TLV320AIC3268: Coefficient Loading, Part 2

Here is my code so far:

void SetLowPass(unsigned int freq)
{
	double Fs = 48000;					// Sample Frequency
	double f0 = freq;					// Corner frequency (provided input)
	double dBGain = 0;					// Gain applied to frequencies within range
	double Q = 0.707;					// Quality constant. 0.707
	double w0, alpha, sinW, cosW;		// Intermediate calculated parameters
	double b0, b1, b2, a0, a1, a2;		// Calculated BiQuad Coefficients
	double B0, B1, B2, A1, A2;			// Normalized BiQuad Coefficients
	double B0new, B1new, B2new;			// Scaled Coefficients so N0, N1, and N2 are less than 1
	double N0, N1, N2, D1, D2;			// Calculated Continuous Coefficients
	double Mx;							// Absolute Maximum Value of 3 Coefficients
	unsigned char NB = 16;				// Number of Databits of Coefficients
	unsigned char Range;
	short N0_1Dot15, N1_1Dot15, N2_1Dot15;	// N0..N2 Conversion to 1.15 Format
	short D1_1Dot15, D2_1Dot15;				// D1 and D2 Conversion to 1.15 Format

	// Calculate intermediate Parameters
	w0 = 2*PI*f0/Fs;
	cosW = cos(w0);
	sinW = sin(w0);
	alpha = sinW/(2*Q);

	// Calculate Low Pass Filter Coefficients
	b0 = pow(10,(dBGain/20))*((1 - cosW)/2);
	b1 = pow(10,(dBGain/20))*(1 - cosW);
	b2 = pow(10,(dBGain/20))*((1 - cosW)/2);
	a0 = 1 + alpha;
	a1 = -2*cosW;
	a2 = (1-alpha);

	// Normalize so that A0 = 1
	B0 = b0/a0;
	B1 = b1/(2*a0);
	B2 = b2/a0;
	A1 = a1/(-2*a0);
	A2 = a2/(-a0);

	// Scale Coefficients for Normalized largest of values is less than 1
	Mx = Maxf(B0, B1, B2);
	if (Mx > 1.0)
	{
		B0new = B0/Mx;
		B1new = B1/Mx;
		B2new = B2/Mx;
	} else
	{
		B0new = B0;
		B1new = B1;
		B2new = B2;
	}

	// Convert Scaled Continuous Coefficients to Quantized Coefficients
	Range = 2^(NB-1)-1;

	N0 = floor(B0new*Range);
	N1 = floor(B1new*Range);
	N2 = floor(B2new*Range);
	D1 = floor(A1*Range);
	D2 = floor(A2*Range);

	// Convert N0-D2 into 1.15 Format Integers (required data format TLV320AIC3268 DSP coefficients
	N0_1Dot15 = Floatto115(N0);
	N1_1Dot15 = Floatto115(N1);
	N2_1Dot15 = Floatto115(N2);
	D1_1Dot15 = Floatto115(D1);
	D2_1Dot15 = Floatto115(D2);

    // Now, upload these new coefficients into the DSP at the location of the Low Bandpass BiQuad Instance.

	// Begin by setting DSP into Adaptive Filtering for DAC (DAC won't accept the new changes otherwise)


}

Correction.

Line 54:  Range = pow(2, (NB-1)) - 1;

In MATLAB the ^ symbol raising x to the y power (exponentiation). In C, this is a logical bitwise XOR. Quite different. One must use the pow() function.

Incidentally, I get the proper low band filter coefficients when compared to the sample in SLAA447. My only question is how to know the register locations.  Since I used PPS to configure the BiQuad, I suspect I need to pay attention to where PPS says we should set the parameters. SLAA447 says these coefficients should be in 1.15 format. It looks from the PPS screenshot above that the coefficients are in 3.23 format. Is PPS back calculating the b0, b1, b2, a1, a2 values from the N0, N1, N2, D1, and D2 values uploaded?  Is this what I am seeing?

Isn't adaptive filtering already enabled in the TLV320AIC3268 since I am using the DAC?  Do I need to still set the adaptive filtering mode on Page 8?

If we are using Adaptive Filtering, shouldn't there be two sets of buffers?  What is the address of the second butter?

Just noticed this discrepancy.  See image below. One is document SLAA447 and the other is SLAS953A (TLV320AIC3268 datasheet. The transfer function in the datasheet is using a 24-bit, 3-Byte data word; more like a 1.23 data format.