WMcD

I'm part of the "zip cord is just fine" club. Also, your local home improvement center is definitely a good source. Zip cord, or lamp cord, with white or black insulation is often offered on spools. A good buy. I do laugh at some of the "specialized" zip cord they sell which have labels like "high definition". There is no merit to the claim. IMHO, 16 gauge is plenty. Please note that as the numbers in the "American Wire Gauge" system go down, the diameter goes up. I do think that 14 or 12 is not necessary. Before you decide on a given gauge, please consider the interconnects, e.g., banana plugs or spade lugs, you may be using. Or, your speakers or amps may have some sort of spring clips. Using large wire like 14 gauge, may make it difficult to hook it up to the spring clips or interconnects. One reason to go to Radio Shack for wire is if you need assistance with the interconnects and hooking up the wire to them. A salesman there may be more helpful than the home improvement center. Maybe. Gil

This was my experience with HT receivers generally. The noise is coming from the digital processing, IMHO. It is not an interconnect or cable problem. Gil

This is a very long run for a high impedance connection. My guess is that even if you solve the noise problem, you're going to have a problem with decrease in high frequencies. If you absolutely want to make a run like this, you might try using tranformers on either end. These transform the high impedance line level output of the preamp to 600 ohms for the long wire run, and then back up to high impedance on the amp end. It seems to me that the XLR microphone transformers sold at Radio Shack will work for this. On the other hand, this is going to take some soldering. I've not use this myself, but it would be the place I'd start. If you need more info, I'll be pleased to answer any questions. Regards, Gil

I'm not absolutely knowledgeable of the hardware involved but I suspect the input to your rig is similar to the miniplug on a pair of Walkman headphones. You should be able to go to the local Radio Shack. A salesman should be able to set you up with an adaptor or two (it may take two)which will work. You probably wind up with a female miniplug to male RCA. Incidentally, this, the "Pro" section of the bbs is for theater type stuff. You should re-post on the "Pro Media" section. You'll probably get better responses. Gil

My guess is that a portable surface, like a 4 x 8 of plywood is not going to do very much good in this situation. First, it is not an appreciable fraction of a wavelength in the bass region, so it is not going to effect things. Second, the pressure wave is just going to run around behind it. That is a non-technical explanation. On the other hand, is there really no corner in the general area. Front, behind, to one side, a bit distant, might not make a big difference. If not, a wall floor intersection might not be so bad. Please do experiment. Gil

To expand your experience, let me suggest "Symphonic Star Trek" on Telarc. You've heard some of the pieces on little TV speakers, it's time to hear it on the Klipsch products. Some neat sound effects too. Gil

I have an old JVC amp with a similar problem. The cure, for me, is to work the input selector switch. Next time the problem develops, please move the input, tape monitor, and any other selector switches. You might as well plug and unplug the RCA connectors feeding the thing, and check the speaker wires for tightness. Power down first when doing this. You should also check the connections to the speakers. Excuse me if this is overly simplistic. But sometimes these things are a problem, particularly in older equipment. Hey, doesn't cost anything. Gil

I'm using 1828 drivers. They tested better than the 1824. Gil

My first observation is the wiring diagram of the cross overs I've seen show no reversal of polarity for the K-Horn. It is intersting to note that the path length from the bass driver to the midrange driver in the K-Horn is about one wavelength at the 400 Hz crossover point (both these are around 33 inches). This is to say, there is about a full wavelength of "delay" and no alteration in phase. On the other hand, in the LaScala, the path is about half a wavelengh. So perhaps the LaScala polarity should be reversed to account for the phase induced by the delay. Naturally, this doesn't take into account the phase change caused by the drivers and cross overs. Given the number of factors involved I wouldn't want to hazard a guess unit unless I measured things with a dual trace scope. Before I get jumped on here, I recognize the terms of polarity, phase, and delay, must be used with great precision. Regards, Gil

I think Stig has the answer. It is more a question of why a cube is not used elsewhere. In other direct radiator speakers, i.e. where the bass cabinet and driver are used up to, say 600 to 2000 Hz, we want to make sure that the internal dimensions of the box H, W, D are different. To some extent, the inside of the box will form a resonant tube (closed on each end) with a length equal to the H, W, and D. (This is a simplied explanation.) The worst case would be if all three dimensions are equal to effectively form three "tubes" of identical length. With a sub though, there is little or no high frequency power being provided by the driver. Therefore, the relative dimensions don't make a difference, as long as they're a small fraction of a wavelength. Even if there is a potential for resonance, it is not excited. As pointed out above, the wavelengths fed to sub are just too long. There is something to be said for avoiding long panels in the sub. The cube shape avoids this. Of course, if you are designing a ported sub, you may wind us using a long pipe. So then you need a longer container in which to fit it, or will have to fold the pipe. That is my read. Gil

Rats, I don't have the book here. The "bible" on the issue is Tomlison Holman's "5.1 Sound, Up and Running", mentioned in previous posts by me. Very worth while the effort to study. And it does take an effort. Getting back to basics of speakers. In a 5.1 set up of speakers, we have L C R (making 3). Then the two surround speakers are 4 and 5. If you use a subwoofer, that is the 0.1. In the classic Dolby Surround or Pro Logic, the surround 4 and 5 just get the same mono L-R signal from the two channel source. This is the L-R delayed by 10 milliseconds. In that conception, the subwoofer just gets sent the bass from the two channels of input. There is no separate channel from the source for bass. However, in later days, we have discrete input sources to our amps (as distinguished from speaker outputs). So by some schemes there are discrete L, C, R, up front. Also, L and R surround. Then there is yet another low frequency effect source, which gets sent to the subwoofer speaker. The sub can be fed with two sources. One is direct from the LFE source. The other is front or surround information which we choose to sent to it. In the 6.1 set up, to my understanding, they add a derived center channel to augment the rear L and R, just like the original Dolby system, or Klipcsh system, derived a center for the front. I'll have to post a diagram. Gil

Here are two pictures. Dr. K holds forth. Ms. Valerie (The Celebrated Mrs. K.) welcomes us. Regards, Gil

Let me clarify a point. It is my understanding that "5.1" actually refers to a speaker set up. Of course you need the amp to go with it. Source material and decoding is a separate issue. As I was pointing out, Dolby and Dolby ProLogic decode from a two channel system of storage or transmission, such as VHS soundtracks. The major technique is L+R for the center and delayed L-R for the surrounds. There's no reason why this couldn't use, tv stereo, vinyl, cassette, or even 8-track, for the two channels. Other systems like Dolby Digital and DTS use true multiple channels. Gil

There is not much I can add to what has been said above. Our many hosts and hostess rolled out the red carpet, Arkansas style. We were careful to not pilfer any items and strip searches were only threated. There was also discussion of Ross V.'s avatar. On close inspection, it does indeed appear to be a cactus. Gil

The original Dolby surround used two channel in the transmission and/or storage system. A regular old Dolby or Dolby Pro-logic equipped HT receiver should be able to "decode" such transmissions or tapes. Please note that if you play a VHS tape which is marked "Dolby", you are really only getting two channels of output from the VHS player. Then the receiver does the rest to produce the center and surround. The 0.1 is just a subwoofer at work. If the station transmits a given movie in stereo, chances are you receive the same sort of audio on the VHS sound track. Therefore, an HT receiver set to Dolby Surround or Dolby Pro-logic will decode it. Regards, Gil

The HT receivers I've worked with all allow 5.1 output with a music CD feed. They do not cut out for lack of a video input. These HT receivers usually have "hall" settings which you can fool with. As Boa points out, there can be some un natural results when they add reverb. Some sound better than others. The less reverb the better, in my experience. If you have questions, you should bring a music CD over to a dealer and try playing on an HT system. I think you'll like it. If you can borrow some speakers, you should try it at home. Do go through the set up tests to balance the outputs of the speakers. Purists say, correctly, it is best if the speakers are factory matched. But you can have a lot of fun without perfection. In my experience, with wide spaced flanking speakers, the center channel is a very nice fill and well worth it. Pop singers place in the center channel. Classical benefits too. You read a lot about "sound stage" with the right amp. But the center speaker really makes it right. The surrounds help with ordinary classical even which is not mixed in Dolby. Some newer recordings are Dolby 5.1 mixed. Remember, the original Dolby system is just a way of getting the most out of a two track recording. The surrounds give hall ambiance to classical. Depending on the recording, you do have to fool with the settings. I wouldn't say that all pop recordings do well with surround. Some do. I think you should try it. I predict you'll like the opportunity to experiment. Gil

I don't know if Klipsch offers touch up stains. I've used some walnut type touch up by Minwax on other speakers. It is like a magic marker. Again, I don't know if Minwax makes it in black. If I was in your sitution I'd touch any scratches in on black stained wood with a black Sharpie. My technique would be to "paint" over the scratch, and then quickly wipe off excess, i.e., the Sharpie dye on the good surface, with a paper towel. It is a bit surprizing to hear the drivers or cross over were damaged by the mechanical mishap. I'd think the most likely thing to fail if there was arcing is the output stages of the amp. It might be worth your while to check. Regards, Gil

I agree with both points. It depends on the room available for identical speakers and you budget. I have some homemade horns with identical squakers and tweeters in the left, center and right. When special effects sound is panned left to center to right, the match is very good. I can't say that non identical speakers would do as well. In fact, the design goal these days is to have identical treble drivers at all five points. This aids the reproduction in pans from rear to front. On the other hand, my observation is that non identical speakers could work well listening to plain old music where there are no "special effects" pans. One last thought is that you might wind up putting whatever center speaker near the TV. This is going to bring up issues of magnetic shielding. Klipsch has some shilding plate available for the Belle, as I recall. You might have to adopt that to your ultimate choice. Gil

Have you run the calibration program on the HT receiver? It circulates band limited noise around the five speakers; each speaker sounds off individually and there is a software setting to get them equal in output. A Radio Shack sound meter is the measurement tool of choice, but ears work well too. It could be that the rears are less sensitive and require more gain. Please excuse if you have already done this. Gil

The short story is that impedance is the "load" presented to the amplifier by the speaker. It is technically voltage divided by current. This is the definition of an ohm, and impedance is measured in ohms. However, speakers don't have a constant impedance. You'll see graphs of how it varies with frequency. However, suppose your speaker presents a 4 ohm load at some frequency. If you crank up the output of the amplifier so it delivers 4 volts, the current is going to be 1 amperes. At 8 volts, it is two amperes. Now, if the amplifier is a perfect "voltage source", it would be able to deliver the required current and voltage under all conditions. But amplifiers are not really perfect. They may run out of voltage supply when driven hard. This is clipping. But they are also limited in the amount of current they can provide. The latter is why you'll see ratings of amps like 100 watts max at 8 ohms, and 150 watts max at 4 ohms. Note, you'd expect the power to double to 200 watts at 4 ohms. The result is that amps said to not "like" low impedances. Indeed, putting speakers in parallel reduces the impedance and you can run into problems. On the other hand, unless you are driving the amp to high output, a 4 ohm load should not be a big problem. It is diffiuclt to generalize further. Others on the bbs might add their comments. Gil

I was thinking. This joke only works if she takes a guess. On the other hand, had she gone up on a hill, her count would have been one over, having mistaken the dog for a sheep, and she would not have won the bet. Gil

I believe most of the following is correct. I'm trying to dispel some misconceptions. I might be wrong in some minor points. Take it with a grain of salt. Point 1. Yes, you can feed the t.v. output to an HT receiver, and it should work well. Point 2. I think the question is "how can this be a "multi-channel" system when we really only have a two channel audio transmission system? This is true in f.m. stereo, vhs tapes, cd's, or even stereo vinyl. There are two channels. Period. The answer is that we have a two channel storage and transmission system in many cases. Please consider that storage or transmission are essentially the same from the user end. (DTS and digital dolby are true multi channel. Maybe 5 or 6 discreet channels, like 5 or 6 phone lines. We are not discussing those here.) The hardware under discussion is pretty simple. There are two channels of audio feed to the user. Yup, the little RCA connector for "left" which is white, and an "right" (the little RCA connector which is red). No joke; these are two channels of audio. If you hooked up a pair of head phones to them, you'd be hearing all the real signal. No hidden mystery channels. What you hear is what you get. Nonetheless, we can fool around with the signals. We can add and subtract to make a L+R channel (for the center) and a L-R for the (surround channel). This is called a matrix in some publications. A poor choice of terms which just obscures what is going on. Lets ignore "steeing logic" and just look at power and math. This is simply analog systems. A. When two identical a.c. voltages (two sine waves in phase) ADD, we get twice the voltage or four times the power. This is 6 dB gain over one signal alone. P=+V*+V/R. B. When we take two identical sine waves and reverse the polarity of one, they cancel entirely. This is SUBTRACTION There is no voltage and no power. P=+V*-V/R =0. C. What happens when we add two a.c. signal which are random in phase? We get twice the power, or a power gain of 3 dB. This is kinda like having one raincloud putting out raindrops in a random way. If you add another raincloud, you get twice the rain. 3 dB power. However, the drops don't fall on top of each other to add up. This is a bit of a strained analogy. We never have cancellation of raindrops like we do with reverse polarity a.c. signals. On the other hand, some of random a.c. in the two channels are fractionally out of phase and might add a bit, or subtract a bit.) Lets look at the decoding end (at home) which just has a "processor" which has an input of a L channel and a R channel. This is total analog. The outputs can be stated in terms of input as L, R, L+R, and L-R. You might say, "Yeah, now we have four outputs, but it is not 4 channels, I just see L's and R's." Ya can't fool me. And you're correct. However, the power equations A,B,C above, tell us some things which might not be obvious. Lets take the L+R Output and feed that processor output to the center main channel. Lets assume the a.c. signal is identical in the L and the R transmission/ storage channels. (Only two of them, natch. My main point.) The power being sent to the center channel is 6 dB higher. Ha. We have to turn down the center amplifier by 6 dB if we expect equal power to the three front channels. What happens when there is just an L signal voltage into the processor as received from the storage/transmission system. The accoustic power comes out full strength from the left speaker, and it is 6 dB down in the center channel speaker. So, our acoustic output is mostly from the left channel speaker, something from the center, and none in the right. There is "leakage" but it is not too bad. Same thing happens when there is just a signal from the right channel of the transmission/storge system. Again leakage, but it is not too bad. PWK calls this a phantom center channel. The movie companies used the set up for years. Bell Labs tested the concept it in the '30s. It is "phantom" because we don't actually have a center channel in the transmision or storage channel. But, it can be created or re-created. Okay, what can we do with the L-R processor output? We imagine an input to the processor which is identical a.c. signals, but one is reversed at the source of the recording or transmission in polarity. Let's call it a positive in the left and the same signal but reversed in polarity in the right. The processor is going to subtract the reversed right channel input. But if the right input is already reversed, now it is reversed again, and it is back in phase. Therefore, if the people making the transmission or recording have planned properly, the L-R voltage encoded input signal appears strongly at the processor's L-R output. You can think of this as being L input minus a negative R input, which comes out of the processor as a strong representation of L-R. This L-R input to the encoding is the "surround" output of the decoder. Of course it doesn't appear at the L+R center speaker at all because the two voltages cancel. And note the L+R, which is supposed to be in front, doesn't show up in the surround. It is cancelled. Now you've spotted a problem. The L part and the R part of any signal is going to show up in the surround L-R. How do we make the brain think that the surround speakers are not putting out "real" L or "real" R from the front. The answer is that we can use some electronics to assist our ear/brain in interpreting the output of surround speakers as not being the true source of direct sound. Essentially, the output to the surround amplifier as delivered to the surround speaker is delayed by at least 10 milliseconds. This means it "sounds" to be at least 10 feet farther away. So, quite correctly, our ear/brain calculates, the L-R output must be distant. However, the undelayed R or L from the front is the true source. This is a balancing problem of intensity and delay. Whales and SSBN's might have better internal computers. Still, if there are two sources of sound and one arrives late and is a bit faint, it is probably an echo and/or from far away. The next question should be what is going on at the encoding end. The recording engineer for Monday Night Football, or Leno, or Star Wars, wants to take advantage of the processor and our set up at home. One thing the engineer can do for "surround" is put a microphone in the stands. Just a mono signal from the mike. Then the engineer puts the microphone output into the left channel which gets recorded or transmitted. The engineer also flips a switch for the feed to the right channel so that it is reversed in polarity going into the right storage/transmission channel. Oh, for the commentators, they go into both the L and the the R transmission channels with no polarity change. The band goes into the left channel, solely. The cheerleeders go into the right, solely. To reiterate, we only have two transmission/storages channels. However, at home, the additive and subtractive processor does its work. The commentator's sound add and are mostly heard in the processor's output (L+R) center, they're not expressed in the surround at all. Band is at the left. To the extent there is some leakage into the surround, their music is delayed and our brain tells us, the band is left, the surround is just an echo. The same for the cheerleaders. Surround from the microphone in the stands? Because it is L-R in the encoding, it is most stong in the surround decode. And since the decoding system has delayed it, our ear/brain tells us it is from far away, even if the microphone is in the middle of the crowd. Next question is: Why do they call it "5.1 Dolby sound." We've only discussed, above, a 4 channel system of encoding and decoding with all these pluses and minuses. How do all these numbers fit together? The answer is that the outputs, note OUTPUTS, are under considered. There is the front left (L) and right ® and center (L+R). That is 3. There are typically two surround speakers even though they are being sent an extracted L-R signal. That makes 5 total. The 0.1 is when we add a subwoofer to simply extract the bass tones and send those to a speaker better able to re-create them. Whoops, someone in the back of the room just asked whether HT systems can be used to play back recordings where there was no effort to mix this Left and Right and Center and Surruound when putting down a two track (channel) recording. Will our HT decoders do good things? The answer is that even if it was not intended during the recording, there is probably some "information" in any two track (stereo) recording which will be decoded by an HT system in our homes. If the two recording microphones were spaced in front of stage, the center channel decode is going to place the center performs pretty well. This was PWK's conclusion decades ago. His little network of resistors feeding a center channel amp, feeding a "Model H" as described in a Dope From Hope, does the L+R, as in an HT system. The next question is whether there is "surround sound" or L-R in old recordings. Yes, there is, potentially. It is not L-R (planned), but it is random. This is why I described the 3 dB gain of random phase, versus 6 dB of in phase, above. Hall sound is going going to be random. So to the extent any hall sound gets into the front center, it will not be getting expressed strongly. The in phase signals (center) will be expressed stongly. Therefore we have a situation where the recording engineer created a recording where L+R is stong because of microphone placement, and reverberant sound is always less. What is the lesson, even if the above is confusing? First, storage and transmission channels are often two channel systems at this date. It is a misconception that there are hidden channels for transmission when they say "surrond" or "Dolby." Second, the addivive and subractive processes can allow us to extract information from a two channel system, nonetheless. Regards, Gil Sorry if I've created confusion.

I have backs on my home made corner horns and they seem to work well. My best guess is that there are two issues. First, getting a good seal on the the last flare section along the sides of the K-Horn is important to good bass response. Placement in a corner is otherwise still beneficial. You have to consider the next piece of wall, floor, and ceiling structure extending out six or ten feet. Those help too. Therefore, do maintain the seal at the back and sides. But, understand, even with a "back" they must go at or near a corner for best results. "Best results" are pretty astounding. So, do it. Regards, Gil

And don't forget, the sheep dog might be offended. =8^o Gil

I must protest. 1. Such humor is ungentlemanly and impolite. Yes, it might be considered mild by some standards, but is still not necessary. 2. This bbs might seem like a small, semi-private circle. But it is not. Through the magic of the internet, it is world wide. You don't know who you might be offending. 3. It puts our hosts, the Klipsch organization, in a very bad position. This is important regardless of anyone's private taste in humor. They are being put into a position of being a forum for humor of questionable taste, or censoring things. I'm sure they don't want to do either. So, let me suggest that we stick to the subject of audio and music. Regards, Gil