Edgar

Definitely looks interesting. Note that Outwater recommends laminating two sheets with the cuts on the inside. That would be a whole lot more stable than a single sheet. Greg

Thanks. How do you insert an URL so that it acts like an URL and not like text? Greg

Has anybody ever tried this?: http://www.outwatercatalogs.com/2006_Master/lg_display.cfm?page_number=322

If you purchase any Altec multicell horn, make certain that you also get the 30210 throats with them.Having said that, I have a pair of Altec 1003b horns (with throats) in fair shape. They were pulled from active duty after at least a couple of decades, and are weary but serviceable. I also have a pair of Altec 290 drivers in unknown (meaning "pretty rough") condition. Any or all are free to a good home, but you have to pick them up in Colchester, Vermont (just north of Burlington). I'll help you load them into the back of your pickup truck, but that's the limit of what I'll do to transport them. Send a private message if interested. Greg

With those T/S parameters, using Keele's equations, I come up with the following for the TAD 1601a: Vb (reactance annulment chamber volume) = 3.18 cubic feet St (throat area) = 74.2 square inches fHM (upper mass rolloff frequency) = 181 Hz By comparison, I have some Eminence numbers for the K33 that show: Fs = 34.46 Hz Vas = 10.65 cubic feet Qts = 0.39 Vb = 5.83 cubic feet St = 114.1 square inches fHM = 181 Hz This shows the 1601 to actually be a better match to the KHorn than the K33. (If anyone has more up-to-date K33 T/S numbers, please post them and I'll run them through the equations.) -- Greg

... and a license should be required to wear spandex!

Mostly it's a noise issue. Balanced cables with unbalanced impedances do not reject common mode noise as well as if the impedances are balanced. So if you have really long cables in a noisy environment (lots of dimmer switches, commutator motors, fluorescent lamps, etc.), you'll pick up noise. But it will still be better than using a plain unbalanced RCA cable. See post 762412: http://forums.klipsch.com/forums/permalink/762412/762412/ShowThread.aspx#762412 It's a lot more important when going from unbalanced outputs to balanced inputs than when going from balanced outputs to unbalanced inputs. Greg

Lots of good info in the old Bruce Edgar interview, in Positive Feedback. http://www.positive-feedback.com/Issue4/edgarinterview.htm Greg

It's just a term that I use to describe it. If you have low-order (shallow slope) crossovers, and the relative phases of the LF and HF drivers vary with frequency, then the angle at which the two sources add in-phase also varies with frequency. If you think of that as a lobe of higher amplitude response, then it moves around as the frequency is changed, like a searchlight. Greg

Um, I'm not an acoustics guy. Signal Processing is my bag. Greg

Essentially, yes. If the delay is constant you get a comb filter. If the delay varies with frequency, you get "searchlight" lobing. Yes, that is one way to deal with it. The other is to use broadband transducers and crossover topologies that are in-phase at all frequencies, like Linkwitz-Riley or Bessel-Derived Matched-Delay Subtractive. You can approximate it with a string of allpass filters. Greg

An example is best. Let's say that the low frequency and high frequency driver responses are 90 degrees apart, and that the drivers are physically separated by some distance -- the HF driver is physically above the LF driver. If the listener is on-axis, the two responses may sum to 1.0. But at some vertical angle above or below the axis (usually below), the phase difference between the drivers is exactly cancelled by the distance between the drivers, so they now add in-phase. At some angle in the opposite direction they add out-of-phase, again because the physicsl distance corresponds exactly to the phase difference. In a living room, the listener is almost always within a few degrees of the vertical axis, so this effect is only noticed in reflections off the floor or ceiling. In a concert hall, though, that axis of "in-phase", and its corresponding axis of "out-of-phase", shine like searchlights over the audience, moving around because phase varies with frequency. Only if both amplitudes are equal, and both are 1/2 the passband amplitude. Oh, I think I understand what you're trying to do. If your amplitudes are correct, but your relative phases are not, then yes, you could conceivably use delay to move the relative phases to 0 or 180 degrees at the crossover frequency. This would eliminate a peak or notch in the overall response at the crossover frequency. But it would do little to smooth the overall phase (or delay, if you prefer) response of the system across all frequencies. Greg

It depends upon the application. In a home situation, where the listener is in the near field of the loudspeaker, a 90 degree phase difference at crossover probably doesn't hurt much because the resultant main lobe of the response gets sent into the floor or ceiling. In a sound reinforcement application, though, that main lobe gets pointed at a section of the audience, and is very audible to them. Essentially true, because the response of the non-dominant driver falls so steeply that it quickly becomes negligible. Correct. I don't completely understand what you are proposing. Determining the response at the crossover frequency (or anywhere else, for that matter) is pretty simple -- express the magnitude and phase of the drivers as complex numbers, and add them together. The magnitude of the sum is the magnitude of the combined response; the phase of the sum is the phase of the combined response. If each of your drivers has an amplitude of .7071 at crossover, and they are 90 degrees apart in phase, then their sum will have a magnitude of 1.0. If each of your drivers has an amplitude of 0.5 at crossover, and they are in-phase, then their sum will have a magnitude of 1.0. If they are 180 degrees out of phase, then you'll have to invert polarity on one of them. This is done in some Linkwitz-Riley crossovers, for example. Greg

To the extent that anything I did influenced the parts that you like, thank you. Greg

I was just one of many who contributed to a long line of digital processors in the Altec Lansing, ElectroVoice, Merlin, Klark-Teknik, and Dynacord lines. The DSP firmware in the Merlin was my personal responsibility, but I borrowed some from my predecessors, and others borrowed from me for later products. After I left, I think that all DSP development went to the Dynacord group in Germany, but they had all of my code and, glancing at some of the features of the Dx38 I wouldn't be surprised if they're still using some of it. Many of the built-in configurations of the Dx38 are dead-ringers for what I did in the Merlin. Greg

The 90 degree relative phase shift is a characteristic of Butterworth crossovers. If a properly-implemented Linkwitz-Riley crossover is used, then the lowpass and highpass sections are in-phase at all frequencies, and each has half-amplitude (-6 dB) at the crossover frequency. In such a situation, perfect time alignment leads to perfect sum-to-allpass frequency response in the crossover, and any remaining anomalies are due to the drivers themselves. I designed the crossovers in the Dx38's great-granddaddy, the Merlin ISP-100, so I know that EV knows how to implement Linkwitz-Riley crossovers properly. I no longer have any affiliation with EV, so I can only hope that they continued the tradition in the Dx38. Group delay is only an issue in a modulated system, which a loudspeaker is not (or should not be). Greg

It doesn't say so in the manual, but I suspect that the 1.927 mSec is the inherent delay through the unit. This is the amount of delay through the A/D converters, processing paths, and D/A converters, and cannot be reduced. I'm not certain exactly how much delay is necessary before lipsync problems become intolerable, but I understand the general rule of thumb to be that it is on the order of 30 mSec. And it's much less noticeable when the audio is delayed relative to the video than it is the other way around. It seems our brains are accustomed to hearing sounds after seeing the events that cause them, because the speed of sound is so slow compared to the speed of light. Greg

I'm assuming that this is a typo, and really should be 96 kHz. Otherwise it's truly absurd because it guarantees that every signal will have to pass through at least one format conversion. If you're referring to Class-D amplifiers, like those from Bel Canto and Tripath, then no. They actually have an analog stage feeding the modulator. The closest thing to a truly all-digital system that I've experienced was here: http://www.audiomn.org/Pages/feb02/HornShow.html The only analog stage in the whole signal chain was the modulator for the Class-D amp. The system sounded REALLY good. It depends. There are some formats that can be converted losslessly (upsampling in integer ratios can be, but usually isn't), others that cannot (DSD to PCM, or vice-versa, for example). Well, for example, signal processing in the DSD native format is difficult, sometimes impossible. Conversion to a form of PCM is almost necessary if you want digital crossovers. I agree with you, though, that the format conversions should be kept to an absolute minimum. Greg

Yesterday at 1:00 PM I saw 104° F (40° C) actual temperature, here in Naperville. Greg

Looking at the manual (http://www.electrovoice.com/download_document.php?doc=67), it appears that there are 5 bands of EQ per input channel before the crossovers, and several configurable bands per output channel after the crossover. Yes. "Each channel employs a digital level control and a polarity switch." There is also delay compensation available. I did all of my ISP-100 settings by ear. It just takes time. Greg

Other than the obvious (the fact that there is an unbalanced connection, with all of its inherent disadvantages), really the only performance issue that I can think of is the fact that this configuration can pass DC, which would be blocked by a transformer or capacitor. It is sometimes difficult to determine the true input impedance of the destination, and even more difficult to determine the true output impedance of the source, but I have found that getting "close enough" works pretty well. If you're WAY off, then it becomes more susceptible to noise and hum pickup. I think that it's actually 6dB, but it's definitely there. I have not found it to be a problem; there is more than enough gain in my amps to drive them to clipping despite the loss in signal level. Greg

There's a quick and simple way to do it using just resistors, that works fine as long as "chassis ground" and "signal ground" are the same on your equipment. It's commonly called a "impedanced balanced" configuration; I've been searching the Web all morning for a diagram but haven't found anything. Basically you always balance the lines, and you don't worry about whether the signals are balanced. ("Balancing" actually refers to impedances, not signals.) So when you connect an unbalanced output to a balanced input, you connect source "hot" and "ground" directly to destination "+hot" and "common", respectively. You connect destination "-hot" to source "ground" THROUGH A RESISTOR EQUAL TO THE OUTPUT IMPEDANCE OF THE SOURCE, with the resistor located on the source side. Going the other way, from a balanced source to an unbalanced destination, you connect source "+hot" and "common" to destination "hot" and "ground", respectively, but you connect source "-hot" to destination "ground" THROUGH A RESISTOR EQUAL TO THE INPUT IMPEDANCE OF THE DESTINATION, with the resistor located on the destination side. It's a lot easier to do than it is to write about. I use this with my Merlin ISP-100 (kind of a grandaddy to the DX38). Greg

PWK's enthusiasm for rubber throats notwithstanding, I still think that constraining the first three folded sections to an exponential or tractrix contour will improve the response below 100 Hz. The increase in distortion should be negligible at the SPLs achieved in the home. Greg

Example of improper installation. Should either be pointed back for thrust or up for traction.

ASIO is one of many possibilities; simply the one with which I am most familiar. Unfortunately ASIO is not as device-independent as it probably could be. There is a moderately complex section of code that must be customized for each sound card; the main portion containing the signal processing can then be generic. Perhaps someone who has more Windows or Mac OS savvy can recommend some other candidates, as well. The good news is that even complicated crossovers and EQ do not significantly load a modern processor. Been wondering what to do with that old Pentium III machine that's gathering dust? It's more than powerful enough for this. Greg