| I've been trying to write a macro with 3 arguments: start time, start
cutoff, and end cutoff (for a filter sweep). It looks like this:
#define SQUARE4(T#SF#EF) #
i1 [$T+0.00] .5 5 6.00 3 $SF 3 1 5
i1 [$T+0.50] .25 5 6.00 3 [$SF+0.50*($EF-$SF)/4] . . .
i1 [$T+0.75] .5 5 6.00 3 [$SF+0.75*($EF-$SF)/4] . . .
i1 [$T+1.25] .5 5 6.00 3 [$SF+1.25*($EF-$SF)/4] . . .
i1 [$T+1.75] .25 5 6.00 3 [$SF+1.75*($EF-$SF)/4] . . .
i1 [$T+2.00] .75 5 6.00 3 [$SF+2.00*($EF-$SF)/4] . . .
i1 [$T+2.75] .25 5 6.00 3 [$SF+2.75*($EF-$SF)/4] . . .
i1 [$T+3.00] .5 5 6.00 3 [$SF+3.00*($EF-$SF)/4] . . .
i1 [$T+3.50] .5 5 6.00 3 [$SF+3.50*($EF-$SF)/4] . . .
#
p7 is the cutoff frequency. I call the macro like this:
$SQUARE4(0#3#5)
...and get this output from perf:
Incorrect evaluation.
FATAL ERROR: terminated perf
I'm using version 3.51 for PowerPC.
I've tried setting all p7s to a constant value, and then it compiled
without any problems. So there must be something wrong with the expression
[$SF+0.50*($EF-$SF)/4].
The weird thing is that I used almost exactly the same expression for
another similar macro (dividing by 8 instead of 4). I'm sorry to waste your
time with something that probably is very simple, but I just can't make it
work. If anyone knows what I'm doing wrong, please tell me.
--ll
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From: Sergey Batov
To: csound@maths.ex.ac.uk
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Subject: WaveIn in DirectCsound
Date: Mon, 8 Feb 1999 02:03:16 +0300
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I try to do something with DirectCsound and microphone.
For the beginning I'd like to print 'rms' of wave in.
; orc
sr=44100
kr=4410
ksmps=10
instr 1
asig in
k_in rms asig
print k_in
endin
; sco
f0 200
with command line:
csound.exe -i -n -b512 --log.txt wavein.orc wavein.sco
But there was no result.
Need I to organize an of infinitive loop? How?
Thank you,
Sergey Batov batov@glasnet.ru
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From: Sergey Batov
To: csound@maths.ex.ac.uk
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Subject: Announce: Bitmap Player
Date: Sun, 7 Feb 1999 23:42:42 +0300
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Hi!
I'd like to introduce to you Bitmap Player. This is a simple program for
convering bitmap images (Windows format without compression) to sounds
(Wav-files).
This program was made by Victor Khashtchansky < father@vsoft.spb.ru >.
He is looking for ways for further development and perfection of his
program,
so he needs your remarks or advices.
The program is available on
http://www.the7hills.com/bmp2wav
Regards,
Sergey Batov batov@glasnet.ru
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From: Sergey Batov
To: Josh Steiner
Cc: csound@maths.ex.ac.uk
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Subject: Re: Announce: Bitmap Player
Date: Mon, 8 Feb 1999 09:04:31 +0300
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Sorry, the link
http://www.the7hills.com/bmp2wav/b2wav.zip
did not works because of capital letter in the name of B2wav.zip on the
server
(it's case sensitive).
Now I fixed it.
Happy downloading!
Sergey Batov batov@glasnet.ru
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From: Servando Valero Castieira
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Thanks for your support. I'll send you the translation as soon as I finish (though I
guess it'll take a while...)
Un piacere leggere il tuo messaggio. Grazie tante per tutto (e per DirectCsound 2.6
beninteso)
Servando
Gabriel Maldonado wrote:
> I think that a spanish translation of Csound manual is a GREAT idea!
>
> When you finish the translation, post me the manual, I will place it in my sites for
> public dowloading.
>
> Any volunteer for an italian translation too?
>
> --
> Gabriel Maldonado
>
> http://www.agora.stm.it/G.Maldonado/home2.htm
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From: jpff@maths.bath.ac.uk
To: rwd@cableinet.co.uk
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In-reply-to: <36BB4D6B.4BDC815B@cableinet.co.uk> (message from Richard Dobson
on Fri, 05 Feb 1999 19:58:35 +0000)
Subject: Re: AIFF files in sndinfo etc
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What must teh text be even length? I cannot see any reason for it.
Anyway I have extended teh AIFF header reader to deal with those
chunks.
==John
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Subject: Re: Spanish translation?
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>>>>> "Servando" == Servando Valero Castieira writes:
Servando> By the way, if I translate the manual
Servando> I won't be able to dispose of a server to place the file, so is there
Servando> any little room over there for it?
No problem in finding space yet. If we had translations into every
language, then there might be problems......
==John ff
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From: Gabriel Maldonado
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Try to call DirectCsound with the following flags:
csound.exe -+K -+q -+e -+p12 -b500 -i wavein.orc wavein.sco
increase the buffer length if necessary
--
Gabriel Maldonado
http://www.agora.stm.it/G.Maldonado/home2.htm
Sergey Batov wrote:
>
> I try to do something with DirectCsound and microphone.
> For the beginning I'd like to print 'rms' of wave in.
>
> ; orc
> sr=44100
> kr=4410
> ksmps=10
>
> instr 1
> asig in
> k_in rms asig
> print k_in
> endin
>
> ; sco
> f0 200
>
> with command line:
> csound.exe -i -n -b512 --log.txt wavein.orc wavein.sco
>
> But there was no result.
> Need I to organize an of infinitive loop? How?
>
> Thank you,
>
> Sergey Batov batov@glasnet.ru
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From: pj richardson
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I'm trying to use csound to process pre-recorded speech to emulate the
classic Texas Instruments speak&spell sound.
Here's an example of the speak&spell speech: (87k, 16bit 8khz)
http://members.xoom.com/icefog/Speakn.au
(your browser's plugin might not play this correctly, you may have to
download it.)
The lpc or melp compression schemes sound as though they'd serve my
purposes, and I suspect there's a way to coax it out of csound, though
as a first-time user I'm a bit intimidated.
More samples of lpc and melp are at:
http://www.ti.com/sc/docs/msp/speech/demo.htm
Info on the Speak&Spell:
http://www.ti.com/corp/docs/history/pmosnf.htm
If anyone has suggestions, hints, pointers toward tutorials or relevant
examples, or any previous experience with this sort of thing I'd be very
grateful for your help.
PJ Richardson
pjrich@ou.edu
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From: jpff@maths.bath.ac.uk
To: valero@cdz.servicom.es
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In-reply-to: <36BC5488.E544F955@cdz.servicom.es> (message from Servando Valero
Castieira on Sat, 06 Feb 1999 15:41:13 +0100)
Subject: Re: Spanish translation?
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>>>>> "Servando" == Servando Valero Castieira writes:
Servando> By the way, if I translate the manual
Servando> I won't be able to dispose of a server to place the file, so is there
Servando> any little room over there for it?
No problem in finding space yet. If we had translations into every
language, then there might be problems......
==John ff
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From: Servando Valero Castieira
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The bigger the translation team the longer my walks along the beach!
I'm no way interested in doing all the work by myself, it's just I'm new =
in
the list and I didn't know your proposal. So, Dami=E1n, if you want to
translate a half PLEASE contact me and we can discuss on the format and
unify translation criteria. Maybe someone else feels like joining us?!
Anyway, after all the e-mails I've been sent off list (thank you all) I'm
sure that a spanish translation of the manual is really needed, and as I =
have
time off now to concentrate on the job (before going to the US in March=
), I
will do it. But, OF COOOOURSE, help is always welcome!
Servando
Damian Keller wrote:
> Sometime ago I suggested to do the Spanish version as team work. This h=
as
> the advantage of letting each one concentrate on his favorite area, and
> gets some work off the shoulders of one person. (Unless he'll get paid =
to
> do it, of course!).
>
> So again, I offer myself to do part of that manual.
>
> ________________________________
> Damian (accent on 2nd a) Keller
> http://www.sfu.ca/~dkeller
> earsay:
> http://www.earsay.com
> ________________________________
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From: David Boothe
To: "'pjrich@ou.edu'"
Cc: "Csound (E-mail)"
Subject: RE: lpc tweaking
Date: Mon, 8 Feb 1999 09:34:48 -0600
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In Csound, the 'lpanal' utility is used to create an analysis file of the
pre-recorded speech to reproduce. The resulting file is used as an input to
the 'lp...' family of opcodes to synthesize the speech, using subractive
synthesis, often with buzz or gbuzz as a source.
Read about these the 'lp...' opcodes and the 'lpanal' utility in the manual.
Also, Dodge/Jerse, "Computer Music" (2nd edition) talks about it.
-David.
P.J. Richardson wrote:
> I'm trying to use csound to process pre-recorded speech to emulate the
> classic Texas Instruments speak&spell sound.
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Date: Mon, 08 Feb 1999 11:02:17 -0800
From: Jonathan Drexler
To: csound@maths.ex.ac.uk
Cc: wright@astro.ocis.temple.edu, rwd@cableinet.co.uk,
phoekje <@ucla.smtp:phoekje@bw.edu>
Subject: Filter clicks: further comment
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I posted a question a couple of weeks ago about
a problem I had with filter clicks in legato
passages and some generous people offered their
responses. I have come up with yet another
solution for this. Apparently the problem is
that, on every new note, when the filter is
reinitialized, it resets the phase to zero, causing
the offending click. This undoes any good which might
have been accomplished by setting the iphs switch
in the oscillator to a negative value:
iPhase=-1
a1 oscil kAmp,kCps,kTbl,iPhase ;-1 skips phase-initialization
f1 tonex a1,kCtf,iNumLayer,1
and setting the filter's "istor" parameter does not
seem to affect this.
Using an igoto seems to solve this:
iPhase=-1
a1 oscil kAmp,kCps,kTbl,iPhase ;-1 skips phase-initialization
if iPhase=-1 igoto SkipFltInit
f1 tonex a1,kCtf,iNumLayer,1
SkipFltInit:
The other solutions people wrote about work quite well, but
this allows you to keep the filter inside the instrument if that
is what you want, and still get a true legato.
Thanks to everyone for responding to my earlier
question.
--Jonathan
!
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Date: Mon, 08 Feb 1999 19:22:34 +0000
From: Richard Dobson
Organization: Composers Desktop Project
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To: csound@maths.ex.ac.uk
Subject: Re: AIFF files in sndinfo etc
References: <36BB4D6B.4BDC815B@cableinet.co.uk>
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It is mandated in the AIFF spec; processors such as the 68000 (possibly
Power PC as well?) cannot read directly from odd addresses, so the
format was designed to be friendly to those machines.
Future formats may even require alignment to 64bits - SDIF is one such,
and even Microsoft may be moving away from byte-alignment for Win64.
Richard Dobson
jpff@maths.bath.ac.uk wrote:
>
> What must teh text be even length? I cannot see any reason for it.
> Anyway I have extended teh AIFF header reader to deal with those
> chunks.
> ==John
--
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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Date: Mon, 8 Feb 1999 18:57:25 +0000
To: csound@maths.ex.ac.uk
Subject: Re: WaveIn in DirectCsound
Message-ID: <19990208185724.E281@scot-mur.demon.co.uk>
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In-Reply-To: ; from Sergey Batov on Mon, Feb 08, 1999 at 02:03:16AM +0300
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On Mon, Feb 08, 1999 at 02:03:16AM +0300, Sergey Batov wrote:
> I try to do something with DirectCsound and microphone.
> For the beginning I'd like to print 'rms' of wave in.
>
> ; orc
> sr=44100
> kr=4410
> ksmps=10
>
> instr 1
> asig in
> k_in rms asig
> print k_in
> endin
>
> ; sco
> f0 200
try using
;sco
i1 0 200
:)
Regards
Rob
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From: Anders Andersson
Reply-To: Anders Andersson
To: CSound list
Date: Mon, 08 Feb 1999 13:37:13 +0100
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Hi there!
I wonder if anyone can give me some tips about how to optimize instruments!
Take this very simple example:
k1 = [blah]
k2 = [blah]
a1 oscil k1*k2-sin(k1),1,1
Would it be faster to insert a k-rate variable to hold the calculation
k1*k2-sin(k1)?
k1 = [blah]
k2 = [blah]
k3 = k1*k2-sin(k1)
a1 oscil k3,1,1
Or does CSound evaluate the expression in k-rate if it contains k-rate
variables only?
Example 2:
kvol linseg 0, 1, p4/3, p3, p4/3, 1, 0
Would this be slower than:
itemp = p4/3
kvol linseg 0, 1, itemp, p3, itemp, 1, 0
(well, i guess not! :D)
.--- -- - -
| Anders "Pipe/Nature" Andersson, pipe@algonet.se
| Proud member of the Amiga community, Nature and Mensa Sweden.
:
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Date: Tue, 09 Feb 1999 18:18:49 +1100
From: Bob Douglas
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Anders Andersson wrote:
> k1 = [blah]
> k2 = [blah]
> a1 oscil k1*k2-sin(k1),1,1
>
> Would it be faster to insert a k-rate variable to hold the calculation
> k1*k2-sin(k1)?
> k1 = [blah]
> k2 = [blah]
> k3 = k1*k2-sin(k1)
> a1 oscil k3,1,1
I haven't done any tests but logically I would imagine only if the same
calculation for a1 is to be repeated later for, say, a different f-table or
phase value, or, if k3 is used within another expression. Even then, I think
there would need to be many recalculations and k3 would have to be a fairly
complex expression (comprising higher order functions) for any detectable
difference to show up on modern computers.
> Or does CSound evaluate the expression in k-rate if it contains k-rate
> variables only?
>
> Example 2:
>
> kvol linseg 0, 1, p4/3, p3, p4/3, 1, 0
>
> Would this be slower than:
>
> itemp = p4/3
> kvol linseg 0, 1, itemp, p3, itemp, 1, 0
The latter would be faster because the calc p4/3 has to be done twice in the
former, and only once in the latter.
The real optimisation occurs less in the realm of machine speed and more in
the fact that if you decide to change the way you calc itemp, then you only
have to alter its definition once. The alternative is wading through the whole
instrument finding (and possibly missing) every instance of p4/3.
Bob Douglas
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From: Anders Andersson
Reply-To: pipe@algonet.se
To: CSound list
Date: Tue, 09 Feb 1999 06:56:49 +0100
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Subject: SID - Resonance data?
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Hello! I'm doing an instrument that will 'emulate' the 6581SID (the famous
music-generating chip in the Commodore64)
My problem is that I can't find any datas about how the resonance is
implemented.
If anyone has any information or links I would be grateful!
.--- -- - -
| Anders "Pipe/Nature" Andersson, pipe@algonet.se
| Proud member of the Amiga community, Nature and Mensa Sweden.
:
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Date: Tue, 9 Feb 99 8:28:25 GMT
From: jpff@maths.bath.ac.uk
Subject: Re: Filter clicks: further comment
To: csound@maths.ex.ac.uk
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Message written at 9 Feb 1999 08:18:12 +0000
--- Copy of mail to jdrexler@support.ucla.edu ---
I have looked at the code, and loosely it is wrong when not
reinitialising. It does not set the initial phase state to zero, but
it always creates a new vector to store the values, thereby throwing
away the previous state. The code fix is easy and I have made it
here. Affects tonex, atonex and resonx.
==John ffitch
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From: jpff@maths.bath.ac.uk
Subject: Re: Filter clicks: further comment
To: csound@maths.ex.ac.uk
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Message written at 8 Feb 1999 21:52:02 +0000
--- Copy of mail to jdrexler@support.ucla.edu ---
Are you really asking that tonex have a 'skip initialisation' option?
I implemented that for biquad the other night. Any others we have missed?
==John ffitch
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From: Sergey Batov
To: csound@maths.ex.ac.uk
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Subject: Re: WaveIn in DirectCsound
Date: Tue, 9 Feb 1999 12:31:18 +0300
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Hi,
now I'm using the following orc+sco
;wavein.orc
sr=44100
kr=4410
ksmps=10
instr 1
asig in
k_in rms asig
print k_in
out asig
endin
;wavein.sco
i1 0 200
and the command line for DirectCsound:
csound.exe -+q -+e -+p12 -b500 -i --log.txt wavein.orc wavein.sco
I can hear sound from microphone but only one meaning of k_in (=0.000) was
printed.
(It's understood.)
If to add "midi in" flag -+K to this command line I can get
renewed meanings of k_in but only pressing keys on midi keaboard.
It's understood too.
Now it seems the entailment of my idea is near.
I'd like to control k_in (to print, at least) every 't' seconds or,
say, with kr rate.
I tried to use 'goto' but did not obtain necessary result.
Can anybody help, please?
Best regards,
Sergey Batov batov@glasnet.ru
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Date: Tue, 9 Feb 1999 12:32:15 +0200 (IST)
From: Erez Webman
To: Richard Dobson
cc: Extended Csound Mailing List
Subject: Re: Sharc and "extended csound" ?
In-Reply-To: <36A78150.2477ECD2@cableinet.co.uk>
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Richard,
Do you know whether the non-beta version of XTCsound is available? I'm
still using the old beta version which is just a little bit too buggy for
me.
Thanks,
Erez Webman
On Thu, 21 Jan 1999, Richard Dobson wrote:
> There are webpages for XTCsound on the ADI site, but they are a little
> hard to find!
> try
>
> http://www.analog.com/industry/systems_solutions/audio/xtcsound.html
>
> and
>
> http://www.analog.com/publications/documentation/C_Sound_Manual/csoundman.html
>
> for the SDK manual.
>
> I have an SDK, and it all basically works very well, though the device
> drivers are somewhat fragile. New 32bit drivers are on the way. We have
> even been told recently that a multi-SHARC version was developed 'for a
> customer'. In the SDK card XTCsound is downloaded as a SHARC executable,
> linked with whatever orchestra modules you have designed.
>
> Richard Dobson
>
>
> Tim Duncan wrote:
> >
> > Extended CSound does indeed exist. You are right about the opcode set,
> > which contains both traditional opcodes and new ones, in particular,
> > opcodes supporting real-time processing. An example is a "harmon," which is
> > a real-time harmonizer.
> >
> > As far as I know, Extended CSound is only available in ROM on the Sphinx card,
>
> >
>
>
> --
> 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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