All makes sense, thx.
And yep, I don't expect headphone results to necessarily translate to speakers. Headphones are just another tool for isolating variables IMO. FWIW, my audio goals center on "outdoor studio quality at concert levels". This may be why I'm more tuned to pursue transients, ie elimination of group delay, than most folks. I like sound that rips air in two
Biggest problem, I find that sound is so much clearer outdoors, it almost becomes the prime factor to acquiring audio excellence
But back to the headphones and continued A/B listening on the setup previously posted. As I listened to more and more tracks, particularly those without alot of major transients, I found I had a hard time hearing a difference between the summed linear-phase and summed LR. With enough kick drum etc, a difference was still there, but it took quite a bit of major transients. This was using sony 7506 phones.
So I switched to stax lamda pros. Now I'd say I hear a difference on almost every track. It's very subtle, there's a small reduction in sibilance with linear-phase, that even seems to apply down into the bass. It's kinda sounds like raspy edges in mid/highs, and rounder notes in bass, both turn thinner and sharper at the same time. Can't say I automatically like it lol
Is that correct? Can someone collaborate this?
I want to try the software if so...
What can correct low-freq, non-linear crossover phase, without introducing latency?
(apart from ab impossible to actually implement, all-pass network)
Minimum latency is specified at 75ms (dependent on operating mode and hardware). I've been looking for a zero phase rumble filter myself, all physically realizable solutions come with latency AFAIK.
I agree. I think many people unfortunately simply associate phase correction with tuning to a spot, instead of seeing it as a better way in speaker design.
The three dimensional problem is IMO very simply, the inability to co-locate all acoustic centers.
I don't think phase correction is about trying to overcome that.
I think it is about trying to restore the natural timing that occurs without having to sum bandpasses.
IOW, completely eliminating group delay.
My guess is group delay will need to be eliminated from mastering studios before we can reliably hear our attempts to take phase flat down low. I mean, the mastering was done to make the recording sound its best on a system with group delay, right? We probably need group delay in our speakers to match. Just like a recording mastered in a flat studio will probably sound best with flat playback...
My hope is when both recording and playback achieve both flat mag and phase, a new sense of realism from restoring natural timing is opened up. Who knows, seems logical at least (to me
My guess is group delay will need to be eliminated from mastering studios before we can reliably hear our attempts to take phase flat down low. I mean, the mastering was done to make the recording sound its best on a system with group delay, right? We probably need group delay in our speakers to match. Just like a recording mastered in a flat studio will probably sound best with flat playback...
My hope is when both recording and playback achieve both flat mag and phase, a new sense of realism from restoring natural timing is opened up. Who knows, seems logical at least (to me
My goodness, you don't want too much, do you Mark?
Dave.
Maybe like a Fender guitar with switching that allows the pickups to be connected in-phase or out-of-phase?
(It's easy to see where you're going with your question.)Audiophile idealism surrounds this topic, but as a practical matter there are numerous trade-offs involved and it's just not possible to check all of the boxes that audiophiles would like to. And, of course, we only have access to the playback portion of this issue.
The tools are there to experiment with though, so it's good fun to challenge a listeners hearing acuity with some practical testing. IF it's done correctly.
Dave.
To me, this Meyer speaker mostly seems to be an attempt to broaden their market into home HiFi. The woofer and waveguide are used in a speaker that's ubiquitous in mastering studios.
The Acheron is 75% cheaper.
What is the ideal directivity pattern for stereo speakers?
"Now days, there are many ways to measure which superseded the TEF machine in most ways. Programs like ARTA are cheap and powerful and allow you to look at the data in many useful ways. The TDS system on the other hand suffered a little first when they went from the hardware implementation to software. The TEF 10, 12 and 12+ all had quadrature phase detectors (something like a Costas loop) which cannot be “fooled” as there are always two phase measurement with a 90 degree difference between the reference.
The TEF20 and 25 are one sided detectors and can occasionally be fooled traversing 0/360 but that is offset by being so much faster and nicer to use (I use a 20 still).
The first problem TDS faced (as I remember) was when a very outspoken originator of a “smart” system took the TEF on as it’s main competition. That FFT based system at the time was fast, had cool color displays and a phase display that looked real but was only the Hilbert view of the amplitude and NOT actual acoustic phase.
I remember arguing at PSW that technically speaking, knowing the actual acoustic phase could be more useful than what it showed. Anyway, that company was also gobbled up by the Harmon and eventually bought out Sam and became very successful and evolved some very useful software in it’s own right.
The TDS process on the other hand was patented and narrowly licensed to Crown, Brujel and Kajer and some other technical users (maybe like Beta tape). While Crown was a top notch company, they too were eventually “Harmonized” and with what the TEF did being a narrow use too, it was sold to Goldline who I have had little contact with.
What I can say is that side by side with ARTA or the MLS part of the TEF software that it is does what it does differently. The biggest difference is dealing with noise, especially at low frequencies, the TDS is better.
How this shows up most obviously is if you examine the lf slope on a speaker. There is a point say 20, 30, 40DB down from the “flat” level where the curve does not continue to roll off.
Instead there is a notch and then a bunch of “free low end” as you continue down.
That is all noise leaking in and to get real data -40dB and more down, takes more time and averages. A major criticism of the TEF by the impulse based systems was that it required a time measurement first and then the response measurement and so “took longer” which if you needed real data doesn’t appear to be the case at all.
It is no problem with the TEF to start with a measurement noise floor 40 or 50 dB down or more if you take the time to adjust the mic gains. For me, it’s strength is that for crossover use, it delivers magnitude and acoustic phase unambiguously in a multi-way system.
I can’t make all the drivers add into one in the computer or the real thing if I don’t have the real data referenced to the tweeter.
I am not up on the technique enough to “know” but I would wonder that since the systems that compute the mag and phase from the impulse response are the ones that tend to have this noise / dynamic limitation in the data, how does that effect using that same impulse for convolution etc?
1Audiohack brings up another of the weird things that the TEF can do, a polar ETC. This is a way to locate where reflections are coming from and so locates where absorption is best placed etc. This was also something Dick imagined and implemented more than 25 years ago.
As you apparently know, one of Doug’s things is stereo imaging and cool acoustics, it has been a lot of fun having him listen to the speakers and recordings from the mic project. This was a imaging test recording he made a zillion years ago;
Online LEDR - Listening Environment Diagnostic Recording Sound Test
Here is one direction “identifying the source” has gone too, this is an amazing piece of hardware and a lot of fun to play with.
Columbia College Chicago - AA+A Acoustics program featured in WGN Channel 9 News
I wish I could find the video from this;
DDT presentation shot - ProAudioSpace
What isn’t widely known yet (but i think i can say it, too late if i can't) is that as of this Thursday, Doug has retired from teaching at Columbia and come to work for us full time. I remember him talking about the Hass kickers although to be honest my world was pretty much below 100Hz back then. What I can do is ask him if he has time to pop in and comment.
Have to run.
Best,
Tom Danley"
To me, this Meyer speaker mostly seems to be an attempt to broaden their market into home HiFi. The woofer and waveguide are used in a speaker that's ubiquitous in mastering studios.
The Acheron is 75% cheaper.
Close, but not the same waveguide (if you'll allow me to nitpick). Here's my post from earlier:
The Bluehorn top cabinet looks identical to that of the Acheron Designer, which has been in production for years and is an Acheron variant. Compare the port dimensions, horn flange, logo size and lack of air intake beneath the woofer. Another difference the trades are reporting is that they're using a new/different compression driver on the Bluehorn so who knows, they may have also manufactured a new horn for the existing cabinet (smaller diaphragm CD for a smaller throat diameter, trading power handling for HF pattern control?). That's just speculation though.
Never did figure out what contour the Meyer "Constant Q" horn was closest to. Always looked more hyperbolic or exponential to me.
Thanks Mark. I'm quite late to this thread but happy to answer any questions about FIR Designer and our other software.
Kind regards,
Michael
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