| Hello everyone !
i'm looking for a software that could allow me to draw graphically
enveloppes.
Do you have any hint ?
Thank you.
Raskizo
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Date: 29 Oct 1999 14:33:42 -0000
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To: csound@maths.ex.ac.uk
Subject: Re: Compressor/Limiter
From: =?ISO-8859-2?Q?Varga_Istv=E1n?=
"Michael A. Thompson" wrote:
> Looking for a good compressor/limiter csound .orc. Anybody messed around
> with doing one?
What sort of compressor/limiter do you want ? For compression, I have some
orc files (and one of the drum instruments also uses compression); for removing
peaks (this is not really compression), I use this:
http://www.geocities.com/SiliconValley/Bit/5683/maximizer-1.0.zip
Note that this program is really for removing large peaks (mostly caused by
bass instruments), it cannot change the average volume. It is also
written in C, so you may not like it. As of compressor orcs, you can find them at
http://www.geocities.com/SiliconValley/Bit/5683/compress.zip
Some simple test files:
http://www.geocities.com/SiliconValley/Bit/5683/testfiles.zip
(It eats up lots of space on the server, so will be removed soon)
Files in the archive are:
test1.mp2: original signal, scaled to 32000
test2.mp2: compressed with Limit412.orc (see below)
test3.mp2: processed with "maximizer", command-line flags:
-verbose -smode 0 -lmode 0 -limit 14000 \
-clpl1 14000 -clpl2 15625 -nband 9 -ovrlp 4 \
-freq2 1000 -scale 2.048 -r_lim a 1.0 \
-clpsn 0 4.0 -clpsn 1 2.0 -clpsn 2 1.4 \
-clpsn 3 1.0 -clpsn 4 0.7 -clpsn 5 1.0 \
-clpsn 6 1.2 -clpsn 7 1.4 -clpsn 8 2.0 \
-clpsn 9 4.0
(this may need some more experimenting)
test4.mp2: simply clipped (see "clip.orc")
-------- Limit412.orc --------
sr = 44100
kr = 44100
ksmps = 1
nchnls = 2
instr 1
iamp = 2.048
imax1 = 14000
idel1 = 20
idec1 = 0.01
iatt1 = 0.0001
ifrq1 = 25
imax2 = 14000
idel2 = 2.5
idec2 = 0.5
iatt2 = 0.0025
ifrq2 = 200
a1,a2 soundin "mix.sf"
idel1 = idel1*0.001
idel2 = idel2*0.001
klv01 init 0
klv02 init 0
ad1 delay a1,idel1
ad2 delay a2,idel1
klx1 downsamp abs(a1)
klx2 downsamp abs(a2)
klx = (klx1>klx2 ? klx1:klx2)
klv01 = (klx>klv01 ? klv01+(klx-klv01)*idec1:klv01+(klx-klv01)*iatt1)
al1 upsamp klv01
al1 butterlp al1,ifrq1
klv01x downsamp al1
a1 = (klv01xklx2 ? klx1:klx2)
klv02 = (klx>klv02 ? klv02+(klx-klv02)*idec2:klv02+(klx-klv02)*iatt2)
al1 upsamp klv02
al1 butterlp al1,ifrq2
klv02x downsamp al1
ad1 = (klv02x; Fri, 29 Oct 1999 08:40:14 -0700
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Date: Fri, 29 Oct 1999 08:49:12 -0700
To: csound@maths.ex.ac.uk
From: Ben McAllister
Subject: Win NT Problems take 2
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Hi,
Well, there wasn't much concrete feedback there on the NT issue. Some
folks HAVE experienced what I have, and some haven't.
For the folks who have Winsound running smoothly on their machines: Could
we get an idea of your system attributes?
To whomever codes the Winsound app: what were the testing conditions for
the app? Win 95? I also have had difficulties under Win 98, and wrote to
the list about those last Oct or Nov. I'm a programmer, though not
familiar with the subtle differences between Win 95 and NT. I assume this
could be part of the problem. Also: are there known conflicts between
Winsound and other programs? Thanks.
Ben Mcallister
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To: csound@lists.bath.AC.UK
Subject: Re: Compressor/Limiter
From: =?ISO-8859-2?Q?Varga_Istv=E1n?=
"Michael A. Thompson" wrote:
> Looking for a good compressor/limiter csound .orc. Anybody messed around
> with doing one?
What sort of compressor/limiter do you want ? For compression, I have some
orc files (and one of the drum instruments also uses compression); for removing
peaks (this is not really compression), I use this:
http://www.geocities.com/SiliconValley/Bit/5683/maximizer-1.0.zip
Note that this program is really for removing large peaks (mostly caused by
bass instruments), it cannot change the average volume. It is also
written in C, so you may not like it. As of compressor orcs, you can find them at
http://www.geocities.com/SiliconValley/Bit/5683/compress.zip
Some simple test files:
http://www.geocities.com/SiliconValley/Bit/5683/testfiles.zip
(It eats up lots of space on the server, so will be removed soon)
Files in the archive are:
test1.mp2: original signal, scaled to 32000
test2.mp2: compressed with Limit412.orc (see below)
test3.mp2: processed with "maximizer", command-line flags:
-verbose -smode 0 -lmode 0 -limit 14000 \
-clpl1 14000 -clpl2 15625 -nband 9 -ovrlp 4 \
-freq2 1000 -scale 2.048 -r_lim a 1.0 \
-clpsn 0 4.0 -clpsn 1 2.0 -clpsn 2 1.4 \
-clpsn 3 1.0 -clpsn 4 0.7 -clpsn 5 1.0 \
-clpsn 6 1.2 -clpsn 7 1.4 -clpsn 8 2.0 \
-clpsn 9 4.0
(this may need some more experimenting)
test4.mp2: simply clipped (see "clip.orc")
-------- Limit412.orc --------
sr = 44100
kr = 44100
ksmps = 1
nchnls = 2
instr 1
iamp = 2.048
imax1 = 14000
idel1 = 20
idec1 = 0.01
iatt1 = 0.0001
ifrq1 = 25
imax2 = 14000
idel2 = 2.5
idec2 = 0.5
iatt2 = 0.0025
ifrq2 = 200
a1,a2 soundin "mix.sf"
idel1 = idel1*0.001
idel2 = idel2*0.001
klv01 init 0
klv02 init 0
ad1 delay a1,idel1
ad2 delay a2,idel1
klx1 downsamp abs(a1)
klx2 downsamp abs(a2)
klx = (klx1>klx2 ? klx1:klx2)
klv01 = (klx>klv01 ? klv01+(klx-klv01)*idec1:klv01+(klx-klv01)*iatt1)
al1 upsamp klv01
al1 butterlp al1,ifrq1
klv01x downsamp al1
a1 = (klv01xklx2 ? klx1:klx2)
klv02 = (klx>klv02 ? klv02+(klx-klv02)*idec2:klv02+(klx-klv02)*iatt2)
al1 upsamp klv02
al1 butterlp al1,ifrq2
klv02x downsamp al1
ad1 = (klv02x
Date: Fri, 29 Oct 1999 19:55:35 +0100
From: Richard Dobson
Organization: Composers Desktop Project
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To: csound@maths.ex.ac.uk
Subject: Re: Win NT Problems take 2
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I have, from time to time, fiddled a bit with the Winsound code (mainly
the playback code), as I have hitherto been running NT4 (with a mere
SoundBlaster 16!), and there were a few pertinent differences from Win95
(on which I also tested - dual-boot setup). I am now running
Windows2000 beta.
At the moment I am still at Csound v3.55 - here I don't find any
problems such as you describe - both 'play at end' and 'remain active'
seem to be working. I have only just moved house, and am still not fully
set up, so it will be a day or two before I can start up with the most
recent version.
The main differences audio-wise are the very different soundcard driver
models - a plain-vanilla MME driver under Win95, and a WDM driver under
Win2K (with vastly lower latency, BTW).
Richard Dobson
Ben McAllister wrote:
>
> Hi,
>
> Well, there wasn't much concrete feedback there on the NT issue. Some
> folks HAVE experienced what I have, and some haven't.
>
> For the folks who have Winsound running smoothly on their machines: Could
> we get an idea of your system attributes?
>
> To whomever codes the Winsound app: what were the testing conditions for
> the app? Win 95? I also have had difficulties under Win 98, and wrote to
> the list about those last Oct or Nov. I'm a programmer, though not
> familiar with the subtle differences between Win 95 and NT. I assume this
> could be part of the problem. Also: are there known conflicts between
> Winsound and other programs? Thanks.
>
> Ben Mcallister
--
Test your DAW with my Soundcard Attrition Page!
http://wkweb5.cableinet.co.uk/rwd (LU: 23rd August 1999)
CDP: http://www.bath.ac.uk/~masjpf/CDP/CDP.htm (LU: 14th June 1999)
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Date: Fri, 29 Oct 1999 13:47:41 -0400
From: Paul Winkler
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To: "Michael A. Thompson"
CC: "csound@maths.ex.ac.uk"
Subject: Re: Compressor/Limiter
References: <37EE97C4.F269A370@jove.acs.unt.edu>
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"Michael A. Thompson" wrote:
>
> Looking for a good compressor/limiter csound .orc. Anybody messed around
> with doing one?
Funny you should ask, I've been working on this lately.
csound has a "dam" opcode that's supposed to be a general-purpose
dynamics processor, but I can't get it to sound good.
Here's where I left off on my compressor instr. THe big difficulty
seems to be implementing "attack" and "decay" in ways that work like
an analog compressor, since it's rarely desirable to have a
compressor that works instantaneously all the time. This version
sort-of does the trick.
Next step is to add a threshold level and different ratios above/
below that level. Not sure how to do that yet. I had some good
exchanges with Paul Barton-Davis where we tried to figure out how
that worked; I think his idea of using an ftable to plot input level
vs. output gain might be the best approach, since it allows multiple
threshold levels and arbitrary dynamics curves.
instr 14 ; do everything in dB.
; Attack/release added using my by-hand linear slope
method.
; Let's see if I can make this work right with ratio (that
is,
; ratio of 1 "turns off" attack/release controls...)
; OK, this is getting there....
; EXTRACT & COMPRESS ENVELOPE
iratio = 1 / p4 ; p4 is "compression ratio" -- 4 means 4:1
ihp = 10 ; lopass filter freq for rms and gain
kenv expseg 1, p3 - .1, 32767, .1, 32767
kdclick linseg 0, .1, 1, p3 -.1, 1
asig, asig2 diskin "arms/salt_notice/nines.wav", 1 ; , 217
avg = (asig + asig2) * .5
krms_raw rms avg, ihp, 0
kdb_raw = 20 * log10(krms_raw + 1) ; + 1 to avoid divide by 0
kdb_comp = kdb_raw * iratio
ioffset = 72 - (72 * iratio) ; I thought it should be 90 d
; but that doesn't work right at high
; ratios.
; 70 catches peaks pretty well.
kdb_comp = kdb_comp + ioffset ; make-up gain.
kdiff = kdb_comp - kdb_raw ; difference
; SLOPE CONTROL (attack & release)
iattack = p6 ; MAX time for level to RISE 60 dB
irelease = p7 ; MIN time for level to DROP 60 dB
; Now we translate that into maximum diff. between
successive
; values of k-rate DIFFERENCE
imaxattd = (90 * iratio) / (kr * iattack ); "max. attack difference"
imaxreld = (90 * iratio) / (kr * irelease)
print iattack, imaxattd
print irelease, imaxreld
kdiff_old init 88
if (kdiff > kdiff_old) goto RELEASE
ATTACK:
; khalftime = iattack ; fast attack
; ; ; this works???
goto OUTPUT
RELEASE:
; This sort-of works...
if (kdiff_old + imaxreld) > kdiff goto OUTPUT
kdiff = kdiff_old + imaxreld
OUTPUT:
kdiff_old = kdiff ; store for next cycle
; FINAL OUTPUT GAIN CONTROL ...
kdb_comp = kdb_raw + kdiff ; construct final target gain
krms_comp = 10 ^ (kdb_comp / 20) ; convert back to rms.
aout1 gain asig, krms_comp, ihp, 1 ; set gain to match RMS
aout2 gain asig2, krms_comp, ihp, 1
; asig, rms, filter hz, initial (0=clear, 1 prevents clicks)
imakeup = p5 ; NOT USED
outs aout1, aout2
endin
;;;;;;;;;; here's a test sco.
; First uncompressed, then compressed with fast attack & release,
; then fast attack / slow release.
f1 0 1024 10 1 ; sine
;14 at dur ratio gain attack release
i14 1 6 1 1 .000001 .0001
i14 7.5 . 8 1 .000001 .00000001
i14 14 . 8 2 .000001 .5
---------------- paul winkler ------------------
slinkP arts: music, sound, illustration, design, etc.
email=========================slinkp AT ulster DOT net
ARMS online=======================http://reacharms.com
personal page===========http://www.ulster.net/~abigoo/
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Date: Fri, 29 Oct 1999 16:19:00 +0000
From: luis jure
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well, i seem to be back in the list again. here's a message i tried to send
twice before, unsuccessfully:
just for learning purposes i've been trying to implement the most basic
types of filters with instruments using just delay lines. everything went
fine with non-recursive filters, but with IIRs i don't get the expected
results.
below is an instrument that i think should produce the following:
y(n) = a*x(n) + b*y(n-1)
; orc =======================================
sr = 44100
kr = 44100
ksmps = 1
instr 1
aout init 0 ; y(n)
a0 rand 20000 ; x(n)
a1 = a0 * .5 ; a * x(n)
a2 delay1 aout ; y(n-1)
a3 = a2 * .5 ; b * y(n-1)
aout = a1 + a3 ; y(n)
out aout
endin
; sco =======================================
i1 0 3
e
it looks to me as if the instrument were right; the problem is that i don't
get a low-pass filter as i should according to the textbook, but rather a
band-reject filter with center freq at sr/4 and a very wide bw. conversely,
with aout = a1 - a3 i get a band-pass filter of similar characteristics
instead of a high-pass. could somebody please point me what i am doing
wrong?
thank you
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Date: Fri, 29 Oct 1999 16:22:33 +0000
From: luis jure
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Subject: feedback FM (with a long delay)
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another old message i had waiting:
from hans mikelson long ago, in response to a question about feedback
fm:
> a1 init 0
>
> aenv adsr iatt, idec, islev, irel
>
> a1 oscil aenv, ifqc*(1+a1), 1
>
> outs a1, a1
>
> endin
>
> The pitch tends to drift somewhat though I think.
>
actually for real feedback fm you need something a bit more complicated
than that, where you can control the modulation index. below is a simple
and naively commented orchestra
; ============================================================
; fdbckfm.orc
sr = 44100
kr = 44100
ksmps = 1
instr 1 ; feedback fm
iamp = p4 ; max amplitude
ifc = p5 ; center frq
indx = p6 ; max. modulation index
afrq init ifc ; initialize oscil frq to center frq
a1 oscil 1,afrq,1 ; oscillator, normalized to 1
kndx line 0,p3,indx ; vary modulation index
afm = afrq * a1 ; calculate modulation frq
adev = afm * kndx ; calculate deviation frq
afrq = ifc + adev ; calculate oscil frq
kamp linseg 0, .1, iamp, p3-.2, iamp, .1, 0
aout = a1 * kamp
out aout
endin
; ============================================================
; fdbckfm.sco
f1 0 8192 10 1
i1 0 3 25000 300 .9
i1 3.2 4 25000 300 2
e
; ============================================================
it's very important to note that the range of usable modulation indexes
in feedback fm is significantly lower than in simple fm, generally I <
1.
in my orchestra pitch tends to raise when I gets close to 1, and for I >
1.2 aprox the system becomes unstable, as you can see in the second
note. i'm not sure if that's something inherent to the technique, or
maybe it's a problem with this particular implementation.
i would appreciate some comments.
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Date: Fri, 29 Oct 1999 12:54:21 -0700
From: Sean Costello
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To: luis jure
CC: csound list
Subject: Re: feedback FM (with a long delay)
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luis jure wrote:
> in my orchestra pitch tends to raise when I gets close to 1, and for I >
> 1.2 aprox the system becomes unstable, as you can see in the second
> note. i'm not sure if that's something inherent to the technique, or
> maybe it's a problem with this particular implementation.
Most of the cool techniques available with feedback FM in the Yamaha FM
synthesizers are not truly based on frequency modulation, but rather on
Phase Modulation. Probably the best way of doing this in Csound is with
a phasor/tablei combination.
Sean Costello
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Date: Fri, 29 Oct 1999 15:28:54 -0700 (PDT)
From: Richard Weisgerber
Message-Id: <199910292228.PAA08348@emerald.oz.net>
To: csound@maths.ex.ac.uk
Reply-To: madgello@oz.net
Subject: Re: Win NT Problems take 2
I tried this with Winsound on NT 4 SP5 wks with following results.
Used tried and true Trapped in Convert composition by Richard Boulanger for
this test.
Remain Active & Play at end unchecked:
Compiled piece and put press a key to close dialog, pressed enter and Winsound
closed. Played results in wave player, success.
Remain Active & Play at end checked:
Compiled piece but DID NOT play the piece when finished. Bug it
Click on logging window, hit a key and up pops the Winsound control dialog.
Played results in wave player, success.
Remain Active checked & Play at end unchecked:
Compile piece, press a key and winsound closed,
Played results in wave player, success.
The only place I noticed weirdness:
Remain Active unchecked & Play at end checked:
Compiled piece, Winsound started playing results, success.
Was about to hit a key to stop playback when I noticed Winsound logging window
title bar says: "Playing.... any character to con(some weird characters)" Bug
it.
DPW 266i w/ ESS 1888 onboard, 128MB RAM, 2x 4GB drives w/ plenty of space.
Clean US NT 4 SP5 install with very few apps installed yet as this is at new
station for day job.
I did have Winsound lock completely when I used (not at home, don't have name
of writer, but is the person that's been putting the drums, strings etc. up
lately) String_11.orc/sco with Remain Active and Play at end checked. BTW, it
says Winsound 3.58 on this version fresh download from
ftp.maths.bath.ac.uk/pub/dream/newest today.
rich out
on with the Csound QA Test crew. ;-)
>From: Ben McAllister
>Hi,
>
> Well, there wasn't much concrete feedback there on the NT issue. Some
>folks HAVE experienced what I have, and some haven't.
>
> For the folks who have Winsound running smoothly on their machines:
Could
>we get an idea of your system attributes?
>
> To whomever codes the Winsound app: what were the testing conditions
for
>the app? Win 95? I also have had difficulties under Win 98, and wrote to
>the list about those last Oct or Nov. I'm a programmer, though not
>familiar with the subtle differences between Win 95 and NT. I assume this
>could be part of the problem. Also: are there known conflicts between
>Winsound and other programs? Thanks.
>
>Ben Mcallister
>
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Date: Fri, 29 Oct 1999 19:08:43 -0400
From: Kirsh Family
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I believe the term "x-class" comes from the fact that these ugens can
produce i-rate, k-rate, or a-rate data without having to have a separate
name (with appropriate beginning letter) for each rate. This notation
is used elsewhere in the manual for the naming of formal arguments which
can take values at any of the three rates; e.g., the "xamp" and "xcps"
arguments of "oscil".
David Kirsh
David Boothe wrote:
>
> ...
> BTW, where does the term "x-class noise generators" come from?
>
> -David.
>
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From: Michael Gogins
To: luis jure ,
csound list
Subject: RE: IIR
Date: Fri, 29 Oct 1999 21:30:38 -0400
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Importance: Normal
IIRs will have to have ksmps = 1, presumably you are already doing that?
-----Original Message-----
From: lj@courier01.adinet.com.uy [mailto:lj@courier01.adinet.com.uy]On
Behalf Of luis jure
Sent: Friday, October 29, 1999 12:19 PM
To: csound list
Subject: IIR
well, i seem to be back in the list again. here's a message i tried to send
twice before, unsuccessfully:
just for learning purposes i've been trying to implement the most basic
types of filters with instruments using just delay lines. everything went
fine with non-recursive filters, but with IIRs i don't get the expected
results.
below is an instrument that i think should produce the following:
y(n) = a*x(n) + b*y(n-1)
; orc =======================================
sr = 44100
kr = 44100
ksmps = 1
instr 1
aout init 0 ; y(n)
a0 rand 20000 ; x(n)
a1 = a0 * .5 ; a * x(n)
a2 delay1 aout ; y(n-1)
a3 = a2 * .5 ; b * y(n-1)
aout = a1 + a3 ; y(n)
out aout
endin
; sco =======================================
i1 0 3
e
it looks to me as if the instrument were right; the problem is that i don't
get a low-pass filter as i should according to the textbook, but rather a
band-reject filter with center freq at sr/4 and a very wide bw. conversely,
with aout = a1 - a3 i get a band-pass filter of similar characteristics
instead of a high-pass. could somebody please point me what i am doing
wrong?
thank you
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From: JM&M DARREMONT
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A nice one for that is Cecilia.
http://mitpress.mit.edu/e-books/csound/frontpage.html
Then click on "Front end" menu.
ras kizo a crit:
> i'm looking for a software that could allow me to draw graphically
> enveloppes.
--
Jean-Michel DARREMONT
|