Jump to content
The Klipsch Audio Community

Chris A

Heritage Members
  • Content Count

    8995
  • Joined

  • Days Won

    3

Chris A last won the day on November 23

Chris A had the most liked content!

Community Reputation

3561 Legendary

About Chris A

  • Rank
    Music Enthusiast

Profile Information

Contact Methods

  • Website URL
    https://drive.google.com/drive/folders/0BwQoJ7rtphauWHhtZEZndUw4M0U

Recent Profile Visitors

39837 profile views
  1. In the cases I've had, the tracking information shows the packages getting to California within 2-3 working days...then it goes black (i.e., I guess that the carriers can't call US Customs out on the delays, or they'll get punished even more). This is a good place to lay out the following about this maligned topic of dual-diaphragm compression drivers, and avoid some notable memeplexes (information that isn't correct and was not actually achieved by first-hand learning) that audiophiles like to perpetuate in this pastime. These type of drivers outperform anything else for home hi-fi duty except perhaps TAD beryllium diaphragm compression drivers--because of their unique design that avoids having to use two separate horns to do one job. This isn't a cone/cone or cone/compression driver co-axial. It's a dual diaphragm ring-radiator compression driver--with two separate electrical circuits, including four connectors on the back of the driver. Please refer to the diagram below for a cross section view of this driver to understand its configuration: red = tweeter diaphragm, green = midrange diaphragm, yellow = compression chamber... This driver doesn't suffer from high frequency chatter that you hear with dome-type titanium or aluminum compression drivers (listening to very clean ride cymbals in jazz and other music genres) due to the elimination of the center dome structure of the diaphragms, which makes the diaphragms effectively much structure stiffer and lighter. You'll also note that the tweeter diaphragm here is a little closer to the center phase plug (the spike-looking object in the middle) than the midrange diaphragm. In the BMS 4592ND, the tweeter diaphragm leads the midrange diaphragm in time by a small amount that's equal to one wavelength at the crossover frequency (nominally 6 kHz). If you bi-amp this driver and EQ the response of each diaphragm to be flat in SPL vs. frequency, and correct the time alignment of the diaphragms to be coincident at the phase plug then you end up with a full-range driver (nominally 375 Hz to 21 kHz) like the TAD TD-4002 or 4001, but without the cost of the beryllium dome diaphragm, and with no concerns about potential toxicity of beryllium. (Note: no one I know would ever use the TADs in a home environment in such a way that would cause the beryllium diaphragm to hit its stops, so there is no real hazard probability with the TADs.) Additionally, the polyester dual diaphragm BMS 4592ND diaphragms have the geometries of ring radiators that precludes diaphragm chatter. The need for extreme stiffness of the diaphragms is eliminated, and the need for the extremely lightweight+stiff beryllium material is thus avoided. Additionally, the diaphragm material itself and its geometry have much better self-damping properties than a single done-shaped diaphragm of the TADs. One additional point to be made: the dual diaphragm driver breaks the frequency spectrum into two portions: ~375 to ~6000 Hz, and ~6000 to ~21000 Hz. So the FM distortion sidebands are much lower than using a single diaphragm to cover that frequency range. This means that you get extremely clean output even at very high output levels. And each diaphragm design can be better optimized to produce more effective and cleaner output for the each of the midrange and tweeter frequencies than a single diaphragm can be designed to handle. The BMS 4592ND also uses neodymium permanent magnets--which is much lighter overall than ferrite because it is more able to concentrate their magnetic fields at the correct locations around the voice coils, with their geometry optimized for a more effective voice coil field strength than a typical dome-type diaphragm compression driver. I do not recommend either version of the driver (the one-diaphragm 4592-MID, or the two diaphragm 4592ND) if you do not use acoustic measurements (I strongly recommend REW or even something more powerful--but not "room correction software") to guide your EQ that is required for either of these two drivers on whatever horn that you choose to fit to them. I strongly recommend using DSP crossover for that purpose, either driver version. To not use a DSP crossover with these drivers and whatever horn you choose would be very much like the old days of mechanical breaker points on engine distributors. You can just plop in a new distributor or replace the points in the engine and it might work, but usually not very well. If instead you use a timing light, then you can dial it in properly (i.e., rotate the distributor housing proper point) and get proper engine performance. These drivers similarly require dialing them in--in order to get the proper performance out of them. If you do not do this, the driver/horn timbre will be off, and the sound of the driver will be oddly irritating in the higher registers and missing other frequencies in its high frequency response. If you need help to measure and use a DSP crossover to dial in your BMS 4592ND (or any other drivers/horns), PM me for measurement and DSP dial-in support. _________________________________________________ One of the things that find curious is that many people accept three-way designs having separate midrange and tweeter drivers and horns. The effects of having those two separate horns and drivers/horns is actually pretty high from an acoustics perspective. The two separate drivers/horns, due to their size and shape, create interference (lobing polar output and cancellations off-axis) between the two outputs because the two sources are separated vertically by much more than a quarter wavelength at the crossover frequency--a distance of about 5/8" at 6 kHz. This small distance is impossible to achieve using two separate drivers and horns at that crossover frequency. My advice is use dual-diagram or concentric drivers if you are trying to cross to a higher frequency driver above 1000 Hz or higher...because you can't get two separate drivers and horns close enough together to avoid polar coverage problems in their in-room output--without causing other more severe issues in the polar output of the drivers. Usually it's vertical axis lobing that's the issue. Chris
  2. I believe that many/most of the TADs running on Jubilees were bought in a group buy in 2009, with MikeTN and Marion I believe slightly leading that group buy pack. As you know, the TAD TD-4002s were discontinued, probably due to the neodymium magnets (prices have risen significantly). Since then, all the 4002's I've seen are coming second hand, and mainly from commercial venues (as I'm also sure you know, and these can be problematic, since the diaphragms alone at running about $1k USD each, and that is the reason why many of the older 4002s are being sold--instead of refurbished). Neodymium magnets also don't like to be overheated--like you get in discos and other very high SPL/long operating period applications. Their Curie points are much lower than ferrite magnets. That's why the TD-4001s (same diaphragm with a 20# ferrite magnet instead of the ~11# neo magnet in the 4002) are still being used, I would guess. The 4001 prices have always been a higher than the discontinued 4002s. So you can get the performance of the 4002s--but using the 4001s. __________________________________________________________________________ I wouldn't go that way now that I've discovered how to dial in and bi-amp a BMS 4592ND to get about the same performance as the TD-4002--and without the FM distortion of having such a wide band of frequencies being reproduced on a single diaphragm. A Hypex FusionAmp (FA123) makes that very easy to do and replace the DSP crossover, two stereo amplifiers, and a couple of speaker wires to each loudspeaker, instead using a signal line from a preamp, player, or computer (running a PCIe to AES card), or an unbalanced RCA analog, balanced XLR analog, or S/PDIF [coaxial or optical] input. Chris
  3. I bought my BMS4592ND from Thomann US. Right now the Euro-to-USD exchange rate isn't terribly good (the price of the driver has risen about 15% since I bought mine two years ago, due mainly to currency exchange rates. I bought mine for right around $461 USD two years ago. Perhaps the currency exchange rate will swing back after late January (if you catch my drift). In any case, I don't think you should buy the drivers first, but put it off until you start to integrate everything. Thomann shipped from Germany to the US and it was at my doorstep three business days later (again, now the US Cusstoms import delays are so affected by politics presently, my guess is that it will take over a month for you to receive them). [I know that my heartworm medication from Canada (actually shipped from New Zealand) is presently taking 4 1/2 weeks on average. This isn't the fault of the manufacturers, sellers or shippers, but the US Gov't.] I'd like to see a little less politics involved in international sales for a change. Chris
  4. You might also compare notes with @mikebse2a3 who I believe has tried a few amplifier combinations on his Jubs. The last picture I saw had him using a McIntosh headphone amplifier (another higher output impedance amplifier like your F1J and likely your Primaluna) on his TAD high frequency channels. I believe that @rigma uses tubes (likely single-ended SETs) for two channel duty with his passive crossovers (the ones mounted on the wall next to the Jubs), but I believe that he's also tried out others, including SS for home theater duty in the past (IIRC). Both of these Jubilee owners still browse the forum fairly often, and sometimes stop to post into a thread. Maybe they'll stop by in this one. I'm going the other direction--trying out Hypex NCore amplifiers (i.e., low output impedance) embedded within FusionAmps, since I'm not much of a tube enthusiast. I may tinker a bit with a transconductance (current feedback/current drive) amplifier schema in a modified Howland circuit. I think it's well within your personal preferences of what you like to hear--and I do believe that the lion's share of what is occurring is related to varying the amplifiers' output impedances. Chris
  5. This is not correct. However, if you drop the K-402s down a bit and allow them to stick out in front of the bass bins about 7" (I call this "shortening down the K-402s"), then the two sources (bass bin and K-402 horn) are closer together. This improves the midbass response vertically even more and produces a little less phase growth through the crossover region. The minimum listening distance drops to something like 3-4 feet when you do this, and the listening involvement (in the case of using zero phase growth crossovers with a DSP crossover that I've talked about on this forum ) goes through the roof, in my experience. The comments about "these loudspeakers are meant for bigger rooms" as a ploy to say they are too big for home hi-fi is totally wrong. In fact, because they have full-range directivity, they are better performing in small rooms than small loudspeakers that spray their acoustic energy around their nearby acoustically reflective objects. In order to get the same response as Jubilees in a small room, the owner would have to completely cover the front of the room with absorption pads/panels, and then the room would probably be too dead. The MEHs are coaxial loudspeakers, so their minimum listening distance is -1 feet (you can put your head inside the loudspeaker and it sounds the same). There is no downside (except for the slightly higher levels of AM distortion that I mentioned above). It actually sounds just like my Jubilees, only more so (more coherent). If you place the MEHs in room corners--there is no downside, because the MEH mouth size couples so well to the room corner that it loads the woofers down to a very low cutoff frequency (well below 40 Hz, even better than the effect of having the woofer ports off-axis in the K-402 which drops the effective expansion rate of the horn that the woofers see--an effect that Danley himself has written about). Chris
  6. If you need support (if you decide to investigate this further), feel free to PM me. The comment that Dave made above about not needing corners is even more true with the K-402-MEH, because of the size/dimensions of the horn's mouth, and the fact that the dual 15" woofers in the K-402 aren't constrained by a exponential horn cutoff frequency (something that the Khorn and Jubilee use in their bass bin designs). The trade-off is slightly higher modulation distortion below 100 Hz, but not harmonic distortion. You actually get even better mid-bass and lower midrange performance than the Jubilee (IMHO) because the woofers are co-axial with the compression driver in the same horn, and thus are allowed to couple with the compression driver acoustically. The MEH is also 1/3 the volume of the Jubilee, 1/3 the height, ~1/3 the weight, and ~1/3 (or less, depending on the quality compression driver used) the cost of new Jubilees. The constraint is that you have to cut the ports in the K-402 horn and mount the woofers using DIY woofer mounting pads and bolts to attach to the horn. The easiest way to do this is via a KPT-305 mid-bass module, then cut the ports, attach the woofers, and enclose the back of the cabinets. The DSP crossover to dial them in is the same as a Jubilee. I'm currently testing Hypex FusionAmp FA122's with my Jubilees and will report back on what I find, probably in the next week or two. If everything goes as planned, this will simplify wiring and issues with having separate amplifiers and DSP crossover, etc. The dual Hypex NCore amplifiers in the FusionAmp have been highly regarded in hi-fi circles, and combined with the on-board DSP crossover on the same assembly, makes the choice a lot cleaner and less expensive than separate amplifiers and a good DSP crossover. It also eliminates dual speaker wires (four conductors per loudspeaker). The FusionAmps require only a two conductor preamp signal-level line, whether it is unbalanced (XLR), balanced (XLR) analog, or coaxial digital input (S/PDIF), TOSLINK optical, or AES3 (AES/EBU using XLR connectors). All input types are available to take an input from a preamp, digital server, disc or streamer player using the FusionAmps (no separate amplifiers are required). This represents unprecedented flexibility in the upsteam signal chain. A three-way FusionAmp is available for $47 more using the FA123 FusionAmp. This can be used with a dual-diaphragm compression driver, such as the BMS 4592ND to give superb performance in a three-way design instead of two-way. I can help to estimate the costs of the different MEH choices/paths that you have, but all are significantly lower compared to the other two Klipsch choices you've discussed, and the sound is like that of a Jubilee (i.e., it's using the same HF horn as the Jubilee). Any way you decide to go, I believe that you're on a very rewarding path. I can recommend the MEH and the two-way home Jubilee quite enthusiastically. The home version of the Jubilee is also slated to be revealed in the coming months, so if you intend to go that direction (and have the extra cash for the nicer looks), that is yet another option that you might consider. I can guess that it will be a bit more turn-key than current home version Jubilees--albeit at a higher price. Chris
  7. Welcome! New from the ASR forum? Yes, this particular review was done by Erin instead of Amir. He has a different measurement rig than Amir's NFS setup, and a slightly different perspective. I hope that both of them start publishing phase and group delay data as a matter of course. Like I mentioned in the ASR forum, that's the other half of the transfer function, and as time passes I've found more and more that this data is almost as important as SPL response. The group delay data posted in that ASR review thread (that I requested) doesn't look like it's terribly accurate yet, and the phase plot is still yet to be posted as of this writing. Perhaps I'll pitch in to his cause when he updates the group delay plot and adds a phase plot that's accurate. Klippel gear is not inexpensive: $10K is about the starting point, and a full-up NFS setup that Amir uses is in the $60K--$100K [USD] range. That's not chicken feed. He must have a very well-paying job. I don't expect that to change anytime soon. Apparently a significant percentage of ASR membership puts most of their money into electronics and "small loudspeakers" (like your typical "audiophile forum"). 😉 That's sort of the opposite of those here that own Klipsch Heritage loudspeakers larger than Heresies. Different priorities, clearly. I'd prefer hi-fi sound in-room myself. Chris
  8. I believe I know why--the same reason why Toole I believe advocates it: direct radiating drivers splash a lot of energy around the room, and that means that a downward sloping response tends to compensate for the increased level of early reflections at higher frequencies that direct radiating loudspeakers typically have. This is the "salt-and-pepper EQ" phenomenon that Roy has talked about with horns that begin to lose their directivity at lower frequencies. This is not necessary with K-402s in my experience: you want flat response as measured at one metre microphone distance. An additional reason is that Toole knows that many/most recordings have accentuated highs during mastering between 1-5 kHz (just below the major sibilance frequencies, since many/most vocalists can't keep their mouths away from the microphones, another interesting subject, and therefore the mixing and mastering guys wind up having to cut somewhere in the 4-7 kHz band. You can see this on many/most popular music recordings, i.e., the difference between the black line and the red line, below, from the John Peter Chapman 1996 JAES article: It's the genres that tend to the right side of the above average dynamic range (crest factor) that experience increased amounts of high frequency boost--and lower dynamic range. The problem is, of course, that the highs above 10 kHz are doubly attenuated using the Toole downward sloping SPL response for loudspeakers. I think there's one other factor--I don't believe that all people are taking their measurements at one metre. If you flatten the SPL response at one metre (easy to do with DSP), then back off to 3-5 metres with the microphone, you will get that down-sloping response automatically. Most people that argue this point I believe are trying to measure loudspeaker response too far away in a noisy listening room full of early reflections if taken that far back. This is also my gripe with "room correction software". The major failing of these applications is that they are taking the measurements with the non-minimum-phase room reflections embedded in the measurements, and they typically do a lousy job of cutting those effects out of their measurements. Unless you're outside on soft ground, I'd strongly advise taking your in-room measurements at one metre microphone distance with the microphone centered on midrange height, and aim for flat response. Unless using half-space microphone measurement techniques with the microphone on the ground, in which case, the advice gets a lot more complicated, and you generally need more and better measurement applications and equipment. Chris
  9. I'd have to look at that .mdat measurement file and your FusionAmp project file to understand what is occurring. It looks like there is a doubled output of the midrange and tweeter above 3 kHz. PM me and I'll send you my email address. Chris
  10. One of the ideas that has been documented has been the use of ESS AMT-1s on top of Cornwalls, La Scalas, and Belles (like my surround loudspeakers have presently). This idea also looks like it might be useful to those running older Heresies and wanting to hear a more spacious soundstage in depth. A couple of AMT-1s plus bi-amping with a DSP crossover (e.g.,a miniDSP 2x4 HD)---just like the AMT-1/Heritage "kit" thread would likely be an interesting and low cost experiment. I'd recommend an 1800 Hz crossover point. A bit noisy horizontal polar spectrogram of a single AMT-1 in-room: Chris
  11. One of the areas that I was most interested in is the horizontal and vertical polar coverage responses of the Heresy midrange horn (nominally 700-5000 Hz, below). First the horizontal polar sonogram (normalized to on-axis SPL): Pretty good -- much better than I would have expected (it has straight-sided horn walls with a truncated tractrix mouth roll out): This is different than the exponential Heresy horns that I own (1981 models). There is a disturbance at the midrange-tweeter interference band (about 4-8 kHz) also. Now let's look at the vertical polar sonogram: A bit less nice than the horizontal. This is a consequence of its small vertical mouth dimension of the midrange horn (which is dictated by the small size of the Heresy itself using separate horns for each of the tweeter and midrange--more on that later), which allows the vertical polars to start to spill out on the ceiling and floor starting just above 2 kHz, and continuing down the the crossover frequency with the woofer (850 Hz)--down to about and octave below that point (425 Hz), with an interesting void in the downward polars between 800-1000 Hz. This says to me that the Heresy IV needs to be sitting on a fairly good pad of absorption material in-room (probably about a 2 feet x 2 feet across in size) to moderate that downward lobe of vertical polars to more closely match the other direct energy coming from the midrange driver/horn. In addition, there is a corresponding loss of vertical polar control in toward the ceiling. It it were me, I'd probably find some absorption material to lay on top of the loudspeaker box and stick out in front of the front to absorb some of that energy headed toward the ceiling in the 1.2-2.4 kHz band. There is also a spike of acoustic energy at about 6.6-6.8 kHz that a pad of absorption material that the loudspeaker is sitting on could absorb , plus a little absorption sticking out on top would help to moderate. These are cheap and easy in-room acoustic treatments that would likely be very effective. I'd recommend using some material and listening for the differences, and move it about and/or add/subtract some the absorption pad area until is sound most neutral and natural with female vocals and the upper registers of piano, etc. I think I'll apply those two ideas to the Heresy I's in my garage. I have a stock of Auralex Sonofiber 2'x2'x1" squares that I can easily use for that duty. Chris
  12. One metre. The only time you might take a measurement at the listening position is when measuring well below ~100 Hz (subwoofers mainly) but not above that frequency. The issue in that case is being in the extreme near field of the loudspeaker due to the very long wavelengths at those low frequencies will affect the measurements. In general, I take measurements almost always at one metre. Chris
  13. With the Hypex FusionAmps, I would first dial-in the crossover filter slopes and the delays, then start to work on flattening the SPL response. When you run a measurement sweep with each individual channel (all the PEQs reset to zero_, then you can use REW to find the suggested PEQs to add to flatten the response. You can use the thread that I linked on the first page of this thread for that portion). With the Hypex FusionAmps, they apparently have no input channel PEQs, so you'll have to run individual sweeps of the woofer, midrange, and tweeter and produce a nominal -3 dB of each driver just at the center crossover frequencies (nominally 400 Hz and 4.9 kHz, or whatever you actually chose). Once you get the individual drivers/horns flattened and carefully attenuated to about -3 dB at the crossover frequencies, then run a sweep with all three drivers playing, then you can set the individual channel gains to bring all three driver channels to the same SPL level, and set any remaining PEQs to further flatten the response. With the exception of attenuating room modes, it is possible to achieve ±1.5 to ±2 dB flatness (using REW's psychoacoustic smoothing) from about 70 Hz to 16-18 kHz. Once that is done, you've got the 99% solution in hand. If you've flattened the SPL response after setting the delays and crossover filters (the SPL response will change), then it's time to listen. If you want any further fine tuning, you can PM me for my email address to send the measurement files (*.mdat). I think you'll find that they sound a lot different now--and better--with the time alignment much closer and the SPL response flat on-axis. You should have a very large soundstage and balanced timbre for your best quality recordings. Chris
  14. V5.20 Beta 61 presently. The beta version of REW really isn't what I'd really call beta--its more mature than that, and many of the better features reside in that version, leading the "released" version by a year or more.
×
×
  • Create New...