| All what you are speaking about is made by Audiosculpt pretty well I think.
It's open to other softwares by text files. I use it with Open Music, CSound,
Patchwork and Diphone.
It can be interesting for analysis then resynthesis in CSound with
modification in Open Music.
It's also very much used as a way to transform a sound into score and make it
played by an orchestra or the reversed way(orchestra sequence be resynthesized
in CSound). In Open Music is a library for connection with Audiosculpt and
making very similar treatments than those you describe in your mail.
Maybe you know it already. Otherwise you can go to:
http://www.ircam.fr/forumnet/
> Michael Gogins wrote:
>
> One of the cool things you can do in computer music is time/frequency
> morphing. I am interested in it for several purposes, but the usual
> techniques don't satisfy me. I'd appreciate some feedback or
> discussion of the underlying issues.
>
> What I want to do, without introducing artifacts, is:
>
> Stretch or shrink time without affecting pitch.
> Transpose pitch without affecting time.
> Combine stretching and shrinking of time with transposition of pitch.
> Translate visual images into sounds.
> Translate sounds into particularly intelligible visual images.
> Generate fractals and translate them into sounds.
> Color one sound with another, perhaps with the two sounds running on
> different scales of time or pitch.
>
> I have done all of the above, sometimes in several different ways,
> usually with Cool Edit Pro or Csound, sometimes with software written
> by me. So, I think I have learned some things:
>
> The phase vocoder can be exact for some transformations, but
> introduces artifacts for many transformations.
>
> Filterbank analysis/resynthesis is not exact, but does not introduce
> artifacts if enough filters and oscillators are used, and can in fact
> be extremely realistic, more so in practice than the phase vocoder.
> However, it does not represent sound using any sort of regular grid
> into which a fractal or image could be quantized in an intuitive way.
>
> It seems that sonograms are effective pictures of sounds, but making
> sonograms and translating them back into sound is not really a very
> good way of working.
>
> The acoustical and musical imagination morphs mental images of sounds
> in some extremely sophisticated way that keeps phase effects, smearing
> from convolution, and so on completely in the background. It seems to
> be possible for the mind to switch back and forth between a "grid" or
> "sonogram" picture of sound and a "harmonic" or "filterbank" picture
> of sound without even thinking about it.
>
> To map a fractal or image onto a time/frequency/phase grid almost
> invariably produces a startling array of artifacts including clicks,
> ringing sounds, smearing, and burbling. However, when I look at a
> picture of the fractal or at the image, I can easily imagine what I
> want the damned thing to sound like.
>
> One of the things I want to do is to decompose a sound into grains, or
> some other kind of elementary "note", that I can manipulate
> compositionally in my Silence software just as I do regular musical
> notes.
>
> In fact I, can ALREADY do this by translating hetro analysis files
> into scores of notes that I synthesize using slurs, so that the result
> for each track of the filterbank analysis is a connected track of
> sinusoidal sound. However, this technique does NOT lend itself to
> representation of fractals. Nor does it allow one to color one sound
> with another except by (a) mixing or (b) convolving the oscillator
> signal with some impulse.
>
> I also want to color sounds with each other in ways that do not
> produce convolution artifacts. For example, I would like to record the
> sound of the wind blowing through the leaves of trees on a summer
> evening and color it with some pitched sounds or with voices, or give
> the sounds themselves some sort of pitch, perhaps by bringing out
> pitches that are inherent in the sounds without otherwise distorting
> their timbre. I can IMAGINE the results but I do NOT know how to
> compute them.
>
> Some of the above is probably confused or uninformed, but - any help?
--
Jean-Michel DARREMONT
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From: Jean-Michel DARRMONT
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Subject: Re: Using ties
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David Meckstroth wrote:
>
> Would someone be kind and send me a short example (sco & orc) (or point me in
> a direction where I could find one) of how to use ties (tival, tigoto,
> fractional i #s). Thank you in advance. -David
I've asked this question few weeks ago and David Kirsh sent me these orc&sco:
;----- Ties orchestra with dclic ----------
sr = 44100
kr = 44100 ; as usual it sounds much better with kr=sr specially on attacks
ksmps = 1
nchnls = 1
instr 5
inote = cpspch(p4) ; This note's pitch
idur = abs(p3)
istied tival
tigoto tieinit
ibegpitch = inote
iprevpitch = inote
goto cont
tieinit:
ibegpitch = iprevpitch
iprevpitch = inote
cont:
kpitchenv linseg ibegpitch, .9, inote, abs(p3), inote
; If this is a tied note, then declick envelope amplitude
; starts out at '1' (i.e., same as ending value of note
; we're tying onto). Otherwise, it starts at '0'.
ibegdclic = (istied == 1 ? 1 : 0)
; If this note is being held (in anticipation of being tied
; onto), then declick amplitude ends at '1' (so it
; matches the beginning declick amplitude of the tied-on note).
; Otherwise, amplitude ends at '0'.
ienddclic = (p3 < 0 ? 1 : 0)
kdclic linseg ibegdclic, 0.1, 1, abs(p3)-0.2, 1, 0.1, ienddclic
a1 buzz 10000, kpitchenv, 6, 1, -1
out a1 * kdclic
endin
;----- Ties score ----------
f1 0 32769 10 1
;p1 decimal allows polyphony.
;
i5.1 0.0 -2 7.00
i5.2 0.0 -2 7.04
i5.1 2.0 -2 7.07
i5.2 2.0 -2 7.11
i5.1 4.0 -2 6.07
i5.2 4.0 -2 7.00
i5.1 6.0 2 7.00
i5.2 6.0 2 6.00
i5 8.1 .8 7.00
i5 9.1 .8 7.04
i5 10.1 .8 7.07
i5 11.1 .8 7.00
e
--
Jean-Michel DARREMONT
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Date: Thu, 04 Mar 1999 15:10:20 +0000
From: Jean-Michel DARRMONT
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To: Sergey Batov
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Sergey Batov wrote:
>
> Hope this is the thing you are looking for (sorry, I can't tell who sent
> this example.
> Is it possible to know who is the author ?)
>
This is from David Kirsh but this one produces clics between the notes.
He sent me later another one wich I put in the previous message and it works fine.
> ; ============== tied notes.orc ========
> sr = 44100
> kr = 882
> ksmps = 50
> nchnls = 1
>
> ; Demonstration of portamento between tied notes
> ; This technique could be applied to other parameters and to
> ; non-tied notes with straightforward changes.
> instr 5
> inote = cpspch(p4) ; This note's pitch
>
> ; If this is not a tied note, then there's no previous
> ; pitch to glide from, so set beginning pitch of
> ; portamento to this note's pitch. Then save this same
> ; pitch in 'iprevpitch' so we can remember what pitch we
> ; are coming from when we have a tied note. We have to
> ; save it both here and in the tied note case because if
> ; we do it in a common place below, Csound complains about
> ; iprevpitch being used before set.
> tigoto tieinit
>
> ; At this point we know we're not in a tied note.
> ; Set beginning and previous pitches to the same thing.
> ibegpitch = inote
> iprevpitch = inote
> goto cont
>
> tieinit:
> ; We're in a tied note.
> ; Set up beginning pitch for portamento to new pitch.
> ; Beginning pitch is pitch of previous note that this one is
> ; tied onto the end of. Save current pitch in case another
> ; note ties onto this one; then we'll be able to portamento
> ; again to that new pitch.
> ibegpitch = iprevpitch
> iprevpitch = inote
>
> cont:
> ; Now set up a pitch envelope that moves from the beginning
> ; pitch (as determined above) to this note's nominal pitch.
> kpitchenv linseg ibegpitch, .9, inote, abs(p3), inote
>
> ; Generate a sound using the pitch envelope. Notice the
> ; last 'buzz' argument (-1), which causes phase
> ; initialization to be skipped. This is important to avoid
> ; clicks on tied notes. (This simple example clicks on
> ; untied notes, though, because there's no amplitude
> ; envelope.)
> a1 buzz 10000, kpitchenv, 6, 1, -1
> out a1
> endin
>
> ;=========== tied notes.sco ====================
>
> ; A Little Test Music
> f1 0 1025 10 1
> ;
> i5.1 0.0 -2 7.00
> i5.2 0.0 -2 7.04
> i5.1 2.0 -2 7.07
> i5.2 2.0 -2 7.11
> i5.1 4.0 -2 6.07
> i5.2 4.0 -2 7.00
> i5.1 6.0 2 7.00
> i5.2 6.0 2 6.00
> e
--
Jean-Michel DARREMONT
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Message-ID: <283AABB8FD0DD21187C200A0C995F5DE0ECE76@neptune.lyrick.com>
From: David Boothe
To: 'Csound'
Subject: Csound on NT
Date: Thu, 4 Mar 1999 08:50:00 -0600
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I'm thinking of moving to a Windows NT machine. Does anyone use Csound under
NT and does it work OK?
Thanks in advance.
-David.
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From: Josep M Comajuncosas
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Subject: Fat pad
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Hi,
I don=B4t see too many orchestras posted to the list these days, so I=B4l=
l
contribute with a nice analogish pad I wrote this afternoon,
notice there is a too obvious tremolo which seems to be caused by the
nonlinear phase response of the cascaded tone filters (I didn =B4t
expected it!).
sr =3D 44100
kr =3D 44100
ksmps =3D 1
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
instr 1;ultrafat FM/analog/waveshape pad
;Another Emphyrical Instrument
;Josep M Comajuncosas / feb=B499
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
aout init 0
kdeclick linen 1, .01,p3,.01
;cascaded FM section (I got this from somewhere)
ifdbk =3D .6;extra feedback to the LPF bank
iamp =3D 1
ipor =3D cpspch(p4)
imod1 =3D 2*ipor
indx1 =3D 3
imod2 =3D 1.33*ipor
indx2 =3D 3
kndx1 linseg 0,p3/3,indx1,p3/2,indx1
kndx2 linseg 0,p3/2,indx2,p3/3,indx2
amod1 oscil3 imod1*kndx1,imod1,1
amod2 oscil3 imod2*kndx2,imod2+amod1,1
apor oscil3 iamp/2,ipor+amod2, 1
kamp linen 1,p3/3,p3,p3/3
;Analog LPF section
kfco linseg 200, p3/4, 10000,p3/4, 4000, p3/2,50
kres linseg 0,p3/3,0,p3/3, .5 ,p3/3, .2
alpf0 moogvcf apor*kamp-ifdbk*aout, kfco,kres
adist =3D .5*tanh(apor*kamp);smooth distortion
alpf1 tonex -adist,sr/4;play with phases ;-)
alpf2 tonex alpf1,sr/8
alpf3 tonex -alpf2,sr/16
alpf4 tonex alpf3,sr/32
;Multiband Waveshaper section
awsh1 table3 .5+alpf1,2,1
awsh2 table3 .5+alpf2,3,1
awsh3 table3 .5+alpf3,4,1
awsh4 table3 .5+alpf4,5,1
;Output
aout =3D awsh1+awsh2+awsh3+awsh4
out 2000*aout * kdeclick
endin
;sine wave
f1 0 8192 10 1
; tr.ft
f2 0 8193 13 1 1 0 6 -2 0 0 0 0 0 0 0 0 0 0 0 0 0 0
f3 0 8193 13 1 1 0 0 0 -2 4 0 0 0 0 0 0 0 0 0 0 0 0
f4 0 8193 13 1 1 0 0 0 0 0 5 -1 -7 3 0 0 0 0 0 0 0 0
f5 0 8193 13 1 1 0 0 0 0 0 0 0 0 0 8 -5 -6 2 4 -1 -2 1
i1 0 5 6.00
i1 4 . 7.00
i1 8 . 8.00
i1 12 . 9.00
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From: Wayne Freno
To: Csound List
Subject: Grain opcode & audio files
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I guess my last post was misunderstood because I didn't
word it right. What I meant to ask was can someone point
me to an example of using an audio file with the grain opcode
in csound? In the manual it just says that the table can be
either a sine wave or a sampled sound, and it provides no
further info. Thanks!
Wayne Freno
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From: Josep M Comajuncosas
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Subject: Re: Using ties
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To run legato instruments I usually define the usual instrument and anoth=
er one
activated during all the performance which generates the amplitude (and p=
itch)
envelopes, reinitialised every time the former instrument starts. You can=
do this with
flags via the zak system. The envelope can ve sent to the main instrument=
via the zak
system also.
Tival and tigoto are dangerous as you can easily get "opcode not initiali=
sed" messages.
Jean-Michel DARR=C9MONT wrote:
> David Meckstroth wrote:
> >
> > Would someone be kind and send me a short example (sco & orc) (or poi=
nt me in
> > a direction where I could find one) of how to use ties (tival, tigoto=
,
> > fractional i #s). Thank you in advance. -David
>
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From: jpff@maths.bath.ac.uk
To: sb07@hotmail.com
CC: rasmuse@hem.passagen.se, csound@maths.ex.ac.uk
In-reply-to: <19990303155051.25173.qmail@hotmail.com> (message from Serge _ on
Wed, 03 Mar 1999 07:50:50 PST)
Subject: Re: compiling files under dos
BCC: jpff@maths.bath.ac.uk
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Date: Thu, 4 Mar 99 17:04:31 GMT
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Make sure you do not include aoscilx.c or delay.c as they are old
code, and should not be in the distribution (if they are).
The 64k data problem is real and they is not a great deal one can do.
You could try removing some opcodes, like all the zak system, or all
the phyical models?.
That is the reason we gave up on 16bit implementations. You would do
much better with a 32bit system, even with a Dos extender. My DOS
version is a 32bit version. that also explains teh long constants.
==John ff
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From: David Boothe
To: 'Andre Bartetzki'
Cc: 'Csound'
Subject: RE: bugs in wg opcodes
Date: Thu, 4 Mar 1999 10:48:47 -0600
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Here are some partial answers. Others are still under investigation.
wgbrass apparently has an undocumnted k-rate argument before iatt.
Documentation should be updated before too long.
The argument knum in shaker opcode was eliminated in Csound vers. 3.49, but
I negelcted to remove it form the pdf manual until vers. 3.51. Current pdf
manual has it right.
Sorry for the confusion.
-David.
Andre Bartetzki wrote:
> Does that mean that there is an undocumented argument before
> ifn, for example
> an idetk-time similar to the other wg opcodes?
-snip-
> -----------
> 5. ar shaker kamp,kfreq,kbeans,kdamp,knum,ktimes [,idecay]
>
> A line similar to the example given in the manual results in
>
> error: too many input args, line 13:
> a1 shaker 20000, 400,8, 0.999, 0,100,0
> 1 syntax errors in orchestra. compilation invalid
>
> (see also Javier Ruiz' message some weeks ago)
>
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In-reply-to: <19990303155051.25173.qmail@hotmail.com> (message from Serge _ on
Wed, 03 Mar 1999 07:50:50 PST)
Subject: Re: compiling files under dos
References: <19990303155051.25173.qmail@hotmail.com>
Date: Thu, 4 Mar 99 17:04:31 GMT
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Make sure you do not include aoscilx.c or delay.c as they are old
code, and should not be in the distribution (if they are).
The 64k data problem is real and they is not a great deal one can do.
You could try removing some opcodes, like all the zak system, or all
the phyical models?.
That is the reason we gave up on 16bit implementations. You would do
much better with a 32bit system, even with a Dos extender. My DOS
version is a 32bit version. that also explains teh long constants.
==John ff
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Thu, 4 Mar 1999 13:43:34 +0100
Date: Thu, 4 Mar 1999 13:43:29 +0100 (ROM )
From: Nicola Bernardini
To: Csound Linux/Unix Development Group ,
Csound mailing list
Cc: "J.P. Fitch (csound bath)" ,
antoine@tiger.dhis.org
Subject: [CUD] rtaudio.c modif for linux (fwd)
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Dear friends,
I am forwarding this mail to you all to ask for advice. Antoine Lefebvre
has done a nice job at separating input and output devices for double-
device linux setups in order to be able to use two cards on the same
machine (full-duplex, hence).
I'd like to include the patch at least in the unofficial distribution
(so jpff may have a chance to see if it works at all before including
it in the official one) but I'm undecided on how to handle it, because
all the solutions I can think of have drawbacks.
solution 1: insert names for I/O devices in the configure process,
all defaulting to the same device (when no two-cards
setups are present), as in:
--output-device=DEV (default: dsp)
--input-device=DEV (default: dsp)
so that whatever is set gets compiled in.
Drawbacks: compile-time option
Plusses: should remain command-line compatible, simple
and clean job
solution 2: allow run-time options to carry the name of the device
(without the '/dev' of course - or perhaps with that too?),
as in: csound -o dsp -i dsp1 etc.
(this could be done like this: when csound reads the -[io]
option, it checks whether the given name matches a possible
device: if it does, 'tis the device, otherwise it's a
file)
Drawbacks: pretty ugly if you ask me, and not command-
line compatible (if that has some meaning),
not so simple
Plusses: run-time option, quite convenient when you are
juggling around with cards...
Advice is more than welcome (it is needed!).
ciao ciao
Nicola
P.S. jpff, I don't get any mail from the csound list since Feb.17; are
we all *that* busy? (I certainly did not write...)
------------------------------------------------------------------------
Nicola Bernardini
E-mail: nicb@axnet.it
Re graphics: A picture is worth 10K words -- but only those to describe
the picture. Hardly any sets of 10K words can be adequately described
with pictures.
---------- Forwarded message ----------
Date: Wed, 3 Mar 1999 17:39:33 -0500
From: Antoine Lefebvre
To: Nicola Bernardini
Subject: rtaudio.c modif for linux
Hello,
it is the first time I use CVS, it is a very great tool! So have have what you said me. I include the diff file of my patch. I don't put any notice about my modification because I don't know how you made that habitually.
The modification is very little as you see. The disavantage is that you have to recompile to change de devices choice...I should be a great idea to add a command line option.
The file rtlinux.diff is include
Thanks
--
Antoine Lefebvre
antoinelefebvre@softhome.net
http://pages.infinit.net/linux/music/music.html
--WIyZ46R2i8wDzkSu
Content-Type: TEXT/PLAIN; CHARSET=us-ascii
Content-ID:
Content-Description:
--- rtlinux.c.orig Wed Mar 3 17:33:37 1999
+++ rtlinux.c Wed Mar 3 17:34:10 1999
@@ -10,8 +10,10 @@
#include
#include
-#define DSP_NAME "/dev/dsp"
-static int dspfd;
+#define DSPIN_NAME "/dev/dsp1"
+#define DSPOUT_NAME "/dev/dsp"
+static int dspinfd;
+static int dspoutfd
void setsndparms(int, int, int, float, unsigned);
void setvolume(unsigned);
@@ -45,11 +47,11 @@
oMaxLag = IODACSAMPS; /* use the default value */
/* Jonathan Mohr 1995 Oct 17 */
/* open DSP device for reading */
- if ( (dspfd = open(DSP_NAME, O_RDONLY)) == -1 )
+ if ( (dspinfd = open(DSPIN_NAME, O_RDONLY)) == -1 )
die("unable to open soundcard for audio input");
/* initialize data format, channels, sample rate, and buffer size */
- setsndparms( dspfd, O.informat, nchnls, esr, oMaxLag * O.insampsiz );
+ setsndparms( dspinfd, O.informat, nchnls, esr, oMaxLag * O.insampsiz );
ishift = getshift(dsize);
}
@@ -67,11 +69,11 @@
channels, but allows the user to set the output volume. */
{
/* open DSP device for writing */
- if ( (dspfd = open(DSP_NAME, O_WRONLY)) == -1 )
+ if ( (dspoutfd = open(DSPOUT_NAME, O_WRONLY)) == -1 )
die("unable to open soundcard for audio output");
/* set sample size/format, rate, channels, and DMA buffer size */
- setsndparms( dspfd, O.outformat, nchnls, esr,
+ setsndparms( dspoutfd, O.outformat, nchnls, esr,
O.outbufsamps * O.outsampsiz);
/* check if volume was specified as command line parameter */
@@ -90,7 +92,7 @@
int rtrecord(char *inbuf, int nbytes) /* get samples from ADC */
{
/* J. Mohr 1995 Oct 17 */
- if ( (nbytes = read(dspfd, inbuf, nbytes)) == -1 )
+ if ( (nbytes = read(dspinfd, inbuf, nbytes)) == -1 )
die("error while reading DSP device for audio input");
return(nbytes);
}
@@ -110,7 +112,7 @@
{
long sampframes = nbytes >> oshift;
/* J. Mohr 1995 Oct 17 */
- if (write(dspfd, outbuf, nbytes) < nbytes)
+ if (write(dspoutfd, outbuf, nbytes) < nbytes)
printf("/dev/dsp: couldn't write all bytes requested\n");
nrecs++;
}
@@ -118,7 +120,9 @@
void rtclose(void) /* close the I/O device entirely */
{ /* called only when both complete */
/* J. Mohr 1995 Oct 17 */
- if (close(dspfd) == -1)
+ if (close(dspinfd) == -1)
+ die("unable to close DSP device");
+ if (close(dspoutfd) == -1)
die("unable to close DSP device");
if (O.Linein) {
#ifdef PIPES
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Thu, 4 Mar 1999 13:43:34 +0100
Date: Thu, 4 Mar 1999 13:43:29 +0100 (ROM )
From: Nicola Bernardini
To: Csound Linux/Unix Development Group ,
Csound mailing list
cc: "J.P. Fitch (csound bath)" ,
antoine@tiger.dhis.org
Subject: rtaudio.c modif for linux (fwd)
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Dear friends,
I am forwarding this mail to you all to ask for advice. Antoine Lefebvre
has done a nice job at separating input and output devices for double-
device linux setups in order to be able to use two cards on the same
machine (full-duplex, hence).
I'd like to include the patch at least in the unofficial distribution
(so jpff may have a chance to see if it works at all before including
it in the official one) but I'm undecided on how to handle it, because
all the solutions I can think of have drawbacks.
solution 1: insert names for I/O devices in the configure process,
all defaulting to the same device (when no two-cards
setups are present), as in:
--output-device=DEV (default: dsp)
--input-device=DEV (default: dsp)
so that whatever is set gets compiled in.
Drawbacks: compile-time option
Plusses: should remain command-line compatible, simple
and clean job
solution 2: allow run-time options to carry the name of the device
(without the '/dev' of course - or perhaps with that too?),
as in: csound -o dsp -i dsp1 etc.
(this could be done like this: when csound reads the -[io]
option, it checks whether the given name matches a possible
device: if it does, 'tis the device, otherwise it's a
file)
Drawbacks: pretty ugly if you ask me, and not command-
line compatible (if that has some meaning),
not so simple
Plusses: run-time option, quite convenient when you are
juggling around with cards...
Advice is more than welcome (it is needed!).
ciao ciao
Nicola
P.S. jpff, I don't get any mail from the csound list since Feb.17; are
we all *that* busy? (I certainly did not write...)
------------------------------------------------------------------------
Nicola Bernardini
E-mail: nicb@axnet.it
Re graphics: A picture is worth 10K words -- but only those to describe
the picture. Hardly any sets of 10K words can be adequately described
with pictures.
---------- Forwarded message ----------
Date: Wed, 3 Mar 1999 17:39:33 -0500
From: Antoine Lefebvre
To: Nicola Bernardini
Subject: rtaudio.c modif for linux
Hello,
it is the first time I use CVS, it is a very great tool! So have have what you said me. I include the diff file of my patch. I don't put any notice about my modification because I don't know how you made that habitually.
The modification is very little as you see. The disavantage is that you have to recompile to change de devices choice...I should be a great idea to add a command line option.
The file rtlinux.diff is include
Thanks
--
Antoine Lefebvre
antoinelefebvre@softhome.net
http://pages.infinit.net/linux/music/music.html
--WIyZ46R2i8wDzkSu
Content-Type: TEXT/PLAIN; CHARSET=us-ascii
Content-ID:
Content-Description:
--- rtlinux.c.orig Wed Mar 3 17:33:37 1999
+++ rtlinux.c Wed Mar 3 17:34:10 1999
@@ -10,8 +10,10 @@
#include
#include
-#define DSP_NAME "/dev/dsp"
-static int dspfd;
+#define DSPIN_NAME "/dev/dsp1"
+#define DSPOUT_NAME "/dev/dsp"
+static int dspinfd;
+static int dspoutfd
void setsndparms(int, int, int, float, unsigned);
void setvolume(unsigned);
@@ -45,11 +47,11 @@
oMaxLag = IODACSAMPS; /* use the default value */
/* Jonathan Mohr 1995 Oct 17 */
/* open DSP device for reading */
- if ( (dspfd = open(DSP_NAME, O_RDONLY)) == -1 )
+ if ( (dspinfd = open(DSPIN_NAME, O_RDONLY)) == -1 )
die("unable to open soundcard for audio input");
/* initialize data format, channels, sample rate, and buffer size */
- setsndparms( dspfd, O.informat, nchnls, esr, oMaxLag * O.insampsiz );
+ setsndparms( dspinfd, O.informat, nchnls, esr, oMaxLag * O.insampsiz );
ishift = getshift(dsize);
}
@@ -67,11 +69,11 @@
channels, but allows the user to set the output volume. */
{
/* open DSP device for writing */
- if ( (dspfd = open(DSP_NAME, O_WRONLY)) == -1 )
+ if ( (dspoutfd = open(DSPOUT_NAME, O_WRONLY)) == -1 )
die("unable to open soundcard for audio output");
/* set sample size/format, rate, channels, and DMA buffer size */
- setsndparms( dspfd, O.outformat, nchnls, esr,
+ setsndparms( dspoutfd, O.outformat, nchnls, esr,
O.outbufsamps * O.outsampsiz);
/* check if volume was specified as command line parameter */
@@ -90,7 +92,7 @@
int rtrecord(char *inbuf, int nbytes) /* get samples from ADC */
{
/* J. Mohr 1995 Oct 17 */
- if ( (nbytes = read(dspfd, inbuf, nbytes)) == -1 )
+ if ( (nbytes = read(dspinfd, inbuf, nbytes)) == -1 )
die("error while reading DSP device for audio input");
return(nbytes);
}
@@ -110,7 +112,7 @@
{
long sampframes = nbytes >> oshift;
/* J. Mohr 1995 Oct 17 */
- if (write(dspfd, outbuf, nbytes) < nbytes)
+ if (write(dspoutfd, outbuf, nbytes) < nbytes)
printf("/dev/dsp: couldn't write all bytes requested\n");
nrecs++;
}
@@ -118,7 +120,9 @@
void rtclose(void) /* close the I/O device entirely */
{ /* called only when both complete */
/* J. Mohr 1995 Oct 17 */
- if (close(dspfd) == -1)
+ if (close(dspinfd) == -1)
+ die("unable to close DSP device");
+ if (close(dspoutfd) == -1)
die("unable to close DSP device");
if (O.Linein) {
#ifdef PIPES
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Date: Thu, 04 Mar 1999 09:45:15 -0800
From: Charles Baker
Reply-To: baker@charlieb.com
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To: Wayne Freno
CC: Csound List
Subject: Re: Grain opcode & audio files
References:
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Wayne Freno wrote:
> I guess my last post was misunderstood because I didn't
> word it right. What I meant to ask was can someone point
> me to an example of using an audio file with the grain opcode
> in csound? In the manual it just says that the table can be
> either a sine wave or a sampled sound, and it provides no
> further info. Thanks!
>
> Wayne Freno
To use a 'sampled sound', you read the soundfile into a large function
table, right?
Gen01?
Unless I'm crazy (which some will assure you is the fact...;-) )
CharlieB
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Date: Thu, 04 Mar 1999 18:05:39 +0000
From: Richard Dobson
Organization: Composers Desktop Project
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To: "csound@maths.ex.ac.uk"
Subject: Re: Using ties
References: <36DE232A.C112423E@concentric.net> <36DEA137.57B9CE11@club-internet.fr> <36DEA49F.52C5F07C@intercom.es>
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That will happen with all the ~igoto commands if you jump over anything
other than other init statements (which is what they are there for).
Used as intended, they are not dangerous at all!
Richard Dobson
Josep M Comajuncosas wrote:
>
> To run legato instruments I usually define the usual instrument and another one
> activated during all the performance which generates the amplitude (and pitch)
> envelopes, reinitialised every time the former instrument starts. You can do this with
> flags via the zak system. The envelope can ve sent to the main instrument via the zak
> system also.
> Tival and tigoto are dangerous as you can easily get "opcode not initialised" messages.
>
--
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, 04 Mar 1999 09:27:43 -0800
To: David Boothe , 'Csound'
From: Ben McAllister
Subject: Re: Csound on NT
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Hi,
I've been using Csound on NT intermittently for about 6 months, and it's
running smooth.
Ben
At 08:50 AM 3/4/99 -0600, David Boothe wrote:
>I'm thinking of moving to a Windows NT machine. Does anyone use Csound under
>NT and does it work OK?
>
>Thanks in advance.
>
>-David.
>
>
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From: Serge _
To: jpff@maths.bath.ac.uk
Cc: rasmuse@hem.passagen.se, csound@maths.ex.ac.uk
Subject: Re: compiling files under dos
Date: Thu, 04 Mar 1999 10:59:59 PST
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Hi John,
Ok, i give up for the dos mode, but even in C++Builder 3.0 from Borland,
it doesn't work, i have so many errors and warnings!
Don't you have any makefile for this compiler?
Thank you
Serge Bland
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From: Serge _
To: jpff@maths.bath.ac.uk
Cc: rasmuse@hem.passagen.se, csound@maths.ex.ac.uk
Subject: Re: compiling files under dos
Date: Thu, 04 Mar 1999 10:59:59 PST
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Hi John,
Ok, i give up for the dos mode, but even in C++Builder 3.0 from Borland,=20
it doesn't work, i have so many errors and warnings!
Don't you have any makefile for this compiler?
Thank you
Serge B=E9land
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From: Paul
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Subject: benchmarks
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For those who care about such things, I have updated my csound
benchmarks page at:
http://members.tripod.com/~slinkP/pw_linux/csbench.html
The main update is that I have added a section on realtime
benchmarking... I've got an orc and sco that should be pretty portable.
Whether this benchmark really tells you anything useful is anyone's
guess. But it fulfilled its primary design goal ... procrastination!
Oh yeah, I think it sounds kind of interesting too. YMMV.
--PW
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From: Tony Ledford
To: 'David Boothe' , 'Csound'
Subject: RE: Csound on NT
Date: Thu, 4 Mar 1999 19:15:01 -0500
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Works fine, David. I've used it with SP3 and SP4 without any issues.
Tony
> -----Original Message-----
> From: owner-csound-outgoing@maths.ex.ac.uk
> [mailto:owner-csound-outgoing@maths.ex.ac.uk]On Behalf Of David Boothe
> Sent: Thursday, March 04, 1999 9:50 AM
> To: 'Csound'
> Subject: Csound on NT
>
>
> I'm thinking of moving to a Windows NT machine. Does anyone
> use Csound under
> NT and does it work OK?
>
> Thanks in advance.
>
> -David.
>
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From: Gabriel Maldonado
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To: 'Csound'
Subject: Re: Csound on NT
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DirectCsound works OK in deferred time, but maybe it has some problems with realtime.
Gab
David Boothe wrote:
>
> I'm thinking of moving to a Windows NT machine. Does anyone use Csound under
> NT and does it work OK?
>
> Thanks in advance.
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