new filter?

el`Ol · 21st September 2006, 04:53 PM

Hello!

Some time ago I made some considerations about filters and I found the following filter interesting:
A subharmonic series of cosine bursts with alternating sign:

sum(-(-1)^i/i cosburst(x/i),(i,1,infinity))
cosburst(x):=cos(x)+1 for (x,-pi,pi), else 0.

would give an infinite impulse response that could be clipped, preferably at zeros.

The response is not strictly periodical, the distance between the zeros increases towards the periphery.
The frequency domain of the stopband is aperiodical.

My questions: Is this new, and would this behaviour be an advantage compared to conventional filters?

Regards, Oliver

Nixie · 22nd September 2006, 02:01 AM

I like the Kaiser-Bessel filter.

el`Ol · 30th September 2007, 05:55 PM

Thread refresh.
Is an aperiodical frequency response in the stop band an advantage?

Eric Juaneda · 30th September 2007, 06:31 PM

Hi Olivier,

Are you able to give simple math formula of your filter like
Y =K1 * S1 + K2 * S2...
where Kx are constant coefficient and Sn, previous and next input sample.

I could insert this filter in my DAC Simulator to have response to your question.

Eric

el`Ol · 1st October 2007, 04:26 AM

Hello Eric!

I converted the (finite) impulse response into numerical form using Mathematica and used the numerical integral to calculate the frequency response. So I already know the filter has this aperiodical behaviour. What I am asking myself is: Does this have a sonical benefit in terms of less audible filter ringing?
But if your simulator has some useful features I can`t easyly implement in Mathematica I can calculate the filter coefficients.

el`Ol · 1st October 2007, 07:05 AM

Or is it more desirable to have zeros at multiples of the Nyquist frequency in order to get least possible aliasing at low frequencies?

Eric Juaneda · 1st October 2007, 06:16 PM

Hello Oliver,

Quote:

Originally posted by el`Ol
...But if your simulator has some useful features I can`t easyly implement in Mathematica I can calculate the filter coefficients.

With the simulator, you could see real output samples. Testing digital filter with various, non cyclic, input waveform is very interesting. Instant distortion is often greater the theory.

The simulator is written in C#, (µsoft .NET 2005) it is not and ended project, I add function to have wide comprehension of digital audio. If you are interested in, we could share this project. Final version will be free to download.

Eric

Eric Juaneda · 1st October 2007, 06:18 PM

On the actual DAC simulator, I implement a simplified digital filter like you could find in Wadia and of course standard Sin(x)/x filter.

el`Ol · 2nd October 2007, 04:43 AM

I will mail you the coefficients. How much oversampling? 8x? 16x?
Center on a sample or between two samples?

Eric Juaneda · 2nd October 2007, 10:31 AM

Hi Olivier,

Quote:

Originally posted by el`Ol
I will mail you the coefficients. How much oversampling? 8x? 16x?
Center on a sample or between two samples?

2x oversampling between two samples. I think, like over filters, to obtain 8x ovsl you cascade filters.

Eric

21st September 2006, 04:53 PM	#1
el`Ol diyAudio Member Join Date: May 2005 Location: Bavarian Forest	new filter? Hello! Some time ago I made some considerations about filters and I found the following filter interesting: A subharmonic series of cosine bursts with alternating sign: sum(-(-1)^i/i cosburst(x/i),(i,1,infinity)) cosburst(x):=cos(x)+1 for (x,-pi,pi), else 0. would give an infinite impulse response that could be clipped, preferably at zeros. The response is not strictly periodical, the distance between the zeros increases towards the periphery. The frequency domain of the stopband is aperiodical. My questions: Is this new, and would this behaviour be an advantage compared to conventional filters? Regards, Oliver

Thread Tools	Search this Thread
Show Printable Version Email this Page	Search this Thread: Advanced Search

Similar Threads
Thread	Thread Starter	Forum	Replies	Last Post
need help wana change 60-Hz twin-T notch filter to 279 Hz filter	prorms	Solid State	0	9th February 2008 11:38 PM
Can Zaphs notch filter be used without the BSC filter?	GuyPanico	Full Range	0	21st July 2007 01:27 AM
Just one big filter cap or more smaller filter caps ...	beppe61	Power Supplies	2	18th July 2006 05:46 PM
active filter from rod elliot and sub filter questions	wuffer	Parts	0	2nd October 2005 01:12 PM
chassis-mounted filter or IEC plug filter	jarthel	Solid State	7	11th March 2002 10:17 AM

New To Site?	Need Help?
Register to Participate Search Privacy Statement Contact Us	Frequently Asked Questions Did you forget your password? Mark Forums Read


	Please consider donating to help us continue to serve you. Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving

22nd September 2006, 02:01 AM	#2
Nixie Banned Join Date: Jan 2006 Location: Vancouver, BC	I like the Kaiser-Bessel filter.

30th September 2007, 05:55 PM	#3
el`Ol diyAudio Member Join Date: May 2005 Location: Bavarian Forest	Thread refresh. Is an aperiodical frequency response in the stop band an advantage?

1st October 2007, 04:26 AM	#5
el`Ol diyAudio Member Join Date: May 2005 Location: Bavarian Forest	Hello Eric! I converted the (finite) impulse response into numerical form using Mathematica and used the numerical integral to calculate the frequency response. So I already know the filter has this aperiodical behaviour. What I am asking myself is: Does this have a sonical benefit in terms of less audible filter ringing? But if your simulator has some useful features I can`t easyly implement in Mathematica I can calculate the filter coefficients.

1st October 2007, 07:05 AM	#6
el`Ol diyAudio Member Join Date: May 2005 Location: Bavarian Forest	Or is it more desirable to have zeros at multiples of the Nyquist frequency in order to get least possible aliasing at low frequencies?

1st October 2007, 06:18 PM	#8
Eric Juaneda diyAudio Member Join Date: Dec 2006 Location: Avignon, France	On the actual DAC simulator, I implement a simplified digital filter like you could find in Wadia and of course standard Sin(x)/x filter.

2nd October 2007, 04:43 AM	#9
el`Ol diyAudio Member Join Date: May 2005 Location: Bavarian Forest	I will mail you the coefficients. How much oversampling? 8x? 16x? Center on a sample or between two samples?

Currently Active Users Viewing This Thread: 1 (0 members and 1 guests)