pauln

Much of what's going on is because of the psycho-acoustic sensory perceptual phenomenon of habituation - variance in the tonal balance presented to your ear is normalized by internal processes of audition. This is not a passive process but an active one that is in constant operation. The effect of this is the well known experience that any boost to bass or treble or volume initially sounds "better" or "more full and complete", but after a qhile it sounds normal. If the boost or variance is extreme enough to unduly load the habituation process, the creeping accumulation leads to a vague discomfort sometimes called "ear fatigue", in severe cases leading to the feeling of a mild headache. Apart from addressing actual anomalies in the room response or real deficiencies in the source recording engineering, the minimal or non-use of tone controls aligns with the idea that if your perception is going to normalize to something the best target toward which to normalize is the actual natural normal flat response, resulting in the least engagement of the habituation effort ( so you enjoy hearing music without the "loading effort" of the habituation process).

The F-M curve is descriptive, not prescriptive. It is a measurement of what is happening, not a suggestion of how one should adjust playback frequency response. People are free to do and enjoy how they please; but don't fool yourself thinking that using loudness compensation is approaching the correct musical picture. Extra boom and twinkle is a marketing feature, not a musical feature. Music at a distance or at a low level is not heard to have as much bass and treble... that is exactly how it is supposed to sound. There is no such natural sound as loud full spectrum frequency soft sound. There is no such natural sound as loud sounding soft sound. Adjusting to make it otherwise is not natural, does not sound natural, and is not a better representation of the music because that sound does not exist except by artificial sounding boost.

Not sure about the history of the Cornwall, but the Heresy has changed through its generations. Heresy I - rated 96dB/W/m and nominally rated 8 ohms impedance, but the minimum impedance is 10.2 ohms at 150Hz.Heresy II - don't knowHeresy III - rated 99dB/W/m but although nominally rated 8 ohms impedance the minimum impedance is 4.2 ohms at 150Hz.The sensitivity spec is with respect to voltage, not watts. The spec is 2.83V, which into 8 ohms is 1 watt, but 2.83V into 4 ohms is 2 W (2.83V into 4.2 ohm is 1.9 watts). The difference between 2 watts and 1 watt is -3dB. And 99dB - 3dB = 96dB. Assuming the Heresy III more sensitive is incorrect. The Heresy III needs close to twice the watts of the Heresy I to hit the same sound level. Maybe the III trades some sensitivity for a little more bass response?The international standard is that the nominal impedance specification is to be no greater than the minimum impedance times a factor of 1.25.For Heresy I, they could have been spec'ed as 12 ohm nominal speakers (12.75 ohm) instead of 8 ohm nominal.Heresy II - don't knowHeresy III, they should have been spec'ed as 5 ohm nominal speakers (5.25 ohm) instead of 8 ohm nominal. I have Heresy I, not saying they are better, just providing info that may or may not be important to your question about matching.

Of the original Heritage Klipsch (KH, LS, BK, CW, and H), all but the Heresy have their little horns elevated pretty well off the floor. The Heresy was designed like the others to be on the floor into a corner and toed-in, but because its little horns' elevations are much lower to the floor, the high end is set to be +3dB compared to the others in order to approach the same tonal balance as the other big Heritage speakers. Which the Heresy does fairly well if placed like that. A lot of modern mainstream hifi advice suggests pulling speakers out into the room and elevating them on stands. This will cause the Heresy to lose boundary bass reinforcement. According to PWK in the "Eight Cardinal Points" 1961 Cavities section - to wit, a 14 inch elevation causes -24dB at 50Hz and great mischief within the two octaves above that. Elevation of the Heresy up closer to ear level will present their sound as too hot, and that in combination with the massive loss of low end will result in a very thin (but clean) tone. A lot of Heresy listeners have "fixed" their speakers with corner floor room placement or a direct internal adjustment to re-balance the high end with respect to the low using the autoformer taps or other circuits. I will say again, anyone who has learned how to make Heresy wonderful, by whatever methods, will be in 7th Heaven with the La Scala.

I have had the Heresy for 40 years, the La Scala for 11 years. A pair of La Scala presents an almost five times greater driven acoustic interface to the air than a pair of Heresy (La Scala 1318 square inches vs Heresy 282 square inches). This difference is further expanded by the interaction with the room. As others mention, there is no way to really describe it; best way is to listen. I will suggest that if you have liked your Heresy, you would really love the La Scala...

I have noticed and enjoyed this effect, too. I think there are two interrelated things that make it happen. One is that within the room, when you are close the predominant field is the speaker, but when you are far the predominant field is the room reflections. There is a "boundary" between these two where the sound locks-in and becomes maximally coherent. The second thing is that this boundary varies with volume level, extending somewhat with increased levels but ultimately falling apart at loud levels. Deliberate room treatments may extend the boundary to allow greater listening distance and higher levels (and may cause the opposite to the effect we are discussing, where close soft listening no longer sounds quite right).

Sometime the simple stuff manages most of it. I don't have tone controls, so it's up to speaker placement, room treatments, and taps selection with tubes. As Chris A mentioned, a room will have the transition frequency range below which the room and speaker placement interact strongly, and above which the sound's reflective boundaries become prominent. Moving the speakers and listening can zone in on the best behavior of the low end. Pushing in and out of corners and towards and away from walls offers a range of response. Then, once the speakers are set, using a small mirror to locate the primary reflections can indicate the spots that need some treatment (high frequency sound and light are both "line-of-sight", so wherever against the wall the mirror is when you can see the reflection of a tweeter in the mirror from where you listen is right where the treatment for that tweeter's primary reflection needs to be - just large enough to occlude the reflected image of the tweeter). With tube amps, if you "tap down" (e.g., using the 4ohm tap for 8ohm speaker load) the high end in general will be slightly reduced and the damping factor increased for control of the low end. If both these effects are beneficial in a particular arrangement, this is an easy "head start" in the right direction.

I vaguely recall the "horns love tubes" thing with respect the speakers having a rising impedance with frequency (typical of horns). The rising impedance with frequency was supposed to have a nice "taming the horns" effect when used with tube amps - in the high mid-range where too much is most noticeable... some kind of synergy resulting in a naturally well behaved frequency response (that was nicer than when using solid state, for which the tonal balance was not originally designed?).

Something else to keep in mind... Expensive stylus diamonds are polished before being mounted to the cantilever; inexpensive ones are not. The first hours of play with a new inexpensive diamond is not going to sound as good until it gets polished from playing. "Real serious" audiophiles use non-critical records with a new diamond because even the pre-polished ones get a much finer polish with playing, but this initial polishing by record play is a higher wear playing than after, when the polish has become very fine. The prime component of stylus wear is the powdered diamond dust in the groove... after a diamond is finely polished with some play hours there is little subsequent accumulation of this (if you clean the record well), but in the first hours, especially with an inexpensive diamond, both the diamond and record get worn, the diamond achieving its final fine polish at the expense of the record suffering marginally more wear. So, some people use a "lessor" importance record to break in a new diamond. And, some people make an evaluation of a new diamond too early. Worse case is using a new diamond on a new record, then having second thoughts about how it sounds and subsequently wondering how the new record got messed up so fast...

I think imagining is mostly stemming from the quality of the source recorded engineering, geometric room/speaker placement, amp topology, and finally the speaker itself, probably in that order. That said, the variation of imaging among speakers I think is a strong indicator of what they get right. Based on my 42 years experience with Heresy and 10 years with La Scala, I would say the former is very good, and the latter is even more very good.

I have CT-120 in my La Scalas with type A crossover, large room, carpet, books, furniture, no special treatments. Everything has always sounded best in this room. CT-120 sounds great with both SET and vintage solid state. I think they sound clearer.

Look at Dope From Hope Volume 9 Number 1 February 1968 "Room Proportions and Distribution of Eigentons". See attached PDF. Included in the supplement (second page) added April 1969 is the result of Bolt's analysis in his paper about frequency statistics for rectangular rooms... PWK reproduces a tool from the paper that summarizes the result and may be used directly for evaluating rooms. It is an X-Y graph in which the X values are the ratios of room width to room height, and the Y values are the ratios of room length to room height. You calculate both ratios and use those as coordinates to plot your point location in the graph... hopefully inside the boundary area defined by his technical results which indicate within what room proportions the sound frequencies are best behaved... PWK comments that this boundary contour encloses "near perfect" rooms, because some rooms slightly outside the boundary sound OK to him. Your W:H is 20/12=1.666... Your L:H is 1.666... The point in the graph represented by those coordinates is well outside the contour boundary, which is not surprising because square rooms are "bad rooms". DOPE from HOPE!!!.pdf

The amps are doing fine compared to speakers, most of which dump 99% of the signal to heat. The automobile V8 dumps enough heat to warm a four bedroom house when it's 15 degrees below (don't try this at home). On a more philosophical note, I think concerns about the "inelegance" of things that seem overly hot is a persistent human projection. Metal is not uncomfortable at 250 degrees despite a strong tendency to think that it must feel least stressed at room temperature. Car engines are designed to operate well over 200 degrees, and tubes need some of their elements to be thousands of degrees. The Earth is not inelegant because its core (268 billion cubic miles of iron) suffers a high temperature, and the Sun feels no pain or shame. ... some things need to be hot.

La Scalas have a reputation of presenting a little less low end (but very cleanly). Using my modest 45 year old solid state 28W rated integrated amplifier, I have never thought the low end was lacking at all. I attributed this to having listened to Heresys for 30 years before getting the La Scalas... I never really thought the Heresys lacked much, and the La Scalas just sounded absolutely complete. Now, over in another audio forum that has a section that is "Exclusively Sansui", a number of members there restore and refurbish vintage amps and make threads in which they take pictures and walk through the process. In one of those, the fellow was working on the phono pre-amp board of the top of the line of the series (AU-9500) and I was interested because mine is the AU-6500. He was doing a couple of things, including doing some calculations, circuit simulations, and measurements of the parts of the phono board that are responsible for reversing the RIAA compensation applied at the time of cutting the record... he found an anomaly. In the AU-9500, the spec is flat but the actual component values on the stock phono board started raising the bass at about 320Hz with a curve that was about +3.5dB upon reaching down to 20Hz. This apparently undiscovered "mistake" was behind the nearly universal appeal of this amp. I provided the info and asked if he could run the figures for the AU-6500. He did, and these also begin to raise bass about 160Hz but pushes to +3.5dB down at 20Hz. It looks like this whole series of amps enjoys this "mistake" and explains their reputation and characteristic desciption of "warm tube sound". This is not a huge boost; the lowest where the La Scala can use it down around 40Hz the boost is +2dB. So three things of note: - this is a nice match with the La Scalas, not totally recovering a flat response, but helping a little bit - this boost in the phono stage rather than doing it with the tone controls may be a nicer more subtle way of raising the low end, in so far as the main amp may take the preamp signal flat without tone adjustment - specs may be misleading for various reasons

I use an SPL meter. The average level over how long a time span is a good question. The meter itself is using a standard summation time for averaging that is about two seconds, so that average reading varies with the music... but if you watch it for a little while you can see what the "long term" average is for most music. Last night I checked and was averaging 75dB playing jazz records for a couple of hours; the loudest I ever listen is about average 80dB which I consider pretty loud, but many people consider that just the beginning. The standard level for control room monitoring in recording studios is slightly above 85dB, which means that the engineer set the tonal balance at that level.

Ultimately, power requirements have to reflect listening level requirements and distortion requirements. Let the listening requirement be 85dB average. Let the distortion requirement be for now, "clean"; un-clipped similar to values at rated power. Let the crest factor requirement be +20dB. Crest factor is a comparison of the peak level to average level with respect to recorded music. A crest factor of +20dB is sufficient to cover all recorded music except for some test records and demonstration albums. This +20dB is a convenient figure because it represents a linear factor of 100. An amp rated at 100W subject to a +20dB crest factor results in a maximum average level of 1W. This means with the 100W amp the maximum average power level that does not exceed 100W when the +20dB crest factor is applied to the average power level is... one watt. Since this 1W level is average, the crest factor above is reflected by a similar range below. 1W + 20dB = 100W 1W - 20dB = 0.01W or 10mW So the 1W level is in the middle of the 10Mw - 100W range... half the music is above, half below. At this point it is clear that even with a 100W amp, the "first watt" is half the music. The listening level of 85dB is also a convenient figure because the average speaker has a corresponding sensitivity specification. Continuing with the above instances, the 85dB speaker means the 1W average power will result in 85dB, and the +20dB crest factor applied means the level from the speaker will vary accordingly. At 10mW the speaker will present 65dB At 1W the speaker will present 85dB At 100W the speaker will present 105dB (if it can) There are some simplifying assumptions in the above, but the point is to compare this to using low power amps and high sensitivity speakers. So, we start again with the same requirements... An amp rated at 1W subject to a +20dB crest factor results in a maximum average level of 0.01W. This means with the 1W amp the maximum average power level that does not exceed 1W when the +20dB crest factor is applied to the average power level is... one hundredth watt. Since this 0.01W level is average, the crest factor above is reflected by a similar range below. 0.01W + 20dB = 1W 0.01W - 20dB = 0.0001W or 0.1mW So the .01W level is in the middle of the 0.0001W - 1W range... half the music is above, half below. At this point it is clear that with a 1W amp, the "first watt" is ALL the music. The listening level of 85dB is also a convenient figure because the sensitive speaker of 105dB is +20dB above that, same as the crest factor. Continuing with the above instances, the 105dB speaker means the 0.01W average power will result in 85dB, and the +20dB crest factor applied means the level from the speaker will vary accordingly. At 0.0001mW the speaker will present 65dB At 0.01W the speaker will present 85dB At 1W the speaker will present 105dB (easily) So at this point, both approaches meet the requirements at first glance. But looking further: In the high power amp low sensitivity speaker combination - class-AB has a constant level of non-harmonically related crossover distortion whose proportion of the total distortion increases with decreasing power output - the need for global negative feedback for solid state amps to be stable is a known detriment to harmonic distortion spectrum (shifting to higher order) - the harmonic distortion rises at lower power output levels (below the 100mW level is difficult to measure and distortion plots cut the line off below 100mW. The few instances when an effort has been made to examine distortion levels below 100mW has given a shocking result; in one case a highly regarded 250lb amp with "perfect specs" that sells for many thousands of dollars was measured to have distortion levels of 8% in the sub 100mW power output. The implication is that modest power amps may be even worse.) - high power levels to the speaker heat the voice coils, increase their resistance, and decrease their marginal output, so resulting in compression of dynamics (so they may not make it linearly to +20dB) In the low power high sensitivity combination - class-A has no crossover distortion - class-A SET amps typically have no global negative feedback - class-A SET amps' distortion curves get cleaner with decreasing output levels - low power levels to speakers are thermally insignificant to their operation (maximal dynamics) Keep in mind that the SPL meters and meters on amps are not perfect. Those that show average level are generally OK in so far as the level shown is close to correct. Those that show peak levels actually are not doing so... the method for summing amplitude over time to get the canonical "peak level" was standardized for male voice early in the radio broadcasting era. The summation time is too long to represent an instantaneous music transient peak and falls short by about 13dB of true peak. This means that when you see the biggest peaks on the meter bumping up to 87db, the actual peaks are up around 100dB.

What is all this talk about changes cause them not to be Klipsch speakers? If I changed the crossovers and the tweeters, does Klipsch advise I remove the badges? When asked, "What kind of speakers are those?", should I say, "I don't know", or "I can't tell you" or "They are my own custom design"? For now, I'm going to go listen to Sonny Rollins' Way Out West on what I have been thinking are Klipsch La Scala...

If you work in certain industries or have access, there are some commercial packing materials that make fine foundations for bass trap construction. For example, in the IT world one might buy hard drives in lots of 60, and they arrive embedded in a giant slab of dark high density foam; typically a big slab like this has three columns of twenty drives, each residing in a 1/2 inch slots already cut into the material to hold each drive... so the slab may be 2 feet high x 8 inches deep x 18 inches wide, with 60 half inch slots... very bass trappy.. Similar types of packing/protection materials are out there, typically discarded as waste.

Let the speed of sound be 343.2 m/s hz wl in m wl in s wl in ms 4.1 ms % 400 0.86 0.003 2.5 164% 283 1.21 0.004 3.5 116% 200 1.72 0.005 5.0 82% 141 2.43 0.007 7.1 58% 100 3.43 0.010 10.0 41% 71 4.85 0.014 14.1 29% 50 6.86 0.020 20.0 21% 35 9.71 0.028 28.3 14% 25 13.73 0.040 40.0 10% First column is woofer Hz Second is wavelength in meters Third is wavelength in seconds Fourth is wavelength in milliseconds Fifth is what percentage of the wavelength in seconds is 4.1 ms - the percentage indicates how much of a full wavelength is being delayed... 100% means a whole wavelength of shift, 164% means a little more than a whole wavelength, etc... So what is interesting is that for very low frequency sounds, the 4.1 ms delay is negligible, especially since low frequencies take more wavelengths to be discriminated in the first place, and you can't determine pitch from a quarter or half wavelength, actually you have to hear quite a few cycles to get it, the lower the frequency the more cycles it takes to determine. So the lower percent portion that is the 4.1 ms delay (potential advantage) is buried in how long it takes hear something down there anyway. On the other hand, the higher frequencies are being delayed an increasing percentage of their wavelength - those exceeding 100% are delaying more than the first cycle. But even at the high end, at 400Hz the 4.1 ms delay is only acting to shift its first approx. one and a half cycles (or maybe as if it cut off the first 1 1/2 cycles and "added" it to the end of the sound). Anyway, the numbers are interesting?

If you are referring to me, I'm a musician that performs live concerts. I played three hours of jazz Saturday night with one of my bands... I know the sound of live performance.

I hear what Deang is saying. I bought one of the very last type "I" La Scalas about the time the II's were first coming out. Finished in piano black, upper grilles, and AL-5 networks. I pulled out the AL-5s and put in Type A, paper in oil. Each of the AL-5s had a higher component count than my 3.5 watt SET mono blocks combined. I am sure they sound fine with high power amps that most are using. I can't speak to the plots, slopes, and curves, but the Type-A sound "right" in the old style La Scala being driven by flea power amps peaking close to 1 watt. From what I recall over ten years ago, it was the AL-5 high end that motivated the change, but it has been so long I have no recollection of details. I just know I've been as happy as can be since then.

Reminds me of a story... I got one of those "come see the property and get a free gift" things in the mail, had free time so went to check it out. The guy showing the place was driving a very old but nice car from the 60's all over the rough still unpaved roads of the development... the car was dead silent, not a squeak was heard. I have been in old cars before that even just rolling down a nice paved road the noise from the interior and dash sounded like the bridge of the Enterprise from the original Star Trek... all chirps, beeps, squeaks, boings, etc. I finally asked him how his car was so quiet... He opened the glove box and inside was a butter knife and a dozen packs of paper matches. He said in the months that he had been doing this he had heard lots of squeaks, but for each one he folded a match over the end of the knife tip and pushed the match into the crack, seam, crevasse, or whatever place was the source of the sound... as many as it took to tighten up the offending parts. Over that time he had stuffed hundreds of matches into the dash and interior until it rode quiet. ....... I'm still curious to know if the selections on the K2 and/or Peach for input sensitivity and/or target input impedance got changed during the move.

Me too; always been ecstatic about my WPA 3.5 SETs (and Wright phono preamp/lineamp). Listening at an average 80dB from the chair, I have +20dB of headroom under the first watt. I seriously doubt the La Scalas have ever seen a peak over 1 watt in the years I've had them.

A possibility...? Some poorly designed power amps "hang" at their rail voltage when they clip, so the amp's voltage level does not instantly and gracefully recover from a clip as the signal voltage drops below the rail voltage, but instead by power supply hysteresis hangs at the clip voltage until a certain drop in the signal voltage occurs. This is a distorted sound. When a pre-amp clips, a perfect power amp even though running well below its own clipping level can sound just like the poor rail hanging amp mentioned above, because it is trying to play the pre-amp clips with fidelity. This is not the power stage clipping - the power amp is trying to amplify the pre-amp signal's flat spots of DC. This is rare in integrated amps because the designers know the pre-amp and the input sensitivity of the power amp stage. It is something to be aware of when matching consumer level audio components, but rarely a problem; happy variations in the right direction are just called "synergy". Pro gear power amps often have diminished input sensitivities (requiring more voltage for rated output), so consumer audio pre-amps may lose headroom with respect to a pro-amp. For example, a very common input sensitivity for consumer audio power amps has been 0.775V for rated output. A common value for pro gear is 2V. That would be a loss of almost 9dB of the pre-amp headroom, which might be a source of clipping in front of the pro amp. Just something to check... edit> just took a quick look and found this... Per the K2: "K2: 32.88 dB gain at 1.4 volt sensitivity; 25 dB gain at 3.0 volt sensitivity" Per the Peach: " There are 2 listening modes, Hi and Lo, and 1 bypass mode. The Hi mode is for amps with a high input impedance and uses only 1 tube in it's gain stage. The Lo mode is for, you guessed it, amps with a low input impedance. This mode includes the other 2 tubes for the gain." If you know what these settings were before the move, check and see if they were changed during the move. If you don't recall the original settings, maybe time to do some careful testing...

Of all the anomalies that may be present in the gear and the room, I agree that imaging will be a primary attribute that suffers. Or to say it the other way around, imaging is a first victim of problems (or inadequacy) of the gear or room. Imaging may be the best gauge, but it may not be the only canary in the mine... but perhaps it is... Imagery and dynamics are closely tied. If the imagery is correct I think that requires the dynamics be correct as well. Maybe inferior dynamics presents a "flatter" presentation of an otherwise well imaging system/room, whereas getting the dynamics better might present some depth and body to the objects of the imaging... more spatial presentation... advancing from a "movie" to a "play", from a 2d presentation to a 3d presentation. So getting the dynamics right may not get best imaging, but when the imaging is right that implies that the dynamics (and a whole host of other things) are right. So I think I agree that imaging quality is the key, and likely the last of a series of unlocking of the system/room potential.