From 7d7cebdd79ca13058b9abbe071ecd49b52e81135 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Sun, 11 Jan 2026 18:19:33 +0100 Subject: [PATCH] minor cleanups * add "override" to all methods that oveeride the base usermod class * fixing a minor copy-paste mistake * ovoid shadowing class attributes * spelling, grammar and other nitpick --- usermods/audioreactive/audio_reactive.h | 81 ++++++++++++------------- 1 file changed, 39 insertions(+), 42 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 2a67fd22..4a03e8c2 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -127,7 +127,7 @@ #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 its shared between tasks. +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 all received packets; 1: auto-purge old packets; 2:only process last received packets @@ -317,7 +317,7 @@ static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = { * Your HiFi equipment should receive its audio input from Line-In, SPDIF, HDMI, or another "undistorted" connection (like CDROM). * Try not to use Bluetooth or MP3 when playing the "pink noise" audio. BT-audio and MP3 both perform "acoustic adjustments" that we don't want now. - * SR WLED: enable AGC ("standard" or "lazy"), set squelch to a low level, check that LEDs don't reacts in silence. + * SR WLED: 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) @@ -328,12 +328,12 @@ static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = { * 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 to target. + * - 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 mutliplier you found for that column. + * - 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. @@ -404,7 +404,7 @@ constexpr float binWidth = SAMPLE_RATE / (float)samplesFFT; // frequency range o // lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2 #if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) // these options actually cause slow-down on -S2 (-S2 doesn't have floating point hardware) -//#define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and an a few other speedups - WLEDMM not faster on ESP32 +//#define FFT_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) @@ -675,11 +675,11 @@ void FFTcode(void * parameter) haveOldSamples = true; #endif - // find highest sample in the batch, and count zero crossings + // 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 our current mic sample - we take the max value from all samples that go into FFT + // 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) { @@ -706,7 +706,7 @@ void FFTcode(void * parameter) micReal_max = datMax; micReal_avg = datAvg / samplesFFT; #if 0 - // compute mix/max again after filering - usefull for filter debugging + // compute mix/max again after filtering - useful for filter debugging for (int i=0; i < samplesFFT; i++) { if (i==0) { datMin = datMax = vReal[i]; @@ -750,7 +750,7 @@ void FFTcode(void * parameter) 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 + 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 @@ -1233,11 +1233,11 @@ class AudioReactive : public Usermod { float soundPressure = 0; // Sound Pressure estimation, based on microphone raw readings. 0 ->5db, 255 ->105db // used to feed "Info" Page - unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket + 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 merasuring + #define CYCLE_SAMPLEMAX 3500 // time window for measuring // strings to reduce flash memory usage (used more than twice) static const char _name[]; @@ -1296,7 +1296,7 @@ class AudioReactive : public Usermod { // // Set true if wanting to see all the bands in their own vertical space on the Serial Plotter, false if wanting to see values in Serial Monitor const bool mapValuesToPlotterSpace = false; - // Set true to apply an auto-gain like setting to to the data (this hasn't been tested recently) + // Set true to apply an auto-gain like setting to the data (this hasn't been tested recently) const bool scaleValuesFromCurrentMaxVal = false; // prints the max value seen in the current data const bool printMaxVal = false; @@ -1343,7 +1343,7 @@ class AudioReactive : public Usermod { /* * A "PI controller" multiplier to automatically adjust sound sensitivity. * - * A few tricks are implemented so that sampleAgc does't only utilize 0% and 100%: + * 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 @@ -1576,12 +1576,12 @@ class AudioReactive : public Usermod { // * 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() { + float estimatePressure() const { // some constants constexpr float logMinSample = 0.8329091229351f; // ln(2.3) - constexpr float sampleMin = 2.3f; + constexpr float sampleRangeMin = 2.3f; constexpr float logMaxSample = 10.1895683436f; // ln(32767 - 6144) - constexpr float sampleMax = 32767.0f - 6144.0f; + 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 @@ -1589,18 +1589,18 @@ class AudioReactive : public Usermod { if (dmType == 0) micSampleMax *= 2.0f; // correction for ADC analog //if (dmType == 4) micSampleMax *= 16.0f; // correction for I2S Line-In if (dmType == 5) micSampleMax *= 2.0f; // correction for PDM - if (dmType == 4) { // I2S Line-In. This is a dirty trick to make sound pressure look interesting for line-in (which doesn't have "sound pressure" as its not a microphone) + 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 <= sampleMin) return 0.0f; - if(micSampleMax >= sampleMax) return 255.0f; + 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(256.0*scaledvalue, 0), 255.0); // scaled value + return fminf(fmaxf(256.0f*scaledvalue, 0.0f), 255.0f); // scaled value } #endif @@ -1765,7 +1765,7 @@ class AudioReactive : public Usermod { 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" - its the legacy value + 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 } @@ -1920,7 +1920,7 @@ class AudioReactive : public Usermod { * You can use it to initialize variables, sensors or similar. * It is called *AFTER* readFromConfig() */ - void setup() + void setup() override { disableSoundProcessing = true; // just to be sure if (!initDone) { @@ -2173,7 +2173,7 @@ class AudioReactive : public Usermod { * connected() is called every time the WiFi is (re)connected * Use it to initialize network interfaces */ - void connected() + void connected() override { if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection udpSyncConnected = false; @@ -2213,7 +2213,7 @@ class AudioReactive : public Usermod { * 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() + void loop() override { static unsigned long lastUMRun = millis(); @@ -2291,7 +2291,7 @@ class AudioReactive : public Usermod { 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. + // 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); @@ -2350,7 +2350,7 @@ class AudioReactive : public Usermod { unsigned timeElapsed = (millis() - last_UDPTime); unsigned maxReadSamples = timeElapsed / AR_UDP_AVG_SEND_RATE; // 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 the receiving queue + // 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 @@ -2447,12 +2447,12 @@ class AudioReactive : public Usermod { } #if defined(_MoonModules_WLED_) && defined(WLEDMM_FASTPATH) - void loop2(void) { + void loop2(void) override { loop(); } #endif - bool getUMData(um_data_t **data) + bool getUMData(um_data_t **data) override { if (!data || !enabled) return false; // no pointer provided by caller or not enabled -> exit *data = um_data; @@ -2461,7 +2461,7 @@ class AudioReactive : public Usermod { #ifdef ARDUINO_ARCH_ESP32 - void onUpdateBegin(bool init) + void onUpdateBegin(bool init) override { #ifdef WLED_DEBUG fftTime = sampleTime = filterTime = 0; @@ -2550,7 +2550,7 @@ class AudioReactive : public Usermod { * handleButton() can be used to override default button behaviour. Returning true * will prevent button working in a default way. */ - bool handleButton(uint8_t b) { + bool handleButton(uint8_t b) override { yield(); // crude way of determining if audio input is analog // better would be for AudioSource to implement getType() @@ -2573,7 +2573,7 @@ class AudioReactive : public Usermod { * 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) + void addToJsonInfo(JsonObject& root) override { #ifdef ARDUINO_ARCH_ESP32 char myStringBuffer[16]; // buffer for snprintf() - not used yet on 8266 @@ -2760,7 +2760,7 @@ class AudioReactive : public Usermod { * 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) + void addToJsonState(JsonObject& root) override { if (!initDone) return; // prevent crash on boot applyPreset() JsonObject usermod = root[FPSTR(_name)]; @@ -2775,7 +2775,7 @@ class AudioReactive : public Usermod { * 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) + void readFromJsonState(JsonObject& root) override { if (!initDone) return; // prevent crash on boot applyPreset() bool prevEnabled = enabled; @@ -2829,8 +2829,7 @@ class AudioReactive : public Usermod { * * I highly recommend checking out the basics of ArduinoJson serialization and deserialization in order to use custom settings! */ - void addToConfig(JsonObject& root) - { + void addToConfig(JsonObject& root) override { JsonObject top = root.createNestedObject(FPSTR(_name)); top[FPSTR(_enabled)] = enabled; #ifdef ARDUINO_ARCH_ESP32 @@ -2901,8 +2900,7 @@ class AudioReactive : public Usermod { * * This function is guaranteed to be called on boot, but could also be called every time settings are updated */ - bool readFromConfig(JsonObject& root) - { + bool readFromConfig(JsonObject& root) override { JsonObject top = root[FPSTR(_name)]; bool configComplete = !top.isNull(); @@ -2974,7 +2972,7 @@ class AudioReactive : public Usermod { if (initDone) { if ((audioSource != nullptr) && (oldDMType != dmType)) errorFlag = ERR_REBOOT_NEEDED; // changing mic type requires reboot if ( (audioSource != nullptr) && (enabled==true) - && ((oldI2SsdPin != i2ssdPin) || (oldI2SsdPin != i2ssdPin) || (oldI2SckPin != i2sckPin)) ) errorFlag = ERR_REBOOT_NEEDED; // changing mic pins requires reboot + && ((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 @@ -2984,8 +2982,7 @@ class AudioReactive : public Usermod { } - void appendConfigData() - { + 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,'');")); @@ -3264,10 +3261,10 @@ class AudioReactive : public Usermod { /* - * getId() allows you to optionally give your V2 usermod an unique ID (please define it in const.h!). + * 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() + uint16_t getId() override { return USERMOD_ID_AUDIOREACTIVE; }