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Natural Science Forum / Physics / Acoustics / March 2004



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Delay of an LC model of a transmission line?

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Svante - 27 Feb 2004 18:01 GMT
I have a way of reasoning spinning around in my head, and I understand
that it is wrong somewhere, I just cannot find out where. It goes like
this.

An acoustic lossless tube can be modeled by a bunch of LC components,
like this:

-L/2-*-L-*-L-*-L-*-L-*-L-*-L/2-
    |   |   |   |   |   |
    C   C   C   C   C   C
    |   |   |   |   |   |
-----*---*---*---*---*---*-----

As far as I understand it models the behavior of the tube perfectly,
if I use an infinite number of elements. If I inject an impulse in the
left end, this impulse will come out delayed in the right end. If I
short the right end (which corresponds to an open tube) there will be
resonances in the tube and these can be seen in the input impedance on
the left side. This filter has only poles, no zeroes.

On the other hand, this contradicts what I know from experience from
inverse filtering. Inverse filtering is a technique used in voice
research, and it aims at cancelling the effects of the acoustic "tube"
between the vocal folds and the mouth. Using this technique, a filter
that is the inverse to the above all-pole filter is connected to the
flow signal out of the mouth, and we can see the waveform of the flow
that passes the vocal folds. BUT: This signal is still delayed, I can
see that if we compare it to when the vocal folds collide. This also
makes sense, if it was not delayed, the inverse filter would be
non-causal.

So, does the above model NOT predict everything that happes in a
transmission line (the delay). If it does not, how come the tube
resonances come out right? They are a consequence of the delay of the
tube. If it does, why is the signal still delayed after inverse
filtering?

This drives me nuts!
dvt - 27 Feb 2004 19:00 GMT
 > An acoustic lossless tube can be modeled by a bunch of LC components,
> like this:
>
[quoted text clipped - 6 lines]
> As far as I understand it models the behavior of the tube perfectly,
> if I use an infinite number of elements.

Yep.

> If I inject an impulse in the
> left end, this impulse will come out delayed in the right end. If I
> short the right end (which corresponds to an open tube) there will be
> resonances in the tube and these can be seen in the input impedance on
> the left side. This filter has only poles, no zeroes.

I suspect you're confusing temporal response (pulse, delay) with
frequency response (poles, zeros). There is a relationship, as you
probably know. The Fourier transform is usually used.

If you put a sinusoidal pressure wave in one end of a tube that is
blocked at the other end, there are some frequencies at which you will
get no net pressure at the input end. If the tube/t-line is 1/4
wavelength long, the wave reflects off the rigid end and completely
cancels at the input end. The pressure is non-zero at the rest of the
tube, but the input impedance appears to be zero (no pressure, lots of
velocity).

> On the other hand, this contradicts what I know from experience from
> inverse filtering. Inverse filtering is a technique used in voice
[quoted text clipped - 9 lines]
> So, does the above model NOT predict everything that happes in a
> transmission line (the delay).

Yes, it represents an ideal transmission line perfectly.

> If it does, why is the signal still delayed after inverse
> filtering?

I can't understand that question. I'll try to talk around the subject a
little bit, but don't be surprised if I don't answer the question.

If you apply a filter and then a filter with the opposite response, the
frequency content of the input and output signals will be the same.
There will be a delay between the input and the output. The inverse
filter can not eliminate the delay, as you mentioned.

If the filters are inverse in magnitude /and/ phase, the input and
output will have the same phase relationships within the signal. But one
will still be delayed. You can delay a signal by one period and the
phase will look exactly the same.

Signature

Dave
dvt at psu dot edu

Svante - 28 Feb 2004 08:35 GMT
>   > An acoustic lossless tube can be modeled by a bunch of LC components,
> > like this:
[quoted text clipped - 27 lines]
> tube, but the input impedance appears to be zero (no pressure, lots of
> velocity).

OK, the input impedance will have zeroes. I must admit that I at some
stage thought of the above model as terminated, ie with a resistance
at the right end. In that case there would be no standing waves, but
still a delay. This filter would be all-pole and minimum phase.

> > On the other hand, this contradicts what I know from experience from
> > inverse filtering. Inverse filtering is a technique used in voice
[quoted text clipped - 11 lines]
>
> Yes, it represents an ideal transmission line perfectly.

Good, I wanted to hear that!

> > If it does, why is the signal still delayed after inverse
> > filtering?
[quoted text clipped - 11 lines]
> will still be delayed. You can delay a signal by one period and the
> phase will look exactly the same.

So, what you are saying is that two signals that are the same can
actually be delayed with respect to one another??? A delayed impulse
would be the same as the impulse??? This is what my reasoning above
leads to and it is the essence of my question. I find this terribly
strange!
dvt - 01 Mar 2004 13:22 GMT
> So, what you are saying is that two signals that are the same can
> actually be delayed with respect to one another??? A delayed impulse
> would be the same as the impulse???

Yes.

> This is what my reasoning above
> leads to and it is the essence of my question. I find this terribly
> strange!

I don't think I completely understand you, since it doesn't seem so
strange to me. The following ASCII art represents two pulses that look
the same, but one (the output) is delayed.

Pulse at input:

   -
  / \
 /   \
-     ------- . . .

Delayed pulse at output:

        -
       / \
      /   \
------     -- . . .

Signature

Dave
dvt at psu dot edu

Svante - 01 Mar 2004 20:35 GMT
> > So, what you are saying is that two signals that are the same can
> > actually be delayed with respect to one another??? A delayed impulse
[quoted text clipped - 23 lines]
>        /   \
> ------     -- . . .

These two don't look *identical* to me. The delay being the
difference. The FTs of these two waveforms will have different phases,
right? Supposedly, the inverse filter reverses all phase and magnitude
effects of the filter, that is the whole point. If phase and magnitude
is restored, the waveform must be the same, without delay. This should
work as long as the filter is minimum phase. The transfer function of
the model (in my first post) can definitely  be minimum phase, and it
can (I think) approximate a delay line if the number of sections is
made high enough (like a bessel filter). But how on earth can the
inverse of this filter know how to cancel the delay and still be
causal?

Am I starting to sound like a nutcase?
dvt - 01 Mar 2004 21:18 GMT
> These two don't look *identical* to me. The delay being the
> difference. The FTs of these two waveforms will have different phases,
> right?

Yes. Assuming, of course, that t=0 is taken to be the leftmost point in
my diagram. If you shift t=0 by the delay, you get the same phase info
for both signals.

> Supposedly, the inverse filter reverses all phase and magnitude
> effects of the filter, that is the whole point.  If phase and magnitude
> is restored, the waveform must be the same, without delay.

I think you're having a problem with steady-state vs. instantaneous. The
frequency response of two causal cascaded filters must have some net
phase shift (and thus delay) given a non-zero bandwidth.

You can replicate a waveform's temporal shape by replicating its
frequency content and adding a linear phase shift (i.e. phase
proportional to frequency). I suspect your "reverse filters" did exactly
that: they reproduced the temporal shape of the signal at the vocal chords.

> But how on earth can the
> inverse of this filter know how to cancel the delay and still be
> causal?

I'd say it can't. Was your inverse filter written in software? You can
make software imitate a noncausal system, but I don't know how to make
such a thing in real life.

> Am I starting to sound like a nutcase?

No, but I'm really beginning to wonder if we're talking about the same
topic. I can't quite grasp your problem yet.

Signature

Dave
dvt at psu dot edu

Eckard Blumschein - 02 Mar 2004 10:54 GMT
> frequency response of two causal cascaded filters must have some net
> phase shift (and thus delay) given a non-zero bandwidth.

Wasn't filtering in case of zero bandwidth anyway pointless?

> You can replicate a waveform's temporal shape by replicating its
> frequency content and adding a linear phase shift (i.e. phase
[quoted text clipped - 9 lines]
> make software imitate a noncausal system, but I don't know how to make
> such a thing in real life.

You know it: It is impossible. I just wonder why so many people take
Fourier analysis a gospel. I see it a tool with some flaws.
I would like more poeple being so critical as Svante, and I hope he will
not get repidly contented with obvious non-causality, etc.

Well, I don't hide offering an alternative: FCT. That's the natural way
our ears analyse the signal into a frequency spectrum with no phase at
all. Doing so, causality is no longer a topic. It is made sure from the
very beginning. Admittedly this concept has been declared too simple
just for lacking insight. Actually, the inner ear does not discard
phase, it does not analyse in terms of magnitude and phase.

Eckard

>> Am I starting to sound like a nutcase?
>
> No, but I'm really beginning to wonder if we're talking about the same
> topic. I can't quite grasp your problem yet.
Ban - 02 Mar 2004 12:16 GMT
The following ASCII art represents two pulses that
>> look the same, but one (the output) is delayed.
>>
[quoted text clipped - 23 lines]
> inverse of this filter know how to cancel the delay and still be
> causal?

What you want is impossible. It reverses time, which in our world is a
no-no. A delay line is also *not* a minimum phase system, so there is no
point reversing the transfer function. You cannot model a delay with a
transfer function, what you model is only the amplitude- and the phase
response without the delay.

> Am I starting to sound like a nutcase?

Yes, reminds me of those hobbyists inventing the perpetuum mobile. There are
constraints like the absolute zero temperature or the speed of light or the
positive axis of time that cannot be overcome. Beam'em up Scotty!

Signature

ciao Ban
Bordighera, Italy

Svante - 02 Mar 2004 21:28 GMT
> The following ASCII art represents two pulses that
> >> look the same, but one (the output) is delayed.
[quoted text clipped - 30 lines]
> transfer function, what you model is only the amplitude- and the phase
> response without the delay.

Well, what I want is an explanation where the error is. I understand
that there is an error *somewhere* in my thinking, I just don't know
where. I know I cannot reverse time, and that a delay cannot be
cancelled with a causal filter, but the LCLCL...LCLCL-model can be
cancelled with an inverse filter, if it is all-pole (and I think it
is, at least if it is properly terminated)..

And I *think* that the delay line model (the LCLCL...LCLCL-model) also
approximates the delay up to a certain frequency (which is determined
by the number of sections I use in the approximation). Otherwise it
could not model pipe resonances.

I am beginning to think that the problem lies in that I have to have
an infinite order bessel filter to model the delay perfectly, and that
is where the error is. If I don't have an infinite number of sections,
the model will not work for high enough frequencies. Hmmm...

> > Am I starting to sound like a nutcase?
>
> Yes,

OK, I had that coming... :-)

> reminds me of those hobbyists inventing the perpetuum mobile. There are
> constraints like the absolute zero temperature or the speed of light or the
> positive axis of time that cannot be overcome. Beam'em up Scotty!

Well, at least I realise that I am wrong *somewhere*... Now where did
I put that flux capacitor?
The Ghost - 03 Mar 2004 00:44 GMT
> Well, what I want is an explanation where the error is. I understand
> that there is an error *somewhere* in my thinking, I just don't know
[quoted text clipped - 13 lines]
> the model will not work for high enough frequencies. Hmmm...
>  

Your error is in the belief/assertion "....but the LCLCL..LCLCL-model can
be cancelled with an inverse filter.  

That belief/assertion is incorrect.  If the inverse filter is causal and
operates in real time, it can only further add to the negative phase shift
associated with the original delay.  You can remove the delay is to take an
FFT of the signal and add a positive phase shift of exp(jwT), where T is
the delay to be removed.  Then inverse FFT and you will get the original
waveform without the dealy.  However, this process amouts to a non-causal
filter and can not be implemented in real time by an inverse filter
constructed with Rs,Ls, and Cs.

 
Ban - 03 Mar 2004 05:36 GMT
>> What you want is impossible. It reverses time, which in our world is
>> a no-no. A delay line is also *not* a minimum phase system, so there
[quoted text clipped - 13 lines]
> by the number of sections I use in the approximation). Otherwise it
> could not model pipe resonances.

This is a good approximation of a lossless transmission line. It is not as
simple as yours, but it is only valid up to a certain maximum frequency(as
yours).

              ||
          +---||----+
          |   ||    |
          |   ___   |
o-------+-+---UUU---+-----+------o
        |                 |      |
        |  ||        ___  |      |
        +--||--------UUU--)--+  .-.
           ||             |  |  | |
                          |  |  | |50R
           ||        ___  |  |  '-'
        +--||--------UUU--+  |   |
        |  ||                |   |
        |     ___            |   |
o-------+-+---UUU---+--------+---o
          |         |
          |   ||    |
          +---||----+
              ||
created by Andy´s ASCII-Circuit v1.24.140803 Beta www.tech-chat.de

> I am beginning to think that the problem lies in that I have to have
> an infinite order bessel filter to model the delay perfectly, and that
> is where the error is. If I don't have an infinite number of sections,
> the model will not work for high enough frequencies. Hmmm...

Exactly, the above model is valid(1% error) for example for 1us delay:
1 element   180kHz
10 elements 2.22MHz
20 elements 5.1 MHz
30 elements 8.3 MHz
Well, then my software came to its limit. :-(
Svante, don't confuse a simulation with the reality. We simplify things to
get valid results within our needs. But a simulation is only what its name
says...
And (again) a transmission line is *not* a minimum phase system. We can
approximate it with some filters, but...
Signature

ciao Ban
Bordighera, Italy

Angelo Campanella - 03 Mar 2004 07:21 GMT
>>The following ASCII art represents two pulses that
>>
[quoted text clipped - 13 lines]
>>>>       /   \
>>>>------     -- . . .

    Not of primary interest here, but good information to be aware of: Such
delay lines have been used since WWII in fast nuclear pulse counters,
especially for very fast radiation (beta and especially gamma rays). The
"scintillation counter" works by positioning a large block (a few inches
across) of plastic in front of (cemented to) a photomultiplier tube
(PMT). The plastic is doped with a fluorescent material that will flash
when a gamma  or beta ray enters it and becomes absorbed.
    These flashes have a nanosecond rise time, and a long tail decay, almost
a microsecond for instance. To get rapid counting rate capability, only
the leading edge of the flash is needed. So on the output of the PMT is
fastened a delay line, shorted at its outer end. When a flash occurs,
the PMT output current follows the brightness of the flash. The voltage
that is found across the delay line first rises, and some time later,
the effect of the short is felt, and pulls the voltage back to zero.
    The result is a short flat-topped voltage pulse whose height is
proportional to the current leading edge slope and strength. The length
of the pulse is proportional to the length of the delay line. A one foot
length would provide a two nanosecond  pulse...

    So you will find such delay line material here and there, as what looks
like a fat coax cable, with a helically wound center conductor. The
helix diameter is a few millimeters, as I recall (it's been a long time
since I have seen and dealt with them).

    Angelo Campanella
Angelo Campanella - 03 Mar 2004 17:47 GMT
>     The result is a short flat-topped voltage pulse whose height is
> proportional to the current leading edge slope and strength. The length
> of the pulse is proportional to the length of the delay line. A one foot
> length would provide a two nanosecond  pulse...

    Correction. That's for regular coax. This special delay-line coax has
sufficient inductance per unit length to significantly reduce the
propagation velocity. Existing values escape me, but I suspect that it
is probably 1/4 or less of the speed of light. The characteristic
impedance of this delay line is quite high... something like 1,500 ohms
or greater. It can be brought down by adding capacitance along the
way... Perhaps someone here can help. I think that today, acoustical
delay lines are more inclined to convert to ultrasounds and, use their
much slower propagation speed to get practical audible acoustics results...

>     Angelo Campanella
dvt - 03 Mar 2004 14:29 GMT
> the LCLCL...LCLCL-model can be
> cancelled with an inverse filter...

I think Gary has it right: this statement of yours is wrong.

Your original post had something about an inverse filter to compensate
the transfer function of the mouth. That inverse filter, unless it were
noncausal, could not have eliminated the delay in the signal. I suspect
you were confused by one of two things: either the filter was noncausal
(i.e. software) or the resulting signal was delayed but the delay was
masked when the data was presented.

> Now where did
> I put that flux capacitor?

In my sock drawer. The bad guys will *never* find it there.

Signature

Dave
dvt at psu dot edu

Svante - 03 Mar 2004 22:17 GMT
> > the LCLCL...LCLCL-model can be
> > cancelled with an inverse filter...
>
> I think Gary has it right: this statement of yours is wrong.

Gary & Dave:

Hmm... I have an old table with bessel filters in it. It has exactly
this configuration of coils and capacitors (but is terminated). This
tells me that the transfer function has only poles. There are other
tables for butterworth and chebychev filter as well with the exact
same network. I suspect that the circuit has only poles regardless of
the values of the components, but that doesn't matter, take the
terminated bessel case. Isn't that invertible?

> Your original post had something about an inverse filter to compensate
> the transfer function of the mouth. That inverse filter, unless it were
> noncausal, could not have eliminated the delay in the signal. I suspect
> you were confused by one of two things: either the filter was noncausal
> (i.e. software) or the resulting signal was delayed but the delay was
> masked when the data was presented.

Actually, I think of the old analogue stuff from the 80's when I think
about these things. However, I have implemented inverse filters on the
computer as well, and they are a cascade of second order FIR filters
(each cancels a formant=resonance=2nd order IIR filter). So there is
no compensating delay and indeed we see the inverse filtered signal
delayed by half a millisecond or so compared to the glottal closures.
This is what makes me realise that I am wrong somewhere.

Right now I have only three candidates;
1: The LCLCLCL model does not include the delay of the TL (=vocal
tract)
2: Something strange happens when going towards infinitely many LC
sections
3: The LCLCLCL network is not minimum phase.

All of them are wrong, but one of them must be right... :-(

I have a vague memory from my education that some teacher said that
the group delay is *not* equivalent to the delay of the information
flow, but rather a delay of stationary signals or something like that.
Maybe there is a solution there? This makes sense since there are
filters (eg first order, to make it really simple) that has a positive
group delay but are invertible. For example the filter

     1+s                         1+2s  
H(s)=------ has the inverse H(s)=------
     1+2s                        1+s  

and cascading these two filters gives a transfer function of 1 without
any delay. (Please say "yes"!) The inverse filter has, as far as I
understand, a negative group delay. Still it is causal. Hmm...

Thank you all for your input!
The Ghost - 04 Mar 2004 00:35 GMT

, I have implemented inverse filters on the
> computer as well, and they are a cascade of second order FIR filters
> (each cancels a formant=resonance=2nd order IIR filter). So there is
> no compensating delay and indeed we see the inverse filtered signal
> delayed by half a millisecond or so compared to the glottal closures.
> This is what makes me realise that I am wrong somewhere.

You have a source signal that is applied to a filter that has a propagation
delay.  You measure the filter output and apply it to a causal inverse
filter.  At the output of the inverse filter you should see the source
signal delayed.  That is exactly what you say you see.    

For example the filter

>       1+s                         1+2s  
> H(s)=------ has the inverse H(s)=------
[quoted text clipped - 3 lines]
> any delay. (Please say "yes"!) The inverse filter has, as far as I
> understand, a negative group delay. Still it is causal. Hmm...

That too is correct, but the filter (1+s)/(1+2s) does not have a
propagation delay and does not model the situation that you previously
described.  

The proper model would be H(s)=[(1+s)/(1+2s)]exp(-sT), which would
represent a filter with a propagation delay.

Now, if you cascaded an inverse filter (1+2s)/(1+s), you would still see
the delay at the output.  In order to back out the delay, the inverse
filter would have to contain the multiplicative term exp(sT), which
corresponds to a negative delay and which is non-causal.



Eckard Blumschein - 04 Mar 2004 08:08 GMT
                     the inverse
> filter would have to contain the multiplicative term exp(sT), which
> corresponds to a negative delay and which is non-causal.

There are not many people who are able and willing to judge so
correctly. Perhaps, I found a treasure in you. Would you be interested
in my somewhat heretical but nonetheless logical and practical point of
view with respect to an alternative reference point of time? Could I
send to you ppt files up to 5 MB?

Eckard
blumschein@et.uni-magdeburg.de
Svante - 04 Mar 2004 08:50 GMT
>  
> , I have implemented inverse filters on the
[quoted text clipped - 30 lines]
> filter would have to contain the multiplicative term exp(sT), which
> corresponds to a negative delay and which is non-causal.

I have been thinking along these lines as well, inspired by your mail,
and it would explain what I see using inverse filtering of the vocal
tract. But then what I don't understand is how can the LCLCLCL model
manage to simulate the tube resonances so well? I mean, the tube
resonances are in essence the effect of the delay in the tube. If the
LCLCLCL model did *not* model this delay, then the resonance
frequencies would come out different from the real tube.
IME, the LCLCLCL model gives resonance frequencies that are very
accurate as long as the number of sections is sufficient.
The Ghost - 05 Mar 2004 22:57 GMT

> Right now I have only three candidates;
> 1: The LCLCLCL model does not include the delay of the TL (=vocal
> tract)


I used pspice to create a 10-section LCLCLCL line.  I calculated the
component vales using the equations Zo=sqrt(L/C) and v=1/sqrt(LC).  For Zo,
I used 92 cgs ohms (which is the characteristic impedance of the human ear
canal) and for v I used 34000 cm/sec, which is the approximate speed of
sound in air.  With the above values for Zo and v, the calculated value for
L was 2.71m, and the calculated value for C was 319n.  At the front and
rear of the line I used L/2.  

When the line was terminated with a resistance equal to Zo (92 ohms), the
output response magnitude was virtually flat up to the cutoff frequency
(approx 10KHz) and the output response phase was linear up to about 5KHz.  
The linear phase lag corresponded to a constant time delay of 0.297msec.  
Using this delay and the velocity of sound, I calculated that the effective
length of the 10-section line is 10.1cm.  Using this length, I then
calculated quarter wave and three quarter wave resonancet frequencies of
842Hz and 2526Hz.  

I then mismatched the line by increasing the terminating resistance from 92
ohms to 200 ohms.  This resulted in resonances in the output, the first two
of which occurred at frequencies of 845 and 2520, in excellent agreement
with the calculated quarter wave and three quarter wave resonant
frequencies. Additionally, the phase response of the mismatched line showed
the same average delay as the Zo terminated line, but contained ripples
associated with the resonances and antiresonances in the magnitude
response.  

In conclusion, the 10-section LCLCLCL line showed BOTH the expected delay
(linear phase) and the expected resonances that are associated with a
continuous transmission line.   This, for me, conclusively rules out
candidate 1.
Bob Cain - 28 Feb 2004 18:33 GMT
>  You can delay a signal by one period and the
> phase will look exactly the same.

Not quite.  It will add a constant to the group delay which
implies adding a tilt to the phase response.

Bob
Signature


"Things should be described as simply as possible, but no
simpler."

                                             A. Einstein

dvt - 01 Mar 2004 13:17 GMT
>>  You can delay a signal by one period and the
>> phase will look exactly the same.
[quoted text clipped - 3 lines]
>
> Bob

Yes, I see your point. When a signal is delayed, my analyses usually
delay the phase reference as well, which gets rid of the tilt in the
phase.  So I often forget about that.

One for the memory book: constant delay = linear phase shift.

Signature

Dave
dvt at psu dot edu

 
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