From 78e0c3dccafca12b5f737f86fe681e4ac20fdf8b Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Tue, 4 Oct 2022 13:47:07 +0200 Subject: [PATCH 1/6] make MIC_LOGGER work again - EVERY_N_MILLIS somehow does not work. Replaced it, - make sure that WLED_DEBUG does not report "fake FFT times" whrn FFT code was not running --- usermods/audioreactive/audio_reactive.h | 49 ++++++++++++++++--------- 1 file changed, 32 insertions(+), 17 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index cc721cea..6237d681 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -197,6 +197,7 @@ void FFTcode(void * parameter) #if defined(WLED_DEBUG) || defined(SR_DEBUG) uint64_t start = esp_timer_get_time(); + bool haveDoneFFT = false; // indicates if second measurement (FFT time) is valid #endif // get a fresh batch of samples from I2S @@ -204,9 +205,10 @@ void FFTcode(void * parameter) #if defined(WLED_DEBUG) || defined(SR_DEBUG) if (start < esp_timer_get_time()) { // filter out overflows - unsigned long sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding + uint64_t sampleTimeInMillis = (esp_timer_get_time() - start +5ULL) / 10ULL; // "+5" to ensure proper rounding sampleTime = (sampleTimeInMillis*3 + sampleTime*7)/10; // smooth } + start = esp_timer_get_time(); // start measuring FFT time #endif xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay @@ -255,6 +257,10 @@ void FFTcode(void * parameter) #endif FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects +#if defined(WLED_DEBUG) || defined(SR_DEBUG) + haveDoneFFT = true; +#endif + } else { // noise gate closed - only clear results as FFT was skipped. MIC samples are still valid when we do this. memset(vReal, 0, sizeof(vReal)); FFT_MajorPeak = 1; @@ -397,8 +403,8 @@ void FFTcode(void * parameter) } #if defined(WLED_DEBUG) || defined(SR_DEBUG) - if (start < esp_timer_get_time()) { // filter out overflows - unsigned long fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding + if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows + uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth } #endif @@ -563,21 +569,21 @@ class AudioReactive : public Usermod { //////////////////// void logAudio() { + if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & 0x02) == 0))) return; // no audio availeable #ifdef MIC_LOGGER // Debugging functions for audio input and sound processing. Comment out the values you want to see Serial.print("micReal:"); Serial.print(micDataReal); Serial.print("\t"); - //Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t"); - //Serial.print("micLev:"); Serial.print(micLev); Serial.print("\t"); - //Serial.print("sampleReal:"); Serial.print(sampleReal); Serial.print("\t"); - //Serial.print("sample:"); Serial.print(sample); Serial.print("\t"); + Serial.print("volumeSmth:"); Serial.print(volumeSmth); Serial.print("\t"); + //Serial.print("volumeRaw:"); Serial.print(volumeRaw); Serial.print("\t"); + //Serial.print("DC_Level:"); Serial.print(micLev); Serial.print("\t"); + //Serial.print("sampleAgc:"); Serial.print(sampleAgc); Serial.print("\t"); //Serial.print("sampleAvg:"); Serial.print(sampleAvg); Serial.print("\t"); + //Serial.print("sampleReal:"); Serial.print(sampleReal); Serial.print("\t"); + //Serial.print("micIn:"); Serial.print(micIn); Serial.print("\t"); + //Serial.print("sample:"); Serial.print(sample); Serial.print("\t"); //Serial.print("sampleMax:"); Serial.print(sampleMax); Serial.print("\t"); //Serial.print("samplePeak:"); Serial.print((samplePeak!=0) ? 128:0); Serial.print("\t"); //Serial.print("multAgc:"); Serial.print(multAgc, 4); Serial.print("\t"); - Serial.print("sampleAgc:"); Serial.print(sampleAgc); Serial.print("\t"); - //Serial.print("volumeRaw:"); Serial.print(volumeRaw); Serial.print("\t"); - //Serial.print("volumeSmth:"); Serial.print(volumeSmth); Serial.print("\t"); - Serial.println(); #endif @@ -1170,9 +1176,11 @@ class AudioReactive : public Usermod { } #if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG) - EVERY_N_MILLIS(20) { - logAudio(); - } + static unsigned long lastMicLoggerTime = 0; + if (millis()-lastMicLoggerTime > 20) { + lastMicLoggerTime = millis(); + logAudio(); + } #endif // Info Page: keep max sample from last 5 seconds @@ -1392,11 +1400,18 @@ class AudioReactive : public Usermod { infoArr = user.createNestedArray(F("Sampling time")); infoArr.add(float(sampleTime)/100.0f); infoArr.add(" ms"); + infoArr = user.createNestedArray(F("FFT time")); - infoArr.add(float(fftTime-sampleTime)/100.0f); - infoArr.add(" ms"); + infoArr.add(float(fftTime)/100.0f); + if ((fftTime/100) >= FFT_MIN_CYCLE) // FFT time over budget -> I2S buffer will overflow + infoArr.add("! ms"); + else if ((fftTime/80 + sampleTime/80) >= FFT_MIN_CYCLE) // FFT time >75% of budget -> risk of instability + infoArr.add(" ms!"); + else + infoArr.add(" ms"); + DEBUGSR_PRINTF("AR Sampling time: %5.2f ms\n", float(sampleTime)/100.0f); - DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", float(fftTime-sampleTime)/100.0f); + DEBUGSR_PRINTF("AR FFT time : %5.2f ms\n", float(fftTime)/100.0f); #endif } } From 7d5ce994ab979cd96a21d26c6a71b16d4469144b Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Tue, 4 Oct 2022 13:50:01 +0200 Subject: [PATCH 2/6] WLED_DEBUG: fix for crash after LittleFS formating This fixes a "division by zero" in WLED_DEBUG code. LittleFS init seems to take some time, so we can arrive at "Loops/sec" with 0 loops executed --> crash. --- wled00/wled.cpp | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/wled00/wled.cpp b/wled00/wled.cpp index 47c0ed02..afe2095a 100644 --- a/wled00/wled.cpp +++ b/wled00/wled.cpp @@ -204,9 +204,11 @@ void WLED::loop() DEBUG_PRINT(F("State time: ")); DEBUG_PRINTLN(wifiStateChangedTime); DEBUG_PRINT(F("NTP last sync: ")); DEBUG_PRINTLN(ntpLastSyncTime); DEBUG_PRINT(F("Client IP: ")); DEBUG_PRINTLN(Network.localIP()); - DEBUG_PRINT(F("Loops/sec: ")); DEBUG_PRINTLN(loops / 30); - DEBUG_PRINT(F("UM time[ms]: ")); DEBUG_PRINT(avgUsermodMillis/loops); DEBUG_PRINT("/");DEBUG_PRINTLN(maxUsermodMillis); - DEBUG_PRINT(F("Strip time[ms]: ")); DEBUG_PRINT(avgStripMillis/loops); DEBUG_PRINT("/"); DEBUG_PRINTLN(maxStripMillis); + if (loops > 0) { // avoid division by zero + DEBUG_PRINT(F("Loops/sec: ")); DEBUG_PRINTLN(loops / 30); + DEBUG_PRINT(F("UM time[ms]: ")); DEBUG_PRINT(avgUsermodMillis/loops); DEBUG_PRINT("/");DEBUG_PRINTLN(maxUsermodMillis); + DEBUG_PRINT(F("Strip time[ms]: ")); DEBUG_PRINT(avgStripMillis/loops); DEBUG_PRINT("/"); DEBUG_PRINTLN(maxStripMillis); + } strip.printSize(); loops = 0; maxUsermodMillis = 0; From 1ae0dd574df3cf77c5a82e54995f0b473f534413 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Tue, 4 Oct 2022 16:00:36 +0200 Subject: [PATCH 3/6] fix for ADC analog the "wait until I2S buffer fills" trick does not work for ADC sources, as the I2S sampling does not run in background for ADC. --- usermods/audioreactive/audio_reactive.h | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 6237d681..2a6efbc6 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -411,8 +411,11 @@ void FFTcode(void * parameter) // run peak detection autoResetPeak(); detectSamplePeak(); - - vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers + + #if !defined(I2S_GRAB_ADC1_COMPLETELY) + if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC + #endif + vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers } // for(;;)ever } // FFTcode() task end From 64441b39ac4f7ce2f4db812e8c9475da9a76daec Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Thu, 6 Oct 2022 15:39:28 +0200 Subject: [PATCH 4/6] critical bugfix it's possible that volume samples become negative. In this scenario, our simple noise gate does stupid things, and it looks like "effects temporarily lost the sound". This fix improves the situation, and makes sure that volume samples are always >= 0. --- usermods/audioreactive/audio_reactive.h | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index 2a6efbc6..d7fbee7c 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -229,7 +229,7 @@ void FFTcode(void * parameter) #ifdef SR_DEBUG if (true) { // this allows measure FFT runtimes, as it disables the "only when needed" optimization #else - if (sampleAvg > 1) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed. + if (sampleAvg > 0.5f) { // noise gate open means that FFT results will be used. Don't run FFT if results are not needed. #endif // run FFT (takes 3-5ms on ESP32, ~12ms on ESP32-S2) @@ -273,7 +273,7 @@ void FFTcode(void * parameter) } // for() // mapping of FFT result bins to frequency channels - if (sampleAvg > 1) { // noise gate open + if (sampleAvg > 0.5f) { // noise gate open #if 0 /* This FFT post processing is a DIY endeavour. What we really need is someone with sound engineering expertise to do a great job here AND most importantly, that the animations look GREAT as a result. * @@ -331,7 +331,7 @@ void FFTcode(void * parameter) // post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling) for (int i=0; i < NUM_GEQ_CHANNELS; i++) { - if (sampleAvg > 1) { // noise gate open + if (sampleAvg > 0.5f) { // noise gate open // Adjustment for frequency curves. fftCalc[i] *= fftResultPink[i]; if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function @@ -678,7 +678,7 @@ class AudioReactive : public Usermod { if (time_now - last_time > 2) { last_time = time_now; - if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) { + if((fabsf(sampleReal) < 2.0f) || (sampleMax < 1.0f)) { // MIC signal is "squelched" - deliver silence tmpAgc = 0; // we need to "spin down" the intgrated error buffer @@ -736,6 +736,7 @@ class AudioReactive : public Usermod { else sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path + sampleAgc = fabsf(sampleAgc); // // make sure we have a positive value last_soundAgc = soundAgc; } // agcAvg() @@ -773,12 +774,13 @@ class AudioReactive : public Usermod { micIn -= micLev; // Let's center it to 0 now // Using an exponential filter to smooth out the signal. We'll add controls for this in a future release. - float micInNoDC = fabs(micDataReal - micLev); + float micInNoDC = fabsf(micDataReal - micLev); expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF); + expAdjF = fabsf(expAdjF); // Now (!) take the absolute value + expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate if ((soundSquelch == 0) && (expAdjF < 0.25f)) expAdjF = 0; // do something meaningfull when "squelch = 0" - expAdjF = fabsf(expAdjF); // Now (!) take the absolute value tmpSample = expAdjF; micIn = abs(micIn); // And get the absolute value of each sample @@ -800,6 +802,7 @@ class AudioReactive : public Usermod { if (sampleMax < 0.5f) sampleMax = 0.0f; sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. + sampleAvg = fabsf(sampleAvg); // make sure we have a positive value } // getSample() From 841758978966f3adc1b007bd802d10b64d67b82a Mon Sep 17 00:00:00 2001 From: Blaz Kristan Date: Tue, 18 Oct 2022 20:54:23 +0200 Subject: [PATCH 5/6] Fix fork version. --- package-lock.json | 2 +- package.json | 2 +- wled00/data/index.css | 4 ++-- wled00/wled.h | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/package-lock.json b/package-lock.json index ce4caa8e..0e231caa 100644 --- a/package-lock.json +++ b/package-lock.json @@ -1,6 +1,6 @@ { "name": "wled", - "version": "0.14.0-b0", + "version": "0.14.0-bl0", "lockfileVersion": 1, "requires": true, "dependencies": { diff --git a/package.json b/package.json index 1e31f689..139ef1a1 100644 --- a/package.json +++ b/package.json @@ -1,6 +1,6 @@ { "name": "wled", - "version": "0.14.0-b0", + "version": "0.14.0-bl0", "description": "Tools for WLED project", "main": "tools/cdata.js", "directories": { diff --git a/wled00/data/index.css b/wled00/data/index.css index 79a15993..a591b3da 100644 --- a/wled00/data/index.css +++ b/wled00/data/index.css @@ -101,7 +101,7 @@ button { position: fixed; bottom: calc(var(--bh) + 6px); right: 6px; - color: var(--c-8); /* should remain bright (--c-d) with dark shadow (see below) to be legible on gray background */ + color: var(--c-d); /* should remain bright (--c-d) with dark shadow (see below) to be legible on gray background */ cursor: pointer; writing-mode: vertical-rl; /* transform: rotate(180deg); */ @@ -1045,7 +1045,7 @@ textarea { /*box-shadow: 0 0 0 5px var(--c-d);*/ } -.qcs/*, #namelabel*/ { /* text shadow for name to be legible on grey backround */ +.qcs, #namelabel { /* text shadow for name to be legible on grey backround */ text-shadow: -1px -1px 0 var(--c-4), 1px -1px 0 var(--c-4), -1px 1px 0 var(--c-4), 1px 1px 0 var(--c-4); } diff --git a/wled00/wled.h b/wled00/wled.h index 8b62413a..e7cbca8f 100644 --- a/wled00/wled.h +++ b/wled00/wled.h @@ -8,7 +8,7 @@ */ // version code in format yymmddb (b = daily build) -#define VERSION 2210101 +#define VERSION 2210181 //uncomment this if you have a "my_config.h" file you'd like to use //#define WLED_USE_MY_CONFIG From ba0bc31525a7f04d39d9fcc7b157518db77a4b85 Mon Sep 17 00:00:00 2001 From: Frank <91616163+softhack007@users.noreply.github.com> Date: Fri, 21 Oct 2022 12:12:02 +0200 Subject: [PATCH 6/6] UDP sound sync: added decoder for legacy packets support decoding of sound sync data from SR version > 0.13.0 --- usermods/audioreactive/audio_reactive.h | 104 +++++++++++++++++------- 1 file changed, 73 insertions(+), 31 deletions(-) diff --git a/usermods/audioreactive/audio_reactive.h b/usermods/audioreactive/audio_reactive.h index d7fbee7c..7f09fc78 100644 --- a/usermods/audioreactive/audio_reactive.h +++ b/usermods/audioreactive/audio_reactive.h @@ -223,7 +223,7 @@ void FFTcode(void * parameter) if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]); } // release highest sample to volume reactive effects early - not strictly necessary here - could also be done at the end of the function - // early release allows the filters (getSample() and agcAvg()) to work with fresh values - we will have matching gain and noise gate values when we want to process the FFT results. micDataReal = maxSample; + // early release allows the filters (getSample() and agcAvg()) to work with fresh values - we will have matching gain and noise gate values when we want to process the FFT results. micDataReal = maxSample; #ifdef SR_DEBUG @@ -552,6 +552,7 @@ class AudioReactive : public Usermod { // used to feed "Info" Page unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket + 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 @@ -887,11 +888,67 @@ class AudioReactive : public Usermod { static bool isValidUdpSyncVersion(const char *header) { return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0; } + static bool isValidUdpSyncVersion_v1(const char *header) { + return strncmp_P(header, PSTR(UDP_SYNC_HEADER_v1), 6) == 0; + } + + void decodeAudioData(int packetSize, uint8_t *fftBuff) { + audioSyncPacket *receivedPacket = reinterpret_cast(fftBuff); + // update samples for effects + volumeSmth = fmaxf(receivedPacket->sampleSmth, 0.0f); + volumeRaw = fmaxf(receivedPacket->sampleRaw, 0.0f); + // update internal samples + sampleRaw = volumeRaw; + sampleAvg = volumeSmth; + rawSampleAgc = volumeRaw; + sampleAgc = volumeSmth; + multAgc = 1.0f; + // Only change samplePeak IF it's currently false. + // If it's true already, then the animation still needs to respond. + autoResetPeak(); + if (!samplePeak) { + samplePeak = receivedPacket->samplePeak >0 ? true:false; + if (samplePeak) timeOfPeak = millis(); + //userVar1 = samplePeak; + } + //These values are only available on the ESP32 + for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i]; + my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f); + FFT_Magnitude = my_magnitude; + 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) { + audioSyncPacket_v1 *receivedPacket = reinterpret_cast(fftBuff); + // update samples for effects + volumeSmth = fmaxf(receivedPacket->sampleAgc, 0.0f); + volumeRaw = volumeSmth; // V1 format does not have "raw" AGC sample + // update internal samples + sampleRaw = fmaxf(receivedPacket->sampleRaw, 0.0f); + sampleAvg = fmaxf(receivedPacket->sampleAvg, 0.0f);; + sampleAgc = volumeSmth; + rawSampleAgc = volumeRaw; + multAgc = 1.0f; + // Only change samplePeak IF it's currently false. + // If it's true already, then the animation still needs to respond. + autoResetPeak(); + if (!samplePeak) { + samplePeak = receivedPacket->samplePeak >0 ? true:false; + if (samplePeak) timeOfPeak = millis(); + //userVar1 = samplePeak; + } + //These values are only available on the ESP32 + for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i]; + my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0); + FFT_Magnitude = my_magnitude; + FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0, 11025.0); // restrict value to range expected by effects + } bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received. { if (!udpSyncConnected) return false; bool haveFreshData = false; + size_t packetSize = fftUdp.parsePacket(); if (packetSize > 5) { //DEBUGSR_PRINTLN("Received UDP Sync Packet"); @@ -900,37 +957,17 @@ class AudioReactive : public Usermod { // VERIFY THAT THIS IS A COMPATIBLE PACKET if (packetSize == sizeof(audioSyncPacket) && (isValidUdpSyncVersion((const char *)fftBuff))) { - audioSyncPacket *receivedPacket = reinterpret_cast(fftBuff); - - // update samples for effects - volumeSmth = fmaxf(receivedPacket->sampleSmth, 0.0f); - volumeRaw = fmaxf(receivedPacket->sampleRaw, 0.0f); - - // update internal samples - sampleRaw = volumeRaw; - sampleAvg = volumeSmth; - rawSampleAgc = volumeRaw; - sampleAgc = volumeSmth; - multAgc = 1.0f; - - autoResetPeak(); - // Only change samplePeak IF it's currently false. - // If it's true already, then the animation still needs to respond. - if (!samplePeak) { - samplePeak = receivedPacket->samplePeak >0 ? true:false; - if (samplePeak) timeOfPeak = millis(); - //userVar1 = samplePeak; - } - - //These values are only available on the ESP32 - for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i]; - - my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f); - FFT_Magnitude = my_magnitude; - FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects - - //DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet"); + decodeAudioData(packetSize, fftBuff); + //DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet v2"); haveFreshData = true; + receivedFormat = 2; + } else { + if (packetSize == sizeof(audioSyncPacket_v1) && (isValidUdpSyncVersion_v1((const char *)fftBuff))) { + decodeAudioData_v1(packetSize, fftBuff); + //DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet v1"); + haveFreshData = true; + receivedFormat = 1; + } else receivedFormat = 0; // unknown format } } return haveFreshData; @@ -1395,12 +1432,17 @@ class AudioReactive : public Usermod { if (audioSyncEnabled) { if (audioSyncEnabled & 0x01) { infoArr.add(F("send mode")); + if ((udpSyncConnected) && (millis() - lastTime < 2500)) infoArr.add(F(" v2")); } else if (audioSyncEnabled & 0x02) { infoArr.add(F("receive mode")); } } else infoArr.add("off"); if (audioSyncEnabled && !udpSyncConnected) infoArr.add(" (unconnected)"); + if (audioSyncEnabled && udpSyncConnected && (millis() - last_UDPTime < 2500)) { + if (receivedFormat == 1) infoArr.add(F(" v1")); + if (receivedFormat == 2) infoArr.add(F(" v2")); + } #if defined(WLED_DEBUG) || defined(SR_DEBUG) infoArr = user.createNestedArray(F("Sampling time"));