What filter are used for Downsampling?

[Daxx]

Hi folks.

I'm using NAudio in WPF Project for downsampling some audiodata. I have to compare the results with matlab. But for that I need to know which filter is applied befor I can rebuild it on Matlab. I asked the guy from NAudio and he said that the resampler is only a port from WDL Resample. So my question is, can anyone tell me what kind of filter (Typ,Order) I have to use in Matlab?

Thx for any help

[Tale]

Well, the WDL resampler can do anything from simple linear resampling, to fast IIR filter resamplng, to HQ sinc resampling using up to a 8192 point FIR kernel at 8192x oversampling.

[Daxx]

Ok, thx. When I look at the sourcecode, I think NAudio only uses IIR Filters at the moment.

resampler = new WdlResampler();
resampler.SetMode(bool interp = true, int filtercnt = 2, bool sinc = false);


At ResampleOut-Method a WDL_Resampler_IIRFilter is created and filtercnt-times applied.

So that would be an IIR-Filter of order 2 right?


SetFilterParms(float filterpos = 0.693f, float filterq = 0.707f);
I don't know exactly how to use this parameters when designing the filter in Matlab. And I think, I also have to know the design-pattern like Chebyshev, or do I understand something wrong?

[Tale]

Originally Posted by Daxx

Ok, thx. When I look at the sourcecode, I think NAudio only uses IIR Filters at the moment.

resampler = new WdlResampler();
resampler.SetMode(bool interp = true, int filtercnt = 2, bool sinc = false);

Yeah, that means it uses linear interpolation with IIR filtering.

Originally Posted by Daxx

At ResampleOut-Method a WDL_Resampler_IIRFilter is created and filtercnt-times applied.

So that would be an IIR-Filter of order 2 right?

Each filter is order 2, so 2 filters would be order 4.

Originally Posted by Daxx

SetFilterParms(float filterpos = 0.693f, float filterq = 0.707f);
I don't know exactly how to use this parameters when designing the filter in Matlab. And I think, I also have to know the design-pattern like Chebyshev, or do I understand something wrong?

Well, I can't help you with Matlab, but I do know that the filter is a standard 2nd order low-pass filter as described here:

http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt

I think the cutoff frequency is at filterpos*srateout/2.

[Daxx]

Thx Tale, you helped me a lot.

I've managed to figure it out. Here is the code if someone is interested:

function [xDown] = WDL_DownSample(x,Fs,FsDown)

%% model filter like WDL-Resampler
m_filterpos = 0.693;
m_filterq = 0.707;

fpos = 1/ (Fs/FsDown) * m_filterpos;
pos = fpos * pi;
cpos = cos(pos);
spos = sin(pos);

alpha = spos / (2.0 * m_filterq);

sc = 1.0 / (1 + alpha);
m_b1 = (1 - cpos) * sc;
m_b0 = m_b1 * 0.5;
m_b2 = m_b1 * 0.5;
m_a1 = -2 * cpos * sc;
m_a2 = (1 - alpha) * sc;

num = [m_b0,m_b1,m_b2];
denom = [1,m_a1,m_a2];
sos = tf2sos(num,denom);

%% DownSampling
step = Fs / FsDown;

xFiltered = sosfilt(sos,x);
xFiltered = sosfilt(sos,xFiltered);

xDown = downsample(xFiltered,step);

end