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Merge branch 'audioreactive-prototype' into dev

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This commit is contained in:
Blaz Kristan 2022-09-02 23:36:44 +02:00
commit cf51892782
2 changed files with 179 additions and 104 deletions
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usermods/audioreactive/audio_reactive.h
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@ -24,7 +24,7 @@
// #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter) // #define MIC_LOGGER // MIC sampling & sound input debugging (serial plotter)
// #define FFT_SAMPLING_LOG // FFT result debugging // #define FFT_SAMPLING_LOG // FFT result debugging
// #define SR_DEBUG // generic SR DEBUG messages // #define SR_DEBUG // generic SR DEBUG messages
// #define NO_MIC_LOGGER // exclude MIC_LOGGER from SR_DEBUG
#ifdef SR_DEBUG #ifdef SR_DEBUG
#define DEBUGSR_PRINT(x) Serial.print(x) #define DEBUGSR_PRINT(x) Serial.print(x)
@ -85,18 +85,25 @@ const float agcSampleSmooth[AGC_NUM_PRESETS] = { 1/12.f, 1/6.f, 1/16.f}; //
static AudioSource *audioSource = nullptr; 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 volatile bool disableSoundProcessing = false; // if true, sound processing (FFT, filters, AGC) will be suspended. "volatile" as its shared between tasks.
// audioreactive variables shared with FFT task
static float micDataReal = 0.0f; // MicIn data with full 24bit resolution - lowest 8bit after decimal point 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 float multAgc = 1.0f; // sample * multAgc = sampleAgc. Our AGC multiplier
static float sampleAvg = 0.0f; // Smoothed Average sample - sampleAvg < 1 means "quiet" (simple noise gate)
// peak detection
static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
static uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
static uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
static unsigned long timeOfPeak = 0; // time of last sample peak detection.
static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[])
static void autoResetPeak(void); // peak auto-reset function
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 // // Begin FFT Code //
//////////////////// ////////////////////
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
// lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2 // lib_deps += https://github.com/kosme/arduinoFFT#develop @ 1.9.2
#define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and an a few other speedups #define FFT_SPEED_OVER_PRECISION // enables use of reciprocals (1/x etc), and an a few other speedups
@ -105,21 +112,22 @@ static float sampleAgc = 0.0f; // Smoothed AGC sample
#endif #endif
#include "arduinoFFT.h" #include "arduinoFFT.h"
// FFT Variables // FFT Output variables shared with animations
#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !!
static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency
static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency
static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects
// FFT Constants
constexpr uint16_t samplesFFT = 512; // Samples in an FFT batch - This value MUST ALWAYS be a power of 2 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. constexpr uint16_t samplesFFT_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
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. // These are the input and output vectors. Input vectors receive computed results from FFT.
static float vReal[samplesFFT] = {0.0f}; static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
static float vImag[samplesFFT] = {0.0f}; static float vImag[samplesFFT] = {0.0f}; // imaginary parts
static float fftBin[samplesFFT_2] = {0.0f};
// the following are observed values, supported by a bit of "educated guessing" // 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 0.65f // 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 FFT_DOWNSCALE 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
#define LOG_256 5.54517744 #define LOG_256 5.54517744
@ -128,13 +136,11 @@ static float windowWeighingFactors[samplesFFT] = {0.0f};
#endif #endif
// Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256. // Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256.
// Oh, and bins 0,1,2 are no good, so we'll zero them out. static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f};
static float fftCalc[16] = {0.0f}; static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON)
static uint8_t fftResult[16] = {0}; // Our calculated result table, which we feed to the animations.
#ifdef SR_DEBUG #ifdef SR_DEBUG
static float fftResultMax[16] = {0.0f}; // A table used for testing to determine how our post-processing is working. static float fftResultMax[NUM_GEQ_CHANNELS] = {0.0f}; // A table used for testing to determine how our post-processing is working.
#endif #endif
static float fftAvg[16] = {0.0f};
#ifdef WLED_DEBUG #ifdef WLED_DEBUG
static unsigned long fftTime = 0; static unsigned long fftTime = 0;
@ -142,7 +148,7 @@ static unsigned long sampleTime = 0;
#endif #endif
// Table of multiplication factors so that we can even out the frequency response. // 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 }; static float fftResultPink[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 };
// Create FFT object // Create FFT object
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT #ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
@ -161,12 +167,12 @@ static float mapf(float x, float in_min, float in_max, float out_min, float out_
static float fftAddAvg(int from, int to) { static float fftAddAvg(int from, int to) {
float result = 0.0f; float result = 0.0f;
for (int i = from; i <= to; i++) { for (int i = from; i <= to; i++) {
result += fftBin[i]; result += vReal[i];
} }
return result / float(to - from + 1); return result / float(to - from + 1);
} }
// FFT main code // FFT main task
void FFTcode(void * parameter) void FFTcode(void * parameter)
{ {
DEBUGSR_PRINT("FFT started on core: "); DEBUGSR_PRINTLN(xPortGetCoreID()); DEBUGSR_PRINT("FFT started on core: "); DEBUGSR_PRINTLN(xPortGetCoreID());
@ -237,9 +243,9 @@ void FFTcode(void * parameter)
#endif #endif
FFT_MajorPeak = constrain(FFT_MajorPeak, 1.0f, 11025.0f); // restrict value to range expected by effects 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. for (int i = 0; i < samplesFFT; i++) {
float t = fabsf(vReal[i]); // just to be sure - values in fft bins should be positive any way 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. vReal[i] = t / 16.0f; // Reduce magnitude. Want end result to be scaled linear and ~4096 max.
} // for() } // for()
// mapping of FFT result bins to frequency channels // mapping of FFT result bins to frequency channels
@ -292,14 +298,14 @@ void FFTcode(void * parameter)
// don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise) // don't use the last bins from 216 to 255. They are usually contaminated by aliasing (aka noise)
#endif #endif
} else { // noise gate closed - just decay old values } else { // noise gate closed - just decay old values
for (int i=0; i < 16; i++) { for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
fftCalc[i] *= 0.85f; // decay to zero fftCalc[i] *= 0.85f; // decay to zero
if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f; if (fftCalc[i] < 4.0f) fftCalc[i] = 0.0f;
} }
} }
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling) // post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
for (int i=0; i < 16; i++) { for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
if (sampleAvg > 1) { // noise gate open if (sampleAvg > 1) { // noise gate open
// Adjustment for frequency curves. // Adjustment for frequency curves.
@ -378,11 +384,43 @@ void FFTcode(void * parameter)
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
} }
#endif #endif
// run peak detection
autoResetPeak();
detectSamplePeak();
} // for(;;) } // for(;;)ever
} // FFTcode() } // FFTcode() task end
////////////////////
// Peak detection //
////////////////////
// peak detection is called from FFT task when vReal[] contains valid FFT results
static void detectSamplePeak(void) {
// Poor man's beat detection by seeing if sample > Average + some value.
if ((sampleAvg > 1) && (maxVol > 0) && (binNum > 1) && (vReal[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;
}
}
static void autoResetPeak(void) {
uint16_t MinShowDelay = MAX(50, strip.getMinShowDelay()); // Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
if (audioSyncEnabled == 0) udpSamplePeak = false; // this is normally reset by transmitAudioData
}
}
////////////////////
// usermod class //
////////////////////
//class name. Use something descriptive and leave the ": public Usermod" part :) //class name. Use something descriptive and leave the ": public Usermod" part :)
class AudioReactive : public Usermod { class AudioReactive : public Usermod {
@ -453,40 +491,36 @@ class AudioReactive : public Usermod {
double FFT_MajorPeak; // 08 Bytes double FFT_MajorPeak; // 08 Bytes
}; };
WiFiUDP fftUdp;
// set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer) // set your config variables to their boot default value (this can also be done in readFromConfig() or a constructor if you prefer)
bool enabled = false; bool enabled = false;
bool initDone = false; bool initDone = false;
// variables for UDP sound sync
WiFiUDP fftUdp; // UDP object for sound sync (from WiFi UDP, not Async UDP!)
bool udpSyncConnected = false;// UDP connection status -> true if connected to multicast group
unsigned long lastTime = 0; // last time of running UDP Microphone Sync
const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED const uint16_t delayMs = 10; // I don't want to sample too often and overload WLED
uint16_t audioSyncPort= 11988;// default port for UDP sound sync
// used for AGC
int last_soundAgc = -1; // used to detect AGC mode change (for resetting AGC internal error buffers)
double control_integrated = 0.0; // persistent across calls to agcAvg(); "integrator control" = accumulated error
// 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
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)
int16_t rawSampleAgc = 0; // not smoothed AGC sample
float sampleAgc = 0.0f; // Smoothed AGC sample
// variables used in effects // variables used in effects
uint8_t maxVol = 10; // Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
uint8_t binNum = 8; // Used to select the bin for FFT based beat detection (deprecated)
bool samplePeak = 0; // Boolean flag for peak. Responding routine must reset this flag
float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample float volumeSmth = 0.0f; // either sampleAvg or sampleAgc depending on soundAgc; smoothed sample
int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc int16_t volumeRaw = 0; // either sampleRaw or rawSampleAgc depending on soundAgc
float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc float my_magnitude =0.0f; // FFT_Magnitude, scaled by multAgc
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
double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler.
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
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
int last_soundAgc = -1;
double control_integrated = 0.0; // 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;
// used to feed "Info" Page // used to feed "Info" Page
unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket unsigned long last_UDPTime = 0; // time of last valid UDP sound sync datapacket
float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds float maxSample5sec = 0.0f; // max sample (after AGC) in last 5 seconds
@ -503,6 +537,10 @@ class AudioReactive : public Usermod {
static const char UDP_SYNC_HEADER_v1[]; static const char UDP_SYNC_HEADER_v1[];
// private methods // private methods
////////////////////
// Debug support //
////////////////////
void logAudio() void logAudio()
{ {
#ifdef MIC_LOGGER #ifdef MIC_LOGGER
@ -525,7 +563,7 @@ class AudioReactive : public Usermod {
#ifdef FFT_SAMPLING_LOG #ifdef FFT_SAMPLING_LOG
#if 0 #if 0
for(int i=0; i<16; i++) { for(int i=0; i<NUM_GEQ_CHANNELS; i++) {
Serial.print(fftResult[i]); Serial.print(fftResult[i]);
Serial.print("\t"); Serial.print("\t");
} }
@ -551,11 +589,11 @@ class AudioReactive : public Usermod {
int maxVal = minimumMaxVal; int maxVal = minimumMaxVal;
int minVal = 0; int minVal = 0;
for(int i = 0; i < 16; i++) { for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
if(fftResult[i] > maxVal) maxVal = fftResult[i]; if(fftResult[i] > maxVal) maxVal = fftResult[i];
if(fftResult[i] < minVal) minVal = fftResult[i]; if(fftResult[i] < minVal) minVal = fftResult[i];
} }
for(int i = 0; i < 16; i++) { for(int i = 0; i < NUM_GEQ_CHANNELS; i++) {
Serial.print(i); Serial.print(":"); Serial.print(i); Serial.print(":");
Serial.printf("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1)); Serial.printf("%04ld ", map(fftResult[i], 0, (scaleValuesFromCurrentMaxVal ? maxVal : defaultScalingFromHighValue), (mapValuesToPlotterSpace*i*scalingToHighValue)+0, (mapValuesToPlotterSpace*i*scalingToHighValue)+scalingToHighValue-1));
} }
@ -574,6 +612,10 @@ class AudioReactive : public Usermod {
} // logAudio() } // logAudio()
//////////////////////
// Audio Processing //
//////////////////////
/* /*
* A "PI controller" multiplier to automatically adjust sound sensitivity. * A "PI controller" multiplier to automatically adjust sound sensitivity.
* *
@ -668,7 +710,7 @@ class AudioReactive : public Usermod {
last_soundAgc = soundAgc; last_soundAgc = soundAgc;
} // agcAvg() } // agcAvg()
// post-processing and filtering of MIC sample (micDataReal) from FFTcode()
void getSample() void getSample()
{ {
float sampleAdj; // Gain adjusted sample value float sampleAdj; // Gain adjusted sample value
@ -729,24 +771,6 @@ class AudioReactive : public Usermod {
if (sampleMax < 0.5f) sampleMax = 0.0f; if (sampleMax < 0.5f) sampleMax = 0.0f;
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples. sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
// Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
uint16_t MinShowDelay = strip.getMinShowDelay();
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
udpSamplePeak = false;
}
//if (userVar1 == 0) samplePeak = 0;
// Poor man's beat detection by seeing if sample > Average + some value.
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;
}
} // getSample() } // getSample()
@ -781,6 +805,26 @@ class AudioReactive : public Usermod {
} }
//////////////////////
// UDP Sound Sync //
//////////////////////
// try to establish UDP sound sync connection
void connectUDPSoundSync(void) {
// This function tries to establish a UDP sync connection if needed
// necessary as we also want to transmit in "AP Mode", but the standard "connected()" callback only reacts on STA connection
static unsigned long last_connection_attempt = 0;
if ((audioSyncPort <= 0) || ((audioSyncEnabled & 0x03) == 0)) return; // Sound Sync not enabled
if (udpSyncConnected) return; // already connected
if (!(apActive || interfacesInited)) return; // neither AP nor other connections availeable
if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds
// if we arrive here, we need a UDP connection but don't have one
last_connection_attempt = millis();
connected(); // try to start UDP
}
void transmitAudioData() void transmitAudioData()
{ {
if (!udpSyncConnected) return; if (!udpSyncConnected) return;
@ -795,7 +839,7 @@ class AudioReactive : public Usermod {
udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it udpSamplePeak = false; // Reset udpSamplePeak after we've transmitted it
transmitData.reserved1 = 0; transmitData.reserved1 = 0;
for (int i = 0; i < 16; i++) { for (int i = 0; i < NUM_GEQ_CHANNELS; i++) {
transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254); transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254);
} }
@ -808,12 +852,10 @@ class AudioReactive : public Usermod {
return; return;
} // transmitAudioData() } // transmitAudioData()
static bool isValidUdpSyncVersion(const char *header) { static bool isValidUdpSyncVersion(const char *header) {
return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0; return strncmp_P(header, PSTR(UDP_SYNC_HEADER), 6) == 0;
} }
bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received. bool receiveAudioData() // check & process new data. return TRUE in case that new audio data was received.
{ {
if (!udpSyncConnected) return false; if (!udpSyncConnected) return false;
@ -839,13 +881,7 @@ class AudioReactive : public Usermod {
sampleAgc = volumeSmth; sampleAgc = volumeSmth;
multAgc = 1.0f; multAgc = 1.0f;
// auto-reset sample peak. Need to do it here, because getSample() is not running autoResetPeak();
uint16_t MinShowDelay = strip.getMinShowDelay();
if (millis() - timeOfPeak > MinShowDelay) { // Auto-reset of samplePeak after a complete frame has passed.
samplePeak = false;
udpSamplePeak = false;
}
//if (userVar1 == 0) samplePeak = 0;
// Only change samplePeak IF it's currently false. // Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond. // If it's true already, then the animation still needs to respond.
if (!samplePeak) { if (!samplePeak) {
@ -855,7 +891,7 @@ class AudioReactive : public Usermod {
} }
//These values are only available on the ESP32 //These values are only available on the ESP32
for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i]; for (int i = 0; i < NUM_GEQ_CHANNELS; i++) fftResult[i] = receivedPacket->fftResult[i];
my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f); my_magnitude = fmaxf(receivedPacket->FFT_Magnitude, 0.0f);
FFT_Magnitude = my_magnitude; FFT_Magnitude = my_magnitude;
@ -869,6 +905,10 @@ class AudioReactive : public Usermod {
} }
//////////////////////
// usermod functions//
//////////////////////
public: public:
//Functions called by WLED or other usermods //Functions called by WLED or other usermods
@ -961,6 +1001,7 @@ class AudioReactive : public Usermod {
disableSoundProcessing = true; disableSoundProcessing = true;
} }
if (enabled) connectUDPSoundSync();
initDone = true; initDone = true;
} }
@ -971,6 +1012,11 @@ class AudioReactive : public Usermod {
*/ */
void connected() void connected()
{ {
if (udpSyncConnected) { // clean-up: if open, close old UDP sync connection
udpSyncConnected = false;
fftUdp.stop();
}
if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) { if (audioSyncPort > 0 && (audioSyncEnabled & 0x03)) {
#ifndef ESP8266 #ifndef ESP8266
udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort); udpSyncConnected = fftUdp.beginMulticast(IPAddress(239, 0, 0, 1), audioSyncPort);
@ -1067,9 +1113,13 @@ class AudioReactive : public Usermod {
if (soundAgc) my_magnitude *= multAgc; 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 limitSampleDynamics();
} } // if (!disableSoundProcessing)
autoResetPeak(); // auto-reset sample peak after strip minShowDelay
if (!udpSyncConnected) udpSamplePeak = false; // reset UDP samplePeak while UDP is unconnected
connectUDPSoundSync(); // ensure we have a connection - if needed
// UDP Microphone Sync - receive mode // UDP Microphone Sync - receive mode
if ((audioSyncEnabled & 0x02) && udpSyncConnected) { if ((audioSyncEnabled & 0x02) && udpSyncConnected) {
@ -1092,7 +1142,7 @@ class AudioReactive : public Usermod {
} }
#endif #endif
// peak sample from last 5 seconds // Info Page: keep max sample from last 5 seconds
if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) { if ((millis() - sampleMaxTimer) > CYCLE_SAMPLEMAX) {
sampleMaxTimer = millis(); sampleMaxTimer = millis();
maxSample5sec = (0.15 * maxSample5sec) + 0.85 *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing maxSample5sec = (0.15 * maxSample5sec) + 0.85 *((soundAgc) ? sampleAgc : sampleAvg); // reset, and start with some smoothing
@ -1100,6 +1150,7 @@ class AudioReactive : public Usermod {
} else { } else {
if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume if ((sampleAvg >= 1)) maxSample5sec = fmaxf(maxSample5sec, (soundAgc) ? rawSampleAgc : sampleRaw); // follow maximum volume
} }
//UDP Microphone Sync - transmit mode //UDP Microphone Sync - transmit mode
if ((audioSyncEnabled & 0x01) && (millis() - lastTime > 20)) { if ((audioSyncEnabled & 0x01) && (millis() - lastTime > 20)) {
// Only run the transmit code IF we're in Transmit mode // Only run the transmit code IF we're in Transmit mode
@ -1137,8 +1188,9 @@ class AudioReactive : public Usermod {
memset(fftCalc, 0, sizeof(fftCalc)); memset(fftCalc, 0, sizeof(fftCalc));
memset(fftAvg, 0, sizeof(fftAvg)); memset(fftAvg, 0, sizeof(fftAvg));
memset(fftResult, 0, sizeof(fftResult)); memset(fftResult, 0, sizeof(fftResult));
for(int i=(init?0:1); i<16; i+=2) fftResult[i] = 16; // make a tiny pattern for(int i=(init?0:1); i<NUM_GEQ_CHANNELS; i+=2) fftResult[i] = 16; // make a tiny pattern
inputLevel = 128; // resset level slider to default inputLevel = 128; // resset level slider to default
autoResetPeak();
if (init && FFT_Task) { if (init && FFT_Task) {
vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash vTaskSuspend(FFT_Task); // update is about to begin, disable task to prevent crash
@ -1186,6 +1238,10 @@ class AudioReactive : public Usermod {
} }
////////////////////////////
// Settings and Info Page //
////////////////////////////
/* /*
* addToJsonInfo() can be used to add custom entries to the /json/info part of the JSON API. * addToJsonInfo() can be used to add custom entries to the /json/info part of the JSON API.
* Creating an "u" object allows you to add custom key/value pairs to the Info section of the WLED web UI. * Creating an "u" object allows you to add custom key/value pairs to the Info section of the WLED web UI.
@ -1307,6 +1363,10 @@ class AudioReactive : public Usermod {
infoArr.add(fftTime-sampleTime); infoArr.add(fftTime-sampleTime);
infoArr.add("ms"); infoArr.add("ms");
#endif #endif
// add a small horizontal line, for better readability
infoArr = user.createNestedArray(F("<hr style=\"height:1px;border-width:0;color:gray;background-color:gray\" />"));
infoArr.add(F(" <hr style=\"height:1px;border-width:0;color:gray;background-color:gray\" /> "));
} }
} }

43
wled00/FX.cpp
View File

@ -6251,7 +6251,7 @@ uint16_t mode_gravcentric(void) { // Gravcentric. By Andrew
return FRAMETIME; return FRAMETIME;
} // mode_gravcentric() } // mode_gravcentric()
static const char _data_FX_MODE_GRAVCENTRIC[] PROGMEM = "Gravcentric@Rate of fall,Sensitivity;!;!;ix=128,mp12=2,ssim=0,1d,vo"; // Circle, Beatsin static const char _data_FX_MODE_GRAVCENTRIC[] PROGMEM = "Gravcentric@Rate of fall,Sensitivity;!;!;ix=128,mp12=3,ssim=0,1d,vo"; // Corner, Beatsin
/////////////////////// ///////////////////////
@ -6387,7 +6387,7 @@ uint16_t mode_midnoise(void) { // Midnoise. By Andrew Tuline.
return FRAMETIME; return FRAMETIME;
} // mode_midnoise() } // mode_midnoise()
static const char _data_FX_MODE_MIDNOISE[] PROGMEM = "Midnoise@Fade rate,Maximum length;,!;!;ix=128,mp12=2,ssim=0,1d,vo"; // Circle, Beatsin static const char _data_FX_MODE_MIDNOISE[] PROGMEM = "Midnoise@Fade rate,Maximum length;,!;!;ix=128,mp12=1,ssim=0,1d,vo"; // Bar, Beatsin
////////////////////// //////////////////////
@ -6512,7 +6512,7 @@ uint16_t mode_plasmoid(void) { // Plasmoid. By Andrew Tuline.
} }
float volumeSmth = *(float*) um_data->u_data[0]; float volumeSmth = *(float*) um_data->u_data[0];
SEGMENT.fadeToBlackBy(64); SEGMENT.fadeToBlackBy(32);
plasmoip->thisphase += beatsin8(6,-4,4); // You can change direction and speed individually. plasmoip->thisphase += beatsin8(6,-4,4); // You can change direction and speed individually.
plasmoip->thatphase += beatsin8(7,-4,4); // Two phase values to make a complex pattern. By Andrew Tuline. plasmoip->thatphase += beatsin8(7,-4,4); // Two phase values to make a complex pattern. By Andrew Tuline.
@ -6664,7 +6664,8 @@ uint16_t mode_blurz(void) { // Blurz. By Andrew Tuline.
SEGENV.aux0 = 0; SEGENV.aux0 = 0;
} }
SEGMENT.fade_out(SEGMENT.speed); int fadeoutDelay = (256 - SEGMENT.speed) / 32;
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(SEGMENT.speed);
SEGENV.step += FRAMETIME; SEGENV.step += FRAMETIME;
if (SEGENV.step > SPEED_FORMULA_L) { if (SEGENV.step > SPEED_FORMULA_L) {
@ -6732,7 +6733,9 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN.
float my_magnitude = *(float*) um_data->u_data[5] / 4.0f; float my_magnitude = *(float*) um_data->u_data[5] / 4.0f;
if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception) if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
SEGMENT.fade_out(SEGMENT.speed); if (SEGENV.call == 0) SEGMENT.fill(BLACK);
int fadeoutDelay = (256 - SEGMENT.speed) / 32;
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(SEGMENT.speed);
int locn = (log10f((float)FFT_MajorPeak) - 1.78f) * (float)SEGLEN/(MAX_FREQ_LOG10 - 1.78f); // log10 frequency range is from 1.78 to 3.71. Let's scale to SEGLEN. int locn = (log10f((float)FFT_MajorPeak) - 1.78f) * (float)SEGLEN/(MAX_FREQ_LOG10 - 1.78f); // log10 frequency range is from 1.78 to 3.71. Let's scale to SEGLEN.
if (locn < 1) locn = 0; // avoid underflow if (locn < 1) locn = 0; // avoid underflow
@ -6747,7 +6750,7 @@ uint16_t mode_freqmap(void) { // Map FFT_MajorPeak to SEGLEN.
return FRAMETIME; return FRAMETIME;
} // mode_freqmap() } // mode_freqmap()
static const char _data_FX_MODE_FREQMAP[] PROGMEM = "Freqmap@Fade rate,Starting color;,!;!;mp12=2,ssim=0,1d,fr"; // Circle, Beatsin static const char _data_FX_MODE_FREQMAP[] PROGMEM = "Freqmap@Fade rate,Starting color;,!;!;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin
/////////////////////// ///////////////////////
@ -6802,7 +6805,7 @@ uint16_t mode_freqmatrix(void) { // Freqmatrix. By Andreas Plesch
return FRAMETIME; return FRAMETIME;
} // mode_freqmatrix() } // mode_freqmatrix()
static const char _data_FX_MODE_FREQMATRIX[] PROGMEM = "Freqmatrix@Time delay,Sound effect,Low bin,High bin,Sensivity;;;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin static const char _data_FX_MODE_FREQMATRIX[] PROGMEM = "Freqmatrix@Time delay,Sound effect,Low bin,High bin,Sensivity;;;mp12=3,ssim=0,1d,fr"; // Corner, Beatsin
////////////////////// //////////////////////
@ -6823,7 +6826,10 @@ uint16_t mode_freqpixels(void) { // Freqpixel. By Andrew Tuline.
if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception) if (FFT_MajorPeak < 1) FFT_MajorPeak = 1; // log10(0) is "forbidden" (throws exception)
uint16_t fadeRate = 2*SEGMENT.speed - SEGMENT.speed*SEGMENT.speed/255; // Get to 255 as quick as you can. uint16_t fadeRate = 2*SEGMENT.speed - SEGMENT.speed*SEGMENT.speed/255; // Get to 255 as quick as you can.
SEGMENT.fade_out(fadeRate);
if (SEGENV.call == 0) SEGMENT.fill(BLACK);
int fadeoutDelay = (256 - SEGMENT.speed) / 64;
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fade_out(fadeRate);
for (int i=0; i < SEGMENT.intensity/32+1; i++) { for (int i=0; i < SEGMENT.intensity/32+1; i++) {
uint16_t locn = random16(0,SEGLEN); uint16_t locn = random16(0,SEGLEN);
@ -6955,7 +6961,7 @@ uint16_t mode_gravfreq(void) { // Gravfreq. By Andrew Tuline.
return FRAMETIME; return FRAMETIME;
} // mode_gravfreq() } // mode_gravfreq()
static const char _data_FX_MODE_GRAVFREQ[] PROGMEM = "Gravfreq@Rate of fall,Sensivity;,!;!;ix=128,mp12=2,ssim=0,1d,fr"; // Circle, Beatsin static const char _data_FX_MODE_GRAVFREQ[] PROGMEM = "Gravfreq@Rate of fall,Sensivity;,!;!;ix=128,mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin
////////////////////// //////////////////////
@ -6969,7 +6975,10 @@ uint16_t mode_noisemove(void) { // Noisemove. By: Andrew Tuli
} }
uint8_t *fftResult = (uint8_t*)um_data->u_data[2]; uint8_t *fftResult = (uint8_t*)um_data->u_data[2];
SEGMENT.fade_out(224); // Just in case something doesn't get faded. if (SEGENV.call == 0) SEGMENT.fill(BLACK);
//SEGMENT.fade_out(224); // Just in case something doesn't get faded.
int fadeoutDelay = (256 - SEGMENT.speed) / 96;
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fadeToBlackBy(4+ SEGMENT.speed/4);
uint8_t numBins = map(SEGMENT.intensity,0,255,0,16); // Map slider to fftResult bins. uint8_t numBins = map(SEGMENT.intensity,0,255,0,16); // Map slider to fftResult bins.
for (int i=0; i<numBins; i++) { // How many active bins are we using. for (int i=0; i<numBins; i++) { // How many active bins are we using.
@ -6995,13 +7004,16 @@ uint16_t mode_rocktaves(void) { // Rocktaves. Same note from eac
float FFT_MajorPeak = *(float*) um_data->u_data[4]; float FFT_MajorPeak = *(float*) um_data->u_data[4];
float my_magnitude = *(float*) um_data->u_data[5] / 16.0f; float my_magnitude = *(float*) um_data->u_data[5] / 16.0f;
SEGMENT.fadeToBlackBy(64); // Just in case something doesn't get faded. if (SEGENV.call == 0) SEGMENT.fill(BLACK);
SEGMENT.fadeToBlackBy(16); // Just in case something doesn't get faded.
float frTemp = FFT_MajorPeak; float frTemp = FFT_MajorPeak;
uint8_t octCount = 0; // Octave counter. uint8_t octCount = 0; // Octave counter.
uint8_t volTemp = 0; uint8_t volTemp = 0;
if (my_magnitude > 32) volTemp = 255; // We need to squelch out the background noise. volTemp = 32.0f + my_magnitude * 1.5f; // brightness = volume (overflows are handled in next lines)
if (my_magnitude < 48) volTemp = 0; // We need to squelch out the background noise.
if (my_magnitude > 144) volTemp = 255; // everything above this is full brightness
while ( frTemp > 249 ) { while ( frTemp > 249 ) {
octCount++; // This should go up to 5. octCount++; // This should go up to 5.
@ -7017,7 +7029,7 @@ uint16_t mode_rocktaves(void) { // Rocktaves. Same note from eac
return FRAMETIME; return FRAMETIME;
} // mode_rocktaves() } // mode_rocktaves()
static const char _data_FX_MODE_ROCKTAVES[] PROGMEM = "Rocktaves@;,!;!;mp12=0,ssim=0,1d,fr"; // Pixels, Beatsin static const char _data_FX_MODE_ROCKTAVES[] PROGMEM = "Rocktaves@;,!;!;mp12=1,ssim=0,1d,fr"; // Bar, Beatsin
/////////////////////// ///////////////////////
@ -7101,10 +7113,13 @@ uint16_t mode_2DGEQ(void) { // By Will Tatam. Code reduction by Ewoud Wijma.
rippleTime = true; rippleTime = true;
} }
SEGMENT.fadeToBlackBy(SEGMENT.speed); if (SEGENV.call == 0) SEGMENT.fill(BLACK);
int fadeoutDelay = (256 - SEGMENT.speed) / 64;
if ((fadeoutDelay <= 1 ) || ((SEGENV.call % fadeoutDelay) == 0)) SEGMENT.fadeToBlackBy(SEGMENT.speed);
for (int x=0; x < cols; x++) { for (int x=0; x < cols; x++) {
uint8_t band = map(x, 0, cols-1, 0, NUM_BANDS - 1); uint8_t band = map(x, 0, cols-1, 0, NUM_BANDS - 1);
if (NUM_BANDS < 16) band = map(band, 0, NUM_BANDS - 1, 0, 15); // always use full range. comment out this line to get the previous behaviour.
band = constrain(band, 0, 15); band = constrain(band, 0, 15);
uint16_t colorIndex = band * 17; uint16_t colorIndex = band * 17;
uint16_t barHeight = map(fftResult[band], 0, 255, 0, rows); // do not subtract -1 from rows here uint16_t barHeight = map(fftResult[band], 0, 255, 0, rows); // do not subtract -1 from rows here