"Pooh Bear" wrote
in message
Arny Krueger wrote:

The same applies to mic preamps. I think that all of the
mic preamps I have on hand are rather agressively
low-pass filtered in and about the inputs, and pretty
fast inside.

Perhaps you'd like to post an illustration of this
aggressive low-pass filtering then ?

Su

http://www.symetrixaudio.com/repository/202_1A0.pdf

"Pooh Bear" wrote
in message
Arny Krueger wrote:

"Pooh Bear"
wrote in message
Paul Stamler wrote:

"Bob Cain" wrote in message
...


Pooh Bear wrote:

When an amplifier is slew limiting, the internal
currents are solely committed to charging /
discharging internal capacitances and it's not
exactly in the linear operating region.

Exactly, and until it reaches just that limit it
operates linearly. It is linear until the point that
it isn't.

No, it ain't. Distortion, measured by high-frequency IM
or high-frequency THD, begins rising at well below the
point where signal slope = slew rate.


Indeed. Because you're pushing the 'drive' beyond the
optimally linear area of operation.

Same basic problem as plain old amplitude modulation
distortion, just with frequncy-sensitive components
added.

I get this feeling I'm the only person who has done
comprehensive measurements of a real-world power amp
circuit that was slew limiting.

What's the point measuring it ? I'm sure it'll be ugly.

Not that bad, except for high frequency nonlinear
distortion.

I simply design to avoid it.

I'll drink to that!

Harder to do back in the days when the silicon was far
slower, or even the devices were germanium.

"Pooh Bear" wrote
in message
Arny Krueger wrote:

"Pooh Bear"
wrote in message
Arny Krueger wrote:

"Les Cargill" wrote in message


I've been told, in non-audio contexts, that dV/dt
behaves more continuously than as a break in the line
slope. I was told that there was a critical region
of hysteresis to avoid.

I don't know if people are trying to talk about
slew-induced latch-up or what.

But, slew-induced latchup can be for real.

I've never seen anything that I'd understand as that.

Can you elaborate ?

It's simple - you feed a power amp with a fast-risetime
signal (a square wave) and it slew limits. When the
square wave flattens out, the stimulus for the slew
limiting is removed. It takes a little while for the amp
to recover its composure, and you get square wave
response that does not look like how the amp performs
when its operating linearly.

BTW Graham, with all the theory you're spouting, I have
to wonder how much time you've actually spent observing
real-world amps that are slew limiting.

I realize that real-world measurements mean nothing to
you after you dismissed that complete set of tests I
pointed you at the Crown amp.

You're quite a piece of work, Graham. If it doesn't fit
into your finite knowlege base, it doesn't exist!

In aus.hi-fi you posted a THD figure for a Crown
Macrotech of *0.0004%*.

I knew it was rubbish and I said so.

It appears you misquoted your own site. You should have
typed *0.004%* !

Phil's analysis is a bit lacking.

He presumes that hum is constant as power goes up. In fact
it tends to rise with higher power levels for a number of
reasons which you probably aleady know about.

Arny Krueger wrote:

"Pooh Bear" wrote
in message
Arny Krueger wrote:

The same applies to mic preamps. I think that all of the
mic preamps I have on hand are rather agressively
low-pass filtered in and about the inputs, and pretty
fast inside.

Perhaps you'd like to post an illustration of this
aggressive low-pass filtering then ?

Su

http://www.symetrixaudio.com/repository/202_1A0.pdf

I'll venture the opinion that those inductors are there for RFI
suppression. The associated capacitance isn't very big. Not
dissimilar to what I use myself.

I certainly don't see an aggressive LPF.


Graham

On Tue, 06 Sep 2005 10:45:58 +0100, Pooh Bear
wrote:

Isn't the elliptical
loadline from partially reactive loads (to individual
devices) while still in the "linear" pre-slewing region
equivalent to the RC circuits model?

A loadline analysis doesn't apply to an amplifier with serious nfb. The
loadline is used for open-loop analysis.

As usual on Usenet we're all talking together
about different topics. (If we were on the
same topic, it'd be YELLING. Arf.)

Or, as Les just recently said "My kingdom
for a whiteboard!"


I could make a very strong case if allowed to
introduce active devices with *no* charge mobility
issues and only a negligible electron transit
time as their only high frequency limitation.
They exist. But it wouldn't be The Cowboy Way.

Of course, I have been wrong before.

Use fets ? Not perfect but no carrier storage in the base region. That's
especially problematic when you have stages that are saturating.

Even worse, I was hinting at vacuum valves, where
"external" (which includes internal parasitic
reactances in this useage) capactitances completely
control high frequency behavior. Their higher intrinsic
impedances help my model by simplifying.

Starting from nearly ideal input transconductance
stages with well defined abilities to drive
compensation capacitors removes a layer of issues.
A good model can be made *before* feedback too.

As always, thanks for your thoughts,

Chris Hornbeck

On Tue, 6 Sep 2005 08:16:42 -0400, "Arny Krueger"
wrote:

I get this feeling I'm the only person who has done
comprehensive measurements of a real-world power amp circuit
that was slew limiting.

Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

Chris Hornbeck

Chris Hornbeck wrote:

On Tue, 6 Sep 2005 08:16:42 -0400, "Arny Krueger"
wrote:

I get this feeling I'm the only person who has done
comprehensive measurements of a real-world power amp circuit
that was slew limiting.

Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

Those beasts are massive. We had them at Neve. Replaced by AP System
Ones rapidly of course in 87/88.

Graham

On Wed, 07 Sep 2005 00:52:22 +0100, Pooh Bear
wrote:

Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

Those beasts are massive. We had them at Neve. Replaced by AP System
Ones rapidly of course in 87/88.

Very challenging to keep up with capacitor, and similar
failures these days, as my interest is failing.

Still, the ability to measure down to .002% (on a very
good day) and up to 100KHz (more or less, with a tailwind,)
gives at least a false confidence.

At my age, a false confidence is often the best. Arf.

Thanks, as always,

Chris Hornbeck

Don Pearce wrote:

On Mon, 05 Sep 2005 17:07:31 GMT, Les Cargill wrote:


I am probably using the word wrong. I am using
it to say that the function is not a straight
line with a break, then another flat line. I am
using "hysteresis" to describe the curve that
replaces the transition where the discontinuous
line break would be.

http://www.phys.ualberta.ca/~gingric...s/node115.html

I'm using "hysteresis" to describe a "soft kneee"
in the curve. Maybe that's a better term? Sure
isn't a fifty dollar word; so that's better.



Thanks, as always,

Chris Hornbeck


"Soft knee" is a good word - stick with that. When you say hysteresis it
means that the curve doesn't get re-traced on the way back down again, but
some different curve is followed. In other words, something somewhere
sticks before it retraces.

d


Geez, I knew that... *cringes*. Thankx!

--
Les Cargill

Some Pommy Dickhead

In aus.hi-fi you posted a THD figure for a Crown Macrotech of
*0.0004%*.

I knew it was rubbish and I said so.

It appears you misquoted your own site. You should have typed
*0.004%* !


** It is far, far worse than that !!!

Arny has screwed up his plots and figures completely !!!

The 400 watt resistive result is really 0.015 % as shown on the plot
itself - not 0.005% !!!!

There simply is no result for 400 watt with simulated speaker load - the
1watt or 10 watt plot has been repeated.





........... Phil

"Arny Krueger"

That's entirely compatible with the distortion spectra shown at:

http://www.pcavtech.com/pwramp/macro.../SNR_010WS.gif

Note that the % shown there which was software-generated, is compatible
with the number I typed in the table.

BTW, you can click any of the specs in the table and it takes you to the
corresponding FFT plot.


** When I click on the 400 watt resistive case - the THD figure is 0.015 %.

Not 0.005% as in your table ????????
---------------------------------------------


The plot for the 1 watt test shows 60 Hz hum at 94 dB down.

The plot for the 10 watt test shows 60 Hz hum at 89 dB down ????

The plot for the 400 watt speaker load test shows 60 Hz hum at 94 dB down
again ??


Clearly, the plots and data are jumbled up !!!!!!!!
--------------------------------------------------------


The 400 watt speaker load plot is a repeat of the 1 watt plot - explaining
the silly low figure.

There simply is NO plot for the 400 watt, speaker load condition.

If there were, the result should be about 0.01% THD.


ALSO:

When I click on the 19/20 kHz two tone for 400 watt resistive the figure is
0.026%

Not 0.0026 % as in the table ??



What a PITA you are - Arny.





............ Phil

"Phil Allison"
Some Pommy Dickhead

In aus.hi-fi you posted a THD figure for a Crown Macrotech of
*0.0004%*.

I knew it was rubbish and I said so.

It appears you misquoted your own site. You should have typed
*0.004%* !


** It is far, far worse than that !!!

Arny has screwed up his plots and figures completely !!!

The 400 watt resistive result is really 0.015 % as shown on the plot
itself - not 0.005% !!!!

There simply is no result for 400 watt with simulated speaker load - the
1watt or 10 watt plot has been repeated.



** Use this page and try clicking on the results boxes one by one:

http://www.pcavtech.com/pwramp/macrot-5000VZ/index.htm


** When I click on the 400 watt resistive case - the THD figure is 0.015 %.

Not 0.005% as in your table ????????
---------------------------------------------


The plot for the 1 watt test shows 60 Hz hum at 94 dB down.

The plot for the 10 watt test shows 60 Hz hum at 89 dB down ????

The plot for the 400 watt speaker load test shows 60 Hz hum at 94 dB down
again ??


Clearly, the plots and data are jumbled up !!!!!!!!
--------------------------------------------------------


The 400 watt speaker load plot is a repeat of the 1 watt plot - explaining
the silly low figure.

There simply is NO plot for the 400 watt, speaker load condition.

If there were, the result should be about 0.01% THD.


ALSO:

When I click on the 19/20 kHz two tone for 400 watt resistive the figure is
0.026%

Not 0.0026 % as in the table ??


What a PITA you are - Arny.




............ Phil









.......... Phil

Looks like my ISPs News server is stuffed again .....

"Alan Rutlidge" wrote in message
...

"Richard Freeman" wrote in message
...

"Alan Rutlidge" wrote in message
...

"Don Pearce" wrote in message
...
On Fri, 2 Sep 2005 15:27:47 +0800, Alan Rutlidge wrote:

"Richard Freeman" wrote in message
...

"Alan Rutlidge" wrote in message
...

"Don Pearce" wrote in message
.. .
On Thu, 1 Sep 2005 17:25:01 +1000, Phil Allison wrote:



- - - - - - -- - - further snipped for brevity - - - - - - - - - - -


loss into their transmission networks to counteract the effects of
instability due to impedance mismatches.

you do not have to build loss into Telephone networks - it is there
anyway -

Not entirely true. Most of the US Inter Exchange Network (IEN) [digital]
is designed to have a nominal 0dB loss exchange to exchange. The only
overall transmission losses occur on the lines between the exchange and
the customers.

True however there is loss (theoretically 6dB - in reality greater than
6dB)
through the Hybrid Transformer - I had meant to flesh this concept out
further yesterday but inadvertantly hit send before I had finished
following
this idea/posting further.

Even before the advent of digital transmission and switching technology,
the old Strowger exchange had a fraction of a dB loss through the exchange
on a local call. IEN echo however was not a problem as it was a two wire
circuit through the exchange and the cable network to the telephone.

so are you starting to suspect that the Hyrbrid has something to do
with the
echo then and not cable reflections ?

it is more a case of keeping your gain down to make sure you do not get
howling due to feedback through the hybrid Transformers at each end -

In a closed 4 wire IEN circuit losses of approximately 7dB occur across
the transformer hybrids at each end of the 4 wire transmission path even
under the worst possible mismatch conditions. This effectively provides a
total of 14dB loss to the singing loop.

Correct - the loop will not be singing if it merely consists of two
sets of
Hybrids - ie no gain in the Transmission system - as per my earlier
comment
that stability is not created by padding a system down (as the basic
system
already contains sufficent loss) but rather about keeping total gain to
a
minimum.

Therefore, provided the total gains in the singing loop don't exceed the
total losses, the circuit will remain unconditionally stable. Any
difference in favour of the losses over the gains in the singing loop is
known as the Stability Margin.
Example : If the losses total 14dB (trans-hybid losses as the worst
possible condition) and the gain only 1dB in each direction of
transmission to overcome adjacent port losses in the transformer hybrids
(total 2dB gain in the singing loop), therefo- 14dB (loss) - 2dB (gain)
= 12dB stability margin. In simple terms the closed 4 wire loop can never
become unstable.

Exactly the basic system - with no added gain - has a total (using your

figures) of 14dB of loss built in and is by its very nature stable (any

signs of instability in this system would be cause for celebration by
physiscts around the world).

In Australia, the loss from 2 wire appearance at the MDF of the exchange
to the same at the other end is designed at 6dB for each direction of
voice transmission. A nice idea, as the echo level is reduced by a
factor of 2 times the loss of the link in the network, resulting in a
minimum stability margin of at least 12dB even under the worst possible
conditions.

Lets see - over 6dB (theoretical) loss each way through the hybrid ....
That sounds about right.

Some other overseas networks aren't anywhere near as good as ours when
it comes to echo performance. In fact a well designed and impedance
matched network requires little or no echo cancellation equipment,
resulting in a clearer network to talk over and minimal VF data
transmission / fax transmission problems.

Hmmm? what about Side tone ? how do these well designed networks provide
Sidetone ?

The sidetone is developed within the telephone. Older phones (pre the
T200 / T400 series) used what was known as an Anti-SideTone Induction Coil
(ASTIC) which is a purposely leaky hybrid, was designed to feedback a
small amount of the speech energy from the transmitter (microphone) to the
receiver. This sidetone is purposefully locally introduced to make the
caller think the phone was "working okay."

The Sidetone was there already as a function of the Hybrid in the
phone, the
ASTIC was wired to cancel the sidetone further than a simple Hybrid did
-
This was only possible as the level (not really the impedance except
where/as
it affected the overall gain) of the signal between the hybrid/Astic
and
the receiver was both known and constant.- In a side note the ASTIC was

actually introduced (around 1939 I believe in Australia) to encourage
people to talk louder into Telephones as it provided less Sidetone than
the normal Hybrid Transformer had previously.
In reality since it is not really impedance mismatches that stop us
supressing Hybrid leakage but rather the fact that we do not know the
overall gain of a System (and hence the exact Signal level we need to
cancel out) - nor does that gain remain constant - largely due
to line length variations etc I would have to argue that in a real
world telephone network no matter how well designed is it not possible
to provide an echo free service without either a VSA (Voice Switched
Amplifier the old method
of echo supression which basically gated the signal in one direction at
a
time) or DSP based echo supression.

The level of sidetone is critical. Too much and the talker will speak
softly (thinking the other party can hear him /her loud enough). Another
problem is background noise (if loud enough) picked up by the transmitter
can tend to drown out incoming speech and make the speech unintelligible.
Conversly, too low a level or absence of sidetone tends to cause the user
to speak too loudly in the false belief the distant party can't hear them.
If they shout loud enough they could overload the A/D converter in the
exchange causing distortion.

ideal Sidetone is considered to be the level we are used to hearing
when we
speak (sorry I don't recall the figure off hand)

BTW. The modern telephone still achieves sidetone, but instead of a bulky
ASTIC, it is achieved with semiconductor technology.

again my apologies I had meant to follow these ideas further before
posting -
However even well designed pure digital Transmission systems require
echo
suppression when the round trip delay exceeds a certain amount of time
(IEC
consider this to be 36mS-
http://www.iec.org/online/tutorials/...cho_cancel.pdf - which is
actually an excellent tutorial on the whole subject and one that I
recommend
highly ) due to the leakage through the hybrid. If rtd is kept below
36mS
echo does still occur however it is heard by the person speaking as
part of
the sidetone (a variant of this effect called 'double tracking' is an
effect
often used in recordings and live concerts to give vocals more power -
but I
digress .....) over 36mS however this starts being noticed as a
separate
echo and becomes a problem.
I suspect that your 'well designed network' is merely one in which no
call
has an RTD of over 36mS. Unfortunately due to the laws of physics this
precludes networks which have paths of over 5,700 Kms (In Australia we
approach that limit before we even consider processes such as those
within
the codec etc which cause additional delay). Of course when/if you add
a
Satellite to the equation you also add 50,000 Kms of path or 166mS and
a
very noticeable echo - I believe (but not working there am unable to
confirm) that India makes (or maybe made) extensive use of satellite
technology as a (relatively) cheap way of providing telecommunications.

- Alternatively ISDN type connections do not naturally have Sidetone or

suffer from echo when both ends terminate on ISDN (try any call through
a
tester such as an IBT 1A) - however this is due to the fact that there
is in
fact no hybrid in such a system - anywhere! TX and RX are maintained as

completely isolated paths through the entire network. Sidetone for ISDN

phones is added deliberately by the handset manufacturer.

Bottom line is (and getting back to the point of the thread - at least
the
point where I joined) echo is not caused by cable reflections but
rather by
Hybrid leakage - which is often incorrectly called a 'reflection - yes
cable
reflections do occur and this is put to good use by the pulse echo
tester -
however they do not occur at levels sufficient to cause noticeable
problems
at VF (Voice Frequency).

regards
Richard Freeman

"Pooh Bear" wrote
in message
Arny Krueger wrote:

"Pooh Bear"
wrote in message
Arny Krueger wrote:

The same applies to mic preamps. I think that all of
the mic preamps I have on hand are rather agressively
low-pass filtered in and about the inputs, and pretty
fast inside.

Perhaps you'd like to post an illustration of this
aggressive low-pass filtering then ?

Su

http://www.symetrixaudio.com/repository/202_1A0.pdf

I'll venture the opinion that those inductors are there
for RFI suppression.

No doubt.

The associated capacitance isn't
very big. Not dissimilar to what I use myself.

Where's the corner frequency and how does that compare to
the slewing capability of the op amp involved?

I certainly don't see an aggressive LPF.

No comment.

"Chris Hornbeck" wrote in
message
On Tue, 6 Sep 2005 08:16:42 -0400, "Arny Krueger"
wrote:

I get this feeling I'm the only person who has done
comprehensive measurements of a real-world power amp
circuit that was slew limiting.

Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

I have long owned a highly modified Heath THD analyzer with
residual under 0.01%. It fell into complete disuse when I
figured out how to get down to 0.001% and then 0.00015% with
computer audio interfaces and software.

So, what do you measure that's relevant to this discusison
of slew rates?

"Phil Allison" wrote in message




The plot for the 1 watt test shows 60 Hz hum at 94 dB
down.
The plot for the 10 watt test shows 60 Hz hum at 89 dB
down ????
The plot for the 400 watt speaker load test shows 60 Hz
hum at 94 dB down again ??


Clearly, the plots and data are jumbled up !!!!!!!!
--------------------------------------------------------

Not a chance.

First off Phil, you need to learn how to write clearly.
There are something like 14 plots at

http://www.pcavtech.com/pwramp/macrot-5000VZ/index.htm that
show data for 60 Hz, at least 3 for each power level you
mention.

So, "the plot for the 1 watt test" is one of no less than
three plots. The plots have URLs and legends but I guess
that fact has escaped you.

Secondly, you seem to take exception to the data shown. That
means that you must know what the right answers are, and be
able to explain why. I see no such thing.

So Phil when you learn how to write a usable informal
technical report, I'll deal with your complaints.

"Arny Krueger"

"Phil Allison"
http://www.pcavtech.com/pwramp/macrot-5000VZ/index.htm
Note that the % shown there which was software-generated, is compatible
with the number I typed in the table.

BTW, you can click any of the specs in the table and it takes you to the
corresponding FFT plot.


** When I click on the 400 watt resistive case - the THD figure is 0.015 %.

Not 0.005% as in your table ????????
---------------------------------------------


The plot for the 1 watt, 1kHz THD test shows 60 Hz hum at 94 dB down.

The plot for the 10 watt 1 kHz, THD test shows 60 Hz hum at 89 dB down
????

The plot for the 400 watt, 1kHz THD speaker load test shows 60 Hz hum at
94 dB down
again ??

Clearly, the plots and data are jumbled up !!!!!!!!
--------------------------------------------------------


The 400 watt, THD speaker load plot is a repeat of the 1 watt plot -
explaining
the silly low figure of 0.0006% THD.

There simply is NO real plot for the 400 watt THD, speaker load condition.

If there were, the result should be about 0.01% THD.

Arny has really ****ED UP BADLY.
-----------------------------------------------------------


ALSO:

When I click on the 19/20 kHz two tone IM test for 400 watt resistive the
figure is
0.026%

Not 0.0026 % as in the table ??
------------------------------------

ALSO :

The 20 Hz , 1 watt & 10 watt THD table figures are WRONG as well.



What a ****ING, LYING , PITA is Arny.

Now he make it worse by trying to BULL**** his way out.







............ Phil

Arny Krueger wrote:

"Pooh Bear" wrote
in message
Arny Krueger wrote:

"Pooh Bear"
wrote in message
Arny Krueger wrote:

The same applies to mic preamps. I think that all of
the mic preamps I have on hand are rather agressively
low-pass filtered in and about the inputs, and pretty
fast inside.

Perhaps you'd like to post an illustration of this
aggressive low-pass filtering then ?

Su

http://www.symetrixaudio.com/repository/202_1A0.pdf

I'll venture the opinion that those inductors are there
for RFI suppression.

No doubt.

The associated capacitance isn't
very big. Not dissimilar to what I use myself.

Where's the corner frequency and how does that compare to
the slewing capability of the op amp involved?

I certainly don't see an aggressive LPF.

No comment.

I'll comment though !

The link you posted showed apparently individual inductors.

This actually isn't the best arrangement since it rolls of the audio
signal as much as any RFI.

You actually got me thinking about this and I put my collected
thoughts on the subject together and wound my own RFI choke for a mic
input.

You can find it at alt.binaries.schematics.electronic

RFI is common mode but the wanted audio signal is differential.

If you wind a toroid bifilar wise you get no net inductance for the
differential audio path but you do get inductance for the common mode
RFI. The same principle is used for EMI filters in power supplies.

I did some quick tests. There was no influence on the differential (
audio ) path even up at 100kHz. Not a sausage !

In comparison, 1MHz common mode was attenuated by 30dB. I should be
able to improve on this. My predecessors' mucking about with ferrite
beads look rather lame in comparison.

Graham

On Wed, 7 Sep 2005 07:55:22 -0400, "Arny Krueger"
wrote:

Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

I have long owned a highly modified Heath THD analyzer with
residual under 0.01%. It fell into complete disuse when I
figured out how to get down to 0.001% and then 0.00015% with
computer audio interfaces and software.

So, what do you measure that's relevant to this discusison
of slew rates?

Please forgive me for not responding earlier. I'd
mistaken your post as rhetorical, or perhaps harsh.
In proper perspective, neither need apply; sorry.

The antique analog hardware certainly has its
limitations, and worse, its creakyness, but also
has some residual strengths. What common digital
computer interface can measure the harmonics of
a 100KHz signal? Or, really, accurately, of any
supersonic signal?

Discussions of slew rates are partly semantic (about
which you and Paul have won me over) and partly
gut-level-design-oriented. Being able to see the
open loop performance of things I'm interested in
helps me. 'S all I'm sayin'. Don't mean I'm either
smart or right, 'cause neither's true.

Thanks, as alwyas,

Chris Hornbeck
"But it's the almostness of Godard's films that makes it special;
if it were too perfect, it would be mechanized and dull. Instead
of dancing, it would be choreography, an applied science."
-rcraig62 commenting on _Bande a part_, 1964

"Chris Hornbeck" wrote in
message
On Wed, 7 Sep 2005 07:55:22 -0400, "Arny Krueger"
wrote:

Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

I have long owned a highly modified Heath THD analyzer
with residual under 0.01%. It fell into complete disuse
when I figured out how to get down to 0.001% and then
0.00015% with computer audio interfaces and software.

So, what do you measure that's relevant to this
discusison of slew rates?

Please forgive me for not responding earlier. I'd
mistaken your post as rhetorical, or perhaps harsh.
In proper perspective, neither need apply; sorry.

The antique analog hardware certainly has its
limitations, and worse, its creakyness, but also
has some residual strengths. What common digital
computer interface can measure the harmonics of
a 100KHz signal? Or, really, accurately, of any
supersonic signal?

To me ultrasonic starts at 16 Khz, so even a M-Audio
Audiophile 24/192 is interesting.

But for the heavy stuff check National Instruments.

http://sine.ni.com/nips/cds/view/p/lang/en/nid/201603

Seems like this interface does 24 bits up to 500 KHz and 16
bits up to 15 MHz.

AFAIK, some of their digitizers cost less than a LynxTWO,
but the one described above is $6K-12K, depending on
features.

Phil Allison wrote:
Graham Stevenon


** Warning do not work on a piles of **** like you.

You are just pure bloody evil.

Go get bowel cancer.



Ah, this list is yet again living up to it's "pro" designation. Wishing
cancer on anyone is disgusting and beyond words. Ever lose a loved one
to cancer? I did, last month. It isn't pretty, and not something I
would ever wish on my worst enemy. Both of you, take it elsewhere and
grow up.

On Fri, 9 Sep 2005 05:10:50 -0400, "Arny Krueger"
wrote:

To me ultrasonic starts at 16 Khz, so even a M-Audio
Audiophile 24/192 is interesting.

Sure, accurate and quiet, too. But it can't
measure harmonic distortion at 100KHz.

But for the heavy stuff check National Instruments.

http://sine.ni.com/nips/cds/view/p/lang/en/nid/201603

Seems like this interface does 24 bits up to 500 KHz and 16
bits up to 15 MHz.

I think it's KS/s and MS/s, but that's still plenty for
our purposes.

AFAIK, some of their digitizers cost less than a LynxTWO,
but the one described above is $6K-12K, depending on
features.

Probably about what a Sound Tech would have to cost
nowdays. Things do get better, on average.

Thanks, as always,

Chris Hornbeck
"But it's the almostness of Godard's films that makes it special;
if it were too perfect, it would be mechanized and dull. Instead
of dancing, it would be choreography, an applied science."
-rcraig62 commenting on _Bande a part_, 1964

"Chris Hornbeck" wrote in
message
On Fri, 9 Sep 2005 05:10:50 -0400, "Arny Krueger"
wrote:

To me ultrasonic starts at 16 Khz, so even a M-Audio
Audiophile 24/192 is interesting.

Sure, accurate and quiet, too. But it can't
measure harmonic distortion at 100KHz.

But for the heavy stuff check National Instruments.

http://sine.ni.com/nips/cds/view/p/lang/en/nid/201603

Seems like this interface does 24 bits up to 500 KHz and
16 bits up to 15 MHz.

I think it's KS/s and MS/s, but that's still plenty for
our purposes.

Well, generally one tests one channel at a time.

AFAIK, some of their digitizers cost less than a LynxTWO,
but the one described above is $6K-12K, depending on
features.

Probably about what a Sound Tech would have to cost
nowdays. Things do get better, on average.

I think the new ones are about 3 times the upper dollar
figure, above.

"Richard" wrote in message
oups.com...
Looks like my ISPs News server is stuffed again .....

"Alan Rutlidge" wrote in message
...

"Richard Freeman" wrote in message
...

"Alan Rutlidge" wrote in message
...

"Don Pearce" wrote in message
...
On Fri, 2 Sep 2005 15:27:47 +0800, Alan Rutlidge wrote:

"Richard Freeman" wrote in message
...

"Alan Rutlidge" wrote in
message
...

"Don Pearce" wrote in message
.. .
On Thu, 1 Sep 2005 17:25:01 +1000, Phil Allison wrote:



- - - - - - -- - - further snipped for brevity - - - - - - - - - - -


loss into their transmission networks to counteract the effects of
instability due to impedance mismatches.

you do not have to build loss into Telephone networks - it is there
anyway -

Not entirely true. Most of the US Inter Exchange Network (IEN) [digital]
is designed to have a nominal 0dB loss exchange to exchange. The only
overall transmission losses occur on the lines between the exchange and
the customers.

True however there is loss (theoretically 6dB - in reality greater than
6dB)
through the Hybrid Transformer - I had meant to flesh this concept out
further yesterday but inadvertantly hit send before I had finished
following
this idea/posting further.

Trans hybrid loss in a two transformer design hybrid (4 Wire Hybrid In port
to 4 Wire Hiybrid Out port) is closer to 7dB loss where the impedance of the
termination connected to the 2 Wire Line port is an infinite mismatch to
that of the impedance of the network connected to the Balance Network port
(ie.e an infinite : 1 ratio). I have measured slightly less (6.7dB) in
proactice on some hybrids, but 7dB is closer to the norm.

The 7dB figure is derived from the fact that a minimum of 3dB loss will
occur between adjacent ports. Add a little extra (say 0.5dB) for
transformer losses and a practical adjacent port loss closely approaches
3.5dB. 3.5dB + 3.5dB = 7dB. A diagram would be easier to illustrate, but
as this is a text only NG, posting a pic is not possible.


Even before the advent of digital transmission and switching technology,
the old Strowger exchange had a fraction of a dB loss through the
exchange
on a local call. IEN echo however was not a problem as it was a two wire
circuit through the exchange and the cable network to the telephone.

so are you starting to suspect that the Hyrbrid has something to do
with the
echo then and not cable reflections ?

No. Albeit that hybrid mismatch is a major contributor to echo in the 4
wire transmission path, it is not the sole cause. Mismatches due to cable
gauge changes onn a simple 2 wire circuit will produce an echo (reflection)
due to impedance mismatch at the point they join. Echo and signal
reflection are one of the same thing. Just we associate echo with long
return path delays of 35mS or more. It's purely an auditory perception
thing as delays less than 28mS are very hard for the human brain to
recognise.


it is more a case of keeping your gain down to make sure you do not get
howling due to feedback through the hybrid Transformers at each end -

In a closed 4 wire IEN circuit losses of approximately 7dB occur across
the transformer hybrids at each end of the 4 wire transmission path even
under the worst possible mismatch conditions. This effectively provides
a
total of 14dB loss to the singing loop.

Correct - the loop will not be singing if it merely consists of two
sets of
Hybrids - ie no gain in the Transmission system - as per my earlier
comment
that stability is not created by padding a system down (as the basic
system
already contains sufficent loss) but rather about keeping total gain to
a
minimum.

It would be pointless creating a 4 wire transmission system containing
hybrids if no active transmission components or ADC / DACs were involved.


Therefore, provided the total gains in the singing loop don't exceed the
total losses, the circuit will remain unconditionally stable. Any
difference in favour of the losses over the gains in the singing loop is
known as the Stability Margin.
Example : If the losses total 14dB (trans-hybid losses as the worst
possible condition) and the gain only 1dB in each direction of
transmission to overcome adjacent port losses in the transformer hybrids
(total 2dB gain in the singing loop), therefo- 14dB (loss) - 2dB
(gain)
= 12dB stability margin. In simple terms the closed 4 wire loop can
never
become unstable.

Exactly the basic system - with no added gain - has a total (using your

figures) of 14dB of loss built in and is by its very nature stable (any

signs of instability in this system would be cause for celebration by
physiscts around the world).

Agreed. If losses exceed gains the circuit must be unconditionally stable.


In Australia, the loss from 2 wire appearance at the MDF of the
exchange
to the same at the other end is designed at 6dB for each direction of
voice transmission. A nice idea, as the echo level is reduced by a
factor of 2 times the loss of the link in the network, resulting in a
minimum stability margin of at least 12dB even under the worst possible
conditions.

Lets see - over 6dB (theoretical) loss each way through the hybrid ....
That sounds about right.

Yes. Especially if transformer hybrids are used. The adjacent port loss in
each hybrid is very close to 3.5dB. As there is a hybrid at each end of the
link, the minimum losses for each direction of transmission would be close
to 7dB.


Some other overseas networks aren't anywhere near as good as ours when
it comes to echo performance. In fact a well designed and impedance
matched network requires little or no echo cancellation equipment,
resulting in a clearer network to talk over and minimal VF data
transmission / fax transmission problems.

Hmmm? what about Side tone ? how do these well designed networks provide
Sidetone ?

The sidetone is developed within the telephone. Older phones (pre the
T200 / T400 series) used what was known as an Anti-SideTone Induction
Coil
(ASTIC) which is a purposely leaky hybrid, was designed to feedback a
small amount of the speech energy from the transmitter (microphone) to
the
receiver. This sidetone is purposefully locally introduced to make the
caller think the phone was "working okay."

The Sidetone was there already as a function of the Hybrid in the
phone, the
ASTIC was wired to cancel the sidetone further than a simple Hybrid did

Once again a circuit diagram would assist in the explanation. A very simple
circuit of t phone would have the transmitter and the receiver in series.
The sidetone level on the receiver would be very high.. The idea of the
ASTIC is to reduce the level of the speech current generated by the
transmitter reaching the receiver, whilst maximising transmitter signal to
line and also maximising the received line incoming speech signal reaching
the receiver. ASTICs are usually single transformer leaky hybrids. Good
examples are to be found in the old Telecom 800 series phones.

-
This was only possible as the level (not really the impedance except
where/as
it affected the overall gain) of the signal between the hybrid/Astic
and
the receiver was both known and constant.

In the old phones (say 800 series and earlier) there was no gain, except in
the hearing aid version of the 800 series which featured a volume control
where the recall button is usually located.

- In a side note the ASTIC was

actually introduced (around 1939 I believe in Australia) to encourage
people to talk louder into Telephones as it provided less Sidetone than
the normal Hybrid Transformer had previously.

As I said previously. Early phones had no ASTIC and sufferred from very
high sidetone. This high sidetone level caused people to talk softer
because they believed the other party could hear them okay based purely on
the effect of the local sidetone level. This became an even bigger problem
on long distance trunk calls which suffered significantly more transmission
loss than local calls.

In reality since it is not really impedance mismatches that stop us
supressing Hybrid leakage but rather the fact that we do not know the
overall gain of a System (and hence the exact Signal level we need to
cancel out) - nor does that gain remain constant - largely due
to line length variations etc I would have to argue that in a real
world telephone network no matter how well designed is it not possible
to provide an echo free service without either a VSA (Voice Switched
Amplifier the old method
of echo supression which basically gated the signal in one direction at
a
time) or DSP based echo supression.

I beg to differ. Getting impedance matching and hybrid balancing correct
negates the requirement for echo cancellation in either the digital or
analogue domains. This is immediately apparent on facsimile and data calls
through the PSTN (not ISDN) where echo cancellation can't be used. To get
reasonable error free data throughput through the PSTN, the echo performance
of the transmission path must be reasonably good to begin with.


The level of sidetone is critical. Too much and the talker will speak
softly (thinking the other party can hear him /her loud enough). Another
problem is background noise (if loud enough) picked up by the transmitter
can tend to drown out incoming speech and make the speech unintelligible.
Conversly, too low a level or absence of sidetone tends to cause the user
to speak too loudly in the false belief the distant party can't hear
them.
If they shout loud enough they could overload the A/D converter in the
exchange causing distortion.

ideal Sidetone is considered to be the level we are used to hearing
when we
speak (sorry I don't recall the figure off hand)

True. When we hold a telephone handset to our head, we cut off some of the
natural sidetone we would normally experience in non telephone coversation.
This means that we need to replace some of that lost natural sidetone with
an equivalent in the telephone. The correct level of sidetone effectively
regulates how loud we speak into the telephone. Ever noticed how spoken but
profoundly deaf people speak - quite often too loud or too soft.

The correct sidetone level in a phone should be approximately 13B below the
speaker's transmission level (local end).


BTW. The modern telephone still achieves sidetone, but instead of a
bulky
ASTIC, it is achieved with semiconductor technology.

again my apologies I had meant to follow these ideas further before
posting -
However even well designed pure digital Transmission systems require
echo
suppression when the round trip delay exceeds a certain amount of time
(IEC
consider this to be 36mS-
http://www.iec.org/online/tutorials/...cho_cancel.pdf - which is
actually an excellent tutorial on the whole subject and one that I
recommend
highly ) due to the leakage through the hybrid. If rtd is kept below
36mS
echo does still occur however it is heard by the person speaking as
part of
the sidetone (a variant of this effect called 'double tracking' is an
effect
often used in recordings and live concerts to give vocals more power -
but I
digress .....) over 36mS however this starts being noticed as a
separate
echo and becomes a problem.
I suspect that your 'well designed network' is merely one in which no
call
has an RTD of over 36mS. Unfortunately due to the laws of physics this
precludes networks which have paths of over 5,700 Kms (In Australia we
approach that limit before we even consider processes such as those
within
the codec etc which cause additional delay). Of course when/if you add
a
Satellite to the equation you also add 50,000 Kms of path or 166mS and
a
very noticeable echo - I believe (but not working there am unable to
confirm) that India makes (or maybe made) extensive use of satellite
technology as a (relatively) cheap way of providing telecommunications.

- Alternatively ISDN type connections do not naturally have Sidetone or

suffer from echo when both ends terminate on ISDN (try any call through
a
tester such as an IBT 1A) - however this is due to the fact that there
is in
fact no hybrid in such a system - anywhere! TX and RX are maintained as

completely isolated paths through the entire network. Sidetone for ISDN

phones is added deliberately by the handset manufacturer.

Bottom line is (and getting back to the point of the thread - at least
the
point where I joined) echo is not caused by cable reflections but
rather by
Hybrid leakage - which is often incorrectly called a 'reflection - yes
cable
reflections do occur and this is put to good use by the pulse echo
tester -
however they do not occur at levels sufficient to cause noticeable
problems
at VF (Voice Frequency).

The IEC tutorial takes a very simplisitic approach to the issues surrpunding
echo cancellation in typical telephone networks. It completely ignores the
issue of VF data (fax and modem calls) through the PSTN and how echo
cancellors are supposed to handle such calls.

Furthermore the document dips to mediocrity with the first sentence on Page
5 which reads " Unfortunately, the hybrid is by nature a leaky device."
What utter crap. I've measured return losses in old 1954 two transformer
hybrids which exceed 60dB! A million to 1 times power isolation is nothing
to be sneezed at. Hybrids, when correctly balanced are supposed to provide
isolation between the 4 wire transmission paths - not leakage.

You've obviously never conducted a Near End return loss measurement on a
customer's telephone line with an EDL423 Network Transmission Quality
Tester. Run a transmission test on a loaded cable (nomimal Z = 1200 ohms)
where it interfaces to a LIB7 LI in an AXE exchange (Zin = 600 ohms fixed).
Where the cable interfaces into the exchange there is a 2 : 1 impedance
mismatch. Even though the mismatch only contributes to 0.5dB additional
forward transmission loss, the return loss at this point is a mere 9.2dB - a
mismatch in anyone's book, with more than 10% of the transmitted signal
reaching this point being reflected back to the source. The only thing
reducing this reflected signal back to the customer is the transmission loss
of the cable itself which by Australian standards must be less than 6.5dB @
820Hz.


Cheers,
Alan



regards
Richard Freeman

Pooh Bear wrote:
Arny Krueger wrote:


"Pooh Bear" wrote
in message

Arny Krueger wrote:


"Pooh Bear"
wrote in message

Arny Krueger wrote:


The same applies to mic preamps. I think that all of
the mic preamps I have on hand are rather agressively
low-pass filtered in and about the inputs, and pretty
fast inside.

Perhaps you'd like to post an illustration of this
aggressive low-pass filtering then ?

Su

http://www.symetrixaudio.com/repository/202_1A0.pdf

I'll venture the opinion that those inductors are there
for RFI suppression.

No doubt.


The associated capacitance isn't
very big. Not dissimilar to what I use myself.

Where's the corner frequency and how does that compare to
the slewing capability of the op amp involved?


I certainly don't see an aggressive LPF.

No comment.


I'll comment though !

The link you posted showed apparently individual inductors.

This actually isn't the best arrangement since it rolls of the audio
signal as much as any RFI.

You actually got me thinking about this and I put my collected
thoughts on the subject together and wound my own RFI choke for a mic
input.

You can find it at alt.binaries.schematics.electronic

RFI is common mode but the wanted audio signal is differential.

If you wind a toroid bifilar wise you get no net inductance for the
differential audio path but you do get inductance for the common mode
RFI. The same principle is used for EMI filters in power supplies.

I did some quick tests. There was no influence on the differential (
audio ) path even up at 100kHz. Not a sausage !

In comparison, 1MHz common mode was attenuated by 30dB. I should be
able to improve on this. My predecessors' mucking about with ferrite
beads look rather lame in comparison.

Graham


Gotta be blunt here, but are you really that far behind Graham? Arrrghh.

Arny Krueger wrote:

"Chris Hornbeck" wrote in
message

On Wed, 7 Sep 2005 07:55:22 -0400, "Arny Krueger"
wrote:


Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

I have long owned a highly modified Heath THD analyzer
with residual under 0.01%. It fell into complete disuse
when I figured out how to get down to 0.001% and then
0.00015% with computer audio interfaces and software.

So, what do you measure that's relevant to this
discusison of slew rates?

Please forgive me for not responding earlier. I'd
mistaken your post as rhetorical, or perhaps harsh.
In proper perspective, neither need apply; sorry.

The antique analog hardware certainly has its
limitations, and worse, its creakyness, but also
has some residual strengths. What common digital
computer interface can measure the harmonics of
a 100KHz signal? Or, really, accurately, of any
supersonic signal?


To me ultrasonic starts at 16 Khz, so even a M-Audio
Audiophile 24/192 is interesting.

But for the heavy stuff check National Instruments.

http://sine.ni.com/nips/cds/view/p/lang/en/nid/201603

Seems like this interface does 24 bits up to 500 KHz and 16
bits up to 15 MHz.

AFAIK, some of their digitizers cost less than a LynxTWO,
but the one described above is $6K-12K, depending on
features.



Yeah, National's cool, especially since you can build test jigs
that do audio one day, and test elevator sensor boards the next with
the same test rig.

Arny Krueger wrote:

"Chris Hornbeck" wrote in
message

On Wed, 7 Sep 2005 07:55:22 -0400, "Arny Krueger"
wrote:


Dunno. I own two Sound Technology distortion analysers
and have since the mid-1970's. I have two because
they get used.

I have long owned a highly modified Heath THD analyzer
with residual under 0.01%. It fell into complete disuse
when I figured out how to get down to 0.001% and then
0.00015% with computer audio interfaces and software.

So, what do you measure that's relevant to this
discusison of slew rates?

Please forgive me for not responding earlier. I'd
mistaken your post as rhetorical, or perhaps harsh.
In proper perspective, neither need apply; sorry.

The antique analog hardware certainly has its
limitations, and worse, its creakyness, but also
has some residual strengths. What common digital
computer interface can measure the harmonics of
a 100KHz signal? Or, really, accurately, of any
supersonic signal?


To me ultrasonic starts at 16 Khz, so even a M-Audio
Audiophile 24/192 is interesting.

But for the heavy stuff check National Instruments.

http://sine.ni.com/nips/cds/view/p/lang/en/nid/201603

Seems like this interface does 24 bits up to 500 KHz and 16
bits up to 15 MHz.

AFAIK, some of their digitizers cost less than a LynxTWO,
but the one described above is $6K-12K, depending on
features.


Where'd Kevin go, by the way?

Dan Kennedy wrote:

In comparison, 1MHz common mode was attenuated by 30dB. I should be
able to improve on this. My predecessors' mucking about with ferrite
beads look rather lame in comparison.

Graham


Gotta be blunt here, but are you really that far behind Graham? Arrrghh.

To be honest it's just a first stab. Literally something I wound using bits
I had handy.

I plan on refining it. What attenuation do you aim for ?

Graham

On Sat, 10 Sep 2005 04:20:59 -0400, "Arny Krueger"
wrote:

http://sine.ni.com/nips/cds/view/p/lang/en/nid/201603

Seems like this interface does 24 bits up to 500 KHz and
16 bits up to 15 MHz.

I think it's KS/s and MS/s, but that's still plenty for
our purposes.

Well, generally one tests one channel at a time.

The two interfaces can be tied together? Really
doesn't matter at these speeds anyway. Plenty
for our purposes, I'd think.

AFAIK, some of their digitizers cost less than a LynxTWO,
but the one described above is $6K-12K, depending on
features.

Probably about what a Sound Tech would have to cost
nowdays. Things do get better, on average.

I think the new ones are about 3 times the upper dollar
figure, above.

Ouch! And I didn't even know any were still made.
All *way* out of my current price range (retired
with a day job - kinda life) anyway.

Thanks,

Chris Hornbeck

Dan Kennedy wrote:

Where'd Kevin go, by the way?

I saw a post of his in s.e.d the other day.

Graham

First off, sorry for being a late night jack-ass.

I went to the shop and looked up the details of the common mode input
filter I've been using, it's not dramatically dissimilar, about 35db
down at 1 mHz, it's bifilar wound on a Siemens pot core.

Some variations on a theme for different inputs, but I've been doing it
for about 4 years on the transformerless mic ins and some
instrumentation inputs as well. What surprised me was that you were just
now looking into it.

Dan Kennedy



Pooh Bear wrote:
Dan Kennedy wrote:


Where'd Kevin go, by the way?


I saw a post of his in s.e.d the other day.

Graham

Dan Kennedy wrote:

First off, sorry for being a late night jack-ass.

S'ok - lol. I'm sure we've all done it. ;-)


I went to the shop and looked up the details of the common mode input
filter I've been using, it's not dramatically dissimilar, about 35db
down at 1 mHz, it's bifilar wound on a Siemens pot core.

Any special reason you chose a pot core ?

Just 'mucking about', I reckon a toroid's easier to wind.

Did I mention I posted a pic in alt.binaries.schematics.electronic btw ?

Some variations on a theme for different inputs, but I've been doing it
for about 4 years on the transformerless mic ins and some
instrumentation inputs as well. What surprised me was that you were just
now looking into it.

It's one of those things that's been on the back burner really. Since I'm
designing low cost equipment there's precious little budget to deal with
something that's rarely an issue.

It just so happens that I've been learning a lot about magnetics recently
and a chance comment made me go and look at it again.

That toroid can be fixed to a pcb using a ty-wrap and it takes next to no
time to wind one - so material cost primarily just boils down to the ferrite
toroid. That seems to be a go-er I reckon.

Graham

Pooh Bear wrote:
Dan Kennedy wrote:

I went to the shop and looked up the details of the common mode input
filter I've been using, it's not dramatically dissimilar, about 35db
down at 1 mHz, it's bifilar wound on a Siemens pot core.

Any special reason you chose a pot core ?

Just 'mucking about', I reckon a toroid's easier to wind.

Toroids are a major pain to wind, and consequently are more expensive.
Pot cores are really easy to do, even by hand or on a simple lathe. You
wrap the bobbin, drop the bobbin in place, and press the core on.

It just so happens that I've been learning a lot about magnetics recently
and a chance comment made me go and look at it again.

That toroid can be fixed to a pcb using a ty-wrap and it takes next to no
time to wind one - so material cost primarily just boils down to the ferrite
toroid. That seems to be a go-er I reckon.

There is a huge variety of pre-wound magnetic stuff available out there,
in the US at Digi-Key at least. Find a Toko dealer in the UK and you will
have hit the jackpot. Common mode stuff is plentiful and cheap, as are
small value inductors. Getting large inductances for audio range equalization
is hard, though. But you just care about RF.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Scott Dorsey wrote:

Pooh Bear wrote:
Dan Kennedy wrote:

I went to the shop and looked up the details of the common mode input
filter I've been using, it's not dramatically dissimilar, about 35db
down at 1 mHz, it's bifilar wound on a Siemens pot core.

Any special reason you chose a pot core ?

Just 'mucking about', I reckon a toroid's easier to wind.

Toroids are a major pain to wind, and consequently are more expensive.
Pot cores are really easy to do, even by hand or on a simple lathe. You
wrap the bobbin, drop the bobbin in place, and press the core on.

It just so happens that I've been learning a lot about magnetics recently
and a chance comment made me go and look at it again.

That toroid can be fixed to a pcb using a ty-wrap and it takes next to no
time to wind one - so material cost primarily just boils down to the ferrite
toroid. That seems to be a go-er I reckon.

There is a huge variety of pre-wound magnetic stuff available out there,
in the US at Digi-Key at least. Find a Toko dealer in the UK and you will
have hit the jackpot. Common mode stuff is plentiful and cheap, as are
small value inductors. Getting large inductances for audio range equalization
is hard, though. But you just care about RF.

The toroid doesn't need a bobbin though.

Such things are a consideration when manufacturing low cost equipiment.

Pre-wound magnetics from the likes of Toko etc are simply way too expensive.
Remember I'll get this made in China.

Graham

I use the pot core because my local magnetics guy winds them day in and
day out, and it's not a quantity/bottom line thing for me.

I also like the more contained field of the pot core, less radiation
into surrounding circuitry, be it adjacent circuit boards, or just close
proximity on the same one. Remember, toroids are better than EI's, but
still have substantial fields on axis and off any irregularities in the
windings.

Also, they mount with four quick solder connections, no dicking around
with stripping wire leads and nylon screws and **** or wire ties.

But again, it's small quantity use here, hundreds, not thousands.

Dan




Scott Dorsey wrote:
Pooh Bear wrote:

Dan Kennedy wrote:


I went to the shop and looked up the details of the common mode input
filter I've been using, it's not dramatically dissimilar, about 35db
down at 1 mHz, it's bifilar wound on a Siemens pot core.

Any special reason you chose a pot core ?

Just 'mucking about', I reckon a toroid's easier to wind.


Toroids are a major pain to wind, and consequently are more expensive.
Pot cores are really easy to do, even by hand or on a simple lathe. You
wrap the bobbin, drop the bobbin in place, and press the core on.


It just so happens that I've been learning a lot about magnetics recently
and a chance comment made me go and look at it again.

That toroid can be fixed to a pcb using a ty-wrap and it takes next to no
time to wind one - so material cost primarily just boils down to the ferrite
toroid. That seems to be a go-er I reckon.


There is a huge variety of pre-wound magnetic stuff available out there,
in the US at Digi-Key at least. Find a Toko dealer in the UK and you will
have hit the jackpot. Common mode stuff is plentiful and cheap, as are
small value inductors. Getting large inductances for audio range equalization
is hard, though. But you just care about RF.
--scott