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Merge remote-tracking branch 'origin/audio-reactive' into mdev

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Frank
2022-10-05 17:58:48 +02:00
parent b404ae1fe4 5ed0bb8c2b
commit c497c6345d
4 changed files with 136 additions and 40 deletions
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154
usermods/audioreactive/audio_reactive.h
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@@ -20,6 +20,8 @@
* ....
*/
#define FFT_PREFER_EXACT_PEAKS // use different FFT wndowing -> results in "sharper" peaks and less "leaking" into other frequencies
// Comment/Uncomment to toggle usb serial debugging
// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
// #define FFT_SAMPLING_LOG // FFT result debugging
@@ -60,8 +62,8 @@ static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1
// user settable parameters for limitSoundDynamics()
static bool limiterOn = true; // bool: enable / disable dynamics limiter
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
static uint16_t attackTime = 50; // int: attack time in milliseconds. Default 0.08sec
static uint16_t decayTime = 300; // int: decay time in milliseconds. New default 300ms. Old default was 1.40sec
// user settable options for FFTResult scaling
static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized sqare root
static uint8_t pinkIndex = 0; // 0: default; 1: line-in; 2: IMNP441
@@ -153,13 +155,54 @@ static uint64_t sampleTime = 0;
#endif
// Table of multiplication factors so that we can even out the frequency response.
#define MAX_PINK 2 // 0 = standard, 1= line-in (pink moise only), 2 = IMNP441, ...
static float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = {
#define MAX_PINK 9 // 0 = standard, 1= line-in (pink moise only), 2..4 = IMNP441, 5..6 = ICS-43434, 6..7 = userdef, 9= flat (no pink noise adjustment)
static const float fftResultPink[MAX_PINK+1][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 }, // default from SR WLED
{ 1.30f, 1.32f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.39f, 3.09f, 4.34f }, // pink noise adjustment only. Good for line-in when there is nomicrophone distortion
{ 2.60f, 2.20f, 1.30f, 1.15f, 1.35f, 2.05f, 2.90f, 2.24f, 2.00f, 2.00f, 2.55f, 2.90f, 2.70f, 2.05f, 4.50f, 8.85f } // optimized for IMNP441
{ 1.30f, 1.32f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.39f, 3.09f, 4.34f }, // Line-In - pink noise adjustment only, without microphone distortion
{ 1.82f, 1.72f, 1.70f, 1.50f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.90f, 3.86f, 6.29f}, // IMNP441 datasheet response profile * pink noise
{ 2.80f, 2.20f, 1.30f, 1.15f, 1.55f, 2.45f, 4.20f, 2.80f, 3.20f, 3.60f, 4.20f, 4.90f, 5.70f, 6.05f,10.50f,14.85f}, // IMNP441 - big speaker, strong bass
// next one has not much visual differece compared to default IMNP441 profile
{ 12.0f, 6.60f, 2.60f, 1.15f, 1.35f, 2.05f, 2.85f, 2.50f, 2.85f, 3.30f, 2.25f, 4.35f, 3.80f, 3.75f, 6.50f, 9.00f}, // IMNP441 - voice, or small speaker
{ 2.75f, 1.60f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 1.75f, 2.55f, 3.60f }, // ICS-43434 datasheet response * pink noise
{ 2.25f, 1.20f, 1.00f, 1.20f, 1.80f, 3.20f, 3.06f, 2.50f, 2.85f, 2.80f, 3.10f, 3.25f, 3.15f, 2.40f, 2.80f, 3.20f }, // ICS-43434 - big speaker, strong bass
{ 2.25f, 1.60f, 1.30f, 1.60f, 2.20f, 3.20f, 3.06f, 2.60f, 2.85f, 3.50f, 3.90f, 4.50f, 3.35f, 3.20f, 3.60f, 4.20f }, // ICS-43434 - userdef #1 for ewowi (enhance median freqs)
{ 4.75f, 3.60f, 2.40f, 2.46f, 3.52f, 1.60f, 1.68f, 3.20f, 2.20f, 2.00f, 2.30f, 2.41f, 2.30f, 1.25f, 4.55f, 6.50f }, // ICS-43434 - userdef #2 for softhack (mic hidden inside mini-shield)
{ 2.38f, 2.18f, 2.07f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.70f, 1.95f, 1.70f, 2.13f, 2.47f } // almost FLAT (IMNP441 but no PINK noise adjustments)
};
/* how to make your own profile:
* ===============================
* preparation: make sure your microphone has direct line-of-sigh with the speaker, 1-2meter distance is best
* Prepare your HiFi equipment: disable all "Sound enhancements" - like Loudness, Equalizer, Bass Boost. Bass/Trebble controls set to middle.
* 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: 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)
* Measure: play Pink Noise for 2-3 minutes - for examples from youtube https://www.youtube.com/watch?v=ZXtimhT-ff4
* Measure: Take a Photo. Make sure that LEDs for each "bar" are well visible (ou need to count them later)
* 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 colums (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.
* * in case of square root scale: multiplier = (target * target) / (barheight * barheight)
* * in case of linear scale: multiplier = target / barheight
*
* - start with a copy of the parameter line "Line-In"
* - go through your new parameter line, multiply each entry with the mutliplier you found for that column.
* Compile + upload
* Test your new profile (same procedure as above). Iterate the process to improve results.
*/
// Create FFT object
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors);
@@ -203,6 +246,7 @@ 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
@@ -210,9 +254,10 @@ 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
@@ -239,8 +284,11 @@ void FFTcode(void * parameter)
// run FFT (takes 3-5ms on ESP32, ~12ms on ESP32-S2)
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
FFT.dcRemoval(); // remove DC offset
FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy
//FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection
#if !defined(FFT_PREFER_EXACT_PEAKS)
FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data using "Flat Top" function - better amplitude accuracy
#else
FFT.windowing(FFTWindow::Blackman_Harris, FFTDirection::Forward); // Weigh data using "Blackman- Harris" window - sharp peaks due to excellent sideband rejection
#endif
FFT.compute( FFTDirection::Forward ); // Compute FFT
FFT.complexToMagnitude(); // Compute magnitudes
#else
@@ -248,8 +296,11 @@ void FFTcode(void * parameter)
//FFT.Windowing( FFT_WIN_TYP_HAMMING, FFT_FORWARD ); // Weigh data - standard Hamming window
//FFT.Windowing( FFT_WIN_TYP_BLACKMAN, FFT_FORWARD ); // Blackman window - better side freq rejection
//FFT.Windowing( FFT_WIN_TYP_BLACKMAN_HARRIS, FFT_FORWARD );// Blackman-Harris - excellent sideband rejection
FFT.Windowing( FFT_WIN_TYP_FLT_TOP, FFT_FORWARD ); // Flat Top Window - better amplitude accuracy
#if !defined(FFT_PREFER_EXACT_PEAKS)
FFT.Windowing( FFT_WIN_TYP_FLT_TOP, FFT_FORWARD ); // Flat Top Window - better amplitude accuracy
#else
FFT.Windowing( FFT_WIN_TYP_BLACKMAN_HARRIS, FFT_FORWARD );// Blackman-Harris - excellent sideband rejection
#endif
FFT.Compute( FFT_FORWARD ); // Compute FFT
FFT.ComplexToMagnitude(); // Compute magnitudes
#endif
@@ -261,6 +312,10 @@ 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;
@@ -345,10 +400,13 @@ void FFTcode(void * parameter)
if(fftCalc[i] > fftAvg[i]) // rise fast
fftAvg[i] = fftCalc[i] *0.75f + 0.25f*fftAvg[i]; // will need approx 2 cycles (50ms) for converging against fftCalc[i]
else { // fall slow
if (decayTime < 1000) fftAvg[i] = fftCalc[i]*0.22f + 0.78f*fftAvg[i]; // approx 5 cycles (225ms) for falling to zero
if (decayTime < 250) fftAvg[i] = fftCalc[i]*0.4f + 0.6f*fftAvg[i];
else if (decayTime < 500) fftAvg[i] = fftCalc[i]*0.33f + 0.67f*fftAvg[i];
else if (decayTime < 1000) fftAvg[i] = fftCalc[i]*0.22f + 0.78f*fftAvg[i]; // approx 5 cycles (225ms) for falling to zero
else if (decayTime < 2000) fftAvg[i] = fftCalc[i]*0.17f + 0.83f*fftAvg[i]; // default - approx 9 cycles (225ms) for falling to zero
else if (decayTime < 3000) fftAvg[i] = fftCalc[i]*0.14f + 0.86f*fftAvg[i]; // approx 14 cycles (350ms) for falling to zero
else fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i]; // approx 20 cycles (500ms) for falling to zero
else if (decayTime < 4000) fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i];
else fftAvg[i] = fftCalc[i]*0.05f + 0.95f*fftAvg[i];
}
// constrain internal vars - just to be sure
fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f);
@@ -373,24 +431,26 @@ void FFTcode(void * parameter)
case 2:
// Linear scaling
currentResult *= 0.30f; // needs a bit more damping, get stay below 255
currentResult -= 4.0; // giving a bit more room for peaks
currentResult -= 2.0; // giving a bit more room for peaks
if (currentResult < 1.0f) currentResult = 0.0f;
currentResult *= 0.85f + (float(i)/1.8f); // extra up-scaling for high frequencies
break;
case 3:
// square root scaling
currentResult *= 0.38f;
//currentResult *= 0.34f; //experiment
currentResult -= 6.0f;
if (currentResult > 1.0) currentResult = sqrtf(currentResult);
else currentResult = 0.0; // special handling, because sqrt(0) = undefined
currentResult *= 0.85f + (float(i)/4.5f); // extra up-scaling for high frequencies
//currentResult *= 0.80f + (float(i)/5.6f); //experiment
currentResult = mapf(currentResult, 0.0, 16.0, 0.0, 255.0); // map [sqrt(1) ... sqrt(256)] to [0 ... 255]
break;
case 0:
default:
// no scaling - leave freq bins as-is
currentResult -= 4; // just a bit more room for peaks
currentResult -= 2; // just a bit more room for peaks
break;
}
@@ -404,16 +464,19 @@ 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
// 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
@@ -536,7 +599,7 @@ class AudioReactive : public Usermod {
// 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 controler.
float micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller
double micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller
float expAdjF = 0.0f; // Used for exponential filter.
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)
@@ -570,21 +633,21 @@ 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
@@ -766,7 +829,8 @@ class AudioReactive : public Usermod {
#endif
#endif
micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
//micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
micLev += (micDataReal-micLev) / 12288.0f;
if(micIn < micLev) micLev = ((micLev * 31.0f) + micDataReal) / 32.0f; // align MicLev to lowest input signal
micIn -= micLev; // Let's center it to 0 now
@@ -1177,9 +1241,11 @@ 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
@@ -1406,11 +1472,18 @@ 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("<b style=\"color:red;\">! ms</b>");
else if ((fftTime/80 + sampleTime/80) >= FFT_MIN_CYCLE) // FFT time >75% of budget -> risk of instability
infoArr.add("<b style=\"color:orange;\"> ms!</b>");
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
}
}
@@ -1608,9 +1681,16 @@ class AudioReactive : public Usermod {
oappend(SET_F("addOption(dd,'Logarithmic (Loudness)',1);"));
oappend(SET_F("dd=addDropdown('AudioReactive','Frequency:Profile');"));
oappend(SET_F("addOption(dd,'standard',0);"));
oappend(SET_F("addOption(dd,'Line-In',1);"));
oappend(SET_F("addOption(dd,'Generic Microphone',0);"));
oappend(SET_F("addOption(dd,'Generic Line-In',1);"));
oappend(SET_F("addOption(dd,'IMNP441',2);"));
oappend(SET_F("addOption(dd,'IMNP441 - big speakers',3);"));
oappend(SET_F("addOption(dd,'IMNP441 - small speakers',4);"));
oappend(SET_F("addOption(dd,'ICS-43434',5);"));
oappend(SET_F("addOption(dd,'ICS-43434 - big speakers',6);"));
oappend(SET_F("addOption(dd,'ICS-43434 - userdef #1',7);"));
oappend(SET_F("addOption(dd,'ICS-43434 - userdef #2',8);"));
oappend(SET_F("addOption(dd,'flat - no adjustments',9);"));
oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');"));
oappend(SET_F("addOption(dd,'Off',0);"));

2
usermods/audioreactive/audio_source.h
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@@ -288,6 +288,8 @@ class I2SSource : public AudioSource {
currSample = (float) newSamples[i]; // 16bit input -> use as-is
#endif
buffer[i] = currSample;
//buffer[i] *= 0.6f; // (ICS-43434): compensate for higher sensitivity (reduce by 2db)
}
}
}

12
wled00/presets.cpp
View File

@@ -47,6 +47,18 @@ void handlePresets()
#if !defined(CONFIG_IDF_TARGET_ESP32C3) && !defined(CONFIG_IDF_TARGET_ESP32S2)
// this does not make sense on single core
core = xPortGetCoreID();
// begin WLEDSR specific
// loopTask (arduino main loop) sometimes runs on core #1
if ((core == 1) && (strncmp(pcTaskGetTaskName(NULL), "loopTask", 8) == 0)) {
DEBUG_PRINTF("[applyPreset] called from loopTask on core %d; forcing core = 0\n", (int)core);
core = 0;
}
// async_tcp (network requests) sometimes runs on core #0
if ((core == 0) && (strncmp(pcTaskGetTaskName(NULL), "async_tcp", 9) == 0)) {
DEBUG_PRINTF("[applyPreset] called from async_tcp on core %d; forcing core = 1\n", (int)core);
core = 1;
}
// end WLEDSR specific
#endif
#endif
//only allow use of fileDoc from the core responsible for network requests (AKA HTTP JSON API)

8
wled00/wled.cpp
View File

@@ -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;