Using LyraT to sample input, IIR functions to process, then output. The dsps_biquad_gen_xxx_f32 function creates garbage coefficients, I can't validate dsps_biquad_f32 as working. (DSP-111) about esp-dsp

[mrSilkie]

I should have spent a little more time investigating before putting this post up. Reducing the change in frequency to 0.001 i have been able to hear the filter working somewhat correctly as i go from f=0.5 and up.

It starts off with okay values (still a lot of distortion)

[filter-dsp] change f 0.69699: 0.69799
[filter-dsp] Coefficients 0: 12.48261
[filter-dsp] Coefficients 1: 24.96522
[filter-dsp] Coefficients 2: 12.48261
[filter-dsp] Coefficients 3: 12.13230
[filter-dsp] Coefficients 4: 36.79814

And as it increases, the values become greater and greater creating a decimation effect

[filter-dsp] change f 0.72899: 0.72999
[filter-dsp] Coefficients 0: 71.31686
[filter-dsp] Coefficients 1: 142.63373
[filter-dsp] Coefficients 2: 71.31686
[filter-dsp] Coefficients 3: 31.77969
[filter-dsp] Coefficients 4: 252.48778

These values continue to climb before i start seeing nan values.

[dmitry1945]

Hi @mrSilkie ,

Thank you for the feedback.
Yes, the frequency is from 0 up to 0.5, where 1 is a sample frequency. The values above 0.5 I this are not correct.
The LPF and HPF filters are very sensitive to the qFactor.

I've planed to release the demo application this week with LyraT board and processing audio. There will be just playing samples with volume/bass/treble.
You are right, first the samples from codec should be converted from int16 to float, and then back from float to int16.
We will implement biquad filter for the stereo samples soon (next few weeks).

Thank you,
Dmitry

[mrSilkie]

I got it working, I wasn't converting the int buffer to a float and back correctly.

Now that I've got it working all I can say is that it sounds like trash. A lot of digital distortion. It's getting tricky to debug the Lyra-T as SD card and Aux in both interfere with JTAG. I'll probably have to code in a file or oscillator to atleast have some test data for verification.

I have been using my button to sweep Q's and filter frequencies. Is the Q in log 10? so values 0.001 and 100 are both valid in some sense? Anyway, here is my code so far.

void int16_to_float_stereo(int16_t* input, float* output, int size, bool mode){
    switch (mode) {
        case true:
			for (int i = 0; i < size;i ++)
			{
				output[i] = input[(2*i)+1] / (float)INT16_MAX;
			}
			break;
        default:
			for (int i = 0; i < size; i++)
			{
				output[i] = input[2*i] / (float)INT16_MAX;
			}
			break;
    }
}

void float_to_int16_stereo(float* input, int16_t* output, int size, bool mode){
    switch (mode) {
        case true:
            for (int i = 1; i < size; i++) {
                output[i] = (int16_t)(input[(2*i)+1] * INT16_MAX);
            }
            break;
        default:
            for (int i = 0; i < size; i++) {
                output[i] = (int16_t)(input[2*i] * INT16_MAX);
            }
    }
}

// Function shows result of IIR filter
void IIRfilter(int16_t* input, int16_t* output, float freq, float qFactor, bool show, uint8_t channelMode)
{
    esp_err_t ret = ESP_OK;
    static float last_freq = -1.0;
    static float coeffs_lpf[5];

	static float filter_History_L[2] = {0};
	static float filter_History_R[2] = {0};

    float FloatDspBuf_IN[I2S_READLEN] = {0};
    float FloatDspBuf_OUT[I2S_READLEN] = {0};

	// INT TO FLOAT
	switch (channelMode)
	{
		case LEFT:
			int16_to_float_stereo(input, FloatDspBuf_IN, I2S_READLEN, 0);
			break;
		case RIGHT:
			int16_to_float_stereo(input, FloatDspBuf_IN, I2S_READLEN, 1);
			break;
		default:
			// no mono right now
			memcpy(output, input, I2S_READLEN);
			return;
	}
	
	// PRE DSP
	if (channelMode != LEFT && freq != last_freq)
	{
		printf("[filter-dsp] New F:%2.5f . . . Current Q: %2.3f\r\n", freq, qFactor);
		last_freq = freq;
		// Calculate iir filter coefficients
		ret = dsps_biquad_gen_hpf_f32(coeffs_lpf, (freq), qFactor);
		// printf all of the coefficients
		printf("[filter-dsp] Coefficients %1.5f\t%1.5f\t%1.5f\t%1.5f\t%1.5f\r\n", 
		coeffs_lpf[0],coeffs_lpf[1],coeffs_lpf[2],coeffs_lpf[3],coeffs_lpf[4]);
		if (ret  != ESP_OK)
		{
			ESP_LOGE(TAG, "Operation error = %i", ret);
			return;
		}
	}

	// DSP
	switch (channelMode)
	{
	case LEFT:
		ret = dsps_biquad_f32_ansi(FloatDspBuf_IN, FloatDspBuf_OUT, I2S_READLEN_HALF, coeffs_lpf, filter_History_L);
		break;
	case RIGHT:
		ret = dsps_biquad_f32_ansi(FloatDspBuf_IN, FloatDspBuf_OUT, I2S_READLEN_HALF, coeffs_lpf, filter_History_R);
		break;
	default:
		break;
	}
    if (ret  != ESP_OK)
    {
        ESP_LOGE(TAG, "Operation error = %i", ret);
        return;
    }

	// POST DSP
	// FLOAT TO INT
	switch (channelMode)
	{
		case LEFT:
			float_to_int16_stereo(FloatDspBuf_OUT, output, I2S_READLEN_HALF, 0);
			break;
		case RIGHT:
			float_to_int16_stereo(FloatDspBuf_OUT, output, I2S_READLEN_HALF, 1);
			break;
		default:
			memcpy(output, input, I2S_READLEN);
		break;
	}

	// 0 out memory
}