Efficient IIR filter bank design?

[drewandre]

Hi,

I was wondering if someone could point me in the right direction for designing an IIR-based filter bank. My goal is to recreate the MSGQ7's 7-band peak detection using the esp-dsp library.

I originally semi-successfully used this library's FFT function, but I couldn't get the resolution I needed in lower frequencies as I noted in this reddit post. I'm no expert in DSP, but a reddit user told me that a 44100fs/1024 point FFT yields a 20ms FFT frame window, and low-period frequencies (<200hz) would cause oscillations in the lower bins. This is exacerbated by spectral leakage.

As suggested in the reddit post, I moved to using this library's IIR filter. For each of the 7 MSGEQ7 bands (datasheet here) I calculated the coefficients using this handy online IIR biquad filter calculator. My "peak detector" logic for 63hz (which is just an average of all frequencies in a bandpass filter output) is as follows:

#define N_SAMPLES (2048)
#define N_SAMPLES_HALF (1024)

float *filter_output = (float *)calloc(N_SAMPLES_HALF, sizeof(float));
float *y_cf = (float *)calloc(N_SAMPLES, sizeof(float));

// x1 continuously updated from bluetooth a2dp i2s stream in my case
float *x1 = (float *)calloc(N_SAMPLES_HALF, sizeof(float));

float coeffs[5] = {
  0.0004485915993143372,
  0,
  -0.0004485915993143372,
  -1.9990222852850883,
  0.9991028168013714,
};

void calculate_average() {
  float w_lpf[5] = {0, 0};
  dsps_biquad_f32(x1, filter_output, N_SAMPLES_HALF, coeffs, w_lpf);

  for (int i = 0; i < N_SAMPLES_HALF; i++) {
    y_cf[i * 2 + 0] = filter_output[i];
    y_cf[i * 2 + 1] = 0;
  }

  dsps_fft2r_fc32(y_cf, N_SAMPLES_HALF);
  dsps_bit_rev_fc32(y_cf, N_SAMPLES_HALF);

  for (int i = 0; i < N_SAMPLES_HALF / 2; i++) {
    avg += (y_cf[i * 2 + 0] * y_cf[i * 2 + 0] + y_cf[i * 2 + 1] * y_cf[i * 2 + 1]) / N_SAMPLES_HALF;
  }
  avg /= N_SAMPLES_HALF;
  printf("%f\n", avg);
}

My question is (and maybe @dmitry1945 you could help me out): is there a more efficient way to do this? Ultimately to get 7 bands (or more), I would need to loop over this 7 times and calculate an IIR filter and FFT before averaging the FFT results. This seems horribly inefficient, but I just don't know enough about IIR filter bank design to know any better.

Alternatively, does anyone know how I could get better lower frequency resolution using the FFT alone? I really liked this solution (I had 1/3rd octave band and a-weighting logic running flawlessly) but I need low frequency resolution. I see that the max sampling rate of esp32 i2s is 48000 so I don't think adjusting this would help with frequency resolution. But is there anything else that I could adjust to get better lower frequency resolution in my FFT?

Any advice would be greatly appreciated! Thanks.

[dmitry1945]

Hi @drewandre For this purpose better if you will use IIR filter bank. You can look to the esp-dsp/examples/iir.

There you will find how to generate a single IIR filter: ret = dsps_biquad_gen_lpf_f32(coeffs_lpf, freq, qFactor);

The frequency is just: your_frequency/sample_frequency. In your case it will be 163/44100, and so on. You will have 7 filters. The Q factor you have to play with this parameter. In this example you can tune your filters.

At the output of your filters you make: average[i] = K*average[i] + (1 - K)*absf(output[i]);

Where: i - bank number from 0 to 6 K - smoothing coefficient 0.9..0.9999, 0.9999 - very smooth. output[i] - output of i'th filter. average[i] - result of your equalizer for i'th bank absf - module operation.

Regards, Dmitry

[drewandre]

@dmitry1945 thanks, this was very helpful!

[generic-beat-detector]

@dmitry1945

Hello sir. I applaud the effort behind this wonderful project!

I'd like some clarification please -- assuming the following for a single filter instance:

dsps_biquad_f32(x, y, N, coeffs_lpf, w_lpf);

How did the output y (array of floats) compute to output[i] (single float value)? I'm I missing something?

Regards.

[drewandre]

It has been a long time since I've worked on this project but I pulled up the function where I averaged the filter output. Hopefully this helps.

void calculate_iir_filter_bank() {
  float avg, x;
  for (int i = 0; i < NUM_AUDIO_ANALYSIS_BANDS; i++) { // NUM_AUDIO_ANALYSIS_BANDS was 7 in my case to match MSGEQ7
    avg = 0.0f;

    dsps_biquad_f32(buffer, filter_output, N_SAMPLES, filter_coefficients[i], filter_delay_lines[i]);

    for (int t = 0; t < N_SAMPLES; t++)
    {
      x = fabs(filter_output[t]) * 4.0f;
      avg += x;
    }
    avg /= (float)N_SAMPLES;

    avg *= filters_amp_adjustments[i];

    if (avg > 1) {
      avg = 1;
    }

    averages[i] = (SMOOTHING_FACTOR * averages[i]) + ((1 - SMOOTHING_FACTOR) * avg);
  }
}