I am now working on simplifying data collection for my students for the accelerometer and microphone, which are a bit tricky as they have a time duration.
The Accelerometer does not seem too bad, in the EdgeImpulse Arduino Library examples. It looks fairly easy, using a buffer to sequence the incoming data. I might change it around a bit using an array which my students are more familiar with. (At this point understanding is more important than speed)
Basic Acceleration Algorithm:
- grab x,y,z samples, convert to accelerations (*9.8)
- wait correct number of milliseconds
- repeat until correct number of samples
- convert samples to a numpy array for classification
// Allocate a buffer here for the values we'll read from the IMU
float buffer[EI_CLASSIFIER_DSP_INPUT_FRAME_SIZE] = { 0 };
for (size_t ix = 0; ix < EI_CLASSIFIER_DSP_INPUT_FRAME_SIZE; ix += 3) {
// Determine the next tick (and then sleep later)
uint64_t next_tick = micros() + (EI_CLASSIFIER_INTERVAL_MS * 1000);
IMU.readAcceleration(buffer[ix], buffer[ix + 1], buffer[ix + 2]);
buffer[ix + 0] *= CONVERT_G_TO_MS2;
buffer[ix + 1] *= CONVERT_G_TO_MS2;
buffer[ix + 2] *= CONVERT_G_TO_MS2;
delayMicroseconds(next_tick - micros());
}
// Turn the raw buffer in a signal which we can the classify
signal_t signal;
int err = numpy::signal_from_buffer(buffer, EI_CLASSIFIER_DSP_INPUT_FRAME_SIZE, &signal);
if (err != 0) {
ei_printf("Failed to create signal from buffer (%d)\n", err);
return;
}
The concern comes with the audio samples which the EdgeImpulse Arduino Library examples are quiet complex using several functions to organize the more complex audio information.
Does anyone have a basic algorithm for what is happening with the audio samples?
In the PDMSerialPlotter it doesn’t seem too confusing
/*
This example reads audio data from the on-board PDM microphones, and prints
out the samples to the Serial console. The Serial Plotter built into the
Arduino IDE can be used to plot the audio data (Tools -> Serial Plotter)
Circuit:
- Arduino Nano 33 BLE board
This example code is in the public domain.
*/
#include <PDM.h>
// buffer to read samples into, each sample is 16-bits
short sampleBuffer[256];
// number of samples read
volatile int samplesRead;
void setup() {
Serial.begin(9600);
while (!Serial);
// configure the data receive callback
PDM.onReceive(onPDMdata);
// optionally set the gain, defaults to 20
// PDM.setGain(30);
// initialize PDM with:
// - one channel (mono mode)
// - a 16 kHz sample rate
if (!PDM.begin(1, 16000)) {
Serial.println("Failed to start PDM!");
while (1);
}
}
void loop() {
// wait for samples to be read
if (samplesRead) {
// print samples to the serial monitor or plotter
for (int i = 0; i < samplesRead; i++) {
Serial.println(sampleBuffer[i]);
}
// clear the read count
samplesRead = 0;
}
}
void onPDMdata() {
// query the number of bytes available
int bytesAvailable = PDM.available();
// read into the sample buffer
PDM.read(sampleBuffer, bytesAvailable);
// 16-bit, 2 bytes per sample
samplesRead = bytesAvailable / 2;
}
but when I look at the EdgeImpulse Arduino microphone Library I get lost
/* Edge Impulse Arduino examples
* Copyright (c) 2020 EdgeImpulse Inc.
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
// If your target is limited in memory remove this macro to save 10K RAM
#define EIDSP_QUANTIZE_FILTERBANK 0
/* Includes ---------------------------------------------------------------- */
#include <PDM.h>
#include <words_inference.h>
/** Audio buffers, pointers and selectors */
typedef struct {
int16_t *buffer;
uint8_t buf_ready;
uint32_t buf_count;
uint32_t n_samples;
} inference_t;
static inference_t inference;
static signed short sampleBuffer[2048];
static bool debug_nn = false; // Set this to true to see e.g. features generated from the raw signal
/**
* @brief Arduino setup function
*/
void setup()
{
// put your setup code here, to run once:
Serial.begin(115200);
Serial.println("Edge Impulse Inferencing Demo");
// summary of inferencing settings (from model_metadata.h)
ei_printf("Inferencing settings:\n");
ei_printf("\tInterval: %.2f ms.\n", (float)EI_CLASSIFIER_INTERVAL_MS);
ei_printf("\tFrame size: %d\n", EI_CLASSIFIER_DSP_INPUT_FRAME_SIZE);
ei_printf("\tSample length: %d ms.\n", EI_CLASSIFIER_RAW_SAMPLE_COUNT / 16);
ei_printf("\tNo. of classes: %d\n", sizeof(ei_classifier_inferencing_categories) / sizeof(ei_classifier_inferencing_categories[0]));
if (microphone_inference_start(EI_CLASSIFIER_RAW_SAMPLE_COUNT) == false) {
ei_printf("ERR: Failed to setup audio sampling\r\n");
return;
}
}
/**
* @brief Arduino main function. Runs the inferencing loop.
*/
void loop()
{
ei_printf("Starting inferencing in 2 seconds...\n");
delay(2000);
ei_printf("Recording...\n");
bool m = microphone_inference_record();
if (!m) {
ei_printf("ERR: Failed to record audio...\n");
return;
}
ei_printf("Recording done\n");
signal_t signal;
signal.total_length = EI_CLASSIFIER_RAW_SAMPLE_COUNT;
signal.get_data = µphone_audio_signal_get_data;
ei_impulse_result_t result = { 0 };
EI_IMPULSE_ERROR r = run_classifier(&signal, &result, debug_nn);
if (r != EI_IMPULSE_OK) {
ei_printf("ERR: Failed to run classifier (%d)\n", r);
return;
}
// print the predictions
ei_printf("Predictions ");
ei_printf("(DSP: %d ms., Classification: %d ms., Anomaly: %d ms.)",
result.timing.dsp, result.timing.classification, result.timing.anomaly);
ei_printf(": \n");
for (size_t ix = 0; ix < EI_CLASSIFIER_LABEL_COUNT; ix++) {
ei_printf(" %s: %.5f\n", result.classification[ix].label, result.classification[ix].value);
}
#if EI_CLASSIFIER_HAS_ANOMALY == 1
ei_printf(" anomaly score: %.3f\n", result.anomaly);
#endif
}
/**
* @brief Printf function uses vsnprintf and output using Arduino Serial
*
* @param[in] format Variable argument list
*/
void ei_printf(const char *format, ...) {
static char print_buf[1024] = { 0 };
va_list args;
va_start(args, format);
int r = vsnprintf(print_buf, sizeof(print_buf), format, args);
va_end(args);
if (r > 0) {
Serial.write(print_buf);
}
}
/**
* @brief PDM buffer full callback
* Get data and call audio thread callback
*/
static void pdm_data_ready_inference_callback(void)
{
int bytesAvailable = PDM.available();
// read into the sample buffer
int bytesRead = PDM.read((char *)&sampleBuffer[0], bytesAvailable);
if (inference.buf_ready == 0) {
for(int i = 0; i < bytesRead>>1; i++) {
inference.buffer[inference.buf_count++] = sampleBuffer[i];
if(inference.buf_count >= inference.n_samples) {
inference.buf_count = 0;
inference.buf_ready = 1;
break;
}
}
}
}
/**
* @brief Init inferencing struct and setup/start PDM
*
* @param[in] n_samples The n samples
*
* @return { description_of_the_return_value }
*/
static bool microphone_inference_start(uint32_t n_samples)
{
inference.buffer = (int16_t *)malloc(n_samples * sizeof(int16_t));
if(inference.buffer == NULL) {
return false;
}
inference.buf_count = 0;
inference.n_samples = n_samples;
inference.buf_ready = 0;
// configure the data receive callback
PDM.onReceive(&pdm_data_ready_inference_callback);
// optionally set the gain, defaults to 20
PDM.setGain(80);
PDM.setBufferSize(4096);
// initialize PDM with:
// - one channel (mono mode)
// - a 16 kHz sample rate
if (!PDM.begin(1, EI_CLASSIFIER_FREQUENCY)) {
ei_printf("Failed to start PDM!");
microphone_inference_end();
return false;
}
return true;
}
/**
* @brief Wait on new data
*
* @return True when finished
*/
static bool microphone_inference_record(void)
{
inference.buf_ready = 0;
inference.buf_count = 0;
while(inference.buf_ready == 0) {
delay(10);
}
return true;
}
/**
* Get raw audio signal data
*/
static int microphone_audio_signal_get_data(size_t offset, size_t length, float *out_ptr)
{
numpy::int16_to_float(&inference.buffer[offset], out_ptr, length);
return 0;
}
/**
* @brief Stop PDM and release buffers
*/
static void microphone_inference_end(void)
{
PDM.end();
free(inference.buffer);
}
#if !defined(EI_CLASSIFIER_SENSOR) || EI_CLASSIFIER_SENSOR != EI_CLASSIFIER_SENSOR_MICROPHONE
#error "Invalid model for current sensor."
#endif
Anyone want to make a stab at the steps for a basic sound recording algorithm going on here?