| As wgpluck is based on waveguide techniques, and the code doesn´t implement
fractional delay interpolation, there is an inherent pitch quantisation
because the delay length must be a multiple of kr, of course this is most
noticeable at high frequencies: at sr = 44100 a delay of 1 sample means
22050 Hz and a delay of 2 samples = 11025, nothing in between!
I would suggest either use an opcode employing resampling techniques (like
pluck) or a waveguide instrument with delay interpolation, like the ones
included in my ks.zip file at
http://members.tripod.com/csound/
Josep M
Larry Troxler escribió:
>
> I do think it's some sort of frequency quantization problem, because it
> gets worse as the pitch gets higher.
>
> There are more than one opcodes, it seems, for a physically modeled
> pluck, so if wgpluck is not the one I should use, let me know, please!
> Thanks :-)
--
Josep M Comajuncosas
C/ Circumval.lacio 75 08790 Gelida - Penedes
Catalunya - SPAIN tel. 93 7792243
e-mail: gelida@intercom.es
ET Informatica de Sistemes
e-mail: jcomajuncosas@campus.uoc.es
http://members.tripod.com/csound/
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From: Grant Covell
To: "'csound@maths.ex.ac.uk'"
Subject: Glad to see soundin fixed...
Date: Thu, 26 Aug 1999 11:22:30 -0400
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Multiple calls to soundin w/in the same instrument (say, within a timout
structure) are now possible. I've been hankering for this for about 1 year!
Thanks
Grant.
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From: Grant Covell
To: "'csound@maths.ex.ac.uk'"
Subject: bug with line?
Date: Thu, 26 Aug 1999 12:21:55 -0400
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Hello--
Here's a problem I've seen using PPC 3.56.
The following sco/orc reveals an odd behavior of line that I can't explain.
When this is rendered, klocinc passes 0 and becomes negative for a short
time (Note that example adds 1 to the random number generated by klocinc so
that the timout doesn't overflow or perpetually get closer to 1 w/o reaching
it, but from the print statement you can see that klocinc is negative).
Grant.
=;; sco
sr = 44100
kr = 44100
ksmps = 1
nchnls = 2
instr 9
idur = p3
klocinc init 10
klocinc line 10, idur, 0
isil unirand klocinc
core:
isil unirand klocinc
timout 0, isil + 1, do
reinit core
do:
print klocinc, isil
endin
;; orc
i9 0 243
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From: Torsten Anders
To: csound@maths.ex.ac.uk
Subject: five args in macro cause "wrong" error
Date: Thu, 26 Aug 1999 16:12:02 +0200
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If one uses five args in _any_ macro definition in the *.orc file, csound
annonces a failure of the _last_ macro. Althought csound happily moves on,
programs as cecilia will stop then.
On something like:
sr = 44100
kr = 441
ksmps = 100
nchnls = 1
; only two dummys
#define FIVEARGS(A'B'C'D'E) #
#
#define DUMMY #
#
instr 1
endin
csound answers:
error: Too many arguments to macroMacro definition for DUMMY
Torsten Anders
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From: Sherlock
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Phasor does not work according the sample.
Does anyone who knows this command inside and out tell me why Csound,
and Winsound, gives a blank sound wave?
I am working on a phaser-like sound. And thought phasor would be perfect
for the job.
Robert Mcnulty Junior
Windows 98 user and Csound Enthusist.
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From: pete moss
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To: Wayne Freno ,
"csound@maths.ex.ac.uk"
Subject: Re: Scores on paper
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i have submitted csound code as well as printouts of FFTs. i have also
drawn up structures on graph paper. most arent too happy to get csound
code (although i would be), but i tell them it is the best
representation of the music. if you really wanted to be an asshole, you
could hand them a list of PCM values ;)
:P
Wayne Freno wrote:
>
> I have heard about the importance of having a printed score
> if one is going to get one's music either heard or performed
> (even tape music I understand) and I am curious about how
> people on the list go about this. Do any of you use a graphic
> representation, (or traditional notation etc), of your csound
> scores on paper?
>
> Wayne
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To: Wayne Freno , csound@maths.ex.ac.uk
MMDF-Warning: Parse error in original version of preceding line at UK.AC.Bath.maths.omphalos
From: tolve
Subject: Re: Scores on paper
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the music publishing (facs!zst) business has not been about publishing
perhaps since tin pan alley. it's all about royalty payments for
recordings. your work can be registered, at least in the u.s., with a
recording (form PA). search for register of copyrights on web.
but from your post it sounds as though you may wish to have your work
performed live and believe it to be possible. traditional notation works
fine for traditional pitched music. finale is probably the most
comprehensive (and complex) software package to deal with scores (not sure
of its graphic capabilities).
as for systems of graphic notation, there have been numerous attempts.
believe this subject has come up on the csound list and a number of
resources were recommended but a search of my own archives have not turned
up anything back to january. you might try searching the csound archives
prior to then. get back to me if you don't find anything and i'll look some
more. but from what i've seen, such systems will probably only provide some
ideas -composers often tend to create symbols and instructions of their own
to best suit their work so don't (ever) hesitate to roll your own rules.
tolve
>I have heard about the importance of having a printed score
>if one is going to get one's music either heard or performed
>(even tape music I understand) and I am curious about how
>people on the list go about this. Do any of you use a graphic
>representation, (or traditional notation etc), of your csound
>scores on paper?
>
>Wayne
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Date: Fri, 27 Aug 1999 00:08:44 +0000
From: Larry Troxler
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To: Josep M Comajuncosas
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Subject: Re: wgpluck : unwanted freq quantization?
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Josep M Comajuncosas wrote:
>
> As wgpluck is based on waveguide techniques, and the code doesn´t implement
> fractional delay interpolation, there is an inherent pitch quantisation
Huh? The code for wgpluck *does* try to implement fractional delay,
unless I'm reading it incorrectly. It uses a fractional delay allpass
filter in series with the waveguide. See pluckPluck() in pluck.c, which
instantiates a waveguide object coded in wavegde.c, which does appear to
set up a tuning filter. But this is as far as I've looked into it -
maybe the filter isn't actually getting used, or the code is just buggy.
There's also wgpluck2, but at first glance it appears that wgpluck2
doesn't implement fractional delays.
If I get stuck, I might look at your instruments, but for performance,
I'd like to try to get one of these ugens to work.
Thanks.
Larry
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Thu, 26 Aug 1999 20:37:38 -0400 (EDT)
From: Michael Gogins
To: music-dsp , CSOUND ,
JavaSound
Subject: Watch out for Noise
Date: Thu, 26 Aug 1999 20:40:14 -0400
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I shall be proceeding to create a version of sfront, John Lazzaro's SAOL to
C compiler, called "Noise" (name copyright or trademark Michael Gogins 1999,
or whatever) that accepts mp4, orc, sco, and/or mid files, as well as
realtime audio and MIDI input, and outputs either realtime or soundfile
audio. Noise may use various external interfaces for drivers (JavaSound,
VST2, DirectShow, etc.). The first of these to be implemented will be
JavaSound beta 0.9 for JDK 1.3 beta. Noise also will have a Java control
interface so that it can plug into Silence, my algorithmic composition
package, for software synthesis. It will of course also work with any other
Java program on JDK 1.3.
To begin with, Noise will shell out to the operating system and run the C
compiler to build a shared library that it will load and use as an
orchestra. In the future, I hope to interface with all-in-one runtime SAOL
compilers or perhaps with SAOL compilers embedded in sound cards.
Noise also will be able to direct Silence scores to the JavaSound MIDI
synthesizer, which has been much improved in quality.
While I am creating Noise, I will be placing my development of a pure Java
synthesizer and further work on AXCsound, my ActiveX/Java interface to
DirectSound, on hold.
My goal is, again, to bring "universal" software synthesis, on a standard
and extensible platform, into Silence in particular, and Java in general.
As long as it is based on sfront, Noise will be licensed under the terms of
the GNU General Public License, as is sfront. And as long as Noise is under
public license, I welcome collaboration on the project. I do not have time
to develop on platforms other than Windows, but I will make every effort to
ensure that the code will compile and run on any JDK 1.3 platform including
Solaris and, in future, the Macintosh and Linux.
Experimental computer music people, allow me to draw your attention to the
fact that SAOL provides for "user-defined opcodes" written in SAOL itself.
In addition to this, which would normally be preferred, the design of sfront
lends itself to the enlargement of the opcode library with additional C or
C++ sources. This, although it exceeds the scope of the MP4 standard, might
be a desirable choice for experiments with, for example, new time/frequency
transforms.
I would prefer sfront and other implementations of SAOL to perform all
computations on real numbers using double-precision floating point numbers
(64 or 80 bit calculations depending on platform, but 8 byte C "double"
storage in any case). There is no single implementation of this format (see
below) either.
The motivation is musical: greater precision, less noise. It is apparent
from some experiments done recently with Csound that the change to higher
precision does make an audible difference.
I have just downloaded JDK1.3 beta for Windows and listened to the JavaSound
demos with the "deluxe" soundbank. There is an immediately audible
improvement in sound quality for MIDI synthesis. I still cannot find any
MIDI input, but it is relatively easy to implement this oneself using either
Robert Marsanyi's JavaMIDI or native methods.
-----Original Message-----
From: John Lazzaro [mailto:lazzaro@CS.Berkeley.EDU]
Sent: Thursday, August 26, 1999 7:19 PM
To: saol-dev@media.mit.edu
Subject: Re: Is SFront deterministic ?
> WRT the last paragraph of your mail below, normative determinism would
make
> conformance-testing a breeze too : decoders producing the right sound file
> (byte for byte) for every single test orc-sco pair would pass, otherwise
they
> would fail.
Note that the FDIS is very careful not to use the words "IEEE Standard
Floating Point" -- I was under the impression that this was done on
purpose, so that DSP chips that have non-standard 32-bit floating point
ALU's could do compliant decoders, as well as SIMD units on x86 processors
that may not be exactly implementing IEEE floating point. Given that
property of the standard, byte-by-byte accuracy isn't going to work for
compliance testing ...
--jl
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From: Michael Gogins
To: music-dsp , CSOUND ,
JavaSound
Subject: Watch out for Noise
Date: Thu, 26 Aug 1999 20:40:14 -0400
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I shall be proceeding to create a version of sfront, John Lazzaro's SAOL to
C compiler, called "Noise" (name copyright or trademark Michael Gogins 1999,
or whatever) that accepts mp4, orc, sco, and/or mid files, as well as
realtime audio and MIDI input, and outputs either realtime or soundfile
audio. Noise may use various external interfaces for drivers (JavaSound,
VST2, DirectShow, etc.). The first of these to be implemented will be
JavaSound beta 0.9 for JDK 1.3 beta. Noise also will have a Java control
interface so that it can plug into Silence, my algorithmic composition
package, for software synthesis. It will of course also work with any other
Java program on JDK 1.3.
To begin with, Noise will shell out to the operating system and run the C
compiler to build a shared library that it will load and use as an
orchestra. In the future, I hope to interface with all-in-one runtime SAOL
compilers or perhaps with SAOL compilers embedded in sound cards.
Noise also will be able to direct Silence scores to the JavaSound MIDI
synthesizer, which has been much improved in quality.
While I am creating Noise, I will be placing my development of a pure Java
synthesizer and further work on AXCsound, my ActiveX/Java interface to
DirectSound, on hold.
My goal is, again, to bring "universal" software synthesis, on a standard
and extensible platform, into Silence in particular, and Java in general.
As long as it is based on sfront, Noise will be licensed under the terms of
the GNU General Public License, as is sfront. And as long as Noise is under
public license, I welcome collaboration on the project. I do not have time
to develop on platforms other than Windows, but I will make every effort to
ensure that the code will compile and run on any JDK 1.3 platform including
Solaris and, in future, the Macintosh and Linux.
Experimental computer music people, allow me to draw your attention to the
fact that SAOL provides for "user-defined opcodes" written in SAOL itself.
In addition to this, which would normally be preferred, the design of sfront
lends itself to the enlargement of the opcode library with additional C or
C++ sources. This, although it exceeds the scope of the MP4 standard, might
be a desirable choice for experiments with, for example, new time/frequency
transforms.
I would prefer sfront and other implementations of SAOL to perform all
computations on real numbers using double-precision floating point numbers
(64 or 80 bit calculations depending on platform, but 8 byte C "double"
storage in any case). There is no single implementation of this format (see
below) either.
The motivation is musical: greater precision, less noise. It is apparent
from some experiments done recently with Csound that the change to higher
precision does make an audible difference.
I have just downloaded JDK1.3 beta for Windows and listened to the JavaSound
demos with the "deluxe" soundbank. There is an immediately audible
improvement in sound quality for MIDI synthesis. I still cannot find any
MIDI input, but it is relatively easy to implement this oneself using either
Robert Marsanyi's JavaMIDI or native methods.
-----Original Message-----
From: John Lazzaro [mailto:lazzaro@CS.Berkeley.EDU]
Sent: Thursday, August 26, 1999 7:19 PM
To: saol-dev@media.mit.edu
Subject: Re: Is SFront deterministic ?
> WRT the last paragraph of your mail below, normative determinism would
make
> conformance-testing a breeze too : decoders producing the right sound file
> (byte for byte) for every single test orc-sco pair would pass, otherwise
they
> would fail.
Note that the FDIS is very careful not to use the words "IEEE Standard
Floating Point" -- I was under the impression that this was done on
purpose, so that DSP chips that have non-standard 32-bit floating point
ALU's could do compliant decoders, as well as SIMD units on x86 processors
that may not be exactly implementing IEEE floating point. Given that
property of the standard, byte-by-byte accuracy isn't going to work for
compliance testing ...
--jl
dupswapdrop -- the music-dsp mailing list and website: subscription info,
FAQ, source code archive, list archive, book reviews, dsp links
http://shoko.calarts.edu/musicdsp/
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From: David Ogborn
To: csound@maths.ex.ac.uk
Date: Thu, 26 Aug 1999 20:43:54 -0500
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Subject: Filter Design
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Hello,
Does anyone on the list have particular suggestions for reference
books on the subject of digital filter design?
David Ogborn
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Date: Thu, 26 Aug 1999 18:53:20 -0700
From: Sean Costello
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Hi all (especially hi list administrator):
Is there a limit to the size of a message that can be posted to this list? I
tried posting an orchestra/score combination to the list earlier today, and it
hasn't shown up yet. The message was about 11 Kb of text (lots of comments,
which it took me the better part of today to do).
Thanks,
Sean Costello
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Date: Fri, 27 Aug 1999 03:27:11 +0000
From: pete moss
Organization: pete moss GmbH
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To: Michael Gogins ,
"csound@maths.ex.ac.uk"
Subject: Re: Watch out for Noise
References:
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Michael Gogins wrote:
>...called "Noise" ... greater precision, less noise.
so you are going to use noise to make less noise?
:P
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Date: Fri, 27 Aug 1999 00:54:51 -0400
From: "Job M. van Zuijlen"
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Don't forget that you you may need silence too ;-)
Job
pete moss wrote:
>
> Michael Gogins wrote:
> >...called "Noise" ... greater precision, less noise.
>
> so you are going to use noise to make less noise?
>
> :P
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Date: Thu, 26 Aug 1999 23:44:51 -0700
From: Sean Costello
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Subject: Orchestras/Scores/MP3s Online
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Hi all:
I tried posting a largish orchestra and score to the list today, but it didn't
come through. I will not be stopped so easily. I have set up a page for some of
my Csound work at
http://www.haint.com/sounds
It contains orchestras and scores for some of my compositions and instruments,
as well as MP3s of the same. Be sure and check out stringphaser.orc and
stringphaser.sco - I just finished these today (these implement a digital
recreation of the string ensemble/phase shifter combination that was common in
many musical spheres during the 1970's and early 1980's). The other compositions
and instruments date from last December to this June. Feel free to send me any
feedback you see fit to send, as this is a rough draft web site.
Sean Costello
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Date: Fri, 27 Aug 1999 00:53:37 -0700 (PDT)
From: Ungur Adrian-Alin
Subject: HPKComposer on Linux
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Dear Sir or madam,
Is there anybody out there who have any ideea how could I install
HPKComposer on my Linux Red Hat 6.0 system ? I tried to make it work
under JDK117_v3 and JRE1.2pre-v2 (glibc 2.1) with swing version 1.1
stolen from jMax's 2.3.16 archive (by the way, is this a good ideea ?).
I am sure it is not a CSound problem (I got a 3.49 version running
without problems under Cecilia).
I will be very happy if someone smarter than me in Java and HPKC could
give me the exact scenario for making the whole stuff work.
Thanks
Best regards,
Ungur Adrian-Alin
ualin@yahoo.com
__________________________________________________
Do You Yahoo!?
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From: Peter =?iso-8859-1?Q?Neubäcker?=
Date: Fri, 27 Aug 99 10:00:20 +0200
To: csound@maths.ex.ac.uk
Subject: Re: Phasor notes
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The name of the opcode may remind you
of an effect box called phasor that modifies
a given sound the way you have in mind.
In fact the phasor opcode has NOTHING
to do with that effect box.
Please forget what you think it should be
and read the manual and the descriptions
I and others posted to you again without
preoccupation!
Peter
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From: richard bowers
To: csound csound
Subject: Fw: Phasor notes
Date: Fri, 27 Aug 1999 09:25:19 +0100
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I'm sure you'll get more authoritative answers from other users but my
experience with phasor can be summed up as follows:
Phasor produces a stream of values at a-rate (k-rate also?) running from 0
to 1 and can cycle through these at a cps rate. The output can then be used
to index a function table (such as one containing a sinusoid - although
other shapes are possible) via the table opcodes. In other words, it can
enable your instrument to read values from a table at the audio (or k?) rate
and keep doing so at a specified cps (ie if cps is 1 then phasor will read
through the table once every second; if cps is 440 you'll get a pitch).
That's how it simulates an oscillator. It can, like so much of CSound, be
used in other ways such as indexing the analysis files for phase vocoders.
I'm not sure how your using it but if its being used on its own you will
only get a signal extending to the value of 1 - which is virtually silence.
If you were to amplify this signal by a large number you would, I imagine,
get some sort of sawtooth wave at the specified cps.
Hope that helps,
Richard.
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Date: Fri, 27 Aug 1999 09:37:40 +0100
From: James Andrews
To: csound@maths.ex.ac.uk
Subject: new to list
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----- Forwarded message from "Prof. Clive Greated" -----
Comments: Authenticated sender is
From: "Prof. Clive Greated"
To: csound-approval@maths.ex.ac.uk
Date: Wed, 25 Aug 1999 13:52:13 +0000
Subject: contact required
Priority: normal
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Subject: Interested to keeping in touch
Date sent: Wed, 25 Aug 1999 13:43:44
I am a staff member at Edinburgh University but am doing a part time
PPhil degree in electro-music composition and I plan to do much of my
work in CSound so I would very much appreciate being kept in touch
with any new developments. Also I have one specific question. I want
to write Score files in different temperaments (not equal
temperament). Is this possible? Clive
----- End forwarded message -----
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