Sloped baffle


Some great speakers have it, some don't. Is it an important feature?
psag
Thanks Roy. Again, you beeped. I'll counter with pop, bang, whack, pow, and all the other 60's Batman fight words that seem to have something else in common. Not arguing you're goal of recreating as accurately as possible but the words describe multiple, changing tones that appear to define our hearing ability more than the actual sound. So, my rhetorical question becomes: Is the b and p really in beep or is that our imagination?
Hi Roy, thanks for coming back. Your contributions are most welcome.
I have few questions for you.
Could you explain the pros and cons of making a speaker time coherent either by analog/digital/active/passive means?
If one were to use digital eq that only deals with room considerations and not speaker refinement, would there be a chance (and if so how much) of altering the time coherence of speakers such as yours?
The late John Dunlavy seemed to be somewhat unique (there might be another but for the life of me I can't remember the name of the manufacturer that was once a regular contributor here) in that he used W M T M W driver, as well as down firing woofer configurations. He told me that because of his previous experience in military antenna array technology he had more experience than most in wave propagation technology. Others who tout their speakers as time coherent seem to stick with more traditional W M T arrays. Is this due to size and marketing considerations, or something else?
Though most manufactures of speakers designed for time coherence seem to make fairly similar placement suggestions, they do vary a bit, from equi-T, to equilateral triangle, to wider than near, etc.. Why would that be?
It would seem to me that ideally a speaker designed for time coherence would have a sealed box, yet none of your current offerings seem to be designed that way. Am I wrong? If not, why aren't they?
Again, I look forward to people coming to understand the concepts behind my waveform illustrations. This understanding is necessary to our discussion here, because we then have agreed on the nature of these concepts at hand and also on some vocabulary.

And I wish my answers could be shorter, but that would leave out necessary details- the same ones glossed over/ignored by the press.

Bifwynne,
DEQX seems fine in theory, and certainly makes a positive difference. For me, it has serious limitations because it cannot measure exactly what needs to be corrected. This leads to results that depend on the music being played and sometimes a limitation in one's seating position.

In particular, DEQX cannot see the immediate reflections from the cabinet surface surrounding the tweeter. It cannot correct properly for anything happening below middle C because of floor-bounce effects on the microphone are not the same as they are to our ears on music.
There are other issues, but to me, those are the two largest ones. I find that a much higher level of coherence is achievable passively.

Regarding Treos-- I've not heard them and it's never good for me to comment on the sound of other speakers. I say to trust your ears above all. And you are right to listen backwards and from another room if possible! I can point out Treo, like other Vandersteens and most speakers for that matter, has a terribly complex crossover circuit, made of what I know are not the most transparent parts.


Ngjockey, thank you for your comments. Our experiences with everyday sounds and noises allows us to use them to imagine the SHAPE of a sound, which means its starting and stopping, and what happens in between. The next step up the chain is to imagine the combination of that sound with another, or literally just hear it via programming a synthesizer.

Remember, in the word 'beep', the opening 'b' has its own shape, since the lips are opening. That 'b' is is a CHANGE that happens along a certain TIMELINE, and we recognize its waveform's CHANGING SHAPE as unique to the letter 'b'. The same happens at the end when the lips close, but like 'p' instead. When you imagine hearing only the the middle 'eee', that is exactly what comes out of a sinewave test tone being switched on and then off. Which is exactly what I illustrate in my drawing.


Unsound, I am glad you find my comments useful. Thanks! Using digital EQ to treat room problems seems like a good idea, but again, just know that what you will measure is not what you are hearing-- not to say there will not be some or even a good amount of improvement. If it is used just for subwoofer correction, there are issues in most every sub's design that look exactly like room problems to the measurement microphone.

I think it best to first measure and correct a sub up close with the mic, then use that correction as your basis for further corrections YOU HEAR out in the room, listening to string bass run the scale and to kick drum (the first for tone balance; the second for transient alignment with the main speakers).

Mr. Dunlavy decided early on that driver symmetry about the ear was important, since it was important to his microphone and to his antenna-derived math. Turns out that when you are seated, the MTM arrangement is not important to the ear. In particular, you hear the tweeter's sound come from the mid when just one tweeter and one mid operate time-coherently without cabinet-surface reflections.

An MTM arrangement, including the infamous D'Appolito arrangement, always places one mid above our heads. This causes the image to be unstable with small head movements, and just plain poor for anyone off center. Why would this be? Because we have a head between our ears and a chest below them. Thus, with a small head movement to the left, much more middle-range sound literally leaks over the TOP of the head to the right ear than it does from the mid placed below the head, so the image jumps to the left speaker. This is also true for any large sound-source, such as a panel speaker or a so-called 'line-array', for the same reason.

When only one mid is used time-coherently with one tweeter, and then placed right in front of us, that single source of sound then leaks over the top of the head by the same amount no matter how much we rotate, move sideways or even stand backwards! So the image remains stable, even when far off-center, if and only if the speaker baffle is also narrow and reflections are not allowed from around the tweeter and mid.

MTM also leads to room placement issues, since sidewall reflections throughout the voice range are more complicated than when only one time-coherent tweeter and mid are used.

Time-coherent Coax operation ala Thiel would be best, except there's no way to avoid the intense tweeter reflections off the mid's cone. Also, a terribly complex crossover is required to get the tweeter's timing right. There are other limitations.

Finally, with two woofers placed high and low, for a WMTMW alignment, the bass response in any room becomes unpredictable, since you are driving bass near TWO boundaries, with your ears trapped in between.

Since everything is a compromise, a one-woofer arrangement works best when the woofer is a certain distance from the floor, in medium-size rooms, with a certain crossover frequency. But in those rooms, the bass output will then be predictable, which helps me. Nothing wrong with having the extra 'slam' from two large woofers- it just requires a very large room to make them perform as one. Then again, a very large room I find uncommon.

Speaker placement/spread is similar for very many speakers using slender front baffles, regardless of their crossover design, when these speakers are placed in 'good' rooms. This is because we need to hear a certain amount of crosstalk for the image to be continuous.

Sidewall reflections and reflections off all the fancy gear piled up between the speakers affects the final spread and the toe-in. Speakers having a large amount of reflections off their fronts are sometimes used with less toe-in, so those reflections are not shot as directly into one's ears. When there are many center-reflections (from that gear or off a video screen), toe-in is reduced. When a speaker is not time-coherent, its particular phase shift may mean those speakers sound best placed close together, pointed nearly straight ahead.

Sealed box is the best for woofers, but the market prefers more efficiency and compact enclosures, so our woofers are smaller, requiring a port. Our new three-way coming out uses twin 6.5-inch woofers, each ported at 40Hz in its own enclosure, for a sensitivity of 91dB with the same cone area as one 11-inch woofer. A single ten-inch sealed-box woofer would be in a cabinet half again larger, with only an 88dB sensitivity (requiring twice the power). The mid and tweeter would also need to be turned down by 3dB -not a great solution.

Again, I hope this helps! I realize other questions still remain, posed earlier in this tread, but I thought it best to get these out of the way right now, so I can look forward to folks' thoughts on my waveform illustrations. I will endeavor to cover the other questions soon.

Best,
Roy
Hi Roy,
good to note that you are back on this thread & have been kind enough to give us your time on this subject. Thanks!

yes, I personally have looked at the waveforms (on the photobucket.com website) you pointed us to. I understand it much better now thanks to your recent post where you explained the diff between time-coherency & phase-coherency. I was looking at the waveforms but did not draw that conclusion; now I have! Also, the 2 cars & 2 cyclists analogy was very helpful.
I have no particular question for you but I'm hoping that many other members who are on the fence re. time-coherence & others you are determined nay-sayers of time-coherence will take this opportunity of your being on this thread to ask their questions....

Bifwynne had a question re. the electrical properties of a driver & how that translated into distortion. Almarg enunciated the issue quite well & I've cut & paste his text below:

"Consider a simple two-way speaker having a first order crossover consisting of a capacitor in series with the tweeter, and an inductor in series with the woofer. For each driver that will result in well behaved 6 db/octave rolloff characteristics, which will result in time and phase coherence if other aspects of the design are also supportive, **IF** the impedances of the woofer and tweeter are purely resistive.

However I believe Bruce has been alluding to the fact that the impedances of the drivers are not purely resistive. And it would be more accurate (if still somewhat oversimplified) to electrically model them as consisting of a resistor and an inductor in series.

So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?"

Can you please address this question for us? thank you.
(My understanding of this question was that the driver is resistive in its pass-band frequency range where its response is flat. I understood that it could be flat response in its pass-band only if it was linear i.e. resistive over that range of frequencies but I could be totally wrong).
Consider a simple two-way speaker having a first order crossover consisting of a capacitor in series with the tweeter, and an inductor in series with the woofer. For each driver that will result in well behaved 6 db/octave rolloff characteristics, which will result in time and phase coherence if other aspects of the design are also supportive, **IF** the impedances of the woofer and tweeter are purely resistive.

However I believe Bruce has been alluding to the fact that the impedances of the drivers are not purely resistive. And it would be more accurate (if still somewhat oversimplified) to electrically model them as consisting of a resistor and an inductor in series.

So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?
thanx to the above question posed by Bifwynne & well enunciated by Almarg, I did some research to try to understand what the issue might be.
As I stated in my prev post, my understanding is that if the driver is used within its flat frequency response range of frequencies then that particular driver can be considered linear or purely resistive. And, some research seemed to indicated exactly that! Allow me to share:

When Bifwynne asked the above question, I discovered that it took the me towards understanding the Thiel/Small (or T/S) parameters of loudspeaker drivers. There's much to learn here but that's a subject for another day. Some info that I thought is pertinent to this subject:
There is a T/S parameter called Re (R-little-e) & a cut/paste from Wikipedia

"
Re
Measured in ohms (Ω), this is the DC resistance (DCR) of the voice coil, best measured with the cone blocked, or prevented from moving or vibrating because otherwise the pickup of ambient sounds can cause the measurement to be unreliable. Re should not be confused with the rated driver impedance, Re can be tightly controlled by the manufacturer, while rated impedance values are often approximate at best.. American EIA standard RS-299A specifies that Re (or DCR) should be at least 80% of the rated driver impedance, so an 8-ohm rated driver should have a DC resistance of at least 6.4 ohms, and a 4-ohm unit should measure 3.2 ohms minimum. This standard is voluntary, and many 8 ohm drivers have resistances of ~5.5 ohms, and proportionally lower for lower rated impedances. "

there's also a T/S parameter called Le (L-little-e)

"
Le
Measured in millihenries (mH), this is the inductance of the voice coil. The coil is a lossy inductor, in part due to losses in the pole piece, so the apparent inductance changes with frequency. Large Le values limit the high frequency output of the driver and cause response changes near cutoff. Simple modeling software often neglects Le, and so does not include its consequences. Inductance varies with excursion because the voice coil moves relative to the polepiece, which acts as a sliding inductor core, increasing inductance on the inward stroke and decreasing it on the outward stroke in typical overhung coil arrangements. This inductance modulation is an important source of nonlinearity (distortion) in loudspeakers. Including a copper cap on the pole piece, or a copper shorting ring on it, can reduce the increase in impedance seen at higher frequencies in typical drivers, and also reduce the nonlinearity due to inductance modulation. "

So, it looks like a significant source of distortion is due to voice-coil inductance modulation (variation) & not so much the fact that the voice-coil has actually a DC resistance associated with it (as Bifwynne & Almarg were thinking).
So, how to tell when viewing/reading a driver's specifications that this inductance modulation is an issue? I don't really know but I took up Roy's advice to look at driver specs on Madisound. On the Madisound I randomly selected "Seas Prestige" - Seas makes good drivers, "Prestige" seems like its upper-end line. Here's the link to one of their 8" woofer drivers:

http://www.madisoundspeakerstore.com/approx-8-woofers/seas-prestige-8-woofer-cd22rn4x-h1192

Lots of good info on this page but reading the specs might be Greek to most of us - I wanted to call your attention to the graph which shows SPL (left vertical axis) vs freq & impedance (right vertical axis) vs freq.

From a Wikipedia page on Speaker Electrical Characterisitics I learnt

".....the effective electrical impedance of the speaker to be at its maximum at Fs, shown as Zmax in the graph. For frequencies just below resonance, the impedance rises rapidly as the frequency approaches Fs and is inductive in nature.

At resonance, the impedance is purely resistive and beyond it—as the impedance drops—it behaves capacitively. The impedance reaches a minimum value (Zmin) at some frequency where the behaviour is fairly (but not perfectly) resistive over some range. A speaker's rated or nominal impedance (Znom) is derived from this Zmin value (see below)."

This Seas driver seems to have a 6.1 Ohms impedance at, say, 150Hz. Using the info from the Wikipedia site, the driver must be mostly resistive at 150Hz to give its minimum impedance at that frequency. Look at this driver's frequency response from 90Hz - 400Hz: practically ruler flat & look at the impedance variation over this same range - goes from 6.1 Ohms to 10 Ohms on both sides of 6.1Ohms, which is a small change in driver impedance compared to the change over the entire 20Hz-20KHz. The driver appears to be mostly resistive in this frequency range.
I *think* the answer to Bifwynne's question is that if you use this driver in the 90Hz-400Hz range, you will get a mostly resistive driver whose impedance varies very little (between 6-10 Ohms), it's frequency response will be flat/linear & the phase distortion will be minimal meaning that the voice-coil inductance modulation/variation (which is a significant source of distortion) will be negligible.

Roy, please correct me if I'm wrong. Thanks.
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