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21st April 2004, 02:54 PM  #1 
diyAudio Member

IIR filters type 1 or 2 – what is best?
Hello,
I’m making a dsp project at the moment. I was wondering what type of filter signal graph is the better one – regarding truncating the numbers in internal registers of the DSP processor. The problem is that with the type 1 signal graph you can implement the difference equation in one go, keeping the numbers in the 32 bit register of the CPU right until the time they are sent to the DA converter. This however I find hard to do with the type 2 signal graph, since it needs 2 difference equations to work. My question is simply, witch signal graph is the better one for precision. Theoretically they describe the same system, but in real life there must be a superior type? Thanks \Jens 
21st April 2004, 03:21 PM  #2 
diyAudio Member
Join Date: May 2003
Location: Norway

Type1 is superior with respect to quantization errors at least with typical lowpass/highpass filters. However it needs more memory and more cycles than a type2 implementation.

21st April 2004, 07:47 PM  #3 
diyAudio Member

Ok thanks, That's what i figured just needed to be sure.
I'll go for the type one equations then.... What about limit cycle oscillation btw? witch one would be better (Suffer less) \Jens 
21st April 2004, 08:25 PM  #4  
diyAudio Member
Join Date: May 2003
Location: Norway

Quote:


21st April 2004, 09:49 PM  #5 
diyAudio Member

limit cycle oscilation
Hello,
I may have this up side down, but I understand limit cycle oscilation as the main factor that the IIR (infinite inpulse response) is called what it is. I belive it to be the noice generated by the algorithme thus making an oscillation because some of the truncation noise is feed back into the summoning point via the a1 and a2 coeffecients. \Jens 
21st April 2004, 09:54 PM  #6 
diyAudio Member

It can look like this with a step of ½ input.
\Jens 
21st April 2004, 10:54 PM  #7 
diyAudio Member
Join Date: May 2003
Location: Norway

Your first graph is correct, but the plot does not look too good! You are correct that the Infinite in IIR is because of feedback through a1 and a2, but not because of selfoscillation due to quantization noise. When a1,a2 and y is quantized this may raise the noisefloor, but should not create oscillation.
Are you using fixed or floating point? How many bits? What are your coefficients? If your a1 and a2 are very small, you might want to do the a1*y(n1)a2*y(n2) in double precision. 
22nd April 2004, 10:35 AM  #8 
diyAudio Member

Hello,
I have not implemented the signal graph on a platfor yet. All the work so fars is done on a PC in a mathprogram. I't a 6'th order system with high Q making a bandstop filter. The freq response is as planned, and it also stops the wanted freq when I sweep it. so it schould be ok i think. \Jens 
22nd April 2004, 05:45 PM  #9  
diyAudio Member
Join Date: Aug 2003
Location: Santa Cruz, California

Quote:
If you're using more than one Type I second order filter, you can use the denominator delays of one section as the numerator delays of the following section. Francois. 

22nd April 2004, 06:00 PM  #10 
diyAudio Member

Hello Francois,
I'm note sure that I understand, can you explain it a bit further please? \Jens  not a DSP geek yet, but hope to be 
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