| On Tue, 2 Feb 1999, Jonathan Drexler wrote:
> I have been trying to set up an instrument in which the filter's
> cutoff-frequency "tracks" the current note being played, similar to
> keyboard tracking in commercial synthesizers. When I program a passage
> which simulates legato, the filter clicks at the beginning of many of
> the notes. The problem is apparently coming from the filter because
> when I remove the filter the legato is quite good. The filter opcode I
> am using is "tonex", but I have had the same problem with butterlp.
>
Try routing the output of the legato instrument to a global instrument
which contains only the tracking filter. You will have to pass the
frequency data via a global variable as well. Turn the filter
"instrument" on at the beginning of performance and let it run throughout.
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To: csound@maths.ex.ac.uk
From: "Peter L. Hoekje"
Subject: Re: Filter clicks
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J. Drexler wrote:
>I have been trying to set up an instrument in which the filter's
>cutoff-frequency "tracks" the current note being played, similar to
>keyboard tracking in commercial synthesizers. When I program a passage
>which simulates legato, the filter clicks at the beginning of many of
>the notes. The problem is apparently coming from the filter because
>...
Whenever you start up a filter, there will always be a startup transient.
Because you are trying to do legato, with a sudden step amplitude for each
frequency, this problem is exacerbated. The smoothest response will come if
you can turn off the old tone and turn on the new one slowly, within the
context of your legato. Here's an example, is this the kind of thing you're
looking for?
peter
;drexler.orc
;Trying to find a problem with J. Drexler's scheme
sr=44100
kr=44100
ksmps=1
nchnls=1
instr 1
; p4=amp, p5=freq, p6=f-table, p7=rise, decay
; p8 = khp/p5
aenv linen p4, p7, p3, p7
a1 oscil aenv, p5, p6
a2 tonex a1, p5*p8, 4, 0
out a2
endin
;drexler.sco
f1 0 2048 10 1
f2 0 2048 10 1 1 1 1 1 1 1 1 1 1 .5 .3 .1
; p4=amp, p5=freq, p6=f-table, p7=rise, decay, p8=khp/p5 for filter
t120
;First example shows problem: a double-size peak in
;filter output during first cycle, due to filter
;initialization. Filter khp = p5.
i1 0 1 300000 440 2 0 1
i. 1 1 . 493.9 . .
i. 2 1 . 523.3 . .
i. 3 1 . 493.9 . .
i. 4 1 . 440 . .
i. 5 1 0 . . .
s
; Filter at 5th harmonic
i1 0 1 90000 440 2 0 5
i. 1 1 . 493.9 . .
i. 2 1 . 523.3 . .
i. 3 1 . 493.9 . .
i. 4 1 . 440 . .
i. 5 1 0 . . .
s
; Try turning the sound on with linen,
; irise = i/p5... better!
i1 0 1 90000 440 2 .00227 5
i. 1 1 . 493.9 . .00202
i. 2 1 . 523.3 . .00191
i. 3 1 . 493.9 . .00202
i. 4 1 . 440 . .00227
i. 5 1 0 . . .
s
; Now try turning down the previous sound
; during new note's attack - best.
i1 0 1.00227 90000 440 2 .00227 5
i. 1 1.00202 . 493.9 . .00202
i. 2 1.00191 . 523.3 . .00191
i. 3 1.00202 . 493.9 . .00202
i. 4 1.00227 . 440 . .00227
i. 5 1 0 . . .
e
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Date: 03 Feb 99 12:21:16 EST
From: "Peter M. Traub"
Subject: Re: sndinfo error
To: csound@maths.ex.ac.uk
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--- You wrote:
Peter M. Traub wrote:
> I'm using sndinfo on various sound files that i have downloaded =
from
> the web. they have all been converted to 44k 16 bit stereo AIFF =
with
> SOX before sndinfo looks at them. with certain files, it outputs =
the
> following error (this is in Linux):
>=20
> 'Floating point exception (core dumped)'
>=20
> does this mean the soundfile is bad or is it something that =
sndinfo
> cannot read (it just looks at the file header, right?). Thanks in
> advance
Peter, what version of SOX are you using ? Earlier versions had =
some
trouble writing AIFF files properly. You can check my soundapps =
page for
the latest version.
=3D=3D Dave Phillips
http://www.bright.net/~dlphilp/index.html
http://www.bright.net/~dlphilp/linux_soundapps.html
--- end of quote ---
Hi Dave,
I'm using 12.15 which I linked to from your page (Chris Bagwell's =
version). I was previously using an older version which did have =
trouble writing AIFFs, but this one seems to be ok. I also tested =
one of the problematic sound files on an SGI, and sndinfo gave me a =
similar error:
Trace/BPT trap (core dumped)
The file played back fine using Peak on a Mac. Thanks again.
peter traub
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I think it depends by jpff, if he decides to include my additional stuff, you will be soon
satisfied.
Gab
Grant Covell wrote:
> Some items here:
>
> I'm eager to see in the Mills PPC.
> Can I be pushy by asking when these might find their way to the PPC?
> Esp. #4 since console messages for errors in macros provide line numbers
> that do not correlate to an extant/known file.
>
> Grant.
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From: Sinan Bokesoy
To: csound@maths.ex.ac.uk
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Subject: filter2 opcode
Date: Wed, 3 Feb 1999 21:17:49 +0200
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Hello everbody,
I need some help for using the filter2 opcode.
I can get the Bh and Ah vectors as the filter coefficents with =
corresponding number of poles and zeros from Matlab.
But after implementing the filter with the filter2 opcode, the result =
has been zero output.
Could someone please send us an example?
Thanks for your help,
Sinan Bokesoy
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Hello =
everbody,
I need some help for =
using the=20
filter2 opcode.
I can get the Bh and Ah vectors as the =
filter=20
coefficents with corresponding number of poles and zeros from=20
Matlab.
But after implementing the filter =
with the=20
filter2 opcode, the result has been zero output.
Could someone please send us an=20
example?
Thanks for your help,
Sinan =
Bokesoy
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From: Michael Gogins
To: Hans Mikelson , csound@maths.ex.ac.uk
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Subject: Re: Benchmarks was Re: About hrtfer
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Winsound should be changed to spawn a high-priority thread for computing
audio while the original thread handles GUI events. That would make it
essentially as fast as DirectSound. It should also be made re-entrant so
that it can stay up and do another run (most of the code is already there).
And it would be handy if it were an ActiveX control that could be embedded
in other apps.
The code for all of the above is available under the GNU General Public
License in AXCsoundAll.zip on my web site,
http://www.pipeline.com/~gogins/Csound/AXCsoundAll.zip.
-----Original Message-----
From: Hans Mikelson
To: csound@maths.ex.ac.uk
Date: Tuesday, February 02, 1999 9:24 PM
Subject: Benchmarks was Re: About hrtfer
>>That's strange... Motivated by the messages about hrtfer I yesterday
>>tried it too, on a 90MHz Pentium running Linux-2.2.1, with csound-3.511.
>
>
>I thought Winsound has been running slowly lately. I just ran the bach.orc
>bach.sco benchmark on a 300 MHz PII.
>
>Winsound 3.511 = 178 seconds
>Directsound 2.11 = 5.82 seconds
>
>Hmmm, something mighty fishy going on here...
>
>Hans Mikelson
>
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To: Sinan Bokesoy ,
csound list
Subject: Re: filter2 opcode
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I've had similar experiences.
Try with zfilter2 instead.
Try to reverse the sign of the coefficents.
Here is an example with both applied.
It worked the last time I checked ( consound version 3.49 I think ), why I don't know.
;10.order butterworth-lp, sr/6
ibgain=0.00001
iagain=-1
aout zfilter2 amix,1,1,10,10,ibgain*.0027,ibgain*.0273,ibgain*.1230,ibgain*.3281,ibgain*.5742,ibgain*.6891,ibgain*.5742,ibgain*.3281,ibgain*.1230,ibgain*.0273,ibgain*.0027,iagain*-7.4907,iagain*25.4989,iagain*-51.8964,iagain*69.8813,iagain*-65.0115,iagain*42.2946,iagain*-18.9909,iagain*5.6301,iagain*-0.9948,iagain*0.0795
> Sinan Bokesoy wrote:
> Hello everbody,
> I need some help for using the filter2 opcode.
> I can get the Bh and Ah vectors as the filter coefficents with corresponding number of poles and
> zeros from Matlab.
> But after implementing the filter with the filter2 opcode, the result has been zero output.
> Could someone please send us an example?
> Thanks for your help,
> Sinan Bokesoy
There seems to be some bugs in filter2/zfilter2
As I am a little interested myself in this I'll try
to look at the sourcecode to see if I understand something
of it.
Last time Erik Spjut wrote:
>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.
to proposed opcode:
>>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.
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From: Richard Dobson
Organization: Composers Desktop Project
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To: csound@maths.ex.ac.uk
Subject: Re: Benchmarks was Re: About hrtfer
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I just checked Winsound 3.50: it runs the bach orc/sco in around 7
seconds, on my PII 333 (NT4), with message-level = 0. So unless
something has gone very wrong with 5.111, the fishiness seems to be
local to your machine. Did you hit the Turbo button, and run with an
8MHz clock? :-)
Richard Dobson
> >
> >I thought Winsound has been running slowly lately. I just ran the bach.orc
> >bach.sco benchmark on a 300 MHz PII.
> >
> >Winsound 3.511 = 178 seconds
> >Directsound 2.11 = 5.82 seconds
> >
> >Hmmm, something mighty fishy going on here...
> >
> >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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From: Jim Stevenson
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To: csound@maths.ex.ac.uk, hljmm@werewolf.net
Subject: Re: notebook sound chips?
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I am planning to buy a new notebook, and want the best sound chips.
Also, what is MS 3d sound?
Thanks. |