| Hi all,
As per the joining recommendation, here's a brief bit about me, and my
first question.
I've just started running CSound under Win95 - I prefer Linux for most
things, but all my other music software (VAZ softsythn, MIDI sequencer etc)
is on windows so it seemed silly to have CSound as the only Linux app -
especially when I haven't really configured my sound card under Linux, and
PnP is a pain with my current kernel :-)
I have no formal musical training, but I have some previous experience with
analogue sythns (I owned an EDP Wasp when they first came out). So, I'm
just winging it here.
My first composition project with CSound is going to be based on additive
synthesis, because I have some understanding of that. The idea is that I'm
going to set up a set of sine wave ocillators at different frequencies
using the oscil opcode and vary the amplitude of each over time to give me
a subtly shifting series of tones. I'm sure this has been done before, but
there you go.
OK - so here's my question. The obvious way to do this is with a separate
instrument for each voice, with each instrument having its own linseg
controlling the amplitude of the wave over time. This is the way I'm
working at the moment.
But I'm open to suggestions as to a more generic way of accomplishing this.
Also, to help me with these explorations, I'm planning to build a Java
front end targetted at this kind of additive synthesis. My plan is to have
a window for each channel, with a selectable number of harmonics in each
window. Each harmonic would have a selectable waveform (limited to sine in
the first instance). The time/amplitude function for each voice could then
be drawn into the window, with each harmonic being displayed in a distinct
colour so that you can get a good overview of the whole channel at once.
Once the compoisition is complete the program would then generate the
appropriate orc and sco files.
If there is already something like this, I wouldn't mind being spared the
effort :-) Otherwise, I'll let folks know how it goes.
Many Thanks
Andreas
________________________________________________________________________
Dr Andreas Schöter, Intertrader Ltd
http://www.intertrader.com
Tel: +44(0)131 475 7108, Fax: +44(0)131 475 7109
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To: pipe@algonet.se
Cc: CSound list
Subject: Re: Amiga
Date: Wed, 23 Dec 1998 16:56:07 GMT
Message-ID: <36811f86.7587882@pop.btx.dtag.de>
References:
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From: Jens Reimer
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On Tue, 22 Dec 1998 18:20:25 +0100, you wrote:
>Hi all of you!
>
>I was right now reading some old postings to the list, when I came across an
>old letter from Fabio Bizzetti, who wrote that he had an Amiga.
>
>Now I wonder if there are some MORE ppl on this list other than we two, that
>either run CSound on Amiga, or use it in any other way!
i use an amiga.
i asked your questions about two years ago, but csound and amiga, that was very
rare here. i got an ftp-adresse for a port of csound, but today, i don`t know
about it.
today i use my amiga not very often, but then for some graphic and music again,
if the long awaited version of prostation (a.k.a audiolab, but much better,as i
read on the homepage) has been released.
>
>I thought I was the only one, but apparently not..
now three :-)
Und Tschüss
Jens Reimer
Email:
jens.reimer@t-online.de
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Subject: Re: Newbie Intro & Question
From: James Correa
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Dear Andreas,
I have a Mac application that do almost the same. It's call "Csound
Additive". So, if you like to see it I can give you the adress to download
(I'm not the author, I think that the author is John Burkhart).
Welcome to the list and Happy Holidays.
Cheers,
James
====================
James Correa
Composer
====================
----------
>From: Andreas Schoter
>To: csound@maths.ex.ac.uk
>Subject: Newbie Intro & Question
>Date: Wed, Dec 23, 1998, 3:43 PM
>
>Hi all,
>
>As per the joining recommendation, here's a brief bit about me, and my
>first question.
>
>I've just started running CSound under Win95 - I prefer Linux for most
>things, but all my other music software (VAZ softsythn, MIDI sequencer etc)
>is on windows so it seemed silly to have CSound as the only Linux app -
>especially when I haven't really configured my sound card under Linux, and
>PnP is a pain with my current kernel :-)
>
>I have no formal musical training, but I have some previous experience with
>analogue sythns (I owned an EDP Wasp when they first came out). So, I'm
>just winging it here.
>
>My first composition project with CSound is going to be based on additive
>synthesis, because I have some understanding of that. The idea is that I'm
>going to set up a set of sine wave ocillators at different frequencies
>using the oscil opcode and vary the amplitude of each over time to give me
>a subtly shifting series of tones. I'm sure this has been done before, but
>there you go.
>
>OK - so here's my question. The obvious way to do this is with a separate
>instrument for each voice, with each instrument having its own linseg
>controlling the amplitude of the wave over time. This is the way I'm
>working at the moment.
>
>But I'm open to suggestions as to a more generic way of accomplishing this
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Date: 23 Dec 98 13:35:23 EST
From: Rick Mealey
To: csound@maths.ex.ac.uk
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Subject: amnesia, stupidity, or missing HRTFcompact?
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Hello all... with your indulgence, several questions, the first of which
follows (the others in separate postings).
I am working with the newest version of Csound for PPC (3.48? 3.49? whichever
was most recently posted to Mills College's FTP site). I am excited about the
new opcodes, it's an early Christmas present. I discover with glee that the
McGill project on head transfers is implemented here in a ginchy new opcode
called HRTFER, and with lab smock on, I attempt to send an AIFF file through
the opcode for output. However, Perf promptly chokes, falls down and dies,
saying it cannot find the file HRTFcompact.
My question, then: should I already have this file? (When I extracted the file
after download, it only contained Csound and Perf, or seemed to.) If
HRTFcompact isn't part of that package and I need to download it separately,
from where would I do that? (I went back to Mills' FTP world and looked for
evidence of that file, but found nothing, but I might not have been looking in
the right place.)
Thanks in advance for the bandwidth...
Rick
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From: Rick Mealey
To: csound@maths.ex.ac.uk
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Subject: more missing files? (PPC)
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Hello again:
The second of several questions.
In his explication of the voice opcode and what the function tables are
supposed to be doing, Mr Ffitch writes:
"The files impuls20.aiff, ahh.aiff eee.aiff or ooo.aiff are
suitable for the first of these, and a sine wave for the second..."
I understand this to mean that the opcode will work properly when using these
particular aiff files. But durned if I could find them anywhere on my hard
drive after I downloaded the latest rev to Csound (per previous post, I'm
using the latest and greatest version for PPC as it sits in the FTP area of
Mills College).
Likewise I find neither hide nor hair of the following concepts on the Mills
servers, though maybe I'm looking in the wrong place or else just plain blind.
Mr Ffitch discusses these in conjunction with the new opcodes on the FM Models
page of the manual:
sinewave fwavblnk twopeaks
My guess is that these mnemonics are supposed to represent either single-cycle
functions derived from instances of GEN09, GEN10, OR GEN19 or else yet more
aiff files to be loaded into instances of GEN01. Am I too far off? If not,
then what exactly are the functions needed to produce these timbres-- or,
again, where are these aiff files...?
Thanks again for letting me bend your ears.
Rick
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From: Rick Mealey
To: csound@maths.ex.ac.uk
MMDF-Warning: Parse error in original version of preceding line at UK.AC.Bath.maths.omphalos
Subject: for the future: more than four channels?
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Last question, I promise. As there seems to be wish list activity on the list
lately, I thought I'd add mine, which is short.
Here it is:
Support for more than four channels. Say, a 5.1 algorithm for film work and/or
Foley processing.
I probably shouldn't be asking for this until I can get more than two channels
to work properly (see previous post on HRTFER). But I can't be the only one
thinking about this... can I?...
Right, cheers, thanks a lot, happy Xmas, war is over (at least for Ramadan).
Rick.
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Date: Wed, 23 Dec 1998 14:30:36 -0800 (PST)
From: "Matt J. Ingalls"
To: Rick Mealey
cc: csound@maths.ex.ac.uk
Subject: Re: for the future: more than four channels?
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should be able to do 8 channels.
5.1 is owned by dolby
plus ambisonics (Bformat) is better (WXYZ)
perhaps joe anderson might post his ambisonics decoder orchestra/cecelia
module...
anyone hear his new piece?
matt
On 23 Dec 1998, Rick Mealey wrote:
> Last question, I promise. As there seems to be wish list activity on the list
> lately, I thought I'd add mine, which is short.
>
> Here it is:
>
> Support for more than four channels. Say, a 5.1 algorithm for film work and/or
> Foley processing.
>
> I probably shouldn't be asking for this until I can get more than two channels
> to work properly (see previous post on HRTFER). But I can't be the only one
> thinking about this... can I?...
>
> Right, cheers, thanks a lot, happy Xmas, war is over (at least for Ramadan).
>
> Rick.
>
> ____________________________________________________________________
> More than just email--Get your FREE Netscape WebMail account today at http://home.netscape.com/netcenter/mail
>
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Files for sound modelling opcodes:
ftp://ftp.maths.bath.ac.uk/pub/dream/documentation/sounds/modelling/
HRTFcompact is at:
ftp://ftp.maths.bath.ac.uk/pub/dream/utilities/Analysis/
or at the mirror
ftp://ftp.musique.umontreal.ca/pub/mirrors/dream/ ...etc
re
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From: Michael Gogins
To: csound@maths.ex.ac.uk, Andreas Schoter
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Use hetro (analysis utility) and adsyn (synthesis opcode) for time/frequency
analysis and resynthesis using a filterback analysis and oscillator bank
resynthesis.
You can read the analysis files with Java, monkey with them, and write them
back again. You can even translate other data (pictures, for example) into
analysis files and synthesize them.
My software Silence (currently in interminable beta release) does all of
this. http://ww.pipeline.com/~gogins.
-----Original Message-----
From: Andreas Schoter
To: csound@maths.ex.ac.uk
Date: Wednesday, December 23, 1998 10:50 AM
Subject: Newbie Intro & Question
Hi all,
As per the joining recommendation, here's a brief bit about me, and my
first question.
I've just started running CSound under Win95 - I prefer Linux for most
things, but all my other music software (VAZ softsythn, MIDI sequencer etc)
is on windows so it seemed silly to have CSound as the only Linux app -
especially when I haven't really configured my sound card under Linux, and
PnP is a pain with my current kernel :-)
I have no formal musical training, but I have some previous experience with
analogue sythns (I owned an EDP Wasp when they first came out). So, I'm
just winging it here.
My first composition project with CSound is going to be based on additive
synthesis, because I have some understanding of that. The idea is that I'm
going to set up a set of sine wave ocillators at different frequencies
using the oscil opcode and vary the amplitude of each over time to give me
a subtly shifting series of tones. I'm sure this has been done before, but
there you go.
OK - so here's my question. The obvious way to do this is with a separate
instrument for each voice, with each instrument having its own linseg
controlling the amplitude of the wave over time. This is the way I'm
working at the moment.
But I'm open to suggestions as to a more generic way of accomplishing this.
Also, to help me with these explorations, I'm planning to build a Java
front end targetted at this kind of additive synthesis. My plan is to have
a window for each channel, with a selectable number of harmonics in each
window. Each harmonic would have a selectable waveform (limited to sine in
the first instance). The time/amplitude function for each voice could then
be drawn into the window, with each harmonic being displayed in a distinct
colour so that you can get a good overview of the whole channel at once.
Once the compoisition is complete the program would then generate the
appropriate orc and sco files.
If there is already something like this, I wouldn't mind being spared the
effort :-) Otherwise, I'll let folks know how it goes.
Many Thanks
Andreas
________________________________________________________________________
Dr Andreas Schöter, Intertrader Ltd
http://www.intertrader.com
Tel: +44(0)131 475 7108, Fax: +44(0)131 475 7109
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From: Richard Dobson
Organization: Composers Desktop Project
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CC: Rick Mealey , csound@maths.ex.ac.uk
Subject: Re: for the future: more than four channels?
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There should be no problem simply writing 5.1 files in Csound (opcodes
permitting); all you are doing is defining (or assuming) speaker
positions, which is exactly what the new Microsoft file format does.
Extended Csound running on the "Sphinx" can output six channels. I think
decoding 5.1 does not incur any license, though the algorithm probably
need to be vetted by Dolby before it can be publicly described as such.
I agree nevertheless that Ambisonics is a more fruitful path. Richard
Elen's G-Format, which is Ambisonics pre-decoded into 5.1, could also be
an interesting path.
The Microsoft format suggests, in time, an alternative form of opcode
for Csound - instead of plain 'outq', 'outh', etc, one could define a
symbolic mapping to speaker positions, e.g 'outh.LFE', and so on,
coupled with an optional init statement which associates speaker
assignmenmts with actual hardware output lines. This may be very useful
now that so many soundcards are standardising on 8-channel outputs.
Then, reconfiguring the hardware would not require any changes to the
Csound code.
I have recently found another soundcard which supports multi-channel
files through a single WAVE device - the Frontier Design 'WaveCenter' -
quite cheap, too. I have more details, and links, on my website.
A Happy Christmas too everyone!
Richard Dobson
"Matt J. Ingalls" wrote:
>
> should be able to do 8 channels.
>
> 5.1 is owned by dolby
> plus ambisonics (Bformat) is better (WXYZ)
...
> > Support for more than four channels. Say, a 5.1 algorithm for film work and/or
> > Foley processing.
> >
>
--
Test your DAW with my Soundcard Attrition Page!
http://wkweb5.cableinet.co.uk/rwd
CDP homepage: http://www.bath.ac.uk/~masjpf/CDP/CDP.htm
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From: Hans Mikelson
To: csound@maths.ex.ac.uk
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Subject: Re: Newbie Intro & Question
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>OK - so here's my question. The obvious way to do this is with a separate
>instrument for each voice, with each instrument having its own linseg
>controlling the amplitude of the wave over time. This is the way I'm
>working at the moment.
Maybe I don't understand something anyway here are some simple instruments &
scores using additive:
; ORCHESTRA
srD100
krD10
ksmps
nchnls=2
;---------------------------------------------------------------------------
-------
; Basic Additive Synthesis
;---------------------------------------------------------------------------
-------
instr 1
idur = p3
iamp = p4
ifqc = cpspch(p5)*2
itable = p6
kdvib linseg 0, .1, 0, .1, .01, idur-.2, .01
kvibr oscil kdvib, 6, 1
kamp linseg 0, .01, iamp, idur-.02, iamp*.8, .01, 0
aamp interp kamp
; Amp Fqc
aoutl oscil aamp, ifqc*(1.001+kvibr), itable
aoutr oscil aamp, ifqc*(0.999+kvibr), itable
outs aoutl, aoutr
endin
;---------------------------------------------------------------------------
-------
; Linear Synthesis
;---------------------------------------------------------------------------
-------
instr 2
idur = p3
iamp = p4
ifqc = cpspch(p5)*2
itable = p6
kdvib linseg 0, .2, 0, .2, .007, idur-.4, .01
kvibr oscil kdvib, 6, 1
kamp linseg 0, .05, iamp, .1, iamp*.7, idur-.25, iamp*.5, .1, 0
aamp interp kamp
attk loscil iamp/4, 440, 10, 440
;asamp1 loscil iamp, ifqc, iwav, ibase
; Amp Fqc Table
aout oscil aamp, ifqc*(1+kvibr), itable
outs attk+aout, attk+aout
endin
;---------------------------------------------------------------------------
-------
; Additive Synthesis 2
;---------------------------------------------------------------------------
-------
instr 3
idur = p3
iamp = p4
ifqc = cpspch(p5)
itabl0 = p6
itab1 = p7
itab2 = p8
itab3 = p9
itab4 = p10
ipch1 = p11
ipch2 = p12
ipch3 = p13
ipch4 = p14
aamp1 oscil 1, 1/idur, itab1
apch1 oscil 1, 1/idur, ipch1
aout1 oscil aamp1, ifqc*apch1, itabl0
aamp2 oscil 1, 1/idur, itab2
apch2 oscil 1, 1/idur, ipch2
aout2 oscil aamp2, ifqc*apch2, itabl0
aamp3 oscil 1, 1/idur, itab3
apch3 oscil 1, 1/idur, ipch3
aout3 oscil aamp3, ifqc*apch3, itabl0
aamp4 oscil 1, 1/idur, itab4
apch4 oscil 1, 1/idur, ipch4
aout4 oscil aamp4, ifqc*apch4, itabl0
aoutl = aout1+aout2+aout3
aoutr = aout1-aout3+aout4
outs aoutl*iamp, aoutr*iamp
endin
;---------------------------------------------------------------------------
-------
; Basic Additive Synthesis
;---------------------------------------------------------------------------
-------
instr 4
idur = p3
iamp = p4
ifqc = cpspch(p5)*2
itable = p6
ipbend = p7
kpbend oscili 1, 1/idur, ipbend
kdvib linseg 0, .1, 0, .1, .01, idur-.2, .01
kvibr oscil kdvib, 6, 1
kamp linseg 0, .01, iamp, idur-.02, iamp*.8, .01, 0
aamp interp kamp
; Amp Fqc
aoutl oscil aamp, ifqc*(1.001+kvibr)*kpbend, itable
aoutr oscil aamp, ifqc*(0.999+kvibr)*kpbend, itable
outs aoutl, aoutr
endin
;---------------------------------------------------------------------------
-------
; Tremelo
;---------------------------------------------------------------------------
-------
instr 20
idur = p3
iamp = p4
ifqc = cpspch(p5)
iwave = p6
ilfofqc = p7
ilfowav = p8
; Amp Fqc Table
kampo oscil iamp, ilfofqc, ilfowav
kamp = (kampo+iamp*.5)
asig oscil kamp, ifqc, iwave
outs asig, asig
endin
;---------------------------------------------------------------------------
-------
; Panning
;---------------------------------------------------------------------------
-------
instr 21
idur = p3
iamp = p4
ifqc = cpspch(p5)
iwave = p6
ilfofqc = p7
ilfowav = p8
; Amp Fqc Table
kampo oscil 1, ilfofqc, ilfowav
kamp = (kampo+.5)
asig oscil iamp, ifqc, iwave
outs asig*sqrt(kamp), asig*sqrt(1-kamp)
endin
; SCORE 1
; Sine Wave
f1 0 16384 10 1
; Saw Wave 1/n series
f2 0 16384 10 1 .5 .33 .25 .2 .167
; Square Wave 1/n odd series
f3 0 16384 10 1 0 .5 0 .333 0 .25 0 .2 0 .167
; Irregular Wave
f4 0 1024 10 1 .3 .1 0 .2 .02 0 .1 .04
; Wave 1/n series 1 point missing
f5 0 16384 10 1 .5 .33 .25 .2 0 .142 .125 .111
; Series with higher harmonics
f6 0 16384 10 1 .5 .33 .25 .2 0 .142 .125 .111 0 .100 0 .09 0 .08 .01
.06 .015
t0 20
; Away In A Manger
; Treble
; Sta Dur Amp Freq Table
i1 0.0 .2 20000 7.07 2
i1 + .3 20000 7.07 2
i1 . .1 20000 7.05 2
i1 . .2 20000 7.04 2
i1 . .2 20000 7.04 2
i1 . .2 20000 7.02 2
i1 . .2 20000 7.00 2
i1 . .2 20000 7.00 2
i1 . .2 20000 6.11 2
i1 . .2 20000 6.09 2
i1 . .4 20000 6.07 2
;
i1 . .2 20000 6.07 2
i1 . .3 20000 6.07 2
i1 . .1 20000 6.09 2
i1 . .2 20000 6.07 2
i1 . .2 20000 6.07 2
i1 . .2 20000 7.02 2
i1 . .2 20000 6.11 2
i1 . .2 20000 6.09 2
i1 . .2 20000 6.07 2
i1 . .2 20000 7.00 2
i1 . .4 20000 7.04 2
;
i1 . .2 20000 7.07 2
i1 . .3 20000 7.07 2
i1 . .1 20000 7.05 2
i1 . .2 20000 7.04 2
i1 . .2 20000 7.04 2
i1 . .2 20000 7.02 2
i1 . .2 20000 7.00 2
i1 . .2 20000 7.00 2
i1 . .2 20000 6.11 2
i1 . .2 20000 6.09 2
i1 . .4 20000 6.07 2
;
i1 . .2 20000 6.07 2
i1 . .3 20000 7.05 2
i1 . .1 20000 7.04 2
i1 . .2 20000 7.02 2
i1 . .2 20000 7.04 2
i1 . .2 20000 7.02 2
i1 . .2 20000 7.00 2
i1 . .2 20000 7.02 2
i1 . .2 20000 6.09 2
i1 . .2 20000 6.11 2
i1 . .4 20000 7.00 2
; SCORE 2
; Sine Wave
f1 0 16384 10 1
; Saw Wave 1/n series
f2 0 16384 10 1 .5 .33 .25 .2 .167
; Square Wave 1/n odd series
f3 0 16384 10 1 0 .5 0 .333 0 .25 0 .2 0 .167
; Irregular Wave
f4 0 1024 10 1 .3 .1 0 .2 .02 0 .1 .04
; Wave 1/n series 1 point missing
f5 0 16384 10 1 .5 .33 .25 .2 0 .142 .125 .111
; Series with higher harmonics
f6 0 16384 10 1 .5 .33 .25 .2 0 .142 .125 .111 0 .100 0 .09 0 .08 .01
.06 .015
; Pitch Bend Table
f10 0 1024 -7 1 250 1 12 2 240 2 12 1.5 120 1.5 10 .5 55 .5 10 2 55 2 20 .5
240 .5
f11 0 1024 -7 1 256 .5 10 1 118 .5 10 1 246 .5 10 1 118 .5 10 1 246 .5
; Treble
; Sta Dur Amp Freq Table PBend
i4 0.0 5 20000 7.02 6 10
i4 + 5 20000 6.07 6 10
i4 . 5 20000 6.04 6 11
; SCORE 3
; Sine Wave
f1 0 16384 10 1
; Amplitude envelopes
f2 0 1025 -7 0 212 1 600 1 213 0
f3 0 1025 -7 0 112 .5 200 1 400 .8 100 .5 113 0
f4 0 1025 -7 0 112 1 200 2 500 .8 213 0
f5 0 1025 -7 0 212 2 100 .5 400 .2 200 1 113 0
; Pitch envelopes
f6 0 1025 -7 1 212 1.01 600 .99 213 1
f7 0 1025 -7 .2 112 .5 200 1 400 .8 100 .5 113 .6
f8 0 1025 -7 2.1 212 2.2 100 2 200 1.8 513 2
f9 0 1025 -7 6 212 6 100 6 400 7 200 7 113 6
; Sta Dur Amp Freq Sine Amp1 Amp2 Amp3 Amp4 Pch1 Pch2 Pch3 Pch4
i3 0.0 11 1500 7.02 1 2 3 4 5 6 7 8 9
i3 2.0 10 . 8.04 . . . . . . . . .
i3 6.0 9 . 8.00 . . . . . . . . .
i3 7.0 8 . 6.04 . . . . . . . . .
i3 8.0 6 . 9.00 . . . . . . . . .
;
i3 12.0 6 2000 7.00 1 2 3 4 5 6 7 8 9
i3 14.0 7 . 8.02 . . . . . . . . .
i3 16.5 9 . 7.10 . . . . . . . . .
i3 17.5 8 . 6.02 . . . . . . . . .
i3 18.5 8 . 9.04 . . . . . . . . .
;
i3 20.0 3 1000 7.00 1 2 3 4 5 6 7 8 9
i3 21.0 4 . 8.02 . . . . . . . . .
i3 22.5 4 . 7.10 . . . . . . . . .
i3 23.5 2 . 6.02 . . . . . . . . .
i3 24.5 2 . 9.04 . . . . . . . . .
i3 25.5 2 . 7.02 . . . . . . . . .
i3 26.5 2 . 9.00 . . . . . . . . .
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Message-ID: <3681F8D3.88CE14AA@agora.stm.it>
Date: Thu, 24 Dec 1998 09:18:27 +0100
From: Gabriel Maldonado
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To: Rick Mealey
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Subject: Re: for the future: more than four channels?
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Santa Claus is arriving!
I've just finished to write some opcodes for multi-channel files.
The new opcode 'fout' allow the output to a file of any number of audio channels (for now
the limit is arbitrary locked to 64, bat can be increased). It supports .wav format for
mono and stereo files in integer, raw format (without header) for 16 bit integer and 32
bit floating-point with any number of channels. In future, when the standard for
multichannel files will be definitively defined, it will support any format.
I will send the sources to JPFF as soon as I can.
A'vincr' and 'clear' opcodes are new useful tools when using 'fout'.
A modification of rdorch.c and entry.c files is needed to allow variant argument list at
a-rate.
fout opcode:
fout "ifilename", iformat, aout1 [, aout2, aout3,....]
DESCRIPTION
Output a list of audio signal to a specified file
INITIALIZATION
ifilename - a double-quote delimited string file name
iformat - a flag to choose output file format:
0 - 16-bit integers without header
1 - 32-bit floating point samples without header
2 - 16-bit integers with .wav type header
PERFORMANCE
aout1,... aoutN - output signals
This opcode allows to write samples of audio signals to a file with any number of
channels. Channel number depends by the
number of aoutN variables (i.e. a mono signal with only an a-rate argument, a stereo
signal with two a-rate arguments etc.)
Maximum number of channels is fixed to 64.
More fout opcodes can be present in the same instrument, allowing to open more audio to
disk streams in parallel.
Notice that, differently by out, outs and outq, fout does not zeroes the audio variable,
so you must provide a zeroing after calling
fout is poliphony is used. You can use incr and clear opcodes for this task.
vincr, clear
vincr asig, aincr
clear avar1 [,avar2, avar3,...,avarN]
DESCRIPTION
vincr increments an audio variable of another signal, i.e. accumulates output.
clear zeroes a list of audio signals.
PERFORMANCE
asig - audio variable to be incremented
aincr - incrementation signal
avar1 [,avar2, avar3,...,avarN] - signals to be zeroed
vincr (variable increment) and clear are thinked to be used togheter. vincr stores the
result of the sum of two audio variables
into the first variable itself (which is thinked to be used as accumulator in case of
polyphony). The accumulator-variable can be
used for output signal by means of fout opcode. After the disk writing operation, the
accumulator-variable should be set to zero
by means of clear opcode (or it will explode).
EXAMPLE OF 'FOUT', 'INCR' AND 'CLEAR':
-------------------------------------
;ORC
sr = 44100
kr = 441
ksmps = 100
nchnls = 1
ga1 init 0
ga2 init 0
ga3 init 0
ga4 init 0
;***************************************************
instr 1
;***************************************************
k1 linseg 0,p3/3,1,p3/6,.5,p3/3,.5,p3/6,0
a1 oscili 10000,p4,1
a2 oscili 10000, p4*1.234567,1
a3 oscili 10000, p4*1.5,1
a4 oscili 10000, p4*.766,1
vincr ga1,a1*k1
vincr ga2,a2*k1
vincr ga3,a3*k1
vincr ga4,a4*k1
out a1
endin
;***************************************************
instr 2
;***************************************************
fout "fout.wav" ,2,ga1,ga2
fout "fout2.wav" ,2,ga3,ga4
clear ga1,ga2,ga3,ga4
endin
;-----------------------
;SCO
f1 0 4096 10 1
i1 0 1.8 106
i1 1.5 2 123.66
i1 3.1 2 51.123335
i2 0 10
--
Gabriel Maldonado
http://www.agora.stm.it/G.Maldonado/home2.htm
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maybe i dont understand, but why use the 'ifilename' variable? why not use the output
filename specified in the command line? why give this fout opcode the power to change the
output filename?
just curious
pete
Gabriel Maldonado wrote:
> Santa Claus is arriving!
>
> I've just finished to write some opcodes for multi-channel files.
> The new opcode 'fout' allow the output to a file of any number of audio channels (for now
> the limit is arbitrary locked to 64, bat can be increased). It supports .wav format for
> mono and stereo files in integer, raw format (without header) for 16 bit integer and 32
> bit floating-point with any number of channels. In future, when the standard for
> multichannel files will be definitively defined, it will support any format.
> I will send the sources to JPFF as soon as I can.
> A'vincr' and 'clear' opcodes are new useful tools when using 'fout'.
> A modification of rdorch.c and entry.c files is needed to allow variant argument list at
> a-rate.
>
> fout opcode:
>
> fout "ifilename", iformat, aout1 [, aout2, aout3,....]
>
> DESCRIPTION
>
> Output a list of audio signal to a specified file
>
> INITIALIZATION
>
> ifilename - a double-quote delimited string file name
> iformat - a flag to choose output file format:
> 0 - 16-bit integers without header
> 1 - 32-bit floating point samples without header
> 2 - 16-bit integers with .wav type header
>
> PERFORMANCE
>
> aout1,... aoutN - output signals
>
> This opcode allows to write samples of audio signals to a file with any number of
> channels. Channel number depends by the
> number of aoutN variables (i.e. a mono signal with only an a-rate argument, a stereo
> signal with two a-rate arguments etc.)
> Maximum number of channels is fixed to 64.
>
> More fout opcodes can be present in the same instrument, allowing to open more audio to
> disk streams in parallel.
>
> Notice that, differently by out, outs and outq, fout does not zeroes the audio variable,
> so you must provide a zeroing after calling
> fout is poliphony is used. You can use incr and clear opcodes for this task.
>
> vincr, clear
>
> vincr asig, aincr
> clear avar1 [,avar2, avar3,...,avarN]
>
> DESCRIPTION
>
> vincr increments an audio variable of another signal, i.e. accumulates output.
> clear zeroes a list of audio signals.
>
> PERFORMANCE
>
> asig - audio variable to be incremented
> aincr - incrementation signal
> avar1 [,avar2, avar3,...,avarN] - signals to be zeroed
>
> vincr (variable increment) and clear are thinked to be used togheter. vincr stores the
> result of the sum of two audio variables
> into the first variable itself (which is thinked to be used as accumulator in case of
> polyphony). The accumulator-variable can be
> used for output signal by means of fout opcode. After the disk writing operation, the
> accumulator-variable should be set to zero
> by means of clear opcode (or it will explode).
>
> EXAMPLE OF 'FOUT', 'INCR' AND 'CLEAR':
> -------------------------------------
> ;ORC
> sr = 44100
> kr = 441
> ksmps = 100
> nchnls = 1
>
> ga1 init 0
> ga2 init 0
> ga3 init 0
> ga4 init 0
>
> ;***************************************************
> instr 1
> ;***************************************************
> k1 linseg 0,p3/3,1,p3/6,.5,p3/3,.5,p3/6,0
> a1 oscili 10000,p4,1
> a2 oscili 10000, p4*1.234567,1
> a3 oscili 10000, p4*1.5,1
> a4 oscili 10000, p4*.766,1
>
> vincr ga1,a1*k1
> vincr ga2,a2*k1
> vincr ga3,a3*k1
> vincr ga4,a4*k1
>
> out a1
> endin
>
> ;***************************************************
> instr 2
> ;***************************************************
> fout "fout.wav" ,2,ga1,ga2
> fout "fout2.wav" ,2,ga3,ga4
> clear ga1,ga2,ga3,ga4
> endin
>
> ;-----------------------
> ;SCO
> f1 0 4096 10 1
> i1 0 1.8 106
> i1 1.5 2 123.66
> i1 3.1 2 51.123335
> i2 0 10
>
> --
> Gabriel Maldonado
>
> http://www.agora.stm.it/G.Maldonado/home2.htm
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Date: Thu, 24 Dec 1998 03:55:11 -0500
From: Paul Winkler
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To: Csound mailing list
Subject: Re: jpff suggestions for the holiday
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> Josep M Comajuncosas wrote:
>
> > Well, I´d really like to see Csound recognizing midi pitch bend messages at
> > last...
> > long time wayting for it!
How about patch change messages? After all, there's only so many
channels, but many more patches...
--PW
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From: Richard Dobson
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To: Gabriel Maldonado , csound@maths.ex.ac.uk
Subject: Re: for the future: more than four channels?
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Don't forget that there are at least two cards (Sonorus STUDI/O and
Creamware TDAT-16 - is anyone on this list using them?) which can
provide a single 16-ch WAVE device, and there are a few more which
offer 8-ch devices, so there is no imperative need to limit WAVE files
to mono or stereo. This also goes for AIFF files. Creamware have
promised that they will provide Linux drivers for the new PULSAR, when
more urgent tasks such as a Mac driver are completed. So there is little
to be lost in providing for at least 8-ch file output now, for both WAV
and AIFF.
I have a pointer to the Microsoft document on my website. It is very
detailed, and enough in fact to begin coding. I doubt if there will be
material changes between that and the final publication. The main change
they made (something I asked for :-) ) was to add the floating-point
format.
If you are going to use a format argument, I think there should then be
a mechanism to #define a format in terms of speaker positions (as
defined in the MS document) an init time (could even be an extension to
the nchnls element of the orc header).
The main thing is to devise a system which can be extended easily, for
when new specific formats appear, and which is flexible in terms of
channel routing. One reason for defining speaker positions is that it
would be plausible on a m/c card to allocate two channels as a stereo
monitor output (or independent pair), and the remaining 6, say, for 5.1
or G-format output. The stero pair could be chans 1 and 2, or 7 and 8.
So people might want Csound output the start at Channel 3, for example.
My main point is that multi-channel file and card support implies a
whole lot more than just have 8-ch opcodes. Speaker assignments,
positional audio and so on demand a whole raft of new software
facilties, most of which I can hardly guess at present!
Millenium? Bah, humbug; this is far more exciting!
Richard Dobson
Gabriel Maldonado wrote:
>
> Santa Claus is arriving!
>
> I've just finished to write some opcodes for multi-channel files.
> The new opcode 'fout' allow the output to a file of any number of audio channels (for now
> the limit is arbitrary locked to 64, bat can be increased). It supports .wav format for
> mono and stereo files in integer, raw format (without header) for 16 bit integer and 32
> bit floating-point with any number of channels. In future, when the standard for
> multichannel files will be definitively defined, it will support any format.
--
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: Thu, 24 Dec 1998 12:11:42 +0100
From: Josep M Comajuncosas
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Hi,
This is just to wish a Happy New Csounding Year (1999 yeah...) to all
the people in the list, and merry Xmas of course!
I´ve been preparing a Carol for Csound but it took soo long that I won´t
probably finish it till the 2000... keep wayting for it!
Josep M Comajuncosas
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Date: Thu, 24 Dec 1998 12:53:34 +0000
From: Leonello Tarabella
Subject: realt time csound
To: csound@maths.ex.ac.uk
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Hi,
I have a problem about midi-real-time on Mac: I suppose this has been
already discussed
but unfortunately, after many unsuccesseful tries to reach the FAQ
website
I'm forced to ask someone to help me:
I use a Mac computer 9600/200Mhz;
I tried to control a very simple archestra via a midi keyboard;
everything works but with an unacceptable delay.
Yes, I tried with different buffers lengths and
taking care of many other things (screen-save and other things off)
Naturally I'm using OMS.
I suppose the problem is known: is the solution as well known?
Thanks a lot.
Leonello Tarabella
Computer Music Lab of CNUCE/C.N.R.
via Santa Maria 36 - 56126 Pisa, ITALY
tel. +39/50/593276 - fax +39/50/904052
http://spcons.cnuce.cnr.it/music/cmd.html
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