Q&A # 192 - July 27, 2000

Staff

Q Regarding bi-amping in your Q&A # 42, you say: "Active is better than passive because, with passive, the crossover networks absorb the energy from the power amplifier that is not being sent to the respective driver. In other words, with the woofer, the inductor(s) in the crossover network absorb high frequency information from the power amplifier, and convert it to heat. With the tweeter, the capacitor(s) in the crossover network absorb low frequency information and convert it to heat. Putting the crossover before the power amplifier allows only high or low frequency information to be sent to the respective power amplifiers, and this is much more efficient. In this case, the crossover networks in the speaker enclosures have to be bypassed." I have Paradigm Studio/100s and Rotel 991 (200 w/ch) and Rotel 1090 (380 w/ch). If I wanted to horizontally bi-amp my speakers using an active crossover to have the 991 driving the mid/hi terminals of the speakers and the 1090 driving the low terminals, how do I "bypass" the crossover networks in the speakers? Also, are these Rotel amplifiers too powerful for bi-amping the Studio/100s - do I risk blowing my speakers?

A Any true losses in the inductor are because of the resistive component of the coil. As the inductor, or capacitor for that matter, becomes most effective in rejecting the frequencies, it not only drops voltage, but generates a phase shift between the current and that voltage, so that while it seems like they're consuming power (apparent power) on a voltage X current sense, real power, calculated by voltage x current x cosine(phase shift) is minimal, as the phase shift reaches 90 degrees. Where most of the power losses occur is within resistor networks used to flatten the frequency response of the drivers and match output between, say, tweeters and woofers. Usually the woofer is the least efficient driver (due to the mass needed to push down the resonant frequency and get deeper bass), and so, in simple terms, the well-tailored passive speaker is only as efficient as its least efficient frequency range of its least efficient driver. It doesn't mean that a passive speaker cannot equal or surpass the performance of an active one, only that everything else being equal (drivers and enclosure), the passive speaker will require more power.

I don't know the specific design features of the Studio 100s, but taking a passive speaker whose crossover has been specifically designed for the drivers in question with a specific enclosure/baffle arrangement, and implementing a generic active crossover, is not a good idea unless you're looking into sound reinforcement applications. Chances are, if the manufacturer did a good job designing the speaker in the first place, that while you might slightly improve the dynamic range (or maybe not), the overall frequency response, blend between drivers, and perhaps dispersion characteristics, would be altered. If you really want to go active, I'd suggest either starting from scratch, or let someone else do the work and buy a loudspeaker designed with active drive in mind. Active speakers are available on the consumer market, though it seems that their popularity has been limited by the high-end mentality that bigger and more powerful are sexier than elegantly and intelligently built to task. If only because of necessity, I'm as guilty as the rest. I really like having a collection of amplifiers that weigh more than me. But, practically speaking, if I didn't have to keep them around for the sake of driving passive speakers provided for review, or as a reference for another amplifier's capabilities, I'd probably go with a set of active speakers if only for the sake of giving myself more space, even if the performance were the same. That's not to say that there aren't active systems which allow outboard amplifiers for the person inclined to keep a stack of boxes. The B&W Nautilus (the original) was a 4-way active design that utilized entirely outboard amplifiers. Meridian's digital speakers use active designs. And, if you're taking a DIY approach, an outboard amplifier would be easier just from a construction standpoint.

Q Ok, so I like to tinker and make things more complicated than they need to be, but I am sure you can relate. I am having a problem driving multiple stereo systems from my main system in the living room. Using Y adapters at the Tape Monitor outputs on my Equalizer, I am driving a small FM transmitter (the legal kind) and the upstairs stereo. I have noticed that when I turn on the stereo upstairs, the sound from the main system in the living room becomes distorted. I am assuming that my equalizer does not have a separate driver amp for the Tape Monitor outputs and "cheated" by doing the equivalent of my Y adapter inside. This seems to result in a situation where the preamp's output is overloaded. So is there such a thing as an distribution preamp? One input with separately amplified outputs (even though there is no net gain in signal level). Kind of a mixer in reverse.

A Try the Xantech AV-61. It has one composite video input, two line level audio inputs, and six sets of identical outputs, all buffered. Gain is unity.

Q I own a middle of the line 5.1 system: A Sony STRDE925 digital receiver with Cambridge Soundworks Tower II fronts. I am fairly happy with the way everything sounds, but the receiver's EQ seems to be set very conservatively (for example, the bass crossover was set at 120 Hz). I want to get the most out of my speakers using the EQ. Cambridge Soundworks hasn't published the specs on the speakers (all I can find is a review of the speakers with some measurements (see  http://www.cambridgesoundworks.com/reviews/twr2rew.html), but I can't make sense of the measurements they made.

A The article's relevant content to this matter is the following: "Our MLS quasi-anechoic frequency-response measurements at distances of 1, 2, and 3 meters were very similar: Except for a sharp dip of about 8 dB at 3 kHz, response between 300 Hz and 20 kHz varied only +3 dB. The manufacturer provided no information on the system's crossover, but the 3-kHz notch appeared to be a crossover artifact between the tweeter and midrange driver and, as is often the case with such effects, was not audible with music."

Now, assuming that you want to do speaker correction with the EQ, as opposed to room correction, (which is more like compensation since you're altering the output at the speaker to compensate for the room), what you're looking to address is the 8 dB dip at 3 kHz. The "artifact" they're talking about between the tweeter and mid-range is probably either a partial cancellation due to phase differences between the drivers at their crossover frequency, or because, either by manufacturing tolerances or by design, the crossover point of the tweeter is higher than the mid-range. If you had access to a quasi-anechoic measuring device, or an anechoic environment, you could verify that the same problem was evident in your set of speakers (which it may or may not be if the problem were due to poor manufacturing tolerances on that particular review set), and then determine the cause of the problem by measuring the output of the speaker under different circumstances. If you simply inverted the polarity of the tweeter and the dip smoothed out, it could be a classic example of a 2nd order crossover causing a notch due to phase cancellation. If the response did not change, or got worse, then if it wasn't due to coincidental differences in loudspeaker manufacturing from the model, it could even be an intentional design feature. In itself, putting an 8 dB dip at 3 kHz would seem to be a pretty inane thing to do, working off of the assumption that you wanted an accurate loudspeaker. However, psychoacoustically, a sharp dip is much harder to detect than a peak, or even a broad dip. A notch at 3 kHz will also decrease the amount of sibilance perceived on many modern recordings that have vocalist record inches away from the microphone, which may make the loudspeaker sound smoother and the vocalist deeper into the sound stage.

So, assuming that the problem does indeed exist with your loudspeakers, and you want to correct it, doing so with EQ becomes a very deliberate exercise, because even if the article did describe the width of the notch (i.e., the curve which would need to be compensated), that particular curve might not apply to your specific speaker. If you could find a method of measuring the anechoic response of your speaker, you will need to set your EQ to be the inverse of the response deviation. That means not only the frequency and amplitude, but the width of the needed boost, technically described as the Q. You could use a test disc or a tone generator to do a sweep, but without an anechoic environment, getting an accurate result might be difficult. If the receiver has no way to adjust the Q, then you might be better off leaving it alone, as any correction has the possibility of simply adding another error.

If the boost was misplaced in frequency, the resulting peak would probably be intolerable to most listeners. If the boost were too narrow, you'd have two smaller dips instead of one. If the boost were too wide, the notch would be replaced by a notch in a large lump of the response, which could sound very much like an alarm clock radio. If you have the equipment and time to really look into this, by all means go for it. Otherwise, my suggestion is to not worry about it too much. (After all, you bought the speakers because of how they sounded, right?) If you want to try to fiddle with it by ear, it won't cost you anything, and you can always undo it. As far as using a test disc and a Radio Shack meter to compensate for the response in your room, I wouldn't do it for two reasons. 1. If the room is the problem, changing the output of the speakers does not change the room. It only results in screwed up output which happens to sum nicely with a screwed up room on paper. Ears don't work quite that way, and you'll be able to tell that something's amiss, e.g., that a blanket has been tossed over the sound. 2. The Radio Shack meter's closest approximation to flat is C-weighting, so that if you used it to balance a system, even in an anechoic environment, the end product would be a lifted treble and bass.

A few suggestions about EQ in general. Become familiar with how each band affects the tonality of what you hear, and how that affects the rest of the spectrum. What seems better at one band, as you've discovered, can upset the total balance of the system. Once you begin to understand the correlation of the bands, you can apply that knowledge to tailoring the response to your personal taste or particular recordings. Use EQ sparingly. Boosts of 10-12 dB are excessive, and due to the logarithmic nature of the dB scale and our hearing, they require far more power when energy is within the band of the boost. Generally speaking, keep boosts below 6 dB for recreational use, although cuts allow more liberal use of the sliders. If you bought your speakers because you liked the way that they sounded, know that EQ will change the way they sound, and although you might like any given iteration more, it's also just as likely that you won't.

Q I am a novice who just started looking for a good receiver to build a decent home-theater system. Fortunately(?), one of my friends left large speakers and a subwoofer in moving out of country, because they were too big to take with him. I paid $400 for the entire set. The model numbers are Cerwin-Vega AT12 and Infinity Subwoofer SSW-212.

My question is whether these are good enough to deserve my investment, or should I go for something more modern in design? Also, what kind of receiver would work with this stuff?
A As long as the speakers are in good condition, and you like their sound quality, you are in good shape. Cerwin-Vega speakers tend to be sensitive (92 dB), so just about any receiver will work with them. Start out with something like the Yamaha RX-V595 for the receiver.

Q I have A Yamaha RX-V2090 5.1-ready receiver, Yamaha DDP-1 DD processor, Millennium 2 4 6 DTS preamp processor, and Sunfire Cinema Grand power amplifier. I am looking to upgrade my system, but I am unsure if the upgrade will help me achieve better sound or just help me reduce the amount of components I presently employ, or both. I want to get the Sunfire Theater Grand preamp processor which is equipped with DD and DTS decoding. Will this upgrade Improve the sound? Is it worth the money in your opinion?

A Outboard decoders did a very good job, pretty much the same as the ones that are now built-into all the new processors and receivers. However, all those cables (at least five for each decoder) were a real hassle and significantly raised the likelihood of ground loop and other noise problems. The decoding circuitry has improved over the past few years too, so the Sunfire, as well as other new processors, will give you a better sound. High-performance processors, such as the Sunfire, tend to have better power supplies than the mass market products, but less features, such as DSP sound fields. So, you should choose your high-performance processor based on its features, because they all are really pretty good.

Q I have a Sunfire True Subwoofwr MK II. It is a Fabulous Subwoofer. I purchased it partly on your recommendation, Thanks! This might be common knowledge, but the only way I could get the sub to stop a bad hum was to put a two-pronged plug adapter between the power cord and the wall to open the ground. Other than this lifting of the ground, everything else is normal and the sub sounds wonderful. I had previously read about this problem and I thought I would be immune to this problem because none of my other equipment uses the ground (the power cords do not have a ground connection). My equipment: Denon AVR-4800 (2 wire power cable), Fisher DVD (2 wire power cable), Sony TV (2 wire power cable), Toshiba VCR (2 wire power cable), Retocon wireless headphones (2 wire power cable), General Instrument Digital Cable Box (2 wire power cable). My concern is that the ground is there for a reason, and I have defeated it. Could I run a separate ground wire to each piece of equipment and tie all of the equipment to the ground?

A The first version of the Sunfire used a two-pronged plug, and the MK II has a three-pronged plug with the ground. I have seen the same thing occur with successive versions of other subwoofer brands too. I think it has to do with the fact that the newer subwoofers, with their high power amplifiers, are using high voltages, so the ground connection is a safety measure. We take a chance by removing that ground connection, but lots of us have done it from time to time. Connecting the chassis of your other components to ground can help. The hum could be induced by any of your components, but it shows up in the subwoofer because it specializes in low frequencies, such as a 60 Hz hum.

Q I have a Sony SDP EP9 ES Processor that I am using currently as my pre/pro for both audio (2 channel) and Dolby Digital. I use a Velodyne HGS-10 subwoofer (has a defeatable crossover), connected to the Sony's subwoofer out . On the bass management modes of the Sony, I set the front speakers for small and set the hi-low pass at 80 Hz in the sub-menu. I hope that in this setup, all frequencies below 80 Hz along with the LFE are passed onto the subwoofer in Dolby Digital mode.  However, in using only the 2 channel mode, does the setup above still send only 80 Hz to the sub, and the rest to the main speakers? I usually defeat the crossover in the Velodyne to avoid double roll-offs. However, I fear that in 2 channel mode, the Sony could be sending a full range signal to the subs and cause lousy response and possibly damage in the future.

A Because bass management is outside of the essential DD decoding, I think the bass settings probably continue into two-channel operation in your receiver. However, the way to test it is to turn the subwoofer volume control down while listening to two-channel stereo. If it still has an 80 Hz high-pass into the main speakers, it will be very obvious in having a lack of bass.

Q Hi ! Thank you for the best Home Theater site in the world! I am in the process of building a home theater. My question is about acoustic treatment behind the main speakers (Acoustic Research HO15). After a few weeks of searching, I found that acoustic foam is too expensive compared to ordinary 6-8 inch thick fiberglass insulation. Does it make any difference in sound control?

A Fiberglass insulation is excellent for acoustic damping. The problem is that it is glass and easy to get small fibers stuck in your fingers. For economy, but still having nice damping, I would suggest getting an eggshell mattress pad. They are sold at bedding stores for about $50 in the king size. Of course, it is not the most attractive thing to hang on a wall, but it works. I have one on the wall behind my main speakers in our audio lab.

� Copyright 2000 Secrets of Home Theater & High Fidelity
Return to Q&A Index.