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csound versions

Date1998-09-17 06:56
Frompete moss
Subjectcsound versions
sorry for the rather lame question, but what is the difference between
the csound.exe file in csound_new.zip and csound_con.zip?  they both
have the same date and time, but the build in csound_con.zip is much
smaller.  also, what version is the most recent for these versions of
csound?
the reason i ask is because i have been using winsound, but i think i
want to switch to the command line version, but i am curious why
winsound sports a later version number and build date.

pete

Date1998-09-18 14:40
Fromjpff@maths.bath.ac.uk
SubjectRe: csound versions
csound_new.zip is an MSDOS build
csound_con.zip is a Windows console-mode build

While I was in the USA this summer I did not have access to the
machine on which I do teh DOS build sdo I have not updated it.  Thanks
for reminding me.  Will get back to it as soon as I can.
==John


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Date: Fri, 18 Sep 1998 06:54:19 -0700 (PDT)
From: Richard Karpen 
To: Josep M Comajuncosas 
cc: csound@maths.ex.ac.uk
Subject: Re: synthesizing with wavelet packets
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I'll be getting back to work soon on convovle and pvoc units that do the
analysis from within the instrument. The pvoc works already in my version,
the convolve might be done in a month or so.

RK

 On Fri, 18 Sep 1998, Josep M Comajuncosas wrote:

> My God, Wavelet resynthesis in orchestra code...!!!
> I wish we could have some wavelet analysis/resynthesis tool, similar to
> that already implemented which uses Short-time fourier transforms
> (pvoc).
> Let me add I think we have now enough computer power to be able to call
> the analysis routines from within the instrument code itself, so for
> example, we could use multiple convolve files without having to
> carefully prepare all the analysis files before (I=B4m thinking of
> implementing an orchestra-based HRTF convolver...), lots of other
> possibilities btw...
>=20
> Josep M Comajuncosas
>=20
>=20
>=20
>=20



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CC: Antoine Lefebvre , 
    Csound mailing list 
Subject: Re: need information
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Antoine,

    Try  "The computer music tutorial" by Curtis Roads (MIT press).
    It's a book that covers "everything" (in a sense) and lets you in on most
synthesis an modulation
    techniques plus all the basic digital-audio
concepts,programming,MIDI,psychoacoustics,sound
    analysis and more.

    It's really thick and makes you come back to it quite often.

        Yair Kass

Gareth Whittock wrote:

> You can find some well written sound theory stuff at:-
> http://www.sfu.ca/sca/Manuals/CLMs.html
> Gareth Whittock
>
> Antoine Lefebvre wrote:
>
> > I work with csound since only few months and now that I understand a
> > great part of the opcodes, I find myself in a biggest problem: I don't
> > have all the theorycal knowledge to be able to reproduce in csound code
> > what I heard in my head. So I would to know if there is some very good
> > theorycal books and reference that can help me with in my works!!!
> >
> > Thanks
> >
> > Antoine Lefebvre





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In-reply-to: <36022502.72322E35@intercom.es> (message from Josep M
	Comajuncosas on Fri, 18 Sep 1998 11:16:50 +0200)
Subject: Re: synthesizing with wavelet packets
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>>>>> "Josep" == Josep M Comajuncosas  writes:

 Josep> My God, Wavelet resynthesis in orchestra code...!!!
 Josep> I wish we could have some wavelet analysis/resynthesis tool, similar to
 Josep> that already implemented which uses Short-time fourier transforms
 Josep> (pvoc).

My student Wafaa is working in it.  No doubt others as well.
==John




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Date: 18 Sep 98 11:38:26 EDT
From: "Matt J. Ingalls" 
Subject: Re: synthesizing with wavelet packets (fwd)
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i thought these were in extended csound (the complete implementation of
spectral data types) and were going to be "donated" to free-csound?? am i
confused here???

-matt

p.s. by the way-if you have noticed my new address i am here @dartmouth
teaching a theory class(of all things) for the fall - jon appleton has been
nice enough to let me use his office while he is away which is part of their
studios so am planning on doing a lot of csound/eamus development/composing!!


>I'll be getting back to work soon on convovle and pvoc units that do the
analysis from within the >instrument. The pvoc works already in my version, the
convolve might be done in a month or so.


 On Fri, 18 Sep 1998, Josep M Comajuncosas wrote:

> Let me add I think we have now enough computer power to be able to call

> the analysis routines from within the instrument code itself, so for





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Great idea, I'll be sending you something soon.

             Yair Kass    (only 5 months on the list)

Tobiah wrote:

> I would like to create a web site where we can
> exhibit the cast of the csound mailing list.
> For inclusion please send me a picture, a text,
> and a sound, or any combination of these.  I will
> assemble the pieces into a nice web site where
> new and old members can get aquainted with one
> another.
>
> Toby
>
>         -There otta be a law-





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Subject: Re: maximizer opcode request
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>to boost or cut and by how much based on the content of the file. i think.
>expect that here again sones come into play.
					yes, sones==subjective so has to be user settable

>maximize opcode should probably also have built in filters to eliminate
>the primary use in mind for a maximize opcode would be for files that are
>already normalized or floating pointed and still somehow do not sound loud
>though already do-able in csound with filters and gain, a maximize opcode
>would be a far faster way of dealing with large files to boost the overall
						why would it be faster?  this all sounds like it would be better as
				an instrument rather than an opcode - you could then easily tweak things
				how you want easily.
>perceived volume, even of an entire mix -and maximizers appear to be in
>wide use for mastering CD's despite the fact that most popular music is of
				is "maximizer" a standardized term?  if so, what exactly does it mean?

-matt


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Date: Fri, 18 Sep 1998 12:14:59 -0600
From: Mike Berry 
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There has been a lot of discussion about maximizers, so I thought I would add
my thoughts.

	A maximizer is also called a band-specific compressor.  An input signal is
run through a filter bank first.  Then each band is separately compressed. 
Certain extraneous bands (above or below a certain freq, for instance) may be
removed or highly suppressed.  Then the signal is recombined.  The output then
is much more even-sounding (i.e. louder) than it probably was before.

	The algorithms for doing this are generally proprietary.  One example (that
we don't need to implement) is the Optimod by Orban.  This is designed for the
final compression stage of a radio broadcast, where there should be less that
5% (!!!) of difference between the loud passages and the quiet ones, so that
listeners in cars don't ever loose the signal.

	Mastering houses often use a less extreme version for CD's, so that it sounds
as loud as possible.  For those of us who sometimes don't want it to be as
loud as possible, you need to specifically request this.

	So where does csound stand in this?  It seems unlikely that any of us is
going to come up with an algorithm that is as good at doing this as your pro
plug-in of choice, like the Ultramaximizer.  These companies exist for the
sole purpose of doing this.  So my suggestion is that instead the focus should
be on a good filter bank opcode.  This is something that would be useful in a
number of situations.  It could look like:

a1 (... an), arest		filtbank	kbands, kbase, kspacing

where:
	kbands = the number of bands
	kbase = the top frequency of the first band
	kspacing = the width in octaves of each band
	a1...an = the output of each band
	arest = the remaining signal, delayed to be in sync with the bands

	Then each band could be manipulated separately and then recombined at will. 
With this, it would be easy to experiment with your own maximizer orchestra. 
This is not a trivial opcode, since designed the necessary filters is rather
tricky.  But there must be some research somewhere on how to do this.

-- 
Mike Berry
mikeb@nmol.com
http://www.nmol.com/users/mikeb




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A while ago, I volunteered to collect csound benchmarks made using the
piece "Xanadu" by Joseph Kung. This piece gives a more reasonable
workout for modern hardware than the old benchmarks did. I got a couple
of replies from folks on the list. I've now collected them into a web
page at:

http://members.tripod.com/~slinkP/pw_linux/csbench.html

Send me more benchmarks! So far I just have Robin Whittle's PPro 180, my
P133, several machines tested by jpff, and a P166 tested by Guto
Caminhoto. Macs are conspicuously absent.

Regards,

PW



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From: jp 
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To: Mike Berry , csound 
Subject: Re: maximizer opcode request
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Mike Berry wrote:
> 
> There has been a lot of discussion about maximizers, so I thought I would add
> my thoughts.


> 
> a1 (... an), arest              filtbank        kbands, kbase, kspacing
> 
> where:
>         kbands = the number of bands
>         kbase = the top frequency of the first band
>         kspacing = the width in octaves of each band
>         a1...an = the output of each band
>         arest = the remaining signal, delayed to be in sync with the bands
> 
>         Then each band could be manipulated separately and then recombined at will.
> With this, it would be easy to experiment with your own maximizer orchestra.
> This is not a trivial opcode, since designed the necessary filters is rather
> tricky.  But there must be some research somewhere on how to do this.

Let me chime in for that. 
optionally run the filtbank at krate and obtain rms of each band as output?


-- 
________________________________________________________
Jean Piche
Universite de Montreal
http://mistral.ere.umontreal.ca/~pichej
http://www.musique.umontreal.ca/electro/CEC/


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Date: 18 Sep 98 15:44:29 EDT
From: "Matt J. Ingalls" 
Subject: Re: maximizer opcode request
To: pichej@ere.umontreal.ca, Mike Berry , 
    csound 
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> a1 (... an), arest              filtbank        kbands, kbase, kspacing
> 
> where:
>         kbands = the number of bands

	would kbands be necessary since number of outputs = kbands?

-mutt


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From: Mike Berry 
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Matt J. Ingalls wrote:
> 
> > a1 (... an), arest              filtbank        kbands, kbase, kspacing
> >
> > where:
> >         kbands = the number of bands
> 
>         would kbands be necessary since number of outputs = kbands?
> 
	I suppose not.  kbands would be the number of outputs - 1 (the 1 being the
rest of the spectrum).
-- 
Mike Berry
mikeb@nmol.com
http://www.nmol.com/users/mikeb




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Date: Sat, 19 Sep 1998 06:53:25 +0100
From: Lex Vanderwal 
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Subject: Any Japanese Csound users on this list?
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Hello

If you are Japanese, please send me an e-mail.
I am on this list for two years, studied computer music at he Institute
of Sonology in Utrecht, the Netherlands (I met Takayuki Rai there - a
great composer...), I am studying Japanese and am going to Japan next
year.

Thanks in advance.
Greetings,
   Lex.



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From: Richard Dobson 
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Subject: Re: maximizer opcode request
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I see what you mean; but I worry that the audio quality will be much less than
what would be expected intuitively. If a filter has a gain of 30dB, which is
added or otherwise combined with existing material without normalizing it, that
will have the effect of pushing the previous material down by the same 30dB,
eventually descending below audibility into quantization noise (hence the need
for a maximiser, I suppose!).

In the worst case, the number cannot accomodate the addition, and will go
denormal, truncate, or whatever, which would lead to even greater distortion.

Of course, if the 'user' is a Csound user, they have control of this at every
stage (and the peak values in the header will add much to that control), but if
that user is a non-technical user of a GUI environment, they may have no control
at all, and will simply be surprised by what comes out of the DAC. 

I admit this is mostly theorizing on my part, as I always ensure everything I do
is within range; I suppose I will have to create some huge-ranged non-normalized
sounds and see what happens. It's good to have a tool like Csound which enbles
me to do just that, without hiding anything!

Richard Dobson

Matt J. Ingalls wrote:
> 
> > you will need to normalize the data in order to play it on a real-world sound
> > card - and be limited by that card's resolution;
> 
>         of course-but from a "user" point of view there would be no
> normalization-the playback would just be "right" leaving her file
> untouched
> 
[etc]