gnarly
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Posts posted by gnarly
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Hi Dave, i took the dcx464 / me464 combo active like shown earlier in this thread https://community.klipsch.com/index.php?/topic/200572-bc-me464-horn-opinions/&tab=comments#comment-2628703
I don't have the passive xover, so can't compare to "non-active".
From raw measurements of the two CD sections, I'd be surprised if sounded very decent without some active help.
Even with the processing i put in place, it hasn't been a combo i've felt was worth pursuing further, as i'm really enjoying DIY conical MEHs.
I agree with your assessment of it sounding more like traditional PA.
So happy to exchange thoughts via PM if you still desire, but i can't say i've had what i'd call really good results.
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Hi uams, thx.
The subs are dual 18 push-push slot loaded (clamshell style PPSL).
I basically took a very successful single 18 bass reflex build, and built the dual 18 PPSL scaling up the single's volume and port specs.
The cabinet vibration reduction vs single 18's has been awesome. I can get rowdy and walking subs have been a problem.
Are you currently using the MicroWreckers i see in your equipment list or the reflex boxes i see in your pict?
I bet both work great.
I agree that diffraction slots have gotten an unfair negative reputation....
No doubt there's some bad implementations, and bad CD/horn combos, using diffraction slots;
but here's some dang good ones too !
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6 hours ago, Chris A said:
You've apparently followed me to this forum from diyAudio, itself looking like a predatory gesture.
Yikes ! That's quite the imagination !
6 hours ago, Chris A said:It's a simple request: could you please start your own threads that you've set up for yourself and please not leapfrog into the threads where I've put in a fair amount of effort over time.
No problem at all, your request will be honored.
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On 8/24/2021 at 3:19 PM, Chris A said:
I do think you've got some issues with interpreting what you've got Mark, and your rationale for going the ways that you do with your MEH designs. This is not new. You may not know this, but perhaps half of the reason why I don't generally post at diyAudio anymore is due to history there with your approaches and methods (among a couple of others there).
I'm sorry to hear i was part of the reason you quit posting at diyaudio.
I don't understand why you would have a problem over approaches and methods that are different from yours.
Nobody's approach is the end all, be all.
Tis healthy to discuss various approaches and methods as long as respect and courtesy is provided to all parties presenting theirs.
On 8/24/2021 at 3:19 PM, Chris A said:I don't believe that your fundamental viewpoints on this subject (if there are, in fact, any) come very close to those that I've developed over time measuring, experimenting with, and developing MEHs. I thought it easier to state this instead of trying to field your "questions that aren't questions".
I hope you can realize how condescending that sounds..."(if there are, in fact, any)"....
You're correct about my "questions that aren't questions".
I didn't ask any questions in this thread, other than what you mean by zeroth order? (which I'd still like to understand)
I didn't disagree or present any alternatives to your first order tuning methods.
I presented observations of measurements that don't reconcile with timings you discussed, and my viewpoint (theory) as to why TOF's measure as they do.
On 8/24/2021 at 3:19 PM, Chris A said:Thanks for being civil in your answers, but I don't believe that we're going to see eye-to-eye on this subject, i.e, MEH design and measurements, and I believe that you'll end up not thanking me if I start to enumerate the "why"--like you have beforehand on this forum, so I'll simply agree to disagree here.
Thank you for appreciating my civility, and for when you provide civility in return.
I need to ask : Why is it important we see to eye-to-eye ?
I don't care if you advocate a particular tuning strategy....why do you care if i advocate a particular different strategy? What is there to agree or disagree about? Both strategies work just fine.
(If i ever hear proof why my strategy doesn't work, i'll happily and very thankfully accept the instruction, knowing better audible results are forthcoming !! )
On 8/24/2021 at 3:19 PM, Chris A said:I recommend that you continue your efforts at diyAudio for those that have been convinced of your efforts there. I don't share your views on MEHs, and in fact have totally different viewpoints for your apparent issues that seem to drive your designs. I think it's best if we leave it at that.
Thanks again.
Chris
I'm sorry it appears you don't like it when folks do not adopt your viewpoints, approaches, and methods.
I don't know if i'll keep posting here or not, as you seem to be disinviting, directing me back to diyAudio.
Seems like a pity to me...I know you have a lot to offer, that there is a lot i could learn from you.
And believe it or not, i have a lot to offer you too.
Anyway, later.. for now.
Sincerely, Mark
Peace...
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47 minutes ago, Chris A said:
It's 90 degrees of net phase lag on the lower frequency drivers that's induced for every order of the paired (high pass and low pass) crossover filters. If you remove the crossover filters and measure again, the lower frequency drivers will lead the higher frequency drivers in an MEH. You can see it in the spectrogram, step, group delay and phase plots for the loudspeaker.
No, i don't think that's it. Phase lag induced for every order of the crossover filters applies to IIR crossovers. I'm using complementary linear phase xovers which have the same phase lag (none), no matter what order used.
47 minutes ago, Chris A said:I think you're having trouble using these extremely high order filters having sharp cutoffs in MEHs, that absolutely don't need those kind of filters--because the reason for using them is simply not there as it is in multiple-aperture driver/horn loudspeakers.
I'm not sure what you're doing using "time of flight", but I don't listen to my MEHs (or any loudspeaker for that matter) using single ways, turning off the other drivers.
Chris
Spectrograms, step, group delay are all just derivatives of mag and phase response....they all key off the same TOF measurements that place the woofer further away that the CD. (REW: TOF = Delay relative to loopback)
26 minutes ago, Chris A said: (This seems to have been edited away or something...???) My reply got hung up Saving...so this is a put back together attempt)
I think you're having trouble using these extremely high order filters having sharp cutoffs in MEHs, that absolutely don't need those kind of filters--because the reason for using them is simply not there as it is in multiple-aperture driver/horn loudspeakers.
The multiple aperture MEH's help mitigate the need for steeper xovers for sure. But MEH's can improve further still with their use. I have tuned them both ways and the steep acoustic rolloff of a large woofer crossing to a CD has been easier to get right using steep, for example.
I use outdoor polars to determine whether to use steep or shallow. Working on such a project right now...
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13 minutes ago, Chris A said:
In the case of the multiple entry horn (MEH), the physical alignment of the drivers has the lower frequency drivers mounted in front (closer to the listener's ear) than the higher frequency drivers:
So, if a zeroth order crossover filter is used, it will allow the lower frequency drivers to lead the higher frequency drivers by 1/4 wavelength at crossover (90 degrees). While this is acceptable for an MEH time alignment, it will actually be a little closer if a first order filter is used to delay the lower frequency drivers by 90 degrees, thereby achieving time alignment without having to "cut and try" the channel delay corrections.
So bottom line: You actually need something in the lower frequency driver channel(s) to delay them in an MEH, albeit very slightly. Just using a first order set of crossover filters (low pass on the lower frequency drivers and high pass on the higher frequency driver[s]) is actually desirable. I've used just straight delay on the lower frequency driver of the K-402-MEH (equivalent to 90 degrees at the crossover frequency) and simple first order filters, and could see no difference in the output of the MEH.
JMTC.
Chris
Hi Chris, i should first ask what do you mean by a "zeroth order crossover" ? Not familiar with that idea...
Here's my two cents...
So far, on every MEH i've built (maybe a dozen counting fully developed prototypes), the woofers have always measured further from the mic, than from the CD. This has occurred despite the fact the woofers are closer to the mic than the CD,....... as in the drawing you posted.
And it occurs whether taking time-of-flight measurements with no filters in place, or with linear-phase low passes of any order (which don't have any group delay).
I've come to the conclusion, perhaps wrongly, that the natural acoustic low pass of the woofer is the source of the delay. Well that, and also i believe that our FFT measurement programs use the Hilbert-Transform to make TOF determinations under the assumption minimum phase represents correct timing.
At any rate, i know all my builds have had to add delay to the CD.
For the first time recently, i tried adding small 4" mids between the Cd and woofer (even though not needed).
The mids also need delay relative to the woofer.
With the woofer at 0.00ms, mids delay is 0.32ms, and CD is 0.83ms & 0.90ms (hf & vhf dcx464 coax sections)
You can see from the spectro, those delays put timing right on target. (other than i got a small bobble at 250Hz, xover between woofer and mid)
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Thx, I saw the AES3 cards and was hoping you were using it for your digital signal processing.
But like you say, anything can be done with the Cores, and maybe you had something else going on with it...
I'm such a huge Q-SYS fanboy too...
Here's the schematic I'm currently listening to. It's for a 4-way MEH on top a dual 18" sub.
500i fed via Q-Lan from a 110f. 16384 taps on all FIR filters! (just to keep from messing with different delays Lol) Now i just need the super amps you guys are talking about 😁
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Langston, may i ask a quick question unrelated to thread topic.........what does the Q-SYS Core do?
Thx, Mark
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20 hours ago, Edgar said:
OK, if demonstrably audible pre-ring is not important, then what is important? There is still plenty of debate over whether phase shift or group delay are sufficiently audible to be problems, yet you are going to great lengths to eliminate them. How do you choose which audible effect to worry about, and which to ignore?
Hi Edgar, in my previous reply, all I did was push against pre-ringing being significant.
And your question did ask what is important (to me)..... and as you said, i do go to great lengths to eliminate phase shift/group delay...
So to try to explain further...
Logically and intuitively, flat mag and phase make sense to me for reproduction.
I think everybody is on board with flat mag (subject to house curves and preferences of course). But we all know how much phase audibility is debated.
Personally, i want to do what i think and intuit, is technically right, whether audible or not. Seems like a can't loose, and will probably win, as well as learn something strategy.
And so far, the flat mag and phase effort, along with extending it to smooth polars, has been paying big audio dividends.
Subjectively I could wax on alot...but i don't really see the point...suffice it to say , tonality, clarity, transients, low end dynamics are making me a very happy camper.
Of course, i can't really separate the processing gains from the speakers' acoustic design gains, but since almost all the considerably varied speaker designs i've attempted have shown similar sonic gains, i have to credit much of the sonics to a common processing technique.
Anyway, if i may go back to the cost of potential pre-ringing vs the gains linear phase might bring...
here's an example of the tradeoff (as i see it), that illustrates the order of magnitudes difference between the tradeoffs plus and minus.
This is a measured not simulated, linear-phase LR 96 dB/oct electrical xover, where both the low-pass and high-pass sides have been summed together.
For me, there isn't much pre-ring, due to complementary cancellation.
What small pre-ringing there is, seems like a tiny price to pay for the outstanding Step Response.
As i'm sure you well know, only first order can achieve a step response like that.
And this is 16th order !! Gives so many degrees of freedom combining drivers acoustically.
I feel like a kid in a DSP candy store...who walks out into a rocking live concert Lol
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19 hours ago, Chris A said:
The example digital audio files (the sound of pre-ringing) need to be lossless files, not AAC, in order to hear what the guy giving the example heard. After he created the example files, he uploaded them to YouTube, which converts the files from lossless to AAC (lossy). That's what I was commenting on--nothing more.
Chris
Gotcha. Thanks 🙂
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1 hour ago, Edgar said:
OK, if demonstrably audible pre-ring is not important, then what is important? There is still plenty of debate over whether phase shift or group delay are sufficiently audible to be problems, yet you are going to great lengths to eliminate them. How do you choose which audible effect to worry about, and which to ignore?
What demonstrably audible pre-ring are you referring to?
I hope not that FabFilter youtube....that thing's ridiculous imo.
I mean, what is the high Q filter in it, min-phase? or lin-phase?
And what is the point of it even being in place?
If it's min-phase it should be fixed with min-phase....applying lin phase to it is a joke.
If it's lin-phase, it's even more of a joke...
Pure contrived marketing imo......
How do I choose which audible effect to worry about, and which to ignore ?....
Good question...🙂
I guess it's as simple as the audible effects i can hear in my real world builds, and my real world testing.
No hear, no worry.
Can't really say i feel i need to read about, what to worry about...
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16 minutes ago, Edgar said:
In a world where people swap wires and roll tubes, searching for that last 0.001% of audio bliss, pre-ring is a big deal. It's certainly a much bigger deal than the sonic difference between crossover capacitors.
Yeppers, the 0.01% hit that pre-ringing might maybe bring, is indeed 10x larger than the 0.001% hit to audio bliss that audiofools typically worry about, huh? ......haha
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The sound degradation in the FabFilter youtube is very easily heard, so i'm not sure what you mean by saying it needs to be lossless to be heard.
I think the worst MP3 would make such a contrived comparison obvious.
That's my point really, it's contrived crappola for marketing purposes....
It's not representative of any real world pre-ringing issues, imho.
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Yes, I've seen and heard many of those studio type presentations regarding pre-ringing.
They all have a simple common trait, imo..... they all use a very high Q electrical filter supposedly linearized (the idea of which is nonsense) ..... carefully constructed to prove their point.
Fabfilter is one of the worst....pure marketing on a fish hook. Lol
Bogus amogus big time! 🙂 imho.
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5 hours ago, Supersteff said:
I don´t know how pre-ringing sounds, but i guess its like muddy waters or something. So the lover the "hold-to-target"-requirement, the "clearer" the sound (waters)? Maybe this is obvious, I just try to picture it for my self!?
Hi again Steffen,
i think you've asked the real question....and i kinda ducked it my previous post.
"What does pre-ringing sound like?"
I truly wish somebody would nail it down ! 😉
The only time i've ever thought i heard pre-ringing, was when i was doing some super long FIR files (64k taps) using j-river convolution, with the same long FIR file size on every output channel. Kinda sounded like a pre-echo maybe, when there was silence before transients. But that experimentation was so short lived, without any real confirmation, it's almost stupid of me to mention it.
The thing with FIR in my mind, is that it is a VERY powerful tool.....which requires VERY judicious use.
FIR questions me, "I don't give a shite what you want to fix. Do you really know what to fix and why?"
Pre-ringing only happens when we can't answer FIR's question, i think.....
Being a lover of "hold-to-target down to -30dB" may be as much BS as pre-ringing 😁
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1 hour ago, Supersteff said:
Hi Mark
That makes sense, and you seem to have succes with keeping pre-ringing-effects low/"inaudible" with
Great, glad that made sense.
QuoteWould that in a way imply, that your tuning technique kind of fixes/moves pre-ringing-effects to sub -30 dB, when the HP- and LP-slopes start to differ from the ideal curves?
I don't think differing from ideal curves matters too much to potential pre-ringing. As long as the difference is relatively minor, and not near the central xover frequency.
With complementary lin-phase xovers , if mag and phase look good thru xover region, pre-ringing is simply not worth thinking about.
-30dB is my standard because I've found it easy enough to do...probably overkill in terms of ring.
What I think matters more to potential pre-ringing, is how complementary is the acoustic coupling both on and off-axis.
The idea that complimentary linear-phase xovers cancel pre-ringing, where the driver on the low pass side negates the driver on the high pass side, depends on acoustic symmetry.
So good old geometric lobing rears it's head again.
Which again gets back to why I like steep xovers....as they minimize the region of potential lobing.
My take on pre-ringing is it's mostly the same type of audio overconcern/overthinking, as with so many other audio things buried deep under the noise floor. Don't mean to offend anyone, but i believe in ranking marginal impacts, maximizing SNR, which sometimes means living with a little more noise i can't hear to gain a whole lot more signal i can hear.
That said, i do think many of the experiments where folks have tried to linearize phase globally on top of existing passive or active designs, have had the capability of bringing pre-ringing up to the surface..
The success of a global phase correction overlay depends on how well the speaker had already achieved acoustically complementary xovers, and also level in-band mag and phase response for each of its drivers. And again, for both on and off-axis.
Otherwise the global correction will be valid only to the mic location.....and complementary ringing cancellation could be sucking wind everywhere else.
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20 hours ago, Supersteff said:
I have been wondering , how far out of band do you have to flatten? If we consider a two-way MEH with a crossover-frequency of say 500 Hz between woofer and CD (I can´t remember how low you cross in your synergy´s?), up to what frequency do you flatten the woofers? Also how far down for the CD? Is it one octave or is half an octave enough? I wonder about the 1/4 wave cancelation-notch, how do you get around that, or do you need much overlap for that reason? just curious.
Hi Steffen,
How far we want flatten to is set by the driver's acoustic target order we want to achieve, and how deep we want to match the target.
For example, say the acoustic target is LR 24 dB/oct. And say the goal is hold-to-target down to -30dB.
Here's a 500Hz LR4 lowpass. As you can see, flattening to -30dB would need to go out to about 1200Hz.
Here's the acoustic target of 500Hz with a much steeper LR96 dB/oct, and goal is the same hold-to-target to -30dB. Flattening to -30dB needs to reach only about 620Hz.
Ok...how far can we actually flatten. The raw measurements speak.
Here's the raw woofer/mid section of syn7, a pair of 10 inchers that reach up to the CD. (with round ports in horn centers)
I think you can see that flattening response up to about 700Hz is pretty easy.
That out-of-band minimum phase frequency flattening to 700Hz, also takes care of the section's out-of-band rising phase (to 700Hz)
The in-band minimum phase frequency flattening (which almost everyone does), takes care of in-band phase flattening.
So when a steep linear phase xover is applied on top of the all the minimum phase flattening, the result is flat phase across the driver's bandwidth down to -30dB.
When the same flattening process is applied to the CD's low end, and uses a complementary linear phase high pass (500Hz LR 96 dB/oct), the result is very flat frequency and phase response across thru xover region (and across the board).
Hopefully, our woofer/mid ports are located where the 1/4 wave notch frequency is well above the -30dB down frequency, which makes the notch mostly immaterial after the steep lowpass.
When I try a shallower acoustic target for the woofer/mids like the LR4 first pictured above, it ends up taking alot of parametric work,along with shelving filters to flatten out to -30dB (or about 1.2kHz). I always end up saying screw this, this is nuts......cause steep works 😁
I've used this technique on everything from MEH's, to co-axials, to PA traps, to line arrays, and all of them with subs (crossed to with same technique).
Iow, anywhere there's a xover ..Lol
Always a cost in audio though, huh ......Price is FIR latency, and need/expense/complexity of multi-channel processing / amplification.
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50 minutes ago, Edgar said:
If you are unable to specify the filter coefficients in your processors' user interface, then you are lost.
If you mean being unable to specify bi-quads and such for IIR work , then I'll prefer to stay lost. 😂
If you mean unable to specify FIR coefficients for a FIR processor, i of course obviously agree.
50 minutes ago, Edgar said:Other than that, what matters is floating-point vs. fixed-point (not many fixed-point processors around any more), and processing power.
I take 24-32 bit floating point, 48-96kHz, with sufficient processing, as a given.
So for me, I/O channel counts, I/O input and output types and their type connectors, are the first screening factors.
Then comes whether I want FIR, and how much.
Then comes the breadth and depth and counts of IIR filter types, and orders.
Precision of delays, gains, limiters if needed...etc. etc
Iow, What can it implement quickly and precisely.
Ease of use is an ever present factor.... whether setup/controlling from software, or from the unit's front panel.
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1 hour ago, Edgar said:
There's a lot of hand-waving in your description, @gnarly, but there doesn't have to be. There are various ways to determine approximate transfer functions of drivers from mag/phase measurements. Once you have their transfer functions, you can mathematically change them to whatever you want (within reason) by means of inverse filtering (for minimum-phase systems), pole-zero cancellation (e.g., Linkwitz Transform), and other means. No need to use Kentucky Windage to try to get the transfer function "just right" -- the math will do it for you.
Sure, there is definitely alot of hand waving. !!! Here's some more !!! 😄
For me, the fact that the transform can be done mathematically in a number of different ways is almost entirely immaterial to choosing a processor,
and figuring out exactly how I'm going to use that processor to implement (what the math says can be done).
The math alone isn't going to do any processing is it ?
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On 6/21/2021 at 5:18 PM, DirtyErnie said:
Can anyone please point out a DSP that runs at more than 48KHZ?
Ideal is 24 bit, 96KHz or greater.
Thanks!
Hey, here's one that might meet your requirements....i forgot about it in my zeal for Open Architecture stuff.
Linea-Research ASC-48. (Or also available rebadged as a Danley SC-48) https://linea-research.co.uk/asc48/
4 in-8 out. analog/AES3. 96kHz. High spec / super clean device. (I have the Danley version.)
One of its really cool features is what it calls LIR xovers, which are essentialy 24dB/oct linear-phase Linkwitz-Riley's.
What makes them cool, is that the LIR xover frequencies can be dialed in on the fly, just like any usual IIR xover.....
AND the processor automatically adjusts internal delays to keep phase alignments spot on
I haven't seen that anywhere else....
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On 6/26/2021 at 11:20 AM, gnarly said:
I also don't feel comfortable linearizing a driver's phase from its natural frequency response rolloff.
There, I like to flatten the driver's out of band response with minimum phase EQs, hopefully to the point that the added electrical xover order becomes the driver's measured acoustic order.
That's a big part of the reason I like steep linear-phase xovers....it makes the out-of-band flattening much easier (and protects against lower-end over excursion from the min-phase flattening).
Hi all, apologies for quoting my own post, but i wanted to build on it and try to get some feedback.
I want to compare the technique of out-of-band fattening combined with complementary 'named crossovers', to building 'un-named' xovers using combinations of parametric EQ's, shelving EQs, perhaps along with high or low pass filters, etc.
Because i believe the two techniques give identical "total filter" results, for a given acoustic order/slope target.
By "total filter" results, I mean a transfer function of the whole electrical filter package of each approach.
The out-of-band flattening when combined with a named xover, becomes the actual xover...and as as such, the combination formed also can't be named.
That combo should look just like the "un-named filter approach", again assuming they both produce the same desired acoustic target.
Imo, the really cool things about this equivalency are:
that it lets us use the named xovers in our processors,
and i find it is much easier to EQ response up to a horizontal flat line, than it is to try to nudge one curve on top another (which is what we have to do when using combinations of parametric EQ's, shelving EQs, etc.) The vertical distances between curves can be quite deceiving. But flat against flat is easy to judge.
Anyway, this technique works so well for me, i keep trying to share it.... (credit for it goes to POS of rePhase)
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34 minutes ago, Edgar said:
My big concern about linear phase is pre-ring. An unavoidable consequence of linear-phase filtering is that the time-domain pre-ring and post-ring are mirror-images of each other. So whatever ringing occurs after the main pulse, also occurs before the main pulse. It may be arguable as to whether this is audible at high frequencies, but at low frequencies it is likely to be, due to the long durations involved.
Yep, been my concern too.
I've come to believe complementary linear-phase xovers cancel each sides preringing, and are completely safe to use.
So even for the low frequencies, like for a sub-to-main xover at 100Hz, i feel comfortable with a linear-phase xover.
Where i don't have comfort with linearizing-phase, are on the spectrum ends, 20Hz and 20KHz. Because there is no complementary offset.
So I've been using IIR for sub high-pass and 20kHz+ low-pass (which for me has helped clean up any tiny digital impulse oscillations)
I also don't feel comfortable linearizing a driver's phase from its natural frequency response rolloff.
There, I like to flatten the driver's out of band response with minimum phase EQs, hopefully to the point that the added electrical xover order becomes the driver's measured acoustic order.
That's a big part of the reason I like steep linear-phase xovers....it makes the out-of-band flattening much easier (and protects against lower-end over excursion from the min-phase flattening).
Steep also has helped me minimize the width of lobing region, making it easier to measure smoother polars thru xover region.
And i think that thereby also minimizes the freq range where complementary linear-phase xovers might not fully cancel pre-ringing acoustically.
Any comments / concerns welcomed....🙂
34 minutes ago, Edgar said:I have been looking for a high definition video delay, but have not found anything that costs less than about a year's salary. 1920x1080x3byte video needs ~6MB per frame, and each frame is ~17 milliseconds at 60 Hz. 4K 60Hz video quadruples that storage requirement.
Gotcha.
Like said, I'm clueless about video.
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Ok, very cool....maybe this is the right thread to continue on with phase some.
I very much share your views and subjective comments about phase audibility. It's a very narrow tightrope to tune to, but when right, for me it's like washing a windshield (to use a visual analogy).
Phase audibility is what has driven me to find the best processor i can, to be able to both implement linear-phase and then easily A-B it against traditional IIR.
Q-sys was the first affordable platform (used) i found that allows instant A-B, FIR vs IIR.
In case folks are interested in the Q-sys Cores' FIR capability, I've found a Core110f can run 8 channels of 4k taps per channel (@48kHz), on stereo 4-ways.
Different tap allocations per band are of course possible, but I haven't been able to put more than 6k on the sub FIRs. Due to that, i recently acquired an older Core500i, that allows 16k taps per channel on at least 16 channels. 16k taps is the current Q-sys maximum, raised from 8k a year of so ago...
Of course like you say, such latencies are a no-go for home theater without video delays.
Do such long video delays exist....? I have no clue as I'm audio only, (and use computer logitech speakers for video. lol)
Anyway, I feel I've had super success with phase linearization, both audibly and with measurements.
I've used it on a number of different type 4-way builds, always on a driver by driver tuning approach, where each driver gets it's own FIR file and amplification channel.
Here's a spectrograph example from a flat-phase MEH build, that uses a CD straight to 10"s. (I used the spectrograph settings you showed in the linked phase flattening thread.)
It's steeply high passed at 100Hz.
Here is a dual 18"s push-push slot loaded sub's transfer function, that I'm using with the MEH.
It was taken outdoors at 4m. (I don't bother with indoor sub measurements.)
Green is raw, blue processed.
It was with 6k taps. Looking forward to dragging it back onto the driveway and retuning it with Core500i.
I think the processed phase trace is already pretty good, but 16k taps will raise frequency resolution to a little better than 6Hz (from the 6k's resolution of 16Hz), to maybe gain some bass definition....who knows 🙂
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Awesome help to folks, like you've been doing.
I wish i wasn't such a slow writer and could try to help more too....
(i ducked every writing class possible in undergrad and grad,.....to my current chagrin ...)
Anyway.... back to to audio.....
I can not stop with A-B comparisons, on what i think you call phase growth. They keep redefining what I think i know/hear.
Are you simply talking phase lag, and re-coining it into 'phase growth' , or are there subtleties I'm missing with your term phase growth?
I keep playing with phase rather intensely.
I'll post some of my typical measurement results, and the tuning techniques that have given best results, on the thread you linked...for critique.
Seems like they would fit better there.
I'll try to stay away from subjective results.....cause we all know well they work 😅
B&C ME464 horn, opinions?
in Technical/Restorations
Posted
Hi uams,
The two horn loaded subs i have are DIY Labhorns (the Tom Danley design on Prosoundweb), and JTR Orbitshifters.
Both are FLH, with about a 9-10ft horn path.
The Orbitshifters use a single long stroke 18" firing into the throat chamber and will definitely walk around when cranked up enough.
The Labhorns use dual opposed 12"s firing into the throat chamber and do not vibrate much, if any, no matter how loud.
I hope that's the principle you meant.....?
The combo definitely improves with active tuning. You've probably seen B&C's recommended filters for use with their passive xover in the 215-DCX suggested build.
When i say "traditional PA", i mean in comparison to high quality boxes like Meyer UPA-1P's and such (which i own and compare to). It's a good sound, just lacking a little definition and openness vs what i've been hearing with the Synergy builds.
I've been working on Synergy iterations for 2+ years, slowly finding improvements that continue to distance their clarity from anything else I've heard. If it weren't for some builds going backwards, i might think it's just a case of continued confirmation bias Lol.
I have every intention of returning to the B&C combo, and giving it the time it deserves.
Maybe even trying the 215 design. I'd like to compare it to the DIY PM90/60's that i think are outstanding high-output boxes..
But happily right now, i've hit on a Synergy build i like so much, I'm in the process if building a couple more of it, for a LCR setup.