| Does anyone know of a program to turn pascal code into c?
Please answer before you quote this.
Thanks.
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Date: Mon, 11 May 1998 08:27:05 -0400 (EDT)
From: Bezkorowajny Mathieu
Subject: Re: pascal to c?
To: Jim Stevenson
Cc: RWD@cableinet.co.uk, csound@maths.ex.ac.uk
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On Mon, 11 May 1998, Jim Stevenson wrote:
> Date: Mon, 11 May 1998 04:44:37 -0700
> From: Jim Stevenson
> To: RWD@cableinet.co.uk, csound@noether.ex.ac.uk
> Subject: Re: pascal to c?
>
> Does anyone know of a program to turn pascal code into c?
>
> Please answer before you quote this.
>
> Thanks.
>
BEZKO
-----------------------------------------------------
"[...]for any musical innovation is full of
danger to the whole State, and ought to be prohibited."
PLATO,("Republica")
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Date: Mon, 11 May 1998 10:13:31 -0700
From: Art Hunkins
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To: csound@maths.ex.ac.uk
Subject: Re: Csound Digest #379 on 11 May 1998
References:
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Hi, John and list--
Score11, a score preprocessor which I have used in 3 of my 4 Csound
pieces, outputs a line beginning with "c" (for comment) every time it
generates a (random) *rest* note. It does this by outputting a normal
looking "i" line and adding a "c" immediately before the "i".
Actually, it is simple enough to go through and manually delete all such
lines (particularly since "" is also added at line's end) prior to
"perfing" the score.
I also note that the old "scot" did output lines beginning with "c" as
well. Anybody still using this?
I imagine that "c" was used back in the days where all new
(non-continuation) score lines had to begin with an *alpha* opcode--and
"c" was the dedicated "Comment" opcode. (This was certainly true back in
the Music11/Music360 days.)
Please let me know what you decide because, as far as Score11 is
concerned, I have material going into the Boulanger CD describing
Score11 and how to use it with Csound. Right now the material states
that "*optionally* you may delete the comment ("c") lines before perf."
This would need to be changed to "you *must* delete the "rest" lines."
I'll look forward to hearing from you as soon as the issue is decided.
jpff@maths.bath.ac.uk wrote:
>
> -------------------------------------------------------
> Date: Mon, 11 May 98 10:34:45 BST
> From: jpff@maths.bath.ac.uk
> Subject: Undocumented facility, anyone use?
> To: csound@maths.ex.ac.uk
>
> Message written at 11 May 1998 09:32:43 +0100
>
> reading the sources of Csound, a habit I really must break, I noticed
> that at least in some cases the character 'c' starts a comment as well
> as ';', but I cannot find this in teh documentation. Actually i
> wanted to use 'c' for a new facility in the score, (which would make
> writing my latest work easier), but then I do not want to destroy
> previously running scores.
>
> Does anyone know about this, or use it? If it has no real existence I
> will release my newer score-reader with c meaning something else.
>
> Advice please.
> ==John ffitch
>
> PS I realise that the [] and changes to \ break some old scores. I
> have an untested fix for that here.
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Date: Mon, 11 May 1998 11:12:11 -0700
From: Jim Stevenson
Message-Id: <199805111812.LAA12052@eos.arc.nasa.gov>
To: jpff@maths.bath.ac.uk, rasmuse@hem.passagen.se
Subject: Re: poscil and backslash
Cc: csound@noether.ex.AC.UK
Please keep it in.
Or is there another pulse generating method?
Thanks.
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From: Jim Stevenson
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Subject: Re: poscil and backslash
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Please keep it in.
Or is there another pulse generating method?
Thanks.
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From: Paul Winkler
To: csound@maths.ex.ac.uk
Subject: HTML list archive: I hereby volunteer
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Dear everyone,
I (and presumably others) sorely miss the online HTML archive of this
mailing list. It's been missing almost all the messages for quite a
while, as of today there's NOTHING in there, and presumable Dave Topper
lacks either the time or the resources to fix it. I've downloaded the
archived messages from
ftp://ftp.musique.umontreal.ca/pub/mirrors/dream/Csound_List_Archive/
... but it's just not the same. <:)
For one thing, I can't browse it easily; for another, the latest chunk
of stuff in there seems to be Csound_99 which is sometime in May of 97.
I'd be willing to do some of the grunt work to try to put back together
the HTML archive, if Dave T. or anyone else is willing to give it a home
online. I have a Perl utility called MHonArc which should allow me to
automatically HTML-ize the messages and index them by date, thread,
and/or author.
Here's the next problem: MHonArc is written to sort mailboxes or
individual messages in "MH mail folders and UUCP/Unix mailbox files." It
won't recognize the files I got from jpff's place which, according to
"file", are MMDF mailbox files. MHonArc can't recognize these files. I
*think* I can figure out how to configure it to do so...
SO...
does anyone by any chance have any/all Csound mailing list messages
stored in either mail types?
--or--
know of a program which will do the same function for MMDF mailbox
files? --or-- convert MMDF files to MH or Unix mailbox files?
The last two points may be academic if I can get MHonArc to work with
the MMDF format, I think it's just a question of specifying a different
message separator type.
Anyone with useful info, please send it my way and I'll try to get this
thing put back together...
Thanks,
PW
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From: Paul Winkler
To: csound@maths.ex.ac.uk
Subject: Instrument won't do what I want.
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Date: Mon, 11 May 1998 13:59:18 PDT
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Here's what looks like an interesting instrument from Charles Dodge's
"Computer Music," second edition, p. 186-187. Unfortunately I can't get
it to do anything like what it's described as. Any suggestions would be
most welcome.
(bad ascii art follows)
amp peak BW
| BW open&close
| | |
+------+ | |
| rand | +------+
+------+ | f1 |
| \ /
| \__/
+-------+ |
0->| filter|<---+
+-------+
|
|
(output)
It says here, "The center frequency is set to 0, thus converting the
filter to a low pass. The instrument produces sound only when the
bandwidth is sufficiently wide to allow noise in the audio range. The
cutoff frequency is changed continuously by the controlling oscillator.
The amplitude of the oscillator sets the maximum cutoff frequency, and
the frequency of the oscillator determines the rate of interruption of
noise. The results fall within a range of sounds described as
'rustling,' 'thumping,' and 'scurrying.'"
Sounds cool to me. Here's how I think this is done in csound:
;orchestra
; sr = 44100 (commented by Cecilia)
; kr = 4410 (commented by Cecilia)
; ksmps = 10 (commented by Cecilia)
; nchnls = 1 (commented by Cecilia)
instr 1
idb= ampdb(p4)
kbw oscil p5, p6, 1 ; set peak freq and rate of filter open/close, use
f1
a1 rand idb ; random signal
afilt reson a1, 0, kbw ; lopass filter random signal
out afilt
endin
And here's a sample score:
f1 0 1024 9 0.5 1 0 ; half-sine
; 2=at 3=dur 4=db 5=peak freq 6=bw rate
i1 0 1 10 10000 1
i1 1 . . 100 1
i1 2 . . 10000 40
i1 3 . . 100 40
i1 4 . . 10000 1000
i1 5 . . 100 1000
e
I can't seem to come up with any useful values of the various pfields.
Either that or something's wrong with my interpretation of the
flowchart. The instrument either produces boring non-scurrying noise or
it's silent. Notice that I've set p4 to 10 decibels, yet this still
produces a few samples out of range. That doesn't seem right.
Ideas?
thanks for any responses,
PW
P.S. Here's a simple instrument from the same book that DOES work
nicely. Have fun with it.
; sr = 44100 (commented by Cecilia)
; kr = 4410 (commented by Cecilia)
; ksmps = 10 (commented by Cecilia)
; nchnls = 1 (commented by Cecilia)
instr 1 ; variably noisy pitched ringmodulator
; by PW based on C. Dodge (2nd ed.) p. 103
;set up an envelope
idur = p3 - p4 -p5
iamp = ampdb (p6)
kenv expseg 1, p4, iamp, p5, iamp, idur, 1
; set up note freq & noise bandwidth
ipit=cpspch(p8)
ibw = .01 * p7 * ipit; p7 sets noise bandwidth as percent of ipitch
;feed amplitude kenv into a random number generator
amp randi kenv, ibw
; random number modulates amp. of oscillator
aout oscil amp, ipit, 1 ; use table 1
out aout
endin
;sample score by PW
f1 0 512 10 1; sine
t 0 30
;p2=time 3=dur 4=atck 5=sus 6=dB 7=bw% 8=pch
i1 0 1.3 .1 0.001 76 10 7.00
i1 1 .5 .05 . 79 12 7.02
i1 1.5 .7 .02 .1 81 8 7.03
i1 1.9 .8 .2 0.001 83 7 6.09
i1 2.6 1 .01 . 66 10 8.09
i1 3.2 1.8 .005 . 73 15 8.10
i1 4 5 .005 .1 78 30 8.07
i1 6 5 1 1 90 80 5.06
i1 8.3 .5 .3 0.001 60 10 10.09
i1 7.4 5 1.5 2 37 60 11.05
i1 7.8 7 1 2.5 33 60 11.08
e
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Date: Mon, 11 May 1998 14:42:23 -0700
From: Erik Spjut
Subject: Re: Instrument won't do what I want.
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At 1:59 PM -0700 5/11/98, Paul Winkler wrote:
>Here's what looks like an interesting instrument from Charles Dodge's
>"Computer Music," second edition, p. 186-187. Unfortunately I can't get
>it to do anything like what it's described as. Any suggestions would be
>most welcome.
Try the following:
;orchestra
sr = 44100
kr = 44100
ksmps = 1
nchnls = 1
instr 1
idb= ampdb(p4)
kbw oscil p5, p6, 1 ;set peak freq and rate of filter open/close, use f1
a1 rand idb ; random signal
afilt reson a1, 0, kbw,1 ; lopass filter random signal
out afilt
endin
f1 0 1024 9 0.5 1 0 ; half-sine
; 2=at 3=dur 4=db 5=peak freq 6=bw rate
i1 0 1 60 10000 1
i1 1 . . 500 1
i1 2 . . 10000 5
i1 3 . . 500 5
i1 4 . . 10000 20
i1 5 . . 500 20
I set the iscl variable in reson to 1 instead of 0. With iscl the filter
gain doesnt change with bandwidth, and you can use a more realistic dB
value. Also kr=sr usually works better. I also changed the 100 Hz to 500 Hz
(It's hard to hear just the bass). I also slowed the rates from 40 and 1000
to 5 and 20. Enjoy.
-------------------------------------------------------------------------------
Erik Spjut (rhymes with cute) - Acting Director,The Center for Design Education
and/or Associate Professor of Engineering
Harvey Mudd College, Claremont, CA 91711 USA
Erik_Spjut@hmc.edu Ph & Voice mail (909) 607-3890 Fax (909) 621-8967
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From: Hans Mikelson
To: csound@maths.ex.ac.uk
MMDF-Warning: Parse error in original version of preceding line at UK.AC.Bath.maths.stork
Subject: Re: Csound Digest #379 on 11 May 1998
Date: Mon, 11 May 1998 18:58:55 -0500
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>I imagine that "c" was used back in the days where all new
Hmmm sounds suspiciously like FORTRAN...
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Date: Mon, 11 May 1998 17:31:21 -0700
From: Toby
Organization: Other than 12-tone
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Subject: Space opcode in 3.48?
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V3.48 notes describe space opcode. My
version 3.48 for Linux never heard of
it. Is it really out there?
Thanks
Toby
--zder autu a+ !$- la--
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Subject: Re: hYdra name
To: Csound mailing list
Date: Mon, 11 May 1998 21:34:05 -0400 (EDT)
From: Eli Brandt
In-Reply-To: <01bd7adb$53112160$37e167ce@hljmm> from "Hans Mikelson" at May 8, 98 06:45:04 pm
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Hans Mikelson wrote:
> Likewise you could manufacture a Moog automobile without violating trademark
> restrictions.
If you picked this example without having heard of Moog Automotive, I
tip my hat to you. :-)
--
Eli Brandt | eli+@cs.cmu.edu | http://www.cs.cmu.edu/~eli/
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Date: Mon, 11 May 1998 22:06:13 -0400
To: csound@maths.ex.ac.uk
From: Charles Starrett
Subject: Ethnography of Csound
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Hello fellow Csounders!
I am a second-year graduate student in Ethnomusicology at who is
particularly interested (as a performer, composer as well as scholar) in
Music Technology. For a seminar in "The Transmission of Musical Tradition"
this semester I am doing a term project on CSound. Although I began by
looking at Csound as the latest stage in a generation of Computer Music
Composition--tracing its heritage back through the Music "N" series to Max
Mathews at Bell Labs, I am now trying to look at other issues of
"transmission" which surround Csound. I am particularly interested in how
CSound as a tool relates to the transmissions of attitudes aesthetics and
compositional style. (I'm sure there are interesting issues related to the
influence of attitudes on CSound and perhaps vice-versa as well!)
Most of my colleagues are studying "World Music" topics, but I have always
had a great interest in electronic and computer composition. I have chosen
CSound for my project mostly because I am already familiar with it through
my attempts to compose with it (and I am making headway, albeit slowly) and
because I have been on the Csound mailing list for over a year now.
Anything you could share of your own experience with Csound and its
predecessors would be great help to me in my project! Some examples might
be:
*As a student
*As a teacher
*As a composer
*As a researcher/engineer
*From a technical perspective
*From an artistic perspective
*Csound as a continuation of the past
*Csound as a continuation into the future
*Compare/Contrast with Music III/IV/V/360/11/
*Compare/Contrast with Cmusic, CLM, etc...
I am particularly interested in issues pertaining to MUSIC and the
TRANSMISSION of MUSICAL TRADITION through Csound (and Music 11, Music 360,
etc.)...
...but, as I said, anything would be helpful! It may be best if you email
me directly at:
starrett@fas.harvard.edu
Thank you in advance for your help. Although the paper is due soon, I hope
to have an expanded and revised copy up on the web by the end of the
summer. To that end, please be sure to tell me if you would prefer to be
referred to by a pseudonym.
Thanks again!
~~Charles
/----Charles D. Starrett-----\ "I do not feel that
| / | ____ | | ____ | my research suffered unduly
| /\--| | |-| | | from the fact that I enjoyed it."
| / \ | |____| | |___ |
| | | | ___|___ | --Daniel Miller,
\--starrett@fas.harvard.edu--/ Modernity--an Ethnographic Approach
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Date: Mon, 11 May 1998 19:33:51 -0700
From: Toby
Organization: Other than 12-tone
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To: Fat_Boy
Cc: csound@maths.ex.ac.uk
Subject: Re: balancing samples
References:
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First, you need to make sure that your loop points are chosen
correctly. I recommend DAP for Linux to accomplish this.
Second, you might try using the loop mode that traverses the
looped section back and forth rather than repeating in the
same direction. This lessens the effect of the starting
and ending amplitude of the loop section being different.
Third, are you sure that you need to loop? It is the
achilles' heal of sampling. If your computer has the
resources, just load obscenely large samples into memory
and play them straight. After all, this is CSOUND!
Let's say you had a piano that you wanted to sample
Generous parameters might be:
44100Hz S/R
Sample every fourth key
each sampled for 10 seconds
So you need (88 / 4) * 88200 * 10 = 19,404,000 Bytes
of RAM available for that sample. This is reasonable
on a recent computer. And of course, the higher
notes really only need half of the length, if that.
By the way, I have been meaning to ask this:
The DAP editor will allow loop point modification,
but seems to be incapable of manipulating the recorded
base frequency in the sound file header, save the action
of nullifying any existing value to middle C. Does
anyone know of a method for altering this value directly
with some utility?
Along that line, what is the most recent doc on the AIFF
format? Any that I have had to date have always had
*DRAFT* printed conspicuously at the top.
Toby
-There otta be a law-
Fat_Boy wrote:
>
> hi,
>
> i was wondering if someone out there could help me with sampling on
> csound, in particular, making loops a little smoother using loscil. say i
> have a recorded piano sample, and take a 2s portion from 4s to 6s, when
> the sound has already decayed somewhat, and is now not dying especially
> rapidly. i want to use this as a looped sample, but the slight decay it
> has emphasizes the looping points. is there a quick and easy way, to get
> the 2s portion evened out? should this make the loop smoother? i can apply
> an amplitude gain envelope (using cooledit), of course, but this is a
> rough solution. can balance be used for this? or someth!ng else?
>
> tia,
>
> f_b
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Reply-To: Ken Locarnini
From: Ken Locarnini
To: CSound Mailing List
Subject: 400 Mhz P 2's
Date: Mon, 11 May 1998 21:26:08 -0700
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Here's a little audio info on the new P 2's:
http://www.harmony-central.com/Computer/Articles/Benchmark/
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From: Paul Winkler
To: csound@maths.ex.ac.uk
Subject: Re: Instrument won't do what I want.
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Thanks for the tips, Eric, I hadn't noticed the iscl option. Your
changes got some more sound out of it... it's a good starting point at
least. Playing with the f1 shape and varying various p-fields over time
should get some more interesting results. I'm curious to see the score
for Dodge's CASCANDO to see what he actually did with this thing, and/or
hear a recording of the piece.
______________________________________________________
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From: Paul Winkler
To: csound@maths.ex.ac.uk
Subject: HTML csound mailing list archive
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Success!
I now have managed to create an HTML archive of this mailing list,
indexed by date, by thread, and by author, consisting of 252 messages
from February to May 1997.
I just noticed there's a lot more mailbox files on the pub/dream ftp
archive that I didn't get last time I was there... that should go a good
step towards completing the archive.
Now all we need is someone to give these files a home on the web!
Any volunteers? Dave T., are you still with us?
best,
PW
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Date: Tue, 12 May 98 11:22:31 BST
From: jpff@maths.bath.ac.uk
Subject: Re: Hi... I'm back...
To: csound@maths.ex.ac.uk
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Message written at 11 May 1998 21:21:26 +0100
--- Copy of mail to psotuyo@equipenet.com.br ---
In-reply-to: <1.5.4.32.19980507200718.00726de8@pop3.equipenet.com.br>
(psotuyo@equipenet.com.br)
References: <1.5.4.32.19980507200718.00726de8@pop3.equipenet.com.br>
If you look at ftp.maths.bath.ac.uk in pub/dream/Csound_Archive/ you
will find all the mail ... may be a few days behind but i will fix
that. The format is gziped and raw mail.
==John ffitch
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From: J P Fitch
To: csound@maths.ex.ac.uk
Subject: Re: Hi... I'm back...
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Sory; location is pub/dream/Csound_List_Archive and it is now uptodate
==John
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From: jpff@maths.bath.ac.uk
To: csound@maths.ex.ac.uk
Subject: Undocumented c in score
Date: Tue, 12 May 98 12:25:30 BST
Source-Info: From (or Sender) name not authenticated.
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Clearly from the mail I have received the c is still in use, so I
recant from my interest in recycling it. I think it might be a good
idea to document it though.
I have grabbed b for my use at present, and what is more, it
works....
==John
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Date: Tue, 12 May 1998 08:36:07 +0100
From: Jamie Bullock
Subject: Joystick-in idea.
To: Csound List
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Is anyone interested in writing an opcode which reads data from the joystick port on PC soundcards (a bit platform-specific I know)?
I've used the joystick port before; I have a few apps which convert joystick port data to MIDI - I know this can then be routed to Csound's midiin opcodes, but it would be nice if Csound could read the port directly - to gain higher resolution and lower latency.
The opcode could take the form:
k1, k2, k3 .... k8 joyin irange1, irange2 ...irange8[,imode]
Where it takes in data, maps it to the range specified by the irange settings (ie 0-irange), and outputs the value at k-rate (k1 etc.)
Is anyone interested in writing such a thing ... or am I asking the impossible?
Cheers,
JamieB
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Is anyone interested in writing an opcode which reads data
from the joystick port on PC soundcards (a bit platform-specific I
know)?
I've used the joystick port before; I have a few
apps which convert joystick port data to MIDI - I know this can then be routed
to Csound's midiin opcodes, but it would be nice if Csound could read the port
directly - to gain higher resolution and lower latency.
The opcode could take the form:
k1, k2, k3 .... k8 joyin irange1, irange2
...irange8[,imode]
Where it takes in data, maps it to the range
specified by the irange settings (ie 0-irange), and outputs the value at k-rate
(k1 etc.)
Is anyone interested in writing such a thing ...
or am I asking the impossible?
Cheers,
JamieB
--Boundary_(ID_aBgZ6dOWKv9U7oWBz5Brwg)--
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From: Karl Greenwood
To: csound@maths.ex.ac.uk
Subject: Few Questions
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Just a couple of questions..
1) highest partial anyone has used or seen used
2) largest table size (sum big ones must be about)
3) are partial strengths integer or real
(can not find source on hard disk at the moment which reminds me of a
joke about an actress and a bishop...)
cheers
kARL
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From: Joseph L ANDERSON
Organization: The University of Birmingham
To: csound@maths.ex.ac.uk
Date: Wed, 13 May 1998 16:08:48 GMT
Subject: (Fwd) Re: Few Questions
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Karl,
A long answer to your first question...
> 1) highest partial anyone has used or seen used
Consider a pitch class, say A, and a sample rate, say 44.1kHz, and a
mic w/ a response up to 22.05kHz.
If the A4 is played on a piano its cps value is 440Hz. If you
record this note you will find you can capture energy at the 50th
harmonic.
Multiply 440Hz by 50 and you get 22kHz. Harmonics 51 (22.44kHz) and
above will not be picked up by the mic, and would be filtered out on
the A/D conversion anyway. (Frequencies above sr/2 can't be
represented w/ digital.)
Play A2 instead. It's cps val is 110Hz. The mic and A/D will be
able to capture harmonic 200 this time. (200*110Hz = 22kHz)
Play A1 w/ cps of 55Hz. This time you can get 400*55Hz = 22kHz.
If you go to higher notes, you find that you get less harmonics
represented in the audio chain.
Try A5 at 880Hz. You'll find you get only 25 harmonics (25*880Hz =
22kHz).
So for a short answer to your question, the lower the fundamental of
your pitch is, the more number of harmonics you get. The higher the
pitch gives less harmonics. Easy, huh??
Joseph L Anderson
Barber Institute of Fine Arts
Music Department
University of Birmingham
Edgbaston
BIRMINGHAM, B17 0NY
England
phn:+44 121 414 5781
fax:+44 121 414 5781
Joseph L Anderson
Barber Institute of Fine Arts
Music Department
University of Birmingham
Edgbaston
BIRMINGHAM, B17 0NY
England
phn:+44 121 414 5781
fax:+44 121 414 5781
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From: David Boothe
To: "'csound@maths.ex.ac.uk'"
Subject: sndwarp problem
Date: Wed, 13 May 1998 16:45:54 -0500
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Hello Csounders-
According the documentation on sndwarp::
"Time and frequency modification are indepentant from one another."
I have not been able to get this to happen. Changing xtimewarp modifies
time and pitch by the same factor, while changing xresample has no
effect. Compiling the orc/sco files below gives a very good
representation of the original sound file, with silence padding out the
end. (soundin.599 is a 440 Hz sine wave 65536 samples long in aiff
format), Changing p5 and p6 to .5 creates a sine wave twice as long and
an octave lower. Changing p7 and p8 to .5, or anything else, has no
effect. I have tried it with RT Sound 1.9 (implementing Csound 3.47) and
Winsound 3.477. What am I doing wrong? Thank you for your help.
-dB
(David M. Boothe, Audio Director
Lyrick Studios, Dallas, TX)
--------------
;orc file using soundin.599
sr = 44100
kr = 4410
ksmps = 10
nchnls = 1
instr 01
aamp = p4 ;amplitude scaling of samp
iatim = p5 ;starting time factor
ibtim = p6 ;ending time factor
iapch = p7 ;starting pitch factor
ibpch = p8 ;ending pitch factor
iwsize = p9 ;window size in samps
irandw = p10 ;rand window size bw (in samps)
ioverlap = p11 ;dens of overlapping windows
ifn1 = 71 ;ftable sound
ifn2 = 81 ;ftable window
ibeg = 0 ;where to start reading ftable
itimemode = 0 ;make atimewarp scaling factor
atimewarp line iatim, p3, ibtim ;time scaling
aresample line iapch, p3, ibpch ;pitch scaling
a1 sndwarp aamp, atimewarp, aresample, ifn1, /
ibeg, iwsize, irandw, ioverlap, ifn2, itimemode
out a1
endin
-----------------
;sco file using soundin.599
f71 0 65536 -1 599 0 0 0 ;soundfile
f81 0 512 20 1 ;Hamming Window
;p4 amplitude scaling of samp
;p5 starting time factor
;p6 ending time factor
;p7 starting pitch factor
;p8 ending pitch factor
;p9 window size in samps
;p10 rand window size bw (in samps)
;p11 dens of overlapping windows
;p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 p11
i1 0 4 1 1 1 1 1 1 0 10
e
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Date: Wed, 13 May 1998 15:07:17 -0700 (PDT)
From: Richard Karpen
To: David Boothe
Cc: "'csound@maths.ex.ac.uk'"
Subject: Re: sndwarp problem
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It looks like your window size is 1 sample long! Try making in 2000 and
then also make the random windowsize something like 200.
Richard Karpen
On Wed, 13 May 1998, David Boothe wrote:
> Hello Csounders-
>
> According the documentation on sndwarp::
>
> "Time and frequency modification are indepentant from one another."
>
> I have not been able to get this to happen. Changing xtimewarp modifies
> time and pitch by the same factor, while changing xresample has no
> effect. Compiling the orc/sco files below gives a very good
> representation of the original sound file, with silence padding out the
> end. (soundin.599 is a 440 Hz sine wave 65536 samples long in aiff
> format), Changing p5 and p6 to .5 creates a sine wave twice as long and
> an octave lower. Changing p7 and p8 to .5, or anything else, has no
> effect. I have tried it with RT Sound 1.9 (implementing Csound 3.47) and
> Winsound 3.477. What am I doing wrong? Thank you for your help.
>
> -dB
> (David M. Boothe, Audio Director
> Lyrick Studios, Dallas, TX)
>
> --------------
> ;orc file using soundin.599
> sr = 44100
> kr = 4410
> ksmps = 10
> nchnls = 1
>
> instr 01
>
> aamp = p4 ;amplitude scaling of samp
> iatim = p5 ;starting time factor
> ibtim = p6 ;ending time factor
> iapch = p7 ;starting pitch factor
> ibpch = p8 ;ending pitch factor
> iwsize = p9 ;window size in samps
> irandw = p10 ;rand window size bw (in samps)
> ioverlap = p11 ;dens of overlapping windows
> ifn1 = 71 ;ftable sound
> ifn2 = 81 ;ftable window
> ibeg = 0 ;where to start reading ftable
> itimemode = 0 ;make atimewarp scaling factor
> atimewarp line iatim, p3, ibtim ;time scaling
> aresample line iapch, p3, ibpch ;pitch scaling
>
> a1 sndwarp aamp, atimewarp, aresample, ifn1, /
> ibeg, iwsize, irandw, ioverlap, ifn2, itimemode
> out a1
> endin
>
> -----------------
> ;sco file using soundin.599
> f71 0 65536 -1 599 0 0 0 ;soundfile
> f81 0 512 20 1 ;Hamming Window
>
> ;p4 amplitude scaling of samp
> ;p5 starting time factor
> ;p6 ending time factor
> ;p7 starting pitch factor
> ;p8 ending pitch factor
> ;p9 window size in samps
> ;p10 rand window size bw (in samps)
> ;p11 dens of overlapping windows
>
> ;p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 p11
> i1 0 4 1 1 1 1 1 1 0 10
> e
>
>
>
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Date: Thu, 14 May 1998 01:46:25 +0200
From: rasmus ekman
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Subject: Re: illegal opcode
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jpff@maths.bath.ac.uk wrote:
>
> There is no noctdft opcode in the Csound that I know -- it is not in
> the manual.
Yes it is. It's in the manuals - both in HTML and the 3.0 one which we all
started from. It also WAS in Csound until fairly recently - I happen to have
some snips from -95, and it's right there in Entry.C (together with octdown)
just before the spec* opcodes. Also I believe I saw them - commented out -
in Entry.C just one or two versions ago, but that may be a hallucination.
So, I guess we remove those too, then...
Regards,
re
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Message-Id:
From: David Boothe
To: 'Richard Karpen' ,
David Boothe
Cc: "'csound@maths.ex.ac.uk'"
Subject: RE: sndwarp problem
Date: Wed, 13 May 1998 18:43:22 -0500
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Thank you, Richard. That fixed it with Winsound 3.47. I downloaded
Winsound 3.481 in the meantime (thinking that might solve the problem),
and get the following compile-time errors (I don't know what they mean):
error: Unknown # option, line 14:
irandw = p10 ;rand window size bw (in samps)
error: Unknown # option, line 15:
ioverlap = p11 ;dens of overlapping windows
In summary: Winsound 3.47 and RT Sound 1.9 compile OK, giving expected
results with different values of p5 through p8. Winsound 3.481 gives
above errors with same orc/sco files, so it seems something has happened
between versions. Guess I'll keep using 3.47 with sndwarp for a while.
-dB
(David M. Boothe, Audio Director
Lyrick Studios, Dallas, TX)
> -----Original Message-----
> From: Richard Karpen [SMTP:karpen@u.washington.edu]
> Sent: May 13, 1998 17:07
> To: David Boothe
> Cc: 'csound@maths.ex.ac.uk'
> Subject: Re: sndwarp problem
>
>
> It looks like your window size is 1 sample long! Try making in 2000
> and
> then also make the random windowsize something like 200.
>
> Richard Karpen
>
>
> On Wed, 13 May 1998, David Boothe wrote:
>
> > ;orc file using soundin.599
> > sr = 44100
> > kr = 4410
> > ksmps = 10
> > nchnls = 1
> >
> > instr 01
> >
> > aamp = p4 ;amplitude scaling of samp
> > iatim = p5 ;starting time factor
> > ibtim = p6 ;ending time factor
> > iapch = p7 ;starting pitch factor
> > ibpch = p8 ;ending pitch factor
> > iwsize = p9 ;window size in samps
> > irandw = p10 ;rand window size bw (in samps)
> > ioverlap = p11 ;dens of overlapping windows
> > ifn1 = 71 ;ftable sound
> > ifn2 = 81 ;ftable window
> > ibeg = 0 ;where to start reading ftable
> > itimemode = 0 ;make atimewarp scaling factor
> > atimewarp line iatim, p3, ibtim ;time scaling
> > aresample line iapch, p3, ibpch ;pitch scaling
> >
> > a1 sndwarp aamp, atimewarp, aresample, ifn1, /
> > ibeg, iwsize, irandw, ioverlap, ifn2, itimemode
> > out a1
> > endin
> >
> > -----------------
> > ;sco file using soundin.599
> > f71 0 65536 -1 599 0 0 0 ;soundfile
> > f81 0 512 20 1 ;Hamming Window
> >
> > ;p4 amplitude scaling of samp
> > ;p5 starting time factor
> > ;p6 ending time factor
> > ;p7 starting pitch factor
> > ;p8 ending pitch factor
> > ;p9 window size in samps
> > ;p10 rand window size bw (in samps)
> > ;p11 dens of overlapping windows
> >
> > ;p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 p11
> > i1 0 4 1 1 1 1 1 1 0 10
> > e
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Date: Wed, 13 May 1998 20:18:30 -0400
From: Dave Phillips
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To: csound@maths.ex.ac.uk
Subject: noctdft & space opcodes
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Greetings:
As Rasmus Ekman has already indicated, noctdft is indeed entered in
the latest HTML manual. It seems to have been last included in Csound
version 3.46 sources and has been absent since. The deleted poscil
opcode is also still in the HTML manual.
Another recent inquiry asked why the 'space' opcode wasn't in version
3.48 (Linux version). I checked the source tree again, and sure enough
the opcode is not included in the entry table. It *is* in version 3.481,
so Linux Csounders may wish to upgrade. 3.481 is available from the BGSU
MusTec server in the USA and from the Bath site in the UK.
== Dave Phillips
http://www.bright.net/~dlphilp/index.html
http://www.bright.net/~dlphilp/linux_soundapps.html
|