How to change order for different filters?

Hi,

I have 2 questions:

• What order filter does y = highpass(x,fpass,fs) and y = lowpass(x,fpass,fs) use? How can I check the filter order what they use? I was using "filtord()" in this manner to find out the filter order, is this correct way?

[y2,fs]=audioread('room.mp3');
y_h2=highpass(y2,1000,fs);
n=filtord(y_h2);

• I have an audio file and I am doing signal processing over that audio file such a passing it though various filters. But I wanted to change order of those filters at my own will. How can I do that?

Thanks.

Answer 1

[Star Strider]

Those functions have a second output, the digitalFilter object. Return that and all the information you want about the type of filter and the order will be available.

Example —

s = rand(1, 5000);
Fs = 1000;
fpass = 50;
[sfilt,df] = lowpass(s, fpass, Fs, 'ImpulseResponse','iir');
df

df = 
 digitalFilter with properties:

           Coefficients: [3×6 double]

   Specifications:
      FrequencyResponse: 'lowpass'
        ImpulseResponse: 'iir'
             SampleRate: 1000
         PassbandRipple: 0.1000
    StopbandAttenuation: 60
      PassbandFrequency: 50
      StopbandFrequency: 120.6500
           DesignMethod: 'ellip'

 Use fvtool to visualize filter
 Use designfilt to edit filter
 Use filter to filter data

.

[Giggs B.]

Thank you. I don't know about this..so I will look into it. Other than that, how can I find the properties of y = highpass(x,fpass,fs) or y = lowpass(x,fpass,fs), like if it is an IIR or FIR filter and what order and all?

[Star Strider]

To get information about the filter, just print it out as I did. It’ll tell you many of those details, as it did here.

Note —

 Use fvtool to visualize filter 
 Use designfilt to edit filter 
 Use filter to filter data

The only exception I have to that is to use it with filtfilt, not filter, if you want to use it to filter other signals.

.

[Giggs B.]

Thanks for the reply. I got it, but the information that is printed doesn't mentions about order.

b = 

 digitalFilter with properties:

           Coefficients: [1×5093 double]

   Specifications:
      FrequencyResponse: 'highpass'
        ImpulseResponse: 'fir'
             SampleRate: 44100
      PassbandFrequency: 200
    StopbandAttenuation: 60
      StopbandFrequency: 168.6000
         PassbandRipple: 0.1000
           DesignMethod: 'kaiserwin'

 Use fvtool to visualize filter
 Use designfilt to edit filter
 Use filter to filter data

Also, I didn't get by what you mean "The only exception I have to that is to use it with filtfilt, not filter, if you want to use it to filter other signals". I used this, is this wrong?

[y,fs]=audioread('fr3.mp3');
% y_b=bandpass(y,[2000 9000],fs);
%[y_h,b]=highpass(y,200,fs);
y_h=filter(b,y)
b

[Star Strider]

It does, actually. Get that information from the coefficients, and for a more detailed explanation, use designfilt to examine its characteristics and understand its performance.

[Giggs B.]

@Star Strider So, when I open the coefficients array...since it's an array its only numbers...how can I derive the order? Also, I used "designfilter" and under the option of "order mode"...when I select it to "specify", it automatically shows "20". So is this the order that is used or it's just default? Because other than that, order mode is selected to "minimum". Thanks.

[Star Strider]

Use designfilt to edit the filter to change its order, if that is what you want to do.

To make this easier, use fvtool instead to find the filter order. Under ‘Analysis’ choose ‘Pole-Zero Plot’ annd count the ‘x’ displayed. Those are the poles, and the filter order are the number of poles.

You can also get this information indirectly from the ‘Coefficients’ displayed in the ‘df’ output. Filters are described by the number of poles the filter has, that being the filter order.

[Giggs B.]

@Star Strider Thank you so much for the explanation.

[Star Strider]

As always, my pleasure!

[Giggs B.]

@Star Strider Sorry, but looks like I am doing some mistake here and can't figure out why this high pass isn't working. Can you help me? I am giving 800 Hz as cut-off frequency.

%Sound 1
[y1,fs]=audioread('ambulance.wav');
designfilt(b)
y_h1=filter(b,y1);

%Spectrum plot
referenceLength=length(y1);
fref=fs*(1:referenceLength)/referenceLength;
R=fft(y_h1,referenceLength);
stor=R(1);
R(1)=0;
absR=abs(R);
plot(fref,absR);
xlim([0 2000]);
ylim([0 200]); 
grid on;
xlabel('Frequency(Hz)');
ylabel('Amplitude');

[Star Strider]

‘I am giving 800 Hz as cut-off frequency.’

That’s not my impression. The stopband frequency is 168.6 Hz and the passband frequency is 800 Hz, giving a 631.4 Hz transition region. Everything below the stopband frequency will not be passed, and everything above the passband frequency will be pased without change. Everything in the transition rebion will be variably attenuated.

I can’t run the posted code (‘b’ is not defined), however I doubt that’s necessary.

[Giggs B.]

Got it, I was able to use the filters this way. Thanks :)

[Star Strider]

As always, my pleasure!

.