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Code shuffling (making bugs)

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This commit is contained in:
Blaz Kristan 2022-06-19 19:15:34 +02:00
parent ac5b3110f2
commit 7ebb58b1fa
1 changed files with 61 additions and 95 deletions
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152
usermods/audioreactive/audio_reactive.h
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@ -40,51 +40,37 @@ constexpr int SAMPLE_RATE = 10240; // Base sample rate in Hz
//#define MAJORPEAK_SUPPRESS_NOISE // define to activate a dirty hack that ignores the lowest + hightest FFT bins
// globals
static byte audioSyncEnabled = 0;
static uint16_t audioSyncPort = 11988;
uint8_t inputLevel = 128; // UI slider value
static uint8_t inputLevel = 128; // UI slider value
static uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value)
static uint8_t sampleGain = 1; // sample gain (config value)
static uint8_t soundAgc = 0; // Automagic gain control: 0 - none, 1 - normal, 2 - vivid, 3 - lazy (config value)
static uint8_t audioSyncEnabled = 0; // bit field: bit 0 - send, bit 1 - receive
//
// AGC presets
// Note: in C++, "const" implies "static" - no need to explicitly declare everything as "static const"
//
#define AGC_NUM_PRESETS 3 // AGC currently has 3 presets: normal, vivid, lazy
// Normal, Vivid, Lazy
const double agcSampleDecay[AGC_NUM_PRESETS] = // decay factor for sampleMax, in case the current sample is below sampleMax
{0.9994, 0.9985, 0.9997};
const float agcZoneLow[AGC_NUM_PRESETS] = // low volume emergency zone
{ 32, 28, 36};
const float agcZoneHigh[AGC_NUM_PRESETS] = // high volume emergency zone
{ 240, 240, 248};
const float agcZoneStop[AGC_NUM_PRESETS] = // disable AGC integrator if we get above this level
{ 336, 448, 304};
const float agcTarget0[AGC_NUM_PRESETS] = // first AGC setPoint -> between 40% and 65%
{ 112, 144, 164};
const float agcTarget0Up[AGC_NUM_PRESETS] = // setpoint switching value (a poor man's bang-bang)
{ 88, 64, 116};
const float agcTarget1[AGC_NUM_PRESETS] = // second AGC setPoint -> around 85%
{ 220, 224, 216};
const double agcFollowFast[AGC_NUM_PRESETS] = // quickly follow setpoint - ~0.15 sec
{ 1.0/192.0, 1.0/128.0, 1.0/256.0};
const double agcFollowSlow[AGC_NUM_PRESETS] = // slowly follow setpoint - ~2-15 secs
{1.0/6144.0, 1.0/4096.0, 1.0/8192.0};
const double agcControlKp[AGC_NUM_PRESETS] = // AGC - PI control, proportional gain parameter
{ 0.6, 1.5, 0.65};
const double agcControlKi[AGC_NUM_PRESETS] = // AGC - PI control, integral gain parameter
{ 1.7, 1.85, 1.2};
const float agcSampleSmooth[AGC_NUM_PRESETS] = // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value)
{ 1.0/12.0, 1.0/6.0, 1.0/16.0};
//
#define AGC_NUM_PRESETS 3 // AGC presets: normal, vivid, lazy
const double agcSampleDecay[AGC_NUM_PRESETS] = { 0.9994f, 0.9985f, 0.9997f}; // decay factor for sampleMax, in case the current sample is below sampleMax
const float agcZoneLow[AGC_NUM_PRESETS] = { 32, 28, 36}; // low volume emergency zone
const float agcZoneHigh[AGC_NUM_PRESETS] = { 240, 240, 248}; // high volume emergency zone
const float agcZoneStop[AGC_NUM_PRESETS] = { 336, 448, 304}; // disable AGC integrator if we get above this level
const float agcTarget0[AGC_NUM_PRESETS] = { 112, 144, 164}; // first AGC setPoint -> between 40% and 65%
const float agcTarget0Up[AGC_NUM_PRESETS] = { 88, 64, 116}; // setpoint switching value (a poor man's bang-bang)
const float agcTarget1[AGC_NUM_PRESETS] = { 220, 224, 216}; // second AGC setPoint -> around 85%
const double agcFollowFast[AGC_NUM_PRESETS] = { 1/192.f, 1/128.f, 1/256.f}; // quickly follow setpoint - ~0.15 sec
const double agcFollowSlow[AGC_NUM_PRESETS] = {1/6144.f,1/4096.f,1/8192.f}; // slowly follow setpoint - ~2-15 secs
const double agcControlKp[AGC_NUM_PRESETS] = { 0.6f, 1.5f, 0.65f}; // AGC - PI control, proportional gain parameter
const double agcControlKi[AGC_NUM_PRESETS] = { 1.7f, 1.85f, 1.2f}; // AGC - PI control, integral gain parameter
const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; // smoothing factor for sampleAgc (use rawSampleAgc if you want the non-smoothed value)
// AGC presets end
//
static AudioSource *audioSource = nullptr;
//static uint16_t micData; // Analog input for FFT
static uint16_t micDataSm; // Smoothed mic data, as it's a bit twitchy
static float micDataReal = 0.0f; // future support - this one has the full 24bit MicIn data - lowest 8bit after decimal point
static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
////////////////////
// Begin FFT Code //
@ -93,23 +79,9 @@ const float agcSampleSmooth[AGC_NUM_PRESETS] = // smoothing factor for sampleA
// FFT Variables
constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2
const uint16_t samples = 512; // This value MUST ALWAYS be a power of 2
//unsigned int sampling_period_us;
//unsigned long microseconds;
static AudioSource *audioSource = nullptr;
static byte soundSquelch = 10; // default squelch value for volume reactive routines
static byte sampleGain = 1; // default sample gain
static uint16_t micData; // Analog input for FFT
static uint16_t micDataSm; // Smoothed mic data, as it's a bit twitchy
static float micDataReal = 0.0f; // future support - this one has the full 24bit MicIn data - lowest 8bit after decimal point
static byte soundAgc = 0; // default Automagic gain control
static float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our multiplier
static double FFT_MajorPeak = 0;
static double FFT_Magnitude = 0;
//static uint16_t mAvg = 0;
// These are the input and output vectors. Input vectors receive computed results from FFT.
static double vReal[samplesFFT];
@ -161,7 +133,7 @@ void FFTcode(void * parameter)
if (audioSource) audioSource->getSamples(vReal, samplesFFT);
// old code - Last sample in vReal is our current mic sample
//micDataSm = (uint16_t)vReal[samples - 1]; // will do a this a bit later
//micDataSm = (uint16_t)vReal[samplesFFT - 1]; // will do a this a bit later
//micDataSm = ((micData * 3) + micData)/4;
const int halfSamplesFFT = samplesFFT / 2; // samplesFFT divided by 2
@ -387,32 +359,37 @@ class AudioReactive : public Usermod {
bool initDone = false;
const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED
// variables used in effects
uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger
uint8_t binNum = 8; // Used to select the bin for FFT based beat detection.
uint8_t myVals[32]; // Used to store a pile of samples because WLED frame rate and WLED sample rate are not synchronized. Frame rate is too low.
bool samplePeak = 0; // Boolean flag for peak. Responding routine must reset this flag
int16_t sample; // either sampleRaw or rawSampleAgc depending on soundAgc
float sampleSmth; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
#ifdef MIC_SAMPLING_LOG
uint8_t targetAgc = 60; // This is our setPoint at 20% of max for the adjusted output (used only in logAudio())
#endif
uint8_t myVals[32]; // Used to store a pile of samples because WLED frame rate and WLED sample rate are not synchronized. Frame rate is too low.
bool samplePeak = 0; // Boolean flag for peak. Responding routine must reset this flag
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 sample; // Current sample. Must only be updated ONCE!!!
int16_t sampleRaw; // Current sample. Must only be updated ONCE!!! (amplified mic value by sampleGain and inputLevel; smoothed over 16 samples)
float sampleMax = 0.0f; // Max sample over a few seconds. Needed for AGC controler.
float sampleReal = 0.0f; // "sample" as float, to provide bits that are lost otherwise. Needed for AGC.
float sampleAvg = 0.0f; // Smoothed Average
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;
uint32_t lastTime = 0;
float micLev = 0.0f; // Used to convert returned value to have '0' as minimum. A leveller
float expAdjF = 0.0f; // Used for exponential filter.
bool udpSyncConnected = false;
uint16_t audioSyncPort = 11988;
// used for AGC
uint8_t lastMode = 0; // last known effect mode
bool agcEffect = false;
int last_soundAgc = -1;
float control_integrated = 0.0f; // "integrator control" = accumulated error
float control_integrated = 0.0f; // persistent across calls to agcAvg(); "integrator control" = accumulated error
unsigned long last_update_time = 0;
unsigned long last_kick_time = 0;
uint8_t last_user_inputLevel = 0;
@ -586,7 +563,7 @@ class AudioReactive : public Usermod {
// NOW finally amplify the signal
tmpAgc = sampleReal * multAgcTemp; // apply gain to signal
if(fabs(sampleReal) < 2.0f) tmpAgc = 0; // apply squelch threshold
if (fabs(sampleReal) < 2.0f) tmpAgc = 0; // apply squelch threshold
//tmpAgc = constrain(tmpAgc, 0, 255);
if (tmpAgc > 255) tmpAgc = 255; // limit to 8bit
if (tmpAgc < 1) tmpAgc = 0; // just to be sure
@ -598,7 +575,7 @@ class AudioReactive : public Usermod {
if (fabs(tmpAgc) < 1.0f)
sampleAgc = 0.5f * tmpAgc + 0.5f * sampleAgc; // fast path to zero
else
sampleAgc = sampleAgc + agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
sampleAgc += agcSampleSmooth[AGC_preset] * (tmpAgc - sampleAgc); // smooth path
//userVar0 = sampleAvg * 4;
//if (userVar0 > 255) userVar0 = 255;
@ -610,7 +587,6 @@ class AudioReactive : public Usermod {
void getSample()
{
float sampleAdj; // Gain adjusted sample value
float expAdjF; // Used for exponential filter.
float tmpSample; // An interim sample variable used for calculatioins.
const float weighting = 0.2f; // Exponential filter weighting. Will be adjustable in a future release.
const int AGC_preset = (soundAgc > 0)? (soundAgc-1): 0; // make sure the _compiler_ knows this value will not change while we are inside the function
@ -620,12 +596,8 @@ class AudioReactive : public Usermod {
micDataReal = micIn;
#else
micIn = micDataSm; // micDataSm = ((micData * 3) + micData)/4;
DEBUGSR_PRINT("micIn:\tmicData:\tmicIn>>2:\tmic_In_abs:\tsample:\tsampleAdj:\tsampleAvg:\n");
DEBUGSR_PRINT(micIn); DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(micData);
// We're still using 10 bit, but changing the analog read resolution in usermod.cpp
//if (digitalMic == false) micIn = micIn >> 2; // ESP32 has 2 more bits of A/D than ESP8266, so we need to normalize to 10 bit.
//DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(micIn);
//DEBUGSR_PRINT("micIn:\tmicData:\tmicIn>>2:\tmic_In_abs:\tsample:\tsampleAdj:\tsampleAvg:\n");
//DEBUGSR_PRINT(micIn); DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(micData);
#endif
// Note to self: the next line kills 80% of sample - "miclev" filter runs at "full arduino loop" speed, following the signal almost instantly!
@ -653,7 +625,7 @@ class AudioReactive : public Usermod {
sampleReal = tmpSample;
sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
sample = (int16_t)sampleAdj; // ONLY update sample ONCE!!!!
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)) {
@ -668,7 +640,7 @@ class AudioReactive : public Usermod {
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sample);
DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sampleRaw);
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(sampleAvg); DEBUGSR_PRINT("\n\n");
// Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
@ -679,7 +651,7 @@ class AudioReactive : public Usermod {
udpSamplePeak = 0;
}
if (userVar1 == 0) samplePeak = 0;
//if (userVar1 == 0) samplePeak = 0;
// Poor man's beat detection by seeing if sample > Average + some value.
// Serial.print(binNum); Serial.print("\t");
// Serial.print(fftBin[binNum]);
@ -713,7 +685,7 @@ class AudioReactive : public Usermod {
}
transmitData.sampleAgc = sampleAgc;
transmitData.sample = sample;
transmitData.sample = sampleRaw;
transmitData.sampleAvg = sampleAvg;
transmitData.samplePeak = udpSamplePeak;
udpSamplePeak = 0; // Reset udpSamplePeak after we've transmitted it
@ -756,7 +728,7 @@ class AudioReactive : public Usermod {
sampleAgc = receivedPacket->sampleAgc;
rawSampleAgc = receivedPacket->sampleAgc;
sample = receivedPacket->sample;
sampleRaw = receivedPacket->sample;
sampleAvg = receivedPacket->sampleAvg;
// Only change samplePeak IF it's currently false.
@ -775,7 +747,7 @@ class AudioReactive : public Usermod {
public:
//Functions called by WLED
//Functions called by WLED or other usermods
/*
* setup() is called once at boot. WiFi is not yet connected at this point.
@ -797,7 +769,7 @@ class AudioReactive : public Usermod {
um_data->u_type[ 1] = UMT_BYTE;
um_data->u_data[ 2] = &sampleAgc; //*used (can be calculated as: sampleReal * multAgc) (..., Juggles, ..., Pixels, Puddlepeak, Freqmatrix)
um_data->u_type[ 2] = UMT_FLOAT;
um_data->u_data[ 3] = &sample; //*used (Matripix, Noisemeter, Pixelwave, Puddles, 2D Swirl, for debugging Gravimeter)
um_data->u_data[ 3] = &sampleRaw; //*used (Matripix, Noisemeter, Pixelwave, Puddles, 2D Swirl, for debugging Gravimeter)
um_data->u_type[ 3] = UMT_INT16;
um_data->u_data[ 4] = &rawSampleAgc; //*used (Matripix, Noisemeter, Pixelwave, Puddles, 2D Swirl)
um_data->u_type[ 4] = UMT_INT16;
@ -821,9 +793,9 @@ class AudioReactive : public Usermod {
um_data->u_type[13] = UMT_FLOAT;
um_data->u_data[14] = myVals; //*used (only once, Pixels)
um_data->u_type[14] = UMT_UINT16_ARR;
um_data->u_data[15] = &soundSquelch; //*used (only once, Binmap)
um_data->u_data[15] = &soundSquelch; //*used (for debugging) (only once, Binmap)
um_data->u_type[15] = UMT_BYTE;
um_data->u_data[16] = fftBin; //*used (only once, Binmap)
um_data->u_data[16] = fftBin; //*used (for debugging) (only once, Binmap)
um_data->u_type[16] = UMT_FLOAT_ARR;
um_data->u_data[17] = &inputLevel; // global UI element!!! (Gravimeter, Binmap)
um_data->u_type[17] = UMT_BYTE;
@ -915,28 +887,23 @@ class AudioReactive : public Usermod {
if (!enabled || strip.isUpdating()) return;
if (!(audioSyncEnabled & 0x02)) { // Only run the sampling code IF we're not in Receive mode
bool agcEffect = false;
if (soundAgc > AGC_NUM_PRESETS) soundAgc = 0; // make sure that AGC preset is valid (to avoid array bounds violation)
getSample(); // Sample the microphone
agcAvg(); // Calculated the PI adjusted value as sampleAvg
myVals[millis()%32] = sampleAgc; // filling values semi randomly
uint8_t knownMode = strip.getMainSegment().mode;
myVals[millis()%32] = sampleAgc; // filling values semi randomly (why?)
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[3];
/* uint8_t printedChars = */ extractModeName(knownMode, JSON_mode_names, lineBuffer, 3); // use of JSON_mode_names is deprecated, use nullptr
//used the following code to reverse engineer this
// Serial.println(lineBuffer);
// for (uint8_t i = 0; i<printedChars; i++) {
// Serial.print(i);
// Serial.print( ": ");
// Serial.println(uint8_t(lineBuffer[i]));
// }
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
// if (agcEffect)
// Serial.println("found ♪ or ♫");
lastMode = knownMode;
}
// update inputLevel Slider based on current AGC gain
@ -968,7 +935,6 @@ class AudioReactive : public Usermod {
last_user_inputLevel = new_user_inputLevel;
}
}
lastMode = knownMode;
#if defined(MIC_LOGGER) || defined(MIC_SAMPLING_LOG) || defined(FFT_SAMPLING_LOG)
EVERY_N_MILLIS(20) {