commit 0ed602f61b17ca05b1362ed5fde88e62a7ad5570
Author: Christoph Sterz <christoph.sterz@kdab.com>
Date: Thu Jul 20 11:39:45 2023 +0200
Initial, Doggo walking.
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..03f4a3c
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1 @@
+.pio
diff --git a/include/README b/include/README
new file mode 100644
index 0000000..194dcd4
--- /dev/null
+++ b/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
diff --git a/lib/README b/lib/README
new file mode 100644
index 0000000..6debab1
--- /dev/null
+++ b/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
diff --git a/platformio.ini b/platformio.ini
new file mode 100644
index 0000000..2db0c68
--- /dev/null
+++ b/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
diff --git a/src/AudioAnalysis.h b/src/AudioAnalysis.h
new file mode 100644
index 0000000..ddff321
--- /dev/null
+++ b/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
diff --git a/src/AudioInI2S.h b/src/AudioInI2S.h
new file mode 100644
index 0000000..3e981a4
--- /dev/null
+++ b/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
diff --git a/src/main.cpp b/src/main.cpp
new file mode 100644
index 0000000..709ba73
--- /dev/null
+++ b/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));
+ }
+ }
+ }
+}
diff --git a/test/README b/test/README
new file mode 100644
index 0000000..9b1e87b
--- /dev/null
+++ b/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