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Merge branch 'main' into 0_15

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This commit is contained in:
Blaz Kristan
2024-02-06 14:47:20 +01:00
parent 0ab2d18b52
commit 7eae8f68d8
83 changed files with 499 additions and 222 deletions
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32
usermods/audioreactive/audio_reactive.h
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@@ -83,7 +83,7 @@ static bool limiterOn = true;
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
static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized square root
//
// AGC presets
@@ -118,9 +118,9 @@ static float sampleAgc = 0.0f; // Smoothed AGC sample
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 = 31; // 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 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 void detectSamplePeak(void); // peak detection function (needs scaled FFT reasults in vReal[])
static void detectSamplePeak(void); // peak detection function (needs scaled FFT results in vReal[])
static void autoResetPeak(void); // peak auto-reset function
@@ -212,7 +212,7 @@ 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
static float fftAddAvg(int from, int to) {
float result = 0.0f;
for (int i = from; i <= to; i++) {
@@ -330,7 +330,7 @@ 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
@@ -389,7 +389,7 @@ void FFTcode(void * parameter)
}
}
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
// post-processing of frequency channels (pink noise adjustment, AGC, smoothing, scaling)
postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS);
#if defined(WLED_DEBUG) || defined(SR_DEBUG)
@@ -436,7 +436,7 @@ 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;
@@ -640,7 +640,7 @@ 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.
@@ -763,13 +763,13 @@ 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
@@ -809,13 +809,13 @@ 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
// 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 energy zone
multAgcTemp = lastMultAgc + agcFollowFast[AGC_preset] * agcControlKp[AGC_preset] * control_error;
multAgcTemp += agcFollowFast[AGC_preset] * agcControlKi[AGC_preset] * control_integrated;
} else { // "normal zone"
@@ -823,7 +823,7 @@ 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;
@@ -853,7 +853,7 @@ 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.
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
@@ -1308,7 +1308,7 @@ class AudioReactive : public Usermod {
// 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
//if ((userloopDelay > 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
@@ -1525,7 +1525,7 @@ 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
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@@ -44,7 +44,7 @@
// benefit: analog mic inputs will be sampled contiously -> better response times and less "glitches"
// WARNING: this option WILL lock-up your device in case that any other analogRead() operation is performed;
// for example if you want to read "analog buttons"
//#define I2S_GRAB_ADC1_COMPLETELY // (experimental) continously sample analog ADC microphone. WARNING will cause analogRead() lock-up
//#define I2S_GRAB_ADC1_COMPLETELY // (experimental) continuously sample analog ADC microphone. WARNING will cause analogRead() lock-up
// data type requested from the I2S driver - currently we always use 32bit
//#define I2S_USE_16BIT_SAMPLES // (experimental) define this to request 16bit - more efficient but possibly less compatible
@@ -378,7 +378,7 @@ 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 {
@@ -429,8 +429,8 @@ 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 {
@@ -475,7 +475,7 @@ 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.
//

8
usermods/audioreactive/readme.md
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@@ -1,6 +1,6 @@
# Audioreactive usermod
Enabless controlling LEDs via audio input. Audio source can be a microphone or analog-in (AUX) using an appropriate adapter.
Enables controlling LEDs via audio input. Audio source can be a microphone or analog-in (AUX) using an appropriate adapter.
Supported microphones range from analog (MAX4466, MAX9814, ...) to digital (INMP441, ICS-43434, ...).
Does audio processing and provides data structure that specially written effects can use.
@@ -19,7 +19,7 @@ This usermod is an evolution of [SR-WLED](https://github.com/atuline/WLED), and
## Supported MCUs
This audioreactive usermod works best on "classic ESP32" (dual core), and on ESP32-S3 which also has dual core and hardware floating point support.
It will compile succesfully for ESP32-S2 and ESP32-C3, however might not work well, as other WLED functions will become slow. Audio processing requires a lot of computing power, which can be problematic on smaller MCUs like -S2 and -C3.
It will compile successfully for ESP32-S2 and ESP32-C3, however might not work well, as other WLED functions will become slow. Audio processing requires a lot of computing power, which can be problematic on smaller MCUs like -S2 and -C3.
Analog audio is only possible on "classic" ESP32, but not on other MCUs like ESP32-S3.
@@ -35,7 +35,7 @@ Customised _arduinoFFT_ library for use with this usermod can be found at https:
### using latest (develop) _arduinoFFT_ library
Alternatively, you can use the latest arduinoFFT development version.
ArduinoFFT `develop` library is slightly more accurate, and slighly faster than our customised library, however also needs additional 2kB RAM.
ArduinoFFT `develop` library is slightly more accurate, and slightly faster than our customised library, however also needs additional 2kB RAM.
* `build_flags` = `-D USERMOD_AUDIOREACTIVE` `-D UM_AUDIOREACTIVE_USE_NEW_FFT`
* `lib_deps`= `https://github.com/kosme/arduinoFFT#develop @ 1.9.2`
@@ -68,7 +68,7 @@ You can use the following additional flags in your `build_flags`
* `-D SR_GAIN=x` : Default "gain" setting (60)
* `-D I2S_USE_RIGHT_CHANNEL`: Use RIGHT instead of LEFT channel (not recommended unless you strictly need this).
* `-D I2S_USE_16BIT_SAMPLES`: Use 16bit instead of 32bit for internal sample buffers. Reduces sampling quality, but frees some RAM ressources (not recommended unless you absolutely need this).
* `-D I2S_GRAB_ADC1_COMPLETELY`: Experimental: continously sample analog ADC microphone. Only effective on ESP32. WARNING this _will_ cause conflicts(lock-up) with any analogRead() call.
* `-D I2S_GRAB_ADC1_COMPLETELY`: Experimental: continuously sample analog ADC microphone. Only effective on ESP32. WARNING this _will_ cause conflicts(lock-up) with any analogRead() call.
* `-D MIC_LOGGER` : (debugging) Logs samples from the microphone to serial USB. Use with serial plotter (Arduino IDE)
* `-D SR_DEBUG` : (debugging) Additional error diagnostics and debug info on serial USB.