Hello! Im currently working on an SPL meter, i found this project made by Ivan Kostoski. He uses an ESP32, well i would like to adapted and use an ESP8266, with the same sensor an INMP441. From what i understand, the problem is that the ESP32 code is made to work in a 2 core boar, whereas the ESP8266 has only 1 core. What should i change to make it work?
Apreciate the help
The code is:
#include <sos-iir-filter.h>// agregada desde el githubb conpleto
/*
* Display A-weighted sound level measured by I2S Microphone
*
* (c)2019 Ivan Kostoski
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <https://www.gnu.org/licenses/>.
*/
/*
* Sketch samples audio data from I2S microphone, processes the data
* with digital IIR filters and calculates A or C weighted Equivalent
* Continuous Sound Level (Leq)
*
* I2S is setup to sample data at Fs=48000KHz (fixed value due to
* design of digital IIR filters). Data is read from I2S queue
* in 'sample blocks' (default 125ms block, equal to 6000 samples)
* by 'i2s_reader_task', filtered trough two IIR filters (equalizer
* and weighting), summed up and pushed into 'samples_queue' as
* sum of squares of filtered samples. The main task then pulls data
* from the queue and calculates decibel value relative to microphone
* reference amplitude, derived from datasheet sensitivity dBFS
* value, number of bits in I2S data, and the reference value for
* which the sensitivity is specified (typically 94dB, pure sine
* wave at 1KHz).
*
* Displays line on the small OLED screen with 'short' LAeq(125ms)
* response and numeric LAeq(1sec) dB value from the signal RMS.
*/
//#include <I2S.h>
//#include "sos-iir-filter.h"
#include <driver/i2s.h>
//
// Configuration
//
#define LEQ_PERIOD 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 1
// NOTE: Some microphones require at least DC-Blocker filter
#define MIC_EQUALIZER ICS43434 // 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 29 // 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 18
#define I2S_SCK 23
#define I2S_SD 19
// I2S peripheral to use (0 or 1)
#define I2S_PORT I2S_NUM_0
//
// Setup your display library (and geometry) here
//
#if (USE_DISPLAY > 0)
// ThingPulse/esp8266-oled-ssd1306, you may need the latest source and PR#198 for 64x48
#include <SSD1306Wire.h>
#define OLED_GEOMETRY GEOMETRY_64_48
//#define OLED_GEOMETRY GEOMETRY_128_32
//#define OLED_GEOMETRY GEOMETRY_128_64
#define OLED_FLIP_V 1
SSD1306Wire display(0x3c, SDA, SCL, OLED_GEOMETRY);
#endif
//
// 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);
}
}
//
// Setup and main loop
//
// Note: Use doubles, not floats, here unless you want to pin
// the task to whichever core it happens to run on at the moment
//
void setup() {
// If needed, now you can actually lower the CPU frquency,
// i.e. if you want to (slightly) reduce ESP32 power consumption
setCpuFrequencyMhz(80); // It should run as low as 80MHz
Serial.begin(112500);
delay(1000); // Safety
#if (USE_DISPLAY > 0)
display.init();
#if (OLED_FLIP_V > 0)
display.flipScreenVertically();
#endif
display.setFont(ArialMT_Plain_16);
#endif
// 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
Serial.printf("%.1f\n", Leq_dB);
// Debug only
//Serial.printf("%u processing ticks\n", q.proc_ticks);
}
#if (USE_DISPLAY > 0)
//
// Example code that displays the measured value.
// You should customize the below code for your display
// and display library used.
//
display.clear();
// It is important to somehow notify when the deivce is out of its range
// as the calculated values are very likely with big error
if (Leq_dB > MIC_OVERLOAD_DB) {
// Display 'Overload' if dB value is over the AOP
display.drawString(0, 24, "Overload");
} else if (isnan(Leq_dB) || (Leq_dB < MIC_NOISE_DB)) {
// Display 'Low' if dB value is below noise floor
display.drawString(0, 24, "Low");
}
// The 'short' Leq line
double short_Leq_dB = MIC_OFFSET_DB + MIC_REF_DB + 20 * log10(sqrt(double(q.sum_sqr_weighted) / SAMPLES_SHORT) / MIC_REF_AMPL);
uint16_t len = min(max(0, int(((short_Leq_dB - MIC_NOISE_DB) / MIC_OVERLOAD_DB) * (display.getWidth()-1))), display.getWidth()-1);
display.drawHorizontalLine(0, 0, len);
display.drawHorizontalLine(0, 1, len);
display.drawHorizontalLine(0, 2, len);
// The Leq numeric decibels
display.drawString(0, 4, String(Leq_dB, 1) + " " + DB_UNITS);
display.display();
#endif // USE_DISPLAY
}
}
void loop() {
// Nothing here..
}