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re: books - volunteering to create/maintain list

Date1998-06-30 02:10
FromChris Brown
Subjectre: books - volunteering to create/maintain list
Hi All,

If John FFitch will agree to put the text file up on the Bath server,
I will compile and maintain a list of Books regarding Electro Acoustic
music, Computer composition etc.

What do you say John.

Chris
_________________________________________________________
DO YOU YAHOO!?
Get your free @yahoo.com address at http://mail.yahoo.com

Date1998-06-30 14:00
Fromjpff@maths.bath.ac.uk
SubjectRe: books - volunteering to create/maintain list
Sound OK by me.  I would suggest placing in a Documentation directory?
==John ff



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Date: Tue, 30 Jun 1998 18:39:37 +0000
From: Jean-Michel DARREMONT 
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To: csound 
Subject: Re: PMAX
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jp wrote:
> 
> Jean-Michel DARREMONT wrote:
> >
> > Dear CSounders,
> >
> > Is it possible to set the score PMAX to more than 5000 and how to do it?
> 
> change it in cs.h
> recompile
> 

I guess I need a C compiler to do that. Where can I find a cheap one.
Where to get the source code?
 
-- 
Jean-Michel DARREMONT



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Date: Tue, 30 Jun 1998 16:06:23 -0500
From: Terry Cast 
Organization: The University of Oklahoma
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To: "csound@noether.ex.ac.uk" 
Subject: Winsound crashes with .mid files (long message)
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This is a multi-part message in MIME format.
--------------F641A72FD872DB8B92D6A8D8
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Hello Csounders,

When I try to compile an orc/sco/mid combination using Winsound 3.482, Windows 95 reports an
illegal function and shuts Winsound down. I have tried two sets of files (well, actually more, but the
ones below are the ones I kept the error messages from). The first is my own (don't laugh too hard):

;ORCHESTRA
sr=44100
kr=44100
ksmps=1
nchnls=2
          instr     1
ipch      pchmidi
icps      cps2pch   ipch,1
iamp      ampmidi   30000
kpan      midictrl  10
kpan2     =         kpan * 0.708
kamp      linsegr   0,.08,.9,.03,1,.1,0
asing     voice     iamp*kamp, icps, 1, 1.2, 5, .01*icps, 20, 3
kdistance           = (iamp*kamp != 0 ? iamp*kamp : .001)
aleft,aright        locsig asing,kpan2,1/(kdistance*.001),20
asndl,asndr         locsend
arvbl     reverb2   asndl,2,0.8
arvbr     reverb2   asndr,2,0.8
          outs      (.7*aleft)+(.3*arvbl),(.7*aright)+(.3*arvbr)
          endin

;SCORE
f1 0 16 -2 1  1.066667 1.125 1.2 1.25 1.33333333  1.40625 1.5 1.6 1.66666667 1.77777778
1.875
  ; Just major scale in C
f3 0 1024 10 1 ; sine wave for vibrato
f20 0 256 1 "c:\csound\samples\impuls20.wav" 0 4 1
f0 15 ; wait 15 seconds for MIDI input
e

The MIDI file is attached.

Thinking the problem might be my orc or score I tried waveshap.orc, waveshap.sco, waveshap.mid, which I found at
ftp://ftp.maths.bath.ac.uk/pub/dream/documentation/orchestras+scores/midi/ , and got a similar error message (see below). I tried ffitch's "Prelude" (in
ftp://ftp.maths.bath.ac.uk/pub/dream/documentation/orchestras+scores/compositions/ffitch/drumsand/ ), which doesn't use MIDI, and it worked fine. Other orc/sco (no MIDI) pieces
generally compile ok.

The error message for my orc/sco/mid reads:

WINSOUND caused a stack fault in module WINSOUND.EXE at 0137:00415ea4.
Registers:
EAX=007d792c CS=0137 EIP=00415ea4 EFLGS=00010246
EBX=007e0504 SS=013f ESP=006ffcf8 EBP=00000000
ECX=007e03d4 DS=013f ESI=00000003 FS=0f27
EDX=00000048 ES=013f EDI=00000001 GS=0000
Bytes at CS:EIP:
83 7d 48 00 0f 84 b9 00 00 00 8b 3b 85 ff 74 0c
Stack dump:
006ffd8c 00000001 00000000 00000003 00000000 0044c234 00000003 007e03d4 007dffc0 006ffd8c 0049769a 004a5353 0048adb8 00000000 41700000 41700000

The error message for waveshap.orc/sco/mid reads:

WINSOUND caused a stack fault in module WINSOUND.EXE at 0137:00415ea4.
Registers:
EAX=007d7770 CS=0137 EIP=00415ea4 EFLGS=00010246
EBX=0085c2a0 SS=013f ESP=006ffcf8 EBP=00000000
ECX=0085c1d0 DS=013f ESI=00000004 FS=0f0f
EDX=00000030 ES=013f EDI=00000001 GS=0000
Bytes at CS:EIP:
83 7d 48 00 0f 84 b9 00 00 00 8b 3b 85 ff 74 0c
Stack dump:
004a8b04 00000001 00000000 00000004 00000000 0044c234 00000004 0085c1d0 007e0538
006ffd8c 0049769a 004a5356 0048adb8 00000000 44160000 44160000

Thank you for your time.

Terry Cast
University of Oklahoma

--------------F641A72FD872DB8B92D6A8D8
Content-Type: audio/x-midi; name="vocal.mid"
Content-Transfer-Encoding: base64
Content-Disposition: inline; filename="vocal.mid"
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--------------F641A72FD872DB8B92D6A8D8--




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From: Sergey 
To: csound@maths.ex.ac.uk
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Subject: Old version of Csound
Date: Wed, 1 Jul 1998 01:27:02 +0400
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Hi,
I try to work with DX7 emulator by Jeff Harrington. An attempt to
run the newest version of Csound with its .orc brings to an infinite
loop with message:
"error:  Characters after \ not comment".
It's happen because of new rules of comments.
The version 3.473  worked without this problem.
Can anybody tell me where can I get this old  version?

Thanks,
   Sergey Batov
batov@glasnet.ru



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From: Sergey 
To: csound@maths.ex.ac.uk
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Hi, Csounders!
I'd like to know your expert opinion or symply any 
comments about the SAOL,  Csound-on based system (if I'm right).
You can find it on http://sound.media.mit.edu/~eds/mpeg4/ , and
http://sound.media.mit.edu/~eds/mpeg4-old


Regards,
   Sergey Batov

batov@glasnet.ru




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From: Michael Gogins 
To: Richard Dobson 
Cc: csound 
Subject: Re: compiler
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>I did embark on just this some while back, adding in lots of init
functions, and
>indeed got Csound running re-entrantly for most of the main opcodes, but I
ran
>out of steam after a while (local statics are a particular problem!). Also,
John
>Fitch told me  a few months ago that
> a few of the things I had done weren't quite right (but I don't yet know
what
>those are!); however, most of what I did is still in the published sources,
>commented out. I have a lot of other work on at the moment, but I do want
to go
>back to this exercise at some point. Once it is done, there is the chance
for
>some ~very~ snappy user interfaces - and it would benefit all platforms.
>
>It would help if people adding new opcodes gave a moment to add init
functions
>(and preferably memory de-allocation functions too), avoided local statics,
and
>so on. Once the modules are re-entrant, there is a sporting chance the
framework
>code can be made so too.
>
>Perhaps we could co-ordinate?


Again, I'd be very willing to do this. As some of you may know, I have a
front end for Csound, WCSound.dll, that already is a COM server.

Here are some features I'd like to see in a COM server version of Csound:

It would be an in-process, apartment-threaded COM server with a dual
interface. This would make it easy to use both from C++ and from Visual
Basic and it could also be used with Microsoft's J++ version of Java, which
can automatically generate a Java proxy for most COM objects.

It would have no self-contained GUI.

It would have plugin opcodes and plugin function tables. These would use
their own COM interfaces. This would allow Csound to be extended without
having to relink it.

Csound would work with standard COM interfaces to MIDI and audio drivers,
i.e. DirectSound or ActiveMovie (I need to do some research about what is
possible).

As far as I know, Csound contains the deepest and broadest library of
synthesis and signal processing code in any cross-platform software
synthesizer. Doing these things would extend the useful life of Csound for
another five or ten years, probably. It would enable Csound to fit into
other music systems the same way the current generation of MIDI software
synthesizers and signal processing plugins does, only of course it would so
much more useful!






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From: David Boothe 
To: "'csound@maths.ex.ac.uk'" 
Subject: Need More RAM?
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Hello Csound community-
 
When performing an instrument that uses several adsyn statements, with
RT Sound or DirectSound, I get a glitch, or repeated section shortly
before the end of the performance. All the adsyn statements use the same
hetro analysis file. The problem seems worse with DirectSound than RT
Sound. Would more RAM help this situation? I am using a Pentium Pro with
32 meg RAM running Win95.

Thanks in advance.

-dB
 (David M. Boothe, Audio Director 
  Lyrick Studios, Dallas, TX)




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Date: Wed, 01 Jul 1998 02:28:15 -0400
From: Brandon Nelson 
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To: "Ruston, Paul" , csound 
Subject: Re: re_books
References: <98Jun26.100040edt.20610@fw.osler.com>
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I bought Roads' _The Computer Music Tutorial_ in the fall, and was EXTREMELY
disappointed. It does touch on everything, but for sound synthesis, gives
overly wordy yet incomplete explanations and almost no examples or models that
are easily implementable. Say bye bye, $100, and Godspeed. 

Brandon Nelson


Ruston, Paul wrote:
> 
> Can anyone recommend one of the previously suggested books over the
> others:
> 
> I found a copy of The Computer Music Tutorial by Roads for about $54.00
> u.s. ($76 canadian - scary exchange rate, eh?) or I can order Computer
> Music: Synthesis, Composition and Performance by Dodge, Charles, etc.
> for $42 u.s. and the same book store can order Elements of Computer
> Music by Moore for $65 u.s..  This last book, which seems to be in high
> demand is listed on the stores computer as being in print but apparently
> this does not guarentee that it actually is - trying to order it is the
> only way to verify for sure.
> 
> I'm hesitant to pick up The Computer Music Tutorial as it consciously
> minimizes the mathematical content for mathematically challenged
> musicians like myself, but I'm working to improve my math skills and
> don't want to be frustrated later by incomplete models and
> explainations, etc.
> 
> Any opinions? If I do confirm that Elements... is indeed in print, I'll
> be certain to notify you all.  The book store I'm dealing with is
> 'Chapters' in Toronto, Canada for anyone concerned.
> 
> Thanks,  Paul Ruston
> 
> >-----Original Message-----
> >From:  pete moss [SMTP:kaeru@flash.net]
> >Sent:  June 23, 1998 3:58 PM
> >To:    herremar@pilot.msu.edu
> >Cc:    csound@noether.ex.ac.uk
> >Subject:       Re: re_books
> >
> >i would bet that ucsd has this book available, as moore is dean of music
> >there
> >or something.
> >
> >pete
> >
> >
> >
> >herremar@pilot.msu.edu wrote:
> >
> >> >Moore's book (Elements of Computer Music, see previous mail) seems to
> >> >be out of print. I'm trying myself to find a copy for our library.
> >> >Perhaps we should start nagging Prentice Hall for a reprint?
> >>
> >> it was the bookstore at either uc san diego or at stanford that told me
> >> they had this in stock.  my local retailers have given me the impression
> >> that they could order it.  i'm hesitating only because soft cover is
> >>$64.00.
> >>
> >> rh
> >>
> >> Ron Herrema
> >>  http://pilot.msu.edu/user/herremar/
> >>
> >> Here's to the was you been,
> >> to the is you in,
> >> to what's deep in deep,
> >> to what's down in down,
> >> to the lost and the blind
> >> and the almost found.
> >>
> >>      - Sekou Sundiata, The Blue Oneness of Dreams
> >
> >
> >
> >



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>I bought Roads' _The Computer Music Tutorial_ in the fall, and was EXTREMELY
>disappointed.
>overly wordy - Say bye bye, $100, and Godspeed.


ekuall+e lak!ng = kompoz!ng [!nzert ke!word] mu.s!k
w!th max. m!t preSS.

altho robert rowe regardz !t az fenomenl.
at what _ !t = rema!nz 2 b [un|d!z]kovrd.






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From: Paul Winkler 
To: guto@sercomtel.com.br
Cc: csound@maths.ex.ac.uk
Subject: Xanadu benchmarks, was Re: Pentium/Linux csound speed boost
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Thanks Guto. If anyone else will send Xanadu test run times to me or 
post them to the list, I volunteer to maintain a results file...

I propose that xanadus.sco be the score (as opposed to xanadul.sco)
and that the orc be xanadu44.orc and optionally xanadu08.orc.

regards,
PW

>Hello Paul,
>
>> .../dream/Csound_List_Archive) ... I haven't seen any other tests of
>> Xanadu for comparison.
>
>I compile xanadu44.orc and xanadu08.orc with xanadus.sco, 
>Pentium 166MHz, 32Mb RAM, 1.6Mb WD Caviar HD, Windows95
>and Maldonado's Csound 3.47:
>
>xanadu44 -  197.180 seconds
>xanadu08 -    35.210 seconds
>
>(the timings that are displayed by csound when finished compiling)
>
>Best wishes,
>
>Guto
>Antonio Augusto Caminhoto Neto (Guto Caminhoto)
>Rua Prof. Joao Candido,434, apto. 110
>86010-000 Londrina PR BRASIL
>guto@sercomtel.com.br
>http://www.sercomtel.com.br/guto
>http://www.uel.br/Web/home/uel/ceca/dart/web/me.htm
>


______________________________________________________
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From: Pedro Batista 
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here's a little coding example, a very rudimentary atempt at using neural 
processing based dsp

This is a very simple neural net, that learns audio patterns, and can 
display memory and recognition when presented with learned patterns (even if 
the input data is incomplete), or simply respond with similarity to unknown 
audio data.
You train the net by presenting two audio files, A and B, respectivelly at 
the input and output of the net; this will associate a pattern at least 
similar to A, with the corresponding response B. You can have a number of 
these pairs, three being a practical maximum, before this particular net 
begins forgeting old patterns
To code the audio in net data, the rudimentary approach of calculating the 
16-bit value of the sample and using binary neurons was used. This is fine 
for visual patterns, but I want to use a more audio oriented mapping, using 
the basic audio info like freq and amp (maybe with the aid of pvoc?), and 
continuous neurons, instead of binary ones.
Using more advanced neural nets, dsp could be performed. For example, 
presenting samples of unfiltered and filtered waves, respectively to the 
input and output of the net, it could theoretically learn how to respond as 
a filter to other audio data (the acurateness would be determined by the 
learning process and data), distinct from what was taught
Problem here is I'm actually training a 16 neuron network for each of the 
samples, what makes this instr run sooo slooow! (this example takes about 4 
min in my P200, for half-a-second of audio...). I really would apreciate 
hints on speeding it up (for instance would table write be faster than zak? 
 - guess not).
I also hope that you can help me develop a more audio-prone coding 
algorithm, but I'll get back on that later
I'm working on a more advanced instr using back propagation, and will post 
results when I get it done
In the mean time, hope this rather useless example amuses you :)

pedro
_____________________________
; start of orchestra

sr   =    44100
kr   =    44100
ksmps     =    1

     zakinit   1, 295
;--------------------------------------------------;
; A simple single-layer bidirectional perceptron,  ;
; using McCulloch-Pitts algorithm                  ;
;                                                  ;
; Coded for CSound by Pedro Batista jun/98         ;
;--------------------------------------------------;

;* * WARNING * * WARNING * * WARNING * * WARNING * *
;* I cannot guarantee bug-free execution,
;* nor can I assure that this instr wont generate
;* high-values, susceptible of causing harm to the
;* listner or listening equipment (it probably will,
;* if unknown patterns are presented, since there's
;* no compression coded yet) - pls use with caution!
;* * WARNING * * WARNING * * WARNING * * WARNING * *

     instr     1
;--------------------------------------------------
inump     init 3 ; number of patterns (maximum 3)
;--------------------------------------------------
iich1     init 1 ; in channel #1
iich2     init 2 ; in channel #2
iich3     init 3 ; in channel #3
ioch1     init 4 ; out channel #1
ioch2     init 5 ; out channel #2
ioch3     init 6 ; out channel #3
irch init 7 ; stimulus/response channel

ibin init  8; [8-23]
ibout     init 24; [24-39]
ipes init 40; [40-295]

ipass     init 0
isigi     init 0
isigi1    init 0
isigi2    init 0
isigi3    init 0
isigi4    init 0
isigi5    init 0
isigo     init 0
isigo1    init 0
isigo2    init 0
isigo3    init 0
isigo4    init 0
isigo5    init 0
indx init 0
indi init 0
indj init 0
ierr init 0
iact init 0
ipow2     init 0

; skip to performance
     igoto     perf

ini: ;-----------------------------------------------

; initialize weights
indi =    0
loopi:
indj =    0
loopj:
     ziw  0, ipes+indi*16+indj
indj =    indj+1
     if (indj<16) igoto loopj
indi =    indi+1
     if (indi<16) igoto loopi

     if (inump<1) igoto off
isigi1    zir  iich1
isigo1    zir  ioch1
     if (inump<2) igoto nosig
isigi2    zir  iich2
isigo2    zir  ioch2
     if (inump<3) igoto nosig
isigi3    zir  iich3
isigo3    zir  ioch3
nosig:

istim     zir  irch

ipass     =    1
restrt:   ;-----------------------------------------------

; select training signals
     if (ipass==1) igoto case1
     if (ipass==2) igoto case2
     if (ipass==3) igoto case3
     igoto     skip
case1:    
isigi     =    isigi1
isigo     =    isigo1
     igoto     skip
case2:    
isigi     =    isigi2
isigo     =    isigo2
     igoto     skip
case3:    
isigi     =    isigi3
isigo     =    isigo3
skip:

; convert input signal to binary
indx =    0
isigi     =    isigi+32768
loop1:
ibit =    (frac(isigi/2)==0 ? 0:1)
     ziw  ibit, ibin+indx
isigi     =    int(isigi/2)
indx =    indx+1
     if (indx<16) igoto loop1

; convert output signal to binary
indx =    0
isigo     =    isigo+32768
loop2:
ibit =    (frac(isigo/2)==0 ? 0:1)
     ziw  ibit, ibout+indx
isigo     =    int(isigo/2)
indx =    indx+1
     if (indx<16) igoto loop2

; train pattern
train:
indi =    0
loop3:
indj =    0
loop4:
ibi  zir  ibin+indi
ibo  zir  ibout+indj
iw   zir  ipes+indi*16+indj
iw   =    iw+((ibi*2-1)*(ibo*2-1))
     ziw  iw, ipes+indi*16+indj
indj =    indj+1
     if (indj<16) igoto loop4
indi =    indi+1
     if (indi<16) igoto loop3

ipass     =    ipass+1

; proceed with next pattern
     if (ipass<=inump) igoto restrt

; convert stimulus signal to binary
indx =    0
istim     =    istim+32768
loop5:
ibit =    (frac(istim/2)==0 ? 0:1)
     ziw  ibit, ibin+indx
istim     =    int(istim/2)
indx =    indx+1
     if (indx<16) igoto loop5

; calculate response from stimulus pattern
prop:
ierr =    0
indi =    0
loop6:
ibo  zir  ibout+indi
iact =    0
indj =    0
loop7:
ibi  zir  ibin+indj
iw   zir  ipes+indj*16+indi
iact =    iact+iw*ibi
indj =    indj+1
     if (indj<16) igoto loop7
     if (iact==0) igoto skp1
iact =    (iact>0 ? 1:0)
ierr =    ierr+abs(iact-ibo)
     ziw  iact, ibout+indi
skp1:
indi =    indi+1
     if (indi<16) igoto loop6

indi =    0
loop8:
ibi  zir  ibin+indi
iact =    0
indj =    0
loop9:
ibo  zir  ibout+indj
iw   zir  ipes+indj*16+indi
iact =    iact+iw*ibo
indj =    indj+1
     if (indj<16) igoto loop9
     if (iact==0) igoto skp2
iact =    (iact>0 ? 1:0)
ierr =    ierr+abs(iact-ibi)
     ziw  iact, ibin+indi
skp2:
indi =    indi+1
     if (indi<16) igoto loop8

     if (ierr>0) igoto prop ; until stable

; convert binary response to audio value
indx =    0
isigo     =    0
ipow2     =    1

loop10:
ibit zir  ibout+indx
isigo     =    isigo+ibit*ipow2
ipow2     =    ipow2*2
indx =    indx+1
     if (indx<16) igoto loop10

isigo     =    isigo-32768

; compress the resulting signal

;iclip    table     ... accepting ideas here :)

off:
; write the output and leave
     ziw  isigo, irch

     rireturn

;============== P E R F O R M A N C E ==============;
perf:     

; change input and output soundin # to whatever you like
; (the soundin # could be passed as a p-field, but for some
; reason my setup doesnt recognise teh SSDIR, so full path
; must be given

ain1 soundin   "c:\csound\samples\soundin.1"
kin1 downsamp ain1
     zkw  kin1, iich1
ain2 soundin   "c:\csound\samples\soundin.2"
kin2 downsamp ain2
     zkw  kin2, iich2
ain3 soundin   "c:\csound\samples\soundin.3"
kin3 downsamp ain3
     zkw  kin3, iich3

; Just use the same pattern for input and output of the net
; A different set of samples could be present at the input and
; output, causing the learning of that association.

;aout1    soundin   "c:\csound\samples\soundin.#"  ; Program three
kout1     =    kin1                           ; patterns in
     zkw  kout1, ioch1                   ; "recognition"
;aout2    soundin   "c:\csound\samples\soundin.#"  ; mode.
kout2     =    kin2                           ; Remember to
     zkw  kout2, ioch2                   ; update var
;aout3    soundin   "c:\csound\samples\soundin.#"  ;   inump
kout3     =    kin3                           ; at the top
     zkw  kout3, ioch3                   ; of the instr

;astim    soundin   "c:\csound\samples\soundin.#"  ; ask the net
kstim     =    kin1                           ; for pattern
     zkw  kstim, irch                    ; number 1
                                         ; (try kin2&3)

; This will successfully (well, almost, try just 2) recognise
; any of the three patterns and hopefully reconstruct any of
; those, if it is slightly damaged.
; What "slightly" means is very little, using this simple
; network, and due to the limited effectiveness of the
; method used for the audio coding in binary form
; I'm studying an implementation with hidden layers and back-
; -prop learning, using better audio coding strategies
;
; One can present a completelly diferent input than those
; that were trained, and see how the net responds, but
; *NOTICE* there is no secure against signal clipping

; process sample at i-rate
     reinit    ini

     ;-----------------------------------------------

kout zkr  irch
aout interp    kout
     out  aout

     endin

; start of score
; the instr just has to be running for the intended duration

i1 0 .5
e



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Hey everyone :)

I've been away for a week, and seem to have lost 2 digests, so I dont know 
if anyone answered my question on goto and loops; anyway, during my absence, 
I had time to ponder on this subject, and came to the conclusion that the 
only way of running loops is at i-time; also if one needs to run one or 
several loops for each sample, it can easily be done, by using the zak's to 
send values between perf and init, and using reinit to control the program 
flux. I'll send an example using this technique next

Here's the deal:

     instr     1
; first time initializations here
; ....

; skip to performance
     igoto     perf

ini:
; following code will be executed for each sample

; read sample value
isig zir  1

; here you can process the sample at i-rate, with the assurance that
; loops will work, and igoto _does_ jump back and forth

; ...do *whatever* with isig

; write the output
     ziw  isig, 1

; return to performance
     rireturn

perf:     

; simple example with soundin
asig soundin   "c:\csound\samples\soundin.1"
ksig downsamp asig  ; I use ksmps=1 for this

; write sample value
     zkw  ksig, 1

; process sample at i-rate
     reinit    ini

; read and play the processed sample
kout zkr  1
aout interp    kout
     out  aout

     endin




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From: Jean-Michel DARREMONT 
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Dear CSounders,


May I increase PMAX parameter that is curently 150 to more than 5000 and how
to do it?
I guess I need a C compiler. As I'm not a C programmer I'd like to find a
cheap or free C compiler
for PowerMac. Does it exist on the net?
I guess I also need CSound source code; where is it available?

My work is stopped now because of this parameter limitation.

Thank you for your help.

Regards.
-- 
Jean-Michel DARREMONT



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To: Brandon Nelson 
From: Hans Timmermans 
Subject: Re: re_books
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>I bought Roads' _The Computer Music Tutorial_ in the fall, and was EXTREMELY
>disappointed. It does touch on everything, but for sound synthesis, gives
>overly wordy yet incomplete explanations and almost no examples or models that
>are easily implementable. Say bye bye, $100, and Godspeed.
>
>Brandon Nelson
>

It is an introduction into a lot of topics.
You need other books to study the topics in detail.

We are using  Roads' _The Computer Music Tutorial_  just as an introduction,
it is very good for that purpose.

Hans Timmermans                          e-mail: hans.timmermans@kmt.hku.nl
Utrecht School of the Arts              phone: (+31) 35 6836464
dep: Music Technology                    fax: (+31) 35 6836480
senior lecturer computer music

PO-BOX 2471
1200 CL HILVERSUM
the NETHERLANDS





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From: pete moss 
Organization: pete moss GmbH
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Subject: oneoverf
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i recently wrote a small program to generate one over f noise.  i know
that the values produced should have a spectrumclose to one over f.  any
ideas how to do this?  what does one over f mean?  what does f mean?
thanks

pete




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From: luis jure 
To: CSound list 
Subject: critical bandwith
Date: Wed, 1 Jul 1998 08:44:31 -0300
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RE critical bandwith, Plomp has demonstrated (R. Plomp, "Aspects of
Tone Sensation", Academic Press, 1976) that the experimental
estimation of the critical bandwith can only be a rough
approximation, since the values found are dependent on the
experimental conditions. According to his research, the values may
differ by a factor of two.

The interesting aspect of the critical bandwith is its non-linear
behaviour along the register: from about 500 Hz upwards it's a
function of the frequency (roughly 1/5 frq); for frequencies below
500 Hz it begins to remain more or less constant. That's why some
musical intervals that are larger than the critical bandwith at
higher frequencies may fall within the critical bandwith in the lower
register.

See also R. Rasch & R. Plomp, "The Perception of Musical Tones",
1982, repr. in D. Deutsch, "The Psychology of Music". This paper and
the others in the book discuss many important aspects of the
psychoacoustics of pitch, timbre, timing, grouping mechanisms in
music, etc. including an article by Risset and Wessel: "Explorations
of Timbre by Analysis and Synthesis".

luis jure