esp32-i2s-slm
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Stuck reading Led_dB
Hello there! Thank you so much for this project, its amaizing!
I've been trying to incorporate the code to a multisensor I'm creating but when the ESP reads Serial.printf("%.1f\n", Leq_dB); Serial.println("dB");
it gets stucks reading forever and won't go back to the loop (see attached picture)
any thoughts?
extra: what does ("%.1f\n", Leq_dB); means? I tried reading a Serial.print(Leq_dB); but i get an error
my code is:
void readNoise() {
// Create FreeRTOS queue samples_queue = xQueueCreate(8, sizeof(sum_queue_t));
// Create the I2S reader FreeRTOS task // NOTE: Current version of ESP-IDF will pin the task // automatically to the first core it happens to run on // (due to using the hardware FPU instructions). // For manual control see: xTaskCreatePinnedToCore xTaskCreate(mic_i2s_reader_task, "Mic I2S Reader", I2S_TASK_STACK, NULL, I2S_TASK_PRI, NULL);
sum_queue_t q; uint32_t Leq_samples = 0; double Leq_sum_sqr = 0; double Leq_dB = 0;
// Read sum of samaples, calculated by 'i2s_reader_task' while (xQueueReceive(samples_queue, &q, portMAX_DELAY)) {
// Calculate dB values relative to MIC_REF_AMPL and adjust for microphone reference
double short_RMS = sqrt(double(q.sum_sqr_SPL) / SAMPLES_SHORT);
double short_SPL_dB = MIC_OFFSET_DB + MIC_REF_DB + 20 * log10(short_RMS / MIC_REF_AMPL);
// In case of acoustic overload or below noise floor measurement, report infinty Leq value
if (short_SPL_dB > MIC_OVERLOAD_DB) {
Leq_sum_sqr = INFINITY;
} else if (isnan(short_SPL_dB) || (short_SPL_dB < MIC_NOISE_DB)) {
Leq_sum_sqr = -INFINITY;
}
// Accumulate Leq sum
Leq_sum_sqr += q.sum_sqr_weighted;
Leq_samples += SAMPLES_SHORT;
// When we gather enough samples, calculate new Leq value
if (Leq_samples >= SAMPLE_RATE * LEQ_PERIOD) {
double Leq_RMS = sqrt(Leq_sum_sqr / Leq_samples);
Leq_dB = MIC_OFFSET_DB + MIC_REF_DB + 20 * log10(Leq_RMS / MIC_REF_AMPL);
Leq_sum_sqr = 0;
Leq_samples = 0;
// Serial output, customize (or remove) as needed
noise = ("%.1f\n", Leq_dB);
Serial.print("Decibeles: ");
Serial.printf("%.1f\n", Leq_dB);
Serial.println("dB");
// Debug only
//Serial.printf("%u processing ticks\n", q.proc_ticks);
}
}
}
thanks!!!
@Xpcker
Serial.printf("%.1f\n", Leq_dB);
Serial.println("dB");
prints a variable to the serial output. You can simply remove or comment those lines.
The code you posted does not compile. The line noise = ("%.1f\n", Leq_dB);
is lacking Serial.print
probably...
thanks! i managed to fix it !
@Xpcker Good to hear! It makes sense to post how you fixed it so others can possibly solve similar problems. You could then also close your issue I guess...
I'm still working on the project and I'll upload everything, for some reason the esp32 Wifi crashes with the XQueueCreate in the FREERTOS, I'm trying to sync it with a neotimer with no luck, I'll make another issue with that
For the fix, I made a function void readNoise() and I just call it from the loop, The code below is in a readNoise.h file so the code is cleaner
below is the code: @HorstBaerbel =)
#include <driver/i2s.h> #include "sos_IRR_Filter.h"
float noise;
// // Configuration MIC //
#define LEQ_PERIOD 0.1 // second(s) #define WEIGHTING C_weighting // Also avaliable: 'C_weighting' or 'None' (Z_weighting) #define LEQ_UNITS "LAeq" // customize based on above weighting used #define DB_UNITS "dBA" // customize based on above weighting used #define USE_DISPLAY 0
// NOTE: Some microphones require at least DC-Blocker filter #define MIC_EQUALIZER INMP441 // See below for defined IIR filters or set to 'None' to disable #define MIC_OFFSET_DB 3.0103 // Default offset (sine-wave RMS vs. dBFS). Modify this value for linear calibration
// Customize these values from microphone datasheet #define MIC_SENSITIVITY -26 // dBFS value expected at MIC_REF_DB (Sensitivity value from datasheet) #define MIC_REF_DB 94.0 // Value at which point sensitivity is specified in datasheet (dB) #define MIC_OVERLOAD_DB 116.0 // dB - Acoustic overload point #define MIC_NOISE_DB 33 // dB - Noise floor #define MIC_BITS 24 // valid number of bits in I2S data #define MIC_CONVERT(s) (s >> (SAMPLE_BITS - MIC_BITS)) #define MIC_TIMING_SHIFT 0 // Set to one to fix MSB timing for some microphones, i.e. SPH0645LM4H-x
// Calculate reference amplitude value at compile time constexpr double MIC_REF_AMPL = pow(10, double(MIC_SENSITIVITY)/20) * ((1<<(MIC_BITS-1))-1);
// // I2S pins - Can be routed to almost any (unused) ESP32 pin. // SD can be any pin, inlcuding input only pins (36-39). // SCK (i.e. BCLK) and WS (i.e. L/R CLK) must be output capable pins // // Below ones are just example for my board layout, put here the pins you will use // #define I2S_WS 19 #define I2S_SCK 18 #define I2S_SD 23
// I2S peripheral to use (0 or 1) #define I2S_PORT I2S_NUM_0
// // IIR Filters //
// DC-Blocker filter - removes DC component from I2S data // See: https://www.dsprelated.com/freebooks/filters/DC_Blocker.html // a1 = -0.9992 should heavily attenuate frequencies below 10Hz SOS_IIR_Filter DC_BLOCKER = { gain: 1.0, sos: {{-1.0, 0.0, +0.9992, 0}} };
// // Equalizer IIR filters to flatten microphone frequency response // See respective .m file for filter design. Fs = 48Khz. // // Filters are represented as Second-Order Sections cascade with assumption // that b0 and a0 are equal to 1.0 and 'gain' is applied at the last step // B and A coefficients were transformed with GNU Octave: // [sos, gain] = tf2sos(B, A) // See: https://www.dsprelated.com/freebooks/filters/Series_Second_Order_Sections.html // NOTE: SOS matrix 'a1' and 'a2' coefficients are negatives of tf2sos output //
// TDK/InvenSense INMP441 // Datasheet: https://www.invensense.com/wp-content/uploads/2015/02/INMP441.pdf // B ~= [1.00198, -1.99085, 0.98892] // A ~= [1.0, -1.99518, 0.99518] SOS_IIR_Filter INMP441 = { gain: 1.00197834654696, sos: { // Second-Order Sections {b1, b2, -a1, -a2} {-1.986920458344451, +0.986963226946616, +1.995178510504166, -0.995184322194091} } };
// // Weighting filters //
// // A-weighting IIR Filter, Fs = 48KHz // (By Dr. Matt L., Source: https://dsp.stackexchange.com/a/36122) // B = [0.169994948147430, 0.280415310498794, -1.120574766348363, 0.131562559965936, 0.974153561246036, -0.282740857326553, -0.152810756202003] // A = [1.0, -2.12979364760736134, 0.42996125885751674, 1.62132698199721426, -0.96669962900852902, 0.00121015844426781, 0.04400300696788968] SOS_IIR_Filter A_weighting = { gain: 0.169994948147430, sos: { // Second-Order Sections {b1, b2, -a1, -a2} {-2.00026996133106, +1.00027056142719, -1.060868438509278, -0.163987445885926}, {+4.35912384203144, +3.09120265783884, +1.208419926363593, -0.273166998428332}, {-0.70930303489759, -0.29071868393580, +1.982242159753048, -0.982298594928989} } };
// // C-weighting IIR Filter, Fs = 48KHz // Designed by invfreqz curve-fitting, see respective .m file // B = [-0.49164716933714026, 0.14844753846498662, 0.74117815661529129, -0.03281878334039314, -0.29709276192593875, -0.06442545322197900, -0.00364152725482682] // A = [1.0, -1.0325358998928318, -0.9524000181023488, 0.8936404694728326 0.2256286147169398 -0.1499917107550188, 0.0156718181681081] SOS_IIR_Filter C_weighting = { gain: -0.491647169337140, sos: { {+1.4604385758204708, +0.5275070373815286, +1.9946144559930252, -0.9946217070140883}, {+0.2376222404939509, +0.0140411206016894, -1.3396585608422749, -0.4421457807694559}, {-2.0000000000000000, +1.0000000000000000, +0.3775800047420818, -0.0356365756680430} } };
// // Sampling // #define SAMPLE_RATE 48000 // Hz, fixed to design of IIR filters #define SAMPLE_BITS 32 // bits #define SAMPLE_T int32_t #define SAMPLES_SHORT (SAMPLE_RATE / 8) // ~125ms #define SAMPLES_LEQ (SAMPLE_RATE * LEQ_PERIOD) #define DMA_BANK_SIZE (SAMPLES_SHORT / 16) #define DMA_BANKS 32
// Data we push to 'samples_queue' struct sum_queue_t { // Sum of squares of mic samples, after Equalizer filter float sum_sqr_SPL; // Sum of squares of weighted mic samples float sum_sqr_weighted; // Debug only, FreeRTOS ticks we spent processing the I2S data uint32_t proc_ticks; }; QueueHandle_t samples_queue;
// Static buffer for block of samples float samples[SAMPLES_SHORT] attribute((aligned(4)));
//
// I2S Microphone sampling setup
//
void mic_i2s_init() {
// Setup I2S to sample mono channel for SAMPLE_RATE * SAMPLE_BITS
// NOTE: Recent update to Arduino_esp32 (1.0.2 -> 1.0.3)
// seems to have swapped ONLY_LEFT and ONLY_RIGHT channels
const i2s_config_t i2s_config = {
mode: i2s_mode_t(I2S_MODE_MASTER | I2S_MODE_RX),
sample_rate: SAMPLE_RATE,
bits_per_sample: i2s_bits_per_sample_t(SAMPLE_BITS),
channel_format: I2S_CHANNEL_FMT_ONLY_LEFT,
communication_format: i2s_comm_format_t(I2S_COMM_FORMAT_I2S | I2S_COMM_FORMAT_I2S_MSB),
intr_alloc_flags: ESP_INTR_FLAG_LEVEL1,
dma_buf_count: DMA_BANKS,
dma_buf_len: DMA_BANK_SIZE,
use_apll: true,
tx_desc_auto_clear: false,
fixed_mclk: 0
};
// I2S pin mapping
const i2s_pin_config_t pin_config = {
bck_io_num: I2S_SCK,
ws_io_num: I2S_WS,
data_out_num: -1, // not used
data_in_num: I2S_SD
};
i2s_driver_install(I2S_PORT, &i2s_config, 0, NULL);
#if (MIC_TIMING_SHIFT > 0)
// Undocumented (?!) manipulation of I2S peripheral registers
// to fix MSB timing issues with some I2S microphones
REG_SET_BIT(I2S_TIMING_REG(I2S_PORT), BIT(9));
REG_SET_BIT(I2S_CONF_REG(I2S_PORT), I2S_RX_MSB_SHIFT);
#endif
i2s_set_pin(I2S_PORT, &pin_config);
//FIXME: There is a known issue with esp-idf and sampling rates, see:
// https://github.com/espressif/esp-idf/issues/2634
// In the meantime, the below line seems to set sampling rate at ~47999.992Hz
// fifs_req=24576000, sdm0=149, sdm1=212, sdm2=5, odir=2 -> fifs_reached=24575996
//NOTE: This seems to be fixed in ESP32 Arduino 1.0.4, esp-idf 3.2
// Should be safe to remove...
//#include <soc/rtc.h>
//rtc_clk_apll_enable(1, 149, 212, 5, 2);
}
// // I2S Reader Task // // Rationale for separate task reading I2S is that IIR filter // processing cam be scheduled to different core on the ESP32 // while main task can do something else, like update the // display in the example // // As this is intended to run as separate hihg-priority task, // we only do the minimum required work with the I2S data // until it is 'compressed' into sum of squares // // FreeRTOS priority and stack size (in 32-bit words) #define I2S_TASK_PRI 4 #define I2S_TASK_STACK 2048 // void mic_i2s_reader_task(void* parameter) { mic_i2s_init();
// Discard first block, microphone may have startup time (i.e. INMP441 up to 83ms) size_t bytes_read = 0; i2s_read(I2S_PORT, &samples, SAMPLES_SHORT * sizeof(int32_t), &bytes_read, portMAX_DELAY);
while (true) { // Block and wait for microphone values from I2S // // Data is moved from DMA buffers to our 'samples' buffer by the driver ISR // and when there is requested ammount of data, task is unblocked // // Note: i2s_read does not care it is writing in float[] buffer, it will write // integer values to the given address, as received from the hardware peripheral. i2s_read(I2S_PORT, &samples, SAMPLES_SHORT * sizeof(SAMPLE_T), &bytes_read, portMAX_DELAY);
TickType_t start_tick = xTaskGetTickCount();
// Convert (including shifting) integer microphone values to floats,
// using the same buffer (assumed sample size is same as size of float),
// to save a bit of memory
SAMPLE_T* int_samples = (SAMPLE_T*)&samples;
for(int i=0; i<SAMPLES_SHORT; i++) samples[i] = MIC_CONVERT(int_samples[i]);
sum_queue_t q;
// Apply equalization and calculate Z-weighted sum of squares,
// writes filtered samples back to the same buffer.
q.sum_sqr_SPL = MIC_EQUALIZER.filter(samples, samples, SAMPLES_SHORT);
// Apply weighting and calucate weigthed sum of squares
q.sum_sqr_weighted = WEIGHTING.filter(samples, samples, SAMPLES_SHORT);
// Debug only. Ticks we spent filtering and summing block of I2S data
q.proc_ticks = xTaskGetTickCount() - start_tick;
// Send the sums to FreeRTOS queue where main task will pick them up
// and further calcualte decibel values (division, logarithms, etc...)
xQueueSend(samples_queue, &q, portMAX_DELAY);
} }
void readNoise() {
// Create FreeRTOS queue samples_queue = xQueueCreate(8, sizeof(sum_queue_t));
// Create the I2S reader FreeRTOS task // NOTE: Current version of ESP-IDF will pin the task // automatically to the first core it happens to run on // (due to using the hardware FPU instructions). // For manual control see: xTaskCreatePinnedToCore xTaskCreate(mic_i2s_reader_task, "Mic I2S Reader", I2S_TASK_STACK, NULL, I2S_TASK_PRI, NULL);
sum_queue_t q; uint32_t Leq_samples = 0; double Leq_sum_sqr = 0; double Leq_dB = 0;
// Read sum of samaples, calculated by 'i2s_reader_task' if (xQueueReceive(samples_queue, &q, portMAX_DELAY)) {
// Calculate dB values relative to MIC_REF_AMPL and adjust for microphone reference
double short_RMS = sqrt(double(q.sum_sqr_SPL) / SAMPLES_SHORT);
double short_SPL_dB = MIC_OFFSET_DB + MIC_REF_DB + 20 * log10(short_RMS / MIC_REF_AMPL);
// In case of acoustic overload or below noise floor measurement, report infinty Leq value
if (short_SPL_dB > MIC_OVERLOAD_DB) {
Leq_sum_sqr = INFINITY;
} else if (isnan(short_SPL_dB) || (short_SPL_dB < MIC_NOISE_DB)) {
Leq_sum_sqr = -INFINITY;
}
// Accumulate Leq sum
Leq_sum_sqr += q.sum_sqr_weighted;
Leq_samples += SAMPLES_SHORT;
// When we gather enough samples, calculate new Leq value
if (Leq_samples >= SAMPLE_RATE * LEQ_PERIOD) {
double Leq_RMS = sqrt(Leq_sum_sqr / Leq_samples);
Leq_dB = MIC_OFFSET_DB + MIC_REF_DB + 20 * log10(Leq_RMS / MIC_REF_AMPL);
Leq_sum_sqr = 0;
Leq_samples = 0;
noise = ("%.1f\n", Leq_dB);
Serial.print("Decibels: ");
Serial.print(Leq_dB);
Serial.println(" dB");
}
}
}
Try a smaller number of queues:
samples_queue = xQueueCreate(3, sizeof(sum_queue_t));
And it would be nice if you could format your source code using the "insert code" button. Makes it much more readable.
You could also close the issue yourself, if you got it fixed...
Hey buddy i have the same issue, but when i call it from loop as you said so i get flash reset problem with esp32, is it possible if you could share your code. it would be so helpful