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audioreactive - spelling corrections in comments

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Frank
2023-10-08 01:36:08 +02:00
parent 28b46d8ee3
commit 1fa5dc8ea8
2 changed files with 38 additions and 38 deletions
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66
usermods/audioreactive/audio_reactive.h
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@@ -21,13 +21,13 @@
* ....
*/
#define FFT_PREFER_EXACT_PEAKS // use different FFT wndowing -> results in "sharper" peaks and less "leaking" into other frequencies
#define FFT_PREFER_EXACT_PEAKS // use different FFT windowing -> results in "sharper" peaks and less "leaking" into other frequencies
//#define SR_STATS
#if !defined(FFTTASK_PRIORITY)
#define FFTTASK_PRIORITY 1 // standard: looptask prio
//#define FFTTASK_PRIORITY 2 // above looptask, below asyc_tcp
//#define FFTTASK_PRIORITY 4 // above asyc_tcp
//#define FFTTASK_PRIORITY 2 // above looptask, below async_tcp
//#define FFTTASK_PRIORITY 4 // above async_tcp
#endif
#if !defined(CONFIG_IDF_TARGET_ESP32S2) && !defined(CONFIG_IDF_TARGET_ESP32C3)
@@ -109,7 +109,7 @@ static float volumeSmth = 0.0f; // either sampleAvg or sampleAgc
static float FFT_MajorPeak = 1.0f; // FFT: strongest (peak) frequency
static float FFT_Magnitude = 0.0f; // FFT: volume (magnitude) of peak frequency
static bool samplePeak = false; // Boolean flag for peak - used in effects. Responding routine may reset this flag. Auto-reset after strip.getMinShowDelay()
static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same tiem as samplePeak, but reset by transmitAudioData
static bool udpSamplePeak = false; // Boolean flag for peak. Set at the same time as samplePeak, but reset by transmitAudioData
static unsigned long timeOfPeak = 0; // time of last sample peak detection.
static uint8_t fftResult[NUM_GEQ_CHANNELS]= {0};// Our calculated freq. channel result table to be used by effects
@@ -123,7 +123,7 @@ static uint16_t decayTime = 300; // int: decay time in milliseconds
// peak detection
#ifdef ARDUINO_ARCH_ESP32
static void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[]) - no used for 8266 receive-only mode
static void detectSamplePeak(void); // peak detection function (needs scaled FFT results in vReal[]) - no used for 8266 receive-only mode
#endif
static void autoResetPeak(void); // peak auto-reset function
static uint8_t maxVol = 31; // (was 10) Reasonable value for constant volume for 'peak detector', as it won't always trigger (deprecated)
@@ -150,7 +150,7 @@ static uint8_t inputLevel = 128; // UI slider value
#endif
// user settable options for FFTResult scaling
static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized sqare root
static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized square root
#ifndef SR_FREQ_PROF
static uint8_t pinkIndex = 0; // 0: default; 1: line-in; 2: IMNP441
#else
@@ -220,7 +220,7 @@ static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels); //
static TaskHandle_t FFT_Task = nullptr;
// Table of multiplication factors so that we can even out the frequency response.
#define MAX_PINK 10 // 0 = standard, 1= line-in (pink moise only), 2..4 = IMNP441, 5..6 = ICS-43434, ,7=SPM1423, 8..9 = userdef, 10= flat (no pink noise adjustment)
#define MAX_PINK 10 // 0 = standard, 1= line-in (pink noise only), 2..4 = IMNP441, 5..6 = ICS-43434, ,7=SPM1423, 8..9 = userdef, 10= flat (no pink noise adjustment)
static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = {
{ 1.70f, 1.71f, 1.73f, 1.78f, 1.68f, 1.56f, 1.55f, 1.63f, 1.79f, 1.62f, 1.80f, 2.06f, 2.47f, 3.35f, 6.83f, 9.55f }, // 0 default from SR WLED
// { 1.30f, 1.32f, 1.40f, 1.46f, 1.52f, 1.57f, 1.68f, 1.80f, 1.89f, 2.00f, 2.11f, 2.21f, 2.30f, 2.39f, 3.09f, 4.34f }, // - Line-In Generic -> pink noise adjustment only
@@ -244,7 +244,7 @@ static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = {
/* how to make your own profile:
* ===============================
* preparation: make sure your microphone has direct line-of-sigh with the speaker, 1-2meter distance is best
* Prepare your HiFi equipment: disable all "Sound enhancements" - like Loudness, Equalizer, Bass Boost. Bass/Trebble controls set to middle.
* Prepare your HiFi equipment: disable all "Sound enhancements" - like Loudness, Equalizer, Bass Boost. Bass/Treble controls set to middle.
* Your HiFi equipment should receive its audio input from Line-In, SPDIF, HDMI, or another "undistorted" connection (like CDROM).
* Try not to use Bluetooth or MP3 when playing the "pink noise" audio. BT-audio and MP3 both perform "acoustic adjustments" that we don't want now.
@@ -258,7 +258,7 @@ static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = {
* Your own profile:
* - Target for each LED bar is 50% to 75% of the max height --> 8(high) x 16(wide) panel means target = 5. 32 x 16 means target = 22.
* - From left to right - count the LEDs in each of the 16 frequency colums (that's why you need the photo). This is the barheight for each channel.
* - From left to right - count the LEDs in each of the 16 frequency columns (that's why you need the photo). This is the barheight for each channel.
* - math time! Find the multiplier that will bring each bar to to target.
* * in case of square root scale: multiplier = (target * target) / (barheight * barheight)
* * in case of linear scale: multiplier = target / barheight
@@ -360,7 +360,7 @@ static float mapf(float x, float in_min, float in_max, float out_min, float out_
return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
}
// compute average of several FFT resut bins
// compute average of several FFT result bins
// linear average
static float fftAddAvgLin(int from, int to) {
float result = 0.0f;
@@ -430,7 +430,7 @@ void FFTcode(void * parameter)
for(uint_fast16_t binInd = 0; binInd < samplesFFT; binInd++) {
float binFreq = binInd * binWidth + binWidth/2.0f;
if (binFreq > (SAMPLE_RATE * 0.42f))
binFreq = (SAMPLE_RATE * 0.42f) - 0.25 * (binFreq - (SAMPLE_RATE * 0.42f)); // supress noise and aliasing
binFreq = (SAMPLE_RATE * 0.42f) - 0.25 * (binFreq - (SAMPLE_RATE * 0.42f)); // suppress noise and aliasing
pinkFactors[binInd] = sqrtf(binFreq) / pinkcenter;
}
pinkFactors[0] *= 0.5; // suppress 0-42hz bin
@@ -482,7 +482,7 @@ void FFTcode(void * parameter)
#endif
xLastWakeTime = xTaskGetTickCount(); // update "last unblocked time" for vTaskDelay
isFirstRun = !isFirstRun; // toggle throtte
isFirstRun = !isFirstRun; // toggle throttle
#ifdef MIC_LOGGER
float datMin = 0.0f;
@@ -619,7 +619,7 @@ void FFTcode(void * parameter)
*
* 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.
* Now, Take the 60Hz and multiply by 1.320367784 to get the next frequency and so on until the end. Then determine the bins.
* End frequency = Start frequency * multiplier ^ 16
* Multiplier = (End frequency/ Start frequency) ^ 1/16
* Multiplier = 1.320367784
@@ -673,7 +673,7 @@ void FFTcode(void * parameter)
fftCalc[13] = fftAddAvg(86,103); // 18 3704 - 4479 high mid
fftCalc[14] = fftAddAvg(104,164) * 0.88f; // 61 4479 - 7106 high mid + high -- with slight damping
}
else if (freqDist == 1) { //WLEDMM: Rightshft: note ewowi: frequencies in comments are not correct
else if (freqDist == 1) { //WLEDMM: Rightshift: note ewowi: frequencies in comments are not correct
if (useInputFilter==1) {
// skip frequencies below 100hz
fftCalc[ 0] = 0.8f * fftAddAvg(1,1);
@@ -717,7 +717,7 @@ void FFTcode(void * parameter)
memcpy(fftCalc, lastFftCalc, sizeof(fftCalc)); // restore last "good" channels
}
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
// post-processing of frequency channels (pink noise adjustment, AGC, smoothing, scaling)
if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK;
//postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS);
postProcessFFTResults((fabsf(volumeSmth)>0.25f)? true : false , NUM_GEQ_CHANNELS); // this function modifies fftCalc, fftAvg and fftResult
@@ -775,7 +775,7 @@ static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // p
// FIR lowpass, to remove high frequency noise
float highFilteredSample;
if (i < (numSamples-1)) highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*sampleBuffer[i+1]; // smooth out spikes
else highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*last_vals[1]; // spcial handling for last sample in array
else highFilteredSample = beta1*sampleBuffer[i] + beta2*last_vals[0] + beta2*last_vals[1]; // special handling for last sample in array
last_vals[1] = last_vals[0];
last_vals[0] = sampleBuffer[i];
sampleBuffer[i] = highFilteredSample;
@@ -1010,7 +1010,7 @@ class AudioReactive : public Usermod {
// variables used by getSample() and agcAvg()
int16_t micIn = 0; // Current sample starts with negative values and large values, which is why it's 16 bit signed
double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controler.
double sampleMax = 0.0; // Max sample over a few seconds. Needed for AGC controller.
double micLev = 0.0; // 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.
@@ -1048,7 +1048,7 @@ class AudioReactive : public Usermod {
////////////////////
void logAudio()
{
if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & 0x02) == 0))) return; // no audio availeable
if (disableSoundProcessing && (!udpSyncConnected || ((audioSyncEnabled & 0x02) == 0))) return; // no audio available
#ifdef MIC_LOGGER
// Debugging functions for audio input and sound processing. Comment out the values you want to see
PLOT_PRINT("volumeSmth:"); PLOT_PRINT(volumeSmth + 256.0f); PLOT_PRINT("\t"); // +256 to move above other lines
@@ -1141,13 +1141,13 @@ class AudioReactive : public Usermod {
* 2. we use two setpoints, one at ~60%, and one at ~80% of the maximum signal
* 3. the amplification depends on signal level:
* a) normal zone - very slow adjustment
* b) emergency zome (<10% or >90%) - very fast adjustment
* b) emergency zone (<10% or >90%) - very fast adjustment
*/
void agcAvg(unsigned long the_time)
{
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
float lastMultAgc = multAgc; // last muliplier used
float lastMultAgc = multAgc; // last multiplier used
float multAgcTemp = multAgc; // new multiplier
float tmpAgc = sampleReal * multAgc; // what-if amplified signal
@@ -1187,13 +1187,13 @@ class AudioReactive : public Usermod {
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
control_integrated += control_error * 0.002 * 0.25; // 2ms = integration time; 0.25 for damping
else
control_integrated *= 0.9; // spin down that beasty integrator
control_integrated *= 0.9; // spin down that integrator beast
// 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
if ((tmpAgc > agcZoneHigh[AGC_preset]) || (tmpAgc < soundSquelch + agcZoneLow[AGC_preset])) { // upper/lower emergency zone
multAgcTemp = lastMultAgc + agcFollowFast[AGC_preset] * agcControlKp[AGC_preset] * control_error;
multAgcTemp += agcFollowFast[AGC_preset] * agcControlKi[AGC_preset] * control_integrated;
} else { // "normal zone"
@@ -1201,7 +1201,7 @@ class AudioReactive : public Usermod {
multAgcTemp += agcFollowSlow[AGC_preset] * agcControlKi[AGC_preset] * control_integrated;
}
// limit amplification again - PI controler sometimes "overshoots"
// limit amplification again - PI controller sometimes "overshoots"
//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;
@@ -1231,7 +1231,7 @@ class AudioReactive : public Usermod {
void getSample()
{
float sampleAdj; // Gain adjusted sample value
float tmpSample; // An interim sample variable used for calculatioins.
float tmpSample; // An interim sample variable used for calculations.
#ifdef WLEDMM_FASTPATH
constexpr float weighting = 0.35f; // slightly reduced filter strength, to reduce audio latency
constexpr float weighting2 = 0.25f;
@@ -1407,7 +1407,7 @@ class AudioReactive : public Usermod {
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
delta_time = constrain(delta_time , 1, 1000); // below 1ms -> 1ms; above 1sec -> silly lil hick-up
float deltaSample = volumeSmth - last_volumeSmth;
if (attackTime > 0) { // user has defined attack time > 0
@@ -1443,11 +1443,11 @@ class AudioReactive : public Usermod {
receivedFormat = 0;
DEBUGSR_PRINTLN(F("AR connectUDPSoundSync(): connection lost, UDP closed."));
}
return; // neither AP nor other connections availeable
return; // neither AP nor other connections available
}
if (udpSyncConnected) return; // already connected
if (millis() - last_connection_attempt < 15000) return; // only try once in 15 seconds
if (updateIsRunning) return; // don't reconect during OTA
if (updateIsRunning) return; // don't reconnect during OTA
// if we arrive here, we need a UDP connection but don't have one
last_connection_attempt = millis();
@@ -1641,7 +1641,7 @@ class AudioReactive : public Usermod {
um_data->u_type[10] = UMT_FLOAT;
#else
// ESP8266
// See https://github.com/MoonModules/WLED/pull/60#issuecomment-1666972133 for explaination of these alternative sources of data
// See https://github.com/MoonModules/WLED/pull/60#issuecomment-1666972133 for explanation of these alternative sources of data
um_data->u_data[6] = &maxVol; // assigned in effect function from UI element!!! (Puddlepeak, Ripplepeak, Waterfall)
um_data->u_type[6] = UMT_BYTE;
@@ -1758,7 +1758,7 @@ class AudioReactive : public Usermod {
if (!audioSource) enabled = false; // audio failed to initialise
#endif
if (enabled) onUpdateBegin(false); // create FFT task, and initailize network
if (enabled) onUpdateBegin(false); // create FFT task, and initialize network
#ifdef ARDUINO_ARCH_ESP32
if (FFT_Task == nullptr) enabled = false; // FFT task creation failed
@@ -1897,9 +1897,9 @@ class AudioReactive : public Usermod {
#if defined(SR_DEBUG)
// complain when audio userloop has been delayed for long time. Currently we need userloop running between 500 and 1500 times per second.
// softhack007 disabled temporarily - avoid serial console spam with MANY leds and low FPS
// softhack007 disabled temporarily - avoid serial console spam with MANY LEDs and low FPS
//if ((userloopDelay > /*23*/ 65) && !disableSoundProcessing && (audioSyncEnabled == 0)) {
//DEBUG_PRINTF("[AR userLoop] hickup detected -> was inactive for last %d millis!\n", userloopDelay);
//DEBUG_PRINTF("[AR userLoop] hiccup detected -> was inactive for last %d millis!\n", userloopDelay);
//}
#endif
@@ -2207,7 +2207,7 @@ class AudioReactive : public Usermod {
} else {
// Analog or I2S digital input
if (audioSource && (audioSource->isInitialized())) {
// audio source sucessfully configured
// audio source successfully configured
if (audioSource->getType() == AudioSource::Type_I2SAdc) {
infoArr.add(F("ADC analog"));
} else {

10
usermods/audioreactive/audio_source.h
View File

@@ -163,7 +163,7 @@ class AudioSource {
SRate_t _sampleRate; // Microphone sampling rate
int _blockSize; // I2S block size
bool _initialized; // Gets set to true if initialization is successful
bool _i2sMaster; // when false, ESP32 will be in I2S SLAVE mode (for devices that only operate in MASTER mode). Only workds in newer IDF >= 4.4.x
bool _i2sMaster; // when false, ESP32 will be in I2S SLAVE mode (for devices that only operate in MASTER mode). Only works in newer IDF >= 4.4.x
float _sampleScale; // pre-scaling factor for I2S samples
I2S_datatype newSampleBuffer[I2S_SAMPLES_MAX+4] = { 0 }; // global buffer for i2s_read
};
@@ -418,7 +418,7 @@ class I2SSource : public AudioSource {
};
/* ES7243 Microphone
This is an I2S microphone that requires ininitialization over
This is an I2S microphone that requires initialization over
I2C before I2S data can be received
*/
class ES7243 : public I2SSource {
@@ -483,8 +483,8 @@ public:
}
};
/* ES8388 Sound Modude
This is an I2S sound processing unit that requires ininitialization over
/* ES8388 Sound Module
This is an I2S sound processing unit that requires initialization over
I2C before I2S data can be received.
*/
class ES8388Source : public I2SSource {
@@ -532,7 +532,7 @@ class ES8388Source : public I2SSource {
// The mics *and* line-in are BOTH connected to LIN2/RIN2 on the AudioKit
// so there's no way to completely eliminate the mics. It's also hella noisy.
// Line-in works OK on the AudioKit, generally speaking, as the mics really need
// amplification to be noticable in a quiet room. If you're in a very loud room,
// amplification to be noticeable in a quiet room. If you're in a very loud room,
// the mics on the AudioKit WILL pick up sound even in line-in mode.
// TL;DR: Don't use the AudioKit for anything, use the LyraT.
//