel/128.0f;
|
|
if (post_gain < 1.0f) post_gain = ((post_gain -1.0f) * 0.8f) +1.0f;
|
|
currentResult *= post_gain;
|
|
}
|
|
fftResult[i] = constrain((int)currentResult, 0, 255);
|
|
}
|
|
}
|
|
////////////////////
|
|
// Peak detection //
|
|
////////////////////
|
|
|
|
// peak detection is called from FFT task when vReal[] contains valid FFT results
|
|
static void detectSamplePeak(void) {
|
|
bool havePeak = false;
|
|
// softhack007: this code continuously triggers while amplitude in the selected bin is above a certain threshold. So it does not detect peaks - it detects high activity in a frequency bin.
|
|
// Poor man's beat detection by seeing if sample > Average + some value.
|
|
// This goes through ALL of the 255 bins - but ignores stupid settings
|
|
// Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync.
|
|
if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 4) && (vReal[binNum] > maxVol) && ((millis() - timeOfPeak) > 100)) {
|
|
havePeak = true;
|
|
}
|
|
|
|
if (havePeak) {
|
|
samplePeak = true;
|
|
timeOfPeak = millis();
|
|
udpSamplePeak = true;
|
|
}
|
|
}
|
|
|
|
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:
|
|
#ifndef AUDIOPIN
|
|
int8_t audioPin = -1;
|
|
#else
|
|
int8_t audioPin = AUDIOPIN;
|
|
#endif
|
|
#ifndef SR_DMTYPE // I2S mic type
|
|
uint8_t dmType = 1; // 0=none/disabled/analog; 1=generic I2S
|
|
#define SR_DMTYPE 1 // default type = I2S
|
|
#else
|
|
uint8_t dmType = SR_DMTYPE;
|
|
#endif
|
|
#ifndef I2S_SDPIN // aka DOUT
|
|
int8_t i2ssdPin = 32;
|
|
#else
|
|
int8_t i2ssdPin = I2S_SDPIN;
|
|
#endif
|
|
#ifndef I2S_WSPIN // aka LRCL
|
|
int8_t i2swsPin = 15;
|
|
#else
|
|
int8_t i2swsPin = I2S_WSPIN;
|
|
#endif
|
|
#ifndef I2S_CKPIN // aka BCLK
|
|
int8_t i2sckPin = 14; /*PDM: set to I2S_PIN_NO_CHANGE*/
|
|
#else
|
|
int8_t i2sckPin = I2S_CKPIN;
|
|
#endif
|
|
#ifndef MCLK_PIN
|
|
int8_t mclkPin = I2S_PIN_NO_CHANGE; /* ESP32: only -1, 0, 1, 3 allowed*/
|
|
#else
|
|
int8_t mclkPin = MCLK_PIN;
|
|
#endif
|
|
|
|
// new "V2" audiosync struct - 40 Bytes
|
|
struct audioSyncPacket {
|
|
char header[6]; // 06 Bytes
|
|
float sampleRaw; // 04 Bytes - either "sampleRaw" or "rawSampleAgc" depending on soundAgc setting
|
|
float sampleSmth; // 04 Bytes - either "sampleAvg" or "sampleAgc" depending on soundAgc setting
|
|
uint8_t samplePeak; // 01 Bytes - 0 no peak; >=1 peak detected. In future, this will also provide peak Magnitude
|
|
uint8_t reserved1; // 01 Bytes - for future extensions - not used yet
|
|
uint8_t fftResult[16]; // 16 Bytes
|
|
float FFT_Magnitude; // 04 Bytes
|
|
float FFT_MajorPeak; // 04 Bytes
|
|
};
|
|
|
|
// old "V1" audiosync struct - 83 Bytes - for backwards compatibility
|
|
struct audioSyncPacket_v1 {
|
|
char header[6]; // 06 Bytes
|
|
uint8_t myVals[32]; // 32 Bytes
|
|
int sampleAgc; // 04 Bytes
|
|
int sampleRaw; // 04 Bytes
|
|
float sampleAvg; // 04 Bytes
|
|
bool samplePeak; // 01 Bytes
|
|
uint8_t fftResult[16]; // 16 Bytes
|
|
double FFT_Magnitude; // 08 Bytes
|
|
double FFT_MajorPeak; // 08 Bytes
|
|
};
|
|
|
|
// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
|
|
bool enabled = false;
|
|
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
|
|
|
|
// 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
|
|
|
|
// variables used in effects
|
|
float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
|
|
int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc
|
|
float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc
|
|
|
|
// used to feed "Info" Page
|
|
unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket
|
|
int receivedFormat = 0; // last received UDP sound sync format - 0=none, 1=v1 (0.13.x), 2=v2 (0.14.x)
|
|
float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds
|
|
unsigned long sampleMaxTimer = 0; // last time maxSample5sec was reset
|
|
#define CYCLE_SAMPLEMAX 3500 // time window for merasuring
|
|
|
|
// strings to reduce flash memory usage (used more than twice)
|
|
static const char _name[];
|
|
static const char _enabled[];
|
|
static const char _inputLvl[];
|
|
static const char _analogmic[];
|
|
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("micReal:"); PLOT_PRINT(micDataReal); PLOT_PRINT("\t");
|
|
PLOT_PRINT("volumeSmth:"); PLOT_PRINT(volumeSmth); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("volumeRaw:"); PLOT_PRINT(volumeRaw); PLOT_PRINT("\t");
|
|
PLOT_PRINT("DC_Level:"); PLOT_PRINT(micLev); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("sampleAgc:"); PLOT_PRINT(sampleAgc); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("sampleAvg:"); PLOT_PRINT(sampleAvg); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("sampleReal:"); PLOT_PRINT(sampleReal); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("micIn:"); PLOT_PRINT(micIn); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("sample:"); PLOT_PRINT(sample); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("sampleMax:"); PLOT_PRINT(sampleMax); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("samplePeak:"); PLOT_PRINT((samplePeak!=0) ? 128:0); PLOT_PRINT("\t");
|
|
//PLOT_PRINT("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t");
|
|
PLOT_PRINTLN();
|
|
#endif
|
|
|
|
#ifdef FFT_SAMPLING_LOG
|
|
#if 0
|
|
for(int i=0; i<NUM_GEQ_CHANNELS; i++) {
|
|
PLOT_PRINT(fftResult[i]);
|
|
PLOT_PRINT("\t");
|
|
}
|
|
PLOT_PRINTLN();
|
|
#endif
|
|
|
|
// OPTIONS are in the following format: Description \n Option
|
|
//
|
|
// Set true if wanting to see all the bands in their own vertical space on the Serial Plotter, false if wanting to see values in Serial Monitor
|
|
const bool mapValuesToPlotterSpace = false;
|
|
// Set true to apply an auto-gain like setting to to the data (this hasn't been tested recently)
|
|
const bool scaleValuesFromCurrentMaxVal = false;
|
|
// prints the max value seen in the current data
|
|
const bool printMaxVal = false;
|
|
// prints the min value seen in the current data
|
|
const bool printMinVal = false;
|
|
// if !scaleValuesFromCurrentMaxVal, we scale values from [0..defaultScalingFromHighValue] to [0..scalingToHighValue], lower this if you want to see smaller values easier
|
|
const int defaultScalingFromHighValue = 256;
|
|
// Print values to terminal in range of [0..scalingToHighValue] if !mapValuesToPlotterSpace, or [(i)*scalingToHighValue..(i+1)*scalingToHighValue] if mapValuesToPlotterSpace
|
|
const int scalingToHighValue = 256;
|
|
// set higher if using scaleValuesFromCurrentMaxVal and you want a small value that's also the current maxVal to look small on the plotter (can't be 0 to avoid divide by zero error)
|
|
const int minimumMaxVal = 1;
|
|
|
|
int maxVal = minimumMaxVal;
|
|
int minVal = 0;
|
|
for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
|
|
if(fftResult[i] > maxVal) maxVal = fftResult[i];
|
|
if(fftResult[i] < minVal) minVal = fftResult[i];
|
|
}
|
|
for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
|
|
PLOT_PRINT(i); PLOT_PRINT(":");
|
|
PLOT_PRINTF("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1));
|
|
}
|
|
if(printMaxVal) {
|
|
PLOT_PRINTF("maxVal:%04d ", maxVal + (mapValuesToPlotterSpace ? 16*256 : 0));
|
|
}
|
|
if(printMinVal) {
|
|
PLOT_PRINTF("%04d:minVal ", minVal); // printed with value first, then label, so negative values can be seen in Serial Monitor but don't throw off y axis in Serial Plotter
|
|
}
|
|
if(mapValuesToPlotterSpace)
|
|
PLOT_PRINTF("max:%04d ", (printMaxVal ? 17 : 16)*256); // print line above the maximum value we expect to see on the plotter to avoid autoscaling y axis
|
|
else {
|
|
PLOT_PRINTF("max:%04d ", 256);
|
|
}
|
|
PLOT_PRINTLN();
|
|
#endif // FFT_SAMPLING_LOG
|
|
} // logAudio()
|
|
|
|
|
|
//////////////////////
|
|
// 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 < 1.0)) {
|
|
// MIC signal is "squelched" - deliver silence
|
|
tmpAgc = 0;
|
|
// we need to "spin down" the intgrated error buffer
|
|
if (fabs(control_integrated) < 0.01) control_integrated = 0.0;
|
|
else control_integrated *= 0.91;
|
|
} else {
|
|
// compute new setpoint
|
|
if (tmpAgc <= agcTarget0Up[AGC_preset])
|
|
multAgcTemp = agcTarget0[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = first setpoint
|
|
else
|
|
multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
|
|
}
|
|
// limit amplification
|
|
if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
|
|
if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
|
|
|
|
// compute error terms
|
|
control_error = multAgcTemp - lastMultAgc;
|
|
|
|
if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
|
|
&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
|
|
control_integrated += control_error * 0.002 * 0.25; // 2ms = 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.
|
|
const float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
|
|
const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function
|
|
|
|
#ifdef WLED_DISABLE_SOUND
|
|
micIn = inoise8(millis(), millis()); // Simulated analog read
|
|
micDataReal = micIn;
|
|
#else
|
|
#ifdef ARDUINO_ARCH_ESP32
|
|
micIn = int(micDataReal); // micDataSm = ((micData * 3) + micData)/4;
|
|
#else
|
|
// this is the minimal code for reading analog mic input on 8266.
|
|
// warning!! Absolutely experimental code. Audio on 8266 is still not working. Expects a million follow-on problems.
|
|
static unsigned long lastAnalogTime = 0;
|
|
static float lastAnalogValue = 0.0f;
|
|
if (millis() - lastAnalogTime > 20) {
|
|
micDataReal = analogRead(A0); // read one sample with 10bit resolution. This is a dirty hack, supporting volumereactive effects only.
|
|
lastAnalogTime = millis();
|
|
lastAnalogValue = micDataReal;
|
|
yield();
|
|
} else micDataReal = lastAnalogValue;
|
|
micIn = int(micDataReal);
|
|
#endif
|
|
#endif
|
|
|
|
micLev += (micDataReal-micLev) / 12288.0f;
|
|
if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
|
|
|
|
micIn -= micLev; // Let's center it to 0 now
|
|
// Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
|
|
float micInNoDC = fabsf(micDataReal - micLev);
|
|
expAdjF = (weighting * micInNoDC + (1.0f-weighting) * expAdjF);
|
|
expAdjF = fabsf(expAdjF); // Now (!) take the absolute value
|
|
|
|
expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
|
|
if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0"
|
|
|
|
tmpSample = expAdjF;
|
|
micIn = abs(micIn); // And get the absolute value of each sample
|
|
|
|
sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment
|
|
sampleReal = tmpSample;
|
|
|
|
sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
|
|
sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
|
|
|
|
// keep "peak" sample, but decay value if current sample is below peak
|
|
if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) {
|
|
sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
|
|
// another simple way to detect samplePeak - cannot detect beats, but reacts on peak volume
|
|
if (((binNum < 12) || ((maxVol < 1))) && (millis() - timeOfPeak > 80) && (sampleAvg > 1)) {
|
|
samplePeak = true;
|
|
timeOfPeak = millis();
|
|
udpSamplePeak = true;
|
|
}
|
|
} else {
|
|
if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0))
|
|
sampleMax += 0.5f * (sampleReal - sampleMax); // over AGC Zone - get back quickly
|
|
else
|
|
sampleMax *= agcSampleDecay[AGC_preset]; // signal to zero --> 5-8sec
|
|
}
|
|
if (sampleMax < 0.5f) sampleMax = 0.0f;
|
|
|
|
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
|
|
sampleAvg = fabsf(sampleAvg); // make sure we have a positive value
|
|
} // getSample()
|
|
|
|
|
|
/* Limits the dynamics of volumeSmth (= sampleAvg or sampleAgc).
|
|
* does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc)
|
|
*/
|
|
// effects: Gravimeter, Gravcenter, Gravcentric, Noisefire, Plasmoid, Freqpixels, Freqwave, Gravfreq, (2D Swirl, 2D Waverly)
|
|
void limitSampleDynamics(void) {
|
|
const float bigChange = 196; // just a representative number - a large, expected sample value
|
|
static unsigned long last_time = 0;
|
|
static float last_volumeSmth = 0.0f;
|
|
|
|
if (limiterOn == false) return;
|
|
|
|
long delta_time = millis() - last_time;
|
|
delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> sily lil hick-up
|
|
float deltaSample = volumeSmth - last_volumeSmth;
|
|
|
|
if (attackTime > 0) { // user has defined attack time > 0
|
|
float maxAttack = bigChange * float(delta_time) / float(attackTime);
|
|
if (deltaSample > maxAttack) deltaSample = maxAttack;
|
|
}
|
|
if (decayTime > 0) { // user has defined decay time > 0
|
|
float maxDecay = - bigChange * float(delta_time) / float(decayTime);
|
|
if (deltaSample < maxDecay) deltaSample = maxDecay;
|
|
}
|
|
|
|
volumeSmth = last_volumeSmth + deltaSample;
|
|
|
|
last_volumeSmth = volumeSmth;
|
|
last_time = millis();
|
|
}
|
|
|
|
|
|
//////////////////////
|
|
// UDP Sound Sync //
|
|
//////////////////////
|
|
|
|
// try to establish UDP sound sync connection
|
|
void connectUDPSoundSync(void) {
|
|
// This function tries to establish a UDP sync connection if needed
|
|
// necessary as we also want to transmit in "AP Mode", but the standard "connected()" callback only reacts on STA connection
|
|
static unsigned long last_connection_attempt = 0;
|
|
|
|
if ((audioSyncPort <= 0) || ((audioSyncEnabled & 0x03) == 0)) return; // Sound Sync not enabled
|
|
if (udpSyncConnected) return; // already connected
|
|
if (!(apActive || interfacesInited)) return; // neither AP nor other connections availeable
|
|
if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds
|
|
|
|
// if we arrive here, we need a UDP connection but don't have one
|
|
last_connection_attempt = millis();
|
|
connected(); // try to start UDP
|
|
}
|
|
|
|
void transmitAudioData()
|
|
{
|
|
if (!udpSyncConnected) return;
|
|
//DEBUGSR_PRINTLN("Transmitting UDP Mic Packet");
|
|
|
|
audioSyncPacket transmitData;
|
|
memset(reinterpret_cast<void *>(&transmitData), 0, sizeof(transmitData)); // make sure that the packet - including "invisible" padding bytes added by the compiler - is fully initialized
|
|
|
|
strncpy_P(transmitData.header, PSTR(UDP_SYNC_HEADER), 6);
|
|
// transmit samples that were not modified by limitSampleDynamics()
|
|
transmitData.sampleRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
|
|
transmitData.sampleSmth = (soundAgc) ? sampleAgc : sampleAvg;
|
|
transmitData.samplePeak = udpSamplePeak ? 1:0;
|
|
udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it
|
|
transmitData.reserved1 = 0;
|
|
|
|
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()
|
|
|
|
static bool isValidUdpSyncVersion(const char *header) {
|
|
return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0;
|
|
}
|
|
static bool isValidUdpSyncVersion_v1(const char *header) {
|
|
return strncmp_P(header, PSTR(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);
|
|
// update internal samples
|
|
sampleRaw = volumeRaw;
|
|
sampleAvg = volumeSmth;
|
|
rawSampleAgc = volumeRaw;
|
|
sampleAgc = volumeSmth;
|
|
multAgc = 1.0f;
|
|
// 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, 0.0f);
|
|
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
|
|
// update internal samples
|
|
sampleRaw = fmaxf(receivedPacket->sampleRaw, 0.0f);
|
|
sampleAvg = fmaxf(receivedPacket->sampleAvg, 0.0f);;
|
|
sampleAgc = volumeSmth;
|
|
rawSampleAgc = volumeRaw;
|
|
multAgc = 1.0f;
|
|
// 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 = fftUdp.parsePacket();
|
|
if (packetSize > 5) {
|
|
//DEBUGSR_PRINTLN("Received UDP Sync Packet");
|
|
uint8_t fftBuff[packetSize];
|
|
fftUdp.read(fftBuff, packetSize);
|
|
|
|
// VERIFY THAT THIS IS A COMPATIBLE PACKET
|
|
if (packetSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftBuff))) {
|
|
decodeAudioData(packetSize, fftBuff);
|
|
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet v2");
|
|
haveFreshData = true;
|
|
receivedFormat = 2;
|
|
} else {
|
|
if (packetSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftBuff))) {
|
|
decodeAudioData_v1(packetSize, fftBuff);
|
|
//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 = 8;
|
|
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;
|
|
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;
|
|
}
|
|
|
|
// 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.
|
|
|
|
useBandPassFilter = false;
|
|
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
|
|
#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)."));
|
|
audioSource = new ES7243(SAMPLE_RATE, BLOCK_SIZE);
|
|
delay(100);
|
|
if (audioSource) audioSource->initialize(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);
|
|
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);
|
|
useBandPassFilter = true; // 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;
|
|
#endif
|
|
case 6:
|
|
DEBUGSR_PRINTLN(F("AR: ES8388 Source"));
|
|
audioSource = new ES8388Source(SAMPLE_RATE, BLOCK_SIZE);
|
|
delay(100);
|
|
if (audioSource) audioSource->initialize(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)."));
|
|
audioSource = new I2SAdcSource(SAMPLE_RATE, BLOCK_SIZE);
|
|
delay(100);
|
|
useBandPassFilter = true; // PDM bandpass filter seems to help for bad quality analog
|
|
if (audioSource) audioSource->initialize(audioPin);
|
|
break;
|
|
#endif
|
|
}
|
|
delay(250); // give microphone enough time to initialise
|
|
|
|
if (!audioSource) enabled = false; // audio failed to initialise
|
|
if (enabled) onUpdateBegin(false); // create FFT task
|
|
if (FFT_Task == nullptr) enabled = false; // FFT task creation failed
|
|
if (enabled) disableSoundProcessing = false; // all good - enable audio processing
|
|
|
|
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
|
|
DEBUGSR_PRINTLN(F("AR: Failed to initialize sound input driver. Please check input PIN settings."));
|
|
#endif
|
|
disableSoundProcessing = true;
|
|
}
|
|
|
|
if (enabled) connectUDPSoundSync();
|
|
initDone = true;
|
|
}
|
|
|
|
|
|
/*
|
|
* 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();
|
|
}
|
|
|
|
if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) {
|
|
#ifndef ESP8266
|
|
udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort);
|
|
#else
|
|
udpSyncConnected = fftUdp.beginMulticast(WiFi.localIP(), IPAddress(239, 0, 0, 1), audioSyncPort);
|
|
#endif
|
|
}
|
|
}
|
|
|
|
|
|
/*
|
|
* 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 {
|
|
#ifdef 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
|
|
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.
|
|
|
|
#ifdef WLED_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 > 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
|
|
|
|
limitSampleDynamics();
|
|
} // if (!disableSoundProcessing)
|
|
|
|
autoResetPeak(); // auto-reset sample peak after strip minShowDelay
|
|
if (!udpSyncConnected) udpSamplePeak = false; // reset UDP samplePeak while UDP is unconnected
|
|
|
|
connectUDPSoundSync(); // ensure we have a connection - if needed
|
|
|
|
// UDP Microphone Sync - receive mode
|
|
if ((audioSyncEnabled & 0x02) && udpSyncConnected) {
|
|
// 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();
|
|
#ifdef ARDUINO_ARCH_ESP32
|
|
else fftUdp.flush(); // Flush udp input buffers if we haven't read it - avoids hickups in receive mode. Does not work on 8266.
|
|
#endif
|
|
lastTime = millis();
|
|
}
|
|
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
|
|
}
|
|
|
|
#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
|
|
if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) {
|
|
sampleMaxTimer = millis();
|
|
maxSample5sec = (0.15f * maxSample5sec) + 0.85f *((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
|
|
}
|
|
|
|
//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();
|
|
}
|
|
|
|
}
|
|
|
|
|
|
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;
|
|
}
|
|
|
|
|
|
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); // give some time for I2S driver to finish sampling before we suspend it
|
|
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();
|
|
}
|
|
} else {
|
|
// update has failed or create task requested
|
|
if (FFT_Task) {
|
|
vTaskResume(FFT_Task);
|
|
connected(); // resume UDP
|
|
} else
|
|
xTaskCreateUniversal( // xTaskCreateUniversal also works on -S2 and -C3 with single core
|
|
FFTcode, // Function to implement the task
|
|
"FFT", // Name of the task
|
|
3592, // Stack size in words // 3592 leaves 800-1024 bytes of task stack free
|
|
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;
|
|
}
|
|
|
|
|
|
/**
|
|
* 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;
|
|
}
|
|
|
|
|
|
////////////////////////////
|
|
// 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)
|
|
{
|
|
char myStringBuffer[16]; // buffer for snprintf()
|
|
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("\"></i>");
|
|
uiDomString += F("</button>");
|
|
infoArr.add(uiDomString);
|
|
|
|
if (enabled) {
|
|
// 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
|
|
} 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);
|
|
}
|
|
|
|
// 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"));
|
|
}
|
|
} 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 {
|
|
infoArr.add(F("I2S digital"));
|
|
}
|
|
// input level or "silence"
|
|
if (maxSample5sec > 1.0f) {
|
|
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");
|
|
}
|
|
|
|
// 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)
|
|
infoArr = user.createNestedArray(F("Sampling time"));
|
|
infoArr.add(float(sampleTime)/100.0f);
|
|
infoArr.add(" ms");
|
|
|
|
infoArr = user.createNestedArray(F("FFT time"));
|
|
infoArr.add(float(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 Sampling time: %5.2f ms\n", float(sampleTime)/100.0f);
|
|
DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", float(fftTime)/100.0f);
|
|
#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);
|
|
}
|
|
if (usermod[FPSTR(_inputLvl)].is<int>()) {
|
|
inputLevel = min(255,max(0,usermod[FPSTR(_inputLvl)].as<int>()));
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/*
|
|
* 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;
|
|
|
|
#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);
|
|
|
|
JsonObject cfg = top.createNestedObject("config");
|
|
cfg[F("squelch")] = soundSquelch;
|
|
cfg[F("gain")] = sampleGain;
|
|
cfg[F("AGC")] = soundAgc;
|
|
|
|
JsonObject dynLim = top.createNestedObject("dynamics");
|
|
dynLim[F("limiter")] = limiterOn;
|
|
dynLim[F("rise")] = attackTime;
|
|
dynLim[F("fall")] = decayTime;
|
|
|
|
JsonObject freqScale = top.createNestedObject("frequency");
|
|
freqScale[F("scale")] = FFTScalingMode;
|
|
|
|
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);
|
|
|
|
#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
|
|
#endif
|
|
#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["config"][F("squelch")], soundSquelch);
|
|
configComplete &= getJsonValue(top["config"][F("gain")], sampleGain);
|
|
configComplete &= getJsonValue(top["config"][F("AGC")], soundAgc);
|
|
|
|
configComplete &= getJsonValue(top["dynamics"][F("limiter")], limiterOn);
|
|
configComplete &= getJsonValue(top["dynamics"][F("rise")], attackTime);
|
|
configComplete &= getJsonValue(top["dynamics"][F("fall")], decayTime);
|
|
|
|
configComplete &= getJsonValue(top["frequency"][F("scale")], FFTScalingMode);
|
|
|
|
configComplete &= getJsonValue(top["sync"][F("port")], audioSyncPort);
|
|
configComplete &= getJsonValue(top["sync"][F("mode")], audioSyncEnabled);
|
|
|
|
return configComplete;
|
|
}
|
|
|
|
|
|
void appendConfigData()
|
|
{
|
|
oappend(SET_F("dd=addDropdown('AudioReactive','digitalmic:type');"));
|
|
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
|
|
oappend(SET_F("addOption(dd,'Generic Analog',0);"));
|
|
#endif
|
|
oappend(SET_F("addOption(dd,'Generic I2S',1);"));
|
|
oappend(SET_F("addOption(dd,'ES7243',2);"));
|
|
oappend(SET_F("addOption(dd,'SPH0654',3);"));
|
|
oappend(SET_F("addOption(dd,'Generic I2S with Mclk',4);"));
|
|
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
|
|
oappend(SET_F("addOption(dd,'Generic I2S PDM',5);"));
|
|
#endif
|
|
oappend(SET_F("addOption(dd,'ES8388',6);"));
|
|
|
|
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);"));
|
|
|
|
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>(♪ effects only)</i>');"));
|
|
oappend(SET_F("addInfo('AudioReactive:dynamics:fall',1,'ms <i>(♪ 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);"));
|
|
|
|
oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');"));
|
|
oappend(SET_F("addOption(dd,'Off',0);"));
|
|
oappend(SET_F("addOption(dd,'Send',1);"));
|
|
oappend(SET_F("addOption(dd,'Receive',2);"));
|
|
oappend(SET_F("addInfo('AudioReactive:digitalmic:type',1,'<i>requires reboot!</i>');")); // 0 is field type, 1 is actual field
|
|
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',0,'<i>sd/data/dout</i>','I2S SD');"));
|
|
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',1,'<i>ws/clk/lrck</i>','I2S WS');"));
|
|
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',2,'<i>sck/bclk</i>','I2S SCK');"));
|
|
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3)
|
|
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',3,'<i>only use -1, 0, 1 or 3</i>','I2S MCLK');"));
|
|
#else
|
|
oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',3,'<i>master clock</i>','I2S MCLK');"));
|
|
#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
|