But if he is not getting a lot of early reflections, dispersion pattern matters less, and on-axis response predominates. For instance, if he is listening fairly close to the speakers in a large room, or if he has substantial acoustical treatment on side walls, etc. I don't know what his situation is, but there is probably a reason why he likes this kind of setup.
post 641...ding ding ding...good answer. It's like studio monitors; not near field, not far field but a very good compromise. A narrower sweet spot but extremely accurate on axis with very little noticeable unwanted "side" effects. Come on wide band TexTreme! My next dream come true!??
Anyone in Asia or Europe have input when the next TexTreme models are being released? Other than the TexTreme tweeter, MW16 and 19; that is all we know in N America unless I missed something in the last week or so...thanks!
Anyone in Asia or Europe have input when the next TexTreme models are being released? Other than the TexTreme tweeter, MW16 and 19; that is all we know in N America unless I missed something in the last week or so...thanks!
I recently listened to another DIY'er setup, he is also using Hypex fusion amps for active crossovers.
He has the Eton 3 inch driver. Eton 3-212/C8/25 HEX Bass-Midrange This one I believe.
I was impressed with the sound of this driver. You might consider it...
He has the Eton 3 inch driver. Eton 3-212/C8/25 HEX Bass-Midrange This one I believe.
I was impressed with the sound of this driver. You might consider it...
Interesting 3" driver. Where were the crossover points/slopes? The driver's FR does not seem all that smooth above 3.5kHz or so.
Attachments
Current listen is Eddie Higgins jazz Trio Christmas songs on YouTube. Lo Fi? yep; does it still sound really good? Yep! The bowed acoustic bass harmonics are very rich and natural here in spite of Lo Fi streaming. Next on my list for a CD I think. I need a SACD player as I have many hybrid SACD discs. Mostly classical but also many jazz. I also have an outboard hard disc for my PC and have purchased many FLAC albums online; mostly classical and mostly from Europe (no surprise...hah!). I think I have what the Recording Engineers hear with my present system. VERY close for sure...highly accurate without any hype or artificial additions or deletions for "effect". Just looking for the next best thing which I am convinced is TexTreme fullrange/broad-band "mids". Fingers crossed...
Indeed
I just wonder... cause alot of nearfield monitors are very keen to describe how smooth their power response are - like Kii, DXT-MON from Heissmann and even the old Behringer monitors. They are all designed with a good even dispersion in mind. Even very good treament of first reflections - would still not beat a well designed speaker. This is why I ask. Cause is the disliking of possible phase issues, worth the trouble with potential off-axis behaviour?
He crossed at about 330 Hz to a 10 inch driver, Seas L26ROY. He crossed at 2.7 kHz (I think?) to a bliesma T25B 25mm berylium dome.
We got together to compare the textreme drivers (MW16 and TW29) to the Eton Hex and Bliesma berylium. Both of us had prototype cabinets at this point, and we evaluated in mono.
Within the limits of the preliminary nature of the comparison, I thought both systems sounded good. really good. Whatever differences there were could easilly be attributed to small differences in EQ.
He follows this site, so perhaps he will share his thoughts...
J.
Good news on the Eton!
I guess maybe I implied I don't have a wide sound stage by using the word focused. I have just the right amount actually, kind of the Goldilocks zone.
I always do fine tuning and final tweaks in mono as well; just 1 channel at a time. For drivers 6 inch and smaller; when I first try them out; I do it at the table in very close range, 1 meter or less to my ears. This way; I am minimizing any room reflections and get a better feel for just the driver itself, many times full range with no filters or crossovers (at first).
There is more than one right way to do things; whatever works best for you doesn't mean everyone agrees and I'm getting used to all the various methods and opinions. (some of these discussions and even arguments are helpful to me, hopefully to others as well).
Back to TexTreme; I know that most of the Papyrus Satoris I have personally had; it usually takes a 2nd order LP on the high end because they all seem to have that rising high frequency response. I am hoping for this to be extended further up in frequency by having a more refined cone material (TexTreme in particular). This is the real reason I got off on a tangent about crossing higher than usual, beaming, etc. Like I said, I am mostly interested in a small wide band driver and would really love it if I could get away with 1st order HP and LP both.
I guess maybe I implied I don't have a wide sound stage by using the word focused. I have just the right amount actually, kind of the Goldilocks zone.
I always do fine tuning and final tweaks in mono as well; just 1 channel at a time. For drivers 6 inch and smaller; when I first try them out; I do it at the table in very close range, 1 meter or less to my ears. This way; I am minimizing any room reflections and get a better feel for just the driver itself, many times full range with no filters or crossovers (at first).
There is more than one right way to do things; whatever works best for you doesn't mean everyone agrees and I'm getting used to all the various methods and opinions. (some of these discussions and even arguments are helpful to me, hopefully to others as well).
Back to TexTreme; I know that most of the Papyrus Satoris I have personally had; it usually takes a 2nd order LP on the high end because they all seem to have that rising high frequency response. I am hoping for this to be extended further up in frequency by having a more refined cone material (TexTreme in particular). This is the real reason I got off on a tangent about crossing higher than usual, beaming, etc. Like I said, I am mostly interested in a small wide band driver and would really love it if I could get away with 1st order HP and LP both.
Well.... if you get the result your looking after.... then simply use the method you find best
I both listend to others creations and build alot myself. Never had succes with the high crossover and for me, the good power response simply does the trick for me
I tried one of those heavily damped rooms. And I think I would still like a speaker that was build by basic theory no matter how few reflections there might be
Driver time alignment
I found this; in the old days; we used a recording with a "tick" or a very short signal. Using a mic. and O'scope; you positioned the tweeter relative to the mid or mid-woofer until the combined signals were on top of one another.
I found this; in the old days; we used a recording with a "tick" or a very short signal. Using a mic. and O'scope; you positioned the tweeter relative to the mid or mid-woofer until the combined signals were on top of one another.
You don't. You measure the offset between drivers.
I use a technique that is based on the interference between two signals, one from driver 1 and the other from driver 2. It's sensitive to below 1mm. The measurement is done with the driver installed and in their final positions.
1. Set up the mic in the listening position, or at least on that axis. Do not move it during the measurements, or adjust the gain.
2. Make a response measurement of driver 1.
3. Make a response measurement of driver 2.
4. Make a response measurement with driver 1 and driver 2 reproducing the same signal, e.g. feed the same signal into both L and R inputs of a 2 channel amp and connect the drivers to the amp's outputs.
That's the measurement part. These measurements should not be SPL only, e.g. with an RTA or the like. You will also need phase information. I save the data to FRD files, for example.
Now to determine the offset. You import the data into a program such as a crossover modeling program. I believe PCD works for this. I use my ACD tools. You need to set up the program to add the two responses, both amplitude and phase. The sum is compared to the measurement (also imported) of the two drivers playing at the same time, e.g. #4 above. The program is simulating the in-air sum by combining measurements #2 and #3. What is missing is the z-offset. You find this by trial and error, via the program, by adding some amount of z-offset to one of the driver (the one farther back from the mic when you did the measurements) and then comparing the sum that the program is calculating with what you measured. When you have the correct z-offset, the two curves will overlap.
Jeff Bagby wrote this up in a whitepaper on this, which I am attaching.
Um, a little light on details there. Can you describe the full Soundeasy procedure please, or provide a reference to the steps e.g. from the manual?
Also, Soundeasy is a very complicated program that costs hundreds of dollars. The method I described uses free software and is a little more DIY friendly IMHO. But I would love to know how Soundeasy does it. It has lots of interesting bells and whistles for sure.
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Any dual channel software can do this as it measures the output line to find the reference impulse, then times the mic's measure of the speaker impulse from that reference IR. The reference is always the same so differences between drivers can be seen directly, in microseconds. I'm at work so don't have time to go more into it.
OK, I see what you are talking about now. Sure, the program will spit out some kind of delay info in dual channel mode. But I have always been very skeptical of that. Some programs use the initial peak of the impulse as a time marker. Exactly how it is done is not always easy to find out. If you have two drivers like a tweeter and a woofer with very different bandwidths, that peak will appear in two different bins with respect to the stimulus due to rise time differences. On the other hand, the method I have outlined is quite reliable, and it is one that can be confident in. Also, not everyone has dual channel measurement capabilities.
i use two channel measurements with arta and it works just fine, this spares me a lot of trouble messing with minimum phase curves to calculate time differencies between the drivers. each time i have tried the minimum phase method i could not get the drivers to really sum correctly, there where always some frequency errors induced relative the reference curve (both drivers measured at once).
with arta the measured phase can still slide slightly between repeated measurements especially in the frequency extremes where the sound level has dropped much relative to the passband
with arta the measured phase can still slide slightly between repeated measurements especially in the frequency extremes where the sound level has dropped much relative to the passband