Sloped baffle


Some great speakers have it, some don't. Is it an important feature?
psag
That's how I was presented this subject. A good link, thank you. Made me flash back to all the horrible homework involved. And then, as the math of physics became ever more advanced during grad school, one wound up using this math daily...

Best,
Roy
Here are my answers to important questions posed earlier, and some clarifications.

To the OP: Psag, you originally asked if a sloped baffle is important. Speaker designs that avoid this are instead using the phase shifts of their crossovers to make sure there are no cancellations/suckouts in frequency response. That is about all their designers look for/measure during the design phase, since they do not make any measurements in the time domain.

I think those designers would have an easier time developing their high-order crossovers if their drivers were first stepped back from each other, as on a sloped baffle, and they got rid of the sonic reflections off their front surfaces.

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Bifwynne, at the beginning, you asked "perhaps someone could explain in layman's terms what causes speaker to operate out of phase. Does it have something to do with the use of caps and chokes in the x-over? Or perhaps the attribute of a dynamic speaker creating its own back EMF by reason of the voice coil moving in a magnetic field??

Incidentally, do all these electrical dynamics operating in tandem cause the electrical phase shifting that gives most amps a headache? "

Let us begin with the phase definition. If a speaker's woofer and tweeter were out of phase more than 'a bit', they would show a dip or even a complete suckout in frequency response, at or near their crossover point, with the microphone placed where your ear would be. As we see from Stereophile's tests, most speakers do not have this issue. So all of those must be "in phase", "phase coherent", "phase linear", or "phased aligned" As I explained earlier, that does not mean they are time-coherent speakers. As a reminder, the opposite IS true: time-coherent speakers are always phase coherent.

What makes the phase go weird?
-- In the speaker cabinet, it is from the drivers' locations/no stepped baffle, and having too many drivers per frequency range.
-- Any crossover circuit's inductors and capacitors delay the signal or advance it, respectively. Resistors do neither. A simple first-order crossover circuit has an inductor going to the woofer, and a capacitor on the way to its tweeter. At their crossover point AND ALL other frequencies, the time-delay created by the woofer's inductor is precisely offset by the time-advance created by the tweeter's capacitor. This is not possible with higher-order crossovers, because the values of their more-numerous inductors and capacitors cannot offset each other.
-- The back-emf from any driver is also a contributor to time-delay in its lower-range, whether woofer or tweeter. Thank you for pointing this out. I should have mentioned this earlier. That back-emf situation is altered by the type and size of the cabinet behind a woofer, and the size of any rear-chamber on a tweeter, and from ferrofluid in its magnet gap.
-- Any cone or dome breakups change the arrival-time as we go up the scale, but mostly we would hear ringing, sibilance, maybe 'dirt' being added to the music. Regardless, the best cones will not show a loud ringing at some frequency (as with most metal cones available in 2014) nor have a ragged frequency response in their upper ranges.
-- And yes, all these phase shifts will talk back to the amp. However, the crossover circuit's design is the primary cause of large swings in a speaker's impedance curve, above 100Hz. Those variations are 'electrical phase shifts' only. These swings in impedance do not reflect the acoustic phase at one's ear- no direct correlation.
The amp gets a headache because large swings in impedance means its output voltage (the pressure it puts on its electrons) is no longer sync'd up with WHEN those electrons are allowed to move by the crossover parts (inductors and capacitors). When the values of those caps and inductors do not offset each other, the result is exactly like pushing a child on a swing at the WRONG time.

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Bifwynne, on the first page, you speculated on the effects of mics, of recording and mastering, processing, playback, etc.

Each of those areas has unique problems, which do not sound like phase shift from a speaker. Each process produces a time delay in the highs and sometimes the lows, but only a speaker can put phase shifts (plural) across the main tone range. Also, whatever that signal is, I see no reason for home- and studio-speaker designs to distort it more.

On that same page you asked
"How are small speaker manufacturers able to design speakers without the benefit of the R&D budget, engineers, and testing facilities that some of the larger manufacturers have at their disposal?"

For me, it's been knowledge, education, and longer, much wider experience. My talent seems to have been expressed as an ability to make the cognitive leap between seemingly unrelated factors, which then made one more link to hearing vs. measurement. All of this has led to me not needing an anechoic chamber (I can always go outdoors for that). I also found the fancy digital test gear gave misleading and often incorrect numbers, compared to analog test gear.

When a designer does not really understand the fundamental physics of how and why drivers move and respond as they do, nor how crossovers delay the signals, then their only recourse TO IMPRESS their board of directors, is the anechoic chamber/digital route, for that is what the AES and any university would also advise those board members responsible for hiring 'a great designer'. Such a designer then blames the sonic differences between his and other speakers as 'we all hear differently'. His board of directors and all reviewers and editors gladly go along with that bullcrap.

We all certainly listen for different things. But here we have found, that as a speaker is made more and more time-coherent, everyone AGREES on the sounds heard in each and every tome range. They all hear 'the bass' in the same way, etc.

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Ohlala, on page one, the possibilities of off-axis cancellations you mention turn out to be non-issues on music, especially when the cabinet is not large, and has little sonic reflections from its surface.

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Timlub, on page one, your speaker design is only phase coherent at its crossover point, not time coherent, as you may know. Your electrical crossover slopes work well because they are combining with the phase shifts of your particular woofer and tweeter, which I am sure you suspect. Thank you for sharing your experiences! Appreciated.

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Bifwynne, you ask too many (good) questions! On page one you ask,

"here the ultimate Q. How can one tell whether a speaker is time and phase coherent? Critical listening? Reviewer comments? Bench test?

If critical listening is that important, the real challenge for us is, as many have written, that it is not easy to meaningfully audition speakers. So what's a person to do?

I'll ask again, how important is time and phase coherence? FWIW, ... really more as an FYI, ... Paradigm's web site states that its 'speakers have phase coherent crossovers designed so that the summed output of the drivers is completely and accurately rejoined.' Is that hype? It is true at all frequencies?"

On my website, I have suggestions on how to audition speakers. I know these work. They are simple, taking only time and effort. The time-coherence part of the audition sounds like clarity and depth, and when time-coherent speakers are designed with the best parts, the musicality is greatly improved.
With the very best, you find yourself never, ever thinking about 'the sound of the bass' or 'the highs'. Instead, you subconsciously always focus on the music and how it is being played, and its emotional and physical connection to you.
When a speaker is time-INcoherent, the music is fragmented, leaving you to hear only 'the details' and 'the soundstage' or 'the air', or 'the impact'. Right now, I see only Green Mountain Audio, certain models from Thiel, and Vandersteeen as making time-coherent speakers. The Audio Machina company is part-way there. With any others claiming time-coherence, I've seen no proof on their websites, or in Stereophile tests.

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Ivan_nosnibor, I appreciated your thoughts, thanks. However, the time delays in your digital crossover circuits are fixed time delays for each driver, when the real problem is the amount of time delays are different at each frequency. You remark on hearing perhaps the highs 'imaging closer to you' on non-time-coherenet speakers, with the mids 'not projecting as far into the room', and so on.

I have found instead it is about the lack of depth in the highs, caused by the smearing of a late-arriving mid, and so on down the musical scale. WHEN the highs arrive is not WHEN you hear the image, but only a portion of that image. One example is hearing the esses and tees of the singer's voice arrive from the tweeter's location above the mid, not from the mid driver's location, where the main part of her voice comes from, listening with eyes closed. That is one sound of a tweeter arriving too soon. It can also sound like the band is leaning forward, for want of a better word. It can sound like the rhythm section is behind the beat (as they would be in those speakers).

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Almarg,

Your described a square wave as "the summation of an infinite number of sine waves, one being at its 'fundamental frequency"' (the frequency with which its pulses repeat), plus others at every odd multiple of that frequency (i.e., the 3rd, 5th, 7th, etc. harmonics). The amplitude of each harmonic decreasing as its order (i.e., its frequency) increases." This is all true, but only of an ongoing series of square waves. The analysis is somewhat different when we examine just the first up-cycle, without even the first down-cycle following it. Just an FYI, seemingly never mentioned on the internet nor in textbooks.

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Mofimadness, the Loudspeaker Design Cookbook is generally excellent, but all previous issues got the concepts of phase time-coherence somewhat wrong. It has been awhile since I looked over a copy, so I can't remember where the problems showed up. I advise to take its advice with a modicum of salt.

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Bfwynne, the Revel 2 and Magico have oodles of phase shift, mostly from their crossovers. What you are seeing in the Stereophile tests is just as John Atkinson says- the mid and woofer take longer for their sounds to arrive. What is not readily apparent is how the phase (time delay) is changing at EVERY frequency. Otherwise, one could fix the Magico and Revel 'problems' by moving their tweeters back, etc. Actually, Almarg gave you a very excellent answer.

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Usermanual, you ask about us proving we are time-coherent.
1) This would not change our sales.
2) It cannot be done in a singular graph or 'scope image useful to a layman, by anyone including us. This is not a case of sour grapes- please read my letter to sixmoons regarding the issues with measurements. Note some of my graphs do not line up correctly with my text on their website.

In the 1994 Stereophile test on our Diamante three-way, remember JA always measures at 50 inches, right in front of a speaker's tweeter. That makes ANYONE'S mid and woofer too far away, relative to the tweeter.

JA then moved his mic straight down, to get farther from our tweeter, closer to the mid and woofer, looking for our claim of time coherence. You see our step response get sharper, more compact. But our frequency response goes to heck because he is now going VERY far off-axis of both mid and tweeter. Again, this test was done in 1994. In the intervening twenty years, every aspect of our sound, and of any measured performance, has improved.

Above all, trust your ears more than measurements and reviewers. My letter to sixmoons shows why this has to be so.

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This covers page one, I think. Perhaps page two will be much, much shorter.

Best,
roy
Thanks Roy. I think I have the instinct, but not the brains for the math and physics. B'li neder (not a vow), I will teach myself the math and physics when I retire as Caeser's tax collector.

So ... I'm all set up. Nice electronics, good music ... listening to some hi-rez redbook CDs right now, ... good looking wife and ok speakers. Why not just go for the DEQX and call it a day?

Btw, just anecdotally and IMO, I think my Paradigm S8s (v3) are made of decent kit: beryllium tweets, aluminum-cobalt alloy mids, good woofies, tweets and mids are ferro-fluid cooled and damped, super neodymium magnets in the tweets and mids, 20,000 gauss magnetic flux density in the tweets and 15K gauss magnetic flux density in mids and woofies. What am I missing except time coherence?

Why not DEQX?

Why not DEQX?
Bifwynne
Bifwynne,Roy has already answered your question some time back on this page 4. Did you miss reading it??
Here's a cut & paste from his 7/16/14 post:
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.

here's more info from Roy on DEQX in his 7/17/14 psot to the whole group
This varying time-delay is what DEQX-type components are trying to correct, and what regular digital crossover circuits never attempt to correct (offering only fixed time delays, such as one millisecond). To correct the varying time delay, a heck of a computer is required, hence the high cost of DEQX type of gear.

Measurement issues and limitations still confuse DEQX type of gear, for two reasons- we cannot (yet) program that computer to how we actually hear on music, and that a measurement microphone cannot resolve the (countless) reflections off the front of a cabinet. If I had spent money on a DEQX, I would first place an "F-11" pure wool felt all around the tweeter, and then run the calibration routine.

Best,
Roy

more info on why DEQX has limitations from Roy's 7/19/14 post:
Putting the measuring mic for DEQX up close to a speaker is pointless (except for fixing up a subwoofer), as what the mic would then be hearing is coming from drivers at much different path-length-differences to the mic compared to the path-lengths to an ear ten feet away. We all know how walking up to a speaker changes everything we hear. Perhaps they are suggesting this for fixing one driver at a time. That has problems too, because any driver's tone balance is different at ten feet away vs. ten inches away.

Bifwynne, this is plenty of info for you to understand why DEQX has limitations & is not a panacea for time-INcoherent speakers. Don't you think?

Roy,

Thanks for your fantastic contribution here. We can only hope for more really knowledgeable folks like you to take the time and educate us hobbyists.

On the 1st or 2nd page I posted about a way I was intending to get at this. I have, at least for this purpose, the advantage of having only an optimized computer as audio source. My plan is to use Acourate software on the server, a multichannel DAC, and independent amps connected directly to each driver, without passive crossovers. Acourate allows the use of a variety of digital crossovers, and allows for time alignment of the drivers. BUT it is limited to a single time delay between any pair of drivers, much like the limitations you describe for DEQX (which I previously considered too, but a needlessly expensive option if the only source is a computer).
Clearly this will not solve 100% of the problem - something I learnt from you. But what's your very educated guess: will it solve maybe 80% of the problem vs a non-time-aligned 3-way speaker?

BTW, would love to get your thoughts about this XO white paper by Dr Uli Bruggemann, the guy behind Acourate.

As of now I'm using B&W 804S. Obviously not time-aligned. Probably not even phase-coherent. So the setup described above would first be used with these speakers. And eventually I'm thinking of building my own speakers using top-notch drivers, the Loudspeaker Cookbook as guide. I'm a mechanical engineer and handy building stuff. Assuming I do a good job selecting drivers and building the cabinets...sounds like I'll end up with very good speakers in terms of bang for buck...what do you think?