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Sync bug fixes.

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Analog input fix.
Code cleanup.
This commit is contained in:
Blaz Kristan 2022-06-16 19:20:04 +02:00
parent 12a94c50b8
commit cf54115077
2 changed files with 81 additions and 113 deletions
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192
usermods/audioreactive/audio_reactive.h
View File

@ -7,10 +7,6 @@
#error This audio reactive usermod does not support the ESP8266.
#endif
//The SCL and SDA pins are defined here.
#define HW_PIN_SCL 22
#define HW_PIN_SDA 21
/*
* Usermods allow you to add own functionality to WLED more easily
* See: https://github.com/Aircoookie/WLED/wiki/Add-own-functionality
@ -170,18 +166,19 @@ void FFTcode(void * parameter) {
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 < samplesFFT; i++) {
for (int i=0; i < halfSamplesFFT; 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 (i < halfSamplesFFT) {
if (fabsf((float)vReal[i]) > maxSample1) maxSample1 = fabsf((float)vReal[i]);
} else {
if (fabsf((float)vReal[i]) > maxSample2) maxSample2 = fabsf((float)vReal[i]);
}
}
if (fabsf((float)vReal[i]) > maxSample1) maxSample1 = 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
micDataSm = (uint16_t)maxSample1;
@ -315,11 +312,11 @@ void FFTcode(void * parameter) {
#ifdef SR_DEBUG
// Looking for fftResultMax for each bin using Pink Noise
// for (int i=0; i<16; i++) {
// fftResultMax[i] = ((fftResultMax[i] * 63.0) + fftResult[i]) / 64.0;
// Serial.print(fftResultMax[i]*fftResultPink[i]); Serial.print("\t");
// }
// Serial.println();
for (int i=0; i<16; i++) {
fftResultMax[i] = ((fftResultMax[i] * 63.0) + fftResult[i]) / 64.0;
DEBUGSR_PRINT(fftResultMax[i]*fftResultPink[i]); DEBUGSR_PRINT("\t");
}
DEBUGSR_PRINTLN();
#endif
} // for(;;)
} // FFTcode()
@ -359,7 +356,7 @@ class AudioReactive : public Usermod {
#else
int8_t sdaPin = ES7243_SDAPIN;
#endif
#ifndef ES7243_SDAPIN
#ifndef ES7243_SCLPIN
int8_t sclPin = -1;
#else
int8_t sclPin = ES7243_SCLPIN;
@ -372,7 +369,7 @@ class AudioReactive : public Usermod {
#define UDP_SYNC_HEADER "00001"
struct audioSyncPacket {
char header[6] = UDP_SYNC_HEADER;
char header[6];
uint8_t myVals[32]; // 32 Bytes
int sampleAgc; // 04 Bytes
int sample; // 04 Bytes
@ -414,8 +411,11 @@ class AudioReactive : public Usermod {
bool udpSyncConnected = false;
// 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
// strings to reduce flash memory usage (used more than twice)
static const char _name[];
@ -424,7 +424,7 @@ class AudioReactive : public Usermod {
// private methods
bool isValidUdpSyncVersion(char header[6]) {
bool isValidUdpSyncVersion(const char *header) {
return strncmp(header, UDP_SYNC_HEADER, 6) == 0;
}
@ -524,14 +524,12 @@ class AudioReactive : public Usermod {
*/
void agcAvg() {
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
static int last_soundAgc = -1;
float lastMultAgc = multAgc; // last muliplier used
float multAgcTemp = multAgc; // new multiplier
float tmpAgc = sampleReal * multAgc; // what-if amplified signal
float control_error; // "control error" input for PI control
static float control_integrated = 0.0f; // "integrator control" = accumulated error
if (last_soundAgc != soundAgc)
control_integrated = 0.0f; // new preset - reset integrator
@ -545,7 +543,7 @@ 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)
//multAgcTemp = multAgc; // keep old control value (no change)
tmpAgc = 0;
// we need to "spin down" the intgrated error buffer
if (fabs(control_integrated) < 0.01f) control_integrated = 0.0f;
@ -617,27 +615,23 @@ class AudioReactive : public Usermod {
micDataReal = micIn;
#else
micIn = micDataSm; // micDataSm = ((micData * 3) + micData)/4;
/*---------DEBUG---------*/
DEBUGSR_PRINT("micIn:\tmicData:\tmicIn>>2:\tmic_In_abs:\tsample:\tsampleAdj:\tsampleAvg:\n");
DEBUGSR_PRINT(micIn); DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(micData);
/*-------END DEBUG-------*/
// 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.
/*---------DEBUG---------*/
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(micIn);
/*-------END DEBUG-------*/
// 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);
#endif
// Note to self: the next line kills 80% of sample - "miclev" filter runs at "full arduino loop" speed, following the signal almost instantly!
//micLev = ((micLev * 31) + micIn) / 32; // Smooth it out over the last 32 samples for automatic centering
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
/*---------DEBUG---------*/
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(micIn);
/*-------END DEBUG-------*/
// Using an exponential filter to smooth out the signal. We'll add controls for this in a future release.
// 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);
expAdjF = (expAdjF <= soundSquelch) ? 0: expAdjF; // simple noise gate
@ -645,17 +639,16 @@ class AudioReactive : public Usermod {
expAdjF = fabs(expAdjF); // Now (!) take the absolute value
tmpSample = expAdjF;
/*---------DEBUG---------*/
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(tmpSample);
/*-------END DEBUG-------*/
micIn = abs(micIn); // And get the absolute value of each sample
sampleAdj = tmpSample * sampleGain / 40 * inputLevel/128 + tmpSample / 16; // Adjust the gain. with inputLevel adjustment
// sampleReal = sampleAdj;
//sampleReal = sampleAdj;
sampleReal = tmpSample;
sampleAdj = fmax(fmin(sampleAdj, 255), 0); // Question: why are we limiting the value to 8 bits ???
sample = (int)sampleAdj; // ONLY update sample ONCE!!!!
sample = (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)) {
@ -670,10 +663,8 @@ class AudioReactive : public Usermod {
sampleAvg = ((sampleAvg * 15.0f) + sampleAdj) / 16.0f; // Smooth it out over the last 16 samples.
/*---------DEBUG---------*/
DEBUGSR_PRINT("\t"); DEBUGSR_PRINT(sample);
DEBUGSR_PRINT("\t\t"); DEBUGSR_PRINT(sampleAvg); DEBUGSR_PRINT("\n\n");
/*-------END DEBUG-------*/
// Fixes private class variable compiler error. Unsure if this is the correct way of fixing the root problem. -THATDONFC
uint16_t MinShowDelay = strip.getMinShowDelay();
@ -700,18 +691,20 @@ class AudioReactive : public Usermod {
void transmitAudioData() {
if (!udpSyncConnected) return;
//DEBUGSR_PRINTLN("Transmitting UDP Mic Packet");
audioSyncPacket transmitData;
strncpy(transmitData.header, UDP_SYNC_HEADER, 6);
for (int i = 0; i < 32; i++) {
transmitData.myVals[i] = myVals[i];
}
transmitData.sampleAgc = sampleAgc;
transmitData.sample = sample;
transmitData.sampleAvg = sampleAvg;
transmitData.sampleAgc = sampleAgc;
transmitData.sample = sample;
transmitData.sampleAvg = sampleAvg;
transmitData.samplePeak = udpSamplePeak;
udpSamplePeak = 0; // Reset udpSamplePeak after we've transmitted it
udpSamplePeak = 0; // Reset udpSamplePeak after we've transmitted it
for (int i = 0; i < 16; i++) {
transmitData.fftResult[i] = (uint8_t)constrain(fftResult[i], 0, 254);
@ -727,6 +720,42 @@ class AudioReactive : public Usermod {
} // transmitAudioData()
void receiveAudioData() {
if (!udpSyncConnected) return;
//DEBUGSR_PRINTLN("Checking for UDP Microphone Packet");
size_t packetSize = fftUdp.parsePacket();
if (packetSize) {
//DEBUGSR_PRINTLN("Received UDP Sync Packet");
uint8_t fftBuff[packetSize];
fftUdp.read(fftBuff, packetSize);
// VERIFY THAT THIS IS A COMPATIBLE PACKET
if (packetSize == sizeof(audioSyncPacket) && !(isValidUdpSyncVersion((const char *)fftBuff))) {
audioSyncPacket *receivedPacket = reinterpret_cast<audioSyncPacket*>(fftBuff);
for (int i = 0; i < 32; i++) myVals[i] = receivedPacket->myVals[i];
sampleAgc = receivedPacket->sampleAgc;
rawSampleAgc = receivedPacket->sampleAgc;
sample = receivedPacket->sample;
sampleAvg = receivedPacket->sampleAvg;
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
if (!samplePeak) samplePeak = receivedPacket->samplePeak;
//These values are only available on the ESP32
for (int i = 0; i < 16; i++) fftResult[i] = receivedPacket->fftResult[i];
FFT_Magnitude = receivedPacket->FFT_Magnitude;
FFT_MajorPeak = receivedPacket->FFT_MajorPeak;
//DEBUGSR_PRINTLN("Finished parsing UDP Sync Packet");
}
}
}
public:
//Functions called by WLED
@ -828,23 +857,10 @@ class AudioReactive : public Usermod {
if (audioSource) audioSource->initialize(audioPin);
break;
}
delay(250);
//sampling_period_us = round(1000000*(1.0/SAMPLE_RATE));
delay(250); // give mictophone enough time to initialise
if (enabled) onUpdateBegin(false); // create FFT task
/*
// Define the FFT Task and lock it to core 0
xTaskCreatePinnedToCore(
FFTcode, // Function to implement the task
"FFT", // Name of the task
5000, // Stack size in words
NULL, // Task input parameter
1, // Priority of the task
&FFT_Task, // Task handle
0); // Core where the task should run
*/
initDone = true;
}
@ -885,8 +901,7 @@ class AudioReactive : public Usermod {
if (lastMode != knownMode) { // only execute if mode changes
char lineBuffer[3];
/* uint8_t printedChars = */ extractModeName(knownMode, JSON_mode_names, lineBuffer, 3); //is this 'the' way to get mode name here?
/* 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++) {
@ -942,62 +957,15 @@ class AudioReactive : public Usermod {
#endif
}
if (millis() - lastTime > 20) {
lastTime = millis();
// Begin UDP Microphone Sync
if ((audioSyncEnabled & 0x02) && millis() - lastTime > delayMs) // Only run the audio listener code if we're in Receive mode
receiveAudioData();
if (millis() - lastTime > 20) {
if (audioSyncEnabled & 0x01) { // Only run the transmit code IF we're in Transmit mode
DEBUGSR_PRINTLN("Transmitting UDP Mic Packet");
transmitAudioData();
}
}
// Begin UDP Microphone Sync
if ((audioSyncEnabled & 0x02) && udpSyncConnected) { // Only run the audio listener code if we're in Receive mode
if (millis()-lastTime > delayMs) {
//Serial.println("Checking for UDP Microphone Packet");
int packetSize = fftUdp.parsePacket();
if (packetSize) {
// Serial.println("Received UDP Sync Packet");
uint8_t fftBuff[packetSize];
fftUdp.read(fftBuff, packetSize);
audioSyncPacket receivedPacket;
memcpy(&receivedPacket, fftBuff, packetSize);
for (int i = 0; i < 32; i++ ){
myVals[i] = receivedPacket.myVals[i];
}
sampleAgc = receivedPacket.sampleAgc;
rawSampleAgc = receivedPacket.sampleAgc;
sample = receivedPacket.sample;
sampleAvg = receivedPacket.sampleAvg;
// VERIFY THAT THIS IS A COMPATIBLE PACKET
char packetHeader[6];
memcpy(&receivedPacket, packetHeader, 6);
if (!(isValidUdpSyncVersion(packetHeader))) {
memcpy(&receivedPacket, fftBuff, packetSize);
for (int i = 0; i < 32; i++ ){
myVals[i] = receivedPacket.myVals[i];
}
sampleAgc = receivedPacket.sampleAgc;
rawSampleAgc = receivedPacket.sampleAgc;
sample = receivedPacket.sample;
sampleAvg = receivedPacket.sampleAvg;
// Only change samplePeak IF it's currently false.
// If it's true already, then the animation still needs to respond.
if (!samplePeak) {
samplePeak = receivedPacket.samplePeak;
}
//These values are only available on the ESP32
for (int i = 0; i < 16; i++) {
fftResult[i] = receivedPacket.fftResult[i];
}
FFT_Magnitude = receivedPacket.FFT_Magnitude;
FFT_MajorPeak = receivedPacket.FFT_MajorPeak;
//Serial.println("Finished parsing UDP Sync Packet");
}
}
}
lastTime = millis();
}
}

2
usermods/audioreactive/audio_source.h
View File

@ -156,7 +156,7 @@ class I2SSource : public AudioSource {
if (_mclkPin != I2S_PIN_NO_CHANGE) pinManager.deallocatePin(_mclkPin, PinOwner::UM_Audioreactive);
}
void getSamples(double *buffer, uint16_t num_samples) {
virtual void getSamples(double *buffer, uint16_t num_samples) {
if (_initialized) {
esp_err_t err;
size_t bytes_read = 0; /* Counter variable to check if we actually got enough data */