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In Reply to: RE: Analysis Audio versus Magnepan posted by JBen on March 16, 2015 at 17:02:34
Very similar to the path I have just embarked on, JBen. :-))After 20+ years of Maggie ownership, I will be adding a pair of 15", custom-made sealed subs to increase my enjoyment of Bach organ music ... and Yello! This won't be happening until November this year - when we move into our new house (the joinery in the listening room for all the equipment / LP storage has an 18" wide space at each front corner, for the subs to hide). So far all I have done is pay a deposit - and ask a lot of questions of the maker.
Answers to these Qs have resulted in a new paradigm for my active Maggies - moving from 3-way analogue active XO to 4-way digital active XO (the '4th way' controls the subs). The sub maker uses either a miniDSP unit or a Behringer 2496 to provide the sub LP / bass panel HP filters. These also provide delay for the bass panels ... to match up with the subs (who have an intrinsic delay resulting from the slope used and the plate amp driving the Dayton Ultimax drivers).
If you delay the woofers ... you have to delay the mids & ribbons by the same amount - so, unfortunately it's out with the analogue XO and in with a digital XO! :-(( For someone who is wedded to vinyl, this need serious consideration! ;-))
However, one thing that pushed me over the edge was that both the miniDSP unit and the DCX2496 will also provide room correction - which is always a good thing for <200Hz.
Regards,Andy
PS: I will be posting at the end of the year, when the subs are in and configured.
Edits: 03/16/15Follow Ups:
I'm not sure I understand, "If you delay the woofers ... you have to delay the mids & ribbons by the same amount".
DSP gets you both phase and impulse alignment typically by delaying the mains, never the subs. That's to offset the time lag from the low pass (subwoofer) filter. An 80Hz 2nd order crossover frequency has an inherent group delay of about 3.125ms equivalent to 3.5ft.
You could have a speaker setup where the subs are several feet closer to the listening position than the mains which would more than offset the LP delay, but that's a pretty unusual case.
I'm looking forward to your experiments!
When I said " delaying the woofers " - I meant delaying the Maggie bass panels, to match the delay inherent in the sub drivers.
And if the Maggie bass panels are delayed ... the mids & ribbons must follow.
I believe you when you say " An 80Hz, 2nd order crossover frequency has an inherent group delay of about 3.125ms equivalent to 3.5ft. ". I would ideally like a 24dB slope but:
* I know this produces more delay ... and the miniDSP unit I would like to use has a maximum delay of only 7ms - so a 24dB slope may not be feasible. :-((
* to make the situation worse, my subs will be 3-4' further away from my ears than the bass panels - so it would seem we are already up to ~6.25mS, using 12dB slopes @ 80Hz!
To keep to 24dB, the guy who builds the subs - who will be installing them and has the equipment to tailor and measure the DSP setup - will try increasing the XO frequency to, say, 120Hz (which I understand will not be a problem, in terms of the directionality of low bass, due to the symmetric sub placement and the fact that the subs are relatively close to the bass panels) ... but this may still not produce a low-enough delay ... so I may have to move to 12dB slopes.
Regards,
Andy
The link at the very bottom is helpful, as is the whole site.
The wall wort power supply of the DSpeaker Antimode 2 Dual Core for room correction that I'm playing with now is the not all that transparent. You'll encounter this with miniDSP.
If you've anticipated this, you may want to build or buy a better one, so here's a download article that compares more than a dozen commercially available & Diy PS designs. There are also charts of noise, PSRR, etc. in the Linear Audio site itself:
http://linearaudio.net/article-detail/2128
Oh, in case you were wondering; "that Barry" is not this Barry.
Have fun.
You don't need to be time aligned in these low frequencies, being in phase in the XO region is good enough. The cycles are long so bass transients are not reproduced by the bass drivers at all. Often you just need to delay or set further the bass so long as you don't high pass or do that at low order.
Some DSPs will have a phaseless XO option so only the DSP's latency and added delay are active, so they may have a lower minimal delay than an analog XO at low freq.
I'm still learning about this and was actually commenting more conceptually. I don't have a good sense of what all various DSPs actually do for bass management. I agree about the XO region needing to be in phase but it's actually easier than that.According to Toole who's saying the same thing that you are in different words, subwoofer frequencies (80 Hz or less), are minimum phase. In English, or your language of choice, that means, for the listener sweet spot, FREQUENCY RESPONSE IS ALL YOU CARE ABOUT about and you want to get it flat. Frequency response contains all the information about phase and transient response that matters at low frequencies. You can move mains and subs, the listening position, or use parametric equalization to accomplish that. If it's flat when you're finished, "mission accomplished" and it will sound great. That's helpful for anyone who doesn't understand all the technical stuff.
Edits: 03/18/15
My own experience is that phase and time alignment are useful for "bass imaging" which was not recognized as something that exists in Toole's day. It is significant in determining acoustic dimensions of the recording space, plays into the way we determine image size
You can have atrocious FR and still obtain amazing lateral imaging. You also can get good depth but somewhat incorrect layering. Event timing in the structure of a transient determines the bulk of imaging cues and is extremely sensitive to precise time response. The localization is entirely over with before pitch is determined by the ear. We are talking 3msec for transient localization (0.1 to 10 ms) then size is determined and "texture" only then is the tone determined (10 to 30 msec).
Prior to Toole and the frequency domain theory taking over as dogma in psycho-acoustics there was active research into transients, that was resumed more recently. I posted an article that consolidates some of the research into transients about a year ago. It is not a difficult read.
http://www.bodziosoftware.com.au/Attributes_Of_Linear_Phase_Loudspeakers.pdf
I do not want the frequency response flat at the listening position. A microphone is also not equal to our ears/hearing. A microphone can give you a hint of what is right/wrong but it cannot give you the whole truth. In the end it is matter of psycho-acoustics and a microphone cannot tell you much about that.
@ Roger G,
You are correct, the microphone will only say flat or not, nothing to do with taste and biases, some like it a bit fat in the Bass. some not...
Regards...
Can You explain more of what You mean by some like it more fat in the bass?
It's easy to interpret it like flat is the right sound.
The truth is rather not if You listen to a recording through flat measured frequency and compare it to real life.
Also, a flat frequency response does not take dynamic compression into account.
A frequency plot is only telling You about the frequency balance at a specific db continuous output.
Cheers!
The one who succeeded was the one who didn't know it was impossible.
A microphone is a very simple device compared to our hearing. An in-room measurement should never be flat at the listening position. A psycho-acoustic flat response is something different! That is what we want and it is not a straight line in a diagram. This could be a good start, http://i47.tinypic.com/rrt8yd.jpg , maybe a bit more in the low bass region. This is for pink noise measurement.
Depends, you want your FM curve anechoic or at the listening position, the microphone is just a tool, what you may or may not like has nothing to do with the FR being flat.I dont think -8 db at 12K would work for me at the listening position, then again it's pink noise from a stereo pr, so it's not the same and the droop at 10K is normal due to being off axis to tweeters and stereo phasing.
Then again you are using a simple microphone with pinknoise ... :)
Edits: 03/28/15 03/28/15
Thanks for the link Roger!
The one who succeeded was the one who didn't know it was impossible.
I will study them both.BTW, re. the wallwart for the DSpeaker Antimode 2 Dual Core ... I notice that its PS is a 12v DC unit - ie. a SMPSU.
I've done quite a few experiments recently with SMPSUs and have the following suggestions for you:
Experiment #1 - replacing the 24v DC SMPSU driving an A.N.T 'Kora' phono stage with a 24v SLA supply.
A mate who owns the 'Kora' was impressed by my own battery-powered 'Muse' phono stage. So I rigged up a couple of 12v SLAs in series for him to try with his 'Kora' - having:
* an on/off switch, and
* a 40uF film cap across the batteries (very important to have this!).This SLA supply made his 'Kora' sound much better ... so he ordered an SLA supply from me. This is actually not a cheap exercise! By the time you include batteries, a nice case, the cap, a charger, a DC plug for the output and a DC socket for the charger ... you're looking at USD300! But he's happy to pay this because it elevates his 'Kora' to another level.
Experiment #2 - putting an isolating transformer plus a Schurter hash-filter between an SMPSU and the wall socket.
The SMPSU in question was the 48v supply to my TT motor speed controller. Just having an isolating tranny between the MeanWell SMPSU and the wall made the TT sound better - but adding the hash-filter (oriented a particular way! o-) ) lifted the sound up another level! :-))
I suspect the hash filter wasn't doing anything to the SMPSU itself ... but what it was doing was preventing the crap which most SMPSUs generate in the mains from getting out to 'infect' other components.
Either approach will enable you to make your DSpeaker Antimode 2 sound better; which one you choose probably depends on how much current it draws and whether it still works when the battery has discharged down to, say, 11v.
Regards,Andy
Edits: 03/17/15
LOL! Yes, you and I are facing a few similar challenges. I actually cannot afford to go back to digital once the initial conversion to analog happens to the music. In fact, your Frankies would remain mightily charming even in the face of something like this. OTOH, my MMGs need every bit of help to keep their sweet charms.
Now, my bass goals are less "deep" for the time being. I live in an apartment. Still, the 10" woofers that I am adding can easily run amok if I am not careful. By going with L/R woofers, two stereo channels and shallow boxes, things are quickly looking up. Time-management success seems to be at hand (ok, more like at arms length : - ))
However, WAF could potentially do me in.
Let's check notes later (visitors on their way from airport).
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