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Merge branch 'MoonModules:mdev' into ES8388S

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netmindz
2022-11-23 21:43:04 +00:00
committed by GitHub
parent 5be6f748bc 17e6a887cb
commit c80e65e106
1 changed files with 137 additions and 120 deletions
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257
usermods/audioreactive/audio_reactive.h
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@@ -4,7 +4,7 @@
#include <driver/i2s.h>
#include <driver/adc.h>
#ifndef ESP32
#ifndef ARDUINO_ARCH_ESP32
#error This audio reactive usermod does not support the ESP8266.
#endif
@@ -28,43 +28,32 @@
// #define FFT_SAMPLING_LOG // FFT result debugging
// #define SR_DEBUG // generic SR DEBUG messages
#ifdef SR_DEBUG
#define DEBUGSR_PRINT(x) Serial.print(x)
#define DEBUGSR_PRINTLN(x) Serial.println(x)
#define DEBUGSR_PRINTF(x...) Serial.printf(x)
#define DEBUGSR_PRINT(x) DEBUGOUT.print(x)
#define DEBUGSR_PRINTLN(x) DEBUGOUT.println(x)
#define DEBUGSR_PRINTF(x...) DEBUGOUT.printf(x)
#else
#define DEBUGSR_PRINT(x)
#define DEBUGSR_PRINTLN(x)
#define DEBUGSR_PRINTF(x...)
#endif
// use audio source class (ESP32 specific)
#include "audio_source.h"
constexpr i2s_port_t I2S_PORT = I2S_NUM_0;
constexpr int BLOCK_SIZE = 128;
constexpr SRate_t SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz is a standard rate. Physical sample time -> 23ms
//constexpr SRate_t SAMPLE_RATE = 16000; // 16kHz - use if FFTtask takes more than 20ms. Physical sample time -> 32ms
//constexpr SRate_t SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental. Physical sample time -> 25ms
//constexpr SRate_t SAMPLE_RATE = 10240; // Base sample rate in Hz - previous default. Physical sample time -> 50ms
#define FFT_MIN_CYCLE 21 // minimum time before FFT task is repeated. Use with 22Khz sampling
//#define FFT_MIN_CYCLE 30 // Use with 16Khz sampling
//#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated. Use with 20Khz sampling
//#define FFT_MIN_CYCLE 46 // minimum time before FFT task is repeated. Use with 10Khz sampling
constexpr i2s_port_t I2S_PORT = I2S_NUM_0; // I2S port to use (do not change !)
constexpr int BLOCK_SIZE = 128; // I2S buffer size (samples)
// globals
static uint8_t inputLevel = 128; // UI slider value
#ifndef SR_SQUELCH
uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value)
uint8_t soundSquelch = 10; // squelch value for volume reactive routines (config value)
#else
uint8_t soundSquelch = SR_SQUELCH; // squelch value for volume reactive routines (config value)
uint8_t soundSquelch = SR_SQUELCH; // squelch value for volume reactive routines (config value)
#endif
#ifndef SR_GAIN
uint8_t sampleGain = 60; // sample gain (config value)
uint8_t sampleGain = 60; // sample gain (config value)
#else
uint8_t sampleGain = SR_GAIN; // sample gain (config value)
uint8_t sampleGain = SR_GAIN; // sample gain (config value)
#endif
static uint8_t soundAgc = 1; // 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 (config value)
@@ -77,11 +66,12 @@ static uint16_t decayTime = 300; // int: decay time in milliseconds
// user settable options for FFTResult scaling
static uint8_t FFTScalingMode = 3; // 0 none; 1 optimized logarithmic; 2 optimized linear; 3 optimized sqare root
#ifndef SR_FREQ_PROF
static uint8_t pinkIndex = 0; // 0: default; 1: line-in; 2: IMNP441
static uint8_t pinkIndex = 0; // 0: default; 1: line-in; 2: IMNP441
#else
static uint8_t pinkIndex = SR_FREQ_PROF; // 0: default; 1: line-in; 2: IMNP441
#endif
//
// AGC presets
// Note: in C++, "const" implies "static" - no need to explicitly declare everything as "static const"
@@ -130,50 +120,14 @@ static void autoResetPeak(void); // peak auto-reset function
// Begin FFT Code //
////////////////////
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
// 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_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt
#define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 50% on ESP32 (as alternative to FFT_SQRT_APPROXIMATION)
#else
// lib_deps += https://github.com/blazoncek/arduinoFFT.git
#endif
#include <arduinoFFT.h>
// some prototypes, to ensure consistent interfaces
void FFTcode(void * parameter); // audio processing task: read samples, run FFT, fill GEQ channels from FFT results
static void runMicFilter(uint16_t numSamples, float *sampleBuffer); // pre-filtering of raw samples (band-pass)
static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels); // post-processing and post-amp of GEQ channels
// 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
#define NUM_GEQ_CHANNELS 16 // number of frequency channels. Don't change !!
// 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_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
// These are the input and output vectors. Input vectors receive computed results from FFT.
static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
static float vImag[samplesFFT] = {0.0f}; // imaginary parts
// 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 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
#define LOG_256 5.54517744
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static float windowWeighingFactors[samplesFFT] = {0.0f};
#endif
// Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256.
static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f};
static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON)
#ifdef SR_DEBUG
static float fftResultMax[NUM_GEQ_CHANNELS] = {0.0f}; // A table used for testing to determine how our post-processing is working.
#endif
#if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS)
static uint64_t fftTime = 0;
static uint64_t sampleTime = 0;
#endif
static TaskHandle_t FFT_Task = nullptr;
// Table of multiplication factors so that we can even out the frequency response.
#define MAX_PINK 9 // 0 = standard, 1= line-in (pink moise only), 2..4 = IMNP441, 5..6 = ICS-43434, 6..7 = userdef, 9= flat (no pink noise adjustment)
@@ -225,20 +179,71 @@ static const float fftResultPink[MAX_PINK+1][NUM_GEQ_CHANNELS] = {
* Test your new profile (same procedure as above). Iterate the process to improve results.
*/
// globals and FFT Output variables shared with animations
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
#if defined(WLED_DEBUG) || defined(SR_DEBUG) || defined(SR_STATS)
static uint64_t fftTime = 0;
static uint64_t sampleTime = 0;
#endif
// FFT Task variables (filtering and post-processing)
static float fftCalc[NUM_GEQ_CHANNELS] = {0.0f}; // Try and normalize fftBin values to a max of 4096, so that 4096/16 = 256.
static float fftAvg[NUM_GEQ_CHANNELS] = {0.0f}; // Calculated frequency channel results, with smoothing (used if dynamics limiter is ON)
#ifdef SR_DEBUG
static float fftResultMax[NUM_GEQ_CHANNELS] = {0.0f}; // A table used for testing to determine how our post-processing is working.
#endif
// audio source parameters and constant
constexpr SRate_t SAMPLE_RATE = 22050; // Base sample rate in Hz - 22Khz is a standard rate. Physical sample time -> 23ms
//constexpr SRate_t SAMPLE_RATE = 16000; // 16kHz - use if FFTtask takes more than 20ms. Physical sample time -> 32ms
//constexpr SRate_t SAMPLE_RATE = 20480; // Base sample rate in Hz - 20Khz is experimental. Physical sample time -> 25ms
//constexpr SRate_t SAMPLE_RATE = 10240; // Base sample rate in Hz - previous default. Physical sample time -> 50ms
#define FFT_MIN_CYCLE 21 // minimum time before FFT task is repeated. Use with 22Khz sampling
//#define FFT_MIN_CYCLE 30 // Use with 16Khz sampling
//#define FFT_MIN_CYCLE 23 // minimum time before FFT task is repeated. Use with 20Khz sampling
//#define FFT_MIN_CYCLE 46 // minimum time before FFT task is repeated. Use with 10Khz sampling
// 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_2 = 256; // meaningfull part of FFT results - only the "lower half" contains useful information.
// 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 0.46f // downscaling factor for FFT results - for "Flat-Top" window @22Khz, new freq channels
#define LOG_256 5.54517744 // log2(256)
// These are the input and output vectors. Input vectors receive computed results from FFT.
static float vReal[samplesFFT] = {0.0f}; // FFT sample inputs / freq output - these are our raw result bins
static float vImag[samplesFFT] = {0.0f}; // imaginary parts
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static float windowWeighingFactors[samplesFFT] = {0.0f};
#endif
// Create FFT object
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
// 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_SQRT_APPROXIMATION // enables "quake3" style inverse sqrt
#define sqrt(x) sqrtf(x) // little hack that reduces FFT time by 50% on ESP32 (as alternative to FFT_SQRT_APPROXIMATION)
#else
// lib_deps += https://github.com/blazoncek/arduinoFFT.git
#endif
#include <arduinoFFT.h>
#ifdef UM_AUDIOREACTIVE_USE_NEW_FFT
static ArduinoFFT<float> FFT = ArduinoFFT<float>( vReal, vImag, samplesFFT, SAMPLE_RATE, windowWeighingFactors);
#else
static arduinoFFT FFT = arduinoFFT(vReal, vImag, samplesFFT, SAMPLE_RATE);
#endif
static TaskHandle_t FFT_Task = nullptr;
// Helper functions
// float version of map()
static float mapf(float x, float in_min, float in_max, float out_min, float out_max){
return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
}
// compute average of several FFT resut bins
#if 1 // linear average
static float fftAddAvg(int from, int to) {
float result = 0.0f;
@@ -247,7 +252,6 @@ static float fftAddAvg(int from, int to) {
}
return result / float(to - from + 1);
}
#else // RMS average
static float fftAddAvg(int from, int to) {
double result = 0.0;
@@ -258,7 +262,9 @@ static float fftAddAvg(int from, int to) {
}
#endif
//
// FFT main task
//
void FFTcode(void * parameter)
{
DEBUGSR_PRINT("FFT started on core: "); DEBUGSR_PRINTLN(xPortGetCoreID());
@@ -297,35 +303,7 @@ void FFTcode(void * parameter)
// band pass filter - can reduce noise floor by a factor of 50
// downside: frequencies below 100Hz will be ignored
if (useBandPassFilter) {
// low frequency cutoff parameter - see https://dsp.stackexchange.com/questions/40462/exponential-moving-average-cut-off-frequency
//constexpr float alpha = 0.04f; // 150Hz
//constexpr float alpha = 0.03f; // 110Hz
constexpr float alpha = 0.0225f; // 80hz
//constexpr float alpha = 0.01693f;// 60hz
// high frequency cutoff parameter
//constexpr float beta1 = 0.75; // 11Khz
//constexpr float beta1 = 0.82; // 15Khz
//constexpr float beta1 = 0.8285; // 18Khz
constexpr float beta1 = 0.85; // 20Khz
constexpr float beta2 = (1.0f - beta1) / 2.0;
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
static float lowfilt = 0.0f; // IIR low frequency cutoff filter
for (int i=0; i < samplesFFT; i++) {
// FIR lowpass, to remove high frequency noise
float highFilteredSample;
if (i < (samplesFFT-1)) highFilteredSample = beta1*vReal[i] + beta2*last_vals[0] + beta2*vReal[i+1]; // smooth out spikes
else highFilteredSample = beta1*vReal[i] + beta2*last_vals[0] + beta2*last_vals[1]; // spcial handling for last sample in array
last_vals[1] = last_vals[0];
last_vals[0] = vReal[i];
vReal[i] = highFilteredSample;
// IIR highpass, to remove low frequency noise
lowfilt += alpha * (vReal[i] - lowfilt);
vReal[i] = vReal[i] - lowfilt;
}
}
if (useBandPassFilter) runMicFilter(samplesFFT, vReal);
// find highest sample in the batch
float maxSample = 0.0f; // max sample from FFT batch
@@ -459,10 +437,68 @@ void FFTcode(void * parameter)
// post-processing of frequency channels (pink noise adjustment, AGC, smooting, scaling)
if (pinkIndex > MAX_PINK) pinkIndex = MAX_PINK;
for (int i=0; i < NUM_GEQ_CHANNELS; i++) {
//postProcessFFTResults((fabsf(sampleAvg) > 0.25f)? true : false , NUM_GEQ_CHANNELS);
postProcessFFTResults((fabsf(volumeSmth)>0.25f)? true : false , NUM_GEQ_CHANNELS);
//if (fabsf(sampleAvg) > 0.25f) { // noise gate open
if (fabsf(volumeSmth) > 0.25f) { // noise gate open
#if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS)
if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows
uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
}
#endif
// run peak detection
autoResetPeak();
detectSamplePeak();
#if !defined(I2S_GRAB_ADC1_COMPLETELY)
if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC
#endif
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
} // for(;;)ever
} // FFTcode() task end
///////////////////////////
// Pre / Postprocessing //
///////////////////////////
static void runMicFilter(uint16_t numSamples, float *sampleBuffer) // pre-filtering of raw samples (band-pass)
{
// low frequency cutoff parameter - see https://dsp.stackexchange.com/questions/40462/exponential-moving-average-cut-off-frequency
//constexpr float alpha = 0.04f; // 150Hz
//constexpr float alpha = 0.03f; // 110Hz
constexpr float alpha = 0.0225f; // 80hz
//constexpr float alpha = 0.01693f;// 60hz
// high frequency cutoff parameter
//constexpr float beta1 = 0.75; // 11Khz
//constexpr float beta1 = 0.82; // 15Khz
//constexpr float beta1 = 0.8285; // 18Khz
constexpr float beta1 = 0.85; // 20Khz
constexpr float beta2 = (1.0f - beta1) / 2.0;
static float last_vals[2] = { 0.0f }; // FIR high freq cutoff filter
static float lowfilt = 0.0f; // IIR low frequency cutoff filter
for (int i=0; i < numSamples; i++) {
// 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
last_vals[1] = last_vals[0];
last_vals[0] = sampleBuffer[i];
sampleBuffer[i] = highFilteredSample;
// IIR highpass, to remove low frequency noise
lowfilt += alpha * (sampleBuffer[i] - lowfilt);
sampleBuffer[i] = sampleBuffer[i] - lowfilt;
}
}
static void postProcessFFTResults(bool noiseGateOpen, int numberOfChannels) // post-processing and post-amp of GEQ channels
{
for (int i=0; i < numberOfChannels; i++) {
if (noiseGateOpen) { // noise gate open
// Adjustment for frequency curves.
fftCalc[i] *= fftResultPink[pinkIndex][i];
if (FFTScalingMode > 0) fftCalc[i] *= FFT_DOWNSCALE; // adjustment related to FFT windowing function
@@ -473,7 +509,7 @@ void FFTcode(void * parameter)
// smooth results - rise fast, fall slower
if(fftCalc[i] > fftAvg[i]) // rise fast
fftAvg[i] = fftCalc[i] *0.75f + 0.25f*fftAvg[i]; // will need approx 2 cycles (50ms) for converging against fftCalc[i]
fftAvg[i] = fftCalc[i] *0.78f + 0.22f*fftAvg[i]; // will need approx 1-2 cycles (50ms) for converging against fftCalc[i]
else { // fall slow
if (decayTime < 250) fftAvg[i] = fftCalc[i]*0.4f + 0.6f*fftAvg[i];
else if (decayTime < 500) fftAvg[i] = fftCalc[i]*0.33f + 0.67f*fftAvg[i];
@@ -537,26 +573,7 @@ void FFTcode(void * parameter)
}
fftResult[i] = constrain((int)currentResult, 0, 255);
}
#if defined(WLED_DEBUG) || defined(SR_DEBUG)|| defined(SR_STATS)
if (haveDoneFFT && (start < esp_timer_get_time())) { // filter out overflows
uint64_t fftTimeInMillis = ((esp_timer_get_time() - start) +5ULL) / 10ULL; // "+5" to ensure proper rounding
fftTime = (fftTimeInMillis*3 + fftTime*7)/10; // smooth
}
#endif
// run peak detection
autoResetPeak();
detectSamplePeak();
#if !defined(I2S_GRAB_ADC1_COMPLETELY)
if ((audioSource == nullptr) || (audioSource->getType() != AudioSource::Type_I2SAdc)) // the "delay trick" does not help for analog ADC
#endif
vTaskDelayUntil( &xLastWakeTime, xFrequency); // release CPU, and let I2S fill its buffers
} // for(;;)ever
} // FFTcode() task end
}
////////////////////
// Peak detection //
////////////////////