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Merge pull request #314 from MoonModules/AR_udpReceivePlus

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AR Sound receive: better handling of outdated packets
* new user option "skip old packets" with default "auto".
* packets are auto-skipped when the receiver is behind, typicially due to slow effects which slow down the main loop.
* improvements to sequence checking
This commit is contained in:
Frank Möhle
2026-01-11 20:01:21 +01:00
committed by GitHub
parent 428becc301 6625e2bb12
commit bf5edbae97
1 changed files with 98 additions and 74 deletions
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172
usermods/audioreactive/audio_reactive.h
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@@ -127,9 +127,10 @@
#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 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
#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !!
@@ -316,7 +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)
@@ -327,12 +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.
@@ -403,7 +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)
@@ -674,11 +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) {
@@ -705,7 +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 min/max again after filtering - useful for filter debugging
for (int i=0; i < samplesFFT; i++) {
if (i==0) {
datMin = datMax = vReal[i];
@@ -749,7 +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
@@ -1188,7 +1189,9 @@ class AudioReactive : public Usermod {
double FFT_MajorPeak; // 08 Bytes
};
#define UDPSOUND_MAX_PACKET 96 // max packet size for audiosync, with a bit of "headroom"
#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)
@@ -1230,11 +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[];
@@ -1293,7 +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;
@@ -1340,7 +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
@@ -1573,12 +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
@@ -1586,18 +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
@@ -1751,24 +1754,18 @@ class AudioReactive : public Usermod {
// 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
// Simpler 8-bit rollover-safe sequence check:
// (int8_t)(cur - prev) > 0 => cur is ahead of prev by 1..127
// 0 => duplicate,
// < 0 => older
// bool sequenceOK = !audioSyncSequence || receivedPacket.frameCounter == 0 || // always accept legacy "0"
// ((int8_t)(receivedPacket.frameCounter - lastFrameCounter) > 0);
// check sequence
bool sequenceOK = false;
if(receivedPacket.frameCounter > lastFrameCounter) sequenceOK = true; // sequence OK
if((lastFrameCounter < 12) && (receivedPacket.frameCounter > 248)) sequenceOK = false; // prevent sequence "roll-back" due to late packets (1->254)
if((lastFrameCounter > 248) && (receivedPacket.frameCounter < 12)) sequenceOK = true; // handle roll-over (255 -> 0)
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" - 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
}
@@ -1848,15 +1845,15 @@ class AudioReactive : public Usermod {
agcSensitivity = 128.0f; // substitute - V1 format does not include this value
}
bool receiveAudioData() {
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};
size_t lastValidPacketSize = 0;
// Loop to read all available packets
while (true) {
// Loop to read available packets
unsigned packetsReceived = 0;
do {
#if __cpp_exceptions
try {
packetSize = fftUdp.parsePacket();
@@ -1867,7 +1864,7 @@ class AudioReactive : public Usermod {
#endif
DEBUG_PRINTLN(F("receiveAudioData: parsePacket out of memory exception caught!"));
USER_FLUSH();
continue; // Skip to next iteration
//continue; // don't skip to next iteration -> we are OOM
}
#else
packetSize = fftUdp.parsePacket();
@@ -1876,31 +1873,36 @@ class AudioReactive : public Usermod {
#ifdef ARDUINO_ARCH_ESP32
if ((packetSize > 0) && ((packetSize < 5) || (packetSize > UDPSOUND_MAX_PACKET))) {
fftUdp.flush();
continue; // Skip invalid packets
continue; // Skip invalid packets -> next iteration
}
#endif
if (packetSize == 0) break; // No more packets available
if (packetSize == 0) break; // No more packets available --> exit loop
if ((packetSize > 5) && (packetSize <= UDPSOUND_MAX_PACKET)) {
fftUdp.read(fftUdpBuffer, packetSize);
lastValidPacketSize = packetSize;
}
}
// Process only the last valid packet
if (lastValidPacketSize > 0) {
if (lastValidPacketSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftUdpBuffer))) {
receivedFormat = 2;
haveFreshData = decodeAudioData(lastValidPacketSize, fftUdpBuffer);
} else if (lastValidPacketSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftUdpBuffer))) {
decodeAudioData_v1(lastValidPacketSize, fftUdpBuffer);
receivedFormat = 1;
haveFreshData = true;
} else {
receivedFormat = 0; // unknown format
// 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;
}
@@ -1916,7 +1918,7 @@ 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) {
@@ -2169,7 +2171,7 @@ 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;
@@ -2209,7 +2211,7 @@ 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();
@@ -2287,7 +2289,7 @@ 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);
@@ -2342,7 +2344,25 @@ class AudioReactive : public Usermod {
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
have_new_sample = receiveAudioData();
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
@@ -2425,12 +2445,12 @@ 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;
@@ -2439,7 +2459,7 @@ class AudioReactive : public Usermod {
#ifdef ARDUINO_ARCH_ESP32
void onUpdateBegin(bool init)
void onUpdateBegin(bool init) override
{
#ifdef WLED_DEBUG
fftTime = sampleTime = filterTime = 0;
@@ -2528,7 +2548,7 @@ 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()
@@ -2551,7 +2571,7 @@ 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
@@ -2738,7 +2758,7 @@ 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)];
@@ -2753,7 +2773,7 @@ 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;
@@ -2807,8 +2827,7 @@ 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
@@ -2859,6 +2878,7 @@ class AudioReactive : public Usermod {
JsonObject sync = top.createNestedObject("sync");
sync[F("port")] = audioSyncPort;
sync[F("mode")] = audioSyncEnabled;
sync[F("skip_old_data")] = audioSyncPurge;
sync[F("check_sequence")] = audioSyncSequence;
}
@@ -2878,8 +2898,7 @@ 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();
@@ -2943,6 +2962,7 @@ class AudioReactive : public Usermod {
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
@@ -2950,7 +2970,7 @@ 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
@@ -2960,8 +2980,7 @@ 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,'<button onclick=\"location.href=&quot;https://mm.kno.wled.ge/soundreactive/Sound-Settings&quot;\" type=\"button\">?</button>');"));
@@ -3171,6 +3190,11 @@ class AudioReactive : public Usermod {
#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);"));
@@ -3235,10 +3259,10 @@ 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;
}