| On Wed, 5 Nov 1997 james@noether.ex.ac.uk wrote:
>
> Hi Art,
> The Extended Csound Software Developer's Kit is now available. Included
> in the kit:
> - 6-channel PCI board
> - Extended Csound software and front-ends
> - manual, documentation and tutorials
> You must have a machine running Windows95 with PCI bus.
>
> To order, please prepare the following:
> - check payable to Analog Devices, for US$1000 plus $50 shipping and
> handling
$50 shipping? For a sound card and some software?
-arlen
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Date: Wed, 05 Nov 1997 15:23:50 +0100
From: Robert Skerjanc
Organization: Heinrich-Hertz-Institut
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Subject: $170 SHARC's from Analog?
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Art Hunkins wrote:
Who knows anything about the "$170 SHARC" board from Analog Devices?
Below is the communication I received regarding the $1000 hand-wired
prototype cards being offered by Analog; this is the only offering I
know of. Have I missed out on something? (I presume this was only
hearsay/rumor.)
Hi Art,
have a look on the page
http://www.analog.com/products/sheets/ADDS2106XSHARCEZKITLITE.html
Robert
--
--------------------------------------------------------------
Dr. Robert Skerjanc Heinrich-Hertz-Institut fuer
Tel. +49 30-31002-257 Nachrichtentechnik Berlin GmbH
Fax +49 30-31002-213 Human factors department
Einsteinufer 37
http://at.hhi.de D-10587 Berlin, Germany
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From: Peter Kearton
Organization: Psychology Dept, Surrey Univ. U.K.
To: csound@maths.ex.ac.uk
Date: Wed, 5 Nov 1997 14:46:27 GMT
Subject: FM timbre
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Hi,
Whilst looking into the effect of the first fraction of time upon
perception of the whole timbre of a sound I have been playing with
some very simple FM synthesis scores. I'm very puzzled now. The .ORC
that follows is a standard single carrier FM orchestra, and two
.SCO's follow. I thought that the difference in tone and timbre
percieved in the two was a result of the initial time segments
differing, but the difference is an actual characterisic of the whole
sound. If the resulting sound files from each are edited to have the
first part removed, they still sound very different. What am I
missing here, how can they be physically different in the latter
stages ??
;
; Single Carrier FM ;
;
; Specific values and basic design come from John Chowning's
article ; ; "The Synthesis of Complex Audio Spectra by Means of
Frequency Modulation" ; ;; ; coded by Richard Boulanger - Berklee
College of Music ; ;
instr 1
; p4 = amplitude of output wave
; p5 = carrier frequency in Hz
; p6 = modulating frequency in Hz
; p7 = modulation index 1
; p8 = modulation index 2
; p9 = carrier envelope function
; p10 = modulator envelope function
i1 = 1/p3 ; one cycle per duration of note
i2 = p7 * p6 ; calculates deviation for index 1
i3 = (p8-p7) * p6 ; calculates deviation for index 2
ampcar oscil p4,i1,p9 ; amplitude envelope for the carrier
ampmod oscil i3,i1,p10 ; amplitude envelope for the
modulator
amod oscili ampmod+i2,p6,1 ; modulating oscillator
asig oscili ampcar,p5+amod,1 ; carrier oscillator
out asig
endin
;score one
f1 0 1024 9 1 1 0
f2 0 513 7 1.04 21.63 .56 149.43 .54 85.37 .54 85.53 .54 85.53 .54
85.53 .54
f3 0 513 7 .54 85.54 .54 85.53 .54 85.37 .54 85.53 .54 85.53
.54 85.53 .54 i1 0 0.5 10000 440 400 10 15 3 2 e
; score two - difference is in p5 of f2 (.55 instead of 1.04)
f1 0 1024 9 1 1 0
f2 0 513 7 .55 21.63 .56 149.43 .54 85.37 .54 85.53 .54 85.53 .54
85.53 .54
f3 0 513 7 .54 85.54 .54 85.53 .54 85.37 .54 85.53 .54 85.53
.54 85.53 .54 i1 0 0.5 10000 440 400 10 15 3 2 e
*************************************************
Peter Kearton,
University of Surrey Psychology Department,
Guildford,
United Kingdom.
GU2 5XH
Tel. UK (01483) 259437
e-mail - P.kearton@surrey.ac.uk
*************************************************
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Subject: Analog SHARC info
To: csound@maths.ex.ac.uk
Date: Wed, 05 Nov 1997 16:00:07 MET
Organization: Hewlett-Packard GmbH
From: Jens Kilian
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Just found the following on Analog's web site:
http://www.analog.com/products/sheets/ADDS2106XSHARCEZKITLITE.html
"How much does it cost to get everything you need,
including a C Compiler, to evaluate SHARCTM
DSPs and develop applications for one of the most
powerful 32-bit digital signal processors in the world? Just $179 US for the
SHARC EZ-KIT LiteTM. [...]"
Be patient; for some reason Analog's web site seems dog slow. Or perhaps it's
the connection to Europe.
Bye,
Jens.
--
mailto:jjk@acm.org phone:+49-7031-14-7698 (HP TELNET 778-7698)
http://www.bawue.de/~jjk/ fax:+49-7031-14-7351
PGP: 06 04 1C 35 7B DC 1F 26 As the air to a bird, or the sea to a fish,
0x555DA8B5 BB A2 F0 66 77 75 E1 08 so is contempt to the contemptible. [Blake]
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To: Peter Kearton
From: Russell Pinkston
Subject: Re: FM timbre
Cc: csound@maths.ex.ac.uk
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You should provide a complete score, not just the f-cards. Otherwise no-one
can test what you're doing.
Regards,
Russell Pinkston
>Hi,
> Whilst looking into the effect of the first fraction of time upon
>perception of the whole timbre of a sound I have been playing with
>some very simple FM synthesis scores. I'm very puzzled now. The .ORC
>that follows is a standard single carrier FM orchestra, and two
>.SCO's follow. I thought that the difference in tone and timbre
>percieved in the two was a result of the initial time segments
>differing, but the difference is an actual characterisic of the whole
>sound. If the resulting sound files from each are edited to have the
>first part removed, they still sound very different. What am I
>missing here, how can they be physically different in the latter
>stages ??
>
>
>
>;
>; Single Carrier FM ;
>;
>; Specific values and basic design come from John Chowning's
>article ; ; "The Synthesis of Complex Audio Spectra by Means of
>Frequency Modulation" ; ;; ; coded by Richard Boulanger - Berklee
>College of Music ; ;
>
>instr 1
> ; p4 = amplitude of output wave
> ; p5 = carrier frequency in Hz
> ; p6 = modulating frequency in Hz
> ; p7 = modulation index 1
> ; p8 = modulation index 2
> ; p9 = carrier envelope function
> ; p10 = modulator envelope function
>
> i1 = 1/p3 ; one cycle per duration of note
> i2 = p7 * p6 ; calculates deviation for index 1
> i3 = (p8-p7) * p6 ; calculates deviation for index 2
>
> ampcar oscil p4,i1,p9 ; amplitude envelope for the carrier
> ampmod oscil i3,i1,p10 ; amplitude envelope for the
> modulator
>
> amod oscili ampmod+i2,p6,1 ; modulating oscillator
> asig oscili ampcar,p5+amod,1 ; carrier oscillator
> out asig
>endin
>
>;score one
>
>f1 0 1024 9 1 1 0
>f2 0 513 7 1.04 21.63 .56 149.43 .54 85.37 .54 85.53 .54 85.53 .54
>85.53 .54
>f3 0 513 7 .54 85.54 .54 85.53 .54 85.37 .54 85.53 .54 85.53
>.54 85.53 .54 i1 0 0.5 10000 440 400 10 15 3 2 e
>
>; score two - difference is in p5 of f2 (.55 instead of 1.04)
>f1 0 1024 9 1 1 0
>f2 0 513 7 .55 21.63 .56 149.43 .54 85.37 .54 85.53 .54 85.53 .54
>85.53 .54
>f3 0 513 7 .54 85.54 .54 85.53 .54 85.37 .54 85.53 .54 85.53
>.54 85.53 .54 i1 0 0.5 10000 440 400 10 15 3 2 e
>
>
>*************************************************
>Peter Kearton,
>University of Surrey Psychology Department,
>Guildford,
>United Kingdom.
>GU2 5XH
>Tel. UK (01483) 259437
>e-mail - P.kearton@surrey.ac.uk
>*************************************************
>
>
----------------------------------
Russell F. Pinkston, D.M.A.
Associate Professor of Composition
Director, Electronic Music Studios
School of Music
The University of Texas at Austin
Austin, TX 78712
[512-471-0865]
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From: Krzysztof Foltman
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To: Khalid
Cc: csound@maths.ex.ac.uk
Subject: Re: More than two channels
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On Tue, 4 Nov 1997, Khalid wrote:
> ActiveSound may work differently. Anyway, there might
> be some clever software that manages this somehow. I
> would definitely like to know ...
You mean DirectSound, perhaps ?
There shouldn't be any problem with "1 input/output" limitation. It means
"1 default input, 1 default output". You can specify device other than
"default" in waveOutOpen/waveInOpen (?).
In DirectSound you can enumerate all wave output (or input) devices, and
use corresponding ones (you get a pointer to the IDirectSound interface,
each of them represents one sound output device).
All of these are AFAIK, of course. My memory isn't very precise :(
...kf
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Date: Wed, 05 Nov 1997 09:43:05 +0100
From: Gabriel Maldonado
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To: Khalid
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Subject: Re: More than two channels
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References: <3.0.3.16.19971024025908.4f07192e@mail.discover-net.net> <345E1959.5B6A4956@agora.stm.it> <345F69CB.25196EDA@uni-essen.de>
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> So you can, let's say, use the inputs of two different cards simul-
> taneously, like if you had a four-track machine? How do you do this?
> I have never seen a Windows software that offers this.
> thank you in advance
Software which manages quad output is very rare, but I can run two or
three instances of a wave editor (Cool editor) in parallel, each one
with a different input and output device. These three instances can be
activated with my 3 cards giving up to 6 channels in input and in
output. The problem is that I create 3 different stereo wav files and it
is difficoult to play them with a good syncronization.
--
Gabriel Maldonado
mailto:g.maldonado@agora.stm.it
http://www.agora.stm.it/G.Maldonado/home2.htm
http://www.geocities.com/SiliconValley/Way/7041/home2.htm
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> I tried to use realtime audio input with Csound using a Tahiti card,
> which is full duplex, in a Pentium 166MHz, 32 Mb Ram,
> but with no success...
> Could you send an example with orc, sco and command line ?
Current version of my Win95 Csound (1.731) does not support realtime
input. Next version (1.8) will support it. I'm waiting for the sources
of the definitive release of 'official' Csound 3.47 to include all new
opcodes before uploading it to the Internet.
--
Gabriel Maldonado
mailto:g.maldonado@agora.stm.it
http://www.agora.stm.it/G.Maldonado/home2.htm
http://www.geocities.com/SiliconValley/Way/7041/home2.htm
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Date: Wed, 5 Nov 1997 08:45:46 -0800 (PST)
From: Mike Berry
Reply-To: Mike Berry
To: Peter Kearton
Cc: csound@maths.ex.ac.uk
Subject: Re: FM timbre
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In FM, the starting phase relationships are going to have an
effect throughout the length of the note. So if you have two notes, one
which is modulated slightly faster just at the beginning, when they settle
down into the part where the should seem to be the same, you are almost
certainly going to have a phase difference between the two modulators.
The only way to avoid this is to do a calculation that ensures that your
changes at the beginning result in no phase shift in the later sections.
Mike Berry
mikeb@mills.edu
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To: csound@maths.ex.ac.uk
Subject: Re: Analog SHARC info
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Jens Kilian wrote:
> Just found the following on Analog's web site:
>
> http://www.analog.com/products/sheets/ADDS2106XSHARCEZKITLITE.html
>
> "How much does it cost to get everything you need,
> including a C Compiler, to evaluate SHARCTM
> DSPs and develop applications for one of the most
> powerful 32-bit digital signal processors in the world? Just $179 US for the
> SHARC EZ-KIT LiteTM. [...]"
>
> Be patient; for some reason Analog's web site seems dog slow. Or perhaps it's
> the connection to Europe.
>
Looks great but any idea as to how useful this card would be for the upcoming new
commercial version of csound? Or the current free version for that matter?
Also, what about using it from Linux? Would this ONLY be useful if I were using
it from dreaded Win95? Can I use this as my primary soundcard for standard audio
applications in win95/linux? Comments....?
Thanks
John
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