pauln

Well, I guess I'll start the questioning with some friendly fire... It may take a moment to formulate my question because first I want to disclose my assumptions, some of which I am sure are incorrect: From a geometric standpoint, I am assuming that with two ears one gets two signals with which to process and extract salient features that we hear is music and sound. With one speaker mono, the channel reaches the ears at a slightly different time and this allows one to infer a first order lateral line of approach - but cannot distinguish whether the signal is from the front or the back. For example, the sound could appear to be coming from 30 degress right of center front or 150 degrees right from behind. With two channel stereo, the above happens for each of the channels, but an additional second order level of this spatial inference occurs because each ear is getting both channels in which the sound source has been "placed" by the inate phase difference (pan) between the channels in the recording. So now each ear is getting four distinct versions of the sound for what was initially meant to be a single sound source originally recorded. The four sounds would be: Left channel to left ear Left channel to right ear Right channel to left ear Right channel to right ear All this for a single sound like a lone flute. And there are higher orders of the same from reflections and reverberation. And none of this is yet sufficient to discern the sound source as coming from the front or the back. Now getting to my question. I know that a sound coming from the front should sound different than from the back due to the shape of the head and ear, but I don't see how one makes the distinction to tell the direction of orientation front from back... If I hear a sound, I don't know in advance what it should really sound like from the front or the back. How is it that with just two ears it is really possible to isolate the true source of sounds coming from the front or back? Bonus question: Does using four playback sources mean that each ear is getting 16 phase distinct versions of a single original sound?

With apologies to "The Ruling Class" 1971; Let me guess... this all started after noticing that when you prey to Him you find you are talking to yourself?

Thanks Gil, I suspected that an analysis that took the extreme case of approaching a square wave might clarify the limits of what could happen In your example you meant that the output is 7.07, not 70.7, right?

I'm sureyou did not hurt anything. Both those tubes are of the same "family" (12AX7, 12AT7, 12AY7, 12AU7) and the pins that connect to the elements inside the tubes are all wired to operate the same way; the tubes just supply different levels of gain, 12AX7 the highest, 12AU7 the lowest. If you noticed and enjoyed the tube sound you are no a greenhorn anymore but well on your way. Looks like you got a real nice amp to me; 12 watts and EL84 output tubes.

Hello Dr Who! So why is it that multiple frequencies require more power? Is it based on the area under the curve (for the positive side, and area over the curve for the negative side)? That would support the idea that square waves need lots of power. Is it based on how much the wave changes direction from moving toward pos to neg and back? This would imply that a single high frequency might need more power than a slow square wave, so this is probably no true? Is it based on the average time the wave is held further away from the zero level (square wave vs sine wave)? This supports the power hungry square wave even thought it is a "single frequency", but then there are the harmonics, right? I'm guessing that the frequencies in question mean all the component sine waves that make up specific sounds, so a low frequency square wave approaching 0Hz would require the maximum power - actually be at the limits of the power available. And white/pink broad spectrum noise would come in a close second place? Just wondering, it all sounds loud enough to me.

I started with a $29 new electric guitar back in 1970, played it for years without an amp, just listening to the radio and playing my records. Now I play the Strat Plus I bought new 20 years ago through a Fender Deluxe Reverb (reissue) tube amp, very slightly modified. I can tell you from experience that it absolutely does not really matter what you start out with. You will know when you need to get something better as you learn and improve. You really only need to worry about your equipment when you begin to play with others and perform, and even then it won't matter much until you are good enough that the difference in guitar and amp makes a difference in the way you sound. Not sure at all what kind of music you are thinking of in terms of crunch... I tried to see how crunchy I could get my Strat a while back playing on a metal backing track - look for the song "MetalMania" here. It's near the end of the list of songs.

Mentioned above: "20 db for headroom for musical peaks would mean that you would need double the wattage for each additional 3 db, or: 5 watts x2x2x2x2x2x2 = 320 watts available for peaks." Where did the 5 watts come from? Is it not more correct to begin with a reference listening level rather than a power level, then do the math upward until you were at a 20dB higher listening level, then see what watts that requires? For example, if I like an average loud listening level of 87dB, with speakers 105db/W@1m that level is acheived with just 15mW (or 0.015W, or 1/64th of a watt), and 20dB of headroom would call for less than 108dB, which would happen at 2W.Now I don't sit 1m from the speakers, but there are two of them and they are in corners, so the level drop with seating distance must be about matched by the doubling with there being two speakers, and room effect. So may I conclude that I have sufficient headroom for peaks with 2W? I still have another 1.5W of rated power beyond the 2W level... If I started with the 5W level that would be starting with a rather loud 112dB wouldn't it?

Yes! I have a little pen knife that has the inscription "Stolen from Paul Klipsch" on the side...

I recall Carver claiming many years ago that a careful live recording in his lab of a pair of scissors snipping paper needed in excess of 800 watts to almost get rid of clipping, and there was still measurable clipping (attributed to transient response) at 800W, but that was the most powerful amp available in the lab. Anyone remember that? I don't think there were any speakers involved at all - the analysis may have been done with just the signal from the amp itself, probably at or near the capability of the equipment of that time. I think the point he was trying to make had to do with the near instantaneous need for tremendous power to capture the leading edge of even a very quiet sound. Assuming this is true, what does this imply about what is required to hear a real pair of scissors snipping paper? Do our ears have the resolution to detect the leading edge of a very fast transient, do speakers? Is the clipping 800W amp really indicating something we can't hear, or something speakers can't reproduce? Do our ears clip?

I experimented with 4, 8, and 16 ohms on my La Scalas (type A networks). This is with SET amps. 4 sounded a little dark, but might be just right for really loud listening, I just don't play that loud. 8 sounded very good and natural, the correct "body" to the sound for medium loud listening. 16 sounded a little strange - light with accentuated high end that made some instruments (harpsichord, some electric guitars, etc) too bright. You could use a different setting on each channel and use a mono signal to hear differences.

Just guessing really, but whatever other characteristics (power, other specs) it should play your speakers clean and clear as loud as you would ever care to hear them when the volume is halfway up. I think there are some technical reasons why the halfway point is the maximum desired reference but I'm not qualified to explain it.

I only use CDs made by other band members that collect the tunes we learn for concert performance. This allows one to play the song a few times to find all the parts and learn the solos. For regular listening I only play records 100%. I never play the same record twice within a few months' span of time. My modest collection of a few hunfred albums provides all the quality and variety of music I desire. I have not been interested in nor have I bought any new music for over 25 years. I have bought about 5 old records since the end of the seventies.

Dave and family were excellent hosts as always, it was great to meet people, and Rick's burgers were amazing. Keeping in mind that I am used to my low power SETs... I was expecting good things from the Fraziers, they were in a bigger room with a pretty good placement. The Klipschorns were in a smaller room with record shelves and equipment tables just a few inches from their sides. We started with piano music on the Fraziers. The first thing I noticed was that the very loud parts seemed to congest and haze out a little. Maybe this is due to the amp (Soundcrafters). The sound was better on the quiet parts, but the low end sounded a little "off", a little different from a piano. The Klipschorns played the loud parts of the piano without any signs of strain and the tone of the low end sounds had the sound of wood, like a piano. The organ music showed a big difference in the low end. The Fraziers I felt did not control the lowest frequencies while the Khorns keep firm all the way down. The Debussy piano piece was interesting to me. On the Fraziers I was annoyed by the performance of the artist - the pauses between phrases seemed too long and affected - I kept wanting him to get on with playing the song. In this piece I noticed a variation in something with the high end. I think the geometry of the Frazier mids and tweets were part of this. There are four mids placed at the corners of a rectangle, vetically about 6-8 inches, horizontally about 18-20 inches (I'm estimating); but I was imagining that the increases on volume made the spaced apart mids (and pair of tweeter spaced apart about 8-10 inches) seem to lose focus. The Khorns sounded better, and I also forgot all about feeling annoyed at the performer playing the pauses too slowly - it was just beautiful music that sounded right. Even in the smaller room with the close-in furnishings so near to the sides of the Khorns they still sounded great. I've read about people unloading their Khorns onto the middle of the bare concrete floor of their garage and hooking them up to find they sound great even there... I almost believe it. I also brought some records to scare everyone. I was curious to hear the low end of The American Metiphysical Circus (Joe Byrd and the Field Hippies), and Steve Hillage's Fish Rising, both old trippy sounding stuff full of Moogs and Mellotrons and delightfully disorienting weirdness from the past. Also brought a cello concerto performed by Rostropovich which the Khorns really seemed to enjoy belting out. Oldtimer thought he recognised it, my have actually performed it many years ago in school. I vote for moving the Khorns out to the bigger space - might have to make a false corner for the left side, though. Thanks Dave, a fine evening of music with a nice crowd.

I should be there in a little while. Looking forward to meeting everyone. Paul

I waited until the end of the auction to jump in, no other bids. I think I'm going to like this one.

Well, I mentioned that this is a perfect example of the kind of MMC6000 that I would look for, but I waited until the end of the auction to see if you might be going for it...I was the only bid and won what I hope is a very excellent MMC6000 for a very low price. Funny, I would not have known about the auction except that I did a search to see if I could show you one. Worried a little that others might see it but I guess it was meant for me this time. So how is your 4002 sounding and behaving, all OK?

The reason I'm asking is because I really don't know, I don't buy them, and was wondering. Just a simple question... how long after buying a CD may you return it for exchange if it goes bad? If I buy a record, I expect to be able to return it within a few days if it has a problem, but after a few weeks or months or years I can understand why a return would be refused - playing it eventually takes it toll, especially with the expected playback on unknown pedestrian equipment and various levels of user care employed in handling, storing, and cleaning them. If I ran a record store I would certainly not take back records for exchange beyond a window of a few days - long enoughto allow the buyer to determine if it has problems in it's new condition. On the other hand, is not one of the fundamental attributes of the CD that it should be like new indefinitely, should not degrade in quality, should not develop problems through time from time itself or playing? This seems to be a whole different kind of music purchase. Furthermore, since there is increased awareness now days that you are paying for the "license" to own and play the music, does not that extend for any length of time? Has anyone ever taken back a 3-4 year old CD to be exchanged because the original went bad? I'm thinking that if I ran a CD store I would have trouble coming up with an argument not to replace bad ones, even old ones. With all the current interest in the music business about illegal music, does not a proper legal purchase of a CD entitle you to ownership of that music despite media failure?

For what's worth, the vintage ones are environmentally "greener"[*-)], but buying the new ones helps the people that make them and supports the product directly in the best way.. For vintage, their environmental impact to have been manufactured and distributed is a sunk cost at a price of low cost energy that has had 30 years to be gently absorbed whereas the footprint of the new ones is immediate and much greater because of the higher price of today's energy. Anytime you can place an existing product, especially an old one, back into service you are "saving the planet" a little bit, unless the old product's ongoing operational environmental cost of impact for it's anticipated service life exceeds the same cost of operation and manufacturing impact of the new version (as in if the new version is cleaner to operate). For speakers, the impact of operation is about the same, so the impact of manufacturing is the primary environmental negative aspect, but for the same reason is the positive impact on those that manufacture and distribute the new product.

You can join the B&O Beolist on Yahoo Groups to exchange questions and answers here You can find out more about your turntable here In my opinion, the very best cartridge for it is the original MMC6000, but these are hard to find. Best of the best ever made by B&O, contact line Pramanik diamond stylus, berylium cantilever, flat response up to 40KHz. See all about it here Here is a perfect example of the kind of MMC6000 that I would look for: new old stock and hopefully perfect and unused: MMC6000 on ebay The best very expensive replacements made by another company now are not even close to the quality of the B&O original MMC6000, primarily because of lessor cantilever quality material (ruby) and less enlightened stylus geometry. The next best would be the original MMC4000 which is similar, easier to find, and maybe just slightly better than the later MMC20EN, depending on the condition. All of the original MMCX000 series work fine on both tangential and radial tracking tables that have the arm made to receive them. The MMC20xx series are the same conformation and connection to the arm, but are engineered "down" in quality a little bit. The later tables used arms made for the smaller series of cartridges with the MMCXxx designation; MMC1, MMC2, MMC3...The better of these with the smaller numbers are a return to higher quality like the originals, but they don't fit the old classic tables. Strangely, the later tables for which these later cartidges were designed to fit were of inferior quality compared to the old classic tables from the seventies. My approach is to use the high quality classic tables with the original highest quality MMCX000 series cartridges, especially the MMC6000.

I'll say it again, not everyone is listening for the same things in their systems. I suspect that there are those that hear music as sound, and those that hear sound as music. I can see why those that hear music as sound would prefer the digital formats. I can certainly see why those of us that hear sound as music prefer the vinyl records. I don't have a credible explanation except that all the technical defects seem to be classed into the "sound" aspect, not the "musical" aspect. Therefore, when listening to music as "sound", the perceived minor defects are either there (records) or not (CDs). When listening to sound as "music", any defects have to be much greater to impose into the listener's enjoyment because the focus is on the musical aspect, not the sound aspect. There may be a better way to express this, but it is clear to me that we listen to different things when we hear the same recording, depending on our perspective from which we approach the thing mentally. Another way of saying it is: since I am a musician my records sound cleaner to me than they should - I naturally hear into the music and hear through the defects. This is not a claim for having Golden Ears, maybe just the opposite - Music Ears! Hope this clears everything up for everyone.

Time alignment does not involve the spacing between performers, but rather the spacing between the drivers in a loudspeaker system. Whenever there are two or more sources reproducing the same sound anomalies will occur. When the multiple sources are separated in time by more than 10 milliseconds or so, two distinct sounds, as in an echo, will be heard. This is not good. If the sources are separated by less of a time interval a comb filter will result. Comb filters cause a rough frequency response curve and lobing. Lobing causes a rough polar response which results in the sound reflecting off of the walls causing time smear and poor imaging. Irregular frequency response, time smear, and poor imaging. Not what we want. Proper time alignment of the drivers within a loudspeaker system eliminates and/or reduces those problems. The time/level shift between the performers in a recording is what allows us to perceive the stereo effect. A system with bad time alignment between its drivers interferes with that effect, affecting the realism of the reproduction. I'm not convinced. Here's my best argument: If time alignnment anomalies were a real thing, wouldn't one suffer hearing it even on a geometrically perfectly aligned set of drivers every time one leans over to pick up one's drink or pipe, scratch an itch, turn one's head to smile at one's girl, or pet the dog? Any sound in other than a free space or anechoic chamber (or head phones, and you know how unnatural they sound) will bounce off surfaces and produce a more complex sound comprised of echos of continous degrees of delay with variable shifts in phase and timbre. Seems to me that all natural listening environments will include some levels of reverberation, combing, frequency response variation, lobing, and many other deformations of the sound. I can see how extreme cases of these will goof up the sound to the point where it sounds lousy, but ultimately these things don't completety ever go away, nor would one want them to totally go away. I think these deformations of the sound are desirable to a natural playback to the degree that they stay under the threshold of not intruding into the music. I don't think time alignment can correct all these deformations anyway, maybe just a small part of them. I guess another way of thinking about it is that I get excellent realistic reproduction with my La Scalas, which should have some time alignment error because of almost a two foot difference between the tweeter and midhorn drivers, yet the integration sounds spot on perfect. Maybe the type A crossovers reduces the effect, or maybe the amount of the effect is still inside the envelop below which it is not possible to hear.

Imagine my shock and horror... As the lead guitar for a couple of bands I have had the opportunity to do some studio recording. Last time was earlier this year. We worked with the engineer during the editing and listened to the results through the studio speakers, but to evaluate the final mixdown the enginner sent us all out to the studio parking lot to audition the mixdown CD in our... CARS!? We moved from car to car listening to the CD on our various systems. That is how it is done nowdays. I took it home and it sounded fine on the home system.

Todd, I have a 4002 that is not in service. I bought it on ebay because it came with THREE cartridges, all of which met my high standards for my collection (I have a bunch of B&O cartridges I cycle through my other B&O table). For best results: Start building a cartridge collection. The way I did it was ebay, originals are no longer made, replacement new ones are not as good as the originals and extremely expensive. Best strategy is to commit a few hundred dollars buying a few on ebay and hoping you get a couple of great ones. Measure and evaluate what you find. With a volt ohm meter, a microscope, and some comparative playing you can rank them by condition and quality. All my stylus guards are numbered to keep track. Look on the various B&O sites and forums for details about maintenance and adjustments. My 4002 came with a problem in the optical sensor that detects arm displacement and converts that to drive the repositioning motor to move the arm across the record. The adjustment is off and pretty impossible to get back - I've spent hours with it to no success - either too fast or too slow. I don't have the shop instruments to make the adjustment fine enough. But I bought it for the cartridges anyway... If you have trouble finding a cartridge let me know.

I buy all of it except the time alignment and phase stipulations. I don't buy the time alignment arguments because I don't see it happening in live music - bands, orchestras, choirs, etc. If it is not critical or even possible to time align live performance, why worry about playback? I don't buy the phase arguments because virtually all recorded music has been phase shifted thousands of degrees between the mic and the tape. A few more degrees can't really hurt, can they? The time alignment and phase things might be important so someone producing an audiophile recording using very special mic placement and other techniques, no mixing, no editing, maybe no mastering. I did not choose my gear to play just a couple of perfect records... not that there is anything wrong with that!

Wm McD, I'm no Paul Klipsch, but I think you are on to something there with the lower bound for rarification. So air is single ended... makes me wonder if the guys are right that believe the absolute polarity of the speakers makes a difference, especially with a single ended amp? That is, does the polarity of the single ended amp signal and the polarity of the horn connection need to "match" so as not to reach the lower boundry for air pressure? Or another way to pose it; which part of the single ended amp signal wave (rounded upper or sharper extended lower) needs to be the half pushing the air on the horn "forward"? Seems like "backwards" would preserve the dynamics - greater amplitude lower wave pushing the diaphram forward for "unbounded" compression, less amplitude upper wave to stay inside the bounds of the rarification... And just how would one go about determing this? Does the network reverse some of the horns' polarity and not other's? I guess the different networks act differently here? For which horns would it be most important, mid and high? As a popular example I have SETs, La Scalas, and type A networks...