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Merge branch 'audioreactive-prototype' of https://github.com/blazoncek/WLED into merge-audio

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Blaž Kristan
2022-08-22 10:34:10 +02:00
parent 844bef9fda be7e7ac274
commit cf0f0d77be
7 changed files with 735 additions and 670 deletions
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6
usermods/PWM_fan/usermod_PWM_fan.h
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@@ -275,12 +275,12 @@ class PWMFanUsermod : public Usermod {
enabled = usermod[FPSTR(_enabled)].as<bool>();
if (!enabled) updateFanSpeed(0);
}
if (!usermod[FPSTR(_speed)].isNull() && usermod[FPSTR(_speed)].is<int>()) {
if (enabled && !usermod[FPSTR(_speed)].isNull() && usermod[FPSTR(_speed)].is<int>()) {
pwmValuePct = usermod[FPSTR(_speed)].as<int>();
updateFanSpeed((MAX(0,MIN(100,pwmValuePct)) * 255) / 100);
updateFanSpeed((constrain(pwmValuePct,0,100) * 255) / 100);
if (pwmValuePct) lockFan = true;
}
if (!usermod[FPSTR(_lock)].isNull() && usermod[FPSTR(_lock)].is<bool>()) {
if (enabled && !usermod[FPSTR(_lock)].isNull() && usermod[FPSTR(_lock)].is<bool>()) {
lockFan = usermod[FPSTR(_lock)].as<bool>();
}
}

502
usermods/audioreactive/audio_reactive.h
View File

@@ -23,11 +23,9 @@
// Comment/Uncomment to toggle usb serial debugging
// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
// #define FFT_SAMPLING_LOG // FFT result debugging
// #define SR_DEBUG // generic SR DEBUG messages (including MIC_LOGGER)
// #define SR_DEBUG // generic SR DEBUG messages
// #define NO_MIC_LOGGER // exclude MIC_LOGGER from SR_DEBUG
// hackers corner
#ifdef SR_DEBUG
#define DEBUGSR_PRINT(x) Serial.print(x)
#define DEBUGSR_PRINTLN(x) Serial.println(x)
@@ -37,21 +35,19 @@
#define DEBUGSR_PRINTLN(x)
#define DEBUGSR_PRINTF(x...)
#endif
// legacy support
// #if defined(SR_DEBUG) && !defined(MIC_LOGGER) && !defined(NO_MIC_LOGGER)
// #define MIC_LOGGER
// #endif
#include "audio_source.h"
constexpr i2s_port_t I2S_PORT = I2S_NUM_0;
constexpr int BLOCK_SIZE = 128;
//constexpr int SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz is a standard rate. Physical sample time -> 23ms
constexpr int SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental. Physical sample time -> 25ms
//constexpr int SAMPLE_RATE = 10240; // Base sample rate in Hz - standard. Physical sample time -> 50ms
constexpr int SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz is a standard rate. Physical sample time -> 23ms
//constexpr int SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental. Physical sample time -> 25ms
//constexpr int SAMPLE_RATE = 10240; // Base sample rate in Hz - previous default. Physical sample time -> 50ms
#define FFT_MIN_CYCLE 22 // minimum time before FFT task is repeated. Must be less than time needed to read 512 samples at SAMPLE_RATE -> not the same as I2S time!!
#define FFT_MIN_CYCLE 18 // minimum time before FFT task is repeated. Use with 22Khz sampling
//#define FFT_MIN_CYCLE 22 // minimum time before FFT task is repeated. Use with 20Khz sampling
//#define FFT_MIN_CYCLE 44 // minimum time before FFT task is repeated. Use with 10Khz sampling
// globals
static uint8_t inputLevel = 128; // UI slider value
@@ -64,6 +60,8 @@ static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1
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
// user settable options for FFTResult scaling
static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized sqare root
//
// AGC presets
@@ -88,8 +86,14 @@ static AudioSource *audioSource = nullptr;
static volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks.
static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point
static float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC.
static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
static int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel)
static int16_t rawSampleAgc = 0; // not smoothed AGC sample
static float sampleAvg = 0.0f; // Smoothed Average sampleRaw
static float sampleAgc = 0.0f; // Smoothed AGC sample
////////////////////
// Begin FFT Code //
////////////////////
@@ -105,7 +109,7 @@ static float multAgc = 1.0f; // sample * multAgc = sampleAgc.
constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2
constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
static float FFT_MajorPeak = 0.0f;
static float FFT_MajorPeak = 1.0f;
static float FFT_Magnitude = 0.0f;
// These are the input and output vectors. Input vectors receive computed results from FFT.
@@ -113,6 +117,12 @@ static float vReal[samplesFFT] = {0.0f};
static float vImag[samplesFFT] = {0.0f};
static float fftBin[samplesFFT_2] = {0.0f};
// the following are observed values, supported by a bit of "educated guessing"
//#define FFT_DOWNSCALE 0.65f // 20kHz - downscaling factor for FFT results - "Flat-Top" window @20Khz, old freq channels
#define FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
#define LOG_256 5.54517744
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static float windowWeighingFactors[samplesFFT] = {0.0f};
#endif
@@ -131,9 +141,6 @@ static unsigned long fftTime = 0;
static unsigned long sampleTime = 0;
#endif
// Table of linearNoise results to be multiplied by soundSquelch in order to reduce squelch across fftResult bins.
static uint8_t linearNoise[16] = { 34, 28, 26, 25, 20, 12, 9, 6, 4, 4, 3, 2, 2, 2, 2, 2 };
// Table of multiplication factors so that we can even out the frequency response.
static float fftResultPink[16] = { 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 };
@@ -146,6 +153,11 @@ static arduinoFFT FFT = arduinoFFT(vReal, vImag, samplesFFT, SAMPLE_RATE);
static TaskHandle_t FFT_Task = nullptr;
// float version of map()
static float mapf(float x, float in_min, float in_max, float out_min, float out_max){
return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
}
static float fftAddAvg(int from, int to) {
float result = 0.0f;
for (int i = from; i <= to; i++) {
@@ -161,27 +173,23 @@ void FFTcode(void * parameter)
// see https://www.freertos.org/vtaskdelayuntil.html
const TickType_t xFrequency = FFT_MIN_CYCLE * portTICK_PERIOD_MS;
//const TickType_t xFrequency_2 = (FFT_MIN_CYCLE * portTICK_PERIOD_MS) / 2;
for(;;) {
TickType_t xLastWakeTime = xTaskGetTickCount();
delay(1); // DO NOT DELETE THIS LINE! It is needed to give the IDLE(0) task enough time and to keep the watchdog happy.
// taskYIELD(), yield(), vTaskDelay() and esp_task_wdt_feed() didn't seem to work.
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
// Only run the FFT computing code if we're not in Receive mode and not in realtime mode
if (disableSoundProcessing || (audioSyncEnabled & 0x02)) {
//delay(7); // release CPU - delay is implemeted using vTaskDelay(). cannot use yield() because we are out of arduino loop context
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, by doing nothing for FFT_MIN_CYCLE millis
continue;
}
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
//vTaskDelayUntil( &xLastWakeTime, xFrequency_2); // release CPU, and let I2S fill its buffers
#ifdef WLED_DEBUG
uint64_t start = esp_timer_get_time();
#endif
// get a fresh batch of samples from I2S
if (audioSource) audioSource->getSamples(vReal, samplesFFT);
#ifdef WLED_DEBUG
@@ -191,29 +199,24 @@ void FFTcode(void * parameter)
}
#endif
const int halfSamplesFFT = samplesFFT / 2; // samplesFFT divided by 2
float maxSample1 = 0.0f; // max sample from first half of FFT batch
float maxSample2 = 0.0f; // max sample from second half of FFT batch
for (int i=0; i < halfSamplesFFT; i++) {
// find highest sample in the batch
float maxSample = 0.0f; // max sample from FFT batch
for (int i=0; i < samplesFFT; i++) {
// set imaginary parts to 0
vImag[i] = 0;
// pick our 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 (fabsf((float)vReal[i]) > maxSample1) maxSample1 = fabsf((float)vReal[i]);
if (fabsf((float)vReal[i]) > maxSample) maxSample = fabsf((float)vReal[i]);
}
for (int i=halfSamplesFFT; i < samplesFFT; i++) {
// set imaginary parts to 0
vImag[i] = 0;
// pick our 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 (fabsf((float)vReal[i]) > maxSample2) maxSample2 = fabsf((float)vReal[i]);
}
// release first sample to volume reactive effects
micDataReal = maxSample1;
// 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;
micDataReal = maxSample;
// run FFT (takes 3-5ms on ESP32)
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
FFT.dcRemoval(); // remove DC offset
FFT.windowing( FFTWindow::Flat_top, FFTDirection::Forward); // Weigh data
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
FFT.compute( FFTDirection::Forward ); // Compute FFT
FFT.complexToMagnitude(); // Compute magnitudes
#else
@@ -226,71 +229,147 @@ void FFTcode(void * parameter)
FFT.Compute( FFT_FORWARD ); // Compute FFT
FFT.ComplexToMagnitude(); // Compute magnitudes
#endif
//
// vReal[3 .. 255] contain useful data, each a 20Hz interval (60Hz - 5120Hz).
// There could be interesting data at bins 0 to 2, but there are too many artifacts.
//
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant
FFT.majorPeak(FFT_MajorPeak, FFT_Magnitude); // let the effects know which freq was most dominant
#else
FFT.MajorPeak(&FFT_MajorPeak, &FFT_Magnitude); // let the effects know which freq was most dominant
FFT.MajorPeak(&FFT_MajorPeak, &FFT_Magnitude); // let the effects know which freq was most dominant
#endif
FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
for (int i = 0; i < samplesFFT_2; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3.
float t = fabs(vReal[i]); // just to be sure - values in fft bins should be positive any way
fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max.
for (int i = 0; i < samplesFFT_2; i++) { // Values for bins 0 and 1 are WAY too large. Might as well start at 3.
float t = fabsf(vReal[i]); // just to be sure - values in fft bins should be positive any way
fftBin[i] = t / 16.0f; // Reduce magnitude. Want end result to be linear and ~4096 max.
} // for()
// mapping of FFT result bins to frequency channels
if (sampleAvg > 1) { // 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.
*
* Andrew's updated mapping of 256 bins down to the 16 result bins with Sample Freq = 10240, samplesFFT = 512 and some overlap.
* Based on testing, the lowest/Start frequency is 60 Hz (with bin 3) and a highest/End frequency of 5120 Hz in bin 255.
* Now, Take the 60Hz and multiply by 1.320367784 to get the next frequency and so on until the end. Then detetermine the bins.
* End frequency = Start frequency * multiplier ^ 16
* Multiplier = (End frequency/ Start frequency) ^ 1/16
* Multiplier = 1.320367784
*/ // Range
fftCalc[ 0] = fftAddAvg(2,4); // 60 - 100
fftCalc[ 1] = fftAddAvg(4,5); // 80 - 120
fftCalc[ 2] = fftAddAvg(5,7); // 100 - 160
fftCalc[ 3] = fftAddAvg(7,9); // 140 - 200
fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
fftCalc[11] = fftAddAvg(64,84); // 1280 - 1700
fftCalc[12] = fftAddAvg(84,111); // 1680 - 2240
fftCalc[13] = fftAddAvg(111,147); // 2220 - 2960
fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900
fftCalc[15] = fftAddAvg(194,250); // 3880 - 5000 // avoid the last 5 bins, which are usually inaccurate
#else
/* new mapping, optimized for 22050 Hz by softhack007 */
// bins frequency range
fftCalc[ 0] = fftAddAvg(1,2); // 1 43 - 86 sub-bass
fftCalc[ 1] = fftAddAvg(2,3); // 1 86 - 129 bass
fftCalc[ 2] = fftAddAvg(3,5); // 2 129 - 216 bass
fftCalc[ 3] = fftAddAvg(5,7); // 2 216 - 301 bass + midrange
fftCalc[ 4] = fftAddAvg(7,10); // 3 301 - 430 midrange
fftCalc[ 5] = fftAddAvg(10,13); // 3 430 - 560 midrange
fftCalc[ 6] = fftAddAvg(13,19); // 5 560 - 818 midrange
fftCalc[ 7] = fftAddAvg(19,26); // 7 818 - 1120 midrange -- 1Khz should always be the center !
fftCalc[ 8] = fftAddAvg(26,33); // 7 1120 - 1421 midrange
fftCalc[ 9] = fftAddAvg(33,44); // 9 1421 - 1895 midrange
fftCalc[10] = fftAddAvg(44,56); // 12 1895 - 2412 midrange + high mid
fftCalc[11] = fftAddAvg(56,70); // 14 2412 - 3015 high mid
fftCalc[12] = fftAddAvg(70,86); // 16 3015 - 3704 high mid
fftCalc[13] = fftAddAvg(86,104); // 18 3704 - 4479 high mid
fftCalc[14] = fftAddAvg(104,165) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
fftCalc[15] = fftAddAvg(165,215) * 0.70f; // 50 7106 - 9259 high -- with some damping
// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
#endif
} else { // noise gate closed - just decay old values
for (int i=0; i < 16; i++) {
fftCalc[i] *= 0.85f; // decay to zero
if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f;
}
}
/* 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.
*
*
* Andrew's updated mapping of 256 bins down to the 16 result bins with Sample Freq = 10240, samplesFFT = 512 and some overlap.
* Based on testing, the lowest/Start frequency is 60 Hz (with bin 3) and a highest/End frequency of 5120 Hz in bin 255.
* Now, Take the 60Hz and multiply by 1.320367784 to get the next frequency and so on until the end. Then detetermine the bins.
* End frequency = Start frequency * multiplier ^ 16
* Multiplier = (End frequency/ Start frequency) ^ 1/16
* Multiplier = 1.320367784
*/
// Range
fftCalc[ 0] = fftAddAvg(3,4); // 60 - 100
fftCalc[ 1] = fftAddAvg(4,5); // 80 - 120
fftCalc[ 2] = fftAddAvg(5,7); // 100 - 160
fftCalc[ 3] = fftAddAvg(7,9); // 140 - 200
fftCalc[ 4] = fftAddAvg(9,12); // 180 - 260
fftCalc[ 5] = fftAddAvg(12,16); // 240 - 340
fftCalc[ 6] = fftAddAvg(16,21); // 320 - 440
fftCalc[ 7] = fftAddAvg(21,29); // 420 - 600
fftCalc[ 8] = fftAddAvg(29,37); // 580 - 760
fftCalc[ 9] = fftAddAvg(37,48); // 740 - 980
fftCalc[10] = fftAddAvg(48,64); // 960 - 1300
fftCalc[11] = fftAddAvg(64,84); // 1280 - 1700
fftCalc[12] = fftAddAvg(84,111); // 1680 - 2240
fftCalc[13] = fftAddAvg(111,147); // 2220 - 2960
fftCalc[14] = fftAddAvg(147,194); // 2940 - 3900
fftCalc[15] = fftAddAvg(194,255); // 3880 - 5120
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
for (int i=0; i < 16; i++) {
// Noise supression of fftCalc bins using soundSquelch adjustment for different input types.
fftCalc[i] = (fftCalc[i] < ((float)soundSquelch * (float)linearNoise[i] / 4.0f)) ? 0 : fftCalc[i];
// Adjustment for frequency curves.
fftCalc[i] *= fftResultPink[i];
// Manual linear adjustment of gain using sampleGain adjustment for different input types.
fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //with inputLevel adjustment
if (sampleAvg > 1) { // noise gate open
// Adjustment for frequency curves.
fftCalc[i] *= fftResultPink[i];
if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
// Manual linear adjustment of gain using sampleGain adjustment for different input types.
fftCalc[i] *= soundAgc ? multAgc : ((float)sampleGain/40.0f * (float)inputLevel/128.0f + 1.0f/16.0f); //apply gain, with inputLevel adjustment
if(fftCalc[i] < 0) fftCalc[i] = 0;
}
// smooth results - rise fast, fall slower
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
fftAvg[i] = fftCalc[i]*0.1f + 0.9f*fftAvg[i]; // will need approx 5 cycles (150ms) for converging against fftCalc[i]
//fftAvg[i] = fftCalc[i]*0.05f + 0.95f*fftAvg[i]; // will need approx 10 cycles (250ms) for converging against fftCalc[i]
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
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
}
// constrain internal vars - just to be sure
fftCalc[i] = constrain(fftCalc[i], 0.0f, 1023.0f);
fftAvg[i] = constrain(fftAvg[i], 0.0f, 1023.0f);
float currentResult;
if(limiterOn == true)
currentResult = fftAvg[i];
else
currentResult = fftCalc[i];
switch (FFTScalingMode) {
case 1:
// Logarithmic scaling
currentResult *= 0.42; // 42 is the answer ;-)
currentResult -= 8.0; // this skips the lowest row, giving some room for peaks
if (currentResult > 1.0) currentResult = logf(currentResult); // log to base "e", which is the fastest log() function
else currentResult = 0.0; // special handling, because log(1) = 0; log(0) = undefined
currentResult *= 0.85f + (float(i)/18.0f); // extra up-scaling for high frequencies
currentResult = mapf(currentResult, 0, LOG_256, 0, 255); // map [log(1) ... log(255)] to [0 ... 255]
break;
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
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 -= 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 = 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
break;
}
// Now, let's dump it all into fftResult. Need to do this, otherwise other routines might grab fftResult values prematurely.
if(limiterOn == true)
fftResult[i] = constrain((int)fftAvg[i], 0, 254);
else
fftResult[i] = constrain((int)fftCalc[i], 0, 254);
if (soundAgc > 0) { // apply extra "GEQ Gain" if set by user
float post_gain = (float)inputLevel/128.0f;
if (post_gain < 1.0f) post_gain = ((post_gain -1.0f) * 0.8f) +1.0f;
currentResult *= post_gain;
}
fftResult[i] = constrain((int)currentResult, 0, 255);
}
#ifdef WLED_DEBUG
@@ -300,11 +379,6 @@ void FFTcode(void * parameter)
}
#endif
//vTaskDelayUntil( &xLastWakeTime, xFrequency_2); // release CPU, by waiting until FFT_MIN_CYCLE is over
// release second sample to volume reactive effects.
// Releasing a second sample now effectively doubles the "sample rate"
micDataReal = maxSample2;
} // for(;;)
} // FFTcode()
@@ -314,7 +388,7 @@ class AudioReactive : public Usermod {
private:
#ifndef AUDIOPIN
int8_t audioPin = 36;
int8_t audioPin = -1;
#else
int8_t audioPin = AUDIOPIN;
#endif
@@ -396,12 +470,7 @@ class AudioReactive : public Usermod {
bool udpSamplePeak = 0; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed
int16_t sampleRaw = 0; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel; smoothed over 16 samples)
double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler.
float sampleReal = 0.0f; // "sampleRaw" as float, to provide bits that are lost otherwise (before amplification by sampleGain or inputLevel). Needed for AGC.
float sampleAvg = 0.0f; // Smoothed Average sampleRaw
float sampleAgc = 0.0f; // Our AGC sample
int16_t rawSampleAgc = 0; // Our AGC sample - raw
uint32_t timeOfPeak = 0;
unsigned long lastTime = 0; // last time of running UDP Microphone Sync
float micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller
@@ -540,7 +609,6 @@ class AudioReactive : public Usermod {
if((fabs(sampleReal) < 2.0f) || (sampleMax < 1.0f)) {
// MIC signal is "squelched" - deliver silence
//multAgcTemp = multAgc; // keep old control value (no change)
tmpAgc = 0;
// we need to "spin down" the intgrated error buffer
if (fabs(control_integrated) < 0.01) control_integrated = 0.0;
@@ -548,27 +616,26 @@ class AudioReactive : public Usermod {
} else {
// compute new setpoint
if (tmpAgc <= agcTarget0Up[AGC_preset])
multAgcTemp = agcTarget0[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = first setpoint
multAgcTemp = agcTarget0[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = first setpoint
else
multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
multAgcTemp = agcTarget1[AGC_preset] / sampleMax; // Make the multiplier so that sampleMax * multiplier = second setpoint
}
// limit amplification
//multAgcTemp = constrain(multAgcTemp, 0.015625f, 32.0f); // 1/64 < multAgcTemp < 32
if (multAgcTemp > 32.0f) multAgcTemp = 32.0f;
if (multAgcTemp < 1.0f/64.0f) multAgcTemp = 1.0f/64.0f;
// compute error terms
control_error = multAgcTemp - lastMultAgc;
if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
control_integrated += control_error * 0.002 * 0.25; // 2ms = intgration time; 0.25 for damping
if (((multAgcTemp > 0.085f) && (multAgcTemp < 6.5f)) //integrator anti-windup by clamping
&& (multAgc*sampleMax < agcZoneStop[AGC_preset])) //integrator ceiling (>140% of max)
control_integrated += control_error * 0.002 * 0.25; // 2ms = intgration time; 0.25 for damping
else
control_integrated *= 0.9; // spin down that beasty integrator
control_integrated *= 0.9; // spin down that beasty integrator
// apply PI Control
tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain
if ((tmpAgc > agcZoneHigh[AGC_preset]) || (tmpAgc < soundSquelch + agcZoneLow[AGC_preset])) { // upper/lower emergy zone
tmpAgc = sampleReal * lastMultAgc; // check "zone" of the signal using previous gain
if ((tmpAgc > agcZoneHigh[AGC_preset]) || (tmpAgc < soundSquelch + agcZoneLow[AGC_preset])) { // upper/lower emergy zone
multAgcTemp = lastMultAgc + agcFollowFast[AGC_preset] * agcControlKp[AGC_preset] * control_error;
multAgcTemp += agcFollowFast[AGC_preset] * agcControlKi[AGC_preset] * control_integrated;
} else { // "normal zone"
@@ -598,9 +665,6 @@ class AudioReactive : public Usermod {
else
sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
//userVar0 = sampleAvg * 4;
//if (userVar0 > 255) userVar0 = 255;
last_soundAgc = soundAgc;
} // agcAvg()
@@ -636,7 +700,7 @@ class AudioReactive : public Usermod {
micLev = ((micLev * 8191.0f) + micDataReal) / 8192.0f; // takes a few seconds to "catch up" with the Mic Input
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
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);
expAdjF = (weighting * micInNoDC + (1.0-weighting) * expAdjF);
@@ -650,12 +714,12 @@ class AudioReactive : public Usermod {
sampleAdj = tmpSample * sampleGain / 40.0f * inputLevel/128.0f + tmpSample / 16.0f; // Adjust the gain. with inputLevel adjustment
sampleReal = tmpSample;
sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
sampleRaw = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
// keep "peak" sample, but decay value if current sample is below peak
if ((sampleMax < sampleReal) && (sampleReal > 0.5f)) {
sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
sampleMax = sampleMax + 0.5f * (sampleReal - sampleMax); // new peak - with some filtering
} else {
if ((multAgc*sampleMax > agcZoneStop[AGC_preset]) && (soundAgc > 0))
sampleMax += 0.5f * (sampleReal - sampleMax); // over AGC Zone - get back quickly
@@ -676,13 +740,12 @@ class AudioReactive : public Usermod {
//if (userVar1 == 0) samplePeak = 0;
// Poor man's beat detection by seeing if sample > Average + some value.
// if (sample > (sampleAvg + maxVol) && millis() > (timeOfPeak + 200)) {
if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) { // This goes through ALL of the 255 bins - but ignores stupid settings
// Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync.
if ((maxVol > 0) && (binNum > 1) && (fftBin[binNum] > maxVol) && (millis() > (timeOfPeak + 100))) {
// This goes through ALL of the 255 bins - but ignores stupid settings
// Then we got a peak, else we don't. The peak has to time out on its own in order to support UDP sound sync.
samplePeak = true;
timeOfPeak = millis();
udpSamplePeak = true;
//userVar1 = samplePeak;
}
} // getSample()
@@ -696,7 +759,7 @@ class AudioReactive : public Usermod {
static unsigned long last_time = 0;
static float last_volumeSmth = 0.0f;
if(limiterOn == false) return;
if (limiterOn == false) return;
long delta_time = millis() - last_time;
delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> sily lil hick-up
@@ -725,9 +788,6 @@ class AudioReactive : public Usermod {
audioSyncPacket transmitData;
strncpy_P(transmitData.header, PSTR(UDP_SYNC_HEADER), 6);
//transmitData.sampleRaw = volumeRaw;
//transmitData.sampleSmth = volumeSmth;
// transmit samples that were not modified by limitSampleDynamics()
transmitData.sampleRaw = (soundAgc) ? rawSampleAgc: sampleRaw;
transmitData.sampleSmth = (soundAgc) ? sampleAgc : sampleAvg;
@@ -757,7 +817,6 @@ class AudioReactive : public Usermod {
bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received.
{
if (!udpSyncConnected) return false;
//DEBUGSR_PRINTLN("Checking for UDP Microphone Packet");
bool haveFreshData = false;
size_t packetSize = fftUdp.parsePacket();
if (packetSize > 5) {
@@ -800,7 +859,8 @@ class AudioReactive : public Usermod {
my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f);
FFT_Magnitude = my_magnitude;
FFT_MajorPeak = fmaxf(receivedPacket->FFT_MajorPeak, 0.0f);
FFT_MajorPeak = constrain(receivedPacket->FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet");
haveFreshData = true;
}
@@ -911,7 +971,7 @@ class AudioReactive : public Usermod {
*/
void connected()
{
if (audioSyncPort > 0 || (audioSyncEnabled & 0x03)) {
if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) {
#ifndef ESP8266
udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort);
#else
@@ -971,7 +1031,7 @@ class AudioReactive : public Usermod {
if (audioSyncEnabled & 0x02) disableSoundProcessing = true; // make sure everything is disabled IF in audio Receive mode
if (audioSyncEnabled & 0x01) disableSoundProcessing = false; // keep running audio IF we're in audio Transmit mode
if(!audioSource->isInitialized()) disableSoundProcessing = true; // no audio source
if (!audioSource->isInitialized()) disableSoundProcessing = true; // no audio source
// Only run the sampling code IF we're not in Receive mode or realtime mode
@@ -998,7 +1058,7 @@ class AudioReactive : public Usermod {
agcAvg(t_now - userloopDelay); // Calculated the PI adjusted value as sampleAvg
userloopDelay -= 2; // advance "simulated time" by 2ms
} while (userloopDelay > 0);
lastUMRun = t_now; // update time keeping
lastUMRun = t_now; // update time keeping
// update samples for effects (raw, smooth)
volumeSmth = (soundAgc) ? sampleAgc : sampleAvg;
@@ -1006,50 +1066,9 @@ class AudioReactive : public Usermod {
// update FFTMagnitude, taking into account AGC amplification
my_magnitude = FFT_Magnitude; // / 16.0f, 8.0f, 4.0f done in effects
if (soundAgc) my_magnitude *= multAgc;
if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
if (volumeSmth < 1 ) my_magnitude = 0.001f; // noise gate closed - mute
limitSampleDynamics(); // optional - makes volumeSmth very smooth and fluent
// update WebServer UI
uint8_t knownMode = strip.getFirstSelectedSeg().mode; // 1st selected segment is more appropriate than main segment
if (lastMode != knownMode) { // only execute if mode changes
char lineBuffer[4];
extractModeName(knownMode, JSON_mode_names, lineBuffer, 3); // use of JSON_mode_names is deprecated, use nullptr
agcEffect = (lineBuffer[1] == 226 && lineBuffer[2] == 153); // && (lineBuffer[3] == 170 || lineBuffer[3] == 171 ) encoding of ♪ or ♫
// agcEffect = (lineBuffer[4] == 240 && lineBuffer[5] == 159 && lineBuffer[6] == 142 && lineBuffer[7] == 154 ); //encoding of 🎚 No clue why as not found here https://www.iemoji.com/view/emoji/918/objects/level-slider
lastMode = knownMode;
}
// update inputLevel Slider based on current AGC gain
if ((soundAgc>0) && agcEffect) {
unsigned long now_time = millis();
// "user kick" feature - if user has moved the slider by at least 32 units, we "kick" AGC gain by 30% (up or down)
// only once in 3.5 seconds
if ( (lastMode == knownMode)
&& (abs(last_user_inputLevel - inputLevel) > 31)
&& (now_time - last_kick_time > 3500)) {
if (last_user_inputLevel > inputLevel) multAgc *= 0.60; // down -> reduce gain
if (last_user_inputLevel < inputLevel) multAgc *= 1.50; // up -> increase gain
last_kick_time = now_time;
}
int new_user_inputLevel = 128.0f * multAgc; // scale AGC multiplier so that "1" is at 128
if (multAgc > 1.0f) new_user_inputLevel = 128.0f * (((multAgc - 1.0f) / 4.0f) +1.0f); // compress range so we can show values up to 4
new_user_inputLevel = MIN(MAX(new_user_inputLevel, 0),255);
// update user interfaces - restrict frequency to avoid flooding UI's with small changes
if (( ((now_time - last_update_time > 3500) && (abs(new_user_inputLevel - inputLevel) > 2)) // small change - every 3.5 sec (max)
||((now_time - last_update_time > 2200) && (abs(new_user_inputLevel - inputLevel) > 15)) // medium change
||((now_time - last_update_time > 1200) && (abs(new_user_inputLevel - inputLevel) > 31))) // BIG change - every second
&& !strip.isUpdating()) // don't interfere while strip is updating
{
inputLevel = new_user_inputLevel; // change of least 3 units -> update user variable
updateInterfaces(CALL_MODE_WS_SEND); // is this the correct way to notify UIs ? Yes says blazoncek
last_update_time = now_time;
last_user_inputLevel = new_user_inputLevel;
}
}
}
@@ -1113,16 +1132,27 @@ class AudioReactive : public Usermod {
volumeRaw = 0; volumeSmth = 0;
sampleAgc = 0; sampleAvg = 0;
sampleRaw = 0; rawSampleAgc = 0;
my_magnitude = 0; FFT_Magnitude = 0; FFT_MajorPeak = 0;
my_magnitude = 0; FFT_Magnitude = 0; FFT_MajorPeak = 1;
multAgc = 1;
// reset FFT data
memset(fftCalc, 0, sizeof(fftCalc));
memset(fftAvg, 0, sizeof(fftAvg));
memset(fftResult, 0, sizeof(fftResult));
for(int i=(init?0:1); i<16; i+=2) fftResult[i] = 16; // make a tiny pattern
inputLevel = 128; // resset level slider to default
if (init && FFT_Task) {
vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash
if (udpSyncConnected) { // close UDP sync connection (if open)
udpSyncConnected = false;
fftUdp.stop();
}
} else {
// update has failed or create task requested
if (FFT_Task)
if (FFT_Task) {
vTaskResume(FFT_Task);
else
connected(); // resume UDP
} else
// xTaskCreatePinnedToCore(
xTaskCreate( // no need to "pin" this task to core #0
FFTcode, // Function to implement the task
@@ -1130,11 +1160,11 @@ class AudioReactive : public Usermod {
5000, // Stack size in words
NULL, // Task input parameter
1, // Priority of the task
&FFT_Task // Task handle
&FFT_Task // Task handle
// , 0 // Core where the task should run
);
}
micDataReal = 0.0f; // just to ber sure
micDataReal = 0.0f; // just to be sure
if (enabled) disableSoundProcessing = false;
}
@@ -1182,69 +1212,77 @@ class AudioReactive : public Usermod {
infoArr.add(uiDomString);
if (enabled) {
infoArr = user.createNestedArray(F("Input level"));
uiDomString = F("<div class=\"slider\"><div class=\"sliderwrap il\"><input class=\"noslide\" onchange=\"requestJson({");
uiDomString += FPSTR(_name);
uiDomString += F(":{");
uiDomString += FPSTR(_inputLvl);
uiDomString += F(":parseInt(this.value)}});\" oninput=\"updateTrail(this);\" max=255 min=0 type=\"range\" value=");
uiDomString += inputLevel;
uiDomString += F(" /><div class=\"sliderdisplay\"></div></div></div>"); //<output class=\"sliderbubble\"></output>
infoArr.add(uiDomString);
// Input Level Slider
if (disableSoundProcessing == false) { // only show slider when audio processing is running
if (soundAgc > 0)
infoArr = user.createNestedArray(F("GEQ Input Level")); // if AGC is on, this slider only affects fftResult[] frequencies
else
infoArr = user.createNestedArray(F("Audio Input Level"));
uiDomString = F("<div class=\"slider\"><div class=\"sliderwrap il\"><input class=\"noslide\" onchange=\"requestJson({");
uiDomString += FPSTR(_name);
uiDomString += F(":{");
uiDomString += FPSTR(_inputLvl);
uiDomString += F(":parseInt(this.value)}});\" oninput=\"updateTrail(this);\" max=255 min=0 type=\"range\" value=");
uiDomString += inputLevel;
uiDomString += F(" /><div class=\"sliderdisplay\"></div></div></div>"); //<output class=\"sliderbubble\"></output>
infoArr.add(uiDomString);
}
// The following can be used for troubleshooting user errors and is so not enclosed in #ifdef WLED_DEBUG
// current Audio input
infoArr = user.createNestedArray(F("Audio Source"));
if (audioSyncEnabled & 0x02) {
// UDP sound sync - receive mode
infoArr.add("UDP sound sync");
if (udpSyncConnected) {
if (millis() - last_UDPTime < 2500)
infoArr.add(" - receiving");
else
infoArr.add(" - idle");
} else {
infoArr.add(" - no network");
}
// UDP sound sync - receive mode
infoArr.add(F("UDP sound sync"));
if (udpSyncConnected) {
if (millis() - last_UDPTime < 2500)
infoArr.add(F(" - receiving"));
else
infoArr.add(F(" - idle"));
} else {
infoArr.add(F(" - no connection"));
}
} else {
// Analog or I2S digital input
if (audioSource && (audioSource->isInitialized())) {
// audio source sucessfully configured
if(audioSource->getType() == AudioSource::Type_I2SAdc) {
infoArr.add("ADC analog");
} else {
infoArr.add("I2S digital");
}
// input level or "silence"
if (maxSample5sec > 1.0) {
float my_usage = 100.0f * (maxSample5sec / 255.0f);
snprintf(myStringBuffer, 15, " - peak %3d%%", int(my_usage));
infoArr.add(myStringBuffer);
} else {
infoArr.add(" - quiet");
}
// audio source sucessfully configured
if (audioSource->getType() == AudioSource::Type_I2SAdc) {
infoArr.add(F("ADC analog"));
} else {
infoArr.add(F("I2S digital"));
}
// input level or "silence"
if (maxSample5sec > 1.0) {
float my_usage = 100.0f * (maxSample5sec / 255.0f);
snprintf_P(myStringBuffer, 15, PSTR(" - peak %3d%%"), int(my_usage));
infoArr.add(myStringBuffer);
} else {
infoArr.add(F(" - quiet"));
}
} else {
// error during audio source setup
infoArr.add("not initialized");
infoArr.add(" - check GPIO config");
// error during audio source setup
infoArr.add(F("not initialized"));
infoArr.add(F(" - check GPIO config"));
}
}
// Sound processing (FFT and input filters)
infoArr = user.createNestedArray(F("Sound Processing"));
if (audioSource && (disableSoundProcessing == false)) {
infoArr.add("running");
infoArr.add(F("running"));
} else {
infoArr.add("suspended");
infoArr.add(F("suspended"));
}
// AGC or manual Gain
if((soundAgc==0) && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
if ((soundAgc==0) && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
infoArr = user.createNestedArray(F("Manual Gain"));
float myGain = ((float)sampleGain/40.0f * (float)inputLevel/128.0f) + 1.0f/16.0f; // non-AGC gain from presets
infoArr.add(roundf(myGain*100.0f) / 100.0f);
infoArr.add("x");
}
if(soundAgc && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
if (soundAgc && (disableSoundProcessing == false) && !(audioSyncEnabled & 0x02)) {
infoArr = user.createNestedArray(F("AGC Gain"));
infoArr.add(roundf(multAgc*100.0f) / 100.0f);
infoArr.add("x");
@@ -1253,13 +1291,14 @@ class AudioReactive : public Usermod {
// UDP Sound Sync status
infoArr = user.createNestedArray(F("UDP Sound Sync"));
if (audioSyncEnabled) {
if (audioSyncEnabled & 0x01) {
infoArr.add("send mode");
} else if (audioSyncEnabled & 0x02) {
infoArr.add("receive mode");
}
if (audioSyncEnabled & 0x01) {
infoArr.add(F("send mode"));
} else if (audioSyncEnabled & 0x02) {
infoArr.add(F("receive mode"));
}
} else
infoArr.add("off");
infoArr.add("off");
if (audioSyncEnabled && !udpSyncConnected) infoArr.add(" <i>(unconnected)</i>");
#ifdef WLED_DEBUG
infoArr = user.createNestedArray(F("Sampling time"));
@@ -1372,6 +1411,9 @@ class AudioReactive : public Usermod {
dynLim[F("Rise")] = attackTime;
dynLim[F("Fall")] = decayTime;
JsonObject freqScale = top.createNestedObject("Frequency");
freqScale[F("Scale")] = FFTScalingMode;
JsonObject sync = top.createNestedObject("sync");
sync[F("port")] = audioSyncPort;
sync[F("mode")] = audioSyncEnabled;
@@ -1418,6 +1460,8 @@ class AudioReactive : public Usermod {
configComplete &= getJsonValue(top["dynamics"][F("Rise")], attackTime);
configComplete &= getJsonValue(top["dynamics"][F("Fall")], decayTime);
configComplete &= getJsonValue(top["Frequency"][F("Scale")], FFTScalingMode);
configComplete &= getJsonValue(top["sync"][F("port")], audioSyncPort);
configComplete &= getJsonValue(top["sync"][F("mode")], audioSyncEnabled);
@@ -1443,11 +1487,15 @@ class AudioReactive : public Usermod {
oappend(SET_F("dd=addDropdown('AudioReactive','dynamics:Limiter');"));
oappend(SET_F("addOption(dd,'Off',0);"));
oappend(SET_F("addOption(dd,'On',1);"));
oappend(SET_F("addInfo('AudioReactive:dynamics:Limiter',0,' Limiter On ');")); // 0 is field type, 1 is actual field
//oappend(SET_F("addInfo('AudioReactive:dynamics:Rise',0,'min. ');"));
oappend(SET_F("addInfo('AudioReactive:dynamics:Rise',1,' ms <br /><i>(volume reactive FX only)</i>');"));
//oappend(SET_F("addInfo('AudioReactive:dynamics:Fall',0,'min. ');"));
oappend(SET_F("addInfo('AudioReactive:dynamics:Fall',1,' ms <br /><i>(volume reactive FX only)</i>');"));
oappend(SET_F("addInfo('AudioReactive:dynamics:Limiter',0,' On ');")); // 0 is field type, 1 is actual field
oappend(SET_F("addInfo('AudioReactive:dynamics:Rise',1,'ms <i>(&#x266A; effects only)</i>');"));
oappend(SET_F("addInfo('AudioReactive:dynamics:Fall',1,'ms <i>(&#x266A; effects only)</i>');"));
oappend(SET_F("dd=addDropdown('AudioReactive','Frequency:Scale');"));
oappend(SET_F("addOption(dd,'None',0);"));
oappend(SET_F("addOption(dd,'Linear (Amplitude)',2);"));
oappend(SET_F("addOption(dd,'Square Root (Energy)',3);"));
oappend(SET_F("addOption(dd,'Logarithmic (Loudness)',1);"));
oappend(SET_F("dd=addDropdown('AudioReactive','sync:mode');"));
oappend(SET_F("addOption(dd,'Off',0);"));