| After playing around with tone and atone, some things puzzle me:
(1) Why is a 0-value ouput signal amplitude obtained when noise (with a
substantial amplitude, say 16000) is passed to tone where khp = 0?
According to the manual, "the variable khp (in cps) determines the response
curve's half-power point. Half power is defined as peak power / root 2.".
So... if noise with an input amplitude of 16000 is passed to tone thus:
afilteredoutput tone anoiseinput,0 ;khp = 0cps
shouldn't the output signal amplitude be *half* of the peak power (=
16000/2 = 8000)? Compiling this example gives me an output signal
amplitude of 0 rather than 8000, suggesting that khp is not the halfpower
point of the response curve, but the point on the response curve beyond
which no input signal is allowed to pass at all. Confusing.
(2) Conversely, why is it not possible to obtain a 0-value ouput amplitude
signal using atone (when such an output *is* possible using tone)?
The manual says "atone, areson are filters whose transfer functions are the
*complements* of tone and reson. atone is thus a form of high-pass
filter...". However, upon implementing the inverse situation described
above:
afilteredoutput atone anoiseinput,22050 ;where khp = sr/2 = 44100/2.
an ouput signal amplitude of 3368.9 is produced (according to the output
terminal) rather than an output signal amplitude of 0 as one would expect
from the experiment with tone above.
Even more confusing, if the sample rate of the piece is changed (say to
10000) then the output signal amplitude is different yet:
afilteredoutput atone anoiseinput,5000 ;where khp = sr/2 = 10000/2.
an output signal amplitude of 3269.2 (rather than 3368.9 (as above) or 0
(as expected)) is produced.
(3) So, apparently, no 0-value output signal amplitude is possible from
atone in the way that 0-value output is possible from tone; moreover, this
has something to do with the sample rate of the piece. Is atone really the
"complement" of tone?
(4) Assuming one knows the input signal amplitude of the signal passed to
tone or atone, how does one predict the output (filtered) signal amplitude?
Output signal amplitude from both tone and atone seems to be a function of
a number of things:
(a) the input signal amplitude.
(b) the halfpower point of the filter (khp in the manual).
(c) the sample rate of the piece.
I was expecting (a) to be the only relevant factor (greater input amp ->
greater output amp); I'm surprised that (b) and (c) are pertinant. Can
anyone provide an exact way (formula) to determine the output signal
amplitude of a signal coming from tone or atone (given that one knows input
signal amp, khp & sr)? By plugging in numerous closely spaced points
(brute force with a short program written in c) I know that the response
curves of both tone and atone are vaguely exponential curves symmetric
about integral multiples of (sr/2)... but I'd like to know how to predict
exact ouput signal amplitude values rather than knowing just the general
shape of the curve.
Any insight would be greatly appreciated, and I apologize for the length of
the letter. Also, I realize that most of my concern about the output
signal amplitude can be assuaged by using balance, but I really want to
understand the working of tone and atone without the aid of balance before
throwing other opcodes into the mix.
Cheers,
Trevor Baca.
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa09580;
19 May 97 19:59 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa25487;
19 May 97 20:04 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Mon, 19 May 1997 20:04:26 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (TAA13897);
Mon, 19 May 1997 19:54:27 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Mon, 19 May 97 19:54:19 +0100
Received: from Thuban.AC.HMC.Edu [134.173.53.8] by hermes via ESMTP (TAA13881);
Mon, 19 May 1997 19:54:06 +0100
Received: from [134.173.38.61] (Spjut.Eng.HMC.Edu)
by THUBAN.AC.HMC.EDU (PMDF V5.1-5 #18017) with ESMTP
id <01IJ1WXR261CE46E0U@THUBAN.AC.HMC.EDU>
for csound@noether.ex.ac.uk; Mon, 19 May 1997 11:54:01 PST
Date: Mon, 19 May 1997 12:00:22 -0700
From: Erik Spjut
Subject: Re: lowpass and highpass filters: 4 questions (long).
In-Reply-To:
X-Sender: spjut@thuban.ac.hmc.edu
To: Trevor Baca , csound@maths.ex.ac.uk
Message-Id:
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
At 1:08 PM -0500 5/19/97, Trevor Baca wrote:
>After playing around with tone and atone, some things puzzle me:
>
>(1) Why is a 0-value ouput signal amplitude obtained when noise (with a
>substantial amplitude, say 16000) is passed to tone where khp = 0?
Tone does not work for khp=0. Try khp=0.001 or khp=0.0000001 or some such
if you truly want a cutoff close to 0. I don't know what you're trying to
do but "integ" may come closer fro very low cutoff frequencies.
>(2) Conversely, why is it not possible to obtain a 0-value ouput amplitude
>signal using atone (when such an output *is* possible using tone)?
Because tone does not work for khp=0 and atone isn't really designed for
khp=sr/2. Compare tone at khp=1 with atone at khp=sr/2-1. Again "diff"
instead of atone may do what you want.
>(3) So, apparently, no 0-value output signal amplitude is possible from
>atone in the way that 0-value output is possible from tone; moreover, this
>has something to do with the sample rate of the piece. Is atone really the
>"complement" of tone?
For reasonable (i.e. non-0 non-sr/2) valuse of khp, yes!
>(4) Assuming one knows the input signal amplitude of the signal passed to
>tone or atone, how does one predict the output (filtered) signal amplitude?
In order to do this calculation one must also know the frequency content of
the input signal and the response-curve of tone and atone. One then
convolves the two. (If the magnitude functions are available in a dB scale,
one can get an estimate by adding up the values for the two curves in dB).
If you don't know how to do these calculations I suggest you check a
reference on signal processing.
One thing to remember is that all digital filters exhibit aliasing in their
frequency response curves. The closer you get to khp=sr/2 the less ideal
the filter response becomes.
-----------------------------------------------------------------------------
Erik Spjut (rhymes with cute) - Associate Professor of Engineering and/or
Associate Director for Engineering Computing, The Center for Design Education
Harvey Mudd College, Claremont, CA 91711 USA
Erik_Spjut@hmc.edu Ph & Voice mail (909) 607-3890 Fax (909) 621-8967
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa09609;
19 May 97 20:09 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa25883;
19 May 97 20:14 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Mon, 19 May 1997 20:14:25 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (UAA14508);
Mon, 19 May 1997 20:05:46 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Mon, 19 May 97 20:05:40 +0100
Received: from [144.91.3.20] by hermes via SMTP (UAA14499);
Mon, 19 May 1997 20:05:37 +0100
Received: (qmail 17221 invoked by uid 2928); 19 May 1997 19:05:26 -0000
Date: Mon, 19 May 1997 12:05:26 -0700 (PDT)
From: Mike Berry
To: Trevor Baca
Cc: csound@maths.ex.ac.uk
Subject: Re: lowpass and highpass filters: 4 questions (long).
In-Reply-To:
Message-Id:
Mime-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
The answer to your first question is that khp is the *frequency*
of the half power point. So by putting in 0, you are filtering everthing
above 0 Hz., naturally resulting in no output.
This should impact all of your other questions, but I would
mention that tone and atone do not perform well without balance, so I
would include that in your experimentation.
Mike Berry
mikeb@mills.edu
http://www.mills.edu/PEOPLE/gr.pages/mikeb.public.html/mikeb.homepage.html
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa09874;
19 May 97 21:16 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa28596;
19 May 97 21:21 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Mon, 19 May 1997 21:21:04 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (VAA17624);
Mon, 19 May 1997 21:09:19 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Mon, 19 May 97 21:09:13 +0100
Received: from typhoon.dial.pipex.net [158.43.128.27] by hermes
via ESMTP (VAA17621); Mon, 19 May 1997 21:09:12 +0100
Received: from ai098.du.pipex.com (193.130.248.98)
by typhoon.dial.pipex.net (8.8.2/UUNET PIPEX simple 1.29) id VAA06506;
Mon, 19 May 1997 21:09:19 +0100 (BST)
Message-Id: <3380B290.18B5@dial.pipex.com>
Date: Mon, 19 May 1997 21:05:36 +0100
From: p robinson
Reply-To: project.x@dial.pipex.com
Organization: Project X
X-Mailer: Mozilla 3.01Gold (Win95; I)
Mime-Version: 1.0
To: csound@maths.ex.ac.uk
Subject: decay & Amplitude problems
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
Can anyone tell me if it's possible to have:
1. The note duration (more to the point the decay) variable on the pitch
contained in a Cscore.
2. Changing Harmonic amplitudes variable on Amplitude of a specific
note. ( say 3 amplitude "ranges" with seperate (GEN 10) Harmonic recipes
for each range)
Thanks,
Pat.
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa10215;
19 May 97 23:05 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id ab04296;
19 May 97 23:10 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Mon, 19 May 1997 23:10:30 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (XAA23379);
Mon, 19 May 1997 23:04:48 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Mon, 19 May 97 23:04:42 +0100
Received: from GS160.SP.CS.CMU.EDU [128.2.203.172] by hermes
via SMTP (XAA23369); Mon, 19 May 1997 23:04:34 +0100
Message-Id: <199705192204.XAA23369@hermes>
Subject: Re: lowpass and highpass filters: 4 questions (long).
To: Csound mailing list
Date: Mon, 19 May 1997 18:04:13 -0400 (EDT)
From: Eli Brandt
In-Reply-To: from "Trevor Baca" at May 19, 97 01:08:47 pm
X-Portmanteau: pantryptaminergeticallysisterrainbowtie
X-Mailer: ELM [version 2.4 PL23]
Mime-Version: 1.0
Content-Type: text/plain; charset=US-ASCII
Content-Transfer-Encoding: 7bit
Content-Length: 1020
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
Trevor Baca wrote:
> So... if noise with an input amplitude of 16000 is passed to tone thus:
>
> afilteredoutput tone anoiseinput,0 ;khp = 0cps
>
> shouldn't the output signal amplitude be *half* of the peak power (=
> 16000/2 = 8000)? Compiling this example gives me an output signal
> amplitude of 0 rather than 8000, suggesting that khp is not the halfpower
> point of the response curve, but the point on the response curve beyond
> which no input signal is allowed to pass at all. Confusing.
As a first approximation, think of "tone" as having no effect below
the cutoff frequency, and attenuating above it. The amount of
attenuation is proportional to how many *octaves* the signal is above
the cutoff. (6 dB/octave for a one-pole filter, specifically.)
But 0 Hz is at minus infinity on the octave scale. If you pick this
as the cutoff, nothing will get through. (Except DC, and in practice
that probably won't work either.)
--
Eli Brandt | eli+@cs.cmu.edu | http://www.cs.cmu.edu/~eli/
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa10339;
20 May 97 0:02 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa06855;
20 May 97 0:08 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Tue, 20 May 1997 00:07:58 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (AAA26382);
Tue, 20 May 1997 00:04:00 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Tue, 20 May 97 00:03:53 +0100
Received: from ella.mills.edu [144.91.3.20] by hermes via SMTP (AAA26373);
Tue, 20 May 1997 00:03:42 +0100
Received: (qmail 14813 invoked by uid 2928); 19 May 1997 23:03:31 -0000
Date: Mon, 19 May 1997 16:03:31 -0700 (PDT)
From: Mike Berry
To: csound list
Subject: csound ppc Manual fix
Message-Id:
Mime-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
There was a problem with the csRef not opening the manual which I
think I have fixed and put back up on the ftp/web sites. The only change
is in the manual. If you have a resource editor, you can make the change
yourself by changing the creator of the manual to 'CSm$'.
Mike Berry
mikeb@mills.edu
http://www.mills.edu/PEOPLE/gr.pages/mikeb.public.html/mikeb.homepage.html
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa10698;
20 May 97 2:17 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa12012;
20 May 97 2:22 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Tue, 20 May 1997 02:22:23 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (CAA01114);
Tue, 20 May 1997 02:17:03 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Tue, 20 May 97 02:16:56 +0100
Received: from lt@westnet.com [206.24.6.2] by hermes via ESMTP (CAA01102);
Tue, 20 May 1997 02:16:54 +0100
Received: from localhost (lt@localhost) by westnet.com (8.8.5/8.8.5) with SMTP
id VAA19768 for ;
Mon, 19 May 1997 21:16:50 -0400 (EDT)
Date: Mon, 19 May 1997 21:16:50 -0400 (EDT)
From: Larry Troxler
To: csound@maths.ex.ac.uk
Subject: FP under/overflows , other mailing lists?
Message-Id:
Mime-Version: 1.0
Content-Type: TEXT/PLAIN; charset=US-ASCII
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
I've been working with Perry Cook's C++ synthesis toolkit, so this is not
Csound exactly. So first question: is there a more general mailing list
for questions related to software sound synthesis and related theoretical
issues (such as the recent loudness perception discussions, for example)?
Second question: In the course of working with this toolkit attempting
real-time synthesis, I've noticed that on my Linux Pentium platform, I've
noticed that floating point multiplications suddenly become a lot slower
when hitting the bottom-rail (~ 1E-43). This came up in the context of a
waveguide string synthesis, when once the wave decays to a certain point,
the performance drops and can no longer keep up.
So, I assume that the general question here is dealing with floating-point
exceptions. I assume that that once the multiplications bottom out, they
start generating exceptions (SIGFPE?), and that overhead is what is
causing the sudden slowdown.
>From a quick test, it seems that overflows also cause a slowdown, and
1./0. doesn't abort the program but generates a floating-point "Inf".
But the overflow and divide-by-zero doesn't matter, I think the main
issue for us is the undeflow. (By which, I mean the exponent becoming too
negative, not the mantissa).
So I imagine that something by default hooks
the SIGFPE but ignores it. Am I on the right track?
Ideally, I think it would be nice if somehow the coprocessor could be
programmed to just return 0 in these cases, without generating an
exception, but I would geuss that this is impossible. So failing, that,
can someone point me to some information about trapping floating-point
exceptions?
Larry
-- Larry Troxler -- lt@westnet.com -- Patterson, NY USA --
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa10953;
20 May 97 5:06 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id ay17650;
20 May 97 5:12 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Tue, 20 May 1997 03:43:09 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (DAA03193);
Tue, 20 May 1997 03:39:26 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Tue, 20 May 97 03:39:20 +0100
Received: from ulysses.Stanford.EDU [36.49.0.124] by hermes
via ESMTP (DAA03190); Tue, 20 May 1997 03:39:13 +0100
Received: (from tkunze@localhost)
by ulysses.stanford.edu (950413.SGI.8.6.12/950213.SGI.AUTOCF)
id TAA15464; Mon, 19 May 1997 19:40:21 -0700
From: Tobias Kunze
Message-Id: <9705191940.ZM15463@ulysses.stanford.edu>
Date: Mon, 19 May 1997 19:40:21 -0700
In-Reply-To: Larry Troxler "FP under/overflows , other mailing lists?" (May 19, 9:16pm)
References:
Reply-To: t@ulysses.stanford.edu
X-Url: http://www.stanford.edu/~tkunze
X-Mailer: Z-Mail (3.2.3 08feb96 MediaMail)
To: Larry Troxler , csound@maths.ex.ac.uk
Subject: Re: FP under/overflows , other mailing lists?
Mime-Version: 1.0
Content-Type: text/plain; charset=us-ascii
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
| Ideally, I think it would be nice if somehow the
| coprocessor could be programmed to just return 0 in these
| cases, without generating an exception, but I would geuss
| that this is impossible. So failing, that, can someone
| point me to some information about trapping floating-point
| exceptions?
you want to flush these numbers to zero as soon as they underflow.
but whether you can do that in software or not depends on your
platform. look at the discussion regarding MIPS processors at
http://reality.sgi.com/employees/cook/audio.apps/dev/fp.underflow
-Tobias
--
______________________________________________________________________
Tobias Kunze t@kunze.stanford.edu
CCRMA, Stanford University http://www.stanford.edu/~tkunze
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa10982;
20 May 97 5:06 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id be17650;
20 May 97 5:12 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Tue, 20 May 1997 04:28:30 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (EAA04752);
Tue, 20 May 1997 04:25:51 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Tue, 20 May 97 04:25:45 +0100
Received: from sparticus.bright.net [205.212.123.14] by hermes
via ESMTP (EAA04738); Tue, 20 May 1997 04:25:43 +0100
Received: from brutus (root@find4-cs-9.dial.bright.net [205.212.145.189])
by mail.bright.net (8.8.5/8.8.5/FNG) with SMTP id XAA03880;
Mon, 19 May 1997 23:25:30 -0400 (EDT)
Message-Id: <33811ACD.310D7801@bright.net>
Date: Mon, 19 May 1997 23:30:21 -0400
From: Dave Phillips
X-Mailer: Mozilla 3.0 (X11; I; Linux 2.0.29 i486)
Mime-Version: 1.0
To: csound@maths.ex.ac.uk
Subject: Linux Csound 3.46
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
Binaries and source for Linux Csound 3.46 are available at
ftp://ftp.maths.bath.ac.uk/pub/dream/platforms/unix/Linux
and at
ftp://mustec.bgsu.edu/pub/linux
Enjoy...
== Dave Phillips
http://www.bright.net/~dlphilp/index.html
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa11247;
20 May 97 7:08 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa22323;
20 May 97 7:14 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Tue, 20 May 1997 07:14:11 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (HAA08483);
Tue, 20 May 1997 07:05:51 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Tue, 20 May 97 07:05:45 +0100
Received: from root@piglet.cc.utexas.edu [128.83.42.61] by hermes
via ESMTP (HAA08477); Tue, 20 May 1997 07:05:39 +0100
Received: from [208.199.177.131] (32-00-dal.dfwmm.net [208.199.177.131]) by piglet.cc.utexas.edu (8.8.5/8.8.5/piglet.mc-1.4)
with SMTP id BAA29201 for ;
Tue, 20 May 1997 01:04:50 -0500 (CDT)
Message-Id:
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
Date: Tue, 20 May 1997 01:08:09 -0500
To: csound@maths.ex.ac.uk
From: Trevor Baca
Subject: amplitude response of tone & atone (cont)
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
Eli Brandt wrote:
>As a first approximation, think of "tone" as having no effect below
>the cutoff frequency, and attenuating above it. The amount of
>attenuation is proportional to how many *octaves* the signal is above
>the cutoff. (6 dB/octave for a one-pole filter, specifically.)
>
>But 0 Hz is at minus infinity on the octave scale. If you pick this
>as the cutoff, nothing will get through. (Except DC, and in practice
>that probably won't work either.)
This works well. However, plugging a bunch of different values into khp
for tone and atone suggests that the amplitude response curves of tone and
atone look something like these (which raise some questions):
AMPLITUDE RESPONSE CURVE OF TONE
|
|
|
ouput |
amp | /-------------\ <-------- just below input amp value (90-97%)
values | / \
| / \ N.B. the plateau isn't really flat;
| / \ the slope of the rise gradually decreases
|/ \
-------------------------
0 sr/2 sr
Question:
(1) Why doesn't tone's passband allow the input signal to pass at 100%
power? (With a noise input of 10000, the maximum output signal amp is
~9700).
AMPLITUDE RESPONSE CURVE OF ATONE
|
|
| /\ /\ |-------- these output amps are *greater*
ouput |/ \ / \ |-------- than the input amp!
amp | -- -- <-------- input amp value
values | -- --
| -- --
| ------
|
---------------------------
0 sr/2 sr
Questions:
(1) How is it possible that an output signal from atone can have a greater
amplitude than the input signal? (When sr = 10000 and noise with an amp of
10000 is filtered with khp = 400, the output amp is ~12100).
(2) Why does the output signal amp constantly decrease as khp -> sr/2?
Shouldn't the passband that is *not* filtered out output at 100% of the
input power? (When sr = 10000 and noise with an amp of 10000 is filtered
with khp = 2500, the output signal is only 5000 rather than 10000).
Trevor Baca
tbaca@ccwf.cc.utexas.edu
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa11294;
20 May 97 8:00 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa24045;
20 May 97 8:05 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Tue, 20 May 1997 08:05:31 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (IAA10126);
Tue, 20 May 1997 08:00:49 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Tue, 20 May 97 08:00:39 +0100
Received: from luga.latrobe.edu.au [131.172.2.2] by hermes
via ESMTP (IAA10121); Tue, 20 May 1997 08:00:25 +0100
Received: from popeye.latrobe.edu.au (popeye.latrobe.edu.au [131.172.4.60])
by luga.latrobe.edu.au (8.8.4/8.6.9) with SMTP id RAA20114
for ; Tue, 20 May 1997 17:00:18 +1000 (EST)
Received: from [131.172.160.54]
by popeye.latrobe.edu.au (5.65v3.2/1.1.10.5/31Oct96-0441PM) id AA13881;
Tue, 20 May 1997 16:58:36 +1000
Date: Tue, 20 May 1997 16:58:36 +1000
X-Sender: musdjgh@pop.latrobe.edu.au
Message-Id:
In-Reply-To: <337D2A04.2781@ere.umontreal.ca>
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
To: csound@maths.ex.ac.uk
From: David Hirst
Subject: Re: csoundscore to midifile???
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
>Thought I'd never ask...
>
>Anyone?
>
>
I have some very rough tools to do this on a Mac PPC, so if anyone else is
interested I can upload it to our ftp site.
David
David Hirst
Senior Lecturer in Music
Music Department
La Trobe University
Bundoora, Vic 3083
Australia
D.Hirst@latrobe.edu.au
Take a look at the latest issue of Mikropolyphonie, the online contemporary
music journal:
http://farben.latrobe.edu.au/mikropol
or Music's WWW Page:
http://farben.latrobe.edu.au/Music_Docs/MusDeptHomePge.html
Received: from stork2.maths.bath.ac.uk by omphalos.maths.Bath.AC.UK id aa12066;
20 May 97 12:53 BST
Received: from goggins.bath.ac.uk by stork.maths.Bath.AC.UK id aa09380;
20 May 97 12:53 BST
Received: from hermes.ex.ac.uk by goggins.bath.ac.uk with SMTP (PP);
Tue, 20 May 1997 12:52:51 +0100
Received: from noether [144.173.8.10] by hermes via SMTP (MAA02080);
Tue, 20 May 1997 12:18:14 +0100
Received: from hermes.ex.ac.uk by maths.exeter.ac.uk;
Tue, 20 May 97 12:18:02 +0100
Received: from pp@goggins.bath.ac.uk [138.38.32.13] by hermes
via ESMTP (MAA02049); Tue, 20 May 1997 12:18:00 +0100
From: jpff@maths.bath.ac.uk
Message-Id: <199705201118.MAA02049@hermes>
Received: from maths.Bath.AC.UK (actually host omphalos.maths.bath.ac.uk)
by goggins.bath.ac.uk with SMTP (PP); Tue, 20 May 1997 12:17:55 +0100
Date: Tue, 20 May 97 12:17:42 BST
Subject: Re: NeXT, latest sources, sread.c, oload.c
To: csound@maths.ex.ac.uk
Sender: owner-csound-outgoing@maths.ex.ac.uk
Precedence: bulk
Message written at 19 May 1997 22:08:36 +0100
--- Copy of mail to sbrandon@music.gla.ac.uk ---
In-reply-to: <199705141606.RAA09066@clarinet.music> (message from Stephen
Brandon - SysAdmin on Wed, 14 May 97 17:06:52 +0100)
I too get an indexing overflow error with your example. The message
means that the index which is calculated as an integer has changed
value when the index is changed to a short index (a 16bit number of
most machines). Indeed I get
MIT Csound: 3.46 (May 17 1997)
indx=-33196 (ffff7e54); (short)indx = 32340 (7e54)
indexing overflow error
so the negative number becomes positive and so it gives up.
I will investigate further as soon as I get time.
If sread.c fails then there is a bug in NeXT's C library as I am
convinced that the code is correct.
==John ff
|