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[Langston Holland]

What made me triggered is on old post of TD:
http://mailman.soundlist.org/pipermail/sound/2007-January/027323.html

If nothing else came out of this thread, that link alone is worth its weight in gold. I never saw that before - thanks Marcel!

I find it interesting that a omni point source with flat magnitude, measures a (fixed) time difference of almost -90 degrees between a in- and output signal.

Technically, this refers to a fixed phase shift of -90 degrees. A fixed time difference (or delay) would result in differing amounts of phase shift at each frequency. It's obvious you know this - I'm just clarifying. Also, this applies to the acoustic phase of most loudspeakers, not just omni point sources. Horn loading if done right can correct this at the cost of a minor fixed time delay. The addition of electrical high and low pass filters can mess it up again. Electrical filters (such as 2nd order all pass) if done right can also correct for the phase shift at a minor cost in fixed time delay. Audio is amazing. :)

What i want to do is to build a circuit (as simple as possible) with some resistor, cap and inductors, that measures the magnitude into that load flat, but with a (fixed) phase delay between input and output. As i see it you have to use some kind of frequency depended resistor to achieve this.

A simple inductor will simulate the current phase lag of -90 degrees throughout it's linear region (use air core), but I don't think this is the best solution.

The circuit is than a good reference for me to use different analyzers, to see how they measure and see the plot, from a known load. I know a can made a compare with an acoustic measurement, but i want to do the electrically, so i can really exclude any uncertainty.

Now for the meat of the "why" behind all this, are you mainly trying to determine the acoustic phase accuracy of different measurement systems?

Edit: if an electrically based test to determine the acoustic phase accuracy of a measurement system is your goal, click here. Also, there is one huge obstacle to phase accuracy with any measurement system - the organic component. :)

[Frank Koenig]

If nothing else came out of this thread, that link alone is worth its weight in gold. I've never saw that before - thanks Marcel!

Indeed! Tom D. is a great explainer. I've ploughed through Beranek a few times (and Thiel, Small, etc.) and while I'm somewhat conversant in electrical/mechanical/acoustic analogies and network analysis, I never came away with that. Gives credence to the notion that only folks with a deep and true understanding of a subject can explain it well to others. Required reading. --Frank

[Marcel de Graaf]

Edit: if an electrically based test to determine the acoustic phase accuracy of a measurement system is your goal, click here. Also, there is one huge obstacle to phase accuracy with any measurement system - the organic component.

Indeed this is what i want to investigate and see it by myself. I have another paper, that was sent from nice fellow at the AA board, that has essentialy the same explanation as TD. The author is Richard Stroh. The paper has some graphics attached to it. If you interested in it you can contact me. 

The way as i read it and i see it now is as follows (correct this if i am wrong):

What is been measured with a microphone is a pressure level of the acoustic power delivered to the air. It`s only the resistive part of the airload (radiation impedance) where this power is dissipated to.

If this power is constant over a range of frequency, while the resistive part is changing, than this is only possible with a compensation of the reactance parts. This would mean there is a phase shift (angle) and it is constant.

Its this phase shift, under that conditions, i want to see in my measurement. I think this is the real acoustic phase phase. I have a little doubt that a measurement systems with a steady state signal would show this. If measuring a electronice device this would show the correct phase, because there is no changing resistive part, but acoustics this can be different. 

It sounds like what you're trying to do is to build an analog Hilbert transformer. If you hit the literature you'll likely find ways to approximate one over a limited range of frequencies. A straight integrator or differentiator will give you a 90 deg phase shift but, of course, the magnitude will vary as 6 dB / octave.

Its indeed kind of sort Hilbert Transform but than totally analog. I can`t find how this can be build with passive parts.

gr. Marcel
 


 

[Langston Holland]

Hi Marcel:

You really don't need to bother with acoustics to verify the phase accuracy of a measurement system. Tom's method will work and it has the secondary benefit of teaching you how to properly remove the time-of-flight (TOF) delay from a measurement so that only the behavior of the DUT remains. That is essential.

If your system measures electrical domain stuff correctly, it'll measure any other domain correctly as long as the transducer you use (measurement mic) doesn't introduce errors.

Another thing that is extremely helpful in this pursuit is to use a measurement system that can display measured phase as well as the two calculated phase plots; minimum and excess. Using Tom's approach, you will know you've removed TOF correctly when your measured phase plot is parallel to the minimum phase plot. The excess phase plot makes things simpler by subtracting the minimum phase from the measured phase to show the difference. If you end up with at flat horizontal line, there is no difference and your measured and minimum phase plots are parallel.

For those playing along at home, this only works for minimum phase devices, such as this electrical filter phase test described by Tom.

[Mark Wilkinson]

Edit: if an electrically based test to determine the acoustic phase accuracy of a measurement system is your goal, click here. Also, there is one huge obstacle to phase accuracy with any measurement system - the organic component.

Hi all,

First, Marcel thanks for the link to Tom's post about current lag and acoustic phase...it's making me realize yet again 'I know nothing'...
Before even trying to ask questions about that link,

I'd like to swerve to the link that Langston gave, which has Tom's test for checking the accuracy of phase measurements.

Hopefully, the Smaart plot below shows the test correctly.  It uses a 4th order HP at 100Hz, a 2nd order LP at 10kHz, and 4ms delay.
The green trace was made with Smaart's delay manually set to 4ms, and I think? looks just  like Tom says it should .

But the red trace is from clicking on Delay Finder and letting Smaart set delay to its determination 4.02ms.
This removes the HF rolloff and pretty much takes phase to flat.

My take is that Smaart's delay finder  sets reference phase=0 to a frequency within the very high end of the measured spectrum.
But that kills the rolloff which I know is there....

So I guess it just comes down to, where in frequency, does phase=0 really reside ?
Seems to me, that 'phase=0 frequency' really needs to be above nyquist, because nyquist itself is a low pass filter, and anything below nyquist has already shifted in phase.....
I's almost like a little bit of time needs to be subtracted from whatever can be measured to see through to above nyquist....

Am I seeing straight?? 
Thanks,  Mark

[Marcel de Graaf]

Mark,

The setting of the TOF a very important thing, because its the reference for the phase plot. Refering the T=0 to the highest frequency that is possible is a good thing. Usually this is around 20khz (or a little higher).

Langston has a good video about removal the TOF:
https://www.youtube.com/watch?v=LFEwxEnpcos

In the case of your example a sec. order low pass at 10khz, means a phase rotation at 180 degrees as 20khz. This would mean it would take 4.025ms before the peak of the impulse response would arrive, from the ref. 20khz. The peak in the impulse is the midband of the magnitude response, where most energy is.

Langston,

I am still a little confused with that acoustic (point source, flat magn.) measurement. In the case the above example of Mark would be that acoustic measurement, the T=0  at 20khz would be a phase plot with a point at -270deg. at 20Khz, and the midband is -90deg. I doubt if Smaart would show this.

gr. Marcel

[Mark Wilkinson]

Mark,

The setting of the TOF a very important thing, because its the reference for the phase plot. Refering the T=0 to the highest frequency that is possible is a good thing. Usually this is around 20khz (or a little higher).

Langston has a good video about removal the TOF:
https://www.youtube.com/watch?v=LFEwxEnpcos

In the case of your example a sec. order low pass at 10khz, means a phase rotation at 180 degrees as 20khz. This would mean it would take 4.025ms before the peak of the impulse response would arrive, from the ref. 20khz. The peak in the impulse is the midband of the magnitude response, where most energy is.

Langston,

I am still a little confused with that acoustic (point source, flat magn.) measurement. In the case the above example of Mark would be that acoustic measurement, the T=0  at 20khz would be a phase plot with a point at -270deg. at 20Khz, and the midband is -90deg. I doubt if Smaart would show this.

gr. Marcel

Hi Marcel, rightly or wrongly, I've come to the conclusion that correct TOF isn't determined by the impulse peak.
I think it's determined by the impulse initial rise point.....that is,  t=0=initial impulse rise.
Because I think a perfect impulse is basically the first sample of all frequencies' initial rise summed together, to form the familiar dirac spike.
And that summation has to include the highest frequency, which has the finest timing and sets the stage for t=0.

So it seems to me t=0 has to be 'peak less one sample time' (or maybe less one-half?)
This is of course no big deal time-wise, peak vs initial rise, at 20kHz.
But it's big at 1kHz, and huge at 100Hz.  Trying to align multi-ways by peak impulse was killing me...which led me to seeing initial rise times.

I don't think phase can be referenced to any part of the spectrum, other than the highest non-rotated frequency. (or perhaps more practically, referenced to the highest  frequency "least rotated" by digital anti-aliasing filters)

Anyway, what's really blowing my mind, is what you linked about speaker movement following current, as opposed to voltage.
With only rudimentary understanding,  it makes me wonder if all our phase measurements are off throughout the spectrum, by a constant 90 degrees.
But such wondering seems pretty ridiculous.....even to think that something so fundamental has been being missed...
It makes me think programs such as Smaart account for current vs voltage....

Hey, if it is true that phase follows unaccounted for current, I've just about finished a filter that simply shifts phase 90 degrees across the spectrum LoL

[Marc Sibilia]

Anyway, what's really blowing my mind, is what you linked about speaker movement following current, as opposed to voltage.
With only rudimentary understanding,  it makes me wonder if all our phase measurements are off throughout the spectrum, by a constant 90 degrees.

Hey, if it is true that phase follows unaccounted for current, I've just about finished a filter that simply shifts phase 90 degrees across the spectrum LoL

Above the speaker resonant frequency, the movement of the cone is mass dominated, and acceleration is proportional to coil force which is proportional to current.  Acoustic pressure, which is what we measure with a microphone, is proportional to cone velocity (remember that (correction: specific) acoustic impedance (resistance for the radiated part) is pressure divided by particle velocity.  So acceleration is proportional to current (which approximates voltage away from the resonant frequency) and pressure is proportional to velocity.  Thus, a 90 degree phase shift.

Marc

[Frank Koenig]

acoustic impedance (resistance for the radiated part) is pressure divided by particle velocity.

Is acoustic impedance not pressure / volume velocity? Not particle velocity. -F

[Marc Sibilia]

Is acoustic impedance not pressure / volume velocity? Not particle velocity. -F

You are correct.  Specific acoustic impedance is pressure/velocity. My mistake.

Marc

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