Instantaneous (analogue) compression of speech signals

[John Woodgate]

I read in sci.electronics.design that Jim Thompson
<thegreatone@example.com> wrote (in <j97ot0l4l7facned07n4f7h091gpismk06@
4ax.com&gt about 'Instantaneous (analogue) compression of speech
signals', on Wed, 5 Jan 2005:

How about TANH?

Sadist! Do you KNOW the series expansion of tanh? I realise that it is
easily implemented with a long-tailed pair, but I can't calculate the
harmonic spectrum; I'll have to write a Mathcad script.

But thanks for the suggestion - I think.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk

 

[John Woodgate]

I read in sci.electronics.design that Ken Smith
<kensmith@green.rahul.net> wrote (in <crh970$n91$1@blue.rahul.net&gt
about 'Instantaneous (analogue) compression of speech signals', on Wed,
5 Jan 2005:

How about this:


CA3080
---------!+\
! >---------+---------
--!+/ !
! !
---------------+
!
---
---
!
GND

The distortion should only be odd order harmonics since the CA3080 puts
out nearly the same current in each direction. Since it is slew rate
limiting, it is equivelent to a trebble boost, clip and then trebble
cut. This should reduce the amplitude of the harmonics. You can vary
the slew rate slope with the pin 5 current on the CA3080.

It looks very interesting in terms of functions. But the device is
obsolete and there is no recommended replacement.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk

 

[Rich Grise]

On Wed, 05 Jan 2005 09:58:54 -0700, Jim Thompson wrote:

On Wed, 5 Jan 2005 16:19:03 +0000, John Woodgate
jmw@jmwa.demon.contraspam.yuk> wrote:

I read in sci.electronics.design that Jim Thompson
thegreatone@example.com> wrote (in <kpunt05gl9dn5v862g759dbb033sbpf2fc@
4ax.com&gt about 'Instantaneous (analogue) compression of speech
signals', on Wed, 5 Jan 2005:
Recalling an ancient project... isn't the ideal clipping case something
like (for example)... a 2dB change in the input produces a 1dB change in
the output?

There was a time when anything more than that introduced unpleasant
artefacts into the signal, but that doesn't happen with modern designs.

Keep us posted... this is becoming a fascinating subject.

It's sounding more and more like John Woodgate is looking for a log amp. I
had a class on them once. ;-)

And to Mr. Dyson - thanks for your exposition; I "get" the concept of FFT
- turn the signal sideways, nothing to it! I just don't want to plow
through all that arithmetic. I once did a bank of 8 bandpass filters, much
like a "graphic equalizer", in an effort to extract formant information
based on the _relationship_ of the components. I saw an experiment on
edjamacayshunal TeeVee, where they did a Pavlov's Dog-style experiment,
with human infants. They used, instead of a bell, a human voice saying one
phoneme, for example, "ah", or "ee". They trained the human infant to
respond to a particular phoneme, and the infant responded to the same
phoneme, _no matter who said it_. Old man, young girl, child, adult man,
adult woman - it didn't matter, as long as they were saying the same
phoneme. So clearly, the phoneme itself is determined by the _pattern_ of
the various formants, without regard to the fundamental, or even the
specific frequencies.

I got bogged down while trying to find some kind of pattern-matching
algorithm, and had to abandon the project because of other considerations.
It was intended to be a speakwrite. Or a first pass at one. To control a
handicap-assist robot.

Thanks,
Rich

 

[John Woodgate]

I read in sci.electronics.design that Jim Thompson
<thegreatone@example.com> wrote (in <p8kot0ls6m59i31qfdm9hoi6h7mu3fvgau@
4ax.com&gt about 'Instantaneous (analogue) compression of speech
signals', on Wed, 5 Jan 2005:

See...

Newsgroups: alt.binaries.schematics.electronic
Subject: Audio Clipping Question, ala S.E.D (Woodgate) -
TanhClipper.pdf
Message-ID: <evjot0h1g9lid521q99blvhcsqc62keeur@4ax.com

It hasn't come through. Could you please email to me at JMW[at]JMWA[dot]
demon.co.uk?
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk

 

[Ben Bradley]

On Tue, 4 Jan 2005 19:19:54 +0000, John Woodgate
<jmw@jmwa.demon.contraspam.yuk> wrote:

I read in sci.electronics.design that Jim Thompson
thegreatone@example.com> wrote (in <n7olt0pimqncoidcohk7mi2jgs8jsjfa82@
4ax.com&gt about 'Instantaneous (analogue) compression of speech
signals', on Tue, 4 Jan 2005:

http://www.semiconductors.philips.com/acrobat_download/applicationnotes/
AN176.pdf

This is helpful for theory but the devices are not now available, I
think.

I'll come back to the file at this link...

...

http://www.onsemi.com/site/products/parts/0,4454,62,00.html

Not instantaneous; these use a rectifier and thus involve at least one
time-constant.

Any VCA-based (or "opto" [LED/Light Bulb and light-dependent
resistor] or vacuum-tube based "Vari-MU") compressor or limiter will
"involve at least one time constant" but the definition of a "limiter"
is (in addition to a near-infinite compression ratio) that the attack
time is short enough to be negligible and no peak will come through.

You didn't object to the circuitry in the first link (AN176.pdf)
not being instantaneous, and for the circuits described, it's clearly
not. Quoting from page 10-6:

"CRECT acts as the rectifier’s filter cap and directly affects the
response time of the circuit. There is a trade-off, though, between
fast attack and decay times and distortion."

It doesn't take too much circuitry to make the attack and decay
times independent (little more than an op-amp as voltage follower) and
have the attack time arbitrarily short, though of course a fast attack
will always distort the first wave at the onset of a louder signal, as
the gain is reduced as the instantaneous input signal goes above the
threshold. This might make a 'click' as the first quarter-cycle or so
is flattened, but I don't think it should be too audible or
objectionable.*

You might ask this on rec.audio.pro where they use this sort of
stuff every day.

* Unless you also set the release time to be very short. Go to
http://www.fmraudio.com/ , click on FAQ, and see the discussion for
the fourth question, "Why does the RNC distort my bass guitar?"

Thanks for your help.

-----
http://mindspring.com/~benbradley

 

[Ken Smith]

If you take a sine wave and run it through a circuit that does:


Y = X ^(17/19)

the sine wave's RMS amplitude will be compressed towards about 0.98V RMS
and there will be some distortion. The 3rd harmonic will be about 2.7%.

Assume that the sine wave we start with is 300Hz.

A phase shifter (all pass filter) can be made with a Q such that the
900Hz, 3rd harmonic is shifted by 180 degree relative to the 300Hz
sinewave.

If we take this shifted signal and do another X^(17/19) operation on it,
the 3rd harmonic will only be about 0.2%

You don't need the phase shift to be exactly 180 degrees. Any non-zero
phase shift and two steps of (17/19) soft clipping will result in less
harmonic content than one step of (17/19)^2 clipping would produce.

If more distortion can be lived with, a lower power such as (11/13) could
be used.

Since the band of interest is 300Hz to 3KHz, we don't have to worry about
the harmonics of the frequencies above 1KHz. Those can be removed with a
simple low pass filter. I haven't verified it yet but it seems to me that
3 stages of phase shifter and 4 clippers should be able to make a
significant compression of amplitude but make less that 5% distortion on a
sine wave.

The intermodulation distortion will not be made zero by this method. If
the input has more than one frequency component, the distortion will be
much higher.

--
--
kensmith@rahul.net forging knowledge

 

[John Woodgate]

I read in sci.electronics.design that Ken Smith
<kensmith@green.rahul.net> wrote (in <crktu7$gpl$1@blue.rahul.net&gt
about '"all pass" thought about (analogue) compression', on Fri, 7 Jan
2005:

Since the band of interest is 300Hz to 3KHz,

It isn't: I've posted that it's 100 Hz to 5 kHz (at least).

we don't have to worry
about the harmonics of the frequencies above 1KHz. Those can be removed
with a simple low pass filter. I haven't verified it yet but it seems
to me that 3 stages of phase shifter and 4 clippers should be able to
make a significant compression of amplitude but make less that 5%
distortion on a sine wave.

Interesting, a bit more ambitious circuit-wise than I really expected,
and how to realise the weird fractional power law?

The intermodulation distortion will not be made zero by this method. If
the input has more than one frequency component, the distortion will be
much higher.

The input is mainly speech, but there could be music as well. In any
case, many frequencies.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk

 

[Robert Baer]

Rich Grise wrote:

On Thu, 06 Jan 2005 04:37:45 +0000, John Woodgate wrote:

I read in sci.electronics.design that John Larkin <john@spamless.usa
wrote (in <dvept093udvtbifbflkkhn80hmtlfsjn9v@4ax.com&gt about
'Instantaneous (analogue) compression of speech signals', on Wed, 5 Jan
2005:
On Mon, 3 Jan 2005 21:21:14 +0000, John Woodgate
jmw@jmwa.demon.contraspam.yuk> wrote:

Does anyone here have any experience of instantaneous (analogue)
compression (aka soft clipping) of speech signals? I've been doing a
little work on it but I'm unable to judge the resulting sound quality.
Why do treble boost controls no longer have any audible effect for me?
(;-)

If you could tolerate a time delay, you could do a very nice smooth
AGC thing without the clipping problem that results from a fast-attack
signal.

True, this technique is well-known, but it's costly. I'm looking for an
ingenious low-cost solution.

A negative delay line (future predictor) would be handy here,
too.

I can do that with a -2 ohm resistor, which has a conductance of -half a
mho.

I guess they haven't been able to get those thiotimoline capacitors off
the production line yet ...

;-)
Rich

The first lab prototype that will be built next month, will go
zonkers, thus producing that Richter 9 earthquake..

 

[Ken Smith]

In article <41DF198C.4F76DB7A@deadend.com>, gwhite <gwhite@deadend.com> wrote:

Ken Smith wrote:

In article <rp0ut0567a2snj680r34esn95mv2ilav1u@4ax.com>,
Jim Thompson <thegreatone@example.com> wrote:
[...]
I'm puzzled by "RF/IF" clipping. How does that work to improve the
demodulated audio?

In a single side band reciever:

It is actually done in the *transmitter* as a purposeful processing technique.

I've also seen it in the receiver as a way to prevent the headphones from
blowing your ears off. It was a simple clamping diode just in front of the
second detector stage. I suspect that if it is done in the transmitter,
and the receiver's clipping level is above the transmitters, it would
greatly reduce the distraction of noise spikes.

FM and AM don't per se "need" RF/IF clipping, but could use it effectively too.

Actually just about all FM receivers clip in the IF strip several times.
The ratio detector also effectively clipps the RF too.




------------>!----+--/\/\/----
( ! !
( --- !
+-- ---C1 +---- Audio
( ! ! !
( ! ! !
- ! ------!<-----+--/\/\/----
!
--------



C1 in my sketch is big enough that the voltage on it does not change at
audio frequencies.


[...]

The distortion ends up being heavy on the IM distortion effects and light
on harmonics. For some reason, this seems to be easier on the hear.


I would say it is likely the same on IM. (Or at least there is no special
reason for it to be different.) The harmonics, as you say, are "gone." So the
total distortion is less.

Yes the total is lower, but I suspect tha in real circuitst the IM is in
fact slighty higher for the same amount of amplitude compression. When
you clip at baseband frequencies, you AM modulate one component with
another. At RF frequencies, there is sure to be some phase modulation
going on too.


--
--
kensmith@rahul.net forging knowledge

 

[gwhite]

Ken Smith wrote:

In article <41DF198C.4F76DB7A@deadend.com>, gwhite <gwhite@deadend.com> wrote:
Ken Smith wrote:

In article <rp0ut0567a2snj680r34esn95mv2ilav1u@4ax.com>,
Jim Thompson <thegreatone@example.com> wrote:
[...]
I'm puzzled by "RF/IF" clipping. How does that work to improve the
demodulated audio?

In a single side band reciever:

It is actually done in the *transmitter* as a purposeful processing technique.

I've also seen it in the receiver as a way to prevent the headphones from
blowing your ears off.

Basically someone spent too much money on the headphone PA. More seriously,
sure, it is the "best known way to clip," but rarely applied in this manner.

It was a simple clamping diode just in front of the
second detector stage. I suspect that if it is done in the transmitter,
and the receiver's clipping level is above the transmitters, it would
greatly reduce the distraction of noise spikes.

FM and AM don't per se "need" RF/IF clipping, but could use it effectively too.

Actually just about all FM receivers clip in the IF strip several times.
The ratio detector also effectively clipps the RF too.

------------>!----+--/\/\/----
( ! !
( --- !
+-- ---C1 +---- Audio
( ! ! !
( ! ! !
- ! ------!<-----+--/\/\/----
!
--------

C1 in my sketch is big enough that the voltage on it does not change at
audio frequencies.

Obviously, and the RF of the TX'er often clips too. That's why I said it must
be "re-basebanded" (taken back to audio to FM/PM modulate, after processing) for
FM. FM is a "constant envelope system." Think about it. Clippers in FM are
pre-modulator for the purpose of limiting peak deviation and thus allowing
higher average deviation. No one is saying "AM a FM system."

The distortion ends up being heavy on the IM distortion effects and light
on harmonics. For some reason, this seems to be easier on the hear.


I would say it is likely the same on IM. (Or at least there is no special
reason for it to be different.) The harmonics, as you say, are "gone." So the
total distortion is less.

Yes the total is lower, but I suspect tha in real circuitst the IM is in
fact slighty higher for the same amount of amplitude compression. When
you clip at baseband frequencies, you AM modulate one component with
another. At RF frequencies, there is sure to be some phase modulation
going on too.

I doubt that. You can "IF clip" at 20 kHz for voice grade systems, if you
want. (A common frequency is 455 kHz, which is practically DC.) The IM should
be the same. For example, term third order term (for two-tone) is:

a3*[cos(w1*t) + cos(w2*t)]^3

This is a general result--no bias regarding what the actual frequencies are.
Memory effects should be vanishing for the typical low frequencies of
processing.

 

[Fred Bartoli]

"John Woodgate" <jmw@jmwa.demon.contraspam.yuk> a écrit dans le message de
news:2pRIs3CKdb2BFwac@jmwa.demon.co.uk...

Does anyone here have any experience of instantaneous (analogue)
compression (aka soft clipping) of speech signals? I've been doing a
little work on it but I'm unable to judge the resulting sound quality.
Why do treble boost controls no longer have any audible effect for me?
(;-)
--

John,

I've not followed the whole thread so I don't know whether sb proposed this
or not.



10n
2.2K
___ || ___
.-|___|---||---+--|___|--+ .--------.
| || | | | |
| ___ | 15K | | |
-+--------|___|-+ | | |
| | | |\| | 15K
15K | |\| | 1K '--|-\ | ___
'---|-\ | ___ | >--'-|___|--+-----
| >--+-|___|--+--+----|+/ |
.---|+/ | | |/| |
| |/| | | ---
=== - V 10n ---
GND 2 diodes ^ - |
| | |
(or diode monuted BJTs) | | ===
====== GND
GNDGND

(created by AACircuit v1.28 beta 10/06/04 www.tech-chat.de)


I didn't tried it in real because I don't have what's required, but in
simulation it has some interesting effects.

I guess 1kHz is about a good corner frequency but of course you can adapt
it.


--
Thanks,
Fred.

 

[John Woodgate]

I read in sci.electronics.design that Fred Bartoli <fred._canxxxel_this_
bartoli@RemoveThatAlso_free.fr_AndThisToo> wrote (in <41e25e76$0$6635$62
6a14ce@news.free.fr&gt about 'Instantaneous (analogue) compression of
speech signals', on Mon, 10 Jan 2005:

I've not followed the whole thread so I don't know whether sb proposed
this or not.

I've got something similar to that at present, although it has *bass-
cut* pre-processing rather than treble-boost. Jim Thompson's tanh
limiter does seem to have advantages, but I'll try your pre-processing
first, because that involves fewer changes to my breadboard.

I planned to do some work on it today, but of course, those damned
'clients' have intervened. -(.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk

 

[gwhite]

John Woodgate wrote:

However, much of the work on articulation index
and speech intelligibility has occurred in the radio/telecom field.
More specifically, radio/telecom work has dealt quite directly with the
concept of speech clippers--probably more so than other fields since it
tends to be a distinctly peak power limited environment (especially
radio). It is a source of info so-to-speak. That is, the effects of
clipping on speech intelligibility has been dealt with directly.

Agreed, although those studies are on signals that can be more seriously
degraded (not by the clipping but by other system characteristics -
bandwidth and noise) than those I'm concerned with.

Well that was my point when I said I doubted the application here. The
*traditional* case for it doesn't seem to exist; the transmission path doesn't
have those "serious degradations" according to you. Nor have you stated that
you have a peak limited system. Again, I realize you are claiming a new or
uncommonly known aspect.

So giving the "benefit of the doubt" to your IP, I wonder if you have considered
preemphasis before clipping and then deemphasis thereafter (unless a boost is
desired anyway). It is claimed that harmonic distortion from clipping can
"cover" the high frequency formants and reduce intelligibility, and in this way
preemphasis helps. (As a side note, this lack of harmonics covering high
frequency formants may help explain why RF/IF clipping is superior.)

 

[John Woodgate]

I read in sci.electronics.design that gwhite <gwhite@deadend.com> wrote
(in <41E2D48B.C13722E3@deadend.com&gt about 'Instantaneous (analogue)
compression of speech signals', on Mon, 10 Jan 2005:

Well that was my point when I said I doubted the application here. The
*traditional* case for it doesn't seem to exist; the transmission path
doesn't have those "serious degradations" according to you. Nor have
you stated that you have a peak limited system.

I'm not sure what you mean by 'peak-limited system' in this context,
since what I am asking about is peak limiting by another name, but there
is a considerable financial incentive to reduce the current and
compliance voltage requirements of the amplifier. I use those terms
because the induction loop load is reactive and is normally driven by a
current-source amplifier.

Again, I realize you
are claiming a new or uncommonly known aspect.

It is generally unknown: several people working in the field know about
it, but not in quantified terms. The IP is in the quantification.

So giving the "benefit of the doubt" to your IP, I wonder if you have
considered preemphasis before clipping and then deemphasis thereafter
(unless a boost is desired anyway). It is claimed that harmonic
distortion from clipping can "cover" the high frequency formants and
reduce intelligibility, and in this way preemphasis helps.

Yes, spectral conditioning is a known technique since the 'infinite
clipping' work of Licklider et al long ago. There are two essentially
different ways of doing it, and I have tried one. The other way has been
proposed in this thread, and I intend to try that as well.
--
Regards, John Woodgate, OOO - Own Opinions Only.
The good news is that nothing is compulsory.
The bad news is that everything is prohibited.
http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk

 

[gwhite]

Ken Smith wrote:

In article <41DEF448.D8A84031@deadend.com>, gwhite <gwhite@deadend.com> wrote:
Ken Smith wrote:


I haven't verified it yet but it seems to me that
3 stages of phase shifter and 4 clippers should be able to make a
significant compression of amplitude but make less that 5% distortion on a
sine wave.

As far as a sine wave goes, the RF clipper eliminates harmonics entirely. A
baseband version has been given. The patent expired.

The intermodulation distortion will not be made zero by this method.

Nor by any method.

FT the signal

Raise each amplitude to the 5/7th power but don't change the phase

That's a general problem where I can't see how you've provided a method. How do
you propose a system resolve two tones and raise power individually. If this
were possible, we really could have amplifiers with *no* IM products. It has
never to my knowledge been accomplished and really seems a contradiction of
terms: non-linearity that has no non-linearity products. ???

The input to the system is random. "Tone frequencies" are unknown beforehand.
More specifically, the inputs are not even tones, and no reasonable filtering
method could have the needed resolution. The actual implementation of a
(ˇ)^(m/n) *frequency domain* "clipper" needs a bit more discussion too.

iFT the new spectrum.

Okay, lets define a standard two-tone equal level signal (let's use analytic
signals for ease):

x(t) = (e^(jˇw1ˇt) + e^(jˇw2ˇt))/(2*pi)

These tones can be arbitrarily "close" or "far" apart.

FT'ing this:

F{x(t)} = X(jw) = dirDel(w-w1) + dirDel(w-w2)

where

dirDel(ˇ) := the dirac delta function

Because applying the rational power function to each tone individually seems to
have no obvious general solution, we apply it to the input generally:

Y(jw) = (dirDel(w-w1) + dirDel(w-w2))^(m/n)

where m/n is some rational fraction; 5/7 if you like.

IFT'ing:

1 /inf
y(t)=invF{Y(jw)}= ----| e^(-jˇwˇt0)ˇ(dirDel(w-w1) + dirDel(w-w2))^(m/n)dw
2ˇpi/-inf

I don't care if you evaluate in the frequency or time domain. It remains to be
shown how this will not produce distortion products, either IM or harmonic
(harmonic is simply a subset of IM anyway.) I would actually like to see the
transform of the integral anyway.

No new frequencies are created and no interaction between the amplitudes
has happened. This method has neither harmonic nor IM distortion.

But you wrote:

"The intermodulation distortion will not be made zero by this method."

"The intermodulation distortion will not be made zero by this method. If
the input has more than one frequency component, the distortion will be
much higher."

So unless I misunderstand you, there is a contradiction.

 

[john jardine]

"Fred Bartoli"
<fred._canxxxel_this_bartoli@RemoveThatAlso_free.fr_AndThisToo> wrote in
message news:41e30653$0$19823$626a14ce@news.free.fr...

I didn't tried this but the sims let me expect something like this.
Can you run some other program through it, like speech and music ?

Maybe you have the opportunity to make a wav and post it back ?
I'd be curious.

For the simulation, run a 500Hz + 4kHz sin waves at various levels, and
observe the output, vs a simple R+diode clipper.
This one is amazing and can handle really **huge** overload levels. (I
didn't tried more than 2 sinus)


--
Thanks,
Fred.

I've only a few *.WAVs on the PC, so I'll see if I can pull something from
the net.
I'll post the (single unit) before and after WAVs to ABSE.
(Input WAV is 8bit at 8k bits per second. Output WAV saved as 8 bits at
11kbits per second. Single channel only.
(Ps, Done via an LTSpice sim, so the diodes suffer perfectly symmetry)
regards
john

 

[Ken Smith]

In article <41E2E7F9.DA987404@deadend.com>, gwhite <gwhite@deadend.com> wrote:

Ken Smith wrote:
[...]
clipper) was used. I really don't know what you have in mind with a single
diode.

A picture's worth 999 words:
This is the 1st example of this sort of circuit I saw.

B+
!
\
/
\ R1
/
!
----+-----+-----
!C1 ! ! -----+-----
--- ! ) ( ! To
--- ! ) ( --- Next stage
! --- ---) ( --- (detector section)
GND --- ! ) ( !
! ! ! -----+------
! ! !
! ! !
B+--!<--+-----------
D1 !
!
Plate


D1 clamps the positive swing of the tank circuit and thereby the amplitude
of the signal on it. Its from a GE communications radio, I had, that got
lost in the great flood.

R1 and C1 may have been already needed in the circuit for other reasons.
The circuit worked fairly well. The crashes of lightning etc were not too
much louder than the folking being listened to.


--
--
kensmith@rahul.net forging knowledge

 

[Fred Bartoli]

"john jardine" <john@jjdesigns.fsnet.co.uk> a écrit dans le message de
news:crv2dv$9e4$1@news6.svr.pol.co.uk...

I've only a few *.WAVs on the PC, so I'll see if I can pull something from
the net.
I'll post the (single unit) before and after WAVs to ABSE.
(Input WAV is 8bit at 8k bits per second. Output WAV saved as 8 bits at
11kbits per second. Single channel only.
(Ps, Done via an LTSpice sim, so the diodes suffer perfectly symmetry)

Ah, yes.
Thanks John, I forgot about that LTSpice capability.


--
Thanks,
Fred.

 

[Jim Thompson]

On Sun, 09 Jan 2005 13:31:52 -0700, Jim Thompson
<thegreatone@example.com> wrote:

On Sun, 9 Jan 2005 19:46:21 +0000, John Woodgate
jmw@jmwa.demon.contraspam.yuk> wrote:

I read in sci.electronics.design that Jim Thompson
thegreatone@example.com> wrote (in <i58ut0h44mmmoumuhismukcei2ao6l5chc@
4ax.com&gt about 'Instantaneous (analogue) compression of speech
signals', on Fri, 7 Jan 2005:
[snip]
Diff pair plus OpAmp, plus a DC loop to keep the diff pair balanced.

Do you mean feeding the op-amp d.c output back through a potential
divider to the base of the tail transistor?

I don't know if you need a divider or not, but definitely a low-pass.

[snip]

John, Circuit E-mailed to you, with an error in it... "quicky" is
asymmetric in the two "halves" gain

But you can fix that easily... I just mildly bungled the math, doing
it in my head :-(

...Jim Thompson
--
| James E.Thompson, P.E. | mens |
| Analog Innovations, Inc. | et |
| Analog/Mixed-Signal ASIC's and Discrete Systems | manus |
| Phoenix, Arizona Voice480)460-2350 | |
| E-mail Address at Website Fax480)460-2142 | Brass Rat |
| http://www.analog-innovations.com | 1962 |

I love to cook with wine. Sometimes I even put it in the food.