A simple MEMS I2S microphone and audio processing library for ESP32.
- Simple I2S sample reading and setup. Just choose the pins, sample size and sample rate.
- Robust audio processing library for analysis.
- Simple FFT compute on your I2S samples.
- Frequency bands in 2, 4, 8, 16, 32 or 64 buckets.
- Volume Unit Meter.
- Set a noise floor to ignore values below it.
- Normalize values into desired min/max ranges.
- Auto level values for noisy/quiet enviroments where you want to keep values around the normalize max.
- Ability to set the peak falloff rates and types. NO_FALLOFF, LINEAR_FALLOFF, ACCELERATE_FALLOFF, EXPONENTIAL_FALLOFF.
- Equalizer to adjust the frequency levels for each bucket. Good for lowering the bass or trebble response depending on the enviroment.
- Easy to follow examples
Basic- Reads I2S microphone data to be viewed in the Serial Plotter.Frequencies- Reads I2S microphone data, processes them into frequency buckets to be viewed in the Serial Plotter.FastLED- Reads I2S microphone data, processes them into frequency buckets and displays them on a WS2812B led strip.OLED- Reads I2S microphone data, processes them into frequency buckets and displays them on a 128x64 OLED display.WIFI- COMING SOON - Reads I2S microphone data, processes them into frequency buckets and sends it wirelessly to a webpage.
- ESP32, ESP32 S2, ESP32 C2, ESP32 C3
- INMP441 - MEMS Microphone
#include <AudioInI2S.h>- AudioInI2S(int bck_pin, int ws_pin, int data_pin, int channel_pin, i2s_channel_fmt_t channel_format) // pin setup
- void begin(int sample_size, int sample_rate = 44100, i2s_port_t i2s_port_number = I2S_NUM_0) - Starts the I2S DMA port.
- void read(int32_t _samples[]) - Stores the current I2S port buffer into samples.
#include <AudioAnalysis.h>- AudioAnalysis()
FFT Functions
- void computeFFT(int32_t samples[], int sample_size, int sample_rate) - 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 band_size = 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 = 255, 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. Array of float percentage values 1.0 = 100% [BAND_SIZE ...]
- **float *getEqualizerLevels(); // gets the last bandEq levels
- 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.
- isNormalize() - is normalize enabled
- isAutoLevel() - is auto level enabled
- isClipping() - is values exceding max
- float *getBands() - gets the last bands calculated from processFrequencies()
- float *getPeaks() - gets the last peaks calculated from processFrequencies()
- 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 processFrequencies()
- float getVolumeUnitPeak() - gets the last volume unit peak calculated from processFrequencies()
- float getVolumeUnitMax() - value of the highest value volume unit
- float getVolumeUnitPeakMax() - value of the highest value volume unit
Checkout the examples/Frequencies, examples/FastLED and examples/OLED_128x64 examples folder for audio analysis.
/* Basic.ino */
#include <AudioInI2S.h>
#define SAMPLE_SIZE 1024 // Buffer size of read samples
#define SAMPLE_RATE 44100 // Audio Sample Rate
// ESP32 S2 Mini
#define BCK_PIN 4 // Clock pin from the mic.
#define WS_PIN 39 // WS pin from the mic.
#define DATA_PIN 5 // Data pin from the mic.
#define CHANNEL_SELECT_PIN 40 // Pin to select the channel output from the mic.
AudioInI2S mic(BCK_PIN, WS_PIN, DATA_PIN, CHANNEL_SELECT_PIN); // defaults to RIGHT channel.
int32_t samples[SAMPLE_SIZE]; // I2S sample data is stored here
void setup() {
Serial.begin(115200);
mic.begin(SAMPLE_SIZE, SAMPLE_RATE); // Starts the I2S DMA port.
}
void loop() {
mic.read(samples); // Stores the current I2S port buffer into samples.
// Send data to serial plotter
for(int i = 0; i < SAMPLE_SIZE; i++) {
Serial.println(samples[i]);
}
}The AudioAnalysis.h library is build on top of ArduinoFF2 V2 develop branch. You can find out more about it here: https://github.com/kosme/arduinoFFT/tree/develop
AudioAnalsis.h is not optimized and uses a lot of helper variables and floats. That said it is still very responsive at 1024 sample size and 44100 sample rate.
This library's audio processing would not be possible without ArduinoFFT https://github.com/kosme/arduinoFFT
MIT Open Source - Free to use anywhere.