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Anyone using Khorns have examples of aligning drivers either physically or with delay etc.....


charlieboy

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1 hour ago, charlieboy said:

Great information everyone. Can anyone show me Klipschorns or similar photos and the dsp setup you are using?

 

On 11/4/2019 at 5:31 PM, Chris A said:

See the following thread for using a DSP crossover to correct the time alignment and EQ of Khorns:

 

 

Because of the degree of time misalignments in the bass bin--midrange (over 2 wavelengths) and midrange-tweeter (over 8 wavelengths), I don't recommend using analog active crossovers with Khorns, which cannot correct those misalignments.  It is the correction of those time misalignments that I've found is the source of the improvement in sound quality.

 

Chris

I believe Greg's settings are in his thread.

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I believe that I have his Yamaha SP2060 crossover used in the above picture.  He used D-75 Crown amplifiers (not D-75As).  The Klark-Teknik DN410 Dual Parametric Equaliser would not be needed in the case of the Yamaha SP2060 crossover, as it has enough biquads internally to handle the EQ and crossover filters required. The JVC K2 is nothing special, IMHO, as is the Sony DVD player.

 

Chris

 

 

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I wouldn't prefer to use one based on comments from others whose opinions that I trust (mostly Jub owners).  But I have no direct experience with the sound quality of the lowest-priced DriveRack crossovers.

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I did some research on the Driverack family and found some comments that you needed a rather hot input level for it to work properly, but having proper gain staging is important for most complex systems. When driven correctly the comments were that they sounded fine.

 

I am planning on a Driverack 260, but I will be taking things out for events. It may be a compromise, but it is one I am willing to live with for a while.

 

Bruce

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Yes, a DBX drive rack would do the job.  I have used the DBX Drive rack in our church to run a biamped setup with subs.........and it has a lot of features.  However, I would not be happy with it in my home setup as it is not what I consider resolving enough and audiophile for home use. 

 

 

Edit.  Sorry Bruce.  Our posts crossed as I was answering his DBX question.  But that is my experience with the 260.  I programmed it and tweaked it quiet a bit.  I used if for other stuff too like compression, feedback removal, pink noise generator..........like I said has a lot of features.  Just not really a great sounding home audio tool.

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The setup pictured does have a Yamaha SP2060. Personally, I would take it over a DBX (or Rane, or Ashly Protea etc) because it allows for a digital input. Avoiding that extra conversion offers a big benefit. If you look at the Yamaha spec sheet (or contact Yamaha directly), they claim it will only accept a AES/EBU digital input (a pro unit with a XLR connector) and not a S/PDIF input (consumer unit usually with a coax connector). However, this is not accurate. The Yamaha will accept a S/PDIF input (build a cable with  an XLR on one end and a RCA/coax on the other end - short pin 1 & 3 and keep the cable length short). It will work fine. In fact you can also do this with a Behringer DCX2496. It avoids an analog input stage and the extra conversion. IOW, you do not need a box that converts a S/PDIF to a AES/EBU signal. The other alternative is a CD player with a AES/EBU output, but they are few and fairly expensive. BTW, a used Yamaha SP2060 can cost well less than a new dBX unit. Buying a new SP2060 is far too expensive. 

 

The reason I am adamant in this recommendation is that the ADC on most DSP units is expecting a 8-10 volt analog signal. Consumer equipment (CD player) is at least 12-15 dB less. When the user then puts a preamp before the ADC, then it is frequently about 24-30 dB less and extra analog stages are inserted. My suggestion is simple: either 1) find a DSP unit with a digital input (and avoid the analog stage and the extra conversion) and use a 4-channel potentiometer after the DSP unit's DAC (and before the amplifiers - effectively eliminating the need for a pre-amp). or 2) be very careful in in feeding the DSP unit with a hot analog signal (most users will not do this and will not read up on the concept of "gain structure").

 

Units with a digital input do NOT include the Ashly Protea, dBX, Rane (IIRC), Electrovoice dx38, Xilica XP series.

 

Units that will accept  a digital input are the Behringer DCX2496 (inexpensive but the analog output section is best bypassed for those of you who are DIYers), Yamaha SP2060 (pricey), Xilica XD series (many interesting filter functions), and the Electrovoice DC one. There are certainly others and they can get even more expensive. 

 

I am done with my rant. (but the problem outlined above, is what folks are frequently describing when they claim DSP units are "noisy" or "thin" etc. - it does not need to be that way).

 

Good Luck,

-Tom

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14 hours ago, PrestonTom said:

...The Yamaha will accept a S/PDIF input (build a cable with  an XLR on one end and a RCA/coax on the other end - short pin 1 & 3 and keep the cable length short). It will work fine. In fact you can also do this with a Behringer DCX2496. It avoids an analog input stage and the extra conversion. IOW, you do not need a box that converts a S/PDIF to a AES/EBU signal...The reason I am adamant in this recommendation is that the ADC on most DSP units is expecting a 8-10 volt analog signal. Consumer equipment (CD player) is at least 12-15 dB less. When the user then puts a preamp before the ADC, then it is frequently about 24-30 dB less and extra analog stages are inserted. My suggestion is simple: either 1) find a DSP unit with a digital input (and avoid the analog stage and the extra conversion) and use a 4-channel potentiometer after the DSP unit's DAC (and before the amplifiers - effectively eliminating the need for a pre-amp). or 2) be very careful in in feeding the DSP unit with a hot analog signal (most users will not do this and will not read up on the concept of "gain structure")...Units that will accept  a digital input are the Behringer DCX2496 (inexpensive but the analog output section is best bypassed for those of you who are DIYers), Yamaha SP2060 (pricey), Xilica XD series (many interesting filter functions), and the Electrovoice DC one. There are certainly others and they can get even more expensive. 

 

I am done with my rant. (but the problem outlined above, is what folks are frequently describing when they claim DSP units are "noisy" or "thin" etc. - it does not need to be that way).

 

These are useful observations and a good suggestion.  User etc6849 (Ellery) uses a Xilica XD series crossover with digital input from a PCIE-AES/EBU conversion card in a PC running JRiver, and reports an excellent noise floor (i.e., completely inaudible) and excellent sound quality--better than he's ever had before.

 

I've had good luck using the Xilica XP series, Yamaha SP2060, and ElectroVoice Dx38 crossovers driven from the analog outputs of a consumer A/V preamp/processor in terms of sound quality and sufficient gain.  The only noise floor that I can detect (when I put my ear into the K-402s) is from the unbalanced connections to the First Watt F3 that I'm running on the TAD drivers.  This is common mode noise (60/120/180 Hz), but it doesn't bother me enough to do anything about it compared to the fidelity of the F3 on the TADs.

 

Chris

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I was able to get the Electrovoice Dx38 to sound fairly good, but it required particular attention to giving it a fairly hot signal to the ADC. Since I was using attenuators after the Dx38's DAC this was not a big problem. However there was still a bit of noise at the speakert, but I did not thoroughly trace whether it was the analog input or output sections on the Dx38 or elsewhere.

 

Ultimately, I was disappointed in the limited number of filters available on the Dx38. At the time I was exploring whether I could substitute (or reduce) the "crossover filter" by substituting a handful of PEQ filters. Since the Jubilees need some EQ already, the limited number of filters became an issue. 

 

Good luck,

-Tom

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10 minutes ago, Chris A said:

Were you using balanced connections (XLR-XLR) between the boxes throughout?

No I was not consistent.

 

I had done both (balanced and unbalanced) on the CD Player to Dx38. I saw little difference on the scope and heard no reliable difference when listening. After the DAC on the Dx38, it was fed as single- ended to the potentiometers (stepped attenuators) and single ended to the amps.

 

In general, I have kept cables short and shielded and avoided nearby power cords (not always easy to do). So I have never been overly concerned about the advantages of balanced connections. Perhaps I should re-think some of my habits.

 

Good Luck,

-Tom

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1 hour ago, PrestonTom said:

...the limited number of filters became an issue. 

This was the reason why I moved on to the Yamaha SP2060SP crossover. I also found that the Dx38 started to act strangely when using all the PEQs simultaneously with large gains.  It would act as if I hadn't made a change in filter settings, then suddenly "anneal itself" to change its overall PEQ curves as they probably should have been acting--but the change happens quite dramatically--and doesn't return to its prior overall output characteristic vs. frequency.  I'd have to start all over again with the PEQ settings (layering them on gingerly so as not to incite the same behavior again--usually unsuccessfully). 

 

I spent many days in my setup fooling with those issues, and finally when I got frustrated enough to swap it out with the Yamaha (which I had on hand meaning to try it), I simply dialed in the PEQ settings in the Jubilee PEQ table from Roy's anechoic chamber settings, and Voila!  Ruler-flat TAD response the first time.  I never used the Dx38 again after that experience.  I still own a Dx38 that I plan to use in a garage-based system, but I I wouldn't recommend it for Jubilee duty today--but only because there are other/better choices that are now on the market which are usually less expensive (even in used condition).

 

Chris

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On 11/4/2019 at 3:21 PM, charlieboy said:

Anyone have the tweeter on top at the rear to match the mid driver in alignment? How where the results? I know it all can be done in the digital world with delay but I was just curious what people were doing or what gave the best results. Or if most just enjoy them stock "as is" like I have been lately. I intended to start down the tri-amp direction and DSP about 13 years ago with a set of Khorns I had but sold them and never pursued it after that.. Since then I have recently purchased a pair of Khorns and was wondering if it was worth trying again? They are the tweakers dream and I don't mind starting that process all over again. It's just hard to get motivated lately since I have just been enjoying my Altec 604H II's so much without all the extra tweaking other than tuning the cabinet and really good crossovers.

I don't have K-Horns, I have KLF-10's. But I have been playing with a top mounted tweeter. I found a titanium dome tweeter at Parts Express on clearance for 6 bucks, I thought "How could I go wrong?". They came with their own 3.3uf cap (which I will upgrade eventually). I mounted them on a piece of 2x4 along with an L-pad. Lining them up with the horn diaphragm is the best position, reflections or not. I'm also trying different angles which can really open up the sound-stage. I recommend doing this, if for no other reason than all the fun of experimentation and associated learning. I think they will become a permanent part of the speakers, at least until I can get a pair of MAHLs.1856304658_2019-11-0721_03_28.thumb.jpg.22c74598e919307301690a263aa9076f.jpg

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On 11/4/2019 at 5:56 PM, Chief bonehead said:

Question. What about the reflection of the horn off the top of the panel?

Should've known Mr. Bonehead wasn't really asking a question here, but rather making a point. Upon further testing, I noticed that lining up the tweeter with the horn diaphragm, while making a big improvement in sound, does cause reflections from the top of the cabinet that cause destructive interference at positions in front of the speaker. Further testing will show if there is a way to get the best of both worlds. I don't think padding or insulation will work. Maybe a structure of some sort... Mr. Bonehead, please chime in.

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On 11/6/2019 at 3:05 PM, PrestonTom said:

2) be very careful in in feeding the DSP unit with a hot analog signal (most users will not do this and will not read up on the concept of "gain structure").

Yes to this for sure. I feed my Xilica from a Presonus mixer. It is a balanced connection. The mixer is capable of putting out +24dBu. I still have the input channel gains up all the way (+15dB) on the Xilica. This gets the input meters to flicker occasionally but only at loud levels. The Xilica is made for professional and live sound systems and can take large and dynamic input signals. Seems like in home reproduction systems it is barely being tickled.

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5 hours ago, MechEngVic said:
On 11/4/2019 at 8:56 PM, Chief bonehead said:

Question. What about the reflection of the horn off the top of the panel?

Should've known Mr. Bonehead wasn't really asking a question here, but rather making a point. Upon further testing, I noticed that lining up the tweeter with the horn diaphragm, while making a big improvement in sound, does cause reflections from the top of the cabinet that cause destructive interference at positions in front of the speaker. Further testing will show if there is a way to get the best of both worlds. I don't think padding or insulation will work. Maybe a structure of some sort...

 

While I mentioned that in this thread before he did, I'm not claiming to posses as much knowledge / experience as he.  That information should rightly be considered as so much "low-hanging fruit."

 

Same for:

 

4 hours ago, babadono said:
On 11/6/2019 at 6:05 PM, PrestonTom said:

2) be very careful in in feeding the DSP unit with a hot analog signal (most users will not do this and will not read up on the concept of "gain structure").

Yes to this for sure. I feed my Xilica from a Presonus mixer. It is a balanced connection. The mixer is capable of putting out +24dBu. I still have the input channel gains up all the way (+15dB) on the Xilica. This gets the input meters to flicker occasionally but only at loud levels. ...

 

If one doesn't make large use of the available bit-depth, they're really limiting themselves to the equivalent of lesser equipment.  It'd be better to flirt with bouncing off the ceiling than playing mainly on the floor.

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Before we go overboard about gain structures and beating one's breast about how much voltage you're feeding your DSP crossover(s), just remember that most DSP crossovers use 24-bit depth words...which is at least 8 bits more than most people use/can possibly hear in their listening rooms, unless one is listening at very high SPL in-room to whispering levels in the recordings.  It exceeds the analog SNR by quite a bit even at 16 bits of word depth.  Also note that unless you're using better-than-CD quality recordings (i.e., "higher resolution" and recording practices that begin to use those extra bits--like extremely dynamic acoustic performances), you're probably working with the placebo effect when talking about being able to hear differences.  The only effects that I can hear are on the decays of transients, and that only works for extremely dynamic recordings.

 

My guess is that we're only talking one or maybe two bits of effective word depth by not driving DSP crossovers to "commercial" voltage levels.

 

Chris

 

 

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