EV HP640 on Klipschorn, now setting up the MINIDSP.

[djk]

" it hasn't been stated in Wikipedia"

That's correct, it hasn't been stated, and it should be stated. B-Radial is a JBL trademark for their constant directivity horns.

http://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=1&cad=rja&ved=0CCwQFjAA&url=http%3A%2F%2Fwww.jblpro.com%2Fpub%2Fobsolete%2F23606566.pdf&ei=1SUwUtKzM8rmrAGnzIGgBQ&usg=AFQjCNGLVJYG_XX0FBWHdIloDBneG9XwLQ&sig2=2syoRSqIYLax6B_RmQz_1w&bvm=bv.51773540,d.aWM

Compare those with the EV HP640, the 640 numbers look better than either the 2360 or 2365, and sounds better due to the lack of the long tail and narrow slot. The 640 will also fit well on top of a Klipschorn bottom. The 640 is not perfect, but I like it.

[PrestonTom]

Re: Electrovoice HP640 horns.

I still have a pair of these in a box in my basement. If someone would like to borrow them for a few weeks and try them out (or buy them), please contact me. I am in Connecticut. You will need some 2 inch throat drivers (eg, JBL 2445/46 or EV DH1A etc)

[mark1101]

For the K402 guys...........I have a couple of pairs of them. One pair is on a 2-way setup with DBB bass cabinets, the other is on an MCM setup. I have used K69, TAD 4002, Faital Pro, and BMS 4592ND coaxial.

Nothing comes close to the clarity and sweetness TADs. No matter what I did with the other drivers or system equipment.

The BMS is next best IMHO and has a ton of HF energy. Actually, these are still pretty new to me and I'm still working with them. I like them a lot. They are not as clear sounding as the TADs though. The strength here is that they are coxials and in a 2-way system there is no way I would consider going 3-way. There are plenty of crispy highs. These are currently on my DBB system.

I liked the Faitals as well. These were very respectable and I was satisified with them on the DBB system. Very good highs and needed less EQ than the K-69. Not close to the clarity of a TAD but similar HF energy. Just a good sounding driver on a 402.

The K69 works pretty well if you use Roy D.'s settings. He has spent a lot of time optimizing this budget driver on the K402, and it DOES have a lot of specific EQ requirements. But as you can read here, every other driver I put on the K402 has sounded quite a bit better.

I took the Faitals off of the Dbb/402s and put them on a pair of 510s and run them above the 402/TADs on my MCM-4 grand (covering 6K-20K). Right now I'm frustrated with this setup. The system sounds much better as an MCM-3 with just TADs on the 402s covering 600-20K......and no 510s. I thought going to an MCM-4 would be great and better than an MCM-3, but I haven't quite figured out setting up the 510/Faitals just yet. Lots of "hash" up top.

I am of the belief the TADs are simply the magic of a K402 setup. If you can ever get a pair with the berylium drivers........Get 'em! You'll love 'em. They are worth every penny if you are serious enough. I may need another pair for my 510s.

[The Dude]

DJK thanks for clearing that up, I will say I like the EV over the P. audio so far,I think it has a very large sound compared to what I am used to. I have a guy locally who thinks he might have a pair of the JBL 2630s for the price and if I like the sound I may just stay with the EVs. I can pick up a pair locally from the same guy for about 200.00 for a pair. I don't know what he wants for the JBLs, but researching them the prices are alot higher. I may all so see if he will let me borrow them to try them out.

I just ordered my minidsp yesterday and it should be here by Monday, I am getting pre excited about finally owning a active processor of some sort I just get to thinking of all the stuff I can do with it. But will definitely need help setting it up, hopefully you guys are up for the task.

Tom you should have pm shortly

Mark thanks for the input. Drivers can get costly, I hope someday to have the extra cash to play around with different ones.

[The Dude]

Well I received my Minidsp, and umk 1 measuring mic today. But I have now clue what to do with it all. I figure I will need something that produces a frequency sweep, that will be produced by the speakers, then after I take a reading I will then need to make adjustments. I all so figure I would have to have a starting point with crossover points. I all so would like to take single driver readings. I am just curios what kind of spectrum this driver/horn covers on its own. I have read some people doing out door measurements, I would like to do that as well but for now lets just try and give it a go in the house until I learn how to play with everything. These are some new toys I have been anxious to get and play with just need some help now. But first things first I need to figure out how it all works and hooks up. Any suggestions would be greatly appreciated, I know the first thing is to read any owners manuals. I do have rew if that helps.

[cjgeraci]

It does sound like you are on the right track. ph4525s can sound good but yes you need to eq them. I run mine with a pair of radian 850s and an active crossover. I actually think that more of my good sound with the 4525s is probably due to the quality of the drivers instead of the horns. But you probably could not use 850s because you need to cross them over lower on the bottom end than I do (unless you used an extreme slope). I get away with crossing them over higher because my mains are A7s with k horn bass bins used as subs.

Have fun. Keep tweaking!!!

Carl.

[The Dude]

Ok this is my first time using REW and umk 1 microphone. Both speakers are Klipschorn bass bins, but left is the EV hp640 horn, and the right is the P audio ph 4525. Both stacks have the same tweeter, and mid driver crossover is Deans AA(I know its probably not ideal to some but seems to work). This first measurement is from my listening position. Now that I have the Mini dsp I can really dial in these horns. I am sure my room has to cause a lot of issues, from what I understand the less equing the better it will be. Crap almost forgot to add the chart. Ok how do I go about making it a jpeg or some way to post it here.

[The Dude]

Ok this might work, now keep in mind I don't know what I am doing, but here goes.

[The Dude]

Hey I did it, now I just have to figure out exactly what I need to do. If it helps I just placed the mic approx 3 feet from the speaker, and approx 4 feet from the ground. I played around with some of the settings to see how it works, you might be able to tell from the plot but it doesn't sound very good.

[The Dude]

I think I should try and read, Does anyone have any good reading for active crossovers. I know Rane has a bunch of stuff on the topics, I was thinking about reading this if I have time tonight.

http://www.rane.com/note160.html

[PrestonTom]

My suggestion would be to use Holm Impulse ( a freeware program over at DIYaudio). The bass bin can be measured with the mic very close to the mouth. You will need to understand the gating function, so go ahead and pour through the threads. Does your DSP allow a L&R filter? If so go ahead and use it. You will need a shelving filter for the horn (either 6 dB or 12 dB).

It is probably best to apply the shelving filter first so the horn is somewhat flat and then tackle the crossover. Also does your DSP allow time alignment?

[The Dude]

Tom I will check it out tonight, what do you mean by L&R filter, and shelving filter, I have Input gain, parametric input, crossover, parametric output, Delay, gain, and RMS. Both parametric's have high shelf and low shelf. Yes on the time alignment, I can go 0-7.5 ms, I thought someone said the bass bin is like 58" is this the correct way of thinking, so I would then delay one or the other approx 4.5' or 1.5 meters roughly. Any ways I have been playing around with things, is there anyway I can blow something, is there some sort of precautions I should do to keep from damaging stuff.

Thanks

[mboxler]

Here's a link that may help....

http://sound.westhost.com/articles/active-filters.htm

[Greg Oshiro]

Duder--

Welcome to the wonderful world of tri-amping. Beware the pitfalls...

The REW help file is very good. Reading it and googling the 'net will help a lot.

Way back in 1978 Siegfried Linkwitz (as in Linkwitz-Riley crossover) triamped his home stereo. He now has a web site:

http://linkwitzlab.com/

This describes the 1978 project:

http://linkwitzlab.com/Removed%20pages/x-sb80-3wy.htm

You have the advantage that the MiniDSP can create filter shapes by tapping numbers into the computer rather than rebuilding analog circuits. I have done both and DSP is easier. I know a bunch of folks who think analog is better, though. Analog is not practical for K-horns.

I have measured K-horns and found that placing the mic 36" from the squawker motorboard and 51" above the floor (midway between the squawker and tweeter heights) works well if a large pile of acoustically absorptive stuff (sofa cushions and wool blankets in my case) is placed on the floor between the K-horn and the mic stand. This reduces the floor reflection. You may find that some absorption is also needed to reduce the ceiling reflection. Use the "Impulse Response" (IR) function in REW to measure the reflections.

When measuring frequency response, also display the phase trace.

I have had good luck with measuring in a manner that attempts to exclude the effects of the room and measure the frequency response of the speaker alone. Using these measurements, select EQ and crossover filters to smooth (and maybe flatten) the frequency response. Then do the rest by ear.

Recommended approach:

1. Remove any and all passive crossovers from the signal path. Amp outputs go directly to driver terminals. Use some sort of multipin connector that prevents patching the LF amp channel to anything other than the LF driver. Neutrik Speakon NL8 connectors are a low-cost, readily available method.
2. Verify polarity of all drivers from the input of the miniDSP to the driver outputs using a phase popper. They should all be the same. If they aren't, fix it. Drivers have been known to leave factories with bad polarity.
3. Set up the mic as described with no fuzzy stuff on the floor. Look at impulse response of the **squawker alone** with no filters in place **at very low level**, maybe 0.10 volts at the squawker driver.. Any peaks in the impulse response after the first arrival are reflections. The time difference between the first arrival and each reflection can be converted into the extra distance the sound had to travel (squawker > reflector > mic). Find out what those surfaces are and place absorption on said surfaces. Usually the first two surfaces are the floor and the ceiling. In my case, I just used absorption on the floor
4. Place fuzzy stuff on floor. Look at impulse response. One of the reflection peaks should have changed.
5. Window the IR beginnning just before the first arrival and ending just before the first "significant" reflection. The longer in time that the window is will make the measurement valid to a lower frequency. Try to reduce enough reflections to get a 10 mSec window. Look at the magnitude and *wrapped* phase of the frequency response.
6. Insert an initial high-pass filter in the squawker signal path, say 400 Hz, 12 dB/octave Butterworth. EQ the squawker as flat as you can in the 800-4000 Hz range using all but 2 of the available EQ filters. I prefer to use cut only with filters wider than 1/6-octave. Your experience may lead you in a different direction.
7. Without moving the microphone or changing any REW settings, do a similar thing to the **tweeter alone** with a 5000 Hz high pass filter. The phase trace will be a mess. Ignore it for the time being. Don't attempt to flatten the tweeter to 20 kHz. Maybe a 6 dB boost to extend the response a bit. There will be holes in the slope below the high pass filter. Boosting worked for me in this case.
8. Once the squawker and tweeter are smoothed, increase the delay on the tweeter until the phase trace cleans up. I ended up with 1.6 mS. YMMV.
9. Experiment with a squawker low-pass filter, tweeter level, tweeter delay and the tweeter high-pass filter to get these results: "flat" portion of the magnitude curves at the same level; magnitude curves crossing over at -6 dB; **the phase curves directly overlapping from one octave below the crossover to one octave above the crossover**. The phase curves are much more important than the magnitude curves at this point. One unused EQ from the squawker and one unused EQ from the tweeter may be needed in the crossover region to make this happen. I couldn't get all the way there, but it's a goal. If required, sacrifice the magnitude criteria to get a better phase overlap. This will take a long time. I do this sort of thing professionally and I spent about 10 hours in total on this (3 tries to get where the system is now). This is where you will learn what filters do to magnitude and phase. I had the benefit of an analog electrical engineering education. You get to learn by experience.
10. Turn on both the squawker and tweeter. The magnitude curve should be smooth (but not neccessarily flat) at crossover and the phase trace should be smooth. If the magnitude curve is not flat through the crossover region, EQ it flat **using the EQ that affects the low, mid and high outputs**. I call this the "overall EQ". Boosting may be required. If there's a narrow, deep notch at crossover, something is wrong. Go back and check polarity and delay. Repeat step 9.

This has taken longer than I thought. When you get this far, we'll work on the woofer/squawker crossover. It gets weird...

This is what I ended up with after my first attempt:

http://community.klipsch.com/forums/t/156476.aspx?PageIndex=1

My measurement system is not REW, so your data will look different.

Since then, I've had the system in 3 different rooms, the LF has been tweaked countless times and various golden-eared folk have helped me tweak by ear.

[The Dude]

Greg thanks for the detailed info, I was thinking tonight to put this project on hold. You see, maybe I forgot to mention is that my room isn't finished and maybe this will hurt me. As I would be trying to fix all the room modes along with the frequency response of the drivers. I was thinking of making some sound panels for the ceilings as they are opened rafters at this point(hence why I feel I should finish my basement first). I remember you posting on your tri amped system a while back. There is lots of info in that thread I think it was well documented, I hope to do the same to help those out there such as my self.

[ZombieWoof]

I myself would continue tweaking & finishing the room ,

by the time the room is done ,

you will have learned how to man handle that miniDSP [H]

you then would be a miniDSP expert & then help us (me)

I plan getting some miniDSP units in the future

[The Dude]

Fast, believe it or not the Minidsp is easy to figure out depending on the plug in. I am using the 2 way crossover plug in. The biggest thing is knowing what numbers to plug in. There are a couple of tools I am trying to figure this out, when I am not being lazy I will post them. OK you got me all I have to do is copy and paste, but man that sounds like a lot of work for some reason. But as far as actually using the minidsp it seems to be simple.

http://www.holmacoustics.com/holmimpulse.php, I haven't dug into this program yet but I think its something like REW just different and more defined.

http://audio.claub.net/software/ACD/ACD.html, I think this is to help calculate the right filters, crossover etc.. not sure yet.

Heres a screen shot of the platform for minidsp 2 way advanced.

[The Dude]

Once your there, you then click on the function you want to setup for instance here is the first one input gain.

[The Dude]

Then the parametric eq.

[The Dude]

Then the crossover, I am sure you get the point. But I just figured out how to use the snipping tool on Vista, and am having fun for some reason. Maybe its the fact of learning something new, don't know but anywho.