Klarinet Archive - Posting 000570.txt from 2005/03

From: "David Kumpf" <dkumpf@-----.com>
Subj: RE: [kl] that nice dark sound
Date: Tue, 22 Mar 2005 21:40:27 -0500

Ormondtoby Montoya wrote:
> Joe, you have raised (indirectly) the same question that I=20
> asked a week
> or so ago. I am hoping that someone who knows the technical details
> will post an answer.
>=20

<snipped>
>=20
> For example, does adjusting the recording equipment's treble=20
> (while recording an instrument with its own mic) cause an=20
> identical effect on the 'ratio of partials' for each note=20
> regardless of the note's position
> in the scale? Or does adjusting the treble merely alter the strength
> of all wave peaks above a certain threshold regardless of=20
> each note's fundamental frequency?
>=20
> (The latter seems more likely to me.)
>=20
I am quite sure that Ben Maas on the list can provide a much more
detailed answer, but here are the basics:

It's been a long time since I have done any analog filter design, but
yes, the latter is correct. A low-pass filter rolls off - decreases the
output amplitude of - frequencies above the cutoff frequency. The cutoff
frequency of a simple filter design, if I remember correctly, is defined
as the point where the output power is down by 3 dB. Simple designs
don't roll off the output power very quickly with respect to increasing
frequency, whereas more complex designs can have steep rolloff.

It is fairly straightforward to add another control to vary the cutoff
frequency of the filter, i.e. to raise or lower it. More on that in a
moment.

Here's a crude schematic example of a low-pass filter (hope it renders
correctly). S is the source signal, R is a resistor, C is a capacitor, O
is the output, G is ground (i.e. "earth").

S ---R-----|--- O
|
C
|
|
G

Assume S is a sinewave. As S increases in frequency, C looks more and
more like a short circuit, decreasing the amplitude of O with respect to
the input at S. As S decreases in frequency, C looks more and more like
an open circuit, increasing the amplitude of O with respect to the input
at S. Making R adjustable can change the cutoff frequency.=20

Fc =3D 1 / 2(pi)RC

More info (and a better drawing!) is at
http://en.wikipedia.org/wiki/Low-pass_filter - what is there are the
basics.

> <snipped>
> For that matter, does recording equipment even have a 'treble'
> adjustment similar to the adjustment on a player?

Analog boards that I worked with (years ago!!) typically had low, mid,
and high-pass filter adjustments. Most serious filtering in the studio
is done with much more sophisticated equalizer combinations. A graphic
equalizer provides boost and cut for a whole set of individual frequency
ranges. See http://www.rane.com/deq60.html for an example. A parametric
equalizer allows you to set the boost or cut for individual ranges that
you define by choosing the center frequency and bandwidth.

Of course, the microphone does not have a flat response curve in the
first place. For example, go look at

http://www.shure.com/images/response/fKSM27_large.gif

This mic is more sensitive between 5 - 15 kHz and then rolls off
starting at 15 kHz, back to flat at what looks like about 17 kHz.

I would think that most sophisticated studio work today is done with
digital filtering that emulates - and goes beyond - what is possible
with analog filter designs.

Which raises a possibility. Your first "ratio of partials" example is
probably feasible using a digital filtering scheme in which the filter
adapts based on tracking the fundamental. (In fact, there is probably an
existing studio or performance effect system that already does something
like this.) It's probably also possible with analog circuits to some
extent as well - I just haven't thought about the problem.

Dave Kumpf
dkumpf@-----.com

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