From 0ac53d83532b048f12d8475f8680b63eaab3afb2 Mon Sep 17 00:00:00 2001 From: gaaat Date: Wed, 8 May 2024 15:33:51 +0200 Subject: [PATCH 1/5] initial port of MoonModules/WLED/pull/60 and related commit --- platformio.ini | 1 + usermods/audioreactive/audio_reactive.h | 229 ++++++++++++++++++------ 2 files changed, 179 insertions(+), 51 deletions(-) diff --git a/platformio.ini b/platformio.ini index 76c4c92d6..d77650bee 100644 --- a/platformio.ini +++ b/platformio.ini @@ -195,6 +195,7 @@ build_flags = ; decrease code cache size and increase IRAM to fit all pixel functions -D PIO_FRAMEWORK_ARDUINO_MMU_CACHE16_IRAM48 ;; in case of linker errors like "section `.text1' will not fit in region `iram1_0_seg'" ; -D PIO_FRAMEWORK_ARDUINO_MMU_CACHE16_IRAM48_SECHEAP_SHARED ;; (experimental) adds some extra heap, but may cause slowdown + -D USERMOD_AUDIOREACTIVE lib_deps = #https://github.com/lorol/LITTLEFS.git diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 442a651ea..2b43de1b2 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -1,6 +1,9 @@ #pragma once #include "wled.h" + +#ifdef ARDUINO_ARCH_ESP32 + #include #include @@ -8,11 +11,9 @@ #error This audio reactive usermod is not compatible with DMX Out. #endif -#ifndef ARDUINO_ARCH_ESP32 - #error This audio reactive usermod does not support the ESP8266. #endif -#if defined(WLED_DEBUG) || defined(SR_DEBUG) +#if defined(ARDUINO_ARCH_ESP32) && (defined(WLED_DEBUG) || defined(SR_DEBUG)) #include #endif @@ -57,6 +58,50 @@ #define MAX_PALETTES 3 +static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks. +static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value) +static bool udpSyncConnected = false; // UDP connection status -> true if connected to multicast group + +#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !! + +// audioreactive variables +#ifdef ARDUINO_ARCH_ESP32 +static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point +static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier +static float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate) +static float sampleAgc = 0.0f; // Smoothed AGC sample +static uint8_t soundAgc = 0; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value) +#endif +//static float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample +static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency +static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency +static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay() +static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData +static unsigned long timeOfPeak = 0; // time of last sample peak detection. +static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects + +// TODO: probably best not used by receive nodes +//static float agcSensitivity = 128; // AGC sensitivity estimation, based on agc gain (multAgc). calculated by getSensitivity(). range 0..255 + +// user settable parameters for limitSoundDynamics() +#ifdef UM_AUDIOREACTIVE_DYNAMICS_LIMITER_OFF +static bool limiterOn = false; // bool: enable / disable dynamics limiter +#else +static bool limiterOn = true; +#endif +static uint16_t attackTime = 80; // int: attack time in milliseconds. Default 0.08sec +static uint16_t decayTime = 1400; // int: decay time in milliseconds. Default 1.40sec + +// peak detection +#ifdef ARDUINO_ARCH_ESP32 +static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[]) - no used for 8266 receive-only mode +#endif +static void autoResetPeak(void); // peak auto-reset function +static uint8_t maxVol = 31; // (was 10) Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated) +static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated) + +#ifdef ARDUINO_ARCH_ESP32 + // use audio source class (ESP32 specific) #include "audio_source.h" constexpr i2s_port_t I2S_PORT = I2S_NUM_0; // I2S port to use (do not change !) @@ -74,18 +119,10 @@ static uint8_t inputLevel = 128; // UI slider value #else uint8_t sampleGain = SR_GAIN; // sample gain (config value) #endif -static uint8_t soundAgc = 1; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value) -static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value) -static bool udpSyncConnected = false; // UDP connection status -> true if connected to multicast group +//static uint8_t soundAgc = 1; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value) +//static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value) +//static bool udpSyncConnected = false; // UDP connection status -> true if connected to multicast group -// user settable parameters for limitSoundDynamics() -#ifdef UM_AUDIOREACTIVE_DYNAMICS_LIMITER_OFF -static bool limiterOn = false; // bool: enable / disable dynamics limiter -#else -static bool limiterOn = true; -#endif -static uint16_t attackTime = 80; // int: attack time in milliseconds. Default 0.08sec -static uint16_t decayTime = 1400; // int: decay time in milliseconds. Default 1.40sec // user settable options for FFTResult scaling static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized square root @@ -109,23 +146,23 @@ const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // // AGC presets end static AudioSource *audioSource = nullptr; -static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks. +//static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks. static bool useBandPassFilter = false; // if true, enables a bandpass filter 80Hz-16Khz to remove noise. Applies before FFT. // audioreactive variables shared with FFT task -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 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 // peak detection -static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay() -static uint8_t maxVol = 31; // 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) -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. -static void detectSamplePeak(void); // peak detection function (needs scaled FFT results in vReal[]) -static void autoResetPeak(void); // peak auto-reset function +// static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay() +// static uint8_t maxVol = 31; // 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) +// 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. +// static void detectSamplePeak(void); // peak detection function (needs scaled FFT results in vReal[]) +// static void autoResetPeak(void); // peak auto-reset function //////////////////// @@ -139,7 +176,7 @@ void FFTcode(void * parameter); // audio processing task: read samples, run static void runMicFilter(uint16_t numSamples, float *sampleBuffer); // pre-filtering of raw samples (band-pass) static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels); // post-processing and post-amp of GEQ channels -#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !! +//#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !! static TaskHandle_t FFT_Task = nullptr; @@ -147,9 +184,9 @@ static TaskHandle_t FFT_Task = nullptr; static float fftResultPink[NUM_GEQ_CHANNELS] = { 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f }; // globals and FFT Output variables shared with animations -static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency -static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency -static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects +//static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency +//static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency +//static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects #if defined(WLED_DEBUG) || defined(SR_DEBUG) static uint64_t fftTime = 0; static uint64_t sampleTime = 0; @@ -521,6 +558,8 @@ static void detectSamplePeak(void) { } } +#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. @@ -538,6 +577,8 @@ static void autoResetPeak(void) { class AudioReactive : public Usermod { private: +#ifdef ARDUINO_ARCH_ESP32 + #ifndef AUDIOPIN int8_t audioPin = -1; #else @@ -569,6 +610,7 @@ class AudioReactive : public Usermod { #else int8_t mclkPin = MCLK_PIN; #endif +#endif // new "V2" audiosync struct - 40 Bytes struct audioSyncPacket { @@ -612,10 +654,14 @@ class AudioReactive : public Usermod { const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED uint16_t audioSyncPort= 11988;// default port for UDP sound sync + bool updateIsRunning = false; // true during OTA. + +#ifdef ARDUINO_ARCH_ESP32 // used for AGC int last_soundAgc = -1; // used to detect AGC mode change (for resetting AGC internal error buffers) double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error + // variables used by getSample() and agcAvg() int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controller. @@ -624,6 +670,7 @@ class AudioReactive : public Usermod { float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC. int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel) int16_t rawSampleAgc = 0; // not smoothed AGC sample +#endif // variables used in effects float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample @@ -644,7 +691,9 @@ class AudioReactive : public Usermod { static const char _dynamics[]; static const char _frequency[]; static const char _inputLvl[]; +#if defined(ARDUINO_ARCH_ESP32) && !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) static const char _analogmic[]; +#endif static const char _digitalmic[]; static const char _addPalettes[]; static const char UDP_SYNC_HEADER[]; @@ -671,11 +720,13 @@ class AudioReactive : public Usermod { //PLOT_PRINT("sampleAgc:"); PLOT_PRINT(sampleAgc); PLOT_PRINT("\t"); //PLOT_PRINT("sampleAvg:"); PLOT_PRINT(sampleAvg); PLOT_PRINT("\t"); //PLOT_PRINT("sampleReal:"); PLOT_PRINT(sampleReal); PLOT_PRINT("\t"); + #ifdef ARDUINO_ARCH_ESP32 //PLOT_PRINT("micIn:"); PLOT_PRINT(micIn); PLOT_PRINT("\t"); //PLOT_PRINT("sample:"); PLOT_PRINT(sample); PLOT_PRINT("\t"); //PLOT_PRINT("sampleMax:"); PLOT_PRINT(sampleMax); PLOT_PRINT("\t"); //PLOT_PRINT("samplePeak:"); PLOT_PRINT((samplePeak!=0) ? 128:0); PLOT_PRINT("\t"); //PLOT_PRINT("multAgc:"); PLOT_PRINT(multAgc, 4); PLOT_PRINT("\t"); + #endif PLOT_PRINTLN(); #endif @@ -731,6 +782,7 @@ class AudioReactive : public Usermod { } // logAudio() +#ifdef ARDUINO_ARCH_ESP32 ////////////////////// // Audio Processing // ////////////////////// @@ -901,6 +953,7 @@ class AudioReactive : public Usermod { sampleAvg = fabsf(sampleAvg); // make sure we have a positive value } // getSample() +#endif /* Limits the dynamics of volumeSmth (= sampleAvg or sampleAgc). * does not affect FFTResult[] or volumeRaw ( = sample or rawSampleAgc) @@ -947,12 +1000,14 @@ class AudioReactive : public Usermod { if (udpSyncConnected) return; // already connected if (!(apActive || interfacesInited)) return; // neither AP nor other connections availeable if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds + if (updateIsRunning) return; // if we arrive here, we need a UDP connection but don't have one last_connection_attempt = millis(); connected(); // try to start UDP } +#ifdef ARDUINO_ARCH_ESP32 void transmitAudioData() { if (!udpSyncConnected) return; @@ -983,11 +1038,13 @@ class AudioReactive : public Usermod { return; } // transmitAudioData() +#endif + static bool isValidUdpSyncVersion(const char *header) { - return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0; + return strncmp_P(header, UDP_SYNC_HEADER, 6) == 0; } static bool isValidUdpSyncVersion_v1(const char *header) { - return strncmp_P(header, PSTR(UDP_SYNC_HEADER_v1), 6) == 0; + return strncmp_P(header, UDP_SYNC_HEADER_v1, 6) == 0; } void decodeAudioData(int packetSize, uint8_t *fftBuff) { @@ -995,12 +1052,14 @@ class AudioReactive : public Usermod { // 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(); @@ -1009,7 +1068,7 @@ class AudioReactive : public Usermod { if (samplePeak) timeOfPeak = millis(); //userVar1 = samplePeak; } - //These values are only available on the ESP32 + //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; @@ -1021,12 +1080,14 @@ class AudioReactive : public Usermod { // 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; + 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(); @@ -1112,6 +1173,9 @@ class AudioReactive : public Usermod { um_data->u_type[7] = UMT_BYTE; } + +#ifdef ARDUINO_ARCH_ESP32 + // Reset I2S peripheral for good measure i2s_driver_uninstall(I2S_NUM_0); // E (696) I2S: i2s_driver_uninstall(2006): I2S port 0 has not installed #if !defined(CONFIG_IDF_TARGET_ESP32C3) @@ -1189,10 +1253,12 @@ class AudioReactive : public Usermod { delay(250); // give microphone enough time to initialise if (!audioSource) enabled = false; // audio failed to initialise - if (enabled) onUpdateBegin(false); // create FFT task - if (FFT_Task == nullptr) enabled = false; // FFT task creation failed - if (enabled) disableSoundProcessing = false; // all good - enable audio processing +#endif + if (enabled) onUpdateBegin(false); // create FFT task, and initailize network + +#ifdef ARDUINO_ARCH_ESP32 + if (FFT_Task == nullptr) enabled = false; // FFT task creation failed if((!audioSource) || (!audioSource->isInitialized())) { // audio source failed to initialize. Still stay "enabled", as there might be input arriving via UDP Sound Sync #ifdef WLED_DEBUG DEBUG_PRINTLN(F("AR: Failed to initialize sound input driver. Please check input PIN settings.")); @@ -1201,7 +1267,8 @@ class AudioReactive : public Usermod { #endif disableSoundProcessing = true; } - +#endif + if (enabled) disableSoundProcessing = false; // all good - enable audio processing if (enabled) connectUDPSoundSync(); if (enabled && addPalettes) createAudioPalettes(); initDone = true; @@ -1220,7 +1287,7 @@ class AudioReactive : public Usermod { } if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) { - #ifndef ESP8266 + #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); @@ -1259,7 +1326,7 @@ class AudioReactive : public Usermod { ||(realtimeMode == REALTIME_MODE_ADALIGHT) ||(realtimeMode == REALTIME_MODE_ARTNET) ) ) // please add other modes here if needed { - #ifdef WLED_DEBUG + #if defined(ARDUINO_ARCH_ESP32) && defined(WLED_DEBUG) if ((disableSoundProcessing == false) && (audioSyncEnabled == 0)) { // we just switched to "disabled" DEBUG_PRINTLN(F("[AR userLoop] realtime mode active - audio processing suspended.")); DEBUG_PRINTF_P(PSTR(" RealtimeMode = %d; RealtimeOverride = %d\n"), int(realtimeMode), int(realtimeOverride)); @@ -1267,7 +1334,7 @@ class AudioReactive : public Usermod { #endif disableSoundProcessing = true; } else { - #ifdef WLED_DEBUG + #if defined(ARDUINO_ARCH_ESP32) && defined(WLED_DEBUG) if ((disableSoundProcessing == true) && (audioSyncEnabled == 0) && audioSource->isInitialized()) { // we just switched to "enabled" DEBUG_PRINTLN(F("[AR userLoop] realtime mode ended - audio processing resumed.")); DEBUG_PRINTF_P(PSTR(" RealtimeMode = %d; RealtimeOverride = %d\n"), int(realtimeMode), int(realtimeOverride)); @@ -1279,6 +1346,7 @@ class AudioReactive : public Usermod { if (audioSyncEnabled & 0x02) disableSoundProcessing = true; // make sure everything is disabled IF in audio Receive mode if (audioSyncEnabled & 0x01) disableSoundProcessing = false; // keep running audio IF we're in audio Transmit mode +#ifdef ARDUINO_ARCH_ESP32 if (!audioSource->isInitialized()) disableSoundProcessing = true; // no audio source @@ -1318,6 +1386,7 @@ class AudioReactive : public Usermod { limitSampleDynamics(); } // if (!disableSoundProcessing) +#endif autoResetPeak(); // auto-reset sample peak after strip minShowDelay if (!udpSyncConnected) udpSamplePeak = false; // reset UDP samplePeak while UDP is unconnected @@ -1351,6 +1420,7 @@ class AudioReactive : public Usermod { #endif // Info Page: keep max sample from last 5 seconds +#ifdef ARDUINO_ARCH_ESP32 if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) { sampleMaxTimer = millis(); maxSample5sec = (0.15f * maxSample5sec) + 0.85f *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing @@ -1358,13 +1428,25 @@ class AudioReactive : public Usermod { } else { if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume } +#else // similar functionality for 8266 receive only - use VolumeSmth instead of raw sample data + if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) { + sampleMaxTimer = millis(); + maxSample5sec = (0.15 * maxSample5sec) + 0.85 * volumeSmth; // reset, and start with some smoothing + if (volumeSmth < 1.0f) maxSample5sec = 0; // noise gate + if (maxSample5sec < 0.0f) maxSample5sec = 0; // avoid negative values + } else { + if (volumeSmth >= 1.0f) maxSample5sec = fmaxf(maxSample5sec, volumeRaw); // follow maximum volume + } +#endif +#ifdef ARDUINO_ARCH_ESP32 //UDP Microphone Sync - transmit mode if ((audioSyncEnabled & 0x01) && (millis() - lastTime > 20)) { // Only run the transmit code IF we're in Transmit mode transmitAudioData(); lastTime = millis(); } +#endif fillAudioPalettes(); } @@ -1377,7 +1459,7 @@ class AudioReactive : public Usermod { return true; } - +#ifdef ARDUINO_ARCH_ESP32 void onUpdateBegin(bool init) override { #ifdef WLED_DEBUG @@ -1426,9 +1508,32 @@ class AudioReactive : public Usermod { } micDataReal = 0.0f; // just to be sure if (enabled) disableSoundProcessing = false; + updateIsRunning = init; } +#else // reduced function for 8266 + void onUpdateBegin(bool init) + { + // gracefully suspend audio (if running) + disableSoundProcessing = true; + // reset sound data + volumeRaw = 0; volumeSmth = 0; + for(int i=(init?0:1); i 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. @@ -1446,7 +1551,7 @@ class AudioReactive : public Usermod { return false; } - +#endif //////////////////////////// // Settings and Info Page // //////////////////////////// @@ -1458,7 +1563,9 @@ class AudioReactive : public Usermod { */ void addToJsonInfo(JsonObject& root) override { - char myStringBuffer[16]; // buffer for snprintf() +#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"); @@ -1476,6 +1583,7 @@ class AudioReactive : public Usermod { infoArr.add(uiDomString); if (enabled) { +#ifdef ARDUINO_ARCH_ESP32 // Input Level Slider if (disableSoundProcessing == false) { // only show slider when audio processing is running if (soundAgc > 0) { @@ -1492,7 +1600,7 @@ class AudioReactive : public Usermod { 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 @@ -1508,6 +1616,11 @@ class AudioReactive : public Usermod { } 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())) { @@ -1552,7 +1665,7 @@ class AudioReactive : public Usermod { 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) { @@ -1571,6 +1684,7 @@ class AudioReactive : public Usermod { } #if defined(WLED_DEBUG) || defined(SR_DEBUG) + #ifdef ARDUINO_ARCH_ESP32 infoArr = user.createNestedArray(F("Sampling time")); infoArr.add(float(sampleTime)/100.0f); infoArr.add(" ms"); @@ -1587,6 +1701,7 @@ class AudioReactive : public Usermod { DEBUGSR_PRINTF("AR Sampling time: %5.2f ms\n", float(sampleTime)/100.0f); DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", float(fftTime)/100.0f); #endif + #endif } } @@ -1625,9 +1740,11 @@ class AudioReactive : public Usermod { if (!prevEnabled && enabled) createAudioPalettes(); } } +#ifdef ARDUINO_ARCH_ESP32 if (usermod[FPSTR(_inputLvl)].is()) { inputLevel = min(255,max(0,usermod[FPSTR(_inputLvl)].as())); } +#endif } if (root.containsKey(F("rmcpal")) && root[F("rmcpal")].as()) { // handle removal of custom palettes from JSON call so we don't break things @@ -1683,6 +1800,7 @@ class AudioReactive : public Usermod { top[FPSTR(_enabled)] = enabled; top[FPSTR(_addPalettes)] = addPalettes; +#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; @@ -1701,13 +1819,15 @@ class AudioReactive : public Usermod { cfg[F("gain")] = sampleGain; cfg[F("AGC")] = soundAgc; + JsonObject freqScale = top.createNestedObject(FPSTR(_frequency)); + freqScale[F("scale")] = FFTScalingMode; +#endif + JsonObject dynLim = top.createNestedObject(FPSTR(_dynamics)); dynLim[F("limiter")] = limiterOn; dynLim[F("rise")] = attackTime; dynLim[F("fall")] = decayTime; - JsonObject freqScale = top.createNestedObject(FPSTR(_frequency)); - freqScale[F("scale")] = FFTScalingMode; JsonObject sync = top.createNestedObject("sync"); sync["port"] = audioSyncPort; @@ -1740,6 +1860,7 @@ class AudioReactive : public Usermod { configComplete &= getJsonValue(top[FPSTR(_enabled)], enabled); configComplete &= getJsonValue(top[FPSTR(_addPalettes)], addPalettes); +#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 @@ -1763,12 +1884,12 @@ class AudioReactive : public Usermod { configComplete &= getJsonValue(top[FPSTR(_config)][F("gain")], sampleGain); configComplete &= getJsonValue(top[FPSTR(_config)][F("AGC")], soundAgc); + configComplete &= getJsonValue(top[FPSTR(_frequency)][F("scale")], FFTScalingMode); + configComplete &= getJsonValue(top[FPSTR(_dynamics)][F("limiter")], limiterOn); configComplete &= getJsonValue(top[FPSTR(_dynamics)][F("rise")], attackTime); configComplete &= getJsonValue(top[FPSTR(_dynamics)][F("fall")], decayTime); - - configComplete &= getJsonValue(top[FPSTR(_frequency)][F("scale")], FFTScalingMode); - +#endif configComplete &= getJsonValue(top["sync"]["port"], audioSyncPort); configComplete &= getJsonValue(top["sync"]["mode"], audioSyncEnabled); @@ -1783,6 +1904,7 @@ class AudioReactive : public Usermod { void appendConfigData() override { +#ifdef ARDUINO_ARCH_ESP32 oappend(SET_F("dd=addDropdown('AudioReactive','digitalmic:type');")); #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S3) oappend(SET_F("addOption(dd,'Generic Analog',0);")); @@ -1814,11 +1936,15 @@ class AudioReactive : public Usermod { 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);")); +#endif oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');")); oappend(SET_F("addOption(dd,'Off',0);")); +#ifdef ARDUINO_ARCH_ESP32 oappend(SET_F("addOption(dd,'Send',1);")); +#endif oappend(SET_F("addOption(dd,'Receive',2);")); +#ifdef ARDUINO_ARCH_ESP32 oappend(SET_F("addInfo('AudioReactive:digitalmic:type',1,'requires reboot!');")); // 0 is field type, 1 is actual field oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',0,'sd/data/dout','I2S SD');")); oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',1,'ws/clk/lrck','I2S WS');")); @@ -1828,6 +1954,7 @@ class AudioReactive : public Usermod { #else oappend(SET_F("addInfo('AudioReactive:digitalmic:pin[]',3,'master clock','I2S MCLK');")); #endif +#endif } From cec72419863a0ce8f19a02aa05bf9073b1302dcd Mon Sep 17 00:00:00 2001 From: gaaat Date: Wed, 8 May 2024 15:42:41 +0200 Subject: [PATCH 2/5] removed commented variables --- usermods/audioreactive/audio_reactive.h | 26 ------------------------- 1 file changed, 26 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index a0a657d68..a718011fe 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -119,10 +119,6 @@ static uint8_t inputLevel = 128; // UI slider value #else uint8_t sampleGain = SR_GAIN; // sample gain (config value) #endif -//static uint8_t soundAgc = 1; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value) -//static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive (config value) -//static bool udpSyncConnected = false; // UDP connection status -> true if connected to multicast group - // user settable options for FFTResult scaling static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized square root @@ -146,25 +142,8 @@ const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // // AGC presets end static AudioSource *audioSource = nullptr; -//static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks. static bool useBandPassFilter = false; // if true, enables a bandpass filter 80Hz-16Khz to remove noise. Applies before FFT. -// audioreactive variables shared with FFT task -// 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 - -// peak detection -// static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay() -// static uint8_t maxVol = 31; // 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) -// 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. -// static void detectSamplePeak(void); // peak detection function (needs scaled FFT results in vReal[]) -// static void autoResetPeak(void); // peak auto-reset function - - //////////////////// // Begin FFT Code // //////////////////// @@ -176,17 +155,12 @@ void FFTcode(void * parameter); // audio processing task: read samples, run static void runMicFilter(uint16_t numSamples, float *sampleBuffer); // pre-filtering of raw samples (band-pass) static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels); // post-processing and post-amp of GEQ channels -//#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !! - static TaskHandle_t FFT_Task = nullptr; // Table of multiplication factors so that we can even out the frequency response. static float fftResultPink[NUM_GEQ_CHANNELS] = { 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f }; // globals and FFT Output variables shared with animations -//static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency -//static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency -//static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects #if defined(WLED_DEBUG) || defined(SR_DEBUG) static uint64_t fftTime = 0; static uint64_t sampleTime = 0; From cc9db0220c87752ec5a0516a76d8827ffe405290 Mon Sep 17 00:00:00 2001 From: gaaat98 <67930088+gaaat98@users.noreply.github.com> Date: Fri, 12 Jul 2024 16:59:08 +0200 Subject: [PATCH 3/5] removed audioreactive usermod build flag --- platformio.ini | 1 - 1 file changed, 1 deletion(-) diff --git a/platformio.ini b/platformio.ini index 641f4baf3..fe8b3a278 100644 --- a/platformio.ini +++ b/platformio.ini @@ -196,7 +196,6 @@ build_flags = ; decrease code cache size and increase IRAM to fit all pixel functions -D PIO_FRAMEWORK_ARDUINO_MMU_CACHE16_IRAM48 ;; in case of linker errors like "section `.text1' will not fit in region `iram1_0_seg'" ; -D PIO_FRAMEWORK_ARDUINO_MMU_CACHE16_IRAM48_SECHEAP_SHARED ;; (experimental) adds some extra heap, but may cause slowdown - -D USERMOD_AUDIOREACTIVE lib_deps = #https://github.com/lorol/LITTLEFS.git From 3a8e19d1b49d7d5210afa496a2041d60bd4189ba Mon Sep 17 00:00:00 2001 From: Frank Date: Fri, 12 Jul 2024 22:09:52 +0200 Subject: [PATCH 4/5] audiosync receive improvements (maintainer edit) * fixed a few typo's in comments * fixed 8266 specific warning about 'comparison of integer expressions of different signedness' based on recommendations made by @willmmiles: * make sure that audioSyncPacket is the same size (44bytes) on all platforms * use static buffer for receiving (avoids heap fragmentation) * copy receive buffer to local audioSyncPacket struct - avoids alignment problems * esp32 only: to stay in sync with UDP, Udp.flush() is needed when Udp.parsePacket() is _not_ followed by Udp.read() --- usermods/audioreactive/audio_reactive.h | 64 ++++++++++++++----------- 1 file changed, 36 insertions(+), 28 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index a718011fe..fe946db9b 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -76,7 +76,7 @@ static uint8_t soundAgc = 0; // Automagic gain control: 0 - n static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay() -static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData +static 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. static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects @@ -94,7 +94,7 @@ static uint16_t decayTime = 1400; // int: decay time in milliseconds // peak detection #ifdef ARDUINO_ARCH_ESP32 -static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[]) - no used for 8266 receive-only mode +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) @@ -587,19 +587,21 @@ class AudioReactive : public Usermod { #endif #endif - // new "V2" audiosync struct - 40 Bytes - struct audioSyncPacket { - char header[6]; // 06 Bytes - float sampleRaw; // 04 Bytes - either "sampleRaw" or "rawSampleAgc" depending on soundAgc setting - float sampleSmth; // 04 Bytes - either "sampleAvg" or "sampleAgc" depending on soundAgc setting - uint8_t samplePeak; // 01 Bytes - 0 no peak; >=1 peak detected. In future, this will also provide peak Magnitude - uint8_t reserved1; // 01 Bytes - for future extensions - not used yet - uint8_t fftResult[16]; // 16 Bytes - float FFT_Magnitude; // 04 Bytes - float FFT_MajorPeak; // 04 Bytes + // new "V2" audiosync struct - 44 Bytes + struct __attribute__ ((packed)) audioSyncPacket { // "packed" ensures that there are no additional gaps + char header[6]; // 06 Bytes offset 0 + uint8_t reserved1[2]; // 02 Bytes, offset 6 - gap required by the compiler - not used yet + 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 reserved2; // 01 Bytes offset 17 - for future extensions - not used yet + uint8_t fftResult[16]; // 16 Bytes offset 18 + uint16_t reserved3; // 02 Bytes, offset 34 - gap required by the compiler - not used yet + float FFT_Magnitude; // 04 Bytes offset 36 + float FFT_MajorPeak; // 04 Bytes offset 40 }; - // old "V1" audiosync struct - 83 Bytes - for backwards compatibility + // old "V1" audiosync struct - 83 Bytes payload, 88 bytes total (with padding added by compiler) - for backwards compatibility struct audioSyncPacket_v1 { char header[6]; // 06 Bytes uint8_t myVals[32]; // 32 Bytes @@ -612,6 +614,8 @@ class AudioReactive : public Usermod { double FFT_MajorPeak; // 08 Bytes }; + #define UDPSOUND_MAX_PACKET 88 // max packet size for audiosync + // set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer) #ifdef UM_AUDIOREACTIVE_ENABLE bool enabled = true; @@ -997,7 +1001,6 @@ class AudioReactive : public Usermod { transmitData.sampleSmth = (soundAgc) ? sampleAgc : sampleAvg; transmitData.samplePeak = udpSamplePeak ? 1:0; udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it - transmitData.reserved1 = 0; for (int i = 0; i < NUM_GEQ_CHANNELS; i++) { transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254); @@ -1023,10 +1026,13 @@ class AudioReactive : public Usermod { } void decodeAudioData(int packetSize, uint8_t *fftBuff) { - 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# + // update samples for effects - volumeSmth = fmaxf(receivedPacket->sampleSmth, 0.0f); - volumeRaw = fmaxf(receivedPacket->sampleRaw, 0.0f); + volumeSmth = fmaxf(receivedPacket.sampleSmth, 0.0f); + volumeRaw = fmaxf(receivedPacket.sampleRaw, 0.0f); #ifdef ARDUINO_ARCH_ESP32 // update internal samples sampleRaw = volumeRaw; @@ -1039,15 +1045,15 @@ class AudioReactive : public Usermod { // If it's true already, then the animation still needs to respond. autoResetPeak(); if (!samplePeak) { - samplePeak = receivedPacket->samplePeak >0 ? true:false; + 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); + 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 + FFT_MajorPeak = constrain(receivedPacket.FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects } void decodeAudioData_v1(int packetSize, uint8_t *fftBuff) { @@ -1084,9 +1090,12 @@ class AudioReactive : public Usermod { bool haveFreshData = false; size_t packetSize = fftUdp.parsePacket(); - if (packetSize > 5) { +#ifdef ARDUINO_ARCH_ESP32 + if ((packetSize > 0) && ((packetSize < 5) || (packetSize > UDPSOUND_MAX_PACKET))) fftUdp.flush(); // discard invalid packets (too small or too big) - only works on esp32 +#endif + if ((packetSize > 5) && (packetSize <= UDPSOUND_MAX_PACKET)) { //DEBUGSR_PRINTLN("Received UDP Sync Packet"); - uint8_t fftBuff[packetSize]; + static uint8_t fftBuff[UDPSOUND_MAX_PACKET+1] = { 0 }; // static buffer for receiving, to reuse the same memory and avoid heap fragmentation fftUdp.read(fftBuff, packetSize); // VERIFY THAT THIS IS A COMPATIBLE PACKET @@ -1229,7 +1238,7 @@ class AudioReactive : public Usermod { if (!audioSource) enabled = false; // audio failed to initialise #endif - if (enabled) onUpdateBegin(false); // create FFT task, and initailize network + if (enabled) onUpdateBegin(false); // create FFT task, and initialize network #ifdef ARDUINO_ARCH_ESP32 @@ -1243,7 +1252,7 @@ class AudioReactive : public Usermod { disableSoundProcessing = true; } #endif - if (enabled) disableSoundProcessing = false; // all good - enable audio processing + if (enabled) disableSoundProcessing = false; // all good - enable audio processing if (enabled) connectUDPSoundSync(); if (enabled && addPalettes) createAudioPalettes(); initDone = true; @@ -1803,7 +1812,6 @@ class AudioReactive : public Usermod { dynLim[F("rise")] = attackTime; dynLim[F("fall")] = decayTime; - JsonObject sync = top.createNestedObject("sync"); sync["port"] = audioSyncPort; sync["mode"] = audioSyncEnabled; @@ -2008,8 +2016,8 @@ CRGB AudioReactive::getCRGBForBand(int x, int pal) { void AudioReactive::fillAudioPalettes() { if (!palettes) return; size_t lastCustPalette = strip.customPalettes.size(); - if (lastCustPalette >= palettes) lastCustPalette -= palettes; - for (size_t pal=0; pal= palettes) lastCustPalette -= palettes; + for (int pal=0; pal Date: Sat, 13 Jul 2024 09:55:59 +0200 Subject: [PATCH 5/5] use fixes-size stack buffer ... protected against array overflow due to previous "if (packetSize <= UDPSOUND_MAX_PACKET)" --- usermods/audioreactive/audio_reactive.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index fe946db9b..088ac880b 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -1095,7 +1095,7 @@ class AudioReactive : public Usermod { #endif if ((packetSize > 5) && (packetSize <= UDPSOUND_MAX_PACKET)) { //DEBUGSR_PRINTLN("Received UDP Sync Packet"); - static uint8_t fftBuff[UDPSOUND_MAX_PACKET+1] = { 0 }; // static buffer for receiving, to reuse the same memory and avoid heap fragmentation + uint8_t fftBuff[UDPSOUND_MAX_PACKET+1] = { 0 }; // fixed-size buffer for receiving (stack), to avoid heap fragmentation caused by variable sized arrays fftUdp.read(fftBuff, packetSize); // VERIFY THAT THIS IS A COMPATIBLE PACKET