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Re: Question about nyquist frequency

Date1999-01-29 13:14
FromJens Groh
SubjectRe: Question about nyquist frequency
Jonathan Drexler  wrote:
>I have been finding that when I generate a sawtooth wave in the 
>treble register (somewhere around an octave or more over 
>middle c) that I am getting some strong undertones as well.  
>I am using a sample rate of 44100.  I have tried it using 
>Gen7 and also Gen10.  
>
>Is this happening because some of the upper
>partials are running into  the Nyquist boundary?  Does 
>anyone know a practical way of handling this?  Would be 
>very grateful for help.
>

Hello list, hello Jonathan!
Yes - you get aliasing of upper harmonics.
You can use a wave table with fewer harmonics to circumvent the problem, but
then it will sound dull in the bass register.
A better, yet more 'expensive' solution is to 'filter at the source'. Here
is a _very_ primitive filter with three taps which are spaced at half the
sample period. The coefficients are {0.5,1.0,0.5}. It is realized by three
weighted sawteeth with appropriate phase shifts. (Note that the phasor
itself is used as a sawtooth generator here.)

ifreq 	=     	....
iampl 	=     	....
asaw0 	phasor	ifreq, 0.0
asaw1u	phasor	ifreq, 0.5 * ifreq / sr
asaw1d	phasor	ifreq, 1.0 - 0.5 * ifreq / sr
asaw  	=     	iampl * (asaw0 + 0.5 * (asaw1d + asaw1u))

A candidate for a new opcode?

Have fun!
Jens



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Date: Fri, 29 Jan 1999 08:20:36 -0500
From: Dave Phillips 
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References: <19990128164731.10F7E382B@toad.ilogic.com.au> <36B0954F.6A9A89EC@bright.net> <36B18AD4.D51576F9@agora.stm.it>
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Greetings:

> It is normal, because you have set kdur = p3, so moscil wait p3 seconds before 
> next
> note-on, but after p3 second the score note is deactivated. Try to set 
> kdur = p3/10

That fixed the problem, thank you.

So, the basic rule is: when using krate envelopes for knum do not use
kdur=p3, is that correct ?

Btw, some wild possibilities exist for realtime audio/MIDI output.
Thanks for the opcodes !

== Dave Phillips

       http://www.bright.net/~dlphilp/index.html
   http://www.bright.net/~dlphilp/linux_soundapps.html


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From: Michael Gogins 
To: Roger Klaveness , csound list 
Subject: Re: TB-303 like Lowpass-Filter
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I think this is a fantastic idea. It might also be interesting to create a
general-purpose filter design program in Java, so all platforms can use it,
that would format the filter coefficients into a line of Csound orc code for
filter2.

-----Original Message-----
From: Roger Klaveness 
To: csound list 
Date: Friday, January 29, 1999 4:30 AM
Subject: Re: TB-303 like Lowpass-Filter


>Josep M Comajuncosas wrote:
>>
>> Talking about general filter design routines... I think it would be a
nice idea to
>> implement a public domain filter design package (Are Meteor or the Pakrs
McLean
>> free?) in the Csound sources, to be able to design filters from Csounbd
itself as
>> needed in the filter2 opcode.
>
>I think that's a great idea.
>In that case one could also have an opcode which
>takes filter-spesifications as an argument, avoiding
>the need to first design filter in an utility-program
>an then using the coefficients with filter2.
>
>maybe something like this
>
>ar filtgen asig, iftype, ibandtype, iripple, iorder, icorner2, icorner2
>
>where
>iftype : filtertype 1=Butterworth,2=Chebychev,3=Bessel
>ibandtype : Bandtype 1=Lowpass , 2=Highpass, 3=Bandpass, 4=Bandstop
>iripple : ripple in dB for Chebychev filter
>iorder : order of filter
>icorner1 : cornerfrequency 1
>icorner2 = cornerfrequency 2, for Bandpass/Bandstop filters
>
>the opcode would then at inittime design the filter and maybe show
>a graphic output of freq ,phase and impulse response.
>
>at http://www.cs.york.ac.uk/~fisher/mkfilter/ there is an online
>interactive filter design page.
>The source code is available to download too, but I haven't found anything
>about license.
>If this is interesting one could off course ask the author if it's okay
>to use his sources in csound.
>
>Roger K



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Date: Fri, 29 Jan 1999 07:45:21 -0800
From: Erik Spjut 
Subject: Re: Question about nyquist frequency
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You can't actually filter out an aliased frequency unless you know the
frequency it's been aliased to. You have to generate the signal at a higher
sample rate, lowpass filter, and then resample at the final rate.

A better approach for Csound is to generate a series of function tables of
band-limited sawtooths (sawteeth?) with different limits and have the
instrument select the correct waveform according to the pitch of the note.
It's very fast at the cost of a little memory.

BTW to check for aliasing start with the frequency of the note, multipy it
by the number of the highest partial. If the result is greater than sr/2
you have aliasing. For example if you have:

F1 0 4096 10 1 0 1 4 1 -1 0 1 1

your highest harmonic is #9. If sr=44100, then sr/2=22050 so 9*pitch must
be less than 22050 or pitch (in cps) must be less than 2450 to prevent
aliasing.

At 2:14 PM +0100 1/29/99, Jens Groh wrote:
>Jonathan Drexler  wrote:
>>I have been finding that when I generate a sawtooth wave in the
>>treble register (somewhere around an octave or more over
>>middle c) that I am getting some strong undertones as well.
>>I am using a sample rate of 44100.  I have tried it using
>>Gen7 and also Gen10.
>>
>>Is this happening because some of the upper
>>partials are running into  the Nyquist boundary?  Does
>>anyone know a practical way of handling this?  Would be
>>very grateful for help.
>>
>
>Hello list, hello Jonathan!
>Yes - you get aliasing of upper harmonics.
>You can use a wave table with fewer harmonics to circumvent the problem, but
>then it will sound dull in the bass register.
>A better, yet more 'expensive' solution is to 'filter at the source'. Here
>is a _very_ primitive filter with three taps which are spaced at half the
>sample period. The coefficients are {0.5,1.0,0.5}. It is realized by three
>weighted sawteeth with appropriate phase shifts. (Note that the phasor
>itself is used as a sawtooth generator here.)
>
>ifreq 	=     	....
>iampl 	=     	....
>asaw0 	phasor	ifreq, 0.0
>asaw1u	phasor	ifreq, 0.5 * ifreq / sr
>asaw1d	phasor	ifreq, 1.0 - 0.5 * ifreq / sr
>asaw  	=     	iampl * (asaw0 + 0.5 * (asaw1d + asaw1u))
>
>A candidate for a new opcode?
>
>Have fun!
>Jens


-------------------------------------------------------------------------------
Erik Spjut (spyoot, rhymes with cute) - Associate Professor of Engineering
and  Associate Director for Engineering Computing,  Center for Design Education
Harvey Mudd College, Claremont, CA 91711-5990  USA
Erik_Spjut@hmc.edu      Ph & Voice mail (909) 607-3890      Fax (909) 621-8967




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Date: Fri, 29 Jan 1999 07:53:23 -0800
From: Erik Spjut 
Subject: Re: Filter Design (was Re: TB-303)
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As usual, the devil is in the details. I used the filter design page to
generate some of the coefficients I used to test filter2 for IIR filters.
The filters work fine in infinite-precision math but fail miserably in
Csound's single-precision math. I agree that the proposed opcode would be
most wonderful. However, whoever does it is in for some tedious work in
finite precision arithmetic.

At 9:28 AM -0500 1/29/99, Michael Gogins wrote:
>I think this is a fantastic idea. It might also be interesting to create a
>general-purpose filter design program in Java, so all platforms can use it,
>that would format the filter coefficients into a line of Csound orc code for
>filter2.
>
>[stuff deleted]
>>
>>maybe something like this
>>
>>ar filtgen asig, iftype, ibandtype, iripple, iorder, icorner2, icorner2
>>
>>where
>>iftype : filtertype 1=Butterworth,2=Chebychev,3=Bessel
>>ibandtype : Bandtype 1=Lowpass , 2=Highpass, 3=Bandpass, 4=Bandstop
>>iripple : ripple in dB for Chebychev filter
>>iorder : order of filter
>>icorner1 : cornerfrequency 1
>>icorner2 = cornerfrequency 2, for Bandpass/Bandstop filters
>>
>>the opcode would then at inittime design the filter and maybe show
>>a graphic output of freq ,phase and impulse response.
>>
>>at http://www.cs.york.ac.uk/~fisher/mkfilter/ there is an online
>>interactive filter design page.
>>The source code is available to download too, but I haven't found anything
>>about license.
>>If this is interesting one could off course ask the author if it's okay
>>to use his sources in csound.
>>
>>Roger K


-------------------------------------------------------------------------------
Erik Spjut (spyoot, rhymes with cute) - Associate Professor of Engineering
and  Associate Director for Engineering Computing,  Center for Design Education
Harvey Mudd College, Claremont, CA 91711-5990  USA
Erik_Spjut@hmc.edu      Ph & Voice mail (909) 607-3890      Fax (909) 621-8967




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Subject: Re: Filter Design (was Re: TB-303)
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Sorry excuse me for coming in half way through this conversation...

Another thing you could consider is the elliptical (IIR I think) filter
available in cmix - although the original code used a fortran coeff
algorithm there are C versions floating around (I can dig up a cleaned f2c
version is someone is interested in dealing with it). This filter allows lp,
bp or hp configurations with good time/accuracy trade-offs.

Ross.

>>>where
>>>iftype : filtertype 1=Butterworth,2=Chebychev,3=Bessel
>>>ibandtype : Bandtype 1=Lowpass , 2=Highpass, 3=Bandpass, 4=Bandstop
>>>iripple : ripple in dB for Chebychev filter
>>>iorder : order of filter
>>>icorner1 : cornerfrequency 1
>>>icorner2 = cornerfrequency 2, for Bandpass/Bandstop filters
>>>
>>>the opcode would then at inittime design the filter and maybe show
>>>a graphic output of freq ,phase and impulse response.
>>>
>>>at http://www.cs.york.ac.uk/~fisher/mkfilter/ there is an online
>>>interactive filter design page.
>>>The source code is available to download too, but I haven't found
anything
>>>about license.
>>>If this is interesting one could off course ask the author if it's okay
>>>to use his sources in csound.
>>>
>>>Roger K
>
>
>---------------------------------------------------------------------------
----
>Erik Spjut (spyoot, rhymes with cute) - Associate Professor of Engineering
>and  Associate Director for Engineering Computing,  Center for Design
Education
>Harvey Mudd College, Claremont, CA 91711-5990  USA
>Erik_Spjut@hmc.edu      Ph & Voice mail (909) 607-3890      Fax (909)
621-8967
>
>



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Hans Mikelson wrote:
> 
> Hi,
> 
> Well the site basically crashed/locked my SGI.  I had to telnet in from
> another system and kill Netscape to get out of it.  I recommend not going to
> that URL unless you want to see your system hang.

I had to do a Ctrl-Alt-Delete in Windows 98 and shut down Netscape in
order to get out of the page.  Why was this site recommended? Does it
have anything to do with Csound, if you have the proper
hardware/software setup to avoid crashing while viewing the page? Or is
it just some annoying "cyberpunk" who can't get attention by creating
something useful, and tries to get noticed by being as annoying as
possible?

Sean Costello

P.S.  On a completely unrelated topic, has anyone here constructed their
own allpass filters in Csound? I am trying to construct a reverb that
uses allpass filters with tapped delay lines, so I can't use the
standard alpass opcode. Any advice?



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To: csound@maths.ex.ac.uk
From: Jens Groh 
Subject: Re: Question about nyquist frequency
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Erik Spjut  wrote:
>You can't actually filter out an aliased unless you know the
>frequency it's been aliased to. You have to generate the signal at a higher
>sample rate, lowpass filter, and then resample at the final rate.

Look again at my code: That's what it does (with a sample rate of sr*2 and
the filter being far from perfect).

>A better approach for Csound is to generate a series of function tables of
>band-limited sawtooths (sawteeth?) with different limits and have the
>instrument select the correct waveform according to the pitch of the note.
>It's very fast at the cost of a little memory.

The better idea, I admit.

Jens



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From: Dave Perry 
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Greetings,

	Documentation for Visual Orchestra 1.95.4 is available for 
download from ,  at present this is in 
Word95 for Windows format - text version to follow.
	
	Also on the same page,  a zipfile containing some example 
projects I created or transcribed using VisOrc over the last few 
months.

	And probably of more interest to those on the VisOrc list,  the 
1.95.5 prerelease is also available from the rebuild's page.   Any 
feedback on this would be greatly appreciated.

Thanks,

Dave


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From: pete moss 
Organization: pete moss GmbH
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which has preference?  the command line parameters or the parameters set
up in CsOptions in a .csd file?  which one overrides the other?

pete



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From: Richard Dobson 
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It was a java bug (his main system is the Mac, and java conformance etc
is still widely inconsistent across platforms - especially for sound!).
The page is now down for revisions. I realized after the event I hadn't
tried ALT-F etc to see if the menus came up! Those who have heard
antiorp's music say it is pretty remarkable - I don't know, but it it
highly possible he uses Csound. It would be difficult to disagree with
the suggestion that he is experimental and non-conmformist. I am not
particularly interested in other sorts of speculations.

Re allpass filters in Csound: Hans Mikelson himself posted several a
while back (relating to nested allpasses for Gardner reverbs). Perhaps
it's time for a revival?

Richard Dobson

Sean Costello wrote:
> 
> Hans Mikelson wrote:
> >
> > Hi,
> >
> > Well the site basically crashed/locked my SGI.  I had to telnet in from
> > another system and kill Netscape to get out of it.  I recommend not going to
> > that URL unless you want to see your system hang.
> 
> I had to do a Ctrl-Alt-Delete in Windows 98 and shut down Netscape in
> order to get out of the page.  Why was this site recommended? Does it
> have anything to do with Csound, if you have the proper
> hardware/software setup to avoid crashing while viewing the page? Or is
> it just some annoying "cyberpunk" who can't get attention by creating
> something useful, and tries to get noticed by being as annoying as
> possible?
> 
> Sean Costello
> 
> P.S.  On a completely unrelated topic, has anyone here constructed their
> own allpass filters in Csound? I am trying to construct a reverb that
> uses allpass filters with tapped delay lines, so I can't use the
> standard alpass opcode. Any advice?

-- 
Test your DAW with my Soundcard Attrition Page!
http://wkweb5.cableinet.co.uk/rwd
CDP homepage: http://www.bath.ac.uk/~masjpf/CDP/CDP.htm


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Richard Dobson wrote:

> Re allpass filters in Csound: Hans Mikelson himself posted several a
> while back (relating to nested allpasses for Gardner reverbs). Perhaps
> it's time for a revival?

Thanks for the tip.  Mikelson's orchestra for the Gardner reverbs is
available at http://www.werewolf.net/~hljmm/csound/gardverb.orc. I was
actually trying to implement a reverb described in a Gardner article,
although the original source was in a JAES article by Joe Dattorro. The
cool aspect of the Dattorro reverb is that it uses interpolating taps in
the allpass delay lines, modulated by a low frequency (~1 Hz) sine wave.
The modulation of the delay lines results in dynamic alterations of the
resonant frequencies, to decreate tonal coloration.  So far, my reverb
has a tendancy to clip internally and/or blow up, so the Mikelson
orchestra should help in properly implementing the allpass stages.

BTW, has anyone on the list tried implementing any of the feedback delay
network (FDN) reverbs as described by Stautner and Puckette, Jot, Julius
Smith, etc.?  They seem promising.  I'll probably try to design such a
reverb this weekend, and will post it if and when I get it working.

Sean Costello


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From: Hans Mikelson 
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>Is this happening because some of the upper
>partials are running into  the Nyquist boundary?  Does 
>anyone know a practical way of handling this?  Would be 
>very grateful for help.


Try the vco opcode (>3.50) for bandlimited sawtooth waveforms.

Hans Mikelson



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Hi,

These are implemented in Gardner Reverbs on my web page:

http://www.werewolf.net/~hljmm/csound/

The nested all-pass filter opcode is in progress, it's taking me a while to
get around to it.  The form is something like:

aout  nestedap  ain, imaxtime, imode, itim1, igain1[, itim2, igain2, itim3,
igain3]

Hans Mikelson



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>From the Gardner descriptions, adding an (optional) pre-delay to the
allpass-filter input might also be worthwhile?

Richard Dobson

Hans Mikelson wrote:
> 
> Hi,
> 
> These are implemented in Gardner Reverbs on my web page:
> 
> http://www.werewolf.net/~hljmm/csound/
> 
> The nested all-pass filter opcode is in progress, it's taking me a while to
> get around to it.  The form is something like:
> 
> aout  nestedap  ain, imaxtime, imode, itim1, igain1[, itim2, igain2, itim3,
> igain3]
> 
> Hans Mikelson

-- 
Test your DAW with my Soundcard Attrition Page!
http://wkweb5.cableinet.co.uk/rwd
CDP homepage: http://www.bath.ac.uk/~masjpf/CDP/CDP.htm


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Thanks John...

jose halac wrote:
>
> I'm trying to use gen1 on Maldonado's CSound (WinSound). I installed it
> according instructions, set up a SSDir folder on the exe and still when I
> calculate a simple fog orchestra that involves a sound file to be
> processed, it give me the message "file not found" or that it can't find
> the sample directory at all...
>
> any help appreciatted
>
> JH

I am using Winsound. I did set up an environment in DOS and it created a
folder called Samples. Then I put a sample inside, open winsound and still
doesn't recognize the folder. Job van Zuijlen suggeted me to check if there
is a space after the = sign. I'll do that right now. I will keep you
posted. Thanks! JH




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From: Hans Mikelson 
To: csound@maths.ex.ac.uk
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Subject: Re: All-Pass filters
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>From the Gardner descriptions, adding an (optional) pre-delay to the
>allpass-filter input might also be worthwhile?


Why not just use delay?

Hans Mikelson



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> Roger Klaveness wrote:

(snip)
> ar filtgen asig, iftype, ibandtype, iripple, iorder, icorner2, icorner2
> 
> where
> iftype : filtertype 1=Butterworth,2=Chebychev,3=Bessel
> ibandtype : Bandtype 1=Lowpass , 2=Highpass, 3=Bandpass, 4=Bandstop
> iripple : ripple in dB for Chebychev filter
> iorder : order of filter
> icorner1 : cornerfrequency 1
> icorner2 = cornerfrequency 2, for Bandpass/Bandstop filters
(snip)

Would it make sense to allow the args to vary at k-rate ?
Clearly some args must be integers only, but it might open up the possibility
of  time-varying filter design, with all the goodies that implies -
conditional filters tested at k-rate, intermediate designs on a continuum
between 2 conventional designs, etc. 

Bob Douglas


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Date: Fri, 29 Jan 1999 21:08:26 -0800
From: Jonathan Drexler 
To: csound@maths.ex.ac.uk
Subject: Problem getting to csound ftp
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I tried to download the new version csound winhelp but 
kept getting an error saying that the directory does not 
exist.  

The url I am connecting to is this:
ftp://ftp.maths.bath.ac.uk/pub/dream/newest/

It is a subdirectory under this URL, containing the
1/99 version of winhelp,  where the
problem is  When I tried to go into the subdirectory 
I got the error.  I wondered if there might be a 
problem with the site?

Thanks.

                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                             


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Date: Fri, 29 Jan 1999 22:52:33 -0800 (PST)
From: Wayne Freno 
To: Jonathan Drexler 
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Subject: Re: Problem getting to csound ftp
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I also could not download csound_win.zip tonight from the
bath site , good thing the Montreal server is there:

ftp://ftp.musique.umontreal.ca/pub/mirrors/dream


On Fri, 29 Jan 1999, Jonathan Drexler wrote:

> I tried to download the new version csound winhelp but 
> kept getting an error saying that the directory does not 
> exist.  
> 
> The url I am connecting to is this:
> ftp://ftp.maths.bath.ac.uk/pub/dream/newest/
> 
> It is a subdirectory under this URL, containing the
> 1/99 version of winhelp,  where the
> problem is  When I tried to go into the subdirectory 
> I got the error.  I wondered if there might be a 
> problem with the site?
> 
> Thanks.
> 
>                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                                              
> 

end soundbite




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Another idea to try could be making the delay time variable...
I really need an allpass unit with variable delay time to easily
implement phaser effects without deltapi units.

Hans Mikelson wrote:

> >From the Gardner descriptions, adding an (optional) pre-delay to the
> >allpass-filter input might also be worthwhile?
>
> Why not just use delay?
>
> Hans Mikelson





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I was thinking about efficiency, and more flexilbility in ordering the
opcodes. And, as John might say, 'it helps the abstraction'.

Richard Dobson

Hans Mikelson wrote:
> 
> >From the Gardner descriptions, adding an (optional) pre-delay to the
> >allpass-filter input might also be worthwhile?
> 
> Why not just use delay?
> 
> Hans Mikelson

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Date1999-01-29 15:45
FromErik Spjut
SubjectRe: Question about nyquist frequency
You can't actually filter out an aliased frequency unless you know the
frequency it's been aliased to. You have to generate the signal at a higher
sample rate, lowpass filter, and then resample at the final rate.

A better approach for Csound is to generate a series of function tables of
band-limited sawtooths (sawteeth?) with different limits and have the
instrument select the correct waveform according to the pitch of the note.
It's very fast at the cost of a little memory.

BTW to check for aliasing start with the frequency of the note, multipy it
by the number of the highest partial. If the result is greater than sr/2
you have aliasing. For example if you have:

F1 0 4096 10 1 0 1 4 1 -1 0 1 1

your highest harmonic is #9. If sr=44100, then sr/2=22050 so 9*pitch must
be less than 22050 or pitch (in cps) must be less than 2450 to prevent
aliasing.

At 2:14 PM +0100 1/29/99, Jens Groh wrote:
>Jonathan Drexler  wrote:
>>I have been finding that when I generate a sawtooth wave in the
>>treble register (somewhere around an octave or more over
>>middle c) that I am getting some strong undertones as well.
>>I am using a sample rate of 44100.  I have tried it using
>>Gen7 and also Gen10.
>>
>>Is this happening because some of the upper
>>partials are running into  the Nyquist boundary?  Does
>>anyone know a practical way of handling this?  Would be
>>very grateful for help.
>>
>
>Hello list, hello Jonathan!
>Yes - you get aliasing of upper harmonics.
>You can use a wave table with fewer harmonics to circumvent the problem, but
>then it will sound dull in the bass register.
>A better, yet more 'expensive' solution is to 'filter at the source'. Here
>is a _very_ primitive filter with three taps which are spaced at half the
>sample period. The coefficients are {0.5,1.0,0.5}. It is realized by three
>weighted sawteeth with appropriate phase shifts. (Note that the phasor
>itself is used as a sawtooth generator here.)
>
>ifreq 	=     	....
>iampl 	=     	....
>asaw0 	phasor	ifreq, 0.0
>asaw1u	phasor	ifreq, 0.5 * ifreq / sr
>asaw1d	phasor	ifreq, 1.0 - 0.5 * ifreq / sr
>asaw  	=     	iampl * (asaw0 + 0.5 * (asaw1d + asaw1u))
>
>A candidate for a new opcode?
>
>Have fun!
>Jens


-------------------------------------------------------------------------------
Erik Spjut (spyoot, rhymes with cute) - Associate Professor of Engineering
and  Associate Director for Engineering Computing,  Center for Design Education
Harvey Mudd College, Claremont, CA 91711-5990  USA
Erik_Spjut@hmc.edu      Ph & Voice mail (909) 607-3890      Fax (909) 621-8967