Instantaneous (analogue) compression of speech signals

I read in sci.electronics.design that Jim Thompson
<thegreatone@example.com> wrote (in <avgot0lpf40j7hq9pj5kubfbbgcsve5dho@
4ax.com>) about 'Instantaneous (analogue) compression of speech
signals', on Wed, 5 Jan 2005:

I've already run a simulation. Want to see it?
Yes, please. 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
 
In article <9DrqGJJlpD3BFwQx@jmwa.demon.co.uk>,
John Woodgate <noone@yuk.yuk> wrote:
I read in sci.electronics.design that Ken Smith
kensmith@green.rahul.net> wrote (in <crh970$n91$1@blue.rahul.net>)
about 'Instantaneous (analogue) compression of speech signals', on Wed,
5 Jan 2005:
How about this:


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

[....]
It looks very interesting in terms of functions. But the device is
obsolete and there is no recommended replacement.
Ok, how about a LT1228?

You even get an added buffer stage.

--
--
kensmith@rahul.net forging knowledge
 
On Wed, 05 Jan 2005 20:55:43 +0800, budgie wrote:

.... I could send you a scan of the relevant
part from one type that we found surpisingly good when overdriven by a large
margin.
I've noticed that Mr. Woodgate said, "yes, please" - for the benefit of us
roobs, would you be so kind as to post it also to
news:alt.binaries.schematics.electronic ?

Thanks,
Rich
 
On Wed, 05 Jan 2005 02:14:57 +0000, Ken Smith wrote:

In article <n7olt0pimqncoidcohk7mi2jgs8jsjfa82@4ax.com>,
Jim Thompson <thegreatone@example.com> wrote:
[...]
How about "compandors" used by most radio stations to keep their
modulation index maxed out....

I like the homomorphic compressor, not because it is better in any way but
because it is unusual.
I just LOVE that new word. "To change into oneself." ;-P
--
The Pig Bladder From Uranus, Still Waiting for
Some Hot Babe to Ask What My Favorite Planet Is.
 
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. A negative delay line (future predictor) would be handy here,
too.

John
 
On Wed, 05 Jan 2005 15:32:44 GMT, nico@puntnl.niks (Nico Coesel) wrote:

budgie <me@privacy.net> wrote:

John, have you looked at the circuits used in commercial two-way radio
equipment? It varies from some pretty harsh diode types (with predictably high
output distortion) to some which do have the effect of improved "effectiveness"
through a better (output) average-to-peak ratio. If it will help (it's a bit
hard to actually describe the circuit) I could send you a scan of the relevant
part from one type that we found surpisingly good when overdriven by a large
margin.

Someone once told me that mixing a speech signal with itself but
shifted up one octave also makes it easier to understand. Never tried
it though. I think you are referring to a similar effect.
nope.
 
On Wed, 05 Jan 2005 22:23:34 GMT, Rich Grise <richgrise@example.net> wrote:

On Wed, 05 Jan 2005 20:55:43 +0800, budgie wrote:

... I could send you a scan of the relevant
part from one type that we found surpisingly good when overdriven by a large
margin.

I've noticed that Mr. Woodgate said, "yes, please" - for the benefit of us
roobs, would you be so kind as to post it also to
news:alt.binaries.schematics.electronic ?
probably - what's a roob? Is that a reclusive noob?
 
On Wed, 05 Jan 2005 22:23:34 GMT, Rich Grise <richgrise@example.net> wrote:

On Wed, 05 Jan 2005 20:55:43 +0800, budgie wrote:

... I could send you a scan of the relevant
part from one type that we found surpisingly good when overdriven by a large
margin.

I've noticed that Mr. Woodgate said, "yes, please" - for the benefit of us
roobs, would you be so kind as to post it also to
news:alt.binaries.schematics.electronic ?
Arrrrgggghhhh! My news service doesn't carry binary groups.
 
I read in sci.electronics.design that John Larkin <john@spamless.usa>
wrote (in <dvept093udvtbifbflkkhn80hmtlfsjn9v@4ax.com>) 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.
--
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
 
On Wed, 05 Jan 2005 12:46:13 -0700, Jim Thompson wrote:

On Wed, 5 Jan 2005 18:58:48 +0000, John Woodgate
jmw@jmwa.demon.contraspam.yuk> wrote:

I read in sci.electronics.design that Jim Thompson
thegreatone@example.com> wrote (in <j97ot0l4l7facned07n4f7h091gpismk06@
4ax.com>) 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.

I've already run a simulation. Want to see it?

I'll show you yours if you'll show me mine. ;^j
--
The Pig Bladder From Uranus, Still Waiting for
Some Hot Babe to Ask What My Favorite Planet Is
..
 
John Woodgate <jmw@jmwa.demon.contraspam.yuk> wrote:
I read in sci.electronics.design that Rick <rik_nntp@dsl.pipex.com
wrote (in <Dq7Dd.34097$g4.634933@news2.nokia.com>) about 'Instantaneous
(analogue) compression of speech signals', on Thu, 6 Jan 2005:
John Woodgate <jmw@jmwa.demon.contraspam.yuk> wrote:
I read in sci.electronics.design that Rick <rik_nntp@dsl.pipex.com
wrote (in <0zTCd.34057$k4.653332@news1.nokia.com>) about 'Instantaneous
(analogue) compression of speech signals', on Wed, 5 Jan 2005:
John Woodgate <jmw@jmwa.demon.contraspam.yuk> wrote:
I read in sci.electronics.design that Rich Grise <richgrise@example.net
wrote (in <pan.2005.01.05.04.17.15.238991@example.net>) about
'Instantaneous (analogue) compression of speech signals', on Wed, 5 Jan
2005:

If a diode clipper is unsatisfactory, would a log amp do?

Not directly, because of the problem with negative-going half-cycles.

What about a "true" log amp (aka Log Video)? These have a sort of
S-shaped response that's symmetrical about 0V.

I wouldn't call that a true log amp. How is it done?

It's done by cascading a series of identical stages - each of which is
two long-tailed pairs with common load resistors and common inputs;
one LTP has a small tail current and is undegenerated (hence high
gain but limits quickly), the other is degenerated to unity gain and
doesn't run out of steam. The composite result of each LTP-pair is a
high gain up to a certain input voltage, and unity gain thereafter.


I think this is called a 'progressive overload' log-amp
Never heard it called that before. It's been called "True" since at
least 1980, when Plessey published their circuit in IEEE JSSC (Vol SC-15,
No.3), "A True Logarithmic Amplifier for Radar IF Applications.". Other
manufacturers such as Philips and Analog Devices also refer to this
circuit topology as "true log".

and gives a
piece-wise linear approximation to a log response. My HP audio wave
analyser has one. It requires numerous stages to get a reasonably
accurate log response.
Yes.

--
Rick
 
I read in sci.electronics.design that Ben Bradley <ben_nospam_bradley@mi
ndspring.com> wrote (in <itoqt0p51rnaj62l9j3lvssvmfrqntr987@4ax.com>)
about 'Instantaneous (analogue) compression of speech signals', on Thu,
6 Jan 2005:
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.
I'm not interested in an argument. I may not have commented on that
point, but it's an overall requirement, as shown by the word
'instantaneous' in the Subject line.
--
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
 
Subject: Re: Instantaneous (analogue) compression of speech signals
From: John Woodgate

I read in sci.electronics.design that John Larkin <john@spamless.usa

On Mon, 3 Jan 2005 21:21:14 +0000, John Woodgate

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.

National Semiconductor has a good audio agc circuit for their dual
transconductance amp (LM13700). How cheap is cheap for you? The next cheapest
thing I could think of would be to use a jfet as a voltage controlled resistor,
however for that to work without an amplifier you would need a fairly large
amplitude audio signal to begin with as the peaks would determine the amount of
resistance in the jfet.
 
I read in sci.electronics.design that Jim Thompson
<thegreatone@example.com> wrote (in <bv0rt0lhne8cu8t3kfm7h1hst8iej548dd@
4ax.com>) about 'Instantaneous (analogue) compression of speech
signals', on Thu, 6 Jan 2005:

Come on now John, are you saying you know what you want more than we do
?:)
How could I possibly be so presumptuous?
--
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
 
I read in sci.electronics.design that Ken Smith
<kensmith@green.rahul.net> wrote (in <crk21g$uh4$2@blue.rahul.net>)
about 'Instantaneous (analogue) compression of speech signals', on Thu,
6 Jan 2005:
Can you stand a little delay in the output signal?


Yes. In fact, if you could make two channels, one having 10 ms more
delay than the other, I could use that for another useful purpose.
--
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
 
In article <pan.2005.01.03.22.37.33.248842@example.net>,
Rich Grise <richgrise@example.net> writes:
On Mon, 03 Jan 2005 21:21:14 +0000, John Woodgate wrote:

Does anyone here have any experience of instantaneous (analogue)
compression (aka soft clipping) of speech signals?

Intentionally? ;-)

I've been doing a
little work on it but I'm unable to judge the resulting sound quality.

Well, isn't that a pretty good indication that your clipper isn't
introducing objectionable distortion? ;-) With speech, AIUI, you can do a

One thing about audio 'clipping' is that more clipping can be done if
intermod is controlled. So, one trick has been to modulate the audio
onto an SSB type carrier, and then clip it there, and demod it (that is
a long way to do it.) Another possibility is to chop up the spectrum,
and apply soft clipping to each chunk.

I did something similar by doing an fft of audio (music, in fact), and
by using a carefully chosen window function, I was able to do evil, nonlinear
processing of the FFT'd signal, and then to reverse FFT the result. With
the proper window, the result is difficult to distingush from the original,
except it is more level compressed. If you choose the wrong window function,
you'll get a buzz effect (because of the overlap of the FFTs isn't optimal.)

With the right method, you can really make audio (even music) much more
dense, yet it is still 'musical.' If you let too much intermod occur, then
it can get ugly and distorted sounding.

John
 
"John S. Dyson" wrote:
In article <pan.2005.01.03.22.37.33.248842@example.net>,
Rich Grise <richgrise@example.net> writes:
On Mon, 03 Jan 2005 21:21:14 +0000, John Woodgate wrote:

Does anyone here have any experience of instantaneous (analogue)
compression (aka soft clipping) of speech signals?

Intentionally? ;-)
Yes. It is standard practice in voice-grade two-way radio. In that application
distortion is less important than long range intelligibility -- so the trade-off
is made.

I've been doing a
little work on it but I'm unable to judge the resulting sound quality.

Well, isn't that a pretty good indication that your clipper isn't
introducing objectionable distortion? ;-) With speech, AIUI, you can do a

One thing about audio 'clipping' is that more clipping can be done if
intermod is controlled. So, one trick has been to modulate the audio
onto an SSB type carrier, and then clip it there, and demod it (that is
a long way to do it.)
I've built one of these and it does not eliminate the odd-order intermod. In
fact, that's the one thing it can't do. However, it totally eliminates both
even- and odd-order harmonics. For example if you do the dual tone test with
800 and 1000 Hz into the RF clipper, you'll get 600, 800, 1000, and 1200 Hz out
(due to third-order effects, higher order effects certainly likely at some
amplitude). However, 1600, 2400, 2000, and 3000 Hz harmonics will be absent.

For SSB radio, conventional clipping in the baseband requires extraordinary PEP
in the transmitter. Of course this is highly undesirable. (Do the Hilbert
transform of a square wave.) While increased intelligibility claims are true,
the best reason for the RF clipper is due to the high PEP required if
conventional and otherwise effective baseband clipping is used.

Craiglow and Werth patented a baseband version of the RF clipper for
Rockwell-Collins. It requires a Hilbert transformer. I wonder if the whole
thing would be better implemented in DSP these days. At any rate, "RF clippers"
eliminate harmonic distortion in instantaneous clippers.

Craiglow-Werth patent (expired):
http://www.freepatentsonline.com/patents/us/441/4410764/4410764.pdf
 
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>) 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.

http://www.portset.co.uk/compand.htm

I have concerns about the 'direct' mode, which is clearly non-linear!

http://www.toko.co.jp/products/ctlg/ic/com_compandor_e.htm

http://www.chipdocs.com/datasheets/datasheet-pdf/Philips-
Semiconductors/NE570.html

Not 'instantaneous' and data only in Japanese :)-(

http://ieeexplore.ieee.org/Xplore/Toclogin.jsp?url=/iel5/4/22551/0105081
4.pdf

Not accessible to me.

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.

Thanks for your help.
I thought you were the audio expert ?:)

What curve would you like and I'll create it in circuitry for you?

...Jim Thompson
--
| James E.Thompson, P.E. | mens |
| Analog Innovations, Inc. | et |
| Analog/Mixed-Signal ASIC's and Discrete Systems | manus |
| Phoenix, Arizona Voice:(480)460-2350 | |
| E-mail Address at Website Fax:(480)460-2142 | Brass Rat |
| http://www.analog-innovations.com | 1962 |

I love to cook with wine. Sometimes I even put it in the food.
 
I read in sci.electronics.design that Nico Coesel <nico@puntnl.niks>
wrote (in <41dc0655.1392136155@news.planet.nl>) about 'Instantaneous
(analogue) compression of speech signals', on Wed, 5 Jan 2005:
In that case, use a voltage controlled amplifier and control the voltage
by the filtering the output of a peak detector. I've used these circuits
many times with excellent results.
This is not instantaneous. There is inevitably a time-constant
associated with the rectifier filter capacitor. I, too, have used this
technique, but it isn't what I want for the present project.
--
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
 
On Thu, 06 Jan 2005 18:50:06 +0000, John Woodgate wrote:

I read in sci.electronics.design that Jim Thompson
thegreatone@example.com> wrote (in <bv0rt0lhne8cu8t3kfm7h1hst8iej548dd@
4ax.com>) about 'Instantaneous (analogue) compression of speech
signals', on Thu, 6 Jan 2005:

Come on now John, are you saying you know what you want more than we do
?:)

How could I possibly be so presumptuous?
Oh, not to worry! It's easy! ;-)

So far, I've gleaned that you don't want to use just plain ol' diodes, or
a log amp; I'm out of suggestions, but the thread is still a good read. :)

Good Luck!
Rich
 

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