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Brainstorming ideas for now...need feedback.


autokelley

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Greetings; Before I begin with my idea's, I want to thank everyone for any info or further expansions upon my idea's. So... what I am researching as of now is the idea of bi-amping my K-Horns and using ALK's trachorn upgrade. I would be using my Sherborn monblocks [solid state] for the bass and using class A tube amps for the horns. An active electronic crossover w/4th order 24dB/octave with -6dB at the crossover points would be used. My idea's originated while reading Rod Elliott. www.sound.westhost.com and http://lenardaudio.com/index.html and much reading on building your own tube amps. I have yet been able to hear my K-Horns through tubes, but I understand they sound beatiful. Also after doing much research on active vs. passive crossovers, I am leaning towards the active approach. The decision for solid state for the bass is mainly do to the power needed. I was investigating a class AB pushpull tube amp project, but cost played a factor and tubes did not have the tightness in the bass I was looking for. And of course the efficiency of the horns do not require high power needs and the tubes should fill my needs for the warm open sound I am looking for in mid to higher frequencies. I am also in the process of building a Belle also w/same concept for my center channel...thanks for the help on locating the blueprints on the belle also. Any insight will be appreciated...Thanks, Mike.

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Are you gonna go 2-way or 3-way, and are you going with a DSP based xover that offers time alignment options?


The
reason I ask is because I've noticed that I'm more sensitive to
time-alignment in the midrange and the way the khorn bass bin shoots
sound into the room, I'm not sure time-alignment between the MF and LF
is going to be as noticeably different.


The advantage to the
linkwitz-riley crossover is that the polar lobe in the passband stays
on-axis, which is described in this article:

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

However, it requires time-alignment for that to be true...


The
linkwitz-riley crossover also assumes that the passband of the
individual drivers is also flat, which is known not to be the case with
the Khorn. Using the natural response / rolloff of the drivers could
allow for a different electrical xover slope/frequency to balance the
speaker response and end up with the benefits offered by the LR xover.


It
should also be noted that the phase response of two different
amplifiers (especially with very different topology) is likely to be
different and adds yet another variable into the mix.


And then
one last comment...the xovers in the stock khorn also include passive
equalization to flatten out the response of the total system. That
equilization should also be emulated in the active xover.


I'm a
huge fan of active xovers, but hopefully you're aware of what you're
getting into since it sounds like you'll have to do a lot of the
engineering stuff on your own. Being able to measure what's going on
has been a huge help for me in the past and is something I would
recommend investing in. I would always argue that it should be possible
to do by ear, but it's a lot harder and takes a lot of time.



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Guest David H

Mike, I just got finished with a similar project, I am running an active crossover with a pair of Decware SE84-DIY amps, and a solid state amp for the lf drivers. The results are excellent. I have not spent any time on dealing with time alignment, because the initial results were so much better than passive crossovers I am still in awe. Switching back and fourth from steeper slopes to first orders, I still prefer the blended sound of the simple first order butterworth between the lf and midrange, and go to steeper slopes on hf.

Here is a link to the Decware amp DIY build I did. http://forums.klipsch.com/forums/p/120412/1212998.aspx#1212998

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Thankyou for the info, and yes I agree with you. I am still reading, reading, reading. I read the link you gave me and found the info very informatve and helpful. I am in the process of learning the measurement process. If you have any links in that area, let me know. Thanks again.

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Guest David H

I am using a modified Behringer DCX2496 with capacitor coupled output stage directly off the DAC.

This is one of many mods available for this unit.

post-24405-13819513795432_thumb.jpg

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Switching back and fourth from steeper slopes to first orders, I still prefer the blended sound of the simple first order butterworth between the lf and midrange, and go to steeper slopes on hf.

I'm assuming you've got phase alignment in the xover band.....but do your polars match pretty well at the xover frequency?

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Thankyou for the info, and yes I agree with you. I am still reading, reading, reading. I read the link you gave me and found the info very informatve and helpful. I am in the process of learning the measurement process. If you have any links in that area, let me know. Thanks again.

You mentioned Rod Elliot....have you seen all the articles he posts on his website?
http://sound.westhost.com/articles.htm

As far as measurement info, I don't know of any single website that really covers everything. However, I would recommend looking into REW, which is a free piece of software that has a good support forum:
http://www.hometheatershack.com/roomeq/

I use the Behringer ECM8000 microphone, but Dayton apparently is offering a slightly better microphone for the same price:
http://www.parts-express.com/pe/showdetl.cfm?Partnumber=390-801
Calibration files can be found for both on the Home Theater Shack forums, and there are also links to people offering relatively cheap calibration services.

For the Microphone preamp and audio interface for my laptop, I'm using the Tascam US-144:
http://www.tascam.com/details;9,15,70,16.html
This is a real nice unit that provides phantom power and runs off the USB power. It's real nice because a battery powered laptop has a floating ground reference, which means no ground loops when you walk up to any sound system to measure it. The only downside is that the gain structure gets a bit complicated when trying to do the loopback feature. I don't think it's a big deal, but it does require some extra thinking.

For what it's worth, I have access to the full version of Smaart, but I prefer using REW for most of the measurements I do. I would highly recommend going through the online manual and getting comfortable with the software and measuring process. I would also highly recommend doing your measurements outdoors as it will be infinitely easier to interpret.

Maybe I can write a short basic tutorial on dialing in a crossover. There are some basic principals that are really easy to demonstrate and show up well in the measurements. Getting acoustic summation through the xover band is usually pretty easy, but the overall voicing of a speaker and all the frequency shaping stuff is really more of an art and something I'm still exploring.

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I am using a modified Behringer DCX2496 with capacitor coupled output stage directly off the DAC.

Is there a schematic for that circuit? "Directly off the DAC" raises all sorts of warning flags, so I'm kinda curious what the actual implementation looks like. It's pretty much impractical to build quality analog reconstruction filters inside an IC, so most DACs rely on a separate lowpass stage after their output to properly reconstruct the signal. I also don't think the output impedance and current sourcing capability of a DAC is generally optimized for driving a line out. Heck, most of the good DACs are differential outputs and require a differential to single ended conversion first.

Maybe there's other stuff going on in the box and you're including the reconstruction filter / single ended conversion as part of the DAC?

Btw, a cheaper way to fix the crossover distortion of a series lytic is to put a DC bias on one side of the cap (or put two caps in series at the output and put the dc bias inbetween them)....and then you wouldn't need to use such a big/expensive film cap (at least I'm assuming that's what that is....). Or if you read Rod Elliot's article on capacitor nonlinearities, you can just make sure you've got a sufficiently low output impedance driving a sufficiently high input impedance...

http://sound.westhost.com/articles/capacitors.htm

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and cost less too.

Not if you want it to have lower distortion and stay flat over the entire audible band...

Well, I used Edcor transformers ($9-12 per channel) and the unit is within a half dB from 20-20K. I ran some test signals and did not see any ringing, but my testing was not exhaustive. As far as distortion ... what specifically would you measure? Steady-state THD may not be that informative.

-Tom

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