The making of: The Two Towers (a 25 driver Full Range line array)

Canvases make aesthetically pleasing damping panels and your boss[ha ha ha] could pick her own pictures?

Already have one of those hanging, though I picked the picture....
doek.jpg


:eek:
 
If I remember correctly, Zaph showed that Vifa TC9 driver to be very flat at the high end, but the Vifa spec sheet showed it to have a bit of a peak in the FR at about 10kHZ. I have that same driver in my computer speakers that I recently built (bi-amp'd, 8 inch woofer X=500HZ), and I think I may be hearing what could be a peak at 10kHZ. The treble in my main system sounds a lot smoother and "lush" (Seas Millenium 1 inch domes). You might try putting a slight dip in the response at 10kHZ.

Another thing I might try is to electronically bandwidth limit the towers to 7kHZ, and use a single (per side, 2/3 the way up to the top) 1.5 inch Fountek Ribbon tweeter (Zaph says it's one of the best) to do the high end. Or possibly a Dayton (affordable) "Air-Motion" tweeter (which I think is di-pole). I think the major advantage of floor to ceiling line arrays may be in the frequencies below about 7kHZ, based on my understandings. I could be wrong, but it's something to try.

For room treatment, I've found that corners are the worse enemy. I nailed 2 inch diameter cotton rope (from a fabric store) into most 2 and 3 surface corners of my living room (not at the floor), and the clap hands test showed a major improvement in the decay response. Bad ringing was virtually eliminated. No large panels or bass traps of any kind. Much better WAF. I better add that cotton rope is very flammable (I didn't know), so soft non-flammable foam rubber would be a wiser choice.

One of the problems with building highly accurate speakers is that the shortcomings of the recording process can be exasperated (made worse or at least more noticeable). I have never regretted designing and building my 4 section Baxandall tone control circuit. Easy to use and makes bad recordings listenable. There are just too many reasons why flat is not likely optimal, and this varies from recording to recording. Especially when you're a musician who might be digging up very old recordings of dubious quality.

Again, congratulations for creating such a beautiful work of art.


I kinda agree with you that I should try something like that at some point. If I look at my waterfall plots I have a very clean mid range, but it is obvious there is still a bit of restlessness up top. I won't show them just yet, I must satisfy myself first :D.
I can EQ flat on the listening axis, but that doesn't mean I don't hear the break up that's thrown into the room off axis. Probably why I have a downwards slope. Remember, I don't get to hear it from one or two cones, there's 50 of them. Another solution is 50x Scan Speak 10F ;).

Most things sound very good, but I do notice once there is more complicated high frequency structures in the song, Especially in the phantom center things fall apart slightly. Not apparent on every song but enough to bug me.

My conjugation network seems to clean up some of that higher frequency and I haven't build my second one yet. So right now I have less balance up top left to right. I should wait till I have both inline to judge.

Yesterday evening I had someone visit me that works in an Audiophile store. He suggested I cut out the domes :eek:. He was impressed with what the arrays are able to do, but based on his experience he figured the domes cause more trouble than they solve up top.

I know member Koldby coated his drivers with Isopunkt. I tried to get it but it has vanished from the market.

Like I said, not a mayor distraction, but when looking for perfection it is noticeable. It took me a while to narrow it down to the most upper frequency range. The key to real greatness is there with these arrays.
 
I kinda agree with you that I should try something like that at some point. If I look at my waterfall plots I have a very clean mid range, but it is obvious there is still a bit of restlessness up top. I won't show them just yet, I must satisfy myself first :D.
I can EQ flat on the listening axis, but that doesn't mean I don't hear the break up that's thrown into the room off axis. Probably why I have a downwards slope. Remember, I don't get to hear it from one or two cones, there's 50 of them. Another solution is 50x Scan Speak 10F ;).

Most things sound very good, but I do notice once there is more complicated high frequency structures in the song, Especially in the phantom center things fall apart slightly. Not apparent on every song but enough to bug me.

My conjugation network seems to clean up some of that higher frequency and I haven't build my second one yet. So right now I have less balance up top left to right. I should wait till I have both inline to judge.

Yesterday evening I had someone visit me that works in an Audiophile store. He suggested I cut out the domes :eek:. He was impressed with what the arrays are able to do, but based on his experience he figured the domes cause more trouble than they solve up top.

I know member Koldby coated his drivers with Isopunkt. I tried to get it but it has vanished from the market.

Like I said, not a mayor distraction, but when looking for perfection it is noticeable. It took me a while to narrow it down to the most upper frequency range. The key to real greatness is there with these arrays.
There's a whole nuther thread about trying to improve the phantom center image with only two speakers. The concensus appears to be that it's not necessarily fixable. It's about the way the short wavelengths add at the listening position (which varies depending on where anyone sits), when coming from two physically displaced drivers. Perhaps a single tweeter up high in the middle(?).Maybe also a tweeter on each side that has just enough time delay to decorrelate, if that's possible.
 
I know bob, I'm heavily involved in that thread.... :)
But in afterthought I was a bit too hasty writing what I did. It is true I'm still looking for ways to improve the upper end of my arrays. But I gave it less credit than it deserved after analysing the track(s) I had troubles with.
On headphones playing back the same track I perceived the same mess in high frequency response. So it wasn't my speakers fault, but a recording flaw....
Probably a bit too much compression or limiters used in the production/mastering.
So my search will continue to see how far I can improve what I have. If I get the opportunity that will also involve experimenting with a tweeter up top to aid that top end.

So it wasn't a complicated high frequency structure but more of a congested part in a song that had me worried, pfew ;).
To check my findings I did send one of the songs I had trouble with to another member on here (maybe he will chime in). He had the same initial findings (on headphones) but will check on another output device to be sure.

A phantom center still is a difficult task to get right. But so far I'm still pleased with the mid/side shuffling I'm playing with. It works very well for me to get the same tonality in the center matching the sides. It worked well enough to relax the window size I use in DRC on top. Right now I use 9 cycles in bass, 4 cycles in mid and 9 cycles up top. Still experimenting with both minimum phase and linear phase settings (and combinations of both).

Meanwhile I will still hunt for better voicing etc. every thing I can think of to get better sound out of these (while obeying the set rules by my significant other). So far not all steps have helped or have been an improvement but slowly but surely I'm getting ahead. Both in measurements and listening tests.

I'm open for any suggestion, though I can't promise to test everything... somehow I don't see myself painting dots on 50 drivers as an experiment for instance ;). No offence meant though.
 
That would make for a fun test! Should work too if crossed high enough!

Right now I'm tied to only 2 channels due to the DAC I use. I do have an expansion card for my Sonar Essence though. So all I'd need is an extra amplifier. Strange, I know. I am not the typical hobbyist you find on here with a crazy supply of speakers, amps and other things lying around :). I do have some spare tweeters somewhere from my Car experiments... not quite as pleasant as the XT25 I replaced them with but they were way more expensive at some point in time. Herz ~1" dome tweeters aimed at the car audio market. Probably still have some pioneer tweeters as well.

First I'm going to finish the second conjugation network and both measure and listen to that extensively.
 
.....So it wasn't a complicated high frequency structure but more of a congested part in a song that had me worried, pfew ;).
To check my findings I did send one of the songs I had trouble with to another member on here (maybe he will chime in). He had the same initial findings (on headphones) but will check on another output device to be sure.....

Checked at other system setup and that recording seems serious flawed and falls apart some passages by a mistake either at mixing or mastering process, line arrays is totally innocent and probably very high end reproducers they just throw out the signal they feeded :D.
 
I think my best suggestion might be to put a single tweeter up high in the middle, and delay it's signal by maybe 10mS (so the cancellations it causes will be so close together that they won't be perceptible), for a de-correlation effect . I haven't tried this, but it seems to make sense in my head.

Do you suggest the upper frequencies of the full rangers be left alone or low passed at some point? Personally, I'd try first to leave the full rangers all alone. I was thinking about experimenting with a rear firing, up-firing ( reflection from the ceiling which would be pretty close from top of the arrays) & lastly a front super-tweeter as you have suggested with 24dB/ octave HP filter. I think 20ms delay would be optimal for front tweeters which, I believe is the threshold for the human brain not to mask it together with the original signal from the full rangers & cause any dips.
It would be a lot better if rear or up-firing super tweeter works here without requiring any delay settings, which is obviously extra work ( & headache) :spin:

Wesayso,
are you using/ trying conjugate network / impedance matching for the high frequencies only or for the low end ( at resonating frequency) as well? It would be awesome if it works!
I have noticed confused & constipated action in the high frequency region of complicated tracks(mostly hard rock playback) in my arrays. Though, I have lots of work to do before making a judgement.
Right now, I am just hugely overwhelmed by the dynamic range of the arrays... which is sometimes too much for some movies. With the loudness set for normal level playback of dialogues, sudden special effects makes me go :eek::eek::eek:. I have never experienced this with any other speakers.

How would you comment on speech ineligibility of your arrays?
 
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Wesayso,
are you using/ trying conjugate network / impedance matching for the high frequencies only or for the low end ( at resonating frequency) as well? It would be awesome if it works!
I have noticed confused & constipated action in the high frequency region of complicated tracks(mostly hard rock playback) in my arrays. Though, I have lots of work to do before making a judgement.
Right now, I am just hugely overwhelmed by the dynamic range of the arrays... which is sometimes too much for some movies. With the loudness set for normal level playback of dialogues, sudden special effects makes me go :eek::eek::eek:. I have never experienced this with any other speakers.

How would you comment on speech ineligibility of your arrays?

Hi rockk19,

Due to my recent encounter with the congested HF I'll be more reserved to lay the blame on the speakers. I guess these speakers are actually more than average in revealing the quality of the mix.

So check the recordings that give you the notion of congestion. It might just be (over)use of compressors and limiters that's bugging you.
The speakers themselves are very clear, anything in the mix that's not clear will stand out.

The speech intelligibility I rate very high. But that needs a bit of explaining. At first I used IIR EQ and was impressed with the dynamics and speech intelligibility but there were some draw backs as well. I noticed using high Q EQ in some places bothered me. Something was off. I noticed some strange effects in songs that shouldn't be there. When I backed off on high Q PEQ cuts/boost things got way better. But it was very difficult to get pleasing tonality dialled in. You get used to the sound after a while though.

After the first use of FIR filters the difference was huge. But I also lost some of that dynamic feel. Things calmed down and my center image was lacking (tonally and in intelligibility) compared to the sides. For a while I was compensating for that with longer correction windows up top and a little mid/side processing to get some body back. It seemed the FIR filters shrunk my Line Arrays down to very small speakers. Using the JRiver Surround Field effect (based on Blumlein shuffle, boosting the side signal in mid side processing) compensated for this shrinking effect.
Measurements with the effect on looked the same as with it off, except for a tiny increase in output. But at least nothing funny was going on so I used that for a while.
Recently I use my own mid/side processing as described here and that gives me back the clear vocals in the center, good tonality and intelligibility.

To get back to the IIR vs FIR, switching between my old settings and the FIR the last one (FIR) gives me way more accurate sounding results.
Though I will admit the IIR processing had me jumping from my seat more than once. It was more restless, busy if you will. Not an unpleasant effect though. Great for building up suspense in the right song.
I remember my heart beating double time just playing "He's a Pirate" from the "Pirates of the Caribbean" soundtrack. Only able to utter a "WOW!" afterwards. It still sounds good (tonally its better, more accurate sounding instruments) but that overpowering feel has diminished somewhat.
Never heard it the same again as in those first weeks with IIR. But which one is more true to the recording? :spin: ;)
 
I noticed I forgot to react on the conjugation network question from rockk19.
It's a complete correction of the impedance of the entire speaker.

Here's a plot of the resulting impedance and electrical phase:
impedance-cor.jpg


I could have done a little better (in hindsight) by changing the value of one resistor a little... but funds are slim so this will have to do.

The schematics:
impedance-nw.jpg


And in real life (not completely finished here):
conjugate-s.jpg

(there are actually a couple of extra resistors on here that I played with in series and parallel)
 
Sure you can ask. Basically because I've read a lot of good things about the effects of compensating networks to straighten the electrical phase. So I got (very) curious about it.
I've seen a lot of reports about the positive effects of these kind of networks, even when used in an all active setup.
I've also read just about as much reports saying it doesn't do anything (positive or negative) for an active setup. A lot of controversy as usual I would say.
But because of the nature of the line array (Acting like an unusual tall full range driver) I figured if there is a difference with or without, I should be able to spot something with these speakers.

I did find a difference in cables, a measurable difference. So why should this be any different? If the speakers play as loud as before after the conjugation network is connected, surely something is different. Is this audible? I don't know yet. The SPL doesn't seem to have changed by adding the network, that much I know. I haven't done enough back to back tests to compare measurements. And have only one conjugate network active right now.

A lot of things in audio gets conflicting views between the so called measurement camp and the subjectivists. I'm just thinking, what if... and try a lot of these theories and myths out for myself.
Like the timing issues, I believe in a time coherent system sounding more real than (the same) one that isn't time coherent. So I did my best to get a time coherent setup with the help of FIR processing. How many speakers are there with a STEP like this:
epstep.jpg


There is still a lot of abracadabra in the world of audio. And I'm not betting on one camp being right. So I do these kind of experiments, mostly to please my own curiosity.

I was hoping the change in electrical phase would also have a similar big change in acoustical phase. But so far it is only little things that I've seen being different. But I need more tests. Even little things can change a lot in sound.

Today I changed the toe-in of my speakers just slightly. World of difference in sound. Probably because of a lot of little things changing. But not always as easy to measure.