| Steve Kersten wrote:
> 1. Is there a way of triggering score events from within an instrument
> (besides the schedule opcode)? More specifically I'm wondering if it is
> possible to 'include' whole score files and thus be able to treat them
> as a single event.
In DirectCsound there is the 'callX' family of opcodes (see manual in
DirectCsound distribution)
--
Gabriel Maldonado
http://web.tiscalinet.it/G-Maldonado/home2.htm
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From: Gabriel Maldonado
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Steve Kersten wrote:
> 1. Is there a way of triggering score events from within an instrument
> (besides the schedule opcode)? More specifically I'm wondering if it is
> possible to 'include' whole score files and thus be able to treat them
> as a single event.
In DirectCsound there is the 'callX' family of opcodes (see manual in
DirectCsound distribution)
--
Gabriel Maldonado
http://web.tiscalinet.it/G-Maldonado/home2.htm
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Steve Kersten wrote:
> 1. Is there a way of triggering score events from within an instrument
> (besides the schedule opcode)? More specifically I'm wondering if it is
> possible to 'include' whole score files and thus be able to treat them
> as a single event.
In DirectCsound there is the 'callX' family of opcodes (see manual in
DirectCsound distribution)
--
Gabriel Maldonado
http://web.tiscalinet.it/G-Maldonado/home2.htm
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From: Gabriel Maldonado
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Steve Kersten wrote:
> 1. Is there a way of triggering score events from within an instrument
> (besides the schedule opcode)? More specifically I'm wondering if it is
> possible to 'include' whole score files and thus be able to treat them
> as a single event.
In DirectCsound there is the 'callX' family of opcodes (see manual in
DirectCsound distribution)
--
Gabriel Maldonado
http://web.tiscalinet.it/G-Maldonado/home2.htm
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From: Gabriel Maldonado
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Steve Kersten wrote:
> 1. Is there a way of triggering score events from within an instrument
> (besides the schedule opcode)? More specifically I'm wondering if it is
> possible to 'include' whole score files and thus be able to treat them
> as a single event.
In DirectCsound there is the 'callX' family of opcodes (see manual in
DirectCsound distribution)
--
Gabriel Maldonado
http://web.tiscalinet.it/G-Maldonado/home2.htm
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From: Steve Kersten
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To: Gabriel Maldonado ,
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Subject: direct csound: commandline file
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Hallo.
I encountered a problem with reading the commandline-options in
DirectCsound (2.8) from a file:
D:\> dsound -+/foo.cmd foo.orc foo.sco
gives the following output:
Csound Version 3.52 (Apr 10 1999)
USING file 'foo.cmd' as command line
orchname: foo.orc
scorename: foo.sco
sorting score ...
... done
orch compiler:
42 lines read
instr 1
MIT Csound: 3.52 (Apr 10 1999)
orch now loaded
displays suppressed
audio buffered in 400 sample-frame blocks
writing 1600-byte blks of shorts to D:\audio\sfdir/test (AIFF)
SECTION 1:
ftable 1:
ftable 2:
[...]
while my test.cmd includes:
-AH -dm0 -+O -+q
which should at least prompt me for the sound device and write to
devaudio (?). Or am I missing something?
Thanks, Steve.
PS: Further investigation led me to the assumption that only the very
first argument is parsed (?)
________________________________________________________________________
K-Labz [a K-Hornz subdivision] - steve-k@gmx.net - http://w3.to/K-Hornz
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From: Steve Kersten
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To: Richard Dobson
CC: Csound
Subject: Re: 'scheduling' scorefiles?
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Richard Dobson schrieb:
> The Windows version doesn't have code to open multiple cards anyway.
> However, the number of channels is passed directly to the MME driver, so
> that if a real multi-channel card is available, Csound should be able to
> use it up to as many channels as they both support. I will be trying
> this very thing out at home soon, using the Creamware Pulsar.
I'd be happy if you could post the results. I suppose no source-code
changes are required to use this feature (on win)? Is it possible at all
(within MME or DirectX) to open multiple devices?
> As latency is generally determined by buffer sizes in bytes, it may be
> somewhat better for quad streams and higher, than it is for mere stereo.
I don't quite understand. Wouldn't latency double since you would have
to circulate through the buffers twice as often (stereo <-> quad)?
Steve.
________________________________________________________________________
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From: David Boothe
Subject: Re: 'scheduling' scorefiles?
To: Richard Dobson , Csound
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Sonorus has Linux drivers available in OSS via 4Front Technologies. Also there
is a Wiindows NT driver, which I am using and works just fine, as well as Mac
and Win9x.
The only problem with the Sonorus card that you have to solve, is that the I/O
is ADAT lightpipe only (16 channels in and out), plus one stereo analog output
for monitoring purposes. In the right environment, though, it is a very nice
card to use. I've been using it for about 4 months, with no problems, once I
got a few start-up glitches worked out. Also, Sonorus tech support is pretty
responsive.
http://www.sonorus.com
-David.
--- Richard Dobson wrote:
> I understand that m/c drivers are planned for Linux, for cards such as
> the Sonorus STUDI/O - but how imminent they are, I don't know.
>
_________________________________________________________
Do You Yahoo!?
Get your free @yahoo.com address at http://mail.yahoo.com
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From: Steven Cook
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Hi,
What are there any advantages to using sum or product instead of + or *, other
than neatness?
Thanks,
Steven Cook.
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From: Torsten Anders
To: Hans van den Tillaart
Subject: Re: Piano sound characteristics
Date: Thu, 15 Jul 1999 22:07:16 +0200
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Hi,
have a look at clm, a sound synthesis language translating common lisp to c.
There is a very well piano programmed by Scott van Duyne. It is not only
velocity sensitive, but also you can tune the strike point, the string
stiffness, add as more sustain pedal as you want and really much more. Some
short texts on it I found in the proceedings of the ICMC, but I'm sure there
will be more.
BTW: I like clm because of it's possibility to normalize automatically a
rendered sound after processing. I miss such a feature in csound: is that in
csound possible too?
Torsten
On Wed, 14 Jul 1999 you wrote:
>Hello,
>I want to know as much there is on piano sound characteristics. I am
>trying to emulate piano or pianolike sounds in CSOUND.
>Which formants are there in play?
>Do they alter when you increase velocity and or playing from low to
>high?
>What does exactly change in the spectrum when velocity is increased?
>How does the sound evolve over time when looking at the spectrum ,
>attackfase release?
>Sure there are more characteristics when trying to emulate ... wich?
>Is there software for investigating formants?
>
>Many thanks in advance!!
>Greetings Hans van den Tillaart.
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From: JM&M DARREMONT
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You should use Sound Hack to display the curve or the spectrum of piano t=
ones.
You'll learn a lot in an intuitive way.
It works on Mac and is freeware.( Maybe on Windows to?)
Hans van den Tillaart a =E9crit:
>=20
> Hello,
> I want to know as much there is on piano sound characteristics. I am
> trying to emulate piano or pianolike sounds in CSOUND.
> Which formants are there in play?
> Do they alter when you increase velocity and or playing from low to
> high?
> What does exactly change in the spectrum when velocity is increased?
> How does the sound evolve over time when looking at the spectrum ,
> attackfase release?
> Sure there are more characteristics when trying to emulate ... wich?
> Is there software for investigating formants?
--=20
Jean-Michel DARREMONT
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From: Richard Dobson
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To: Csound
Subject: Re: 'scheduling' scorefiles?
References: <199907142244.SAA15906@renoir.op.net> <378D1B83.5066B96B@cableinet.co.uk> <378E17D8.C1F08E90@gmx.net>
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Well, I was not quite correct before - there is still some code which
only accepts audio which complies with the old SoundBlaster standard, so
no quad files, and no 48KHz srate either! This code is really
unnecessary as any unacceptable formats will be rejected by
waveOutOpen() anyway.
By cutting out the format code bits (long switch blocks) I have Csound
playing a quad stream through the Pulsar card. So it can be done.
It is possible programmatically to open multiple devices, but for
multi-channel streams (for which you want perfect synchronisation) this
is a most unsatisfactory way of doing things.
Latency is not the same as write speed. It is true that multi-channel
data must be written to a card that much more quickly than a mono or
stereo stream, bt this is hardly an issue on modern machines. Latency
relates to the smallest block size you can send to the card, and is an
issue for real-time audio processing (in and out) and for MIDI control.
If you have a buffer size of 2048 bytes, for example, this means that
you won't hear anything until those bytes have all been sent to the
card. The latency is related to the 'frame rate', where one frame =
(2048 / wordsize_in_bytes / nchans).
Thus for 16bit stereo the latency is 512 frames, which is some 11.6
msecs at 44100 hz.
A quad stream will fit 256 frames into the same space, so the latency is
a pleasant 5.8 msecs.
The more devious readers of this list, in posession of a multi-channel
card, might conjecture that low latency (notwithstanding other technical
obstacles such as interrrupts) could be achieved even with large block
sizes, by writing their stereo stream packed into an eight-channel one.
Richard Dobson
Steve Kersten wrote:
>
> Richard Dobson schrieb:
> > The Windows version doesn't have code to open multiple cards anyway.
> > However, the number of channels is passed directly to the MME driver, so
> > that if a real multi-channel card is available, Csound should be able to
> > use it up to as many channels as they both support. I will be trying
> > this very thing out at home soon, using the Creamware Pulsar.
>
> I'd be happy if you could post the results. I suppose no source-code
> changes are required to use this feature (on win)? Is it possible at all
> (within MME or DirectX) to open multiple devices?
>
> > As latency is generally determined by buffer sizes in bytes, it may be
> > somewhat better for quad streams and higher, than it is for mere stereo.
>
> I don't quite understand. Wouldn't latency double since you would have
> to circulate through the buffers twice as often (stereo <-> quad)?
>
> Steve.
> ________________________________________________________________________
> K-Labz [a K-Hornz subdivision] - steve-k@gmx.net - http://w3.to/K-Hornz
--
Test your DAW with my Soundcard Attrition Page!
http://wkweb5.cableinet.co.uk/rwd (LU: 6th July 1999)
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From: ras kizo
To: csound@maths.ex.ac.uk
Subject: Midi and drums
Date: Fri, 16 Jul 1999 02:49:07 PDT
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Hello !
I wonder wether someone knows a way to write an orchestra that will read
several midi event on channel 10 and will play then using soundin.
I have been trying to write such an orchestra but it never worked.
What i want to do is for example read a midi drum pattern with a snare and
a bass drum sound and assign them wave samples.
How can i do that ?
Thanks in advance !
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Message-ID: <378425B5.C40D7A37@usa.net>
Date: Wed, 07 Jul 1999 23:14:45 -0500
From: "craig-routt@usa.net"
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i know i'm being redundant but i've got this vision
about pluck that it needs to be frequency modulated
in the time domain.
for instance you run 2 pluck parrallel and feed back
the out put with weighted wet or dry into feedback
ok you got that . but the time domain must shift the feed back time.
this is not my idea . this comes from a paper i read by bose.
who just happens to sell the best speaker system in the world
called the bode 802 with a sound cannon for base.
but in a conversation with the fellow he said if you use one
speaker and run a 100 hz. signal into it with a 1000 hz signal
you get doppler shift. in a pluck this means time domain must change
in the feed back loop.
this is a problem which has plagued me for 20 years. when i would
synthesizie a drum on my moog it was not the same as the actual
drum in my recording suite.
WHY. WHY WHY WHY WHY WHY WHY
THEN IT OCCURRED TO ME
DOPPLER SHIFT OF THE TIME DOMAIN OF FEED BACK!!!!!!
OK now you got that.
the partial s must fm .
this is mandatory.
perhaps you fellows have solved this allready. i have been
negligent in my reading of cmj recently (2 years now)
but this question has bothered me for 20 years.
PLEASE PLEASE HELP.
thank you
craig
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To: csound@maths.ex.ac.uk
From: James McCartney
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At 10:14 PM -0600 7/7/99, craig-routt@usa.net wrote:
>WHY. WHY WHY WHY WHY WHY WHY
>
>THEN IT OCCURRED TO ME
>
>DOPPLER SHIFT OF THE TIME DOMAIN OF FEED BACK!!!!!!
Bartender, I'll have whatever he's having.
--- james mccartney james@audiosynth.com http://www.audiosynth.com
If you have a PowerMac check out SuperCollider2, a real time synth program:
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Date: 7 Jul 99 23:34:51 MDT
From: craig routt
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oh , for those of you who are not drummers or studio owners
with drum kits or have not analized a drum hit.
the first hit of a drum is direct output.
the second hit is a doppler shift of the first
strike of the head. which would indicate a logic
if 2ond note switch on doppler shift, that is also apparrent.
but it would be fm but by a doppler shift of which would
give an amplitude modulation in accordance with the
resonant body this is what i base my supposition on
=2E so to do this would i need to rewrite
pluck to do time base/feedback modulation but related
to not necessarily fundamental but a mix of resonance
as well as feedback and fundamental. and with a couple of =
more p field for resonant/time domain shift,
parrallel pluck units in the resonator path,
(oh boy right here i remember reading this somewhere)
with curves weighted for each parallel harmonic to
be thrown into the feedback. but the feedback is not =
just time function. true?
the parrallels wouldgive resonant body shape.
(later you could do a gui to draw shapes in theory)
i'm still thinking on this.
i'm sort of thinking out loud to the csound
group . hope you do not mind.
thank you =
craig
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