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In Reply to: Re: Tom Danley at AES Chicago posted by Paul Eizik on February 21, 2007 at 11:21:14:
"Due to the fact that a direct radiator design is dominated by mass in most of it's pass band, it necessarily introduces phase errors which don't track the input signal accurately....."From what I've heard, loudspeakers are minimum phase systems and thus will not (and should not) give less phase than what is dictated by the amplitude transfer.
Follow Ups:
RapidThe point you raise (i.e.: the asumption that direct radiator speakers are minimum phase) is exactly what Tom Danley has called into question and provided evidence against. If you take a theoretical speaker on an infinite baffle, it will show an increasing acoustic load with increasing frequency. To make the driver measure nominally flat, the mass of the moving system will have to be increased to the point that it counteracts this tendency. At this point the driver can modeled as an electrical analogous circuit, or an R/C filter to be specific. This is where the R=the voice coil resistance lumped with the amplifier source impedance, and the C is the moving mass of the driver reflected through the driver motor. This C behaves like a capacitor (but is a seperate factor from the actual voice coil capacitance of course). So basicly what you have in this model is an R/C crossover which adds phase shift. Now let's say that you increase the strength of the magnet to a great degree (or conversely drasticly lighten the moving mass) to the point that you have an "overdamped driver", or where the magnet becomes dominant. You then get a rising high frequency response as the mass becomes less dominant. You can see this in a typical big magnet musical instrument driver (which is over damped) as the bass will roll off typicaly over an octave above Fs or, to look at the other way, a rising high frequency response. The large magnet Lowther drivers are typical of this family too, and their rising high freq. response is generally noted (however their whizzer cones are a factor in this too). Adding a horn to this type of driver introduces an acoustic load which is not like a capacitance in that it is (theoreticly if done correctly) consistant over the horn's pass band, and not frequency dependent. So you are substituting the R/C filter effect for the more consistant acoustic resistant load of the horn ( and some bandwidth too, no doubt). Hopefully Tom will come in here if I have mangled any of the concepts.
Hi Paul,Paul, you hit the nail on the head.
When a direct radiator is below the knee in its radiation resistance curve, it feels a changing radiation resistance vs frequency.
In order to make “flat” response in that range, one must roll off the radiator velocity to compensate.
The filter that does this is internal to the driver, consisting of the VC’s Rdc and the driver’s moving mass which is converted by the motor to a capacitance.
Some stuff on line.Fig#3
http://www.akabak.de/Texte/aes102.pdfhttp://www.silcom.com/~aludwig/Sysdes/Thiele_equiv_circuit.gif
When you have a minimum phase problem, you can fix it with a complementary minimum phase fix, this is when EQ works perfectly.
Here, one is fixing the effect of a frequency dependent resistance which does not have the same reactance as a filter with the same slope. Compensating that with an RC filter, does fix the amplitude response BUT leaves the phase of the RC filter which is not canceled out by anything.
At the meeting, I had a TEF measurement of a small woofer, showing the acoustic phase mid band is around –90 degrees over a wide range.Now, while this phase shift prevents the driver from replicating a complex input waveshape (like a square wave, impulse or even music for example), some direct radiators operate PAST the knee in the curve. RCA fan sent me a curve once of a 8 inch full range driver that made a smooth transition thru that range and above, could reproduce a square wave over some range.
Now, this is a rare exception by a very skillful design and component properties.The problem is that when drivers operate in breakup, there behavior becomes very frequency dependant so this is normally avoided.
On the other hand, the absence of perturbation at crossover and some range of proper operation is likely what makes single drivers so attractive for the home.Keep in mind that some very popular measurement systems make the assumption that the speaker IS minimum phase and in which case, the phase response can be found by the amplitude response. On the other hand, the Time Delay Spectrometry process measures both amplitude and phase separately and so it shows what the driver actually does.
When one has things like a changing radiation resistance that does not have the normal phase change associated with it, that this is not seen properly unless actual acoustic phase is measured.
Lastly, unlike electronics speakers spread signals out in time significantly, keep in mind that it is nothing like universally agreed that speakers shouldn’t do this or that a speaker even should be able to preserve the waveshape of the input signal.Best,
Hi there I am Svante, and I am new to this forum. I am a teacher in electro acoustics at university level, and the writer of the Basta! loudspeaker simulation software, just to give you an idea of my background. I just had to jump in on this because this post was pointed out to me and that I find the statements in Tom's post fundamentally wrong (no offence intended).The idea of a radiation resistance that is proportional to frequency ie Ra~f^2 comes from the radiation impedance of a baffled piston. Here folows an explanation, skip to **** if you already know it.
The analytical expression for this impedance is rather complicated and involves Bessel fuctions. Fortunately, for low frequencies, the impedance is in essence identical to that of the pulsating sphere, and this impedance in turn has a rather simple expression:
Za=rho0*c/S*jkr/(1+jkr)
k=w/c, c=344 m/s, rho0=1.2kg/m3, S= surface of the sphere, r= radius of the sphere, j=sqrt(-1).
The interesting part is jkr/(1+jkr).
Now if we want to see this impedance as a series connection of a resistive part and a reactive (mass) part we can multiply both the numerator and denominator by (1-jkr), and the result then becomes jkr(1-jkr)/(1+(kr)^2)=(jkr+(kr)^2)/(1+(kr)^2).
The real part of this expression is (kr)^2/(1+(kr)^2) which for low frequencies can be approximated by (kr)^2, ie the resistive part is proportional t f^2 (since k = w/c).
****
Now, this line of thinking would produce a resistor, the radiation resistance, which would absorb an amount of power that corresponds to the power radiated to the air. The model is not, however, suited to find the phase relation between the applied velocity of the cone and the pressure in the air, since there also is a reactive part of the radiation impedance present that affects the pressure.
The solution to find this phase relation is to see the radiation impedance as a parallel circuit instead of a series circuit. Doing so it will have the admittance:
Ya = 1/Za = S/rho0*c*(1+jkr)/jkr.
Here the part (1+jkr)/jkr is the interesting one, and the real part of it is 1, ie it is frequency independent. In other words, the radiation impedance can also be seen as a parallel circuit with an acoustic mass, and an acoustic resistance, and in this case both are well-behaved ordinary impedances.
If this is done, the analog for the loudspeaker in an infinite baffle can be set up, transformed to the acoustical side and reduced for the mass controlled range like in the attached image.
From the reduced analog it is clear that the end result is a pressure (~voltage) divider, with normal well-behaved impedances, and it can be seen that the pressure (voltage) across the radiation impedance is in phase with the driving pressure.
This is also supported by simple measurements of complex waveforms with the main energy in the driver's mass-controlled range.
Now, I was not present at the AES meeting where you presented this 90 degree off measurements, but I cannot see how you managed to get them. In my experience, measurements with a square wave or similar on a single driver typically results in a a square wave from the microphone, any small discrepancies can usually be explained with the frequency response of the driver.
My bottom line here is that there is NO phase shif introduced by the combination of the radiation resistance and the mass reactance. The acoustic signal and the electrical signal are identical and in phase(neglecting other errors in the driver).
“Hi there I am Svante, and I am new to this forum. I am a teacher in electro acoustics at university level, and the writer of the Basta! loudspeaker simulation software, just to give you an idea of my background. I just had to jump in on this because this post was pointed out to me and that I find the statements in Tom's post fundamentally wrong (no offence intended).”Hi Svante
A couple thoughts. You should read up on the work of Richard Heyser and the Time Delay Spectrometry measurement approach and measurement of loudspeaker arrival times.
Many popular measurements systems do not actually measure acoustic phase although they have a display for it. Rather, it is assumed that the speaker is always minimum phase and the phase is calculated based on amplitude response (Hilbert).“Now, I was not present at the AES meeting where you presented this 90 degree off measurements, but I cannot see how you managed to get them”
See above.“My bottom line here is that there is NO phase shif introduced by the combination of the radiation resistance and the mass reactance. The acoustic signal and the electrical signal are identical and in phase(neglecting other errors in the driver).”
I guess your math says one thing and acoustic measurements of the real thing another.
I have had a TEF machine to measure transducers I built for over 25 years now, it is unlikely to have fundamentally wrong all that time.
No phase shift means the acoustic phase is zero degrees, which I have never seen in any speaker other than a Manger. How do you make a woofer etc with what is in effect zero variable time delay and resistively controlled?
If you have real measurements where the woofer is at zero degrees phase, I would like to see them and the know the details.On the other hand, one can set up a “reality test” with a speaker processor and show that many measurement systems do not measure phase correctly while the TEF does.
Or, one can take any small normal woofer outdoors and find that it does not unilaterally radiate a square wave when driven with one. By unilaterally, not tuning F and mic position to “find” a square wave, I mean radiating that.
If speakers preserved acoustic phase at zero degrees and then reproduced the input signal, active sound cancellation would be a breeze, instead it is not.The flaw in your argument (in my opinion) is precisely that the changing radiation resistance does not have the same amount of reactance, as its slope would normally cause “IF” it were a normal reactive circuit. Instead, this is more like a changing resistance.
Thus while the amplitudes do cancel out leaving “flat” one is left with the residual phase.
Consider for a simple sealed box, that the accelerating force which is in phase with radiated pressure is the current. Acoustic phase (mid band) has the general appearance of the current phase in the drive Voltage. It looks Capacitive in its midband well below Rmin and above Fb and by capacitive, it is lagging behind the input Voltage over a broad range. That non-zero phase prevents the wave shape from being reproduced as this represents a different delay for each frequency.
A horn like the one described, can preserve waveshape because going from low to high, it traverse the compliance dominated range, thru zero, becoming a direct radiator but then continuing on to near 180 degrees. A proper size horn is resistance dominated, pressure is velocity, not acceleration. A full horn acts as an inverter (being two quarter wave stubs) so once the phase is around 180 and is resistively controlled, it can reproduce the wave shape. Here, like with an electrostatic speaker, the force is the input voltage, not the input current.
Also, one can find that with the changing resistive portion, that the lf roll off on a horn may not have the normal corresponding phase change one would get from a filter or direct radiator (where N order slope has N times 90 degrees phase shift).
As a result, the Group delay at the low corner of such a system may be superior to one with the same response, made of direct radiators
Best,Tom Danley
---------
"A couple thoughts. You should read up on the work of Richard Heyser and the Time Delay Spectrometry measurement approach and measurement of loudspeaker arrival times.
Many popular measurements systems do not actually measure acoustic phase although they have a display for it. Rather, it is assumed that the speaker is always minimum phase and the phase is calculated based on amplitude response (Hilbert)."
---------Hmm, I don't want to turn this into a discussion on measurement techniques. Really, I am not involving any advanced measurement methods at all, I am just looking on the square waveform on an oscilloscope, and apart from obvious shortcomings due to diffraction or limitations in the driver frequency response, the square wave is perfectly reproduced. This would not happen if all frequencies were shifted by 90 degrees. Of course it will not have a phase shift of 0 degrees due to the acoustic delay between the driver and the microphone, but I assume that that is not what you mean?
So, skipping the advanced methods for a while, are you saying that a normal woofer cannot reproduce a square wave, mounted in an infinite baffle?
Are you also saying the the normal loudspeaker shifts all frequencies by 90 degrees? That would in fact mean that the driver is an Hilbert transformer, see
http://en.wikipedia.org/wiki/Hilbert_transform
...and as far as I know, the square wave response looks nowhere like the graph on that page.
---------
"The flaw in your argument (in my opinion) is precisely that the changing radiation resistance does not have the same amount of reactance, as its slope would normally cause “IF” it were a normal reactive circuit. Instead, this is more like a changing resistance.
Thus while the amplitudes do cancel out leaving “flat” one is left with the residual phase."
---------Hmm... "My" model of the radiation resistance IS independent of frequency, and this is perfectly OK if a parallel circuit is assumed (or a series circuit for mobility analogies).
Do you understand that the impedance of the pulsating sphere can be seen as either a series equivalent, or a parallel one? And that they are equivalent? Do you also understand that the parallel equivalent has a resistance that is NOT varying with frequency?
If you understand this, I can not see how it fits with the above.
I was using impedance analogies in my previous post, using the mobility analogy everything becomes "the other way".
---------
"Consider for a simple sealed box, that the accelerating force which is in phase with radiated pressure is the current. Acoustic phase (mid band) has the general appearance of the current phase in the drive Voltage. It looks Capacitive in its midband well below Rmin and above Fb and by capacitive, it is lagging behind the input Voltage over a broad range. That non-zero phase prevents the wave shape from being reproduced as this represents a different delay for each frequency.
---------Ok, so this seems like a mobility analogy to me, and of course this is fine too. I have redrawn the diagrams in my previous post using mobility analogies, see the attached image, and the result is the same. A key to understanding it is that the radiation resistance is NOT frequency dependent when seeing it as a series equivalent (in the mobility analogy).
In this case the circuit ends up as a pressure (current) divider, where the current from the source is divided in two capacitive branches, and since the resistor is small and independent of frequency, the voltage across it will be proporional to the driving current, ie NO phase shift (apart from the delay between driver and microphone).
HiIf you don’t measure it is pretty hard to argue with theory and this may leave us little common ground.
My approach is strongly based on what is going on in the measurements and many box simulators don’t even not show this effect in the first place (don’t know about yours).
Also, while there are different ways to look at a loudspeakers equivalent circuit, the one I found on line shown here gives an impedance curve which can match the loudspeakers.
In the mid band, the circuit simplifies to the series R and parallel C with the output across the C.http://www.du.edu/~jcalvert/tech/speak.htm
If you use a network analyzer like the one next to my head in the long url below, you know you can also derive the same electrical equivalent circuit with the right data.
Some peripheral discussion of why one can’t use the simple approach for scientific uses..
http://www.dsprelated.com/showmessage/131/1.php
“Are you also saying the the normal loudspeaker shifts all frequencies by 90 degrees?”Yes and No. Look up Richard Heyser’s work on loudspeaker arrival times and Marshal Leach’s work on excess phase in loudspeakers.
For a radiator that is acoustically small, to have “flat response” it has to have a constant acceleration profile IE: falling Velocity to off set the slope in the radiation resistance curve. The motor on the other hand produces back EMF proportional to Velocity AND force proportional to Current. The “filter is the series Rdc and the reflected mass as a C.So, in the complete circuit, well below Fb, acoustic phase may have a positive value, at Fb it is at/near zero, midway between Fb and Rmin it tends to be around –90 degrees. Around Rmin, it is usually back towards zero.
By a careful trade off of inductive roll off point with the radiation resistance curve and the collapsing directivity, it is possible sometimes to have what appears to be and I guess is, resistive operation higher up and these drivers can preserve waveshape above the radiation knee. I saw a curve for an 8 inch jbl full range driver that was good like that.The bottom line is that if you go outdoors to eliminate room effects and put say a typical 10 inch woofer in an appropriate sealed box and got a cutoff of say flat to 40Hz, that between 60 to say 200Hz, it normally cannot reproduce a square wave because its acoustic phase lags significantly and mutilates a square wave (broad band signal ) by being dispersive in time..
Take a 6 inch low inductance miracle woofer tuned to be flat to 20 Hz, and it cannot reproduce a square wave, over an even wider bandwidth.
Try it.
Best,Tom Danley
Hi,------
"If you don’t measure it is pretty hard to argue with theory and this may leave us little common ground. "
------Don't get me wrong, I measure a lot, I just didn't want this discussion to be about measurement techniques. I think I manage to make a case without involving more advanced measurement techniques (a simple oscilloscope and a square wave generator will do). Nothing more is needed.
------
"My approach is strongly based on what is going on in the measurements and many box simulators don’t even not show this effect in the first place (don’t know about yours)."
------
Ok, my simulator doesn't, and I don't think it should :-) . The effect that you describe is not real. You might have seen square wave responses that are not perfect, but how do you conclude that the cause is the sloping radiation resistance? Without proper theory you cannot. It is your theory that I have opinions on, not the imperfect square wave responses.
To get some common ground for the discussions:
As I understand it you make a case of that the flat response of a driver comes about by the fact that the frequency dependance of the radiation resistance is balanced by the increasing mass reactance of the cone. This balance is, according to you, only perfect for the amplitude; there will be a residual phase shift due to the fact that the mass reactance introduces a shift of 90 degrees but the radiation resistance is at 0 degrees. Right?
It is this line of thinking I oppose to. It is based on that the radiated sound pressure is proportional to the pressure at the series equivalent radiation resistance (using the impedance analogy). This is where your theory is fundamentally wrong, and you can understand that by looking into the analogies that I drew in my previous posts. It is the POWER in the radiation resistance that is equivalent to the radiated POWER, but with a frequency dependent radiation resistance, you don't find the pressue in the analog diagram. It is not there, and therefore it can't tell you anything about the phase.
------
"Also, while there are different ways to look at a loudspeakers equivalent circuit, the one I found on line shown here gives an impedance curve which can match the loudspeakers.
In the mid band, the circuit simplifies to the series R and parallel C with the output across the C.http://www.du.edu/~jcalvert/tech/speak.htm
If you use a network analyzer like the one next to my head in the long url below, you know you can also derive the same electrical equivalent circuit with the right data.
Some peripheral discussion of why one can’t use the simple approach for scientific uses..
http://www.dsprelated.com/showmessage/131/1.php
"
------Hmm, this text makes me think that you are not familiar with analogies, is that so? I am NOT talking about equivalents to the electrical impedance, these are analogies with force and pressure represented as voltage and velocity and volume flow represented as current. The components in my diagrams were mostly mechanical and acoustical.
------
"
“Are you also saying the the normal loudspeaker shifts all frequencies by 90 degrees?”Yes and No. Look up Richard Heyser’s work on loudspeaker arrival times and Marshal Leach’s work on excess phase in loudspeakers.
For a radiator that is acoustically small, to have “flat response” it has to have a constant acceleration profile IE: falling Velocity to off set the slope in the radiation resistance curve. The motor on the other hand produces back EMF proportional to Velocity AND force proportional to Current. The “filter is the series Rdc and the reflected mass as a C.So, in the complete circuit, well below Fb, acoustic phase may have a positive value, at Fb it is at/near zero, midway between Fb and Rmin it tends to be around –90 degrees. Around Rmin, it is usually back towards zero.
By a careful trade off of inductive roll off point with the radiation resistance curve and the collapsing directivity, it is possible sometimes to have what appears to be and I guess is, resistive operation higher up and these drivers can preserve waveshape above the radiation knee. I saw a curve for an 8 inch jbl full range driver that was good like that.The bottom line is that if you go outdoors to eliminate room effects and put say a typical 10 inch woofer in an appropriate sealed box and got a cutoff of say flat to 40Hz, that between 60 to say 200Hz, it normally cannot reproduce a square wave because its acoustic phase lags significantly and mutilates a square wave (broad band signal ) by being dispersive in time..
Take a 6 inch low inductance miracle woofer tuned to be flat to 20 Hz, and it cannot reproduce a square wave, over an even wider bandwidth. "
------Ok, so I take that as a "no" for most cases then. This is strange, I just hooked up a small 4" midwoofer and mounted it on a piece of cardboard to reduce the diffraction effects, and the square wave response was actually quite good. The flat parts were sloping a bit, but that can be explained by the rolloff of lower frequencies in the driver.
Might I add that a device that shifts all frequencies by 90 degrees but maintains a constant amplitude is in fact a Hilbert transformer. A Hilbert transformer is non-causal, ie the impulse response starts at t <0. So if the loudspeaker should be considered to shift all frequencies by 90 degrees, it will actually have to predict the future. I don't buy that. ;-)
Please give my thoughts a shot before rejecting them. I am not just another silly loudspeaker nerd with his own "revolutionary" theory, I really do think I have this right after 18 years of teaching the subject. And this IS a theory discussion. It is NOT a discussion on measured data, but on your theory that the effects of mass reactance and frequency dependent radiation resistance do not cancel (with regard to phase). At least it is to me.
“Ok, my simulator doesn't, and I don't think it should :-) . The effect that you describe is not real. You might have seen square wave responses that are not perfect, but how do you conclude that the cause is the sloping radiation resistance? Without proper theory you cannot. It is your theory that I have opinions on, not the imperfect square wave responses.”Funny (as in odd) an educator would describe something that is easy to measure and described by some for many years as “not real”.
“As I understand it you make a case of that the flat response of a driver comes about by the fact that the frequency dependance of the radiation resistance is balanced by the increasing mass reactance of the cone. This balance is, according to you, only perfect for the amplitude; there will be a residual phase shift due to the fact that the mass reactance introduces a shift of 90 degrees but the radiation resistance is at 0 degrees. Right?”
More or less, the crux being that the radiation resistance is mostly a changing R not a reactance which would be perfectly canceled in mag AND phase by the R-C filter (Richard Small) in the woofer mid band.
http://www.akabak.de/Texte/aes102.pdf
http://www.silcom.com/~aludwig/Sysdes/Thiele_equiv_circuit.gif“Ok, so I take that as a "no" for most cases then. This is strange, I just hooked up a small 4" midwoofer and mounted it on a piece of cardboard to reduce the diffraction effects, and the square wave response was actually quite good. The flat parts were sloping a bit, but that can be explained by the rolloff of lower frequencies in the driver.”
Ok, carefully re-read what I said, now try what I described, take a woofer in a sealed box with flat response down low, go outside and check in its mid band. It is normal to be able to find a place in room and some frequency where you get a square wave, this is not the same as the driver radiating that over a band.
“Please give my thoughts a shot before rejecting them . loudspeaker nerd with his own "revolutionary" theory, I really do think I have this right after 18 years of teaching the subject. And this IS a theory discussion. It is NOT a discussion on measured data, but on your theory that the effects of mass reactance and frequency dependent radiation resistance do not cancel (with regard to phase). At least it is to me.”
Ok, now you really have my curiosity, you say several things which really make red flags pop up.“Please give my thoughts a shot before rejecting them. I am not just another silly loudspeaker nerd with his own "revolutionary" theory. I really do think I have this right after 18 years of teaching the subject.”
I am not sure how you could teach this at a serious level and not be aware of Heyser’s work and his discussions of loudspeaker acoustic phase as well as the work of others on the subject. Not only that, you should be superficially familiar with active cancellation of random noise as a popular academic issue where loudspeaker acoustic phase is one primary hang up.
Also, you should know, the Hilbert transform as used in audio processing is a delay mechanism at –90 degrees.“And this IS a theory discussion. It is NOT a discussion on measured data”
I also find it highly odd an educator would dismiss a large difference between theory and measured reality so easily. One does not have that expedient in designing something that has to produce a specific waveshape signal, even something goofy like this.http://pdf.aiaa.org/preview/1993/PV1993_4430.pdf
Where exactly do you teach anyway?
I have to ask, are you really V in disguise?Tom
Ok, so I don't get through, I won't repeat my theoretical arguments. They are there in my previous posts if you or anyone else would want them. So let's talk measurements.I won't go outdoors to do the measurement, there is to much snow here, but if I did, and if I dug down the speaker in the ground in order to get rid of the baffle diffraction effects, I am convinced that I would get a square wave good enough to convince me.
If that was not the case, how can software that measure the impulse response show a near impulse as the impulse response? I have written such software myself, down to the very last multiplication inside the FFTs, and this software is not using the Hilbert transform or assuming minimum phase at all, and room effects can easily be neglected as you probaly know; they just show up as late impulses in the impulse response.
If the loudspeaker was actually introducing a 90 degree phase shift for all frequencies (ie a Hilbert transform), that would for sure show up in the impulse response, right?
As I understand it this was a presentation at an AES meeting in Chicago? Did you write a conference paper on this, if so maybe this could shed some light on things I might have missed. Have you published anything about your theory?
I teach at the Royal Institute of Technology (KTH) in Stockholm among other things the course in electro acoustics. You will probably even be able to a photo of me if you search the web for a while. I admit to being terribly poor at namedropping, but I think my understanding of the topic is good enough anyway.
Hi SvanteOk, it looks like you are who you say, I saw your program web site.
While you didn’t “get through” I am convinced you are sincere.
So lets talk about this in more depth but from a different angle.
One has the Voice coil motor which produces X force per amp of current AND linked to that force sensitivity is a Voltage generated proportional to Velocity.Lets say speaker A was the kind we have been talking about, its radiator velocity has to fall to account for the radiation resistance.
This is the condition in the AKABAK link I posted.
In this case, we disagree on what the acoustic phase response would look like due to the presence (or not) of the phase change normally associated with a change in level (radiation resistance)..
I say that this case the acoustic phase, when the response is flat BUT the radiator acoustically small, generally lags behind in the middle of its band on the TEF machine.Alternately, lets consider a different system with the same motor.
Now one has a Horn system, lets say it is “full size” and high efficiency so that its operation is simple to describe.
Now, one has an acoustically inverting system that is resistively controlled, where the radiated power is proportional to the radiator velocity.
Electrically, this speaker is also much “different” looking than the first case in that the impedance is nearly resistive and at 50% efficiency, is about 2X the Rdc while 50% efficient.So, in one case you have a system in which the electrical terminals look like a tank circuit above resoance, a reactive load over much of its range. Here the force accelerating the mass IS proportional to current, NOT input voltage waveshape.
In the other case, one has a system, which appears to be largely a resistive load, where the motor Velocity is proportional to output and so proportional to the input voltage.
Since the domains of these two systems are 90 degrees apart (constant acceleration vs constant velocity), which one is more likely to be around zero and the other more reactive looking?Lastly, it is cold and snowing here to right now, nude arc welding weather for sure.
For fun, experimentally alter the phase in a test response and see how visible this kind of phase shift is in the appearance of the impulse response.
I can tell you that some measurement systems do not pass a reality check I devised to see if a technique actually can be trusted or not.
If you have a loudspeaker controller and pspice or other filter way to model filters do this check.
Set up an imaginary perfect speaker, start with a 3 ms time delay, add a 2 nd or 4 th order high pass at say 50 or 100Hz and 2 nd order low pass at 5Khz.
Model that response and phase in the computer as the “should be” case.
Now, measure full band and see if the amplitude and phase are proper relative to the “should be”.To answer another question, this is not “my theory”; I exploited what I saw so make a multi-way horn loudspeaker.
I didn’t submit a paper for this yet, it was a local meeting, a presentation with measurements and then music with fireworks recording at the end.
Actually I dropped out of AES about 10 years ago but I will probably re-join as I had fun.
Here are few old papers I gave, most of which aren’t available for free.
Also the blurb on the talk and a link to a white paper that gives a better idea what I am up to.
http://www.aes.org/e-lib/browse.cfm?elib=11721http://www.aes.org/e-lib/browse.cfm?elib=5023
http://scitation.aip.org/getabs/servlet/GetabsServlet?prog=normal&id=JASMAN000096000006003828000003&idtype=cvips&gifs=yes
http://adsabs.harvard.edu/abs/1994ASAJ...96.3828D
http://www.aes.org/sections/chicago/aes_notice_feb2007.pdfhttp://www.danleysoundlabs.com/pdf/danley_tapped.pdf
Tom Danley
-------
"Hi Svante"
-------Hi!
-------
"Ok, it looks like you are who you say, I saw your program web site.
While you didn’t “get through” I am convinced you are sincere.
So lets talk about this in more depth but from a different angle.
One has the Voice coil motor which produces X force per amp of current AND linked to that force sensitivity is a Voltage generated proportional to Velocity.Lets say speaker A was the kind we have been talking about, its radiator velocity has to fall to account for the radiation resistance.
This is the condition in the AKABAK link I posted.
In this case, we disagree on what the acoustic phase response would look like due to the presence (or not) of the phase change normally associated with a change in level (radiation resistance)..
I say that this case the acoustic phase, when the response is flat BUT the radiator acoustically small, generally lags behind in the middle of its band on the TEF machine."
-------Yes, and here is the core of the "dispute". We should concentrate on this. I see no reason to doubt your observations, but there are a number of possible explanations to what you see in the measurements, for example the voice coil inductance, the fact that it is hard to estimate the acoustic centre and thus compensate for the time delay etc. The baffle step is also there introducing a phase shift, but that should actially work in the opposite direction. I argue that the "zero phase frequency dependent radiation resistance" is not an explanation to the lag.
-------
"Alternately, lets consider a different system with the same motor.
Now one has a Horn system, lets say it is “full size” and high efficiency so that its operation is simple to describe.
Now, one has an acoustically inverting system that is resistively controlled, where the radiated power is proportional to the radiator velocity.
Electrically, this speaker is also much “different” looking than the first case in that the impedance is nearly resistive and at 50% efficiency, is about 2X the Rdc while 50% efficient."
-------No problem with this.
-------
"So, in one case you have a system in which the electrical terminals look like a tank circuit above resoance, a reactive load over much of its range. Here the force accelerating the mass IS proportional to current, NOT input voltage waveshape."
-------Well, above resonance and below the range where voice coil inductance start to make an influence, the electrical load is largely resistive, so current and voltage will be more or less proportional to oneanother. Also, the increase in electrical impedance near resonance is compensated by an increased mobility (due to mechanical resonance), so near resonance the acceleration is actually closer related to voltage than current.
-------
"In the other case, one has a system, which appears to be largely a resistive load, where the motor Velocity is proportional to output and so proportional to the input voltage."
-------No problem here.
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"Since the domains of these two systems are 90 degrees apart (constant acceleration vs constant velocity), which one is more likely to be around zero and the other more reactive looking?"
-------Well... The direct drive speaker will of course have a 90 degree lag and a -6 dB/octave tilt when it comes to cone velocity, but this is compensated for by the different radiation impedance (which is something more than the radiation resistance).
This is what I feel is not getting through. You have not commented on "my" (they are not mine actually) two different models of the radiation impedance.
It is true that radiation resistance is frequency dependent in one of the models (the series model using the impedance analogy). In that model, it is hopeless to say anything about phase relations, since the sound pressure does not have a simple representation in the analog diagram here. Yet, this is what you assume, and this is an easy mistake to do if one is comparing to normal resistive systems.
Using the other model, where the radiation resistance is independent if frequency, the pressure IS present in the analog diagram, and phase relations can be derived. Using this model, the phase lag is completely compensated for.
If this is Gibberish to you, here is an alternative approach, which builds on the same fundaments as the above, but is free from the concept of a radiation resistance:
The sound pressure from a point source can be expressed as:
p=jwQ*rho0/(4*pi*r)
where Q is the volume flow, w is angular frequency and r the distance, the rest are constants. The equation neglects the delay due to distance, but that is perfectly in order here.
Note that the pressure is proportional to the DERVATIVE of volume flow (multiplication by jw). So while the mass integrates the force in terms of velocity and volume flow, the point source introduces a differentiation. The end result is a perfect compensation for the amplitude slopes AND the phase lag.
-------
"Lastly, it is cold and snowing here to right now, nude arc welding weather for sure.
For fun, experimentally alter the phase in a test response and see how visible this kind of phase shift is in the appearance of the impulse response."
-------This has happened to me many times measuring speakers, and the result is that the impulse response points downwards instead as upwards, just as I would expect.
-------
I can tell you that some measurement systems do not pass a reality check I devised to see if a technique actually can be trusted or not.
If you have a loudspeaker controller and pspice or other filter way to model filters do this check.
Set up an imaginary perfect speaker, start with a 3 ms time delay, add a 2 nd or 4 th order high pass at say 50 or 100Hz and 2 nd order low pass at 5Khz.
Model that response and phase in the computer as the “should be” case.
Now, measure full band and see if the amplitude and phase are proper relative to the “should be”.
-------Hmm, I am not sure what you mean by this. Is it a test of measurement systems? Or is it a test of an actual loudspeaker?
If we are talking about measurement systems I usually use home written stuff (I still haven't come to a software release of this, but I have experimented quite a lot with it), and they produce impulse responses that are just as would be expected both from electrical filters and loudspeakers.
I could imagine that there are software, however, that make assumptions about minimum phase (as you say) that are not perfectly valid.
-------
"To answer another question, this is not “my theory”; I exploited what I saw so make a multi-way horn loudspeaker.
I didn’t submit a paper for this yet, it was a local meeting, a presentation with measurements and then music with fireworks recording at the end.
Actually I dropped out of AES about 10 years ago but I will probably re-join as I had fun.
Here are few old papers I gave, most of which aren’t available for free.
Also the blurb on the talk and a link to a white paper that gives a better idea what I am up to.
http://www.aes.org/e-lib/browse.cfm?elib=11721http://www.aes.org/e-lib/browse.cfm?elib=5023
http://scitation.aip.org/getabs/servlet/GetabsServlet?prog=normal&id=JASMAN000096000006003828000003&idtype=cvips&gifs=yes
http://adsabs.harvard.edu/abs/1994ASAJ...96.3828D
http://www.aes.org/sections/chicago/aes_notice_feb2007.pdfhttp://www.danleysoundlabs.com/pdf/danley_tapped.pdf
Tom Danley"
-------
Ok, so I can tell that you are a "horn guy" :-) . It seems as if you have done great things with horns, it is just this little thing with the direct radiating speakers that I disagree with you on.
I'm sorry if my entering into this discussion was a bit blunt.
/Svante
Hi Tom,With your square wave comments, you seem to be implying that the synergy horn system IS capable of passing a square wave, at least over a certain square wave frequency range. Have you had a chance to publish any square wave measurements? Thanks for all the great information!
HiYes they can over some range.
At the presentation, I had photo’s from an oscilloscope taken at the standard third octave centers. The SH-50 does it anywhere from very good to fair from 220Hz to about 2600Hz.
There is some info at the web site and a picture of what is inside Danleysoundlabs.com
Best,Tom Danley
In all fairness, the diaphragm in a horn is not immune to breakup and the additional acoustic load may actually exacerbate the problem. Further, horns provide uneven loading at low frequencies where the horn is less than 1/4 size. This is linear, but still represents rapidly shifting phase. Throat distortion isn't an issue except at very high SPL, but it does introduce non-linear behavior. Magnetic non-linearities from the movement of the voice coil introduce non-linear behavior. All these things prevent a horn from operating with zero phase shift, same as a direct radiator. There is only the advantage of acoustic load, which is significant, but certainly not the only contributor to transient performance.
Cripes V, are you becoming a Stalker who shows up in every thread that Tom Danley participates in to spread disease? Please don't drive him away from this and other forums, he has so much to share and is generous with his knowledge.
Benford's law of controversy - Passion is inversely proportional to the amount of real information available.
Wasn't trying to offend, sorry if I put you off. What exactly did I say that was offensive to you?
Hi V
Hey hey, its you again.
I remember you, do you remember this thread where you “suggested” I used Martin Kings work to develop the Tapped horn?http://www.audioasylum.com/forums/hug/messages/118585.html
Did you also happen to catch Martin King, commenting on that same thread and you, “not exactly” backing you up on that ?
http://www.decware.com/cgi-bin/yabb2/YaBB.pl?num=1168762375/15
I don’t know it just stuck me as ironic that you would begin by saying “in all fairness”.
I mean “in all fairness”, with the “fatal problems” you seem to see, how would you explain the fact that the speaker actually does (as in the mic signal viewed on an oscilloscope) reproduce a square wave over a range spanning all three sets of drivers?
You talk nonlinearity as a show stopper, just what effect do you think gobs of headroom has?
These speakers were independently measured and they found they change their response less than 3 dB up to 56Volts shaped pink noise (about 780 Watts average) which has peaks +6 dB (3120 Watts peak) over that average. .
At a little over 100dB 1W 1M, how hard do you think do you need to push them at an AES meeting at a Hotel?
The only thing that reached any significant peak power (where I had set the level just below clip on peaks) was the fire works and there I needed the subs and could have used more power.
Fireworks are not like music however, they are technically very hard to reproduce and they have a huge peak to average power ratio. They also don’t sound right unless all the energy arrives like in real life and so were part of the AES demo.
I would like to think that time part also helps with music and voice reproduction too.
Tom
Regarding the current topic, which is loudspeaker non-linearities, my point is that if one were to be objective, they would find that all the same non-linearities that caused a direct radiator to become non-minimum phase also cause a horn loudspeaker to be non-minimum phase. That doesn't make anything a "show stopper" except of course, marketing blather.About the MLK spreadsheets, the beauty of them is they model standing waves in transmission lines with driver offset. They are able to calculate nodes at 1/4, 3/4, etc and this allows the DIY builder to position the driver some distance down the line to reduce notches from standing wave nodes. The flare is able to be modeled too, although the fold isn't considered, which will matter at higher frequencies if you make a folded horn. Still, the advanced spreadsheet is very close to being able to model exactly the configuration of your tapped horn. I would have thought this would be interesting to the DIY community, and flattering to you.
Martin King said he hadn't made a spreadsheet specifically to model your "tapped horn" but did say he thought his spreadsheet would just need to be modified slightly for that purpose. He went on to say he didn't think he wanted to make a spreadsheet that would model it exactly because he wasn't sure there was any value in doing so.
Quoting MJK from here:
http://www.diyaudio.com/forums/showthread.php?postid=879165#post879165
I thought about the problem yesterday and concluded that a worksheet to model this type of enclosure is possible. It would require some rearranging and extension of the math but I believe it could be done. Unfortunately, it is not on my list of priorities and can only be classified as something interesting to look at if I ever run out of things to work on (there are a few other enclosures on that list also).
So let me ask a couple of questions :
1. What is special about the performance of this design compared to a classic TL?
2. What performance advantages does it offer?
3. Where is this design used and over what frequency range?
If there is some performance advancement possible with this geometry that would be interesting, if this is just an old design cobbled together by a very creative thinking individual that sounded good compared to state of the art at that point in time then I am not sure this is worth pursuing. Is the Jensen Transflex just another Karlson or Hageman style of exotic, outside the norm, or curiousity enclosure that appeals to a small group of enthusiasts or is it truely a high performance design that has fallen through the cracks?
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