#pragma once /* @title MoonModules WLED - audioreactive usermod @file audio_reactive.h @repo https://github.com/MoonModules/WLED-MM, submit changes to this file as PRs to MoonModules/WLED-MM @Authors https://github.com/MoonModules/WLED-MM/commits/mdev/ @Copyright © 2024,2025 Github MoonModules Commit Authors (contact moonmodules@icloud.com for details) @license Licensed under the EUPL-1.2 or later */ #include "wled.h" #ifdef ARDUINO_ARCH_ESP32 #include #include #include #endif #if defined(ARDUINO_ARCH_ESP32) && (defined(WLED_DEBUG) || defined(SR_DEBUG)) #include #endif /* * Usermods allow you to add own functionality to WLED more easily * See: https://github.com/Aircoookie/WLED/wiki/Add-own-functionality * * This is an audioreactive v2 usermod. * .... */ #if defined(WLEDMM_FASTPATH) && defined(CONFIG_IDF_TARGET_ESP32S3) || defined(CONFIG_IDF_TARGET_ESP32) #define FFT_USE_SLIDING_WINDOW // perform FFT with sliding window = 50% overlap #endif #define FFT_PREFER_EXACT_PEAKS // use different FFT windowing -> results in "sharper" peaks and less "leaking" into other frequencies //#define SR_STATS #if !defined(FFTTASK_PRIORITY) #if defined(WLEDMM_FASTPATH) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && defined(ARDUINO_ARCH_ESP32) // FASTPATH: use higher priority, to avoid that webserver (ws, json, etc) delays sample processing //#define FFTTASK_PRIORITY 3 // competing with async_tcp #define FFTTASK_PRIORITY 4 // above async_tcp #else #define FFTTASK_PRIORITY 1 // standard: looptask prio //#define FFTTASK_PRIORITY 2 // above looptask, below async_tcp #endif #endif #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) // this applies "pink noise scaling" to FFT results before computing the major peak for effects. // currently only for ESP32-S3 and classic ESP32, due to increased runtime #define FFT_MAJORPEAK_HUMAN_EAR #endif // high-resolution type for input filters #if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) #define SR_HIRES_TYPE double // ESP32 and ESP32-S3 (with FPU) are fast enough to use "double" #else #define SR_HIRES_TYPE float // prefer faster type on slower boards (-S2, -C3) #endif // Comment/Uncomment to toggle usb serial debugging // #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter) // #define FFT_SAMPLING_LOG // FFT result debugging // #define SR_DEBUG // generic SR DEBUG messages #ifdef SR_DEBUG #define DEBUGSR_PRINT(x) DEBUGOUT(x) #define DEBUGSR_PRINTLN(x) DEBUGOUTLN(x) #define DEBUGSR_PRINTF(x...) DEBUGOUTF(x) #else #define DEBUGSR_PRINT(x) #define DEBUGSR_PRINTLN(x) #define DEBUGSR_PRINTF(x...) #endif #if defined(SR_DEBUG) #define ERRORSR_PRINT(x) DEBUGSR_PRINT(x) #define ERRORSR_PRINTLN(x) DEBUGSR_PRINTLN(x) #define ERRORSR_PRINTF(x...) DEBUGSR_PRINTF(x) #else #if defined(WLED_DEBUG) #define ERRORSR_PRINT(x) DEBUG_PRINT(x) #define ERRORSR_PRINTLN(x) DEBUG_PRINTLN(x) #define ERRORSR_PRINTF(x...) DEBUG_PRINTF(x) #else #define ERRORSR_PRINT(x) #define ERRORSR_PRINTLN(x) #define ERRORSR_PRINTF(x...) #endif #endif #if defined(MIC_LOGGER) || defined(FFT_SAMPLING_LOG) #define PLOT_PRINT(x) DEBUGOUT(x) #define PLOT_PRINTLN(x) DEBUGOUTLN(x) #define PLOT_PRINTF(x...) DEBUGOUTF(x) #define PLOT_FLUSH() DEBUGOUTFlush() #else #define PLOT_PRINT(x) #define PLOT_PRINTLN(x) #define PLOT_PRINTF(x...) #define PLOT_FLUSH() #endif // sanity checks #ifdef ARDUINO_ARCH_ESP32 // we need more space in for oappend() stack buffer -> SETTINGS_STACK_BUF_SIZE and CONFIG_ASYNC_TCP_STACK_SIZE #if SETTINGS_STACK_BUF_SIZE < 3904 // 3904 is required for WLEDMM-0.14.0-b28 #warning please increase SETTINGS_STACK_BUF_SIZE >= 3904 #endif #if (CONFIG_ASYNC_TCP_STACK_SIZE - SETTINGS_STACK_BUF_SIZE) < 4352 // at least 4096+256 words of free task stack is needed by async_tcp alone #error remaining async_tcp stack will be too low - please increase CONFIG_ASYNC_TCP_STACK_SIZE #endif #endif // audiosync constants #define AUDIOSYNC_NONE 0x00 // UDP sound sync off #define AUDIOSYNC_SEND 0x01 // UDP sound sync - send mode #define AUDIOSYNC_REC 0x02 // UDP sound sync - receiver mode #define AUDIOSYNC_REC_PLUS 0x06 // UDP sound sync - receiver + local mode (uses local input if no receiving udp sound) #define AUDIOSYNC_IDLE_MS 2500 // timeout for "receiver idle" (milliseconds) static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as it is shared between tasks. static uint8_t audioSyncEnabled = AUDIOSYNC_NONE; // bit field: bit 0 - send, bit 1 - receive, bit 2 - use local if not receiving static bool audioSyncSequence = true; // if true, the receiver will drop out-of-sequence packets static uint8_t audioSyncPurge = 1; // 0: process each packet (don't purge); 1: auto-purge old packets; 2: only process latest received packet (always purge) static bool udpSyncConnected = false; // UDP connection status -> true if connected to multicast group static volatile bool isOOM = false; // FFTask: not enough memory for buffers (audio processing failed to start) #define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !! // audioreactive variables #ifdef ARDUINO_ARCH_ESP32 #ifndef SR_AGC // Automatic gain control mode #ifdef SR_SQUELCH #define SR_AGC 1 // default "squelch" was provided --> default mode = on #else #define SR_AGC 0 // default mode = off #endif #endif static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier static float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate) static float sampleAgc = 0.0f; // Smoothed AGC sample static uint8_t soundAgc = SR_AGC; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value) - enable AGC if default "squelch" was provided #endif static float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay() static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same time as samplePeak, but reset by transmitAudioData static unsigned long timeOfPeak = 0; // time of last sample peak detection. volatile bool haveNewFFTResult = false; // flag to directly inform UDP sound sender when new FFT results are available (to reduce latency). Flag is reset at next UDP send static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0}; // Our calculated freq. channel result table to be used by effects static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f}; // Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256. (also used by dynamics limiter) static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON) static uint16_t zeroCrossingCount = 0; // number of zero crossings in the current batch of 512 samples // TODO: probably best not used by receive nodes static float agcSensitivity = 128; // AGC sensitivity estimation, based on agc gain (multAgc). calculated by getSensitivity(). range 0..255 // user settable parameters for limitSoundDynamics() #ifdef UM_AUDIOREACTIVE_DYNAMICS_LIMITER_OFF static bool limiterOn = false; // bool: enable / disable dynamics limiter #else static bool limiterOn = true; #endif [[maybe_unused]] static uint8_t micQuality = 0; // input filtering; 0 normal, 1 minimal filtering, 2 no filtering - unused on 8266 #ifdef FFT_USE_SLIDING_WINDOW static uint16_t attackTime = 24; // int: attack time in milliseconds. Default 0.024sec static uint16_t decayTime = 250; // int: decay time in milliseconds. New default 250ms. #else static uint16_t attackTime = 50; // int: attack time in milliseconds. Default 0.08sec static uint16_t decayTime = 300; // int: decay time in milliseconds. New default 300ms. Old default was 1.40sec #endif // peak detection #ifdef ARDUINO_ARCH_ESP32 static void detectSamplePeak(void); // peak detection function (needs scaled FFT results in vReal[]) - no used for 8266 receive-only mode #endif static void autoResetPeak(void); // peak auto-reset function static uint8_t maxVol = 31; // (was 10) Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated) static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated) #ifdef ARDUINO_ARCH_ESP32 // use audio source class (ESP32 specific) #include "audio_source.h" constexpr int BLOCK_SIZE = 128; // I2S buffer size (samples) // globals static uint8_t inputLevel = 128; // UI slider value #ifndef SR_SQUELCH uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value) #else uint8_t soundSquelch = SR_SQUELCH; // squelch value for volume reactive routines (config value) #endif #ifndef SR_GAIN uint8_t sampleGain = 60; // sample gain (config value) #else uint8_t sampleGain = SR_GAIN; // sample gain (config value) #endif // user settable options for FFTResult scaling static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized square root #ifndef SR_FREQ_PROF static uint8_t pinkIndex = 0; // 0: default; 1: line-in; 2: IMNP441 #else static uint8_t pinkIndex = SR_FREQ_PROF; // 0: default; 1: line-in; 2: IMNP441 #endif // // AGC presets // Note: in C++, "const" implies "static" - no need to explicitly declare everything as "static const" // #define AGC_NUM_PRESETS 3 // AGC presets: normal, vivid, lazy const double agcSampleDecay[AGC_NUM_PRESETS] = { 0.9994f, 0.9985f, 0.9997f}; // decay factor for sampleMax, in case the current sample is below sampleMax const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65% const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang) const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85% const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // quickly follow setpoint - ~0.15 sec const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter #if defined(WLEDMM_FASTPATH) const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/8.f, 1/5.f, 1/12.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value) #else 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) #endif // AGC presets end static AudioSource *audioSource = nullptr; static uint8_t useInputFilter = 0; // enables low-cut filtering. Applies before FFT. //WLEDMM experimental settings static uint8_t micLevelMethod = 1; // 0=old "floating" miclev, 1=new "freeze" mode, 2=fast freeze mode (mode 2 may not work for you) #if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) static constexpr uint8_t averageByRMS = false; // false: use mean value, true: use RMS (root mean squared). use simpler method on slower MCUs. #else static constexpr uint8_t averageByRMS = true; // false: use mean value, true: use RMS (root mean squared). use better method on fast MCUs. #endif static uint8_t freqDist = 0; // 0=old 1=rightshift mode static uint8_t fftWindow = 0; // FFT windowing function (0 = default) #ifdef FFT_USE_SLIDING_WINDOW static uint8_t doSlidingFFT = 1; // 1 = use sliding window FFT (faster & more accurate) #endif // variables used in effects //static int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc //static float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc // shared vars for debugging #ifdef MIC_LOGGER static volatile float micReal_min = 0.0f; // MicIn data min from last batch of samples static volatile float micReal_avg = 0.0f; // MicIn data average (from last batch of samples) static volatile float micReal_max = 0.0f; // MicIn data max from last batch of samples #if 0 static volatile float micReal_min2 = 0.0f; // MicIn data min after filtering static volatile float micReal_max2 = 0.0f; // MicIn data max after filtering #endif #endif //////////////////// // Begin FFT Code // //////////////////// // some prototypes, to ensure consistent interfaces static float mapf(float x, float in_min, float in_max, float out_min, float out_max); // map function for float static float fftAddAvg(int from, int to); // average of several FFT result bins void FFTcode(void * parameter); // audio processing task: read samples, run FFT, fill GEQ channels from FFT results static void runMicFilter(uint16_t numSamples, float *sampleBuffer); // pre-filtering of raw samples (band-pass) static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels, bool i2sFastpath); // post-processing and post-amp of GEQ channels static TaskHandle_t FFT_Task = nullptr; // Table of multiplication factors so that we can even out the frequency response. #define MAX_PINK 10 // 0 = standard, 1= line-in (pink noise only), 2..4 = IMNP441, 5..6 = ICS-43434, ,7=SPM1423, 8..9 = userdef, 10= flat (no pink noise adjustment) static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = { { 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f }, // 0 default from SR WLED // { 1.30f, 1.32f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.39f, 3.09f, 4.34f }, // - Line-In Generic -> pink noise adjustment only { 2.35f, 1.32f, 1.32f, 1.40f, 1.48f, 1.57f, 1.68f, 1.80f, 1.89f, 1.95f, 2.14f, 2.26f, 2.50f, 2.90f, 4.20f, 6.50f }, // 1 Line-In CS5343 + DC blocker { 1.82f, 1.72f, 1.70f, 1.50f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.90f, 3.86f, 6.29f}, // 2 IMNP441 datasheet response profile * pink noise { 2.80f, 2.20f, 1.30f, 1.15f, 1.55f, 2.45f, 4.20f, 2.80f, 3.20f, 3.60f, 4.20f, 4.90f, 5.70f, 6.05f,10.50f,14.85f}, // 3 IMNP441 - big speaker, strong bass // next one has not much visual differece compared to default IMNP441 profile { 12.0f, 6.60f, 2.60f, 1.15f, 1.35f, 2.05f, 2.85f, 2.50f, 2.85f, 3.30f, 2.25f, 4.35f, 3.80f, 3.75f, 6.50f, 9.00f}, // 4 IMNP441 - voice, or small speaker { 2.75f, 1.60f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 1.75f, 2.55f, 3.60f }, // 5 ICS-43434 datasheet response * pink noise { 2.90f, 1.25f, 0.75f, 1.08f, 2.35f, 3.55f, 3.60f, 3.40f, 2.75f, 3.45f, 4.40f, 6.35f, 6.80f, 6.80f, 8.50f,10.64f }, // 6 ICS-43434 - big speaker, strong bass { 1.65f, 1.00f, 1.05f, 1.30f, 1.48f, 1.30f, 1.80f, 3.00f, 1.50f, 1.65f, 2.56f, 3.00f, 2.60f, 2.30f, 5.00f, 3.00f }, // 7 SPM1423 { 2.25f, 1.60f, 1.30f, 1.60f, 2.20f, 3.20f, 3.06f, 2.60f, 2.85f, 3.50f, 4.10f, 4.80f, 5.70f, 6.05f,10.50f,14.85f }, // 8 userdef #1 for ewowi (enhance median/high freqs) { 4.75f, 3.60f, 2.40f, 2.46f, 3.52f, 1.60f, 1.68f, 3.20f, 2.20f, 2.00f, 2.30f, 2.41f, 2.30f, 1.25f, 4.55f, 6.50f }, // 9 userdef #2 for softhack (mic hidden inside mini-shield) { 2.38f, 2.18f, 2.07f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.95f, 1.70f, 2.13f, 2.47f } // 10 almost FLAT (IMNP441 but no PINK noise adjustments) }; /* how to make your own profile: * =============================== * preparation: make sure your microphone has direct line-of-sigh with the speaker, 1-2meter distance is best * Prepare your HiFi equipment: disable all "Sound enhancements" - like Loudness, Equalizer, Bass Boost. Bass/Treble controls set to middle. * Your HiFi equipment should receive its audio input from Line-In, SPDIF, HDMI, or another "undistorted" connection (like CDROM). * Try not to use Bluetooth or MP3 when playing the "pink noise" audio. BT-audio and MP3 both perform "acoustic adjustments" that we don't want now. * SR WLED: enable AGC ("standard" or "lazy"), set squelch to a low level, check that LEDs don't react in silence. * SR WLED: select "Generic Line-In" as your Frequency Profile, "Linear" or "Square Root" as Frequency Scale * SR WLED: Dynamic Limiter On, Dynamics Fall Time around 4200 - makes GEQ hold peaks for much longer * SR WLED: Select GEQ effect, move all effect slider to max (i.e. right side) * Measure: play Pink Noise for 2-3 minutes - for examples from youtube https://www.youtube.com/watch?v=ZXtimhT-ff4 * Measure: Take a Photo. Make sure that LEDs for each "bar" are well visible (ou need to count them later) * Your own profile: * - Target for each LED bar is 50% to 75% of the max height --> 8(high) x 16(wide) panel means target = 5. 32 x 16 means target = 22. * - From left to right - count the LEDs in each of the 16 frequency columns (that's why you need the photo). This is the barheight for each channel. * - math time! Find the multiplier that will bring each bar to the target. * * in case of square root scale: multiplier = (target * target) / (barheight * barheight) * * in case of linear scale: multiplier = target / barheight * * - replace one of the "userdef" lines with a copy of the parameter line for "Line-In", * - go through your new "userdef" parameter line, multiply each entry with the multiplier you found for that column. * Compile + upload * Test your new profile (same procedure as above). Iterate the process to improve results. */ // globals and FFT Output variables shared with animations static float FFT_MajPeakSmth = 1.0f; // FFT: (peak) frequency, smooth #if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS) static float fftTaskCycle = 0; // avg cycle time for FFT task static float fftTime = 0; // avg time for single FFT static float sampleTime = 0; // avg (blocked) time for reading I2S samples static float filterTime = 0; // avg time for filtering I2S samples #endif // FFT Task variables (filtering and post-processing) static float lastFftCalc[NUM_GEQ_CHANNELS] = {0.0f}; // backup of last FFT channels (before postprocessing) #if !defined(CONFIG_IDF_TARGET_ESP32C3) // audio source parameters and constant constexpr SRate_t SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz is a standard rate. Physical sample time -> 23ms //constexpr SRate_t SAMPLE_RATE = 16000; // 16kHz - use if FFTtask takes more than 20ms. Physical sample time -> 32ms //constexpr SRate_t SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental. Physical sample time -> 25ms //constexpr SRate_t SAMPLE_RATE = 10240; // Base sample rate in Hz - previous default. Physical sample time -> 50ms #ifndef WLEDMM_FASTPATH #define FFT_MIN_CYCLE 21 // minimum time before FFT task is repeated. Use with 22Khz sampling #else #ifdef FFT_USE_SLIDING_WINDOW #define FFT_MIN_CYCLE 8 // we only have 12ms to take 1/2 batch of samples #else #define FFT_MIN_CYCLE 15 // reduce min time, to allow faster catch-up when I2S is lagging #endif #endif //#define FFT_MIN_CYCLE 30 // Use with 16Khz sampling //#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated. Use with 20Khz sampling //#define FFT_MIN_CYCLE 46 // minimum time before FFT task is repeated. Use with 10Khz sampling #else // slightly lower the sampling rate for -C3, to improve stability //constexpr SRate_t SAMPLE_RATE = 20480; // 20Khz; Physical sample time -> 25ms //#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated. constexpr SRate_t SAMPLE_RATE = 18000; // 18Khz; Physical sample time -> 28ms #define FFT_MIN_CYCLE 25 // minimum time before FFT task is repeated. // try 16Khz in case your device still lags and responds too slowly. //constexpr SRate_t SAMPLE_RATE = 16000; // 16Khz -> Physical sample time -> 32ms //#define FFT_MIN_CYCLE 30 // minimum time before FFT task is repeated. #endif // FFT Constants constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2 constexpr uint16_t samplesFFT_2 = 256; // meaningful part of FFT results - only the "lower half" contains useful information. // the following are observed values, supported by a bit of "educated guessing" //#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels //#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels #define FFT_DOWNSCALE 0.40f // downscaling factor for FFT results, RMS averaging #define LOG_256 5.54517744f // log(256) // These are the input and output vectors. Input vectors receive computed results from FFT. static float* vReal = nullptr; // FFT sample inputs / freq output - these are our raw result bins static float* vImag = nullptr; // imaginary parts #ifdef FFT_MAJORPEAK_HUMAN_EAR static float* pinkFactors = nullptr; // "pink noise" correction factors constexpr float pinkcenter = 23.66; // sqrt(560) - center freq for scaling is 560 hz. constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range of each FFT result bin #endif // Create FFT object // lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2 #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) // these options actually cause slow-down on -S2 (-S2 doesn't have floating point hardware) //#define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and a few other speedups - WLEDMM not faster on ESP32 //#define FFT_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt - WLEDMM slower on ESP32 #endif #define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 10-50% on ESP32 (as alternative to FFT_SQRT_APPROXIMATION) #define sqrt_internal sqrtf // see https://github.com/kosme/arduinoFFT/pull/83 #include // Helper functions // float version of map() static float mapf(float x, float in_min, float in_max, float out_min, float out_max){ return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min; } // compute average of several FFT result bins // linear average static float fftAddAvgLin(int from, int to) { float result = 0.0f; for (int i = from; i <= to; i++) { result += vReal[i]; } return result / float(to - from + 1); } // RMS average static float fftAddAvgRMS(int from, int to) { double result = 0.0; for (int i = from; i <= to; i++) { result += vReal[i] * vReal[i]; } return sqrtf(result / float(to - from + 1)); } static float fftAddAvg(int from, int to) { if (from == to) return vReal[from]; // small optimization if (averageByRMS) return fftAddAvgRMS(from, to); // use RMS else return fftAddAvgLin(from, to); // use linear average } #if defined(CONFIG_IDF_TARGET_ESP32C3) constexpr bool skipSecondFFT = true; #else constexpr bool skipSecondFFT = false; #endif // allocate FFT sample buffers from heap static bool alocateFFTBuffers(void) { #ifdef SR_DEBUG USER_PRINT(F("\nFree heap ")); USER_PRINTLN(ESP.getFreeHeap()); #endif if (vReal) d_free(vReal); vReal = nullptr; // should not happen if (vImag) d_free(vImag); vImag = nullptr; // should not happen if ((vReal = (float*) d_calloc(samplesFFT, sizeof(float))) == nullptr) return false; // calloc or die if ((vImag = (float*) d_calloc(samplesFFT, sizeof(float))) == nullptr) return false; #ifdef FFT_MAJORPEAK_HUMAN_EAR if (pinkFactors) p_free(pinkFactors); if ((pinkFactors = (float*) p_calloc(samplesFFT, sizeof(float))) == nullptr) return false; #endif #ifdef SR_DEBUG USER_PRINTLN("\nalocateFFTBuffers() completed successfully."); USER_PRINT(F("Free heap: ")); USER_PRINTLN(ESP.getFreeHeap()); USER_PRINT("FFTtask free stack: "); USER_PRINTLN(uxTaskGetStackHighWaterMark(NULL)); USER_FLUSH(); #endif return(true); // success } // de-allocate FFT sample buffers from heap static void destroyFFTBuffers(bool panicOOM) { #ifdef FFT_MAJORPEAK_HUMAN_EAR if (pinkFactors) p_free(pinkFactors); pinkFactors = nullptr; #endif if (vImag) d_free(vImag); vImag = nullptr; if (vReal) d_free(vReal); vReal = nullptr; if (panicOOM && !isOOM) { // notify user isOOM = true; errorFlag = ERR_LOW_MEM; USER_PRINTLN("AR startup failed - out of memory!"); } #ifdef SR_DEBUG USER_PRINTLN("\ndestroyFFTBuffers() completed successfully."); USER_PRINT(F("Free heap: ")); USER_PRINTLN(ESP.getFreeHeap()); USER_FLUSH(); #endif } // High-Pass "DC blocker" filter // see https://www.dsprelated.com/freebooks/filters/DC_Blocker.html static void runDCBlocker(uint_fast16_t numSamples, float *sampleBuffer) { constexpr float filterR = 0.990f; // around 40hz static float xm1 = 0.0f; static SR_HIRES_TYPE ym1 = 0.0f; for (unsigned i=0; i < numSamples; i++) { float value = sampleBuffer[i]; SR_HIRES_TYPE filtered = (SR_HIRES_TYPE)(value-xm1) + filterR*ym1; xm1 = value; ym1 = filtered; sampleBuffer[i] = filtered; } } // // FFT runner - "return" from a task function causes crash. This wrapper keeps the task alive // void runFFTcode(void * parameter) __attribute__((noreturn,used)); void runFFTcode(void * parameter) { bool firstFail = true; // prevents flood of warnings do { if (!disableSoundProcessing) { FFTcode(parameter); if (firstFail) {USER_PRINTLN(F("warning: unexpected exit of FFT main task."));} firstFail = false; } else firstFail = true; // re-enable warning message vTaskDelay(1000); // if we arrive here, FFcode has returned due to OOM. Wait a bit, then try again. } while (true); } // // FFT main task // void FFTcode(void * parameter) { #ifdef SR_DEBUG USER_FLUSH(); USER_PRINT("AR: "); USER_PRINT(pcTaskGetTaskName(NULL)); USER_PRINT(" task started on core "); USER_PRINT(xPortGetCoreID()); // causes trouble on -S2 USER_PRINT(" [prio="); USER_PRINT(uxTaskPriorityGet(NULL)); USER_PRINT(", min free stack="); USER_PRINT(uxTaskGetStackHighWaterMark(NULL)); USER_PRINTLN("]"); USER_FLUSH(); #endif // see https://www.freertos.org/vtaskdelayuntil.html const TickType_t xFrequency = FFT_MIN_CYCLE * portTICK_PERIOD_MS; const TickType_t xFrequencyDouble = FFT_MIN_CYCLE * portTICK_PERIOD_MS * 2; static bool isFirstRun = false; #ifdef FFT_USE_SLIDING_WINDOW static float* oldSamples = nullptr; // previous 50% of samples static bool haveOldSamples = false; // for sliding window FFT bool usingOldSamples = false; if (!oldSamples) oldSamples = (float*) d_calloc(samplesFFT_2, sizeof(float)); // allocate on first run if (!oldSamples) { disableSoundProcessing = true; haveOldSamples = false; destroyFFTBuffers(true); return; } // no memory -> die #endif bool success = true; if ((vReal == nullptr) || (vImag == nullptr)) success = alocateFFTBuffers(); // allocate sample buffers on first run if (success == false) { // no memory -> clean up heap, then suspend disableSoundProcessing = true; destroyFFTBuffers(true); return; } // create FFT object - we have to do if after allocating buffers #if defined(FFT_LIB_REV) && FFT_LIB_REV > 0x19 // arduinoFFT 2.x has a slightly different API static ArduinoFFT FFT = ArduinoFFT( vReal, vImag, samplesFFT, SAMPLE_RATE, true); #else // recommended version optimized by @softhack007 (API version 1.9) #if defined(WLED_ENABLE_HUB75MATRIX) && defined(CONFIG_IDF_TARGET_ESP32) static float* windowWeighingFactors = nullptr; if (!windowWeighingFactors) windowWeighingFactors = (float*) d_calloc(samplesFFT, sizeof(float)); // cache for FFT windowing factors - use heap if (!windowWeighingFactors) { disableSoundProcessing = true; haveOldSamples = false; destroyFFTBuffers(true); return; } // alloc failed #else static float windowWeighingFactors[samplesFFT] = {0.0f}; // cache for FFT windowing factors - use global RAM #endif static ArduinoFFT FFT = ArduinoFFT( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors); #endif #ifdef FFT_MAJORPEAK_HUMAN_EAR // pre-compute pink noise scaling table for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++) { float binFreq = binInd * binWidth + binWidth/2.0f; if (binFreq > (SAMPLE_RATE * 0.42f)) binFreq = (SAMPLE_RATE * 0.42f) - 0.25 * (binFreq - (SAMPLE_RATE * 0.42f)); // suppress noise and aliasing pinkFactors[binInd] = sqrtf(binFreq) / pinkcenter; } pinkFactors[0] *= 0.5; // suppress 0-42hz bin #endif TickType_t xLastWakeTime = xTaskGetTickCount(); for(;;) { delay(1); // DO NOT DELETE THIS LINE! It is needed to give the IDLE(0) task enough time and to keep the watchdog happy. // taskYIELD(), yield(), vTaskDelay() and esp_task_wdt_feed() didn't seem to work. // Don't run FFT computing code if we're in Receive mode or in realtime mode if (disableSoundProcessing || (audioSyncEnabled == AUDIOSYNC_REC)) { isFirstRun = false; #ifdef FFT_USE_SLIDING_WINDOW haveOldSamples = false; #endif vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers continue; } #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) // timing uint64_t start = esp_timer_get_time(); bool haveDoneFFT = false; // indicates if second measurement (FFT time) is valid static uint64_t lastCycleStart = 0; static uint64_t lastLastTime = 0; if ((lastCycleStart > 0) && (lastCycleStart < start)) { // filter out overflows uint64_t taskTimeInMillis = ((start - lastCycleStart) +5ULL) / 10ULL; // "+5" to ensure proper rounding fftTaskCycle = (((taskTimeInMillis + lastLastTime)/2) *4 + fftTaskCycle*6)/10.0; // smart smooth lastLastTime = taskTimeInMillis; } lastCycleStart = start; #endif // get a fresh batch of samples from I2S memset(vReal, 0, sizeof(float) * samplesFFT); // start clean #ifdef FFT_USE_SLIDING_WINDOW uint16_t readOffset; if (haveOldSamples && (doSlidingFFT > 0)) { memcpy(vReal, oldSamples, sizeof(float) * samplesFFT_2); // copy first 50% from buffer usingOldSamples = true; readOffset = samplesFFT_2; } else { usingOldSamples = false; readOffset = 0; } // read fresh samples, in chunks of 50% do { // this looks a bit cumbersome, but it onlyworks this way - any second instance of the getSamples() call delivers junk data. if (audioSource) audioSource->getSamples(vReal+readOffset, samplesFFT_2); readOffset += samplesFFT_2; } while (readOffset < samplesFFT); #else if (audioSource) audioSource->getSamples(vReal, samplesFFT); #endif #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) // debug info in case that stack usage changes static unsigned int minStackFree = UINT32_MAX; unsigned int stackFree = uxTaskGetStackHighWaterMark(NULL); if (minStackFree > stackFree) { minStackFree = stackFree; DEBUGSR_PRINTF("|| %-9s min free stack %d\n", pcTaskGetTaskName(NULL), minStackFree); //WLEDMM } // timing if (start < esp_timer_get_time()) { // filter out overflows uint64_t sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10.0; // smooth } start = esp_timer_get_time(); // start measuring filter time #endif xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay isFirstRun = !isFirstRun; // toggle throttle #ifdef MIC_LOGGER float datMin = 0.0f; float datMax = 0.0f; double datAvg = 0.0f; for (int i=0; i < samplesFFT; i++) { if (i==0) { datMin = datMax = vReal[i]; } else { if (datMin > vReal[i]) datMin = vReal[i]; if (datMax < vReal[i]) datMax = vReal[i]; } datAvg += vReal[i]; } #endif #if defined(WLEDMM_FASTPATH) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && defined(ARDUINO_ARCH_ESP32) // experimental - be nice to LED update task (trying to avoid flickering) - dual core only #if FFTTASK_PRIORITY > 1 if (strip.isServicing()) delay(1); #endif #endif // normal mode: filter everything float *samplesStart = vReal; uint16_t sampleCount = samplesFFT; #ifdef FFT_USE_SLIDING_WINDOW if (usingOldSamples) { // sliding window mode: only latest 50% need filtering samplesStart = vReal + samplesFFT_2; sampleCount = samplesFFT_2; } #endif // band pass filter - can reduce noise floor by a factor of 50 // downside: frequencies below 100Hz will be ignored bool doDCRemoval = false; // DCRemove is only necessary if we don't use any kind of low-cut filtering if ((useInputFilter > 0) && (useInputFilter < 99)) { switch(useInputFilter) { case 1: runMicFilter(sampleCount, samplesStart); break; // PDM microphone bandpass case 2: runDCBlocker(sampleCount, samplesStart); break; // generic Low-Cut + DC blocker (~40hz cut-off) default: doDCRemoval = true; break; } } else doDCRemoval = true; #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) // timing measurement if (start < esp_timer_get_time()) { // filter out overflows uint64_t filterTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding filterTime = (filterTimeInMillis*3 + filterTime*7)/10.0; // smooth } start = esp_timer_get_time(); // start measuring FFT time #endif // set imaginary parts to 0 memset(vImag, 0, sizeof(float) * samplesFFT); #ifdef FFT_USE_SLIDING_WINDOW memcpy(oldSamples, vReal+samplesFFT_2, sizeof(float) * samplesFFT_2); // copy last 50% to buffer (for sliding window FFT) haveOldSamples = true; #endif // find the highest sample in the batch, and count zero crossings float maxSample = 0.0f; // max sample from FFT batch uint_fast16_t newZeroCrossingCount = 0; for (int i=0; i < samplesFFT; i++) { // pick our current mic sample - we take the max value from all samples that go into FFT if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) { //skip extreme values - normally these are artefacts #ifdef FFT_USE_SLIDING_WINDOW if (usingOldSamples) { if ((i >= samplesFFT_2) && (fabsf(vReal[i]) > maxSample)) maxSample = fabsf(vReal[i]); // only look at newest 50% } else #endif if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]); } // WLED-MM/TroyHacks: Calculate zero crossings // if (i < (samplesFFT-1)) { if (__builtin_signbit(vReal[i]) != __builtin_signbit(vReal[i+1])) // test sign bit: sign changed -> zero crossing newZeroCrossingCount++; } } newZeroCrossingCount = (newZeroCrossingCount*2)/3; // reduce value so it typically stays below 256 zeroCrossingCount = newZeroCrossingCount; // update only once, to avoid that effects pick up an intermediate value // release highest sample to volume reactive effects early - not strictly necessary here - could also be done at the end of the function // early release allows the filters (getSample() and agcAvg()) to work with fresh values - we will have matching gain and noise gate values when we want to process the FFT results. micDataReal = maxSample; #ifdef MIC_LOGGER micReal_min = datMin; micReal_max = datMax; micReal_avg = datAvg / samplesFFT; #if 0 // compute min/max again after filtering - useful for filter debugging for (int i=0; i < samplesFFT; i++) { if (i==0) { datMin = datMax = vReal[i]; } else { if (datMin > vReal[i]) datMin = vReal[i]; if (datMax < vReal[i]) datMax = vReal[i]; } } micReal_min2 = datMin; micReal_max2 = datMax; #endif #endif float wc = 1.0; // FFT window correction factor, relative to Blackman_Harris // run FFT (takes 3-5ms on ESP32) if (fabsf(volumeSmth) > 0.25f) { // noise gate open if ((skipSecondFFT == false) || (isFirstRun == true)) { // run FFT (takes 2-3ms on ESP32, ~12ms on ESP32-S2, ~30ms on -C3) if (doDCRemoval) FFT.dcRemoval(); // remove DC offset switch(fftWindow) { // apply FFT window case 1: FFT.windowing(FFTWindow::Hann, FFTDirection::Forward); // recommended for 50% overlap wc = 0.66415918066; // 1.8554726898 * 2.0 break; case 2: FFT.windowing( FFTWindow::Nuttall, FFTDirection::Forward); wc = 0.9916873881f; // 2.8163172034 * 2.0 break; case 5: FFT.windowing( FFTWindow::Blackman, FFTDirection::Forward); wc = 0.84762867875f; // 2.3673474360 * 2.0 break; case 3: FFT.windowing( FFTWindow::Hamming, FFTDirection::Forward); wc = 0.664159180663f; // 1.8549343278 * 2.0 break; case 4: FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude preservation, low frequency accuracy wc = 1.276771793156f; // 3.5659039231 * 2.0 break; case 0: // falls through default: FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman - Harris" window - sharp peaks due to excellent sideband rejection wc = 1.0f; // 2.7929062517 * 2.0 } #ifdef FFT_USE_SLIDING_WINDOW if (usingOldSamples) wc = wc * 1.10f; // compensate for loss caused by averaging #endif FFT.compute( FFTDirection::Forward ); // Compute FFT FFT.complexToMagnitude(); // Compute magnitudes vReal[0] = 0; // The remaining DC offset on the signal produces a strong spike on position 0 that should be eliminated to avoid issues. float last_majorpeak = FFT_MajorPeak; float last_magnitude = FFT_Magnitude; #ifdef FFT_MAJORPEAK_HUMAN_EAR // scale FFT results for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++) vReal[binInd] *= pinkFactors[binInd]; #endif #if defined(FFT_LIB_REV) && FFT_LIB_REV > 0x19 // arduinoFFT 2.x has a slightly different API FFT.majorPeak(&FFT_MajorPeak, &FFT_Magnitude); #else FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant #endif FFT_Magnitude *= wc; // apply correction factor if (FFT_MajorPeak < (SAMPLE_RATE / samplesFFT)) {FFT_MajorPeak = 1.0f; FFT_Magnitude = 0;} // too low - use zero if (FFT_MajorPeak > (0.42f * SAMPLE_RATE)) {FFT_MajorPeak = last_majorpeak; FFT_Magnitude = last_magnitude;} // too high - keep last peak #ifdef FFT_MAJORPEAK_HUMAN_EAR // undo scaling - we want unmodified values for FFTResult[] computations for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++) vReal[binInd] *= 1.0f/pinkFactors[binInd]; //fix peak magnitude if ((FFT_MajorPeak > (binWidth/1.25f)) && (FFT_MajorPeak < (SAMPLE_RATE/2.2f)) && (FFT_Magnitude > 4.0f)) { unsigned peakBin = constrain((int)((FFT_MajorPeak + binWidth/2.0f) / binWidth), 0, samplesFFT -1); FFT_Magnitude *= fmaxf(1.0f/pinkFactors[peakBin], 1.0f); } #endif FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects FFT_MajPeakSmth = FFT_MajPeakSmth + 0.42 * (FFT_MajorPeak - FFT_MajPeakSmth); // I like this "swooping peak" look } else { // skip second run --> clear fft results, keep peaks memset(vReal, 0, sizeof(float) * samplesFFT); } #if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS) haveDoneFFT = true; #endif } else { // noise gate closed - only clear results as FFT was skipped. MIC samples are still valid when we do this. memset(vReal, 0, sizeof(float) * samplesFFT); FFT_MajorPeak = 1; FFT_Magnitude = 0.001; } if ((skipSecondFFT == false) || (isFirstRun == true)) { for (int i = 0; i < samplesFFT; i++) { float t = fabsf(vReal[i]); // just to be sure - values in fft bins should be positive any way vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max. } // for() // mapping of FFT result bins to frequency channels //if (fabsf(sampleAvg) > 0.25f) { // noise gate open if (fabsf(volumeSmth) > 0.25f) { // noise gate open //WLEDMM: different distributions if (freqDist == 0) { /* new mapping, optimized for 22050 Hz by softhack007 --- update: removed overlap */ // bins frequency range if (useInputFilter==1) { // skip frequencies below 100hz fftCalc[ 0] = wc * 0.8f * fftAddAvg(3,3); fftCalc[ 1] = wc * 0.9f * fftAddAvg(4,4); fftCalc[ 2] = wc * fftAddAvg(5,5); fftCalc[ 3] = wc * fftAddAvg(6,6); // don't use the last bins from 206 to 255. fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping } else { fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass fftCalc[ 2] = wc * fftAddAvg(3,4); // 2 129 - 216 bass fftCalc[ 3] = wc * fftAddAvg(5,6); // 2 216 - 301 bass + midrange // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping } fftCalc[ 4] = wc * fftAddAvg(7,9); // 3 301 - 430 midrange fftCalc[ 5] = wc * fftAddAvg(10,12); // 3 430 - 560 midrange fftCalc[ 6] = wc * fftAddAvg(13,18); // 5 560 - 818 midrange fftCalc[ 7] = wc * fftAddAvg(19,25); // 7 818 - 1120 midrange -- 1Khz should always be the center ! fftCalc[ 8] = wc * fftAddAvg(26,32); // 7 1120 - 1421 midrange fftCalc[ 9] = wc * fftAddAvg(33,43); // 9 1421 - 1895 midrange fftCalc[10] = wc * fftAddAvg(44,55); // 12 1895 - 2412 midrange + high mid fftCalc[11] = wc * fftAddAvg(56,69); // 14 2412 - 3015 high mid fftCalc[12] = wc * fftAddAvg(70,85); // 16 3015 - 3704 high mid fftCalc[13] = wc * fftAddAvg(86,103); // 18 3704 - 4479 high mid fftCalc[14] = wc * fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping } else if (freqDist == 1) { //WLEDMM: Rightshift: note ewowi: frequencies in comments are not correct if (useInputFilter==1) { // skip frequencies below 100hz fftCalc[ 0] = wc * 0.8f * fftAddAvg(1,1); fftCalc[ 1] = wc * 0.9f * fftAddAvg(2,2); fftCalc[ 2] = wc * fftAddAvg(3,3); fftCalc[ 3] = wc * fftAddAvg(4,4); // don't use the last bins from 206 to 255. fftCalc[15] = wc * fftAddAvg(165,205) * 0.75f; // 40 7106 - 8828 high -- with some damping } else { fftCalc[ 0] = wc * fftAddAvg(1,1); // 1 43 - 86 sub-bass fftCalc[ 1] = wc * fftAddAvg(2,2); // 1 86 - 129 bass fftCalc[ 2] = wc * fftAddAvg(3,3); // 2 129 - 216 bass fftCalc[ 3] = wc * fftAddAvg(4,4); // 2 216 - 301 bass + midrange // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) fftCalc[15] = wc * fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping } fftCalc[ 4] = wc * fftAddAvg(5,6); // 3 301 - 430 midrange fftCalc[ 5] = wc * fftAddAvg(7,8); // 3 430 - 560 midrange fftCalc[ 6] = wc * fftAddAvg(9,10); // 5 560 - 818 midrange fftCalc[ 7] = wc * fftAddAvg(11,13); // 7 818 - 1120 midrange -- 1Khz should always be the center ! fftCalc[ 8] = wc * fftAddAvg(14,18); // 7 1120 - 1421 midrange fftCalc[ 9] = wc * fftAddAvg(19,25); // 9 1421 - 1895 midrange fftCalc[10] = wc * fftAddAvg(26,36); // 12 1895 - 2412 midrange + high mid fftCalc[11] = wc * fftAddAvg(37,45); // 14 2412 - 3015 high mid fftCalc[12] = wc * fftAddAvg(46,66); // 16 3015 - 3704 high mid fftCalc[13] = wc * fftAddAvg(67,97); // 18 3704 - 4479 high mid fftCalc[14] = wc * fftAddAvg(98,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping } } else { // noise gate closed - just decay old values isFirstRun = false; for (int i=0; i < NUM_GEQ_CHANNELS; i++) { fftCalc[i] *= 0.85f; // decay to zero if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f; } } memcpy(lastFftCalc, fftCalc, sizeof(lastFftCalc)); // make a backup of last "good" channels } else { // if second run skipped memcpy(fftCalc, lastFftCalc, sizeof(fftCalc)); // restore last "good" channels } // post-processing of frequency channels (pink noise adjustment, AGC, smoothing, scaling) if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK; #ifdef FFT_USE_SLIDING_WINDOW postProcessFFTResults((fabsf(volumeSmth) > 0.25f)? true : false, NUM_GEQ_CHANNELS, usingOldSamples); // this function modifies fftCalc, fftAvg and fftResult #else postProcessFFTResults((fabsf(volumeSmth) > 0.25f)? true : false, NUM_GEQ_CHANNELS, false); // this function modifies fftCalc, fftAvg and fftResult #endif #if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS) // timing static uint64_t lastLastFFT = 0; if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding fftTime = (((fftTimeInMillis + lastLastFFT)/2) *3 + fftTime*7)/10.0; // smart smooth lastLastFFT = fftTimeInMillis; } #endif // run peak detection autoResetPeak(); detectSamplePeak(); haveNewFFTResult = true; #if !defined(I2S_GRAB_ADC1_COMPLETELY) if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC #endif { #ifdef FFT_USE_SLIDING_WINDOW if (!usingOldSamples) { vTaskDelayUntil( &xLastWakeTime, xFrequencyDouble); // we need a double wait when no old data was used } else #endif if ((skipSecondFFT == false) || (fabsf(volumeSmth) < 0.25f)) { vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers } else if (isFirstRun == true) { vTaskDelayUntil( &xLastWakeTime, xFrequencyDouble); // release CPU after performing FFT in "skip second run" mode } } } // for(;;)ever } // FFTcode() task end /////////////////////////// // Pre / Postprocessing // /////////////////////////// static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // pre-filtering of raw samples (band-pass) { // low frequency cutoff parameter - see https://dsp.stackexchange.com/questions/40462/exponential-moving-average-cut-off-frequency //constexpr float alpha = 0.04f; // 150Hz //constexpr float alpha = 0.03f; // 110Hz constexpr float alpha = 0.0225f; // 80hz //constexpr float alpha = 0.01693f;// 60hz // high frequency cutoff parameter //constexpr float beta1 = 0.75f; // 11Khz //constexpr float beta1 = 0.82f; // 15Khz //constexpr float beta1 = 0.8285f; // 18Khz constexpr float beta1 = 0.85f; // 20Khz constexpr float beta2 = (1.0f - beta1) / 2.0; static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter static float lowfilt = 0.0f; // IIR low frequency cutoff filter for (int i=0; i < numSamples; i++) { // FIR lowpass, to remove high frequency noise float highFilteredSample; if (i < (numSamples-1)) highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*sampleBuffer[i+1]; // smooth out spikes else highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*last_vals[1]; // special handling for last sample in array last_vals[1] = last_vals[0]; last_vals[0] = sampleBuffer[i]; sampleBuffer[i] = highFilteredSample; // IIR highpass, to remove low frequency noise lowfilt += alpha * (sampleBuffer[i] - lowfilt); sampleBuffer[i] = sampleBuffer[i] - lowfilt; } } static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels, bool i2sFastpath) // post-processing and post-amp of GEQ channels { for (int i=0; i < numberOfChannels; i++) { if (noiseGateOpen) { // noise gate open // Adjustment for frequency curves. fftCalc[i] *= fftResultPink[pinkIndex][i]; if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function // Manual linear adjustment of gain using sampleGain adjustment for different input types. fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //apply gain, with inputLevel adjustment if(fftCalc[i] < 0) fftCalc[i] = 0; } float speed = 1.0f; // filter correction for sampling speed -> 1.0 in normal mode (43hz) if (i2sFastpath) speed = 0.6931471805599453094f * 1.1f; // -> ln(2) from math, *1.1 from my gut feeling ;-) in fast mode (86hz) if(limiterOn == true) { // Limiter ON -> smooth results if(fftCalc[i] > fftAvg[i]) { // rise fast fftAvg[i] += speed * 0.78f * (fftCalc[i] - fftAvg[i]); // will need approx 1-2 cycles (50ms) for converging against fftCalc[i] } else { // fall slow if (decayTime < 150) fftAvg[i] += speed * 0.50f * (fftCalc[i] - fftAvg[i]); else if (decayTime < 250) fftAvg[i] += speed * 0.40f * (fftCalc[i] - fftAvg[i]); else if (decayTime < 500) fftAvg[i] += speed * 0.33f * (fftCalc[i] - fftAvg[i]); else if (decayTime < 1000) fftAvg[i] += speed * 0.22f * (fftCalc[i] - fftAvg[i]); // approx 5 cycles (225ms) for falling to zero else if (decayTime < 2000) fftAvg[i] += speed * 0.17f * (fftCalc[i] - fftAvg[i]); // default - approx 9 cycles (225ms) for falling to zero else if (decayTime < 3000) fftAvg[i] += speed * 0.14f * (fftCalc[i] - fftAvg[i]); // approx 14 cycles (350ms) for falling to zero else if (decayTime < 4000) fftAvg[i] += speed * 0.10f * (fftCalc[i] - fftAvg[i]); else fftAvg[i] += speed * 0.05f * (fftCalc[i] - fftAvg[i]); } } else { // Limiter OFF if (i2sFastpath) { // fast mode -> average last two results float tmp = fftCalc[i]; fftCalc[i] = 0.7f * tmp + 0.3f * fftAvg[i]; fftAvg[i] = tmp; // store current sample for next run } else { // normal mode -> no adjustments fftAvg[i] = fftCalc[i]; // keep filters up-to-date } } // 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 = limiterOn ? fftAvg[i] : fftCalc[i]; // continue with filtered result (limiter on) or unfiltered result (limiter off) switch (FFTScalingMode) { case 1: // Logarithmic scaling currentResult *= 0.42; // 42 is the answer ;-) currentResult -= 8.0; // this skips the lowest row, giving some room for peaks if (currentResult > 1.0) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function else currentResult = 0.0; // special handling, because log(1) = 0; log(0) = undefined currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies currentResult = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255] break; case 2: // Linear scaling currentResult *= 0.30f; // needs a bit more damping, get stay below 255 currentResult -= 2.0; // giving a bit more room for peaks if (currentResult < 1.0f) currentResult = 0.0f; currentResult *= 0.85f + (float(i)/1.8f); // extra up-scaling for high frequencies break; case 3: // square root scaling currentResult *= 0.38f; //currentResult *= 0.34f; //experiment currentResult -= 6.0f; if (currentResult > 1.0) currentResult = sqrtf(currentResult); else currentResult = 0.0; // special handling, because sqrt(0) = undefined currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies //currentResult *= 0.80f + (float(i)/5.6f); //experiment currentResult = mapf(currentResult, 0.0, 16.0, 0.0, 255.0); // map [sqrt(1) ... sqrt(256)] to [0 ... 255] break; case 0: default: // no scaling - leave freq bins as-is currentResult -= 2; // just a bit more room for peaks break; } // Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely. if (soundAgc > 0) { // apply extra "GEQ Gain" if set by user float post_gain = (float)inputLevel/128.0f; if (post_gain < 1.0f) post_gain = ((post_gain -1.0f) * 0.8f) +1.0f; currentResult *= post_gain; } fftResult[i] = max(min((int)(currentResult+0.5f), 255), 0); // +0.5 for proper rounding } } //////////////////// // Peak detection // //////////////////// // peak detection is called from FFT task when vReal[] contains valid FFT results static void detectSamplePeak(void) { bool havePeak = false; #if 1 // softhack007: this code continuously triggers while volume in the selected bin is above a certain threshold. So it does not detect peaks - it detects volume in a frequency bin. // Poor man's beat detection by seeing if sample > Average + some value. // This goes through ALL of the 255 bins - but ignores stupid settings // Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync. if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 4) && (vReal[binNum] > maxVol) && ((millis() - timeOfPeak) > 100)) { havePeak = true; } #endif #if 0 // alternate detection, based on FFT_MajorPeak and FFT_Magnitude. Not much better... if ((binNum > 1) && (maxVol > 8) && (binNum < 10) && (sampleAgc > 127) && (FFT_MajorPeak > 50) && (FFT_MajorPeak < 250) && (FFT_Magnitude > (16.0f * (maxVol+42.0)) /*my_magnitude > 136.0f*16.0f*/) && (millis() - timeOfPeak > 80)) { havePeak = true; } #endif if (havePeak) { samplePeak = true; timeOfPeak = millis(); udpSamplePeak = true; } } #endif static void autoResetPeak(void) { uint16_t MinShowDelay = MAX(50, strip.getMinShowDelay()); // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed. samplePeak = false; if (audioSyncEnabled == AUDIOSYNC_NONE) udpSamplePeak = false; // this is normally reset by transmitAudioData } } //////////////////// // usermod class // //////////////////// //class name. Use something descriptive and leave the ": public Usermod" part :) class AudioReactive : public Usermod { private: #ifdef ARDUINO_ARCH_ESP32 // HUB75 workaround - audio receive only #ifdef WLED_ENABLE_HUB75MATRIX #undef SR_DMTYPE #define SR_DMTYPE 254 // "network receive only" #endif #ifndef AUDIOPIN int8_t audioPin = -1; #else int8_t audioPin = AUDIOPIN; #endif #ifndef SR_DMTYPE // I2S mic type uint8_t dmType = 1; // 0=none/disabled/analog; 1=generic I2S #define SR_DMTYPE 1 // default type = I2S #else uint8_t dmType = SR_DMTYPE; #endif #ifndef I2S_SDPIN // aka DOUT int8_t i2ssdPin = 32; #else int8_t i2ssdPin = I2S_SDPIN; #endif #ifndef I2S_WSPIN // aka LRCL int8_t i2swsPin = 15; #else int8_t i2swsPin = I2S_WSPIN; #endif #ifndef I2S_CKPIN // aka BCLK int8_t i2sckPin = 14; /*PDM: set to I2S_PIN_NO_CHANGE*/ #else int8_t i2sckPin = I2S_CKPIN; #endif #ifndef ES7243_SDAPIN int8_t sdaPin = -1; #else int8_t sdaPin = ES7243_SDAPIN; #endif #ifndef ES7243_SCLPIN int8_t sclPin = -1; #else int8_t sclPin = ES7243_SCLPIN; #endif #ifndef MCLK_PIN int8_t mclkPin = I2S_PIN_NO_CHANGE; /* ESP32: only -1, 0, 1, 3 allowed*/ #else int8_t mclkPin = MCLK_PIN; #endif #endif // new "V2" audiosync struct - 44 Bytes struct __attribute__ ((packed)) audioSyncPacket { // WLEDMM "packed" ensures that there are no additional gaps char header[6]; // 06 Bytes offset 0 - "00002" for protocol version 2 ( includes \0 for c-style string termination) uint8_t pressure[2]; // 02 Bytes, offset 6 - sound pressure as fixed point (8bit integer, 8bit fraction) float sampleRaw; // 04 Bytes offset 8 - either "sampleRaw" or "rawSampleAgc" depending on soundAgc setting float sampleSmth; // 04 Bytes offset 12 - either "sampleAvg" or "sampleAgc" depending on soundAgc setting uint8_t samplePeak; // 01 Bytes offset 16 - 0 no peak; >=1 peak detected. In future, this will also provide peak Magnitude uint8_t frameCounter; // 01 Bytes offset 17 - rolling counter to track duplicate/out of order packets uint8_t fftResult[16]; // 16 Bytes offset 18 - 16 GEQ channels, each channel has one byte (uint8_t) uint16_t zeroCrossingCount; // 02 Bytes, offset 34 - number of zero crossings seen in 23ms float FFT_Magnitude; // 04 Bytes offset 36 - largest FFT result from a single run (raw value, can go up to 4096) float FFT_MajorPeak; // 04 Bytes offset 40 - frequency (Hz) of largest FFT result }; // old "V1" audiosync struct - 83 Bytes payload, 88 bytes total - for backwards compatibility struct audioSyncPacket_v1 { char header[6]; // 06 Bytes uint8_t myVals[32]; // 32 Bytes int32_t sampleAgc; // 04 Bytes int32_t sampleRaw; // 04 Bytes float sampleAvg; // 04 Bytes bool samplePeak; // 01 Bytes uint8_t fftResult[16]; // 16 Bytes double FFT_Magnitude; // 08 Bytes double FFT_MajorPeak; // 08 Bytes }; #define UDPSOUND_MAX_PACKET 96 // max packet size for audiosync, with a bit of "headroom" #define AR_UDP_READ_INTERVAL_MS 18 // 23ms = time for reading one new batch of samples @ 22kHz; minus 5ms margin for network overhead #define AR_UDP_FLUSH_ALL 255 // tells receiveAudioData to purge the network input queue // set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer) #if defined(SR_ENABLE_DEFAULT) || defined(UM_AUDIOREACTIVE_ENABLE) bool enabled = true; // WLEDMM #else bool enabled = false; #endif bool initDone = false; // variables for UDP sound sync WiFiUDP fftUdp; // UDP object for sound sync (from WiFi UDP, not Async UDP!) unsigned long lastTime = 0; // last time of running UDP Microphone Sync #if defined(WLEDMM_FASTPATH) const uint16_t delayMs = 5; // I don't want to sample too often and overload WLED #else const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED #endif uint16_t audioSyncPort= 11988;// default port for UDP sound sync bool updateIsRunning = false; // true during OTA. #ifdef ARDUINO_ARCH_ESP32 // used for AGC int last_soundAgc = -1; // used to detect AGC mode change (for resetting AGC internal error buffers) double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error // variables used by getSample() and agcAvg() double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controller. double micLev = 0.0; // Used to convert returned value to have '0' as minimum. A leveller float expAdjF = 0.0f; // Used for exponential filter. float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC. int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel) int16_t rawSampleAgc = 0; // not smoothed AGC sample #endif // variables used in effects int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc float soundPressure = 0; // Sound Pressure estimation, based on microphone raw readings. 0 ->5db, 255 ->105db // used to feed "Info" Page unsigned long last_UDPTime = 0; // time of last valid UDP sound sync data packet 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 measuring // strings to reduce flash memory usage (used more than twice) static const char _name[]; static const char _enabled[]; static const char _inputLvl[]; #if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) static const char _analogmic[]; #endif static const char _digitalmic[]; static const char UDP_SYNC_HEADER[]; static const char UDP_SYNC_HEADER_v1[]; // private methods //////////////////// // Debug support // //////////////////// void logAudio() { if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & AUDIOSYNC_REC) == 0))) return; // no audio available #ifdef MIC_LOGGER // Debugging functions for audio input and sound processing. Comment out the values you want to see PLOT_PRINT("volumeSmth:"); PLOT_PRINT(volumeSmth + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines //PLOT_PRINT("volumeRaw:"); PLOT_PRINT(volumeRaw + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines //PLOT_PRINT("samplePeak:"); PLOT_PRINT((samplePeak!=0) ? 128:0); PLOT_PRINT("\t"); #ifdef ARDUINO_ARCH_ESP32 PLOT_PRINT("micMin:"); PLOT_PRINT(0.5f * micReal_min); PLOT_PRINT("\t"); // scaled down to 50%, for better readability PLOT_PRINT("micMax:"); PLOT_PRINT(0.5f * micReal_max); PLOT_PRINT("\t"); // scaled down to 50% //PLOT_PRINT("micAvg:"); PLOT_PRINT(0.5f * micReal_avg); PLOT_PRINT("\t"); // scaled down to 50% //PLOT_PRINT("micDC:"); PLOT_PRINT(0.5f * (micReal_min + micReal_max)/2.0f);PLOT_PRINT("\t"); // scaled down to 50% PLOT_PRINT("micReal:"); PLOT_PRINT(micDataReal + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines PLOT_PRINT("DC_Level:"); PLOT_PRINT(micLev + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines // //PLOT_PRINT("filtmicMin:"); PLOT_PRINT(0.5f * micReal_min2); PLOT_PRINT("\t"); // scaled down to 50% // //PLOT_PRINT("filtmicMax:"); PLOT_PRINT(0.5f * micReal_max2); PLOT_PRINT("\t"); // scaled down to 50% //PLOT_PRINT("sampleAgc:"); PLOT_PRINT(sampleAgc); PLOT_PRINT("\t"); //PLOT_PRINT("sampleAvg:"); PLOT_PRINT(sampleAvg); PLOT_PRINT("\t"); //PLOT_PRINT("sampleReal:"); PLOT_PRINT(sampleReal); PLOT_PRINT("\t"); //PLOT_PRINT("sample:"); PLOT_PRINT(sample); PLOT_PRINT("\t"); //PLOT_PRINT("sampleMax:"); PLOT_PRINT(sampleMax); PLOT_PRINT("\t"); //PLOT_PRINT("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t"); #endif PLOT_PRINTLN(); PLOT_FLUSH(); #endif #ifdef FFT_SAMPLING_LOG #if 0 for(int i=0; i maxVal) maxVal = fftResult[i]; if(fftResult[i] < minVal) minVal = fftResult[i]; } for(int i = 0; i < NUM_GEQ_CHANNELS; i++) { PLOT_PRINT(i); PLOT_PRINT(":"); PLOT_PRINTF("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1)); } if(printMaxVal) { PLOT_PRINTF("maxVal:%04d ", maxVal + (mapValuesToPlotterSpace ? 16*256 : 0)); } if(printMinVal) { PLOT_PRINTF("%04d:minVal ", minVal); // printed with value first, then label, so negative values can be seen in Serial Monitor but don't throw off y axis in Serial Plotter } if(mapValuesToPlotterSpace) PLOT_PRINTF("max:%04d ", (printMaxVal ? 17 : 16)*256); // print line above the maximum value we expect to see on the plotter to avoid autoscaling y axis else { PLOT_PRINTF("max:%04d ", 256); } PLOT_PRINTLN(); #endif // FFT_SAMPLING_LOG } // logAudio() #ifdef ARDUINO_ARCH_ESP32 ////////////////////// // Audio Processing // ////////////////////// /* * A "PI controller" multiplier to automatically adjust sound sensitivity. * * A few tricks are implemented so that sampleAgc doesn't only utilize 0% and 100%: * 0. don't amplify anything below squelch (but keep previous gain) * 1. gain input = maximum signal observed in the last 5-10 seconds * 2. we use two setpoints, one at ~60%, and one at ~80% of the maximum signal * 3. the amplification depends on signal level: * a) normal zone - very slow adjustment * b) emergency zone (<10% or >90%) - very fast adjustment */ void agcAvg(unsigned long the_time) { const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function float lastMultAgc = multAgc; // last multiplier used float multAgcTemp = multAgc; // new multiplier float tmpAgc = sampleReal * multAgc; // what-if amplified signal float control_error; // "control error" input for PI control if (last_soundAgc != soundAgc) control_integrated = 0.0; // new preset - reset integrator // For PI controller, we need to have a constant "frequency" // so let's make sure that the control loop is not running at insane speed static unsigned long last_time = 0; unsigned long time_now = millis(); if ((the_time > 0) && (the_time < time_now)) time_now = the_time; // allow caller to override my clock if (time_now - last_time > 2) { last_time = time_now; if((fabsf(sampleReal) < 2.0f) || (sampleMax < 1.0f)) { // MIC signal is "squelched" - deliver silence tmpAgc = 0; // we need to "spin down" the intgrated error buffer if (fabs(control_integrated) < 0.01) control_integrated = 0.0; else control_integrated *= 0.91; } else { // compute new setpoint if (tmpAgc <= agcTarget0Up[AGC_preset]) multAgcTemp = agcTarget0[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = first setpoint else multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint } // limit amplification if (multAgcTemp > 32.0f) multAgcTemp = 32.0f; if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f; // compute error terms control_error = multAgcTemp - lastMultAgc; if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping && (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max) control_integrated += control_error * 0.002 * 0.25; // 2ms = integration time; 0.25 for damping else control_integrated *= 0.9; // spin down that integrator beast // 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 emergency zone multAgcTemp = lastMultAgc + agcFollowFast[AGC_preset] * agcControlKp[AGC_preset] * control_error; multAgcTemp += agcFollowFast[AGC_preset] * agcControlKi[AGC_preset] * control_integrated; } else { // "normal zone" multAgcTemp = lastMultAgc + agcFollowSlow[AGC_preset] * agcControlKp[AGC_preset] * control_error; multAgcTemp += agcFollowSlow[AGC_preset] * agcControlKi[AGC_preset] * control_integrated; } // limit amplification again - PI controller sometimes "overshoots" //multAgcTemp = constrain(multAgcTemp, 0.015625f, 32.0f); // 1/64 < multAgcTemp < 32 if (multAgcTemp > 32.0f) multAgcTemp = 32.0f; if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f; } // NOW finally amplify the signal tmpAgc = sampleReal * multAgcTemp; // apply gain to signal if (fabsf(sampleReal) < 2.0f) tmpAgc = 0.0f; // apply squelch threshold //tmpAgc = constrain(tmpAgc, 0, 255); if (tmpAgc > 255) tmpAgc = 255.0f; // limit to 8bit if (tmpAgc < 1) tmpAgc = 0.0f; // just to be sure // update global vars ONCE - multAgc, sampleAGC, rawSampleAgc multAgc = multAgcTemp; if (micQuality > 0) { if (micQuality > 1) { rawSampleAgc = 0.95f * tmpAgc + 0.05f * (float)rawSampleAgc; // raw path sampleAgc += 0.95f * (tmpAgc - sampleAgc); // smooth path } else { rawSampleAgc = 0.70f * tmpAgc + 0.30f * (float)rawSampleAgc; // min filtering path sampleAgc += 0.70f * (tmpAgc - sampleAgc); } } else { #if defined(WLEDMM_FASTPATH) rawSampleAgc = 0.65f * tmpAgc + 0.35f * (float)rawSampleAgc; #else rawSampleAgc = 0.8f * tmpAgc + 0.2f * (float)rawSampleAgc; #endif // update smoothed AGC sample if (fabsf(tmpAgc) < 1.0f) sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero else sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path } sampleAgc = fabsf(sampleAgc); // // make sure we have a positive value last_soundAgc = soundAgc; } // agcAvg() // post-processing and filtering of MIC sample (micDataReal) from FFTcode() void getSample() { float sampleAdj; // Gain adjusted sample value float tmpSample; // An interim sample variable used for calculations. const float weighting = 0.18f; // Exponential filter weighting. Will be adjustable in a future release. const float weighting2 = 0.073f; // Exponential filter weighting, for rising signal (a bit more robust against spikes) const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function static bool isFrozen = false; static bool haveSilence = true; static unsigned long lastSoundTime = 0; // for delaying un-freeze static unsigned long startuptime = 0; // "fast freeze" mode: do not interfere during first 12 seconds (filter startup time) if (startuptime == 0) startuptime = millis(); // fast freeze mode - remember filter startup time if ((micLevelMethod < 1) || !isFrozen) { // following the input level, UNLESS mic Level was frozen micLev += (micDataReal-micLev) / 12288.0f; } if(micDataReal < (micLev-0.24)) { // MicLev above input signal: micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // always align MicLev to lowest input signal if (!haveSilence) isFrozen = true; // freeze mode: freeze micLevel so it cannot rise again } // Using an exponential filter to smooth out the signal. We'll add controls for this in a future release. float micInNoDC = fabsf(micDataReal - micLev); if ((micInNoDC > expAdjF) && (expAdjF > soundSquelch)) // MicIn rising, and above squelch threshold? expAdjF = (weighting2 * micInNoDC + (1.0f-weighting2) * expAdjF); // rise slower else expAdjF = (weighting * micInNoDC + (1.0f-weighting) * expAdjF); // fall faster expAdjF = fabsf(expAdjF); // Now (!) take the absolute value if ((micLevelMethod == 2) && !haveSilence && (expAdjF >= (1.5f * float(soundSquelch)))) isFrozen = true; // fast freeze mode: freeze micLevel once the volume rises 50% above squelch // simple noise gate if ((expAdjF <= soundSquelch) || ((soundSquelch == 0) && (expAdjF < 0.25f))) { expAdjF = 0.0f; micInNoDC = 0.0f; } if (expAdjF <= 0.5f) haveSilence = true; else { lastSoundTime = millis(); haveSilence = false; } // un-freeze micLev if (micLevelMethod == 0) isFrozen = false; if ((micLevelMethod == 1) && isFrozen && haveSilence && ((millis() - lastSoundTime) > 4000)) isFrozen = false; // normal freeze: 4 seconds silence needed if ((micLevelMethod == 2) && isFrozen && haveSilence && ((millis() - lastSoundTime) > 6000)) isFrozen = false; // fast freeze: 6 seconds silence needed if ((micLevelMethod == 2) && (millis() - startuptime < 12000)) isFrozen = false; // fast freeze: no freeze in first 12 seconds (filter startup phase) tmpSample = expAdjF; // Adjust the gain. with inputLevel adjustment. if (micQuality > 0) { sampleAdj = micInNoDC * sampleGain / 40.0f * inputLevel/128.0f + micInNoDC / 16.0f; // ... using unfiltered sample sampleReal = micInNoDC; } else { sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // ... using pre-filtered sample sampleReal = tmpSample; } sampleAdj = fmax(fmin(sampleAdj, 255.0f), 0.0f); // Question: why are we limiting the value to 8 bits ??? sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!! // keep "peak" sample, but decay value if current sample is below peak if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) { sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering #if 1 // another simple way to detect samplePeak - cannot detect beats, but reacts on peak volume if (((binNum < 12) || ((maxVol < 1))) && (millis() - timeOfPeak > 80) && (sampleAvg > 1)) { samplePeak = true; timeOfPeak = millis(); udpSamplePeak = true; } #endif } else { if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0)) sampleMax += 0.5f * (sampleReal - sampleMax); // over AGC Zone - get back quickly else sampleMax *= agcSampleDecay[AGC_preset]; // signal to zero --> 5-8sec } if (sampleMax < 0.5f) sampleMax = 0.0f; if (micQuality > 0) { if (micQuality > 1) sampleAvg += 0.95f * (sampleAdj - sampleAvg); else sampleAvg += 0.70f * (sampleAdj - sampleAvg); } else { #if defined(WLEDMM_FASTPATH) sampleAvg = ((sampleAvg * 11.0f) + sampleAdj) / 12.0f; // make reactions a bit more "crisp" in fastpath mode #else sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. #endif } sampleAvg = fabsf(sampleAvg); // make sure we have a positive value } // getSample() // current sensitivity, based on AGC gain (multAgc) float getSensitivity() { // start with AGC gain factor float tmpSound = multAgc; // experimental: this gives you a calculated "real gain" // if ((sampleAvg> 1.0) && (sampleReal > 0.05)) tmpSound = (float)sampleRaw / sampleReal; // calculate gain from sampleReal // else tmpSound = ((float)sampleGain/40.0f * (float)inputLevel/128.0f) + 1.0f/16.0f; // silence --> use values from user settings if (soundAgc == 0) tmpSound = ((float)sampleGain/40.0f * (float)inputLevel/128.0f) + 1.0f/16.0f; // AGC off -> use non-AGC gain from presets else tmpSound /= (float)sampleGain/40.0f + 1.0f/16.0f; // AGC ON -> scale value so 1 = middle value // scale to 0..255. Actually I'm not absolutely happy with this, but it works if (tmpSound > 1.0) tmpSound = sqrtf(tmpSound); if (tmpSound > 1.25) tmpSound = ((tmpSound-1.25f)/3.42f) +1.25f; // we have a value now that should be between 0 and 4 (representing gain 1/16 ... 16.0) return fminf(fmaxf(128.0*tmpSound -6.0f, 0), 255.0); // return scaled non-inverted value // "-6" to ignore values below 1/24 } // estimate sound pressure, based on some assumptions : // * sample max = 32676 -> Acoustic overload point --> 105db ==> 255 // * sample < squelch -> just above hearing level --> 5db ==> 0 // see https://en.wikipedia.org/wiki/Sound_pressure#Examples_of_sound_pressure // use with I2S digital microphones. Expect stupid values for analog in, and with Line-In !! float estimatePressure() const { // some constants constexpr float logMinSample = 0.8329091229351f; // ln(2.3) constexpr float sampleRangeMin = 2.3f; constexpr float logMaxSample = 10.1895683436f; // ln(32767 - 6144) constexpr float sampleRangeMax = 32767.0f - 6144.0f; // take the max sample from last I2S batch. float micSampleMax = fabsf(sampleReal); // from getSample() - nice results, however a bit distorted by MicLev processing //float micSampleMax = fabsf(micDataReal); // from FFTCode() - better source, but more flickering if (dmType == 0) micSampleMax *= 2.0f; // correction for ADC analog //if (dmType == 4) micSampleMax *= 16.0f; // correction for I2S Line-In if (dmType == 5) micSampleMax *= 2.0f; // correction for PDM if (dmType == 4) { // I2S Line-In. This is a dirty trick to make sound pressure look interesting for line-in (which doesn't have "sound pressure" as it is not a microphone) micSampleMax /= 11.0f; // reduce to max 128 micSampleMax *= micSampleMax; // blow up --> max 16000 } // make sure we are in expected ranges if(micSampleMax <= sampleRangeMin) return 0.0f; if(micSampleMax >= sampleRangeMax) return 255.0f; // apply logarithmic scaling float scaledvalue = logf(micSampleMax); scaledvalue = (scaledvalue - logMinSample) / (logMaxSample - logMinSample); // 0...1 return fminf(fmaxf(256.0f*scaledvalue, 0.0f), 255.0f); // scaled value } #endif /* Limits the dynamics of volumeSmth (= sampleAvg or sampleAgc). * does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc) */ // effects: Gravimeter, Gravcenter, Gravcentric, Noisefire, Plasmoid, Freqpixels, Freqwave, Gravfreq, (2D Swirl, 2D Waverly) void limitSampleDynamics(void) { const float bigChange = 196; // just a representative number - a large, expected sample value static unsigned long last_time = 0; static float last_volumeSmth = 0.0f; if (limiterOn == false) return; long delta_time = millis() - last_time; delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> silly 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(); } // MM experimental: limiter to smooth GEQ samples (only for UDP sound receiver mode) // target value (if gotNewSample) : fftCalc // last filtered value: fftAvg void limitGEQDynamics(bool gotNewSample) { constexpr float bigChange = 202; // just a representative number - a large, expected sample value constexpr float smooth = 0.8f; // a bit of filtering static unsigned long last_time = 0; if (limiterOn == false) return; if (gotNewSample) { // take new FFT samples as target values for(unsigned i=0; i < NUM_GEQ_CHANNELS; i++) { fftCalc[i] = fftResult[i]; fftResult[i] = fftAvg[i]; } } long delta_time = millis() - last_time; delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> silly lil hick-up float maxAttack = (attackTime <= 0) ? 255.0f : (bigChange * float(delta_time) / float(attackTime)); float maxDecay = (decayTime <= 0) ? -255.0f : (-bigChange * float(delta_time) / float(decayTime)); for(unsigned i=0; i < NUM_GEQ_CHANNELS; i++) { float deltaSample = fftCalc[i] - fftAvg[i]; if (deltaSample > maxAttack) deltaSample = maxAttack; if (deltaSample < maxDecay) deltaSample = maxDecay; deltaSample = deltaSample * smooth; fftAvg[i] = fmaxf(0.0f, fminf(255.0f, fftAvg[i] + deltaSample)); fftResult[i] = fftAvg[i]; } 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 == AUDIOSYNC_NONE)) return; // Sound Sync not enabled if (!(apActive || WLED_CONNECTED || interfacesInited)) { if (udpSyncConnected) { udpSyncConnected = false; fftUdp.stop(); receivedFormat = 0; DEBUGSR_PRINTLN(F("AR connectUDPSoundSync(): connection lost, UDP closed.")); } return; // neither AP nor other connections available } if (udpSyncConnected) return; // already connected if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds if (updateIsRunning) return; // don't reconnect during OTA // if we arrive here, we need a UDP connection but don't have one last_connection_attempt = millis(); connected(); // try to start UDP } #ifdef ARDUINO_ARCH_ESP32 void transmitAudioData() { if (!udpSyncConnected) return; static uint8_t frameCounter = 0; //DEBUGSR_PRINTLN("Transmitting UDP Mic Packet"); audioSyncPacket transmitData; memset(reinterpret_cast(&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.frameCounter = frameCounter; transmitData.zeroCrossingCount = zeroCrossingCount; for (int i = 0; i < NUM_GEQ_CHANNELS; i++) { transmitData.fftResult[i] = fftResult[i]; } // WLEDMM transmit soundPressure as 16 bit fixed point uint32_t pressure16bit = max(0.0f, soundPressure) * 256.0f; // convert to fixed point, remove negative values uint16_t pressInt = pressure16bit / 256; // integer part uint16_t pressFract = pressure16bit % 256; // faction part if (pressInt > 255) pressInt = 255; // saturation at 255 transmitData.pressure[0] = (uint8_t)pressInt; transmitData.pressure[1] = (uint8_t)pressFract; 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(&transmitData), sizeof(transmitData)); fftUdp.endPacket(); } frameCounter++; } // transmitAudioData() #endif static bool isValidUdpSyncVersion(const char *header) { return strncmp_P(header, UDP_SYNC_HEADER, 6) == 0; } static bool isValidUdpSyncVersion_v1(const char *header) { return strncmp_P(header, UDP_SYNC_HEADER_v1, 6) == 0; } bool decodeAudioData(int packetSize, uint8_t *fftBuff) { if((0 == packetSize) || (nullptr == fftBuff)) return false; // sanity check //audioSyncPacket *receivedPacket = reinterpret_cast(fftBuff); audioSyncPacket receivedPacket; memset(&receivedPacket, 0, sizeof(receivedPacket)); // start clean memcpy(&receivedPacket, fftBuff, min((unsigned)packetSize, (unsigned)sizeof(receivedPacket))); // don't violate alignment - thanks @willmmiles // validate sequence, discard out-of-sequence packets static uint8_t lastFrameCounter = 0; int lastReceivedFormat = receivedFormat; // add info for UI if ((receivedPacket.frameCounter > 0) && (lastFrameCounter > 0)) receivedFormat = 3; // v2+ else receivedFormat = 2; // v2 // check sequence bool sequenceOK = false; if ((int8_t)(receivedPacket.frameCounter - lastFrameCounter) > 0) sequenceOK = true; // 8-bit rollover-safe sequence check if (millis()- last_UDPTime >= AUDIOSYNC_IDLE_MS) sequenceOK = true; // receiver timed out - resync needed if (lastReceivedFormat < 2) sequenceOK = true; // first or second V2 packet - accept anything (prevents delay when re-enabling AR) if(audioSyncSequence == false) sequenceOK = true; // sequence checking disabled by user if((sequenceOK == false) && (receivedPacket.frameCounter != 0)) { // always accept "0" as the legacy value DEBUGSR_PRINTF("Skipping audio frame out of order or duplicated - %u vs %u\n", lastFrameCounter, receivedPacket.frameCounter); return false; // reject out-of sequence frame } else { lastFrameCounter = receivedPacket.frameCounter; } // update samples for effects volumeSmth = fmaxf(receivedPacket.sampleSmth, 0.0f); volumeRaw = fmaxf(receivedPacket.sampleRaw, 0.0f); #ifdef ARDUINO_ARCH_ESP32 // update internal samples sampleRaw = volumeRaw; sampleAvg = volumeSmth; rawSampleAgc = volumeRaw; sampleAgc = volumeSmth; multAgc = 1.0f; #endif // Only change samplePeak IF it's currently false. // If it's true already, then the animation still needs to respond. autoResetPeak(); if (!samplePeak) { samplePeak = receivedPacket.samplePeak >0 ? true:false; if (samplePeak) timeOfPeak = millis(); //userVar1 = samplePeak; } //These values are only computed by ESP32 for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket.fftResult[i]; my_magnitude = fmaxf(receivedPacket.FFT_Magnitude, 0.0f); FFT_Magnitude = my_magnitude; FFT_MajorPeak = constrain(receivedPacket.FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects #ifdef ARDUINO_ARCH_ESP32 FFT_MajPeakSmth = FFT_MajPeakSmth + 0.42f * (FFT_MajorPeak - FFT_MajPeakSmth); // simulate smooth value #endif agcSensitivity = 128.0f; // substitute - V2 format does not include this value zeroCrossingCount = receivedPacket.zeroCrossingCount; // WLEDMM extract soundPressure if ((receivedPacket.pressure[0] != 0) || (receivedPacket.pressure[1] != 0)) { // found something in gap "reserved2" soundPressure = float(receivedPacket.pressure[1]) / 256.0f; // fractional part soundPressure += float(receivedPacket.pressure[0]); // integer part } else { soundPressure = volumeSmth; // fallback } return true; } void decodeAudioData_v1(int packetSize, uint8_t *fftBuff) { audioSyncPacket_v1 *receivedPacket = reinterpret_cast(fftBuff); // update samples for effects volumeSmth = fmaxf(receivedPacket->sampleAgc, 0.0f); volumeRaw = volumeSmth; // V1 format does not have "raw" AGC sample #ifdef ARDUINO_ARCH_ESP32 // update internal samples sampleRaw = fmaxf(receivedPacket->sampleRaw, 0.0f); sampleAvg = fmaxf(receivedPacket->sampleAvg, 0.0f);; sampleAgc = volumeSmth; rawSampleAgc = volumeRaw; multAgc = 1.0f; #endif // Only change samplePeak IF it's currently false. // If it's true already, then the animation still needs to respond. autoResetPeak(); if (!samplePeak) { samplePeak = receivedPacket->samplePeak >0 ? true:false; if (samplePeak) timeOfPeak = millis(); //userVar1 = samplePeak; } //These values are only available on the ESP32 for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i]; my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0); FFT_Magnitude = my_magnitude; FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0, 11025.0); // restrict value to range expected by effects soundPressure = volumeSmth; // substitute - V1 format does not include this value agcSensitivity = 128.0f; // substitute - V1 format does not include this value } bool receiveAudioData( unsigned maxSamples) { // maxSamples = AR_UDP_FLUSH_ALL (255) means "purge complete input queue" if (!udpSyncConnected) return false; bool haveFreshData = false; size_t packetSize = 0; static uint8_t fftUdpBuffer[UDPSOUND_MAX_PACKET + 1] = {0}; // Loop to read available packets unsigned packetsReceived = 0; do { #if __cpp_exceptions try { packetSize = fftUdp.parsePacket(); } catch (...) { packetSize = 0; #ifdef ARDUINO_ARCH_ESP32 fftUdp.flush(); #endif DEBUG_PRINTLN(F("receiveAudioData: parsePacket out of memory exception caught!")); USER_FLUSH(); //continue; // don't skip to next iteration -> we are OOM } #else packetSize = fftUdp.parsePacket(); #endif #ifdef ARDUINO_ARCH_ESP32 if ((packetSize > 0) && ((packetSize < 5) || (packetSize > UDPSOUND_MAX_PACKET))) { fftUdp.flush(); continue; // Skip invalid packets -> next iteration } #endif if (packetSize == 0) break; // No more packets available --> exit loop if ((packetSize > 5) && (packetSize <= UDPSOUND_MAX_PACKET)) { fftUdp.read(fftUdpBuffer, packetSize); } // Process each received packet: last value will persist, intermediate ones needed to update sequence counters if (packetSize > 0) { if (packetSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftUdpBuffer))) { //receivedFormat = max(receivedFormat, 2); // format V2 or V2+ - will be set in decodeAudioData() haveFreshData |= decodeAudioData(packetSize, fftUdpBuffer); } else if (packetSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftUdpBuffer))) { decodeAudioData_v1(packetSize, fftUdpBuffer); receivedFormat = 1; haveFreshData = true; } else { receivedFormat = 0; // unknown format } } packetsReceived++; } while ((packetSize > 0) && ((packetsReceived < maxSamples) || (maxSamples == AR_UDP_FLUSH_ALL))); // repeat until we have read enough packets, or no more packets available #if defined(WLED_DEBUG) || defined(SR_DEBUG) if ((packetsReceived > 1) && haveFreshData) {DEBUGSR_PRINTF("AR UDP: dropped %d packets [%ums]\t%d maxDrop.\n", packetsReceived-1, millis() - last_UDPTime, maxSamples-1);} // for debugging #endif 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() override { disableSoundProcessing = true; // just to be sure isOOM = false; 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 = 12; 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; #ifdef ARDUINO_ARCH_ESP32 um_data->u_data[8] = &FFT_MajPeakSmth; // new um_data->u_type[8] = UMT_FLOAT; #else um_data->u_data[8] = &FFT_MajorPeak; // substitute for 8266 um_data->u_type[8] = UMT_FLOAT; #endif um_data->u_data[9] = &soundPressure; // used (New) um_data->u_type[9] = UMT_FLOAT; um_data->u_data[10] = &agcSensitivity; // used (New) - dummy value on 8266 um_data->u_type[10] = UMT_FLOAT; um_data->u_data[11] = &zeroCrossingCount; // for auto playlist usermod um_data->u_type[11] = UMT_UINT16; } #ifdef ARDUINO_ARCH_ESP32 // Reset I2S peripheral for good measure - not needed in esp-idf v4.4.x and later. #if ESP_IDF_VERSION < ESP_IDF_VERSION_VAL(4, 4, 0) 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 #endif delay(100); // Give that poor microphone some time to setup. #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) if ((i2sckPin == I2S_PIN_NO_CHANGE) && (i2ssdPin >= 0) && (i2swsPin >= 0) && ((dmType == 1) || (dmType == 4)) ) dmType = 51; // dummy user support: SCK == -1 --means--> PDM microphone #endif useInputFilter = 2; // default: DC blocker switch (dmType) { #if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) || defined(CONFIG_IDF_TARGET_ESP32S3) // stub cases for not-yet-supported I2S modes on other ESP32 chips case 0: //ADC analog #if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) case 5: //PDM Microphone case 51: //legacy PDM Microphone #endif #endif case 1: DEBUGSR_PRINT(F("AR: Generic I2S Microphone - ")); DEBUGSR_PRINTLN(F(I2S_MIC_CHANNEL_TEXT)); audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE); delay(100); if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin); break; case 2: DEBUGSR_PRINTLN(F("AR: ES7243 Microphone (right channel only).")); //useInputFilter = 0; // in case you need to disable low-cut software filtering audioSource = new ES7243(SAMPLE_RATE, BLOCK_SIZE); delay(100); // WLEDMM align global pins if ((sdaPin >= 0) && (i2c_sda < 0)) i2c_sda = sdaPin; // copy usermod prefs into globals (if globals not defined) if ((sclPin >= 0) && (i2c_scl < 0)) i2c_scl = sclPin; if (i2c_sda >= 0) sdaPin = -1; // -1 = use global if (i2c_scl >= 0) sclPin = -1; 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); //audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f/24.0f, false); // I2S SLAVE mode - does not work, unfortunately delay(100); if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin, mclkPin); break; #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) case 5: DEBUGSR_PRINT(F("AR: Generic PDM Microphone - ")); DEBUGSR_PRINTLN(F(I2S_PDM_MIC_CHANNEL_TEXT)); audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f/4.0f); useInputFilter = 1; // PDM bandpass filter - this reduces the noise floor on SPM1423 from 5% Vpp (~380) down to 0.05% Vpp (~5) delay(100); if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin); break; case 51: DEBUGSR_PRINT(F("AR: Legacy PDM Microphone - ")); DEBUGSR_PRINTLN(F(I2S_PDM_MIC_CHANNEL_TEXT)); audioSource = new I2SSource(SAMPLE_RATE, BLOCK_SIZE, 1.0f); useInputFilter = 1; // PDM bandpass filter delay(100); if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin); break; #endif case 6: #ifdef use_es8388_mic DEBUGSR_PRINTLN(F("AR: ES8388 Source (Mic)")); #else DEBUGSR_PRINTLN(F("AR: ES8388 Source (Line-In)")); #endif audioSource = new ES8388Source(SAMPLE_RATE, BLOCK_SIZE, 1.0f); //useInputFilter = 0; // to disable low-cut software filtering and restore previous behaviour delay(100); // WLEDMM align global pins if ((sdaPin >= 0) && (i2c_sda < 0)) i2c_sda = sdaPin; // copy usermod prefs into globals (if globals not defined) if ((sclPin >= 0) && (i2c_scl < 0)) i2c_scl = sclPin; if (i2c_sda >= 0) sdaPin = -1; // -1 = use global if (i2c_scl >= 0) sclPin = -1; if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin, mclkPin); break; case 7: #ifdef use_wm8978_mic DEBUGSR_PRINTLN(F("AR: WM8978 Source (Mic)")); #else DEBUGSR_PRINTLN(F("AR: WM8978 Source (Line-In)")); #endif audioSource = new WM8978Source(SAMPLE_RATE, BLOCK_SIZE, 1.0f); //useInputFilter = 0; // to disable low-cut software filtering and restore previous behaviour delay(100); // WLEDMM align global pins if ((sdaPin >= 0) && (i2c_sda < 0)) i2c_sda = sdaPin; // copy usermod prefs into globals (if globals not defined) if ((sclPin >= 0) && (i2c_scl < 0)) i2c_scl = sclPin; if (i2c_sda >= 0) sdaPin = -1; // -1 = use global if (i2c_scl >= 0) sclPin = -1; if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin, mclkPin); break; case 8: DEBUGSR_PRINTLN(F("AR: AC101 Source (Line-In)")); audioSource = new AC101Source(SAMPLE_RATE, BLOCK_SIZE, 1.0f); //useInputFilter = 0; // to disable low-cut software filtering and restore previous behaviour delay(100); // WLEDMM align global pins if ((sdaPin >= 0) && (i2c_sda < 0)) i2c_sda = sdaPin; // copy usermod prefs into globals (if globals not defined) if ((sclPin >= 0) && (i2c_scl < 0)) i2c_scl = sclPin; if (i2c_sda >= 0) sdaPin = -1; // -1 = use global if (i2c_scl >= 0) sclPin = -1; if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin, mclkPin); break; case 9: DEBUGSR_PRINTLN(F("AR: ES8311 Source (Mic)")); audioSource = new ES8311Source(SAMPLE_RATE, BLOCK_SIZE, 1.0f); //useInputFilter = 0; // to disable low-cut software filtering and restore previous behaviour delay(100); // WLEDMM align global pins if ((sdaPin >= 0) && (i2c_sda < 0)) i2c_sda = sdaPin; // copy usermod prefs into globals (if globals not defined) if ((sclPin >= 0) && (i2c_scl < 0)) i2c_scl = sclPin; if (i2c_sda >= 0) sdaPin = -1; // -1 = use global if (i2c_scl >= 0) sclPin = -1; if (audioSource) audioSource->initialize(i2swsPin, i2ssdPin, i2sckPin, mclkPin); break; case 255: // falls through case 254: // dummy "network receive only" driver if (audioSource) delete audioSource; audioSource = nullptr; disableSoundProcessing = true; audioSyncEnabled = AUDIOSYNC_REC; // force udp sound receive mode break; #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) // ADC over I2S is only possible on "classic" ESP32 case 0: default: DEBUGSR_PRINTLN(F("AR: Analog Microphone (left channel only).")); useInputFilter = 1; // PDM bandpass filter seems to work well for analog, too audioSource = new I2SAdcSource(SAMPLE_RATE, BLOCK_SIZE); delay(100); if (audioSource) audioSource->initialize(audioPin); break; #endif } delay(250); // give microphone enough time to initialise if (!audioSource && (dmType < 254)) enabled = false; // audio failed to initialise #endif if (enabled) onUpdateBegin(false); // create FFT task, and initialize network #ifdef ARDUINO_ARCH_ESP32 if (audioSource && FFT_Task == nullptr) enabled = false; // FFT task creation failed if((!audioSource) || (!audioSource->isInitialized())) { // audio source failed to initialize. Still stay "enabled", as there might be input arriving via UDP Sound Sync if (dmType < 254) { USER_PRINTLN(F("AR: Failed to initialize sound input driver. Please check input PIN settings."));} else { USER_PRINTLN(F("AR: No sound input driver configured - network receive only."));} disableSoundProcessing = true; } else { USER_PRINTLN(F("AR: sound input driver initialized successfully.")); } #endif if (enabled) disableSoundProcessing = false; // all good - enable audio processing // try to start UDP last_UDPTime = 0; receivedFormat = 0; delay(100); if (enabled) connectUDPSoundSync(); initDone = true; DEBUGSR_PRINT(F("AR: init done, enabled = ")); DEBUGSR_PRINTLN(enabled ? F("true.") : F("false.")); USER_FLUSH(); // dump audiosync data layout #if defined(SR_DEBUG) { audioSyncPacket data; USER_PRINTF("\naudioSyncPacket_v1 size = %d\n", sizeof(audioSyncPacket_v1)); // size 88 USER_PRINTF("audioSyncPacket size = %d\n", sizeof(audioSyncPacket)); // size 44 USER_PRINTF("| char header[6] offset = %2d size = %2d\n", offsetof(audioSyncPacket, header[0]), sizeof(data.header)); // offset 0 size 6 USER_PRINTF("| uint8_t pressure[2] offset = %2d size = %2d\n", offsetof(audioSyncPacket, pressure[0]), sizeof(data.pressure)); // offset 6 size 2 USER_PRINTF("| float sampleRaw offset = %2d size = %2d\n", offsetof(audioSyncPacket, sampleRaw), sizeof(data.sampleRaw)); // offset 8 size 4 USER_PRINTF("| float sampleSmth offset = %2d size = %2d\n", offsetof(audioSyncPacket, sampleSmth), sizeof(data.sampleSmth)); // offset 12 size 4 USER_PRINTF("| uint8_t samplePeak offset = %2d size = %2d\n", offsetof(audioSyncPacket, samplePeak), sizeof(data.samplePeak)); // offset 16 size 1 USER_PRINTF("| uint8_t frameCounter offset = %2d size = %2d\n", offsetof(audioSyncPacket, frameCounter), sizeof(data.frameCounter)); // offset 17 size 1 USER_PRINTF("| uint8_t fftResult[16] offset = %2d size = %2d\n", offsetof(audioSyncPacket, fftResult[0]), sizeof(data.fftResult)); // offset 18 size 16 USER_PRINTF("| uint16_t zeroCrossingCount offset = %2d size = %2d\n", offsetof(audioSyncPacket, zeroCrossingCount), sizeof(data.zeroCrossingCount)); // offset 34 size 2 USER_PRINTF("| float FFT_Magnitude offset = %2d size = %2d\n", offsetof(audioSyncPacket, FFT_Magnitude), sizeof(data.FFT_Magnitude));// offset 36 size 4 USER_PRINTF("| float FFT_MajorPeak offset = %2d size = %2d\n", offsetof(audioSyncPacket, FFT_MajorPeak), sizeof(data.FFT_MajorPeak));// offset 40 size 4 USER_PRINTLN(); USER_FLUSH(); } #endif #if defined(ARDUINO_ARCH_ESP32) && defined(SR_DEBUG) DEBUGSR_PRINTF("|| %-9s min free stack %d\n", pcTaskGetTaskName(NULL), uxTaskGetStackHighWaterMark(NULL)); //WLEDMM #endif } /* * connected() is called every time the WiFi is (re)connected * Use it to initialize network interfaces */ void connected() override { if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection udpSyncConnected = false; fftUdp.stop(); receivedFormat = 0; DEBUGSR_PRINTLN(F("AR connected(): old UDP connection closed.")); } if ((audioSyncPort > 0) && (audioSyncEnabled > AUDIOSYNC_NONE)) { #ifdef ARDUINO_ARCH_ESP32 udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort); #else udpSyncConnected = fftUdp.beginMulticast(WiFi.localIP(), IPAddress(239, 0, 0, 1), audioSyncPort); #endif receivedFormat = 0; if (udpSyncConnected) last_UDPTime = millis(); if (apActive && !(WLED_CONNECTED)) { DEBUGSR_PRINTLN(udpSyncConnected ? F("AR connected(): UDP: connected using AP.") : F("AR connected(): UDP is disconnected (AP).")); } else { DEBUGSR_PRINTLN(udpSyncConnected ? F("AR connected(): UDP: connected to WIFI.") : F("AR connected(): UDP is disconnected (Wifi).")); } } #if defined(ARDUINO_ARCH_ESP32) && defined(SR_DEBUG) DEBUGSR_PRINTF("|| %-9s min free stack %d\n", pcTaskGetTaskName(NULL), uxTaskGetStackHighWaterMark(NULL)); //WLEDMM #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() override { 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.isServicing() && (millis() - lastUMRun < 2)) return; // WLEDMM isServicing() is the critical part (be nice, but not too nice) // sound sync "receive or local" bool useNetworkAudio = false; if (audioSyncEnabled > AUDIOSYNC_SEND) { // we are in "receive" or "receive+local" mode if (udpSyncConnected && ((millis() - last_UDPTime) <= AUDIOSYNC_IDLE_MS)) useNetworkAudio = true; else useNetworkAudio = false; if (audioSyncEnabled == AUDIOSYNC_REC) useNetworkAudio = true; // don't fall back to local audio in standard "receive mode" } // 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 < AUDIOSYNC_REC)) { // 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; useNetworkAudio = false; } else { #if defined(ARDUINO_ARCH_ESP32) && defined(WLED_DEBUG) if ((disableSoundProcessing == true) && (audioSyncEnabled < AUDIOSYNC_REC) && 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 != AUDIOSYNC_REC)) lastUMRun = millis(); // just left "realtime mode" - update timekeeping disableSoundProcessing = false; } if (audioSyncEnabled == AUDIOSYNC_REC) disableSoundProcessing = true; // make sure everything is disabled IF in audio Receive mode if (audioSyncEnabled == AUDIOSYNC_SEND) disableSoundProcessing = false; // keep running audio IF we're in audio Transmit mode #ifdef ARDUINO_ARCH_ESP32 if (!audioSource || !audioSource->isInitialized()) { // no audio source disableSoundProcessing = true; if (audioSyncEnabled > AUDIOSYNC_SEND) useNetworkAudio = true; } if ((audioSyncEnabled == AUDIOSYNC_REC_PLUS) && useNetworkAudio) disableSoundProcessing = true; // UDP sound receiving - disable local audio #ifdef SR_DEBUG // debug info in case that task stack usage changes static unsigned int minLoopStackFree = UINT32_MAX; unsigned int stackFree = uxTaskGetStackHighWaterMark(NULL); if (minLoopStackFree > stackFree) { minLoopStackFree = stackFree; DEBUGSR_PRINTF("|| %-9s min free stack %d\n", pcTaskGetTaskName(NULL), minLoopStackFree); //WLEDMM } #endif // Only run the sampling code IF we're not in Receive mode or realtime mode if ((audioSyncEnabled != AUDIOSYNC_REC) && !disableSoundProcessing && !useNetworkAudio) { if (soundAgc > AGC_NUM_PRESETS) soundAgc = 0; // make sure that AGC preset is valid (to avoid array bounds violation) unsigned long t_now = millis(); // remember current time int userloopDelay = int(t_now - lastUMRun); if (lastUMRun == 0) userloopDelay=0; // startup - don't have valid data from last run. #if defined(SR_DEBUG) // complain when audio userloop has been delayed for long time. Currently, we need userloop running between 500 and 1500 times per second. // softhack007 disabled temporarily - avoid serial console spam with MANY LEDs and low FPS //if ((userloopDelay > /*23*/ 65) && !disableSoundProcessing && (audioSyncEnabled == AUDIOSYNC_NONE)) { //DEBUG_PRINTF("[AR userLoop] hiccup detected -> was inactive for last %d millis!\n", userloopDelay); //} #endif // run filters, and repeat in case of loop delays (hick-up compensation) if (userloopDelay <2) userloopDelay = 0; // minor glitch, no problem if (userloopDelay >200) userloopDelay = 200; // limit number of filter re-runs do { getSample(); // run microphone sampling filters agcAvg(t_now - userloopDelay); // Calculated the PI adjusted value as sampleAvg userloopDelay -= 2; // advance "simulated time" by 2ms } while (userloopDelay > 0); lastUMRun = t_now; // update time keeping // update samples for effects (raw, smooth) volumeSmth = (soundAgc) ? sampleAgc : sampleAvg; volumeRaw = (soundAgc) ? rawSampleAgc: sampleRaw; // update FFTMagnitude, taking into account AGC amplification my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects if (soundAgc) my_magnitude *= multAgc; if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute // get AGC sensitivity and sound pressure static unsigned long lastEstimate = 0; #ifdef WLEDMM_FASTPATH if (millis() - lastEstimate > 7) { #else if (millis() - lastEstimate > 12) { #endif lastEstimate = millis(); agcSensitivity = getSensitivity(); if (limiterOn) soundPressure = soundPressure + 0.38f * (estimatePressure() - soundPressure); // dynamics limiter on -> some smoothing else soundPressure = soundPressure + 0.95f * (estimatePressure() - soundPressure); // dynamics limiter on -> raw value } limitSampleDynamics(); } // if (!disableSoundProcessing) #endif autoResetPeak(); // auto-reset sample peak after strip minShowDelay if (!udpSyncConnected) udpSamplePeak = false; // reset UDP samplePeak while UDP is unconnected connectUDPSoundSync(); // ensure we have a connection - if needed // UDP Microphone Sync - receive mode if ((audioSyncEnabled & AUDIOSYNC_REC) && 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) { // DEBUG_PRINTF(F("AR reading at %d compared to %d max\n"), millis() - lastTime, delayMs); // TroyHacks unsigned timeElapsed = (millis() - last_UDPTime); unsigned maxReadSamples = timeElapsed / AR_UDP_READ_INTERVAL_MS; // estimate how many packets arrived since last receive maxReadSamples = max(1U, min(maxReadSamples, 20U)); // constrain to [1...20] = max 380ms drop // check if we should purge the receiving queue switch (audioSyncPurge) { case 0: maxReadSamples = 1; break; // never drop packets, unless new connection or timed out case 2: maxReadSamples = AR_UDP_FLUSH_ALL; break; // always drop - process latest packet only default: // falls through case 1: // auto drop when silence detected, or when receiver loop is slower than sender if (fabsf(volumeSmth) < 0.25f) maxReadSamples = AR_UDP_FLUSH_ALL; break; } if (receivedFormat == 0) maxReadSamples = AR_UDP_FLUSH_ALL; // new connection -> always flush queue if (timeElapsed >= AUDIOSYNC_IDLE_MS) maxReadSamples = AR_UDP_FLUSH_ALL; // too long since last run - always flush queue // try to get fresh data have_new_sample = receiveAudioData(maxReadSamples); if (have_new_sample) { last_UDPTime = millis(); useNetworkAudio = true; // UDP input arrived - use it } lastTime = millis(); } else { #ifdef ARDUINO_ARCH_ESP32 fftUdp.flush(); // WLEDMM: Flush this if we haven't read it. Does not work on 8266. #endif } if (useNetworkAudio) { 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 limitGEQDynamics(have_new_sample); // WLEDMM experimental: smooth FFT (GEQ) samples } } else { receivedFormat = 0; } if ( (audioSyncEnabled & AUDIOSYNC_REC) // receive mode && udpSyncConnected // connected && (receivedFormat > 0) // we actually received something in the past && ((millis() - last_UDPTime) > 25000)) { // close connection after 25sec idle udpSyncConnected = false; receivedFormat = 0; fftUdp.stop(); volumeSmth =0.0f; volumeRaw =0; my_magnitude = 0.1; FFT_Magnitude = 0.01; FFT_MajorPeak = 2; soundPressure = 1.0f; agcSensitivity = 64.0f; #ifdef ARDUINO_ARCH_ESP32 multAgc = 1; #endif DEBUGSR_PRINTLN(F("AR loop(): UDP closed due to inactivity.")); } #if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG) static unsigned long lastMicLoggerTime = 0; if (millis()-lastMicLoggerTime > 20) { lastMicLoggerTime = millis(); logAudio(); } #endif // Info Page: keep max sample from last 5 seconds #ifdef ARDUINO_ARCH_ESP32 if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) { sampleMaxTimer = millis(); maxSample5sec = (0.15 * maxSample5sec) + 0.85 *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing if (sampleAvg < 1) maxSample5sec = 0; // noise gate } else { if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume } #else // similar functionality for 8266 receive only - use VolumeSmth instead of raw sample data if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) { sampleMaxTimer = millis(); maxSample5sec = (0.15 * maxSample5sec) + 0.85 * volumeSmth; // reset, and start with some smoothing if (volumeSmth < 1.0f) maxSample5sec = 0; // noise gate if (maxSample5sec < 0.0f) maxSample5sec = 0; // avoid negative values } else { if (volumeSmth >= 1.0f) maxSample5sec = fmaxf(maxSample5sec, volumeRaw); // follow maximum volume } #endif #ifdef ARDUINO_ARCH_ESP32 //UDP Microphone Sync - transmit mode #if defined(WLEDMM_FASTPATH) if ((audioSyncEnabled & AUDIOSYNC_SEND) && (haveNewFFTResult || (millis() - lastTime > 24))) { // fastpath: send data once results are ready, or each 25ms as fallback (max sampling time is 23ms) #else if ((audioSyncEnabled & AUDIOSYNC_SEND) && (millis() - lastTime > 20)) { // standard: send data each 20ms #endif haveNewFFTResult = false; // reset notification // Only run the transmit code IF we're in Transmit mode transmitAudioData(); lastTime = millis(); } #endif } #if defined(_MoonModules_WLED_) && defined(WLEDMM_FASTPATH) void loop2(void) override { loop(); } #endif bool getUMData(um_data_t **data) override { if (!data || !enabled) return false; // no pointer provided by caller or not enabled -> exit *data = um_data; return true; } #ifdef ARDUINO_ARCH_ESP32 void onUpdateBegin(bool init) override { #ifdef WLED_DEBUG fftTime = sampleTime = filterTime = 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 don't process audio updateIsRunning = init; } #endif #ifdef ARDUINO_ARCH_ESP32 /** * handleButton() can be used to override default button behaviour. Returning true * will prevent button working in a default way. */ bool handleButton(uint8_t b) override { yield(); // crude way of determining if audio input is analog // better would be for AudioSource to implement getType() if (enabled && dmType == 0 && audioPin>=0 && (buttonType[b] == BTN_TYPE_ANALOG || buttonType[b] == BTN_TYPE_ANALOG_INVERTED) ) { return true; } return false; } #endif //////////////////////////// // Settings and Info Page // //////////////////////////// /* * addToJsonInfo() can be used to add custom entries to the /json/info part of the JSON API. * Creating an "u" object allows you to add custom key/value pairs to the Info section of the WLED web UI. * Below it is shown how this could be used for e.g. a light sensor */ void addToJsonInfo(JsonObject& root) override { #ifdef ARDUINO_ARCH_ESP32 char myStringBuffer[16]; // buffer for snprintf() - not used yet on 8266 #endif JsonObject user = root["u"]; if (user.isNull()) user = root.createNestedObject("u"); JsonArray infoArr = user.createNestedArray(FPSTR(_name)); String uiDomString = F(""); infoArr.add(uiDomString); if (enabled) { bool audioSyncIDLE = false; // true if sound sync is not receiving #ifdef ARDUINO_ARCH_ESP32 // audio sync status if ((audioSyncEnabled & AUDIOSYNC_REC) && (!udpSyncConnected || (millis() - last_UDPTime > AUDIOSYNC_IDLE_MS))) // connected and nothing received in 2.5sec audioSyncIDLE = true; if ((audioSource == nullptr) || (!audioSource->isInitialized())) // local audio not configured audioSyncIDLE = false; // Input Level Slider if (disableSoundProcessing == false) { // only show slider when audio processing is running if (soundAgc > 0) { infoArr = user.createNestedArray(F("GEQ Input Level")); // if AGC is on, this slider only affects fftResult[] frequencies // show slider value as a number float post_gain = (float)inputLevel/128.0f; if (post_gain < 1.0f) post_gain = ((post_gain -1.0f) * 0.8f) +1.0f; post_gain = roundf(post_gain * 100.0f); snprintf_P(myStringBuffer, 15, PSTR("%3.0f %%"), post_gain); infoArr.add(myStringBuffer); } else { infoArr = user.createNestedArray(F("Audio Input Level")); } uiDomString = F("
"); // infoArr.add(uiDomString); } #endif // The following can be used for troubleshooting user errors and is so not enclosed in #ifdef WLED_DEBUG // current Audio input infoArr = user.createNestedArray(F("Audio Source")); if ((audioSyncEnabled == AUDIOSYNC_REC) || (!audioSyncIDLE && (audioSyncEnabled == AUDIOSYNC_REC_PLUS))){ // UDP sound sync - receive mode infoArr.add(F("UDP sound sync")); if (udpSyncConnected) { if (millis() - last_UDPTime < AUDIOSYNC_IDLE_MS) infoArr.add(F(" - receiving")); else infoArr.add(F(" - idle")); } else { infoArr.add(F(" - no connection")); } #ifndef ARDUINO_ARCH_ESP32 // substitute for 8266 } else { infoArr.add(F("sound sync Off")); } #else // ESP32 only } else { // Analog or I2S digital input if (audioSource && (audioSource->isInitialized()) && !isOOM) { // audio source successfully configured if (audioSource->getType() == AudioSource::Type_I2SAdc) { infoArr.add(F("ADC analog")); } else { if (dmType != 51) { if (dmType == 5) infoArr.add(F("PDM digital")); else infoArr.add(F("I2S digital")); } else infoArr.add(F("legacy PDM")); } // input level or "silence" if (maxSample5sec > 1.0) { float my_usage = 100.0f * (maxSample5sec / 255.0f); snprintf_P(myStringBuffer, 15, PSTR(" - peak %3d%%"), int(my_usage)); infoArr.add(myStringBuffer); } else { infoArr.add(F(" - quiet")); } } else { // error during audio source setup infoArr.add(F("not initialized")); if (isOOM) infoArr.add(F(" - out of memory")); else if (dmType < 254) 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 == AUDIOSYNC_REC)) { 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 > 0) && (disableSoundProcessing == false) && !(audioSyncEnabled == AUDIOSYNC_REC)) { infoArr = user.createNestedArray(F("AGC Gain")); infoArr.add(roundf(multAgc*100.0f) / 100.0f); infoArr.add("x"); } #endif // UDP Sound Sync status infoArr = user.createNestedArray(F("UDP Sound Sync")); if (audioSyncEnabled) { if (audioSyncEnabled & AUDIOSYNC_SEND) { infoArr.add(F("send mode")); if ((udpSyncConnected) && (millis() - lastTime < AUDIOSYNC_IDLE_MS)) infoArr.add(F(" v2+")); } else if (audioSyncEnabled == AUDIOSYNC_REC) { infoArr.add(F("receive mode")); } else if (audioSyncEnabled == AUDIOSYNC_REC_PLUS) { infoArr.add(F("receive+local mode")); } } else infoArr.add("off"); if (audioSyncEnabled && !udpSyncConnected) infoArr.add(" (unconnected)"); if (audioSyncEnabled && udpSyncConnected && (millis() - last_UDPTime < AUDIOSYNC_IDLE_MS)) { if (receivedFormat == 1) infoArr.add(F(" v1")); if (receivedFormat == 2) infoArr.add(F(" v2")); if (receivedFormat == 3) { if (audioSyncSequence) infoArr.add(F(" v2+")); // Sequence checking enabled else infoArr.add(F(" v2")); } } #if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS) #ifdef ARDUINO_ARCH_ESP32 infoArr = user.createNestedArray(F("I2S cycle time")); infoArr.add(roundf(fftTaskCycle)/100.0f); infoArr.add(" ms"); infoArr = user.createNestedArray(F("Sampling time")); infoArr.add(roundf(sampleTime)/100.0f); infoArr.add(" ms"); infoArr = user.createNestedArray(F("Filtering time")); infoArr.add(roundf(filterTime)/100.0f); infoArr.add(" ms"); infoArr = user.createNestedArray(F("FFT time")); infoArr.add(roundf(fftTime)/100.0f); #ifdef FFT_USE_SLIDING_WINDOW unsigned timeBudget = doSlidingFFT ? (FFT_MIN_CYCLE) : fftTaskCycle / 115; #else unsigned timeBudget = (FFT_MIN_CYCLE); #endif if ((fftTime/100) >= timeBudget) // FFT time over budget -> I2S buffer will overflow infoArr.add("! ms"); else if ((fftTime/85 + filterTime/85 + sampleTime/85) >= timeBudget) // FFT time >75% of budget -> risk of instability infoArr.add(" ms!"); else infoArr.add(" ms"); DEBUGSR_PRINTF("AR I2S cycle time: %5.2f ms\n", roundf(fftTaskCycle)/100.0f); DEBUGSR_PRINTF("AR Sampling time : %5.2f ms\n", roundf(sampleTime)/100.0f); DEBUGSR_PRINTF("AR filter time : %5.2f ms\n", roundf(filterTime)/100.0f); DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", roundf(fftTime)/100.0f); #endif #endif } } /* * addToJsonState() can be used to add custom entries to the /json/state part of the JSON API (state object). * Values in the state object may be modified by connected clients */ void addToJsonState(JsonObject& root) override { 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) override { if (!initDone) return; // prevent crash on boot applyPreset() bool prevEnabled = enabled; JsonObject usermod = root[FPSTR(_name)]; if (!usermod.isNull()) { if (usermod[FPSTR(_enabled)].is()) { enabled = usermod[FPSTR(_enabled)].as(); if (prevEnabled != enabled) onUpdateBegin(!enabled); } #ifdef ARDUINO_ARCH_ESP32 if (usermod[FPSTR(_inputLvl)].is()) { inputLevel = min(255,max(0,usermod[FPSTR(_inputLvl)].as())); } #endif } } /* * addToConfig() can be used to add custom persistent settings to the cfg.json file in the "um" (usermod) object. * It will be called by WLED when settings are actually saved (for example, LED settings are saved) * If you want to force saving the current state, use serializeConfig() in your loop(). * * CAUTION: serializeConfig() will initiate a filesystem write operation. * It might cause the LEDs to stutter and will cause flash wear if called too often. * Use it sparingly and always in the loop, never in network callbacks! * * addToConfig() will make your settings editable through the Usermod Settings page automatically. * * Usermod Settings Overview: * - Numeric values are treated as floats in the browser. * - If the numeric value entered into the browser contains a decimal point, it will be parsed as a C float * before being returned to the Usermod. The float data type has only 6-7 decimal digits of precision, and * doubles are not supported, numbers will be rounded to the nearest float value when being parsed. * The range accepted by the input field is +/- 1.175494351e-38 to +/- 3.402823466e+38. * - If the numeric value entered into the browser doesn't contain a decimal point, it will be parsed as a * C int32_t (range: -2147483648 to 2147483647) before being returned to the usermod. * Overflows or underflows are truncated to the max/min value for an int32_t, and again truncated to the type * used in the Usermod when reading the value from ArduinoJson. * - Pin values can be treated differently from an integer value by using the key name "pin" * - "pin" can contain a single or array of integer values * - On the Usermod Settings page there is simple checking for pin conflicts and warnings for special pins * - Red color indicates a conflict. Yellow color indicates a pin with a warning (e.g. an input-only pin) * - Tip: use int8_t to store the pin value in the Usermod, so a -1 value (pin not set) can be used * * See usermod_v2_auto_save.h for an example that saves Flash space by reusing ArduinoJson key name strings * * If you need a dedicated settings page with custom layout for your Usermod, that takes a lot more work. * You will have to add the setting to the HTML, xml.cpp and set.cpp manually. * See the WLED Soundreactive fork (code and wiki) for reference. https://github.com/atuline/WLED * * I highly recommend checking out the basics of ArduinoJson serialization and deserialization in order to use custom settings! */ void addToConfig(JsonObject& root) override { JsonObject top = root.createNestedObject(FPSTR(_name)); top[FPSTR(_enabled)] = enabled; #ifdef ARDUINO_ARCH_ESP32 #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) JsonObject amic = top.createNestedObject(FPSTR(_analogmic)); amic["pin"] = audioPin; #endif JsonObject dmic = top.createNestedObject(FPSTR(_digitalmic)); dmic[F("type")] = dmType; // WLEDMM: align with globals I2C pins if ((dmType == 2) || (dmType == 6)) { // only for ES7243 and ES8388 if (i2c_sda >= 0) sdaPin = -1; // -1 = use global if (i2c_scl >= 0) sclPin = -1; // -1 = use global } JsonArray pinArray = dmic.createNestedArray("pin"); pinArray.add(i2ssdPin); pinArray.add(i2swsPin); pinArray.add(i2sckPin); pinArray.add(mclkPin); pinArray.add(sdaPin); pinArray.add(sclPin); JsonObject cfg = top.createNestedObject("config"); cfg[F("squelch")] = soundSquelch; cfg[F("gain")] = sampleGain; cfg[F("AGC")] = soundAgc; //WLEDMM: experimental settings JsonObject poweruser = top.createNestedObject("experiments"); poweruser[F("micLev")] = micLevelMethod; poweruser[F("Mic_Quality")] = micQuality; poweruser[F("freqDist")] = freqDist; //poweruser[F("freqRMS")] = averageByRMS; poweruser[F("FFT_Window")] = fftWindow; #ifdef FFT_USE_SLIDING_WINDOW poweruser[F("I2S_FastPath")] = doSlidingFFT; #endif JsonObject freqScale = top.createNestedObject("frequency"); freqScale[F("scale")] = FFTScalingMode; freqScale[F("profile")] = pinkIndex; //WLEDMM #endif JsonObject dynLim = top.createNestedObject("dynamics"); dynLim[F("limiter")] = limiterOn; dynLim[F("rise")] = attackTime; dynLim[F("fall")] = decayTime; JsonObject sync = top.createNestedObject("sync"); sync[F("port")] = audioSyncPort; sync[F("mode")] = audioSyncEnabled; sync[F("skip_old_data")] = audioSyncPurge; sync[F("check_sequence")] = audioSyncSequence; } /* * 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) override { JsonObject top = root[FPSTR(_name)]; bool configComplete = !top.isNull(); #ifdef ARDUINO_ARCH_ESP32 // remember previous values auto oldEnabled = enabled; auto oldDMType = dmType; auto oldI2SsdPin = i2ssdPin; auto oldI2SwsPin = i2swsPin; auto oldI2SckPin = i2sckPin; auto oldI2SmclkPin = mclkPin; #endif configComplete &= getJsonValue(top[FPSTR(_enabled)], enabled); #ifdef ARDUINO_ARCH_ESP32 #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) configComplete &= getJsonValue(top[FPSTR(_analogmic)]["pin"], audioPin); #else audioPin = -1; // MCU does not support analog mic #endif configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["type"], dmType); #if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) || defined(CONFIG_IDF_TARGET_ESP32S3) if (dmType == 0) dmType = SR_DMTYPE; // MCU does not support analog #if defined(CONFIG_IDF_TARGET_ESP32S2) || defined(CONFIG_IDF_TARGET_ESP32C3) if (dmType == 5) dmType = SR_DMTYPE; // MCU does not support PDM if (dmType == 51) dmType = SR_DMTYPE; // MCU does not support legacy PDM #endif #else if (dmType == 5) useInputFilter = 1; // enable filter for PDM if (dmType == 51) useInputFilter = 1; // switch on filter for legacy PDM #endif configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][0], i2ssdPin); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][1], i2swsPin); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][2], i2sckPin); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][3], mclkPin); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][4], sdaPin); configComplete &= getJsonValue(top[FPSTR(_digitalmic)]["pin"][5], sclPin); configComplete &= getJsonValue(top["config"][F("squelch")], soundSquelch); configComplete &= getJsonValue(top["config"][F("gain")], sampleGain); configComplete &= getJsonValue(top["config"][F("AGC")], soundAgc); //WLEDMM: experimental settings configComplete &= getJsonValue(top["experiments"][F("micLev")], micLevelMethod); configComplete &= getJsonValue(top["experiments"][F("Mic_Quality")], micQuality); configComplete &= getJsonValue(top["experiments"][F("freqDist")], freqDist); //configComplete &= getJsonValue(top["experiments"][F("freqRMS")], averageByRMS); configComplete &= getJsonValue(top["experiments"][F("FFT_Window")], fftWindow); #ifdef FFT_USE_SLIDING_WINDOW configComplete &= getJsonValue(top["experiments"][F("I2S_FastPath")], doSlidingFFT); #endif configComplete &= getJsonValue(top["frequency"][F("scale")], FFTScalingMode); configComplete &= getJsonValue(top["frequency"][F("profile")], pinkIndex); //WLEDMM #endif configComplete &= getJsonValue(top["dynamics"][F("limiter")], limiterOn); configComplete &= getJsonValue(top["dynamics"][F("rise")], attackTime); configComplete &= getJsonValue(top["dynamics"][F("fall")], decayTime); configComplete &= getJsonValue(top["sync"][F("port")], audioSyncPort); configComplete &= getJsonValue(top["sync"][F("mode")], audioSyncEnabled); configComplete &= getJsonValue(top["sync"][F("skip_old_data")], audioSyncPurge); configComplete &= getJsonValue(top["sync"][F("check_sequence")], audioSyncSequence); // WLEDMM notify user when a reboot is necessary #ifdef ARDUINO_ARCH_ESP32 if (initDone) { if ((audioSource != nullptr) && (oldDMType != dmType)) errorFlag = ERR_REBOOT_NEEDED; // changing mic type requires reboot if ( (audioSource != nullptr) && (enabled==true) && ((oldI2SsdPin != i2ssdPin) || (oldI2SwsPin != i2swsPin) || (oldI2SckPin != i2sckPin)) ) errorFlag = ERR_REBOOT_NEEDED; // changing mic pins requires reboot if ((audioSource != nullptr) && (oldI2SmclkPin != mclkPin)) errorFlag = ERR_REBOOT_NEEDED; // changing MCLK pin requires reboot if ((oldDMType != dmType) && (oldDMType == 0)) errorFlag = ERR_POWEROFF_NEEDED; // changing from analog mic requires power cycle if ((oldDMType != dmType) && (dmType == 0)) errorFlag = ERR_POWEROFF_NEEDED; // changing to analog mic requires power cycle } #endif return configComplete; } void appendConfigData() override { oappend(SET_F("ux='AudioReactive';")); // ux = shortcut for Audioreactive - fingers crossed that "ux" isn't already used as JS var, html post parameter or css style oappend(SET_F("uxp=ux+':digitalmic:pin[]';")); // uxp = shortcut for AudioReactive:digitalmic:pin[] oappend(SET_F("addInfo(ux+':help',0,'');")); #ifdef ARDUINO_ARCH_ESP32 //WLEDMM: add defaults #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) // -S3/-S2/-C3 don't support analog audio #ifdef AUDIOPIN oappend(SET_F("xOpt(ux+':analogmic:pin',1,' ⎌',")); oappendi(AUDIOPIN); oappend(");"); #endif oappend(SET_F("aOpt(ux+':analogmic:pin',1);")); //only analog options #endif oappend(SET_F("dd=addDropdown(ux,'digitalmic:type');")); #if SR_DMTYPE==254 oappend(SET_F("addOption(dd,'None - network receive only (⎌)',254);")); #else oappend(SET_F("addOption(dd,'None - network receive only',254);")); #endif #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) #if SR_DMTYPE==0 oappend(SET_F("addOption(dd,'Generic Analog (⎌)',0);")); #else oappend(SET_F("addOption(dd,'Generic Analog',0);")); #endif #endif #if SR_DMTYPE==1 oappend(SET_F("addOption(dd,'Generic I2S (⎌)',1);")); #else oappend(SET_F("addOption(dd,'Generic I2S',1);")); #endif #if SR_DMTYPE==2 oappend(SET_F("addOption(dd,'ES7243 (⎌)',2);")); #else oappend(SET_F("addOption(dd,'ES7243',2);")); #endif #if SR_DMTYPE==3 oappend(SET_F("addOption(dd,'SPH0654 (⎌)',3);")); #else oappend(SET_F("addOption(dd,'SPH0654',3);")); #endif #if SR_DMTYPE==4 oappend(SET_F("addOption(dd,'Generic I2S with Mclk (⎌)',4);")); #else oappend(SET_F("addOption(dd,'Generic I2S with Mclk',4);")); #endif #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) #if SR_DMTYPE==5 oappend(SET_F("addOption(dd,'Generic PDM (⎌)',5);")); #else oappend(SET_F("addOption(dd,'Generic PDM',5);")); #endif #if SR_DMTYPE==51 oappend(SET_F("addOption(dd,'.Legacy PDM ☾ (⎌)',51);")); #else oappend(SET_F("addOption(dd,'.Legacy PDM ☾',51);")); #endif #endif #if SR_DMTYPE==6 oappend(SET_F("addOption(dd,'ES8388 ☾ (⎌)',6);")); #else oappend(SET_F("addOption(dd,'ES8388 ☾',6);")); #endif #if SR_DMTYPE==7 oappend(SET_F("addOption(dd,'WM8978 ☾ (⎌)',7);")); #else oappend(SET_F("addOption(dd,'WM8978 ☾',7);")); #endif #if SR_DMTYPE==8 oappend(SET_F("addOption(dd,'AC101 ☾ (⎌)',8);")); #else oappend(SET_F("addOption(dd,'AC101 ☾',8);")); #endif #if SR_DMTYPE==9 oappend(SET_F("addOption(dd,'ES8311 ☾ (⎌)',9);")); #else oappend(SET_F("addOption(dd,'ES8311 ☾',9);")); #endif #ifdef SR_SQUELCH oappend(SET_F("addInfo(ux+':config:squelch',1,'⎌ ")); oappendi(SR_SQUELCH); oappend("');"); // 0 is field type, 1 is actual field #endif #ifdef SR_GAIN oappend(SET_F("addInfo(ux+':config:gain',1,'⎌ ")); oappendi(SR_GAIN); oappend("');"); // 0 is field type, 1 is actual field #endif oappend(SET_F("dd=addDropdown(ux,'config:AGC');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'Normal',1);")); oappend(SET_F("addOption(dd,'Vivid',2);")); oappend(SET_F("addOption(dd,'Lazy',3);")); //WLEDMM: experimental settings oappend(SET_F("xx='experiments';")); // shortcut oappend(SET_F("dd=addDropdown(ux,xx+':micLev');")); oappend(SET_F("addOption(dd,'Floating',0);")); oappend(SET_F("addOption(dd,'Freeze (⎌)',1);")); oappend(SET_F("addOption(dd,'Fast Freeze',2);")); oappend(SET_F("addInfo(ux+':'+xx+':micLev',1,'☾');")); oappend(SET_F("dd=addDropdown(ux,xx+':Mic_Quality');")); oappend(SET_F("addOption(dd,'average (standard)',0);")); oappend(SET_F("addOption(dd,'low noise',1);")); oappend(SET_F("addOption(dd,'perfect',2);")); oappend(SET_F("dd=addDropdown(ux,xx+':freqDist');")); oappend(SET_F("addOption(dd,'Normal (⎌)',0);")); oappend(SET_F("addOption(dd,'RightShift',1);")); oappend(SET_F("addInfo(ux+':'+xx+':freqDist',1,'☾');")); //oappend(SET_F("dd=addDropdown(ux,xx+':freqRMS');")); //oappend(SET_F("addOption(dd,'Off (⎌)',0);")); //oappend(SET_F("addOption(dd,'On',1);")); //oappend(SET_F("addInfo(ux+':experiments:freqRMS',1,'☾');")); oappend(SET_F("dd=addDropdown(ux,xx+':FFT_Window');")); oappend(SET_F("addOption(dd,'Blackman-Harris (MM standard)',0);")); oappend(SET_F("addOption(dd,'Hann (balanced)',1);")); oappend(SET_F("addOption(dd,'Nuttall (more accurate)',2);")); oappend(SET_F("addOption(dd,'Blackman',5);")); oappend(SET_F("addOption(dd,'Hamming',3);")); oappend(SET_F("addOption(dd,'Flat-Top (AC WLED, inaccurate)',4);")); #ifdef FFT_USE_SLIDING_WINDOW oappend(SET_F("dd=addDropdown(ux,xx+':I2S_FastPath');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'On (⎌)',1);")); oappend(SET_F("addInfo(ux+':'+xx+':I2S_FastPath',1,'☾');")); #endif oappend(SET_F("dd=addDropdown(ux,'dynamics:limiter');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'On',1);")); oappend(SET_F("addInfo(ux+':dynamics:limiter',0,' On ');")); // 0 is field type, 1 is actual field oappend(SET_F("addInfo(ux+':dynamics:rise',1,'ms (♪ effects only)');")); oappend(SET_F("addInfo(ux+':dynamics:fall',1,'ms (♪ effects only)');")); oappend(SET_F("dd=addDropdown(ux,'frequency:scale');")); oappend(SET_F("addOption(dd,'None',0);")); oappend(SET_F("addOption(dd,'Linear (Amplitude)',2);")); oappend(SET_F("addOption(dd,'Square Root (Energy)',3);")); oappend(SET_F("addOption(dd,'Logarithmic (Loudness)',1);")); //WLEDMM add defaults oappend(SET_F("dd=addDropdown(ux,'frequency:profile');")); #if SR_FREQ_PROF==0 oappend(SET_F("addOption(dd,'Generic Microphone (⎌)',0);")); #else oappend(SET_F("addOption(dd,'Generic Microphone',0);")); #endif #if SR_FREQ_PROF==1 oappend(SET_F("addOption(dd,'Generic Line-In (⎌)',1);")); #else oappend(SET_F("addOption(dd,'Generic Line-In',1);")); #endif #if SR_FREQ_PROF==5 oappend(SET_F("addOption(dd,'ICS-43434 (⎌)',5);")); #else oappend(SET_F("addOption(dd,'ICS-43434',5);")); #endif #if SR_FREQ_PROF==6 oappend(SET_F("addOption(dd,'ICS-43434 - big speakers (⎌)',6);")); #else oappend(SET_F("addOption(dd,'ICS-43434 - big speakers',6);")); #endif #if SR_FREQ_PROF==7 oappend(SET_F("addOption(dd,'SPM1423 (⎌)',7);")); #else oappend(SET_F("addOption(dd,'SPM1423',7);")); #endif #if SR_FREQ_PROF==2 oappend(SET_F("addOption(dd,'IMNP441 (⎌)',2);")); #else oappend(SET_F("addOption(dd,'IMNP441',2);")); #endif #if SR_FREQ_PROF==3 oappend(SET_F("addOption(dd,'IMNP441 - big speakers (⎌)',3);")); #else oappend(SET_F("addOption(dd,'IMNP441 - big speakers',3);")); #endif #if SR_FREQ_PROF==4 oappend(SET_F("addOption(dd,'IMNP441 - small speakers (⎌)',4);")); #else oappend(SET_F("addOption(dd,'IMNP441 - small speakers',4);")); #endif #if SR_FREQ_PROF==10 oappend(SET_F("addOption(dd,'flat - no adjustments (⎌)',10);")); #else oappend(SET_F("addOption(dd,'flat - no adjustments',10);")); #endif #if SR_FREQ_PROF==8 oappend(SET_F("addOption(dd,'userdefined #1 (⎌)',8);")); #else oappend(SET_F("addOption(dd,'userdefined #1',8);")); #endif #if SR_FREQ_PROF==9 oappend(SET_F("addOption(dd,'userdefined #2 (⎌)',9);")); #else oappend(SET_F("addOption(dd,'userdefined #2',9);")); #endif oappend(SET_F("addInfo(ux+':frequency:profile',1,'☾');")); #endif oappend(SET_F("dd=addDropdown(ux,'sync:mode');")); oappend(SET_F("addOption(dd,'Off',0);")); // AUDIOSYNC_NONE #ifdef ARDUINO_ARCH_ESP32 oappend(SET_F("addOption(dd,'Send',1);")); // AUDIOSYNC_SEND #endif oappend(SET_F("addOption(dd,'Receive',2);")); // AUDIOSYNC_REC #ifdef ARDUINO_ARCH_ESP32 oappend(SET_F("addOption(dd,'Receive or Local',6);")); // AUDIOSYNC_REC_PLUS #endif // Receiver drops old packets and processes the latest packet only oappend(SET_F("dd=addDropdown(ux,'sync:skip_old_data');")); oappend(SET_F("addOption(dd,'Never',0);")); oappend(SET_F("addOption(dd,'Auto (recommended)',1);")); // auto = drop during silence, or when last receive happened too long ago oappend(SET_F("addOption(dd,'Always',2);")); // check_sequence: Receiver skips out-of-sequence packets when enabled oappend(SET_F("dd=addDropdown(ux,'sync:check_sequence');")); oappend(SET_F("addOption(dd,'Off',0);")); oappend(SET_F("addOption(dd,'On',1);")); oappend(SET_F("addInfo(ux+':sync:check_sequence',1,'when receiving ☾
Sync audio data with other WLEDs');")); // must append this to the last field of 'sync' oappend(SET_F("addInfo(ux+':digitalmic:type',1,'requires reboot!');")); // 0 is field type, 1 is actual field #ifdef ARDUINO_ARCH_ESP32 oappend(SET_F("addInfo(uxp,0,'sd/data/dout','I2S SD');")); #ifdef I2S_SDPIN oappend(SET_F("xOpt(uxp,0,' ⎌',")); oappendi(I2S_SDPIN); oappend(");"); #endif oappend(SET_F("addInfo(uxp,1,'ws/clk/lrck','I2S WS');")); oappend(SET_F("dRO(uxp,1);")); // disable read only pins #ifdef I2S_WSPIN oappend(SET_F("xOpt(uxp,1,' ⎌',")); oappendi(I2S_WSPIN); oappend(");"); #endif oappend(SET_F("addInfo(uxp,2,'sck/bclk','I2S SCK');")); oappend(SET_F("dRO(uxp,2);")); // disable read only pins #ifdef I2S_CKPIN oappend(SET_F("xOpt(uxp,2,' ⎌',")); oappendi(I2S_CKPIN); oappend(");"); #endif oappend(SET_F("addInfo(uxp,3,'master clock','I2S MCLK');")); oappend(SET_F("dRO(uxp,3);")); // disable read only pins #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) oappend(SET_F("dOpt(uxp,3,2,2);")); //only use -1, 0, 1 or 3 oappend(SET_F("dOpt(uxp,3,4,39);")); //only use -1, 0, 1 or 3 #endif #ifdef MCLK_PIN oappend(SET_F("xOpt(uxp,3,' ⎌',")); oappendi(MCLK_PIN); oappend(");"); #endif oappend(SET_F("addInfo(uxp,4,'','I2C SDA');")); oappend(SET_F("rOpt(uxp,4,'use global (")); oappendi(i2c_sda); oappend(")',-1);"); #ifdef ES7243_SDAPIN oappend(SET_F("xOpt(uxp,4,' ⎌',")); oappendi(ES7243_SDAPIN); oappend(");"); #endif oappend(SET_F("addInfo(uxp,5,'','I2C SCL');")); oappend(SET_F("rOpt(uxp,5,'use global (")); oappendi(i2c_scl); oappend(")',-1);"); #ifdef ES7243_SCLPIN oappend(SET_F("xOpt(uxp,5,' ⎌',")); oappendi(ES7243_SCLPIN); oappend(");"); #endif oappend(SET_F("dRO(uxp,5);")); // disable read only pins #endif } /* * handleOverlayDraw() is called just before every show() (LED strip update frame) after effects have set the colors. * Use this to blank out some LEDs or set them to a different color regardless of the set effect mode. * Commonly used for custom clocks (Cronixie, 7 segment) */ //void handleOverlayDraw() //{ //strip.setPixelColor(0, RGBW32(0,0,0,0)) // set the first pixel to black //} /* * getId() allows you to optionally give your V2 usermod a 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() override { 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