Initial, Doggo walking.
This commit is contained in:
Christoph Sterz
commit
0ed602f61b
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.pio
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This directory is intended for project header files.
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A header file is a file containing C declarations and macro definitions
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to be shared between several project source files. You request the use of a
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header file in your project source file (C, C++, etc) located in `src` folder
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by including it, with the C preprocessing directive `#include'.
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||||
|
||||
```src/main.c
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||||
|
||||
#include "header.h"
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||||
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int main (void)
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||||
{
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...
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}
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```
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Including a header file produces the same results as copying the header file
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into each source file that needs it. Such copying would be time-consuming
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and error-prone. With a header file, the related declarations appear
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in only one place. If they need to be changed, they can be changed in one
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place, and programs that include the header file will automatically use the
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new version when next recompiled. The header file eliminates the labor of
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finding and changing all the copies as well as the risk that a failure to
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find one copy will result in inconsistencies within a program.
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In C, the usual convention is to give header files names that end with `.h'.
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It is most portable to use only letters, digits, dashes, and underscores in
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header file names, and at most one dot.
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Read more about using header files in official GCC documentation:
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* Include Syntax
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* Include Operation
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* Once-Only Headers
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* Computed Includes
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https://gcc.gnu.org/onlinedocs/cpp/Header-Files.html
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This directory is intended for project specific (private) libraries.
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PlatformIO will compile them to static libraries and link into executable file.
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The source code of each library should be placed in a an own separate directory
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("lib/your_library_name/[here are source files]").
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For example, see a structure of the following two libraries `Foo` and `Bar`:
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|--lib
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| |
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| |--Bar
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| | |--docs
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| | |--examples
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| | |--src
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| | |- Bar.c
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| | |- Bar.h
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| | |- library.json (optional, custom build options, etc) https://docs.platformio.org/page/librarymanager/config.html
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| |
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| |--Foo
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| | |- Foo.c
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| | |- Foo.h
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| |
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| |- README --> THIS FILE
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|
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|- platformio.ini
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|--src
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|- main.c
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|
||||
and a contents of `src/main.c`:
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```
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||||
#include <Foo.h>
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||||
#include <Bar.h>
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||||
|
||||
int main (void)
|
||||
{
|
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...
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}
|
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```
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PlatformIO Library Dependency Finder will find automatically dependent
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libraries scanning project source files.
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More information about PlatformIO Library Dependency Finder
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- https://docs.platformio.org/page/librarymanager/ldf.html
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; PlatformIO Project Configuration File
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; Build options: build flags, source filter
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;
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; Please visit documentation for the other options and examples
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; https://docs.platformio.org/page/projectconf.html
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|
||||
[env:wemos_d1_mini32]
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||||
board = wemos_d1_mini32
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||||
platform = espressif32
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||||
framework = arduino
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||||
monitor_speed = 115200
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lib_deps = ledc
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#ifndef AudioAnalysis_H
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#define AudioAnalysis_H
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#include "Arduino.h"
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|
||||
// 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
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|
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#include <arduinoFFT.h>
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#ifndef SAMPLE_SIZE
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#define SAMPLE_SIZE 1024
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#endif
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#ifndef BAND_SIZE
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#define BAND_SIZE 8
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||||
#endif
|
||||
|
||||
class AudioAnalysis
|
||||
{
|
||||
public:
|
||||
enum falloff_type
|
||||
{
|
||||
NO_FALLOFF,
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||||
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.
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// min and max are based on pre-normalized values.
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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
|
||||
|
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bool
|
||||
isNormalize(); // is normalize enabled
|
||||
bool isAutoLevel(); // is auto level enabled
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||||
bool isClipping(); // is values exceding max
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|
||||
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.
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float *getBands(); // gets the last bands calculated from computeFrequencies()
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||||
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;
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||||
falloff_type _autoLevelFalloffType = ACCELERATE_FALLOFF;
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||||
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
|
||||
|
|
@ -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
|
||||
|
|
@ -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));
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
@ -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
|
||||
Loading…
Reference in New Issue