But I am pretty sure that there is no velocity field,
page 35
But I am pretty sure that there is no velocity field,
Basically, type of the crossover filter transfer function and order of the filter are most important for directivity plots, together with driver axial distances and height difference. There would be a lot to speak about, however better to get the literature.
2nd order crossover, point source radiators, at crossover frequency.
3kHz crossover frequency, 14cm distance between point sources
Indeed that distance makes things "extremely interesting".
If you have transfer functions for each step then you have what you need. Of course there will uncertainties, but the principle is as stated.And would the sum of many steps/stages not create and unpredictable result? COuld you not daisy chain many filters to simulate it and then listen for perceived differences?
If there are pressure gradients then there are particle velocities. This velocity is different from the velocity of wave propagation.I was hoping that on a forum you could explain it.
But I am pretty sure that there is no velocity field, and that it is only a pressure field… with the speed of propagation being a constant at any temperature.
The pressure field may have a vector describing the direction, which is radial for a point source… and it beams for a larger driver at a high frequency.
Where is the volume? Is it a Spatial volume? Or is it a Sound volume?
Some more examples of vertical directivity, again with ideal point sources, at 3kHz crossover frequency
1) 2nd order Butterworth filter, distance of sources d = 140mm (5/4 lambda)
View attachment 237071
2) same as (1) but d = 113mm (d = lambda)
View attachment 237072
3) d = 140mm, but 1st order crossover
View attachment 237073
Odd order filters are usually not a good choice because of non-symmetry in the vertical directivity.
These examples still deal with ideal point sources that produce a spherical wave regardless of frequency. In a real speaker, the directivity is a result of such plots as shown here with the directivity of speaker drivers used. Yes, "interesting". So, is phase important, or not?
Newsgroups: rec.audio.high-end
Subject: More on loudspeaker performance
From: John Dunlavy <[email protected]>
Date: 23 Jan 1997 17:44:58 -0500
A recent comment stated that I, "... regularly veer off into the time domain and phase matters without offering any new (or even old) evidence of why distortion (or the absence of distortion) in these areas should really much matter to the human ear auditioning playback in an enclosed space, especially a domestic-sized one.", is good and deserving of a detailed answer.
First, the time and frequency domains are related to each other, mathematically; sometimes in a simple way and sometimes in a more complex manner. For example, events in the time domain (such as changes of signal amplitude with time) always produce events in the frequency domain, with components that are generally related by the reciprocal of the variations in amplitude with respect to time. Further, a short impulse in the time domain contains implicit information regarding frequency response, phase response, step response, etc., all of which may derived by FFT analysis of the impulse. (Doug Rife's now famous MLSSA system is an excellent example of using FFT analysis of an impulse to determine most loudspeaker performance attributes.) Once understood, the step response (merely the integral of the impulse response) provides a quick visual evaluation of several time and frequency domain properties of a loudspeaker.
Of course, most audiophiles, without an appropriate technical and math background, probably cannot visualize how so many of these seemingly diverse performance properties can be so directly related to each other - but they are! The only exceptions are the impedance, radiation patterns and non-linear distortions (THD and IMD).
Linear signal distortions, such as a poor impulse/step, large variations in the modulus of amplitude and phase Vs. frequency, etc., can directly affect the "perceived realism" of complex musical waveforms reproduced by any component within the audio chain, including the loudspeaker. The acoustical reflections from the floor, ceiling, walls, etc. of the listening room add another set of "time and frequency domain distortions" onto those of the audio equipment, though with a very different signature in the time domain. The difference is that most time-domain distortions created by equipment, including loudspeakers, typically occur within a time window of less than a mSec, compared to several milli-seconds for an average "floor reflection" and even longer for wall and ceiling reflections within most rooms.
Also, a property of human hearing, known as the "fusion time" (which is the interval of separation between short transients required to perceive whether one or two transients are present), permits a "critical listener" to discern between room reflections (normally arriving more than about 5 mSec after the direct sound) and time-domain distortion created within audio components, including the loudspeaker (which occur within a time window typically much less than 0.5 mSec.). Thus, fusion time helps us to become familiar with and ignore most reflections from room boundaries, while letting us discern time domain distortions produced by equipment/loudspeakers as a blurring/smearing of musical transients or an alteration of spectral balance, etc.
With respect to your question as to what "... blind music-playback tests you and your colleagues have performed with the ***only*** variable being changes in "phase" behavior, with results showing a preference for one kind of behavior over others?", we have conducted quite a wide spectrum of tests during the past 20-plus years. We spent the time and money to do the research for two reasons: 1) it appeared to have been ignored by other investigators, and 2) it represented a unique technical challenge, was intellectually interesting and potentially commercially rewarding. (Commercially rewarding in the context of permitting us to better understand certain design goals and performance constraints, leading to better performing and more salable products.)
With regard to the audibility of changes in phase Vs frequency of a loudspeaker (or any other audio component, for that matter), our investigations have tended to show, pretty conclusively, that small changes of less than about 30 degrees per octave are probably not audible - except when listening to certain complex signals such as impulses and square waves. However, phase changes exceeding 180 degrees within an octave (or less), created by a loudspeaker crossover network, non time-coherent drivers, etc., are usually audible with complex waveforms and musical transients, when compared to the unaltered original signal. This is especially true if the 180 degree phase shift occurs within a octave or less within the frequency range from about 200 Hz to 5 kHz (often referred to as the mid-range region of most loudspeakers). I could go on and on about this interesting subject but time presently is not available to do so. It is shame that more investigators have not explored this generally neglected but very important ground.
Last, but not least, I would like to remark that if I had to choose but one measurement of a loudspeaker from which I would be required to determine and describe most of its performance properties, including frequency response, phase response, impulse response, etc., it would be "step response". For those familiar with step response and what it has to reveal about so many important properties (at a mere glance), it is a true gem in disguise.
Best regards, John Dunlavy
Dr Toole addressed this "issue" years ago in a post in another forum. Quote from his AVSForum post:...
So, is phase important, or not?
*It seems more recently researchers have managed to find signals that lowered the detection threshold of group delay to ~0.6 ms using headphones. (Reference)Phase absolutely matters in the design of loudspeaker systems because in the crossover region the output from the woofer (say) must add appropriately with that from the tweeter if there is to be a smooth transition between the two. It has to do with the performance of the overall system, achieving good on and off axis frequency responses in the crossover region. So the transfer function of the woofer (amplitude and phase) is measured, and that of the tweeter, and nowadays the data can be fed into a computer simulator/optimizer that will help achieve the best acoustical summation. However, once the system is designed, the phase (shifts) in the radiated sounds are unimportant. In fact, group delay up to about 2 ms is simply not audible - surprising, but verified more than once by serious researchers
The concept of acoustic impedance draws the relationship between acoustic particle velocity and pressure. (The screen snips are from this: https://www4.uwsp.edu/physastr/kmenning/Phys115/Link5-09_acoustic_impedance.pdf)I was hoping that on a forum you could explain it.
But I am pretty sure that there is no velocity field, and that it is only a pressure field… with the speed of propagation being a constant at any temperature.
The pressure field may have a vector describing the direction, which is radial for a point source… and it beams for a larger driver at a high frequency.
Where is the volume? Is it a Spatial volume? Or is it a Sound volume?
Dr Toole addressed this "issue" years ago
This. I was going to post a link to my main acoustics text, from grad school, but the equations in this paper/text are the same as we waded through. I can't decide whether to shudder anew at the complexity or be amazed that once upon a time I understood all the math.
Odd order filters are usually not a good choice because of non-symmetry in the vertical directivity.
These examples still deal with ideal point sources that produce a spherical wave regardless of frequency. In a real speaker, the directivity is a result of such plots as shown here with the directivity of speaker drivers used. Yes, "interesting". So, is phase important, or not?
If the design of crossover is not good enough due to cost limitation,Dr Toole addressed this "issue" years ago in a post in another forum. Quote from his AVSForum post:
However, once the system is designed, the phase (shifts) in the radiated sounds are unimportant.
At some point we left the idea of electric signals and how the drivers combine from a crossover to get to acoustic fields.
Did that seem to happen a bit suddenly?
Some more examples of vertical directivity, again with ideal point sources, at 3kHz crossover frequency
1) 2nd order Butterworth filter, distance of sources d = 140mm (5/4 lambda)
View attachment 237071
2) same as (1) but d = 113mm (d = lambda)
View attachment 237072
3) d = 140mm, but 1st order crossover
View attachment 237073
Odd order filters are usually not a good choice because of non-symmetry in the vertical directivity.
These examples still deal with ideal point sources that produce a spherical wave regardless of frequency. In a real speaker, the directivity is a result of such plots as shown here with the directivity of speaker drivers used. Yes, "interesting". So, is phase important, or not?
MTM designs do help in some regards. In some cases pointing the pattern zero in a particular direction has also proven quite handy.To get around the asymmetry with odd order crossovers you use vertical symmetrical array as with Duntech and Dunlavy speakers
cheers
That's some dedication!I'm an admitted "phase-oholic", who strives for flat phase (constant delay) in all my DIY speaker builds. (Flat magnitude is taken as a given.)
I truly think i hear superior results from doing so, but cannot say the results are solely/mainly due to achieving flat phase.
A big part of the excellent results may simply be how much easier it is to tune a speaker with complementary linear phase xovers, than with traditional IIR xovers.
Tying drivers together with xovers that don't have phase rotation .....is laughably easier than working with IIR, ime.
Especially so, if the complementary linear phase xovers are very steep.....then the driver's critical summation/overlap regions become narrower, and waay easier to make well behaved for excellent magnitude summation. Let me repeat waaaay easier....
That ease of obtaining excellent magnitude measurement results with steep xovers, may have as much/more to do with my perception of audible benefit, as does the flat phase....i dunno.
But If nothing else, steep reduces lobing by confining lobing potential to a narrower frequency range.
I don't think steep IIR xovers are viable for all the well known reasons (which makes a case for phase audibility in it's own right, huh?)
Anyway, that's one of my two tactics for trying to achieve flat phase....which can be summed up as first tuning individual drivers both in-band and out-of-band with minimum phase EQs, and then tying the drivers together with steep complementary linear phase xovers.
The other main tactic is simply trying to minimize reduce the speaker's c-2-c distances between driver sections.
Synergy horns are the best method I've found for that so far. I'm measuring polars on one right now...
If you look closely, there are 4 mics on the mast, vertically set at 0, -10, +15, and -20 deg. It's my first attempt at getting serious with vertical polars for my synergy builds.
Horizontal polars have been easy to capture with a lazy-susan spinorama,
but the way i build synergies doesn't allow them to rest on their sides on the spinorama....hence the staggered mics.
Anyway again, I added the synergy project and measurement rig to show all my 'wind about working to achieve flat phase' is true wind, even if ultimately hot air
View attachment 237377