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Re: Frequency Shifting & Hilbert Transforms

Date1999-03-18 14:53
FromJens Groh
SubjectRe: Frequency Shifting & Hilbert Transforms
Roger Klaveness  wrote:

>Will this do the trick ?
[......]
>
> ; Hilbert transform, 11-order FIR
> igain =     1.570796327
>
> axv0 init 0
> axv1 init 0
> axv2 init 0
> axv3 init 0
> axv4 init 0
> axv5 init 0
> axv6 init 0
> axv7 init 0
> axv8 init 0
> axv9 init 0
> axv10 init 0
>
> axv0 = axv1
> axv1 = axv2
> axv2 = axv3
> axv3 = axv4
> axv4 = axv5
> axv5 = axv6
> axv6 = axv7
> axv7 = axv8
> axv8 = axv9
> axv9 = axv10
> axv10 = ain/igain;
>
> ahilb = .0160000000 * (axv0 - axv10) + .1326173942 * (axv2 - axv8) + .9121478174 * (axv4 - axv6)
> ; ------------------------------------------------------------------
>
> amod1 = ahilb*asinmod
> amod2 = ain*acosmod
>
> outs amod2-amod1,amod1+amod2
>
[......]

Hi Roger!
Probably it will, if you replace your delay line with one that has its taps at every sample, not every ksmps samples. Try it with the delay1 opcode. Or try to use the filter2 opcode which should be suitable to implement a FIR filter, too.
Jens Groh



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Date: Thu, 18 Mar 1999 10:39:34 -0600 (CST)
From: Tesch David 
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Subject: off the subject
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sorry to get off the subject, but has anyone heard that lame ORGY tune
that Joy Division did a long time ago?  I think it's called Blue Monday.
anyway, the guitar tone is really cool.  I think it sounds like the
engineer processed the soundfile in Soundhack spectral compressor.  i
inevitably had to try it (im not a guitar player-what do i know) but did
not get even similiar results.  ANYONE KNOW?

On the subject>  I have built two compressors in Csound (for soundfiles)
but they are rms type compressors.  the rms command is not suitable for
limiting, however.  i have done some research and not found any command
for finding the instantaniuos peak value on a sample by sample basis.  i
like the way digital compression sounds, as i have built a orc+sco that
sets all sample values to the same value (except it's RMS value so there
is still some amplitude variation) 

anyone's help would be appreciated.

-Tracy Love 



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Date: Thu, 18 Mar 1999 11:38:38 -0800
From: Sean Costello 
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Sergey Batov wrote:

> Here are two examples of Karplus Strong Algorithm (from List).
> 
> It seems that the pitches of notes  are good enough in first example ,
> but not in second example. Why?
> In the beginning I thought that this is because of LPF y[n]=(x[n]+x[n-1])/2
> in the second example, but using the 'tone' brings worst results.
> 
> May be the pair "delayr - delayw" works more exactly than "delay"?

Try using deltapi in conjunction with a delayr/delayw pair.  delay will
round the pitch off to the nearest integer sample number, which is bad
for pitch accuracy at higher pitches.  deltapi will interpolate between
samples to get a more exact pitch, although the slight amount of delay
of the LPF in the feedback loop will probably add a slight bit of pitch
change.  You should also note that the linear interpolation in deltapi
will act very close to the LPF you describe in your example, so the tone
and sustain of pitches will probably vary in some unpredictable ways.

Sean Costello


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Date: Thu, 18 Mar 1999 11:42:04 -0800
From: Sean Costello 
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To: Jens Groh 
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Jens Groh wrote:

> P.S. Dear Sean Costello, would you like to tell us what you want to use the frequency shifter for?

Frequency Shifting!! It's just a beautiful sound, plain and simple. I've
heard it in Hyperprism, and it sounds beautiful, especially on voices.
Plus, some nice paradoxical sounds can be generated with feedback - i.e.
a phasing sound that goes in one direction only.  Lots of stuff I want
to explore.  I'll be testing out the algorithms people posted early next
week - just finished up some compositions for class last week, so ran
out of time.  

Thanks for all the input on this one, folks!  I've got some nice stuff
to try out.  I also have a friend at CCRMA who wrote a frequency shifter
in CLM, so I might try to get that from him and translate it into Csound
or C.  I'll keep people posted.

Sean Costello


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From: rasmus ekman 
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Tesch David wrote:
> 
> i have done some research and not found any command
> for finding the instantaniuos peak value on a sample by sample basis. 

You may be looking for peak/peakk. 
Not sure if they're in the main docs.


	re


********************

kpeak	peakk	ksig
kpeak	peak	asig

These opcodes maintain the output k rate variable as the peak absolute
level so far received. 

peakk takes k-rate inputs and peak takes a-rate inputs.


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From: J P Fitch 
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Subject:  peakk/peak
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Is there any reason why these are not both called peak as the discrimination on
teh first argument is easy.
==John
> 
> i have done some research and not found any command
> for finding the instantaniuos peak value on a sample by sample basis. 

You may be looking for peak/peakk. 
Not sure if they're in the main docs.


        re



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J P Fitch wrote:
> 
> Is there any reason why [peak/peakk] are not both called peak 

Not at all in my opinion. Quite a few of the more recent
opcodes have been unnecessarily split by type.

Opcode writers, look at the pile of existing ugens to see how to
make your code handle i/a/k-rate arguments transparently
to the end-users!

(and of course, thanks for all your hard work!!)


Cheers,

	r the enduser


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We now have an upgrade of the XTCsound card drivers.  To download, visit:
http://www.geocities.com/researchtriangle/thinktank/2138/xtc.html

thanks for your patience,
Scotty Vercoe
XTCsound Applications Consultant
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We now have an upgrade of the XTCsound card drivers.  To download, visit:
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thanks for your patience,
Scotty Vercoe
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are anyone of you csounders out there going to be at seamus99 next week
in san jose?  i will be there.  i have a piece called 'descent' on the
first concert at 1:30 on friday.  look for me.  it would be nice to meet
some of my fellow csound users face to face.

pete



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Date: Fri, 19 Mar 1999 09:37:56 +0100
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>Jens Groh wrote:
> 
> Hi Roger!
> Probably it will, if you replace your delay line with one that has its taps at every sample,
?? 
I'm using a-rate for the filter
 
>not every ksmps samples. Try it with the delay1 opcode. Or try to use the filter2 opcode which 
>should be suitable to implement a FIR filter, too.
> Jens Groh

I have had difficulties with filter2 and zfilter2.
Sometimes they just doesen't work ( coeffs generated 
by matlab and http://www.cs.york.ac.uk/~fisher/mkfilter )
That was with some Butterworth IIR-filters, I didn't 
try them out with the hilbert-FIR though.

Anyway I think it is funny to implement a filter directly 
in csound-code ;-)


> 
> Roger Klaveness  wrote:
> 
> >Will this do the trick ?
> [......]
> >
> > ; Hilbert transform, 11-order FIR
> > igain =     1.570796327
> >
> > axv0 init 0
> > axv1 init 0
> > axv2 init 0
> > axv3 init 0
> > axv4 init 0
> > axv5 init 0
> > axv6 init 0
> > axv7 init 0
> > axv8 init 0
> > axv9 init 0
> > axv10 init 0
> >
> > axv0 = axv1
> > axv1 = axv2
> > axv2 = axv3
> > axv3 = axv4
> > axv4 = axv5
> > axv5 = axv6
> > axv6 = axv7
> > axv7 = axv8
> > axv8 = axv9
> > axv9 = axv10
> > axv10 = ain/igain;
> >
> > ahilb = .0160000000 * (axv0 - axv10) + .1326173942 * (axv2 - axv8) + .9121478174 * (axv4 - axv6)
> > ; ------------------------------------------------------------------
> >
> > amod1 = ahilb*asinmod
> > amod2 = ain*acosmod
> >
> > outs amod2-amod1,amod1+amod2
> >
> [......]


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To: csound@maths.ex.ac.uk
From: Jens Groh 
Subject: Re: Frequency Shifting & Hilbert Transforms
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> > > axv0 = axv1
> > > axv1 = axv2
> > > axv2 = axv3
> > > axv3 = axv4
> > > axv4 = axv5
> > > axv5 = axv6
> > > axv6 = axv7
> > > axv7 = axv8
> > > axv8 = axv9
> > > axv9 = axv10
> > > axv10 = ain/igain;
> >
> > [...] replace your delay line with one that has its taps at every sample,
> > not every ksmps samples. Try it with the delay1 opcode. 
>  
> ?? 
> I'm using a-rate for the filter
>  

Roger, I'm afraid there is a big misunderstanding! (Hope it's not on my side.)
The a-rate operations in Csound code are _not_ executed every a-rate cycle.
They are executed every k-rate cycle, but apply to a _vector_ of values.
(The dimension of this vector is ksmps.)

Jens



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From: Sergey Batov 
To: csound@maths.ex.ac.uk
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Subject: simple LPF
Date: Fri, 19 Mar 1999 13:15:04 +0300
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Hi,
sorry for "loop" but I'd like to ask it again:
is it possible to know characteristics  (cutoff frequency,
e.t.c.) of the LPF:

a3 delay1 a2
afeedback=(a2+a3)*.5 ?

Thank you,
Sergey Batov   batov@glasnet.ru


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From: Sergey Batov 
To: csound@maths.ex.ac.uk
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Subject: parameters in K-S?
Date: Fri, 19 Mar 1999 12:34:10 +0300
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Dear Csounders!
I'd like to know: is it possible to change "place of plucking" 
and time of decay in a simplest form of Karplus-Strong algorithm 
("cycling" delay line + LPF)?

Thanks in advance,
Sergey Batov   batov@glasnet.ru


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To: Sergey Batov 
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Subject: Re: parameters in K-S?
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Sergey Batov wrote:

> Dear Csounders!
> I'd like to know: is it possible to change "place of plucking"
> and time of decay in a simplest form of Karplus-Strong algorithm
> ("cycling" delay line + LPF)?

Yes, simply prefilter the excitation signal with a FIR comb filter.Time
decay: loop filter gain at DC & filter cutoff (harmonic decay rate)