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minor cleanups

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* 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
This commit is contained in:
Frank
2026-01-11 18:19:33 +01:00
parent d813bc430c
commit 7d7cebdd79
1 changed files with 39 additions and 42 deletions
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81
usermods/audioreactive/audio_reactive.h
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@@ -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_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) #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 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 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 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). * 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. * 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: 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: 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) * 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: * 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. * - 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. * - 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 square root scale: multiplier = (target * target) / (barheight * barheight)
* * in case of linear scale: multiplier = target / 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", * - 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 * Compile + upload
* Test your new profile (same procedure as above). Iterate the process to improve results. * 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 // lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3) #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) // 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 //#define FFT_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt - WLEDMM slower on ESP32
#endif #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(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; haveOldSamples = true;
#endif #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 float maxSample = 0.0f; // max sample from FFT batch
uint_fast16_t newZeroCrossingCount = 0; uint_fast16_t newZeroCrossingCount = 0;
for (int i=0; i < samplesFFT; i++) { 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 if ((vReal[i] <= (INT16_MAX - 1024)) && (vReal[i] >= (INT16_MIN + 1024))) { //skip extreme values - normally these are artefacts
#ifdef FFT_USE_SLIDING_WINDOW #ifdef FFT_USE_SLIDING_WINDOW
if (usingOldSamples) { if (usingOldSamples) {
@@ -706,7 +706,7 @@ void FFTcode(void * parameter)
micReal_max = datMax; micReal_max = datMax;
micReal_avg = datAvg / samplesFFT; micReal_avg = datAvg / samplesFFT;
#if 0 #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++) { for (int i=0; i < samplesFFT; i++) {
if (i==0) { if (i==0) {
datMin = datMax = vReal[i]; datMin = datMax = vReal[i];
@@ -750,7 +750,7 @@ void FFTcode(void * parameter)
break; break;
case 0: // falls through case 0: // falls through
default: 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 wc = 1.0f; // 2.7929062517 * 2.0
} }
#ifdef FFT_USE_SLIDING_WINDOW #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 float soundPressure = 0; // Sound Pressure estimation, based on microphone raw readings. 0 ->5db, 255 ->105db
// used to feed "Info" Page // 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) 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 float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds
unsigned long sampleMaxTimer = 0; // last time maxSample5sec was reset 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) // strings to reduce flash memory usage (used more than twice)
static const char _name[]; 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 // 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; 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; const bool scaleValuesFromCurrentMaxVal = false;
// prints the max value seen in the current data // prints the max value seen in the current data
const bool printMaxVal = false; const bool printMaxVal = false;
@@ -1343,7 +1343,7 @@ class AudioReactive : public Usermod {
/* /*
* A "PI controller" multiplier to automatically adjust sound sensitivity. * 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) * 0. don't amplify anything below squelch (but keep previous gain)
* 1. gain input = maximum signal observed in the last 5-10 seconds * 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 * 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 // * sample < squelch -> just above hearing level --> 5db ==> 0
// see https://en.wikipedia.org/wiki/Sound_pressure#Examples_of_sound_pressure // 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 !! // use with I2S digital microphones. Expect stupid values for analog in, and with Line-In !!
float estimatePressure() { float estimatePressure() const {
// some constants // some constants
constexpr float logMinSample = 0.8329091229351f; // ln(2.3) 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 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. // 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(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 == 0) micSampleMax *= 2.0f; // correction for ADC analog
//if (dmType == 4) micSampleMax *= 16.0f; // correction for I2S Line-In //if (dmType == 4) micSampleMax *= 16.0f; // correction for I2S Line-In
if (dmType == 5) micSampleMax *= 2.0f; // correction for PDM 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 /= 11.0f; // reduce to max 128
micSampleMax *= micSampleMax; // blow up --> max 16000 micSampleMax *= micSampleMax; // blow up --> max 16000
} }
// make sure we are in expected ranges // make sure we are in expected ranges
if(micSampleMax <= sampleMin) return 0.0f; if(micSampleMax <= sampleRangeMin) return 0.0f;
if(micSampleMax >= sampleMax) return 255.0f; if(micSampleMax >= sampleRangeMax) return 255.0f;
// apply logarithmic scaling // apply logarithmic scaling
float scaledvalue = logf(micSampleMax); float scaledvalue = logf(micSampleMax);
scaledvalue = (scaledvalue - logMinSample) / (logMaxSample - logMinSample); // 0...1 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 #endif
@@ -1765,7 +1765,7 @@ class AudioReactive : public Usermod {
if (millis()- last_UDPTime >= AUDIOSYNC_IDLE_MS) sequenceOK = true; // receiver timed out - resync needed 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 (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(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); DEBUGSR_PRINTF("Skipping audio frame out of order or duplicated - %u vs %u\n", lastFrameCounter, receivedPacket.frameCounter);
return false; // reject out-of sequence frame 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. * You can use it to initialize variables, sensors or similar.
* It is called *AFTER* readFromConfig() * It is called *AFTER* readFromConfig()
*/ */
void setup() void setup() override
{ {
disableSoundProcessing = true; // just to be sure disableSoundProcessing = true; // just to be sure
if (!initDone) { if (!initDone) {
@@ -2173,7 +2173,7 @@ class AudioReactive : public Usermod {
* connected() is called every time the WiFi is (re)connected * connected() is called every time the WiFi is (re)connected
* Use it to initialize network interfaces * Use it to initialize network interfaces
*/ */
void connected() void connected() override
{ {
if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection
udpSyncConnected = false; 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. * 2. Try to avoid using the delay() function. NEVER use delays longer than 10 milliseconds.
* Instead, use a timer check as shown here. * Instead, use a timer check as shown here.
*/ */
void loop() void loop() override
{ {
static unsigned long lastUMRun = millis(); 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 (lastUMRun == 0) userloopDelay=0; // startup - don't have valid data from last run.
#if defined(SR_DEBUG) #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 // softhack007 disabled temporarily - avoid serial console spam with MANY LEDs and low FPS
//if ((userloopDelay > /*23*/ 65) && !disableSoundProcessing && (audioSyncEnabled == AUDIOSYNC_NONE)) { //if ((userloopDelay > /*23*/ 65) && !disableSoundProcessing && (audioSyncEnabled == AUDIOSYNC_NONE)) {
//DEBUG_PRINTF("[AR userLoop] hiccup detected -> was inactive for last %d millis!\n", userloopDelay); //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 timeElapsed = (millis() - last_UDPTime);
unsigned maxReadSamples = timeElapsed / AR_UDP_AVG_SEND_RATE; // estimate how many packets arrived since last receive 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 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) { switch (audioSyncPurge) {
case 0: maxReadSamples = 1; break; // never drop packets, unless new connection or timed out 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 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) #if defined(_MoonModules_WLED_) && defined(WLEDMM_FASTPATH)
void loop2(void) { void loop2(void) override {
loop(); loop();
} }
#endif #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 if (!data || !enabled) return false; // no pointer provided by caller or not enabled -> exit
*data = um_data; *data = um_data;
@@ -2461,7 +2461,7 @@ class AudioReactive : public Usermod {
#ifdef ARDUINO_ARCH_ESP32 #ifdef ARDUINO_ARCH_ESP32
void onUpdateBegin(bool init) void onUpdateBegin(bool init) override
{ {
#ifdef WLED_DEBUG #ifdef WLED_DEBUG
fftTime = sampleTime = filterTime = 0; fftTime = sampleTime = filterTime = 0;
@@ -2550,7 +2550,7 @@ class AudioReactive : public Usermod {
* handleButton() can be used to override default button behaviour. Returning true * handleButton() can be used to override default button behaviour. Returning true
* will prevent button working in a default way. * will prevent button working in a default way.
*/ */
bool handleButton(uint8_t b) { bool handleButton(uint8_t b) override {
yield(); yield();
// crude way of determining if audio input is analog // crude way of determining if audio input is analog
// better would be for AudioSource to implement getType() // 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. * 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 * 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 #ifdef ARDUINO_ARCH_ESP32
char myStringBuffer[16]; // buffer for snprintf() - not used yet on 8266 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). * 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 * 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() if (!initDone) return; // prevent crash on boot applyPreset()
JsonObject usermod = root[FPSTR(_name)]; 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). * 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 * 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() if (!initDone) return; // prevent crash on boot applyPreset()
bool prevEnabled = enabled; 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! * 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)); JsonObject top = root.createNestedObject(FPSTR(_name));
top[FPSTR(_enabled)] = enabled; top[FPSTR(_enabled)] = enabled;
#ifdef ARDUINO_ARCH_ESP32 #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 * 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)]; JsonObject top = root[FPSTR(_name)];
bool configComplete = !top.isNull(); bool configComplete = !top.isNull();
@@ -2974,7 +2972,7 @@ class AudioReactive : public Usermod {
if (initDone) { if (initDone) {
if ((audioSource != nullptr) && (oldDMType != dmType)) errorFlag = ERR_REBOOT_NEEDED; // changing mic type requires reboot if ((audioSource != nullptr) && (oldDMType != dmType)) errorFlag = ERR_REBOOT_NEEDED; // changing mic type requires reboot
if ( (audioSource != nullptr) && (enabled==true) 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 ((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) && (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 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("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("uxp=ux+':digitalmic:pin[]';")); // uxp = shortcut for AudioReactive:digitalmic:pin[]
oappend(SET_F("addInfo(ux+':help',0,'<button onclick=\"location.href=&quot;https://mm.kno.wled.ge/soundreactive/Sound-Settings&quot;\" type=\"button\">?</button>');")); oappend(SET_F("addInfo(ux+':help',0,'<button onclick=\"location.href=&quot;https://mm.kno.wled.ge/soundreactive/Sound-Settings&quot;\" type=\"button\">?</button>');"));
@@ -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. * 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; return USERMOD_ID_AUDIOREACTIVE;
} }