Initial, Doggo walking.

.gitignore

@@ -0,0 +1 @@
+.pio

include/README

@@ -0,0 +1,39 @@
+
+This directory is intended for project header files.
+
+A header file is a file containing C declarations and macro definitions
+to be shared between several project source files. You request the use of a
+header file in your project source file (C, C++, etc) located in `src` folder
+by including it, with the C preprocessing directive `#include'.
+
+```src/main.c
+
+#include "header.h"
+
+int main (void)
+{
+ ...
+}
+```
+
+Including a header file produces the same results as copying the header file
+into each source file that needs it. Such copying would be time-consuming
+and error-prone. With a header file, the related declarations appear
+in only one place. If they need to be changed, they can be changed in one
+place, and programs that include the header file will automatically use the
+new version when next recompiled. The header file eliminates the labor of
+finding and changing all the copies as well as the risk that a failure to
+find one copy will result in inconsistencies within a program.
+
+In C, the usual convention is to give header files names that end with `.h'.
+It is most portable to use only letters, digits, dashes, and underscores in
+header file names, and at most one dot.
+
+Read more about using header files in official GCC documentation:
+
+* Include Syntax
+* Include Operation
+* Once-Only Headers
+* Computed Includes
+
+https://gcc.gnu.org/onlinedocs/cpp/Header-Files.html

lib/README

@@ -0,0 +1,46 @@
+
+This directory is intended for project specific (private) libraries.
+PlatformIO will compile them to static libraries and link into executable file.
+
+The source code of each library should be placed in a an own separate directory
+("lib/your_library_name/[here are source files]").
+
+For example, see a structure of the following two libraries `Foo` and `Bar`:
+
+|--lib
+|  |
+|  |--Bar
+|  |  |--docs
+|  |  |--examples
+|  |  |--src
+|  |     |- Bar.c
+|  |     |- Bar.h
+|  |  |- library.json (optional, custom build options, etc) https://docs.platformio.org/page/librarymanager/config.html
+|  |
+|  |--Foo
+|  |  |- Foo.c
+|  |  |- Foo.h
+|  |
+|  |- README --> THIS FILE
+|
+|- platformio.ini
+|--src
+   |- main.c
+
+and a contents of `src/main.c`:
+```
+#include <Foo.h>
+#include <Bar.h>
+
+int main (void)
+{
+  ...
+}
+
+```
+
+PlatformIO Library Dependency Finder will find automatically dependent
+libraries scanning project source files.
+
+More information about PlatformIO Library Dependency Finder
+- https://docs.platformio.org/page/librarymanager/ldf.html

platformio.ini

@@ -0,0 +1,16 @@
+; PlatformIO Project Configuration File
+;
+;   Build options: build flags, source filter
+;   Upload options: custom upload port, speed and extra flags
+;   Library options: dependencies, extra library storages
+;   Advanced options: extra scripting
+;
+; Please visit documentation for the other options and examples
+; https://docs.platformio.org/page/projectconf.html
+
+[env:wemos_d1_mini32]
+board = wemos_d1_mini32
+platform = espressif32
+framework = arduino
+monitor_speed = 115200
+lib_deps = ledc

src/AudioAnalysis.h

@@ -0,0 +1,669 @@
+#ifndef AudioAnalysis_H
+#define AudioAnalysis_H
+
+#include "Arduino.h"
+
+// arduinoFFT V2
+// See the develop branch on GitHub for the latest info and speedups.
+// https://github.com/kosme/arduinoFFT/tree/develop
+// if you are going for speed over percision uncomment the lines below.
+//#define FFT_SPEED_OVER_PRECISION
+//#define FFT_SQRT_APPROXIMATION
+
+#include <arduinoFFT.h>
+
+#ifndef SAMPLE_SIZE
+#define SAMPLE_SIZE 1024
+#endif
+#ifndef BAND_SIZE
+#define BAND_SIZE 8
+#endif
+
+class AudioAnalysis
+{
+public:
+  enum falloff_type
+  {
+    NO_FALLOFF,
+    LINEAR_FALLOFF,
+    ACCELERATE_FALLOFF,
+    EXPONENTIAL_FALLOFF,
+  };
+
+  AudioAnalysis();
+  /* FFT Functions */
+  void computeFFT(int32_t *samples, int sampleSize, int sampleRate); // calculates FFT on sample data
+  float *getReal();                                                  // gets the Real values after FFT calculation
+  float *getImaginary();                                             // gets the imaginary values after FFT calculation
+
+  /* Band Frequency Functions */
+  void setNoiseFloor(float noiseFloor);                                // threshold before sounds are registered
+  void computeFrequencies(uint8_t bandSize = BAND_SIZE);               // converts FFT data into frequency bands
+  void normalize(bool normalize = true, float min = 0, float max = 1); // normalize all values and constrain to min/max.
+
+  void autoLevel(falloff_type falloffType = ACCELERATE_FALLOFF, float falloffRate = 0.01, float min = 10, float max = -1); // auto ballance normalized values to ambient noise levels.
+                                                                                                                            // min and max are based on pre-normalized values.
+  void setEqualizerLevels(float low = 1, float mid = 1, float high = 1 );   // adjust the frequency levels for a given range - low, medium and high.
+                                                                            // 0.5 = 50%, 1.0 = 100%, 1.5 = 150%  the raw value etc.
+  void setEqualizerLevels(float *bandEq);                                   // full control over each bands eq value.
+  float *getEqualizerLevels(); // gets the last bandEq levels
+
+  bool
+  isNormalize();      // is normalize enabled
+  bool isAutoLevel(); // is auto level enabled
+  bool isClipping();  // is values exceding max
+
+  void bandPeakFalloff(falloff_type falloffType = ACCELERATE_FALLOFF, float falloffRate = 0.05); // set the falloff type and rate for band peaks.
+  void vuPeakFalloff(falloff_type falloffType = ACCELERATE_FALLOFF, float falloffRate = 0.05);   // set the falloff type and rate for volume unit peak.
+
+  float *getBands(); // gets the last bands calculated from computeFrequencies()
+  float *getPeaks(); // gets the last peaks calculated from computeFrequencies()
+
+  float getBand(uint8_t index); // gets the value at bands index
+  float getBandAvg();           // average value across all bands
+  float getBandMax();           // max value across all bands
+  int getBandMaxIndex();        // index of the highest value band
+  int getBandMinIndex();        // index of the lowest value band
+
+  float getPeak(uint8_t index); // gets the value at peaks index
+  float getPeakAvg();           // average value across all peaks
+  float getPeakMax();           // max value across all peaks
+  int getPeakMaxIndex();        // index of the highest value peak
+  int getPeakMinIndex();        // index of the lowest value peak
+
+  /* Volume Unit Functions */
+  float getVolumeUnit();        // gets the last volume unit calculated from computeFrequencies()
+  float getVolumeUnitPeak();    // gets the last volume unit peak calculated from computeFrequencies()
+  float getVolumeUnitMax();     // value of the highest value volume unit
+  float getVolumeUnitPeakMax(); // value of the highest value volume unit
+
+protected:
+  /* Library Settings */
+  bool _isAutoLevel = false;
+  bool _isClipping = false;
+  float _autoMin = 10; // lowest raw value the autoLevel will fall to before stopping. -1 = will auto level down to 0.
+  float _autoMax = -1; // highest raw value the autoLevel will rise to before clipping. -1 = will not have any clipping.
+
+  bool _isNormalize = false;
+  float _normalMin = 0;
+  float _normalMax = 1;
+
+  falloff_type _bandPeakFalloffType = ACCELERATE_FALLOFF;
+  float _bandPeakFalloffRate = 0.05;
+  falloff_type _vuPeakFalloffType = ACCELERATE_FALLOFF;
+  float _vuPeakFalloffRate = 0.05;
+  falloff_type _autoLevelFalloffType = ACCELERATE_FALLOFF;
+  float _autoLevelFalloffRate = 0.01;
+
+  float calculateFalloff(falloff_type falloffType, float falloffRate, float currentRate);
+  template <class X>
+  X mapAndClip(X x, X in_min, X in_max, X out_min, X out_max);
+
+  /* FFT Variables */
+  int32_t *_samples;
+  int _sampleSize;
+  int _sampleRate;
+  float _real[SAMPLE_SIZE];
+  float _imag[SAMPLE_SIZE];
+  float _weighingFactors[SAMPLE_SIZE];
+
+  /* Band Frequency Variables */
+  float _noiseFloor = 0;
+  int _bandSize = BAND_SIZE;
+  float _bands[BAND_SIZE];
+  float _peaks[BAND_SIZE];
+  float _peakFallRate[BAND_SIZE];
+  float _peaksNorms[BAND_SIZE];
+  float _bandsNorms[BAND_SIZE];
+  float _bandEq[BAND_SIZE];
+
+  float _bandAvg;
+  float _peakAvg;
+  int8_t _bandMinIndex;
+  int8_t _bandMaxIndex;
+  int8_t _peakMinIndex;
+  int8_t _peakMaxIndex;
+  float _bandMin;
+  float _bandMax; // used for normalization calculation
+  float _peakMin;
+  float _autoLevelPeakMax; // used for normalization calculation
+  // float _peakMinFalloffRate;
+  float _autoLevelPeakMaxFalloffRate; // used for auto level calculation
+
+  /* Volume Unit Variables */
+  float _vu;
+  float _vuPeak;
+  float _vuPeakFallRate;
+  float _vuMin;
+  float _vuMax; // used for normalization calculation
+  float _vuPeakMin;
+  float _autoLevelVuPeakMax; // used for normalization calculation
+  // float _vuPeakMinFalloffRate;
+  float _autoLevelMaxFalloffRate; // used for auto level calculation
+  ArduinoFFT<float> *_FFT = nullptr;
+};
+
+AudioAnalysis::AudioAnalysis()
+{
+  // set default eq levels;
+  for (int i = 0; i < _bandSize; i++)
+  {
+    _bandEq[i] = 1.0;
+  }
+}
+
+void AudioAnalysis::computeFFT(int32_t *samples, int sampleSize, int sampleRate)
+{
+  _samples = samples;
+  if (_FFT == nullptr || _sampleSize != sampleSize || _sampleRate != sampleRate)
+  {
+    _sampleSize = sampleSize;
+    _sampleRate = sampleRate;
+    _FFT = new ArduinoFFT<float>(_real, _imag, _sampleSize, _sampleRate, _weighingFactors);
+  }
+
+  // prep samples for analysis
+  for (int i = 0; i < _sampleSize; i++)
+  {
+    _real[i] = samples[i];
+    _imag[i] = 0;
+  }
+
+  _FFT->dcRemoval();
+  _FFT->windowing(FFTWindow::Hamming, FFTDirection::Forward, false); /* Weigh data (compensated) */
+  _FFT->compute(FFTDirection::Forward);                              /* Compute FFT */
+  _FFT->complexToMagnitude();                                        /* Compute magnitudes */
+}
+
+float *AudioAnalysis::getReal()
+{
+  return _real;
+}
+
+float *AudioAnalysis::getImaginary()
+{
+  return _imag;
+}
+
+void AudioAnalysis::setNoiseFloor(float noiseFloor)
+{
+  _noiseFloor = noiseFloor;
+}
+
+float getPoint(float n1, float n2, float percent)
+{
+  float diff = n2 - n1;
+
+  return n1 + (diff * percent);
+}
+
+void AudioAnalysis::setEqualizerLevels(float low, float mid, float high)
+{
+  float xa, ya, xb, yb, x, y;
+  // low curve
+  float x1 = 0;
+  float lowSize = _bandSize / 4;
+  float y1 = low;
+  float x2 = lowSize / 2;
+  float y2 = low;
+  float x3 = lowSize;
+  float y3 = (low + mid)/2.0;
+  for (int i = x1; i < lowSize; i++)
+  {
+    float p = (float)i / (float)lowSize;
+    //xa = getPoint(x1, x2, p);
+    ya = getPoint(y1, y2, p);
+    //xb = getPoint(x2, x3, p);
+    yb = getPoint(y2, y3, p);
+
+    //x = getPoint(xa, xb, p);
+    y = getPoint(ya, yb, p);
+
+    _bandEq[i] = y;
+  }
+
+  // mid curve
+  x1 = lowSize;
+  float midSize = (_bandSize-lowSize) / 2;
+  y1 = y3;
+  x2 = x1 + midSize / 2;
+  y2 = mid;
+  x3 = x1 + midSize;
+  y3 = (mid + high) / 2.0;
+  for (int i = x1; i < x1+midSize; i++)
+  {
+    float p = (float)(i - x1) / (float)midSize;
+    // xa = getPoint(x1, x2, p);
+    ya = getPoint(y1, y2, p);
+    // xb = getPoint(x2, x3, p);
+    yb = getPoint(y2, y3, p);
+
+    // x = getPoint(xa, xb, p);
+    y = getPoint(ya, yb, p);
+
+    _bandEq[i] = y;
+  }
+
+  // high curve
+  x1 = lowSize + midSize;
+  float highSize = midSize;
+  y1 = y3;
+  x2 = x1 + highSize / 2;
+  y2 = high;
+  x3 = x1 + highSize;
+  y3 = high;
+  for (int i = x1; i < x1+highSize; i++)
+  {
+    float p = (float)(i - x1) / (float)highSize;
+    // xa = getPoint(x1, x2, p);
+    ya = getPoint(y1, y2, p);
+    // xb = getPoint(x2, x3, p);
+    yb = getPoint(y2, y3, p);
+
+    // x = getPoint(xa, xb, p);
+    y = getPoint(ya, yb, p);
+
+    _bandEq[i] = y;
+  }
+}
+
+void AudioAnalysis::setEqualizerLevels(float *bandEq)
+{
+  // blind copy of eq percentages
+  for(int i = 0; i < _bandSize; i++) {
+    _bandEq[i] = bandEq[i];
+  }
+}
+
+float *AudioAnalysis::getEqualizerLevels()
+{
+  return _bandEq;
+}
+
+void AudioAnalysis::computeFrequencies(uint8_t bandSize)
+{
+  // TODO: use maths to calculate these offset values. Inputs being _sampleSize and _bandSize output being similar exponential curve below.
+  // band offsets helpers based on 1024 samples
+  const static uint16_t _2frequencyOffsets[2] = {24, 359};
+  const static uint16_t _4frequencyOffsets[4] = {6, 18, 72, 287};
+  const static uint16_t _8frequencyOffsets[8] = {2, 4, 6, 12, 25, 47, 92, 195};
+  const static uint16_t _16frequencyOffsets[16] = {1, 1, 2, 2, 2, 4, 5, 7, 11, 14, 19, 28, 38, 54, 75, 120}; // initial 
+  // 32 and 64 frequency offsets are low end biased because of int math... the first 4 and 8 buckets should be 0.5f but we cant do that here.
+  const static uint16_t _32frequencyOffsets[32] = {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 3, 3, 5, 5, 7, 7, 8, 8, 14, 14, 19, 19, 27, 27, 37, 37, 60, 60};
+  const static uint16_t _64frequencyOffsets[64] = {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 7, 7, 7, 7, 8, 8, 8, 8, 13, 13, 13, 13, 18, 18, 18, 18, 30, 30, 30, 30}; // low end biased because of int
+  const uint16_t *_frequencyOffsets;
+try_frequency_offsets_again:
+  switch (bandSize)
+  {
+  case 2:
+    _frequencyOffsets = _2frequencyOffsets;
+    break;
+  case 4:
+    _frequencyOffsets = _4frequencyOffsets;
+    break;
+  case 8:
+    _frequencyOffsets = _8frequencyOffsets;
+    break;
+  case 16:
+    _frequencyOffsets = _16frequencyOffsets;
+    break;
+  case 32:
+    _frequencyOffsets = _32frequencyOffsets;
+    break;
+  case 64:
+    _frequencyOffsets = _64frequencyOffsets;
+    break;
+  default:
+    bandSize = BAND_SIZE;
+    goto try_frequency_offsets_again;
+  }
+  _bandSize = bandSize;
+  _isClipping = false;
+  // for normalize falloff rates
+  if (_isAutoLevel)
+  {
+    if (_autoLevelPeakMax > _autoMin)
+    {
+      _autoLevelPeakMaxFalloffRate = calculateFalloff(_autoLevelFalloffType, _autoLevelFalloffRate, _autoLevelPeakMaxFalloffRate);
+      _autoLevelPeakMax -= _autoLevelPeakMaxFalloffRate;
+    }
+    if (_autoLevelVuPeakMax > _autoMin * 1.5)
+    {
+      _autoLevelMaxFalloffRate = calculateFalloff(_autoLevelFalloffType, _autoLevelFalloffRate, _autoLevelMaxFalloffRate);
+      _autoLevelVuPeakMax -= _autoLevelMaxFalloffRate;
+    }
+  }
+  _vu = 0;
+  _bandMax = 0;
+  _bandAvg = 0;
+  _peakAvg = 0;
+  _bandMaxIndex = -1;
+  _bandMinIndex = -1;
+  _peakMaxIndex = -1;
+  _peakMinIndex = -1;
+  int offset = 2; // first two values are noise
+  for (int i = 0; i < _bandSize; i++)
+  {
+    _bands[i] = 0;
+    // handle band peaks fall off
+    _peakFallRate[i] = calculateFalloff(_bandPeakFalloffType, _bandPeakFalloffRate, _peakFallRate[i]);
+    if (_peaks[i] - _peakFallRate[i] <= _bands[i])
+    {
+      _peaks[i] = _bands[i];
+    }
+    else
+    {
+      _peaks[i] -= _peakFallRate[i]; // fall off rate
+    }
+    for (int j = 0; j < _frequencyOffsets[i]; j++)
+    {
+      // scale down factor to prevent overflow
+      int rv = (_real[offset + j] / (0xFFFF * 0xFF));
+      int iv = (_imag[offset + j] / (0xFFFF * 0xFF));
+      // some smoothing with imaginary numbers.
+      rv = sqrt(rv * rv + iv * iv);
+      // add eq offsets
+      rv = rv * _bandEq[i];
+      // combine band amplitudes for current band segment
+      _bands[i] += rv;
+      _vu += rv;
+    }
+    offset += _frequencyOffsets[i];
+
+    // remove noise
+    if (_bands[i] < _noiseFloor)
+    {
+      _bands[i] = 0;
+    }
+
+    if (_bands[i] > _peaks[i])
+    {
+      _peakFallRate[i] = 0;
+      _peaks[i] = _bands[i];
+    }
+
+    // handle min/max band
+    if (_bands[i] > _bandMax && _bands[i] > _noiseFloor)
+    {
+      _bandMax = _bands[i];
+      _bandMaxIndex = i;
+    }
+    if (_bands[i] < _bandMin)
+    {
+      _bandMin = _bands[i];
+      _bandMinIndex = i;
+    }
+    // handle min/max peak
+    if (_peaks[i] > _autoLevelPeakMax)
+    {
+      _autoLevelPeakMax = _peaks[i];
+      if (_isAutoLevel && _autoMax != -1 && _peaks[i] > _autoMax)
+      {
+        _isClipping = true;
+        _autoLevelPeakMax = _autoMax;
+      }
+      _peakMaxIndex = i;
+      _autoLevelPeakMaxFalloffRate = 0;
+    }
+    if (_peaks[i] < _peakMin && _peaks[i] > _noiseFloor)
+    {
+      _peakMin = _peaks[i];
+      _peakMinIndex = i;
+    }
+
+    // handle band average
+    _bandAvg += _bands[i];
+    _peakAvg += _peaks[i];
+  } // end bands
+  // handle band average
+  _bandAvg = _bandAvg / _bandSize;
+  _peakAvg = _peakAvg / _bandSize;
+
+  // handle vu peak fall off
+  _vu = _vu / 8.0; // get it closer to the band peak values
+  _vuPeakFallRate = calculateFalloff(_vuPeakFalloffType, _vuPeakFalloffRate, _vuPeakFallRate);
+  _vuPeak -= _vuPeakFallRate;
+  if (_vu > _vuPeak)
+  {
+    _vuPeakFallRate = 0;
+    _vuPeak = _vu;
+  }
+  if (_vu > _vuMax)
+  {
+    _vuMax = _vu;
+  }
+  if (_vu < _vuMin)
+  {
+    _vuMin = _vu;
+  }
+  if (_vuPeak > _autoLevelVuPeakMax)
+  {
+    _autoLevelVuPeakMax = _vuPeak;
+    if (_isAutoLevel && _autoMax != -1 && _vuPeak > _autoMax)
+    {
+      _isClipping = true;
+      _autoLevelVuPeakMax = _autoMax;
+    }
+    _autoLevelMaxFalloffRate = 0;
+  }
+  if (_vuPeak < _vuPeakMin)
+  {
+    _vuPeakMin = _vuPeak;
+  }
+}
+
+template <class X>
+X AudioAnalysis::mapAndClip(X x, X in_min, X in_max, X out_min, X out_max)
+{
+  if (_isAutoLevel && _autoMax != -1 && x > _autoMax)
+  {
+    // clip the value to max
+    x = _autoMax;
+  }
+  else if (x > in_max)
+  {
+    // value is clipping
+    x = in_max;
+  }
+  return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min;
+}
+
+void AudioAnalysis::normalize(bool normalize, float min, float max)
+{
+  _isNormalize = normalize;
+  _normalMin = min;
+  _normalMax = max;
+}
+void AudioAnalysis::bandPeakFalloff(falloff_type falloffType, float falloffRate)
+{
+  _bandPeakFalloffType = falloffType;
+  _bandPeakFalloffRate = falloffRate;
+}
+
+void AudioAnalysis::vuPeakFalloff(falloff_type falloffType, float falloffRate)
+{
+  _vuPeakFalloffType = falloffType;
+  _vuPeakFalloffRate = falloffRate;
+}
+
+float AudioAnalysis::calculateFalloff(falloff_type falloffType, float falloffRate, float currentRate)
+{
+  switch (falloffType)
+  {
+  case LINEAR_FALLOFF:
+    return falloffRate;
+  case ACCELERATE_FALLOFF:
+    return currentRate + falloffRate;
+  case EXPONENTIAL_FALLOFF:
+    if (currentRate == 0)
+    {
+      currentRate = falloffRate;
+    }
+    return currentRate + currentRate;
+  case NO_FALLOFF:
+  default:
+    return 0;
+  }
+}
+
+void AudioAnalysis::autoLevel(falloff_type falloffType, float falloffRate, float min, float max)
+{
+  _isAutoLevel = falloffType != NO_FALLOFF;
+  _autoLevelFalloffType = falloffType;
+  _autoLevelFalloffRate = falloffRate;
+  _autoMin = min;
+  _autoMax = max;
+}
+
+bool AudioAnalysis::isNormalize()
+{
+  return _isNormalize;
+}
+
+bool AudioAnalysis::isAutoLevel()
+{
+  return _isAutoLevel;
+}
+
+bool AudioAnalysis::isClipping()
+{
+  return _isClipping;
+}
+
+float *AudioAnalysis::getBands()
+{
+  if (_isNormalize)
+  {
+    for (int i = 0; i < _bandSize; i++)
+    {
+      _bandsNorms[i] = mapAndClip(_bands[i], 0.0f, _autoLevelPeakMax, _normalMin, _normalMax);
+    }
+    return _bandsNorms;
+  }
+  return _bands;
+}
+
+float AudioAnalysis::getBand(uint8_t index)
+{
+  if (index >= _bandSize || index < 0)
+  {
+    return 0;
+  }
+  if (_isNormalize)
+  {
+    return mapAndClip(_bands[index], 0.0f, _autoLevelPeakMax, _normalMin, _normalMax);
+  }
+  return _bands[index];
+}
+
+float AudioAnalysis::getBandAvg()
+{
+  if (_isNormalize)
+  {
+    return mapAndClip(_bandAvg, 0.0f, _autoLevelPeakMax, _normalMin, _normalMax);
+  }
+  return _bandAvg;
+}
+
+float AudioAnalysis::getBandMax()
+{
+  return getBand(getBandMaxIndex());
+}
+
+int AudioAnalysis::getBandMaxIndex()
+{
+  return _bandMaxIndex;
+}
+
+int AudioAnalysis::getBandMinIndex()
+{
+  return _bandMinIndex;
+}
+
+float *AudioAnalysis::getPeaks()
+{
+  if (_isNormalize)
+  {
+    for (int i = 0; i < _bandSize; i++)
+    {
+      _peaksNorms[i] = mapAndClip(_peaks[i], 0.0f, _autoLevelPeakMax, _normalMin, _normalMax);
+    }
+    return _peaksNorms;
+  }
+  return _peaks;
+}
+
+float AudioAnalysis::getPeak(uint8_t index)
+{
+  if (index >= _bandSize || index < 0)
+  {
+    return 0;
+  }
+  if (_isNormalize)
+  {
+    return mapAndClip(_peaks[index], 0.0f, _autoLevelPeakMax, _normalMin, _normalMax);
+  }
+  return _peaks[index];
+}
+
+float AudioAnalysis::getPeakAvg()
+{
+  if (_isNormalize)
+  {
+    return mapAndClip(_peakAvg, 0.0f, _autoLevelPeakMax, _normalMin, _normalMax);
+  }
+  return _peakAvg;
+}
+
+float AudioAnalysis::getPeakMax()
+{
+  return getPeak(getPeakMaxIndex());
+}
+
+int AudioAnalysis::getPeakMaxIndex()
+{
+  return _peakMaxIndex;
+}
+
+int AudioAnalysis::getPeakMinIndex()
+{
+  return _peakMinIndex;
+}
+
+float AudioAnalysis::getVolumeUnit()
+{
+  if (_isNormalize)
+  {
+    return mapAndClip(_vu, 0.0f, _autoLevelVuPeakMax, _normalMin, _normalMax);
+  }
+  return _vu;
+}
+
+float AudioAnalysis::getVolumeUnitPeak()
+{
+  if (_isNormalize)
+  {
+    return mapAndClip(_vuPeak, 0.0f, _autoLevelVuPeakMax, _normalMin, _normalMax);
+  }
+  return _vuPeak;
+}
+
+float AudioAnalysis::getVolumeUnitMax()
+{
+  if (_isNormalize)
+  {
+    return mapAndClip(_vuMax, 0.0f, _autoLevelVuPeakMax, _normalMin, _normalMax);
+  }
+  return _vuMax;
+}
+
+float AudioAnalysis::getVolumeUnitPeakMax()
+{
+  if (_isNormalize)
+  {
+    return _normalMax;
+  }
+  return _autoLevelVuPeakMax;
+}
+
+#endif // AudioAnalysis_H

src/AudioInI2S.h

@@ -0,0 +1,87 @@
+#ifndef AudioInI2S_H
+#define AudioInI2S_H
+
+#include "Arduino.h"
+#include <driver/i2s.h>
+
+class AudioInI2S
+{
+public:
+  AudioInI2S(int bck_pin, int ws_pin, int data_pin, int channel_pin = -1, i2s_channel_fmt_t channel_format = I2S_CHANNEL_FMT_ONLY_RIGHT);
+  void read(int32_t _samples[]);
+  void begin(int sample_size, int sample_rate = 44100, i2s_port_t i2s_port_number = I2S_NUM_0);
+
+private:
+  int _bck_pin;
+  int _ws_pin;
+  int _data_pin;
+  int _channel_pin;
+  i2s_channel_fmt_t _channel_format;
+  int _sample_size;
+  int _sample_rate;
+  i2s_port_t _i2s_port_number;
+
+  i2s_config_t _i2s_config = {
+      .mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_RX),
+      .sample_rate = 0, // set in begin()
+      .bits_per_sample = I2S_BITS_PER_SAMPLE_32BIT,
+      .channel_format = I2S_CHANNEL_FMT_ONLY_RIGHT,
+      .communication_format = I2S_COMM_FORMAT_I2S,
+      .intr_alloc_flags = ESP_INTR_FLAG_LEVEL1,
+      .dma_buf_count = 4,
+      .dma_buf_len = 0, // set in begin()
+      .use_apll = false,
+      .tx_desc_auto_clear = false,
+      .fixed_mclk = 0};
+
+  i2s_pin_config_t _i2s_mic_pins = {
+      .bck_io_num = I2S_PIN_NO_CHANGE, // set in begin()
+      .ws_io_num = I2S_PIN_NO_CHANGE,  // set in begin()
+      .data_out_num = I2S_PIN_NO_CHANGE,
+      .data_in_num = I2S_PIN_NO_CHANGE // set in begin()
+  };
+};
+
+AudioInI2S::AudioInI2S(int bck_pin, int ws_pin, int data_pin, int channel_pin, i2s_channel_fmt_t channel_format)
+{
+  _bck_pin = bck_pin;
+  _ws_pin = ws_pin;
+  _data_pin = data_pin;
+  _channel_pin = channel_pin;
+  _channel_format = channel_format;
+}
+
+void AudioInI2S::begin(int sample_size, int sample_rate, i2s_port_t i2s_port_number)
+{
+  if (_channel_pin >= 0)
+  {
+    pinMode(_channel_pin, OUTPUT);
+    digitalWrite(_channel_pin, _channel_format == I2S_CHANNEL_FMT_ONLY_RIGHT ? LOW : HIGH);
+  }
+
+  _sample_rate = sample_rate;
+  _sample_size = sample_size;
+  _i2s_port_number = i2s_port_number;
+
+  _i2s_mic_pins.bck_io_num = _bck_pin;
+  _i2s_mic_pins.ws_io_num = _ws_pin;
+  _i2s_mic_pins.data_in_num = _data_pin;
+
+  _i2s_config.sample_rate = _sample_rate;
+  _i2s_config.dma_buf_len = _sample_size;
+  _i2s_config.channel_format = _channel_format;
+
+  // start up the I2S peripheral
+  i2s_driver_install(_i2s_port_number, &_i2s_config, 0, NULL);
+  i2s_set_pin(_i2s_port_number, &_i2s_mic_pins);
+}
+
+void AudioInI2S::read(int32_t _samples[])
+{
+  // read I2S stream data into the samples buffer
+  size_t bytes_read = 0;
+  i2s_read(_i2s_port_number, _samples, sizeof(int32_t) * _sample_size, &bytes_read, portMAX_DELAY);
+  int samples_read = bytes_read / sizeof(int32_t);
+}
+
+#endif // AudioInI2S_H

src/main.cpp

@@ -0,0 +1,168 @@
+/*
+  ESP32 I2S Microphone Sample
+  esp32-i2s-mic-sample.ino
+  Sample sound from I2S microphone, display on Serial Plotter
+  Requires INMP441 I2S microphone
+
+  DroneBot Workshop 2022
+  https://dronebotworkshop.com
+*/
+
+// Include I2S driver
+#include <driver/i2s.h>
+ #include <HardwareSerial.h>
+// Connections to INMP441 I2S microphone
+#define I2S_WS 23
+#define I2S_SD 33
+#define I2S_SCK 27
+
+// Use I2S Processor 0
+#define I2S_PORT I2S_NUM_0
+
+// Define input buffer length
+#define bufferLen 64
+int16_t sBuffer[bufferLen];
+
+
+// Settings for LEDC PWM 
+#include "driver/ledc.h"
+#include "esp_err.h"
+
+#define LEDC_TIMER              LEDC_TIMER_0
+#define LEDC_MODE               LEDC_LOW_SPEED_MODE
+#define LEDC_OUTPUT_IO          (0) // Define the output GPIO
+#define LEDC_CHANNEL            LEDC_CHANNEL_0
+#define LEDC_DUTY_RES           LEDC_TIMER_10_BIT // Set duty resolution to 13 bits
+#define LEDC_DUTY               (4095) // Set duty to 50%. ((2 ** 13) - 1) * 50% = 4095
+#define LEDC_FREQUENCY          (20000) // Frequency in Hertz. Set frequency at 20 kHz
+
+void i2s_install() {
+  // Set up I2S Processor configuration
+  const i2s_config_t i2s_config = {
+    .mode = i2s_mode_t(I2S_MODE_MASTER | I2S_MODE_RX),
+    .sample_rate = 44100,
+    .bits_per_sample = i2s_bits_per_sample_t(16),
+    .channel_format = I2S_CHANNEL_FMT_ONLY_LEFT,
+    .communication_format = i2s_comm_format_t(I2S_COMM_FORMAT_STAND_I2S),
+    .intr_alloc_flags = 0,
+    .dma_buf_count = 8,
+    .dma_buf_len = bufferLen,
+    .use_apll = false
+  };
+
+  i2s_driver_install(I2S_PORT, &i2s_config, 0, NULL);
+}
+
+void i2s_setpin() {
+  // Set I2S pin configuration
+  const i2s_pin_config_t pin_config = {
+    .bck_io_num = I2S_SCK,
+    .ws_io_num = I2S_WS,
+    .data_out_num = -1,
+    .data_in_num = I2S_SD
+  };
+
+  i2s_set_pin(I2S_PORT, &pin_config);
+}
+
+static void example_ledc_init(void)
+{
+    // Prepare and then apply the LEDC PWM timer configuration
+    ledc_timer_config_t ledc_timer = {
+        .speed_mode       = LEDC_MODE,
+        .duty_resolution  = LEDC_DUTY_RES,
+        .timer_num        = LEDC_TIMER,
+        .freq_hz          = LEDC_FREQUENCY,  // Set output frequency at 5 kHz
+        .clk_cfg          = LEDC_AUTO_CLK
+    };
+    ESP_ERROR_CHECK(ledc_timer_config(&ledc_timer));
+
+    // Prepare and then apply the LEDC PWM channel configuration
+    ledc_channel_config_t ledc_channel = {
+        .gpio_num       = LEDC_OUTPUT_IO,
+        .speed_mode     = LEDC_MODE,
+        .channel        = LEDC_CHANNEL,
+        .intr_type      = LEDC_INTR_DISABLE,
+        .timer_sel      = LEDC_TIMER,
+        .duty           = 0, // Set duty to 0%
+        .hpoint         = 0
+    };
+    ESP_ERROR_CHECK(ledc_channel_config(&ledc_channel));
+}
+
+//static void save_power(void)
+//{
+//
+//    
+//    ESP_ERROR_CHECK(esp_pm_configure(&power_configuration));
+//}
+
+void setup() {
+
+  // Set up Serial Monitor
+  Serial.begin(115200);
+  Serial.println(" ");
+  delay(500);
+
+  // Set up I2S
+  i2s_install();
+  i2s_setpin();
+  i2s_start(I2S_PORT);
+
+  delay(500);
+
+  // Set the LEDC peripheral configuration
+  example_ledc_init();
+}
+
+
+
+float thresholdMax = 1000;
+float thresholdMin = 30;
+float threshold=thresholdMax;
+
+void loop() {
+
+  // Get I2S data and place in data buffer
+  size_t bytesIn = 0;
+  esp_err_t result = i2s_read(I2S_PORT, &sBuffer, bufferLen, &bytesIn, portMAX_DELAY);
+
+  if (result == ESP_OK)
+  {
+    // Read I2S data buffer
+    int16_t samples_read = bytesIn / 8;
+    if (samples_read > 0) {
+      float mean = 0;
+      for (int16_t i = 0; i < samples_read; ++i) {
+        mean += (sBuffer[i]);
+      }
+
+      // Average the data reading
+      mean /= samples_read;
+
+      // Print to serial plotter
+      Serial.print(-threshold);
+      Serial.print(" ");
+      Serial.print(threshold);
+      Serial.print(" ");
+      Serial.println(mean);
+      if ( mean < threshold) threshold = std::max(thresholdMin, threshold * 0.9999f); 
+     
+      auto dogOnDuty = std::max(0.0f, threshold - thresholdMin);
+      if(dogOnDuty < 500) dogOnDuty = 1; 
+      ESP_ERROR_CHECK(ledc_set_duty(LEDC_MODE, LEDC_CHANNEL, dogOnDuty));
+      // Update duty to apply the new value
+      ESP_ERROR_CHECK(ledc_update_duty(LEDC_MODE, LEDC_CHANNEL));
+      
+      if ( std::abs(mean) > threshold) {
+	 // DOG WAKEUP
+	 threshold = thresholdMax;
+          // Set the LEDC peripheral configuration
+    	// Set duty to 50%
+   	 //ESP_ERROR_CHECK(ledc_set_duty(LEDC_MODE, LEDC_CHANNEL, LEDC_DUTY));
+    	// Update duty to apply the new value
+    	//ESP_ERROR_CHECK(ledc_update_duty(LEDC_MODE, LEDC_CHANNEL));
+      }
+    }
+  }
+}

test/README

@@ -0,0 +1,11 @@
+
+This directory is intended for PlatformIO Test Runner and project tests.
+
+Unit Testing is a software testing method by which individual units of
+source code, sets of one or more MCU program modules together with associated
+control data, usage procedures, and operating procedures, are tested to
+determine whether they are fit for use. Unit testing finds problems early
+in the development cycle.
+
+More information about PlatformIO Unit Testing:
+- https://docs.platformio.org/en/latest/advanced/unit-testing/index.html