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In Reply to: RE: Analysis Audio versus Magnepan posted by Norman M on March 14, 2015 at 11:11:32
Hello Norman,
The Tympani iva has three separate panels. One for the mid planar and ribbon tweeter, one for upper bass, and one for lower bass. The mg20.1 uses one larger panel and places the lower and upper bass and mid planers in the same panel along with the ribbon tweeter. According to magnepan the basic planar panels are improved, the structural support of the panels is improved, the crossover is improved, and the ribbon voiced to solve brightness issues. At the same time the bass panels have less area and the panels are not as free from the enclosure as in the tympani iva. Although I agree the mg20.1 has improved the basic sound the mg20.1 lost two characteristics of the tympani iva, soundstage and palpability (can't explain this but the tympani iva has a certain you are there feeling). Thinking that this could be the 3 panel config and crossovers of the tympani iva I wanted to recreate the configuration using the improved mid/highs of the mg20.1. I basically am trying to recreate the tympani configuration with the current improved mid panels and ribbon tweeters.
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I think the basic difference between the 20.1 and Tympani IVa performance is in the midrange and tweeter - where spatial performance is created - the 3 panels can be oriented to have the lower mids covered by the mid and bass and the upper mids/low treble produced by mids and tweeter to be time aligned at a variety of angles. The MG20.1 can only be time aligned at one toe in angle fairly close to face forwards, which may not be a favorable one for the backwave reflections providing good ambiance. The time alignment has an impact on the size of the soundstage and on image precision.
The TIVa midrange compresses at lower volumes than does the midrange of the 20.1 so provides a better dynamic performance. Also useful is the extension of the bottom end cuttoff of the midrange in the 20.1 to an octave lower.
I use a line of Neo8 drivers to replace the mids of my TIV which allows the mids to operate down to a bit over 200hz and up into the top octave. The Neo8 is far more dynamic and precise than the TIVa and 20.1 midrange.. In one successful arrangement the XO is a symmetrical 1st order and the panels are time aligned. XO is at 275 and 6 khz and the imaging is the best I have ever heard on any system. A somewhat less successful arrangement in this regard is the Limage setup for Tympani with great bass but difficult time alignment issues which I solved by having the bass rolled off at LR4 and kept the mid HP filter appx at 100hz (while output does not come into play till an octave higher up) and dropped the LP filter entirely. Then I XO the tweeter as a supertweeter at appx 11Khz (I tried both 1st order which is better for clarity and a low q 5th order which is better for imaging).
The dispersion of the TIVa bass panels in the upper bass in multiple directions is a contributor to the sense of envelopment.
Oh, "The Impossible Dream".
I tried doing exactly what you have described. Only substituting a 3.6 which I do have, didn't ever see my way clear of justifying the cost of a 20.X, (neither would wifey) plus all four Tympani IV-A Bass panels, which I also do have. IIRC my Mrs. bought a fur coat, before the days of PETA!, so I could buy a pair of IV-As.) Although satisfying at first, I grew weary of the sound from all four IV-A Bass panels + my 3.6Rs over the long haul, so I'm back to the IV-As all by themselves. IME experience the IV-A 'loads' a room with sound in a way none other can, right out of the box using Magnepan's placement instructions, and no tweaks required. Believe it or not, you might find using a Carver C-9 Hologram Generator beneficial, often seen and inexpensive on fleaBay. It can be easily 'inserted' or by-passed in a system without wiring/rewiring, so nearly instantaneous comparisons are possible (depending upon how far away you are seated from the C-9, mine is less than an arm's length away. I've owned a C-9 from the time of its introduction and find it 'cool' for many recordings, and not beneficial for others. E.g., synthesizer type recordings, such as those of Wendy Carlos or Don Dorsey's Telarcs can be both mind and ear bending, appearing as if stretching sound out beyond the edges of my speakers but all across the width of the room, and just as Bob Carver claimed, it can seem to wrap sound around me.
Impossible Dream? Oh, you mean that we all have TWO of you to assist on this endeavor! LOL! It would be great to see you being able to ferry those panels around just to try new angles, Norman. (More for health reasons than just audio...it is great exercise...if the back can take it...mine would burst in pieces, I think : - )) .Somehow, however, I get the impression that you are happy enough as things are, and for good reasons.
Edits: 03/16/15
Hi Ben,
Thanks for your reply. In good health, and presently in aged poorer health, I've been ferrying those speakers around my listening room for 23 years (that anniversary came up this week). After listening at each and every new location/angle, etc., etc., I've returned to the present one so, I'm resigned to having them as they are.
Norman, I am guessing that if even you had such a hard time, perhaps the challenge is analogous to mating ordinary woofers to any planar...maybe even worse.
In recent weeks I finally embarked on adding cone woofers for use under 40Hz with my modded MMGs. Darn, the closer one gets to the goal, the harder it gets! However, I am applying some acoustic measuring and...there's light at the end of the tunnel (I am almost afraid to say).
To both you and the OP, I would not mind helping in implementing the software in a PC. The required calibrated microphone and associated items are so cheap these days that <$30 could get one in the game. With this, time alignment would be less cumbersome and the feedback on impulse, frequencies & distortion can help tremendously.
(LOL, I almost know what you are going to say. If I am right, there's a gentleman in PA that may throw some light on it.)
JBen,
Do you have a recommendation for a calibrated microphone? My system has been in boxes for a year and a half but will be resurrected in the new place soon. I'd like to take a look at using the mic with my DCX 2496, which has XLR connections.
Thanks,
Steve
Hi Steve! I have two Behringer ECM8000 and one Dayton EMM-6, all with their individual calibration files. However, one of the Behringer mics fell on a bare floor and became erratic...the Dayton has taken similar punishment and kept going.
These above are all XLR and require the phantom power supply. They can be bought for <$100 professionally calibrated. In fact, Cross-Spectrum supplies the Dayton EMM-6 with their own great "basic" custom calibration for about $75. OTOH, Dayton offers the same mic for about 50 bucks, with a quite decent calibration included. Dayton's offer is more than enough for most home audio measurements. Perhaps important for the DCX, if a calibration file can't be loaded, it is very likely that these mics are linear enough anyway.
Ironically, one can get a decent-enough calibrated microphone for <20 bucks. I have tried a couple of the Dayton iMM-6. One is now elsewhere, reading tests that we will show here soon. The other is at home, as backup if my main gear flames out suddenly. This model lacks the S/N ratio of the larger models but the ones that I tried were clean enough. These inexpensive little suckers are also very sensitive. Just this weekend one of them "heard" things that we missed while doing some sweeps with REW. The iMM-6 works on PCs, laptops, Android devices and iThings. There is some audio software that can use its calibration file. However, this mic is not XLR; the DCX 2496 may not be able to use one?
There is a Dayton USB mic, uMM-6. I have not tested it myself but it is getting good reviews also. (I hardly think that the DCX 2496 could use it but some folks may need one for other hardware.)
Let me know if you need more detail. Good luck!
I see that Cross-Spectrum has stopped working with the Behringer ECM8000 due to poor quality. The DCX 2496 has a feature called auto align that is supposed to adjust for phase cancellations, and yes, the mic has to be XLR and the DCX supplies phantom power. My new music room is on the small side for my MGIII's - 12'x 17' if I remember right - and some level of detection/correction could be beneficial. Along with some placement of book shelves to dampen wall refelctions, etc. Maybe even a Magnepan-approved fake ficus plant or two!
I'll look into the PC-based software and mics as well.
Thanks again for the good information!
Regards,
Steve
If you can go via PC, you would likely be better off than limiting to the DCX, like Davey suggests. You would get a wider range of options and better software choices. REW alone is a little powerhouse just by itself, and costs nothing. It helped me to tune my Maggies promptly, starting in 2008.
My room is 12x25x8h. Depending on the height, we probably will observe some similar room modes. Remind me later, when you can measure, and I'll see what may serve you. I do have notched (or truncated) corners behind the Maggies. They make a positive SQ contribution.
My room isn't quite rectangular as there is a bump out for a window behind the left speaker position. The ceiling is ~8'. What do you mean by notched corners? Having a shorter long wall than you I'll be more near field in my listening position, which I've had to do since I bought my Maggies ~30 years ago. I'll have to put a baffle behind the left speaker to mimic the wall behind the right speaker so the reflected sound delay is the same. There is a recessed window behind the left and wall behind the right. I suppose I could use the DCX to change the delay on the left channel, but the reflections off the glass and sheetrock could be different. I don't have much in the way of WAF issues this time around (in more ways than one...)as the room is my mine to use for music.
Regards,
Steve
Steve, I think I have an old diagram where I drew the corners. I'll post it when I get home this evening.
It does sound like we have a little challenge with that nook? in your room.
The DCX auto-alignment feature is fairly crude and targeted for the professional (longer delays) and not a typical domestic environment. Because of the small acoustic space the results it will give will not be consistent and accurate.
Much better to use one of the dedicated PC-based systems like REW or similar.
For low-frequency room correction it hardly matters worrying about a calibration file. The minor corrections contained within will be MUCH less than any peaks/valleys created from room modes or boundaries.
If you haven't yet purchased, I would suggest one of the newer USB-connected microphones. These bypass some of the pitfalls of microphone preamps and level settings.
http://www.minidsp.com/products/acoustic-measurement/umik-1
Cheers,
Dave.
Sounds like I should be looking at using PC-based applications to help tune the DCX settings. Of course, what it really comes down to is if I like it, it's good. However, it would be interesting to know, for my room and equipment, what speaker placement, room mods, and crossover settings combine to give me a pleasuable listening experience. It's conceivable I could come up with different XO settings for different volume levels. Maybe flatter at higher volumes and a boost here and there for lower levels. Now if I plug the source directly into the DCX, use the DCX volume control, and automatically adjust the settings based on the output level... Hopefully after we move and get settled in I can do some experimenting.
Regards,
Steve
Hi. J
You posted..."The required microphone and associated items are so cheap these days that <$30 could get you in the game". This is 100% true.
I also confirm the "ton" of help that you have provided this no-tech old man!!!
Thanks again J....old guy
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: 03/16/15
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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