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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