| It will be useful only if the audio cards are hardware-synchronized.
Gab
Ricardo MadGello wrote:
>
> I just wish that it was possible to talk to an arbitrary number of Windows
> WAVE and/or DirectX I/O devices for RT purposes.
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From: David Boothe
To: 'Csound'
Subject: Minor Revision to pdf Manual
Date: Wed, 24 Mar 1999 10:24:30 -0600
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Csounders-
I have replaced the Manual files on the server. If you downloaded the Manual
before about 15:45 UT today (March 24), there are about ten page numbers in
the Finder (index) which are wrong. The new Finder (three pages total) is
available separately, also.
Sorry for any inconvenience this causes.
-David.
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Date: Wed, 24 Mar 1999 08:48:32 -0800
From: Sean Costello
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jpff@maths.bath.ac.uk wrote:
> Following the number of people who have pointed out that winsound
> takes the audio file name to be what is said, and does not add .wav or
> .aif to teh name, do you'll think that it shoudl add suffix if one is
> not given explicitely? Woudl be easy enough to do but I worry that it
> will then be hard to create an audio file without a suffix.
When would it be useful to have an audio file without a suffix? Even if
you are using the generated result as a control file (i.e. to fill a
table), it still makes sense to have a suffix. You can always remove
the suffix in Windows Explorer.
On the other hand, how hard is it to type .wav or .aiff on the end of a
file? I would fear that having an automatic suffix would produce strange
results, such as .aiff files being renamed .wav, or files having names
like composition.wav.wav (I've had enough random things like this happen
in Winsound, using involving a mismatch between file suffix and the true
file format). Also, just as you can remove the suffix in Windows
Explorer, you can always add one there, too.
Sean Costello
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Date: Wed, 24 Mar 1999 08:48:32 -0800
From: Sean Costello
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jpff@maths.bath.ac.uk wrote:
> Following the number of people who have pointed out that winsound
> takes the audio file name to be what is said, and does not add .wav or
> .aif to teh name, do you'll think that it shoudl add suffix if one is
> not given explicitely? Woudl be easy enough to do but I worry that it
> will then be hard to create an audio file without a suffix.
When would it be useful to have an audio file without a suffix? Even if
you are using the generated result as a control file (i.e. to fill a
table), it still makes sense to have a suffix. You can always remove
the suffix in Windows Explorer.
On the other hand, how hard is it to type .wav or .aiff on the end of a
file? I would fear that having an automatic suffix would produce strange
results, such as .aiff files being renamed .wav, or files having names
like composition.wav.wav (I've had enough random things like this happen
in Winsound, using involving a mismatch between file suffix and the true
file format). Also, just as you can remove the suffix in Windows
Explorer, you can always add one there, too.
Sean Costello
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From: David Boothe
To: 'Sean Costello' , jpff@maths.bath.ac.uk
Cc: csound@maths.ex.ac.uk
Subject: RE: Winsound Extensions (was: C sound/No sound)
Date: Wed, 24 Mar 1999 11:10:08 -0600
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My vote is to leave things the way they are. The first time this happened to
me, it took all of 30 seconds to figure out it was pilot error, and that I
am responsible for the extension. (The next time it happened, I was able to
get that figure down to about 15 seconds!)
-David.
Sean Costello wrote
> On the other hand, how hard is it to type .wav or .aiff on
> the end of a
> file? I would fear that having an automatic suffix would
> produce strange
> results, such as .aiff files being renamed .wav, or files having names
> like composition.wav.wav (I've had enough random things like
> this happen
> in Winsound, using involving a mismatch between file suffix
> and the true
> file format). Also, just as you can remove the suffix in Windows
> Explorer, you can always add one there, too.
>
> Sean Costello
>
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RE: Winsound Extensions (was: C sound/No sound)
My vote is to leave things the way they are. The =
first time this happened to me, it took all of 30 seconds to figure out =
it was pilot error, and that I am responsible for the extension. (The =
next time it happened, I was able to get that figure down to about 15 =
seconds!)
-David.
Sean Costello wrote
> On the other hand, how hard is it to type .wav =
or .aiff on
> the end of a
> file? I would fear that having an automatic =
suffix would
> produce strange
> results, such as .aiff files being renamed =
.wav, or files having names
> like composition.wav.wav (I've had enough =
random things like
> this happen
> in Winsound, using involving a mismatch between =
file suffix
> and the true
> file format). Also, just as you can remove the =
suffix in Windows
> Explorer, you can always add one there, =
too.
>
> Sean Costello
>
------_=_NextPart_001_01BE7619.2C3EE272--
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From: David Boothe
To: 'Sean Costello' , jpff@maths.bath.ac.uk
Cc: csound@maths.ex.ac.uk
Subject: RE: Winsound Extensions (was: C sound/No sound)
Date: Wed, 24 Mar 1999 11:10:08 -0600
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My vote is to leave things the way they are. The first time this happened to
me, it took all of 30 seconds to figure out it was pilot error, and that I
am responsible for the extension. (The next time it happened, I was able to
get that figure down to about 15 seconds!)
-David.
Sean Costello wrote
> On the other hand, how hard is it to type .wav or .aiff on
> the end of a
> file? I would fear that having an automatic suffix would
> produce strange
> results, such as .aiff files being renamed .wav, or files having names
> like composition.wav.wav (I've had enough random things like
> this happen
> in Winsound, using involving a mismatch between file suffix
> and the true
> file format). Also, just as you can remove the suffix in Windows
> Explorer, you can always add one there, too.
>
> Sean Costello
>
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RE: Winsound Extensions (was: C sound/No sound)
My vote is to leave things the way they are. The =
first time this happened to me, it took all of 30 seconds to figure out =
it was pilot error, and that I am responsible for the extension. (The =
next time it happened, I was able to get that figure down to about 15 =
seconds!)
-David.
Sean Costello wrote
> On the other hand, how hard is it to type .wav =
or .aiff on
> the end of a
> file? I would fear that having an automatic =
suffix would
> produce strange
> results, such as .aiff files being renamed =
.wav, or files having names
> like composition.wav.wav (I've had enough =
random things like
> this happen
> in Winsound, using involving a mismatch between =
file suffix
> and the true
> file format). Also, just as you can remove the =
suffix in Windows
> Explorer, you can always add one there, =
too.
>
> Sean Costello
>
------_=_NextPart_001_01BE7619.2C3EE272--
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Date: Wed, 24 Mar 1999 09:43:24 -0800
From: Erik Spjut
Subject: Re: Frequency Shifting & Hilbert Transforms
In-reply-to: <199903181159.MAA08047@nett.irt.de>
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To: Jens Groh , csound@maths.ex.ac.uk
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I forget that not everyone reads all of the messages on the list. Useful
IIR filters have poles very near to the unit circle. To keep the filters
from going unstable requires some very sophisticated numerical algorithms.
John ff. and I made some attempts to improve the numerical stability of
filter2 et al. but they didn't help enough. The filter2 family works fine
for two-or-fewer-pole filters but is numerically unstable at higher pole
counts when the poles are near the unit circle. When someone with numerical
methods experienec has time, we'll try to fix the filter2 family for higher
orders. Any volunteers?
At 12:59 PM +0100 3/18/99, Jens Groh wrote:
>Hello Josep, hello list!
>
>Josep M Comajuncosas wrote:
>> I=B4m rather intrigued with your code though I din =B4t have had engough
>>time to test it.
>> Why are you using biquad sections instead of filter2 which implement a
>>transposed form-II digital filter lattice?
>> How did you design the elliptical filter?
>
>Thanks - I'm happy to read a positive reply.
>I didn't get filter2 nor zfilter2 to work. Actually, I would have
>preferred them, especially zfilter2 because of its tuneability. But I
>encountered strange problems: The filters were unstable - no idea, why.
>Even when partitioned in biquad sections with the same coefficients that I
>use with the biquad opcode, their output signals swung up to infinity. I
>didn't try Csound versions other than DirectCsound 2.7 nor tried anyhow to
>track the problem down. Maybe I misunderstood something about (z)filter2.
>The coefficients were designed with a commercial DOS program from the days
>of yore. Not a good one, but the only one I had access to at that time. It
>uses probably the bilinear transform method. The filter and its design
>procedure may be improvable, but for the moment, I just wanted to have
>some not-too-bad filter.
>
>Happy frequency shifting!
>Jens
>
>P.S. Dear Sean Costello, would you like to tell us what you want to use
>the frequency shifter for?
Dr. R. Erik Spjut
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From: mark williamson
To: Csound
Subject: Help with midi/realtime and ukelales(sp?)
Date: Wed, 24 Mar 1999 23:34:48 -0000
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Hi,
a quick question - I've been messing about with
real time stuff in direct CSound - one of the things
I am not clear about is how you figure out how long
things last. What I am doing is physical modeling - I want
to a) change the timing of envelopes based on note length
(ie. not a fixed envelope no matter what) and b) have
global elements - ie. reverb and resonaters. are there any
example .orcs or has anyone got a quick tutorial?
The second question is more specific does any one have
any physical modeling info on ukelalea's - ie string tunings
and body resonance info... (bit of a long shot this one...)
cheers
mark
_____________________________________
junklight http://dspace.dial.pipex.com/mark.williamson
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Date: Wed, 24 Mar 1999 19:29:35 -0500 (EST)
From: Greg Judelman
To: CSOUND
Subject: HetroDyne analysis of piano. (fwd)
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Here is a message forwarded from Fabio Bertolotti
----------------------------------------
----- Begin Included Message -----
A sound that is dominated by a large initial ``spike''
in amplitude, followed by a sustain portion with an
identifyable pitch, should be hard to reproduce with
adsyn + hetrodyne analysis, in my opinion.
This is so because the spike (i.e. the attack portion
created by a hammer striking a piano string) has a Fourier
transform containing energy at many frequencies, without
particular peaks at specific frequencies that could be
identified as ``fundamental'' and ``harmonic'' components
of the sound. Hetro + Adsyn work well when such particular peaks
are present. Furthermore, hetrodyne + adsyn neglect phase
information that is important in the reproduction of
spikes and rapid changes. In the sustain part of the sound,
phase is not important to the ear.
In my opinion, it would be better to reproduce the piano
sound using the addition of two signals:
1) a wavetable synthesis for the brief, but strong, attack
portion,
2) adsyn + hetrodyne data of the remaining sustain + release.
I have developed a hetrodyne analysis tool-kit that should facilitate
with the attack-sustain-release partitioning of the hetrodyne data,
plus plotting, editing & mixing + other steps. I have sent the
source + documentation + binaries fo Richard Boulanger, for the
upcomming Csound book. I hope this toolkit will help those using
adsyn.
Regards
Fabio P. Bertolotti
----- End Included Message -----
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Message-ID: <36F982B3.5E66F818@seanet.com>
Date: Wed, 24 Mar 1999 16:26:27 -0800
From: Sean Costello
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To: csound
Subject: Frequency Shifter Orchestra and Score Again
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Hi:
See prior email for comments. I am pasting the orchestra and score into
the email, for those people who can't read attachments. Lots of
reformatting will need to be done to get these to work, as the email
program will probably put carriage returns in between the coefficients
for the filters.
Sean Costello
; frequencyshifter.orc
;
; Sean Costello
; costello@seanet.com
; March 24, 1999
;
; An implementation of single-sideband (SSB) amplitude modulation.
; Essentially similar to the Bode/Moog implementation, but using
; an FIR Hilbert transformer to obtain the 90 degree phase shift,
; as opposed to the allpass phase differencing network ("Dome filter")
; used by Bode and Moog. The drawback of the FIR approach is that
; the output is delayed with regards to the input, but in this
; implementation the delay is relatively minor (100 samples, or
; about 2.3 milliseconds).
sr = 44100 ; Hilbert transformer designed to work at 44100.
kr = 4410
ksmps = 10
nchnls = 2
instr 1 ; Frequency Shifter, Attempt #4
idur = p3
ibegshift = p4 ; Initial amount of frequency shift. Can be positive or
negative number.
iendshift = p5 ; Final amount of frequency shift. Can be positive or
negative number.
; I use a simple linear envelope to control frequency shift. Can be
modified to be
; controlled by an oscillator, random signals, another signal, etc.
kfreq linseg ibegshift, idur, iendshift
; I use soundin for my input, but any input should work. My particular
test file was a John
; Fahey sample, but I also used a simple sine oscillator during the
testing of the
; orchestra in order to test how much of the unwanted band was
rejected. You might want to
; try running signals generated by pluck through the frequency shifter,
in order to
; get drum-like sounds, as the frequency shifter will "detune" the
harmonic partials
; generated by pluck. Voice works great, too.
ain soundin "C:\\Csound\\Waves\\despair.wav", 0
; Gain has to be adjusted to compensate for loss of the FIR Hilbert
Transformer.
ain2 = ain / 1.570796327
; First part of a 201 sample length FIR Hilbert Transformer. I found
that the longer the Hilbert
; transformer, the better, as you obtained more accuracy throughout the
frequency range. With lower
; order Hilbert transformers, you get less rejection of the unwanted
passband, due to the
; Hilbert transformer not covering the entire frequency range accurately
(there is unacceptable
; rolloff at low and high frequencies). A 201-tap Hilbert transformer
works nicely, with quite
; acceptable rejection of the unwanted passband. I am using filter2 to
implement the transformer,
; as per the recommendation of Jens Groh. Coefficients obtained from
; http://www.cs.york.ac.uk/~fisher/mkfilter/ (thanks to Roger Klaveness
for the pointer).
afilt1 filter2 ain2, 48, 0, +0.0000000000, +0.0008103736,
+0.0000000000, +0.0008457886, +0.0000000000, +0.0009017196,
+0.0000000000, +0.0009793364, +0.0000000000, +0.0010798341,
+0.0000000000, +0.0012044365, +0.0000000000, +0.0013544008,
+0.0000000000, +0.0015310235, +0.0000000000, +0.0017356466,
+0.0000000000, +0.0019696659, +0.0000000000, +0.0022345404,
+0.0000000000, +0.0025318040, +0.0000000000, +0.0028630784,
+0.0000000000, +0.0032300896, +0.0000000000, +0.0036346867,
+0.0000000000, +0.0040788644, +0.0000000000, +0.0045647903,
+0.0000000000, +0.0050948365, +0.0000000000, +0.0056716186,
+0.0000000000, +0.0062980419, +0.0000000000, +0.0069773575,
+0.0000000000, +0.0077132300, +0.0000000000, +0.0085098208,
+0.0000000000, +0.0093718901
; Unfortunately, filter2 can only accept 50 coefficients for an FIR
filter. I came up with a bit
; of a kludgy solution, but it works well: Seperate the Hilbert
transformer into several FIR
; filters, and delay the additional FIR filters by the amount that they
would be delayed if
; they were in one continuous filter, then add together the results.
For example, afilt2 would
; originally be taps #49-96 in the Hilbert transformer, so the output of
afilt2 is run through
; a delay set to 49 samples.
afilt2 filter2 ain2, 48, 0, +0.0000000000, +0.0103049226,
+0.0000000000, +0.0113152847, +0.0000000000, +0.0124104218,
+0.0000000000, +0.0135991079, +0.0000000000, +0.0148917649,
+0.0000000000, +0.0163008758, +0.0000000000, +0.0178415242,
+0.0000000000, +0.0195321089, +0.0000000000, +0.0213953037,
+0.0000000000, +0.0234593652, +0.0000000000, +0.0257599469,
+0.0000000000, +0.0283426636, +0.0000000000, +0.0312667947,
+0.0000000000, +0.0346107648, +0.0000000000, +0.0384804823,
+0.0000000000, +0.0430224431, +0.0000000000, +0.0484451086,
+0.0000000000, +0.0550553725, +0.0000000000, +0.0633242001,
+0.0000000000, +0.0740128560, +0.0000000000, +0.0884368322,
+0.0000000000, +0.1090816773, +0.0000000000, +0.1412745301,
+0.0000000000, +0.1988673273
; Delay for afilt2.
adel2 delay afilt2, 49/sr
; See above for explanation of afilt3, afilt4, afilt5, and adel3, adel4,
adel5
afilt3 filter2 ain2, 48, 0, +0.0000000000, +0.3326528346,
+0.0000000000, +0.9997730178, +0.0000000000, -0.9997730178,
+0.0000000000, -0.3326528346, +0.0000000000, -0.1988673273,
+0.0000000000, -0.1412745301, +0.0000000000, -0.1090816773,
+0.0000000000, -0.0884368322, +0.0000000000, -0.0740128560,
+0.0000000000, -0.0633242001, +0.0000000000, -0.0550553725,
+0.0000000000, -0.0484451086, +0.0000000000, -0.0430224431,
+0.0000000000, -0.0384804823, +0.0000000000, -0.0346107648,
+0.0000000000, -0.0312667947, +0.0000000000, -0.0283426636,
+0.0000000000, -0.0257599469, +0.0000000000, -0.0234593652,
+0.0000000000, -0.0213953037, +0.0000000000, -0.0195321089,
+0.0000000000, -0.0178415242, +0.0000000000, -0.0163008758,
+0.0000000000, -0.0148917649
adel3 delay afilt3, 97/sr
afilt4 filter2 ain2, 48, 0, +0.0000000000, -0.0135991079,
+0.0000000000, -0.0124104218, +0.0000000000, -0.0113152847,
+0.0000000000, -0.0103049226, +0.0000000000, -0.0093718901,
+0.0000000000, -0.0085098208, +0.0000000000, -0.0077132300,
+0.0000000000, -0.0069773575, +0.0000000000, -0.0062980419,
+0.0000000000, -0.0056716186, +0.0000000000, -0.0050948365,
+0.0000000000, -0.0045647903, +0.0000000000, -0.0040788644,
+0.0000000000, -0.0036346867, +0.0000000000, -0.0032300896,
+0.0000000000, -0.0028630784, +0.0000000000, -0.0025318040,
+0.0000000000, -0.0022345404, +0.0000000000, -0.0019696659,
+0.0000000000, -0.0017356466, +0.0000000000, -0.0015310235,
+0.0000000000, -0.0013544008, +0.0000000000, -0.0012044365,
+0.0000000000, -0.0010798341
adel4 delay afilt4, 145/sr
afilt5 filter2 afilt4, 9, 0, +0.0000000000, -0.0009793364,
+0.0000000000, -0.0009017196, +0.0000000000, -0.0008457886,
+0.0000000000, -0.0008103736, +0.0000000000
adel5 delay afilt5, 154/sr
; The complete Hilbert transformer is the sum of afilt1, plus the
delayed outputs of the other filters.
ahilb = afilt1 + adel2 + adel3 + adel4 + adel5
; The original input signal is delayed by the amount of delay found in
the Hilbert transformer.
adelay delay ain, 100/sr
; Generation of sine and cosine signals for frequency shifting
asin oscili 1, kfreq, 1
acos oscili 1, kfreq, 1, .25
; Hilbert transform multiplied by sine modulation signal, while
; delayed input is multiplied by cosine modulation signal.
amod1 = ahilb * asin
amod2 = adelay * acos
aout1 = (amod1 + amod2) * 0.7 ; Upshift output (although this varies on
sign of modulation signal).
aout2 = (amod1 - amod2) * 0.7 ; Downshift output (although this varies
on sign of modulation signal).
; Gains vary with shifting amount, so adjust as needed. Gains of .7
; to .9 should provide unity gain with most settings.
outs aout1, aout1 ; Try small shifting amounts (0-6 Hz), with aout1 in
one channel,
; and aout2 in another. Makes for nice stereo effects.
endin
; frequencyshifter.sco
; Sine table for modulation oscillators
f1 0 16384 10 1
i1 0 30 -50 50
e
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From: Sean Costello
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To: mark williamson
CC: Csound
Subject: Re: Help with midi/realtime and ukelales(sp?)
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mark williamson wrote:
> The second question is more specific does any one have
> any physical modeling info on ukelalea's - ie string tunings
> and body resonance info... (bit of a long shot this one...)
You could probably adapt much of the physical modelling work for
classical guitar to model a ukelele. CCRMA is where most of the
physical modelling work started - check http://ccrma-www.stanford.edu. I
know that there has also been some work done in Finland (can't remember
any names at the moment) on guitar modelling, using commuted physical
modelling. Basically, the impulse response of the body, as excited by a
plucking motion, is sampled, and this sampled impulse is used as the
excitation input to the waveguide. The result is identical to sending
the output of a waveguide through body resonators, but is much less
computationally intensive.
You could probably start by using four pluck generators, one for each
string of the ukelele. Have the strings "plucked" at slightly different
times for a chord, a few milliseconds apart, to get a proper strumming
sound (Common Lisp would be a great environment to work with for
generating the note lists for this type of thing). Run the output of
all of the pluck instruments into a few resonators (reson), to simulate
the body resonances. A good musical instrument physics textbook, like
Fletcher and Rossing's "The Physics of Musical Instruments," might have
a listing of ukelele body resonances; otherwise, just start with a
listing of guitar resonances, and scale them up.
Having said that...why the hell would you want to use state of the art
computer technology to simulate a freaking UKELELE?!?! :)
I guess it is just my own bias. I like using the computer to simulate
"real-world" instruments like bowed instruments and choral voices.
However, as a guitarist, the Karplus-Strong stuff doesn't do that much
for me. If I wanted that sound, I'd just pick up a guitar. I'm sure
that pianists, trombonists, etc. might get really excited about the
Karplus-Strong stuff.
Sean Costello
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From: rasmus ekman
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Subject: Re: C sound/No sound
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Vote:
Add standard suffix to standard output format files unless
there is a dot after the first alphanumeral in the name.
IRCAM format being non-standard for PC it would get nothing (?)
re
jpff@maths.bath.ac.uk wrote:
>
> Following the number of people who have pointed out that winsound
> takes the audio file name to be what is said, and does not add .wav or
> .aif to teh name, do you'll think that it shoudl add suffix if one is
> not given explicitely? Woudl be easy enough to do but I worry that it
> will then be hard to create an audio file without a suffix.
>
> Advice? Thoughts?
>
> ==John ffitch
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From: Lonce LaMar Wyse
Organization: Kent Ridge Digital Labs
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Subject: Cakewalk and CSound
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I'm also a Cakewalk+CSound user.
They don't like to run on the same machine, the symptom being sound
"breakup" (my guess is the problem is over-taxing the easily over-taxed
Windows timer mechanism), so I run Cakewalk on an older slower machine
and run a MIDI cable to my laptop running csound. A keybaord, Peavy 1600
MIDI controller or WX-5 windcontroller are all great for jamming with
prerecorded tracks.
One nice thing about Cakewalk is the "instrument definition files" which
allow you to assign names to note numbers, and names to controllers that
show up in the cakewalk environment. Very very handy.
Cakewalk 8 allows you to import videos - then you can frame-by-fram
synch video to Csound events and parameter changes.... are we having fun
yet?
- lonce
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Date: Thu, 25 Mar 1999 09:57:02 +0100 (MET)
From: Thomas Huber
Subject: Re: Second thoughts
To: csound@maths.ex.ac.uk
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[About compressing with csound]
> The results? A drum track which to be sure sounded nicely compressed (only
> breathed a little bit), but sounded as though you were listening to it on your
> favorite local college radio station... tuned just enough off station for the
> noise floor to be annoying. Eh bien.
Yesterday I just tested the 'dam' opcode to compress the output of
all my instruments (the instruments add to a global audio variable which
is then read out by the 'mixer' instrument which then outputs with
'outs'). You are right, it sounds like I would be 'listening to it on my
favorite college station...'. So 'dam' produces extreme noise, it a total
catastrophe. I will implement a compressor directly in csound, using
'rms' opcode to measure the sound level etc.
I hope this will sound better...
Thomas
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From: mark williamson
To: Csound
Subject: RE: Help with midi/realtime and ukelales(sp?)
Date: Thu, 25 Mar 1999 10:40:31 -0000
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>Having said that...why the hell would you want to use state of the art
>computer technology to simulate a freaking UKELELE?!?! :)
Just a starting point - I'm not after perfection here - what I'm looking
for is very short strings and harsh resonances - the Ukelele is such an
offensive instrument. Part of my musical explorations at the moment
is looking at the sounds of folk/traditional music (not the content though)
some of the instruments are really bizarre.
Thanks for your help
mark
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Date: 25 Mar 99 07:37:45 EST
From: Rick Mealey
To: huber@iamexwi.unibe.ch
Subject: Still not ready for prime time [Was Re: Second thoughts]
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huber@iamexwi.unibe.ch (Thomas Huber) wrote, in part:
> You are right, it sounds like I would be 'listening to it on my
> favorite college station...'. So 'dam' produces extreme noise, it a tot=
al
> catastrophe. =
Just so that we're all clear on the subject, the point I (obviously) fail=
ed to
make in my last post is that I may have been trying to use the dam opcode=
improperly.
As I understand the opcode, I can have my signal compress (or expand) at =
one
ratio below the threshold, and compress/expand at a second ratio above th=
at
threshold. I haven't done enough work with the opcode to determine whethe=
r it
will produce that level of noise no matter what settings I use. Thomas, s=
hould
I infer that from your post...?
And for that matter I probably haven't put the sweat equity into Cool Edi=
t to
get the best results there, either. As I recall, I had originally sampled=
my
individual tracks at 32 bits (OK, really 24-bit plus mantissa). But when =
I had
CSound read the 32-bit samples, I discovered that, for example, the vocal=
s
sounded kinda-sorta like the distortion Beck got shouting "Two turntables=
and
a microphone" into a stomp box-- without any of the musicality. I wound u=
p
converting to 16-bit, and that may have introduced some noise and/or alia=
sing,
because my hi-fi audio suddenly sounded a little less hi ~and~ a little l=
ess
fi.... Bottom line is, I'm still not ready to unleash any of my CSound on=
the
world....
Back to the drawing board,
Rick
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