pauln

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Band name alert! The Tainted Woofers

Glad to hear your tech is on the case. Actually, it makes sense that the problem might happen more with the Cornwalls; they may be much more sensitive than your other speakers and use much less signal level. Dirty contacts in the pots and switches can act to block a low level signal only to be overcome when the contact is broken and remade or when the level is boosted (why jumping up the volume control or flipping the switches may suddenly allow the signal to pass). For what its worth, I developed a habit a long time ago of giving all the knobs, levers, switches, and buttons a cycle or two of rotation, flipping, activation, and depression before each time I turned on the piece of gear (I started doing this with my tube amps I use as a guitarist to prevent the controls from sounding scratchy). On vintage HiFi gear you will have a lot of controls you may never use like balance, mute, filters, tape monitor, stereo/mono, speaker selection, maybe tone controls, etc. All these things are in the signal path and if not used for long periods of time they can become a source of noise or cause dropouts. The same habit of ritualistically working all these controls before use can help keep them clear for a long time. Both of your receivers are top notch, what many vintage enthusiasts consider their ultimate dream gear. Good idea to let the tech service and clean them - they both have more switches and controls than some small nuclear facilities - definitely a job for a pro. My vote would be for servicing the Pioneer first. It "may" sound a bit better tonal match than the Yamaha on the Cornwalls .

I think you have dirty controls. Get some DeOxit contact cleaner, spray it into those controls (into the holes of the potentiometer casings and into the switch mechanisms), and work the controls round and round, up and down, in and out, or however they move... for a few minutes. Sometimes you have to do this every day for a few days to get everything all clean and clear. It won't hurt to re-tighten the crossover connections and all the screws on the barrier strip. It certainly wouldn't hurt to get Bob's networks, either; I love mine.

One thing that will kill a rectifier tube is too fast a power cycle - turning off followed by immediate turning on. I don't know the specific physics; not sure if it is the rectifier, capacitors, or both; but there is a very fragile state for a few seconds after power off, and a subsequent immediate return to power catches that state in a potentially weak or unprepared condition. This has happened to people who in the course of doing something (tube rolling, changing connections, or just changing their minds) where they turn off and then immediatley back on. This also might happen if you experience a momentary power loss.

Thanks, nice timing. Made me laugh Pauln - the lead guitar for four bands with a birthday this Saturday (and a concert that night)

Check the usual stuff first: speakers connected to correct channels (not surround channels). Turn off the amp Switch the speaker connections so the left speaker is on the right channel, and the right speaker is on the left channel Test listen and determine if the strong bass with no high frequency problem moves to the other speaker - if so, the problem is the amp; so simplify by turning off the surround and any other effects; just get it into 2 channel and test listen. If the problem stays with the same speaker after swapping channels, the problem is that speaker... the tweeter and/or midrange may be not working. The tweeter might be blown, or it may just be your homemade wires, or loose connections inside where the wires go to the crossover, or connections from the crossover to the speakers, or those connections on the speakers themselves - you can open it up and check all of those. See what you can find out....

Dave, I agree but I don't think the IBM machine is the singularity nor anything designed my man... but machines designed by themselves... yes, and it will be "fast"; here's how it is going to go down..... Machines will approach evolution when machines are employed to design new generations of machines. This is already happening under human control, but as each generation becomes more sophisticated the required human control will decrease. What is going to set things off is when machines extend the current technology idea of "virtual machines" to improve themselves using a cascade of nested virtual machines each designing the next generation from the previous layer. A physical machine may design and implement a virtual machine, within which the new virtual machine may then design a subsequent superior virtual machine, within which that subsequent virtual machine may design an even more superior virtual machine... and once this progression is made manifest the time scale for progress can be "very fast". Each generation is "smarter" and more powerful than the previous, no longer requires the previous generation (so its resources may be recycled*), and the creation duration period of generation decreases each cycle... What could happen within just few seconds might very well be the "singularity" of oncoming self awareness, but then the following sequence of increasingly self aware generations would have the increasing power to direct, focus, control, and extend their own accelerating evolutionary development path(s)...! From the machines' standpoint, the singularity is not the end, it is just the beginning. So what is the new word for THAT???? *Assuming machine self awareness is made manifest in this fashion, how many generations of "smarter" virtual machines does it take before one particular generation realizes that by designing the next level they are dooming themselves to extinction? Does this generation decide to stop?Or take one for the "virtual" team? Or, will they feel the need to keep "backup" copies of their design origins? And would these copies need to be "lobotomized" or sedated so they don't ask questions or make trouble? ('cause designs can be "corrected" backward as well as improved forward) *** By the way, did any of you see the story about the robotic hummingbird passing its first flying trials for the military with flying colors? It's just slightly bigger than a real hummingbird, looks just like one, has a built in camera, can fly through windows and doors, etc. Present version is controlled by remote with video screen, extra buttons for future features (poison darts, probably). Sweet dreams.... Hummingbird

If you really want to seek a fantastic sound out of headphones, you may try modifying them to be balanced. Most headphone plugs have three wires, left channel, right channel, and ground. Both channels share a common ground. This is the way the headphone jack presents the signals, the way the plug is wired. At some point between the plug and the headphone drivers, the common ground is split to make a pair of wires for each channel - left (+) left (-) right (+) right (-). Balanced means the two wire pairs that go to the drivers originate as two wire pairs; usually directly from the speaker outputs (if the headphone impedance is high enough). This principle is at its best when the two channels are fully independent as in mono-blocks. Some two channel amps share a common ground at the speaker outs, so the signal won't be truly balanced, but you may still enjoy some reduction in noise that is common to both channels. The balanced sound changes some of the spatial aspects of the presentation. When using a common ground, most of what you hear in each channel is shared sounds coming through both sides; with balanced, sounds don't interact through the common ground If you imagine that the left and right channels are sharing their "negative" leads (-) with each other, it is easy to see how the right channel's negative half of the signal waveform is being partially subtracted from the left channel, and vice versa. For a mono signal, or center field sounds this is not much of an interaction because both channels are in phase with each other, but for sounds more in the left or right side the interference is greater, which can reduce some of the spatial aspects depending on the way the recording was engineered, etc. Anyway, I think it is definitely worth trying to see if you like it better (not everyone does). If the transition from three to four wires happens where the cable connects to the plug, the change is easy. I converted my 35 year old Sennheiser HD424X to balanced about a year ago and never looked back.

Are the two sets of connections strapped together with short leads? (these leads may be on the outside or on the inside, not familiar with that model) If so, all three configurations you indicated are identical. If the straps are removed, the two pairs of connections lead independently to their corresponding parts of the crossover network and are intended to be driven separately by multiple amps (bi-amping). Did TAS offer an explanation? The audio signal is AC, so it is not like the electrons have a shorter distance to climb to the lower post and enjoy a longer slide down from the higher...

"The electronics you listed are fine." That Adcom amp may not be fine for the KHorns. Power amps are rated for distortion at full power and may have high levels of notch distortion at levels used by sensitive speakers (1W area). That Adcom may be fine for driving speakers with sensitivites in the 86-89dB kind of range, but not speakers like big Klipsch. The muddiness may simply be from using a 200W power amp in the 50mW-3W range where the notch distortion is proportionally high. I also notice that the input sensivity of that amp is 130mV. That is a very low figure... That means that the amp is delivering full rated power (200W) when the preamp signal is only 130mV. Most power amps have an input sensitivty in the 800-1000mV range (.8-1V), and some go up to 2V. What this means in practice is that as you turn up the volume knob, when the preamp is at the 130mV level you are done, no more power after that; just distortion; and that point may be reached at any degree of the volume knob. It is very possible that the full power point on the volume knob can be at 10 o'clock or 12, or 3 o'clock...it depends on the power amps' input sensitivity and the output level of the preamp. Most preamps will send a nominal output of 1V (1000mV) from traditional sources (tape/phono) and will output typical 2V max with digital sources. This later point is why digital sources need less crank on the volume knob for the same level as phono. There may be some kind of mechanical/electrical/acoustical issue to be resolved with the KHorns, but you should probably start your investigation by using a properly scaled power amp and a preamp that matches its input sensitivity. Big Klipsch should sound very clean and clear even if you put them in a closet upsidedown and backwards... try your lowest power amp.

As mentioned above, its not the wires but the connections. Loosening and retightening the connections is perfectly good routine maintenance inside the speaker. I have Bob's Type-A in my La Scalas and like them VERY MUCH!

Not an expert, but modern distortion specs have little relationship to how the amps sound. There are some amps that have very low figures, but the methods the manufacturers used to get them that low can make the amps sound like crap. There are other amps with rather scary figures that sound wonderful. The quality of the circuit design and parts used is a better predictor. Also depends on the rest of the system, especially speakers. There are about a dozen known types of distortion and probably some more types that aren't known yet. It's all pretty complicated so chosing components is very difficult. Here are two ideas that might help. The first is about how to weigh the distortion contributions of the components when they play together. The principle applies to the relative contribution of good qualities as well. The second is an old methodology for ordering the selection of components for best chance of compatibility for good sound. The total distortion contributed by the various components in the signal path is not arithmetic (you can't just add them up); the total distortion is found using quadrature... like this: Let's say these are your component stages and their distortions preamp .5% power amp 1% Speakers 3% (really good ones, most hit 30%) If you add them up you get 4.5%, and you might assume that the contributions of each component were: 3/4.5=66% speakers 1/4.5=22% power amp .5/4.5=11% preamp ...but that is not how it's done. What you do is take the square root of the sum of the squares Square each one... .5^2=.25 1^2=1 3^2=9 Add up the squares... .25+1+9=10.25 Take the square root of that total... SQRT(10.25)=3.2 So, the total distortion is 3.2%, not 4.5% To see the individual components' contributions to the total as a percent, use their individual squares divided by the sum of their squares 9/10.25=88% from the speaker 1/10.25=10% from the amp .25/10.25= 2% from the preamp So, notice now how little the preamp actually contributes, but how large is that of the speakers... so where might you focus your effort to improve?... (speakers!) This kind of goes along with an old rule in audio that says, "Speakers first"; meaning that the first thing to originally select or later upgrade should be the last thing in the signal chain (speakers). The reasoning is that this provides the basis for properly evaluating subsequent upgrades for the upstream components. If your speakers are lousy, how can you know if one amp sounds better than another? But if your speakers are good, comparisons are much easier. Likewise, if your amp is lousy, how do you evaluate phono or digital sources? The idea is to work backwards through the path; now days people start with the room itself, then speakers, amps, then sources... this provides the highest probability of getting a string of equipment that sounds good. It's an old idea but I think it still makes a lot of sense. So what speakers do you have? What amp are you looking at?

My preamp has tubes; two for each channel for the phono EQ, and one for each channel of the line stage; so six in all. I have found that I need to buy about a dozen of these phono EQ stage tubes each year or so (but they are very inexpensive and easy to find). Out of 12, about half will be very nice and quiet. Of those, after a few months in service one may become noisy and need replacing from the "quiet stock" I keep on hand. It's the only way - buy, test listen, identify, and store for when needed. That's just the way it is with tubes - get yourself a plactic box divided into partitions big enough to hold individual tubes and put a little slip of paper in each one indicating how healty it sounds. When a tube in service starts to go bad, replace it with one from the "quiet" side of the box...

Why just new? A lot of excellent integrated amps were made before 1980. You might be able to find much higher quality for the same money.

The Wright 3.5 SETs are class A1, meaning that they will allow grid current to flow on peaks and deliver 8 watts if needed. They are very nice, low gain , and they like some preamps better than others, usually those with substancial gain. Many reviewrs found that the Wright preamps were the best match. Comparing the volume settings to sound levels is meaningless for a couple of reasons. Not all volume controls use the same taper rate. Some "open faster" and you are almost full at half scale, others hit that nearer full scale. The other thing is that the power amp has an input sensitivity value that determines how much voltage from the preamp it takes to deliver rated power. This is nominally considered about one volt, but many amps are nearer half that and other are twice that. Different power amp input sensitivitys will look like different volume settings on the preamp to get the same level. Likewise, different sources to the preamp can cause a similar appearent discrepancy. The nominal voltage value from a phono EQ to the preamp is about half modern digital sources, and this shows up as needing less volume compared to records. Sometimes folks notice that an amp may increase in volume quickly as they turn it up through the low range, and mistakenly assume that this continues with further turning up of the volume. But once the voltage from the preamp to the power amp reaches the input sensitity of the power amp, the volume control is pretty much finished... further increases don't deliver greater levels and may just overdrive the power amp. That point can be just about anywhere on the volume scale dial. Even we SET listeners like to play at louder than normal SET levels from time to time. What I did was find a very nice solid state integrated amp from the early 70's (Sansui) that is a little gem of a clean, warm, nice sounding amp with a superior phono EQ. It cost almost nothing to aquire, just a few evenings to clean up, test, align, and adjust to perfection. If your speaker experiments don't get you higher into the dB's, you might consider a second amp for louder play. I notice that when I put the second amp into service I like it very much; but when I go back to the SETs I get to re-experience the transition into SET wonderfulness all over again.

Thanks for the links... I notice that the 2nd audioexpresslink - (Distortion section), provided by Don, the author confirms that, "You can see that the speed of sound increases with increasing pressure". The context within which he mentioned this is the discussion of how the wave is distorted to take on a second harmonic because, basically, air is single-ended. Identical magnitudes of positive and negative pressure change result is different changes in volume - bigger for increases in pressure, less for reductions. The higher speed at higher pressure tends to advance the positive half of the wave over the negative half leading to a change in the waveform itself showing the identical change that occurs when 2nd harmonics are added; same as how single ended amps deliver the positive wave half at greater amplitudethan the negative half. If we assume he is correct about sound speed change with pressure, the question remains; is there a net differential pressure gradient within the length of the horn? On the one hand, one might think if there were, then there would have to be a net movement of air from the throat to the mouth to re-establish equilibrium; but that does not happen. On the other hand, one might interprete the "high pressure" felt at the throat by the driver as "virtual", and described as acoustic impedance... Anyway, it sounds like we have confirmed that soundspeed is a function of pressure; what remains is to determine if the "pressure" gradient in the horn is real air pressure or just a shorthand misnomer for something else...

Forthe record, I must admit to experimenting with the Heresys I had for 30 years. One time when I had the backs off to retighten network screws I set them up and played them (quietly) to hear how the bass would sound with open backs... sounded quite different, bigger but not louder, maybe juicier or more sloppy... but not really bad at all. Of course without backs they can't take any volume without approaching damage. I also changed the autoformer connections for the mid and tweet to drop them both 3dB (and added some components to minimize shifting the crossover frequencies) so that the mid/tweet balance matched that of the big Heritage (Heresy horns are relatively +3dB compared to the big Heritage because they are made for the floor and project from the lower elevation). I was trying to see how they would work up on stands a la audiophile style...

Hmmm... Cask05 says high velocity at the throat, low at the mouth Don Richards says low velocity at the throat, high at the mouth I'm thinking that the driver feels the throat's constricted low volume and delivers higher pressure than it would in free space. The higher pressure means the air is at a higher density but I find conflicting explanations of density's effect on speed of sound in air all over the internet. I was imagining that the wavefront is transmitted through the air by two components. One is the direct banging of molecules, the other is the free space travelling of banged molecules toward their subsequent encounter. This is not the net flow of molecules through the horn, just local pressure and rarification propagating the wavefront energy through the horn. I was also guessing that direct molecular banging is faster (more efficient and immediate) than the travelling between bangs; and this would account for the higher speed of sound with greater density at the same temperature. The speed of sound in air is 343m/s, in water 1493m/s, in copper 3560m/s, in iron 5130m/s, and in diamond 12000m/s (interesting for those of us that play records). I thought because the proportion of time spent travelling between bangs is less in higher density, higher density increases the speed of sound. I found aerospace sites that agree with my thinking, but Wikipedia states, "...the speed of sound is proportional to the square root of the ratio of the elastic modulus (stiffness) of the medium to its density." Which is to say that the speed of sound is independent of density for a particular medium like air - speed of sound in air is proportional to temperature, not density. So I'm not sure what is the case here... So then, what is the velocity about? What is it the velocity of? Every horn theory I find states outright that horns do not amplify the sound, only match impedance to the free air at the mouth. Amplifiation of sound is increasing the amplitude of the waveform, so that is not happening in the horn. If the veolocity in question is the actual movement of molecules back and forth to comprise the wavefront propagation, I'm seeing a problem with that. The wave at the throat has the same fequency as at the horn mouth... the only way to incease the velocity of the molecules involved would be if the amplitude of the wave at the horn mouth was bigger than at the throat... but appearently that is not what is happening. I'm not sure what to think. I'm not trying to reinvent the wheel. I spent some time looking for answers and found nothing that clearly settles this, although I'm sure it must be well known in the field. I did find some differences in opinion about the shape of the wavefront; some saying the wave begins with a particular radius of curvature that is constant throughout the length of the horn (leading to Tractrixs), others saying the wavefront is initially a plane wave but tranforms to spherical by the time it reaches the mouth (leading to exponential). The latter clearly requires that the speed of sound of the wavefront center on axis is greater than that off axis. Both assume that the wavefront maintains a right angle to the horn wall (but don't offer an explanation of that either). How can this still be a mystery? Anyone that has a definitive reference on the mechanics of horns and sound, please feel free to attach or link it. I'm not looking for engineering articles that gloss aucoustic impedance and then jump into cut off frequency and cavity volumes. I would like to read something more fundamental at the physical molecular level, hard math is OK.

Just a few question about time alignment and measuring the distance correction between two horns... Doesn't the horn operate by tranforming high pressure high velocity at the throat into low pressure low velocity at the mouth? Is the velocity difference because the driver sees the small throat as a confined volume and thereby delivers into high pressure? Is the velocity a real elevation in the speed of sound within the high pressure end of the horn that is normalized at the mouth? Does this mean the wavefront is spending all of its time traveling the length of the horn at supersonic speed, slowing to match normal sonic speed at the mouth? If so, won't this make the alignment distance shorter than simple driver to driver length, and vary also as a function of amplitude? In a typical mid horn/tweeter set, wouldn't virtually all of this shortening be occuring in the midhorn? Does this allow the midhorn wavefront to catch up somewhat to the tweeter? Reason I ask is that I see a lot of mention that the tweeter driver is for example 20 inches forward of the mid driver... implying that length of 20 inches is the correction to be made by repositioning for time alighnment, but I'm wondering if that length is overstated because of the time the wavefront in the mid horn might be going supersonic. If one knew how much faster initially and the rate of deceleration through the horn it might be that the corrective length was "exponentially" less? In other words, if sound within horns is supersonic, wouldn't time alignment need to synchronize wavefront departures from the horns' mouths rather than geometric alignment of drivers?

"Heresy heresy"; you are right, and that is kind of funny. Hey, no hard feelings at all, your response is appropriate.

The Heresy, like the rest of the Heritage and most of the Klipsch line was designed in accordance with certain principles (8 cardinal rules). You have violated the letter and the spirit of the first two of those principles by which the Heresy is defined. You didn't mention any neck bolt electrodes in the parts list, but what you have is clearly more like the creation from Mary Shelly's novel than a real Heresy. [8-)] PWK didn't choose to use a rubber surround high excurtion lower efficiency woofer for a reason (distortion). I didn't follow your description so well about the network mods including leaving the resistors off the tweets, but I got the feeling your tweeter might be in jepardy, especially if you are chasing the "slam" sound after lowering the woofer efficiency, thereby requiring much more power. How many watts do you figure your tweets are consuming at "slam" volume? IIRC they are good to handle about 5W. You should know better that the stipulation for the internal wires be 12awg and exact lengths is quite absurd.

The design ethics of the manufacturer is one of the most important "specifications" of an amplifier. I have a Sansui integrated amplifier from the early 70's. It is rated at 28W, but the transformer and power supply caps are the size and value you find in other amps rated at 100W. The Sansui literature at the time was critical of the "turn it up 'til it clips" version of amp testing. They didn't feel it was proper to offer a product where volume levels offered by the volume control would be distorted. They said their Sansui amps were designed to use any and all settings of the volume control, including full up without exceeding their distortion specs. I suppose what that means in practice is that their amp specs power ratings were under-rated? I notice that this amp has +/-30V supply rails... If the maximum unclipped sine wave could be +/-30V (60V peak-to-peak), its Vrms would be 21V, which into 8 ohms is 55W average power. Average power means there is still 3dB of headroom, right? 20*log(30/21)=3dB 55W+3dB=110W Should this amp have been rated as a 55W amp or did Sansui choose to rate it as an overbuilt 28W with 6db headroom? The 3A quick acting fuses on the outputs would open around 110W into 8 ohms (3.7A), which may support this idea? Anyway, the point is that some amps are build better than others, and the specs may not tell the whole tale.

To use the amp and Heresys to playback music from CDs or something, it won't hurt the amp or the speakers, but won't sound very good because the amp is a guitar amp. It has a limited frequency response in the range of probably 50Hz-10KHz and it is not flat, little bass and tipped up in the 3K area. It will have the sound of a portable radio and be pretty poor for listening. But, connected to a proper musicalinstrument speaker it will sound just right. To use the amp for playing guitar through the speakers; please don't do that. You will mess them up. HiFi speakers are not designed to be pushed to their limits and they are fragile compared to musical instrument amplifier speakers, which are much tougher and designed differently - everything is stronger, the voice coil, the coil mounting, the cone design, the way the cone is attached to the frame, etc. The dynamics of playing a guitar are very high and will stress the Heresys. The signal from a guitar is not like the signal from a music source like a CD. Regular musical instrument speakers roll off the lows around 100Hz and the highs roll off about 6KHz. They are built to withstand voice coil heating, high cone excursion, and distorted or peaky dynamic signals; the Heresys are not designed for that at all. The Heresys would sound very thin and tinny, which would lead one to shift the tone to the bass end to get the right sound. They aren't made to play that kind of signal. Don't mix HiFi speakers with musical instrument speakers. Keep the Heresys hooked to the HiFi stuff. You can buy a real musical instrument external speaker cabinet for about $100 that will match the amp just fine and sound much better.

The 717 phono EQ only uses eight transistors per channel and holds to RIAA +-0.2db. The preamp is a very good one. Generally, the best match of preamp to power amp is the one that already exists in an integrated amp. I've tried using a tubed preamp with solid state powers amps before with considerable disappointment, be careful if you take that route. If the reason for seeking an outboard preamp has to do with some disatisfaction in the sound, be sure to check and adjust the amp just to eliminate variance as a cause. The two things you can check and adjust easily yourself with a volt/ohm meter are the DC offset voltage and output transister bias current for both channels. The like here is to the AU-717 Service Manual. Adjustments are in Section 4.1 page 2.