add some code
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idf_component_register(SRCS "dsps_fir_main.c")
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/*
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* SPDX-FileCopyrightText: 2022 Espressif Systems (Shanghai) CO LTD
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*
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* SPDX-License-Identifier: Apache-2.0
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include "freertos/FreeRTOS.h"
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#include "freertos/task.h"
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#include "esp_system.h"
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#include "driver/spi_master.h"
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#include "soc/gpio_struct.h"
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#include "driver/gpio.h"
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#include "driver/uart.h"
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#include "soc/uart_struct.h"
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#include <math.h>
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#include <malloc.h>
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#include "esp_dsp.h"
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static const char *TAG = "main";
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// This example shows how to use FIR filters from esp-dsp library
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#define FIR_COEFFS_LEN 64 // Number of FIR filter coefficients
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#define DECIMATION 2 // Decimation ratio of the FIR filter
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#define N_SAMPLES 1024 // Input samples
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#define FIR_DELAY (FIR_COEFFS_LEN / DECIMATION) // Amount of samples not being considered for the FFT
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#define FIR_BUFF_OUT_LEN (N_SAMPLES + FIR_DELAY) // Total length of samples
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// Function shows the result of the FIR filter
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void show_FFT(float *input_signal, const unsigned int fft_len)
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{
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dsps_fft2r_fc32(input_signal, fft_len >> 1);
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dsps_bit_rev2r_fc32(input_signal, fft_len >> 1);
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dsps_cplx2real_fc32(input_signal, fft_len >> 1);
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// Correction factor for the FFT spectrum
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const float correction_factor = fft_len * 3;
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// Calculating power of spectrum in dB
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for (int i = 0 ; i < fft_len / 2 ; i++) {
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input_signal[i] = 10 * log10f((input_signal[i * 2 + 0] * input_signal[i * 2 + 0] + input_signal[i * 2 + 1] * input_signal[i * 2 + 1] + 0.0000001) / correction_factor);
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}
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// Display power spectrum
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dsps_view(input_signal, fft_len / 2, 64, 10, -120, 40, '|');
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}
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// Generate Windowed-Sinc filter coefficients
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void generate_FIR_coefficients(float *fir_coeffs, const unsigned int fir_len, const float ft)
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{
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// Even or odd length of the FIR filter
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const bool is_odd = (fir_len % 2) ? (true) : (false);
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const float fir_order = (float)(fir_len - 1);
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// Window coefficients
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float *fir_window = (float *)malloc(fir_len * sizeof(float));
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dsps_wind_blackman_f32(fir_window, fir_len);
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for (int i = 0; i < fir_len; i++) {
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if ((i == fir_order / 2) && (is_odd)) {
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fir_coeffs[i] = 2 * ft;
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} else {
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fir_coeffs[i] = sinf((2 * M_PI * ft * (i - fir_order / 2))) / (M_PI * (i - fir_order / 2));
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}
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fir_coeffs[i] *= fir_window[i];
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}
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free(fir_window);
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}
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static __attribute__((aligned(16))) float tone_combined[FIR_BUFF_OUT_LEN];
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static __attribute__((aligned(16))) float fir_coeffs[FIR_COEFFS_LEN];
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static __attribute__((aligned(16))) float delay_line[FIR_COEFFS_LEN];
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void app_main()
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{
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const int32_t fir_len = FIR_COEFFS_LEN;
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const float fir_ft = 0.5 / DECIMATION; // Transition frequency of the FIR filter
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const int32_t N = N_SAMPLES; // Number of input samples
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const int32_t fir_decim = DECIMATION; // FIR filter decimation
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const int32_t N_buff = FIR_BUFF_OUT_LEN; // Total length of samples with ignored
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fir_f32_t fir1;
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esp_err_t ret;
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// Ignoring the first set of samples, due to the delay line of the FIR filter
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const int fir_out_offset = ((FIR_DELAY / 2) - 1);
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ESP_LOGI(TAG, "Start Example.");
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// If a user doesn't care about buffer allocation, the default
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// initialization could be used as shown here:
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ret = dsps_fft2r_init_fc32(NULL, CONFIG_DSP_MAX_FFT_SIZE);
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if (ret != ESP_OK) {
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ESP_LOGE(TAG, "Not possible to initialize FFT. Error = %i", ret);
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return;
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}
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ret = dsps_fft4r_init_fc32(NULL, CONFIG_DSP_MAX_FFT_SIZE);
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if (ret != ESP_OK) {
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ESP_LOGE(TAG, "Not possible to initialize FFT. Error = %i", ret);
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return;
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}
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// Generate input signal as 2 sine waves
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float *tone_1 = (float *)malloc(N_buff * sizeof(float));
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float *tone_2 = (float *)malloc(N_buff * sizeof(float));
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dsps_tone_gen_f32(tone_1, N_buff, 5, 0.2, 0);
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dsps_tone_gen_f32(tone_2, N_buff, 5, 0.4, 0);
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// Generate windowing coefficients
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float *window = (float *)malloc(N * sizeof(float));
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dsps_wind_blackman_harris_f32(window, N);
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// Add the two waves together
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for (int i = 0 ; i < N_buff ; i++) {
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tone_combined[i] = tone_1[i] + tone_2[i];
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}
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free(tone_1);
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free(tone_2);
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// Apply the windowing
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for (int i = 0 ; i < N ; i++) {
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window[i] *= tone_combined[i];
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}
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// Show FFT spectrum
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show_FFT(window, N);
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// Calculate coefficients for the FIR filter
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generate_FIR_coefficients(fir_coeffs, fir_len, fir_ft);
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ESP_LOGI(TAG, "\n");
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// Filter the input signal with FIR filter
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float *fir_out = (float *)malloc( N_buff * sizeof(float));
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dsps_fird_init_f32(&fir1, fir_coeffs, delay_line, fir_len, fir_decim);
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uint32_t start_b = dsp_get_cpu_cycle_count();
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dsps_fird_f32(&fir1, tone_combined, fir_out, N_buff / fir_decim);
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uint32_t end_b = dsp_get_cpu_cycle_count();
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// Generate windowing coefficients and apply the windowing
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dsps_wind_blackman_harris_f32(window, (N / fir_decim));
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for (int i = 0 ; i < N / fir_decim ; i++) {
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window[i] *= fir_out[fir_out_offset + i];
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}
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// Show FFT spectrum, ignoring first samples from the delay line
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show_FFT(window, N / fir_decim);
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ESP_LOGI(TAG, "FIR for %"PRId32" samples and decimation %"PRId32" takes %"PRId32" cycles", N, fir_decim, (int32_t)(end_b - start_b));
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ESP_LOGI(TAG, "End Example.");
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free(fir_out);
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free(window);
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}
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@@ -0,0 +1,4 @@
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dependencies:
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espressif/esp-dsp:
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override_path: "../../../"
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version: "*"
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