TONEARM DAMPING : DAMPED OR NOT ? ? USELESS ? ? WELCOMED ? ?


Dear friends: This tonearm critical subject sometimes can be controversial for say the least. Some audiophiles swear for non damped tonearms as the FR designs or SAEC or even the SME 3012 that is not very well damped in stock original status.

Some other audiophiles likes good damped tonearms.


In other thread a gentleman posted:


"  If a cartridge is properly matched to the tonearm damping is not required. " and even explained all what we know about the ideal resonance frequency range between tonearm and cartridge ( 8hz to 12hz. ). He refered to this when said: " properly matched to the tonearm ".


In that same thread that a Triplanar tonearm owner posted:


" This is the one thing about the Triplanar that I don't like. I never use the damping trough...... I imagine someone might have a use for it; I removed the troughs on my Triplanars; its nice to imagine that it sounds better for doing so. "


At the other side here it's a very well damped tonearm:


https://audiotraveler.wordpress.com/tag/townshend/


Now, after the LP is in the spining TT platter ( everything the same, including well matched cartridge/tonearm.  ) the must critical issue is what happens once the cartridge stylus tip hits/track the LP grooves modulations.

The ideal is that those groove modulations can pass to the cartridge motor with out any additional kind of developed resonances/vibrations and that the transducer makes its job mantaining the delicated and sensible signal integrity that comes in those recorded groove modulations.

 That is the ideal and could be utopic because all over the process/trip of the cartridge signal between the stylus tip ride and the output at the tonearm cable the signal suffers degradation (  resonances/vibrations/feedback ) mainly developed through all that " long trip " .


So, DAMPING IS NEED IT AT THE TONEARM/HEADSHELL SIDE OR NOT?


I'm trying to find out the " true " about and not looking if what we like it or not like it is rigth or not but what should be about and why of that " should be ".


I invite all of you analog lovers audiophiles to share your points of view in this critical analog audio subject. WHAT DO YOU THINK ABOUT?


Thank's in advance.



Regards and enjoy the MUSIC NOT DISTORTIONS,
R.






Ag insider logo xs@2xrauliruegas
Post removed 
Dear friends: This is a clear example of the benefits of a tonearm damping mechanism as this Technics EPA250 mounted in a SP-10MK2 in the Ortofon extremely LOMC cartridge MC-2000 reviewed by Pisha in Audio magazyne ( page 83. )  where live measurements said the resonance frequency between cartridge/tonearm was/is 5.1hz and this combination tracks with out trouble the Telarc very high velocity cannon shots in the 1812 recording and this I can attest it because I tested  the MC-2000 several times with that great Telarc recording and only with well damped tonearms ( GST-801, MAX,EPA 100, AT1010. ) made the LP tracking with out problems ( with SAEC/FR/Grace/AT 1503 even that the resonance frequency was in the SAEC/FR/AT inside/nearest the ideal frequency range just can't do it. ): 

https://worldradiohistory.com/Archive-All-Audio/Archive-Audio/80s/Audio-1984-12.pdf

R.
Tonearm superiority isn't meant by tracking everything with all cartridges. It should do it also proper way controlling resonances and going fluently via deformations. 
If to use only tracking test Telarc, HiFI news or even better Ortofon test record you'll find that most of the high end cartridges don't track everything. For designer it's quite easy to make suspension softer and to get 100 μm peak although sound generated by coils and in case of too much fluency in suspension sound level will be uneven. So designer has to put into one sound profile - size of coils, impedance of coils and magnet type and shape, weight of cartridge and decide about suspension material, type and shape of cantilever it's material, stylus length and diamond shape. 
Company which built their reputation on tracking is Shure with V15 series If you'll check Shure Ultra 500 which is the best in the line you'll find that it has brush at the front. It's not intended as anti-static brush, it's part of suspension keeps it  more steady.

Dear friends: "  "  If a cartridge is properly matched to the tonearm damping is not required. "

through this thread that statement not only can't be corroborated/confirmed but the other way around: is totally false.
It's easy for any one of us make a critic or adverse opinion in an audio subject and unfortunatelly  this kind of posts are very often in the forums where the person that post a critic never gives any prove/facts that can confirm with out doubt that he is rigth, any facts that be the foundation of his opinion. 

So that kind of statements are not only false ( till can proved. ) but totally useless for all of us.

In the articles linked we can read:

"  The first (A) is the result measured with an arm/cartridge resonance of 7 Hz. In (B) the resonance is around 9,5 Hz and in (C) it has been put at 16 Hz and some damping applied. The lack of sidebands in (C) compared with (A) gives a clear improvement in sound quality in terms of increased stability and transparency in the stereo picture. From this it is clear to see that to improve audible quality the main problem IS TO REDUCE THE RELATIVE MOVEMENTS BETWEEN CARTRIDGE AND RECORD AS MUCH IS POSIBLE.
 !n other words, ONE HAS TO DAMP THE TONEARM RESONANCE . 
In pursuit of this goal one should not make trade offs with respect to rigidity of the tonearm tube and fixture. Flexing in the arm and other spurious resonances could then be the result and destroy the stability of the stereo image. "



""  However, one must realize that these resonances build up when hit by transients in the music, either direct from the groove or indirect via the loudspeaker. When the transient is gone the resonances deliver their stored energy BACK to the cartridge and IS NOW CONVERTED TO ELECTRICAL SIGNALS AT A TIME WHERE THERE SHOUILD BE NO SIGNAL. ""



"""  Lastly we demonstrated the influence on tracking force giving distortion in the midrange during playback of high frequencies. As a parallel to the now widely used term TIM (Transient Intermodulation Distortion) which indicates the distortion components falling into the audible band when high level and high frequency (out of band) signals are fed to a feed-back amplifier — we could introduce the word BIM (Ref.5). Bass Intermodulation — a result of a high level low frequency (out of band) signals from a record boosted by an UNDAMPED tonearm resonance. The last conclusion we can draw from these investigations is the means of avoiding BIM. Since we have to accept that practical records (Ref.2) contain a large amount of "rubbish" centred around 4 — 5 Hz including warps, the optimum solution is clear... In addition some DAMPING should be applied to eliminate oscillations and influence on the frequency response above 20 Hz. """

Btw, the capital letters came from me but comes in the articles.


""""  When looking a iittfe closer to the oscillograms in Fig.28 it can be seen that in the case of arm nr. 3, the tracking force 20% of the time is below 5 mN (half of the preset value). It follows then that the cartridge is not able to track high frequencies without distortion for a considerable part of the total playback time. In this connection it could be mentioned that in a corresponding time interval the Fig.27. Set-up for recording the tracking force variations during play-back of ordinary records tracking force is far above what it is Fig.29. Here we have shown on the B&K Type 2131 1/3 Octave Analyzer, the distortion from the playback of a 1/3 octave pink noise at 20kHz (from test record B&K OP 2011). supposed to be with possible acceleration of record wear. The actual increase in distortion due to mistracking is illustrated in ...""""


SO, Tonearm and Cartridge must be well damped no matters what till some one comes here and proves with facts/live measurements damping is not necessary to improve the quality performance of any cartridge/tonearm combination.


Regards and enjoy the MUSIC NOT DISTORTIONS,
R.




Dear friends: In this thread I posted the importance to have deep first hand experiences listening live MUSIC seated at near field position and this " condition "  is need it to any one of us can make any kind of listened evaluation of the quality of our room/system with the LP tracks we are listening.

I posted all those from several years now and posted often in different threads and that I remember only @mikelavigne made comments on that issue and I posted the very first time when my common sense told me that the recording microphones are " seated " at really near field to pick up the MUSIC source information and that information is the one recorded in the cutted/pressed LPs.

THose adjectives used for we audiophiles as: warm, organic, sweet,  and the like just does not exist in near field live MUSIC but what we like in our room/system quality performance levels are  what is inside the overall meaning of those adjectives and many more and we always are looking for that " nice " sound ( that's the way we make evakluations/test/comparison and what defines ourdecisions to buy this or that audio item. ) that does not exist in near field  live MUSIC, so we are just wrong it does not matters that that is what we like it

I have those kind of experiences through many years in different venues with different kind of MUSIC when attend to do it and before I learned my take on the issue was exactly as the one of any audiophile.
Even when I make an audio items evaluations my LP tracks listening time I made it seated in my room at near field.

Well, I just found out an artricle that I have to paste here because the site just does not works to link it. In this article for the first time I confirmed that I was and am to wrong in this important issue and in the subjects of this thread: we have to know what to look for when listening damped against undamped tonearm/cartridges combinations.
The article autor is a :Professor Mathematics Ph.D. University of California, Berkeley  and a symphonic orchestra musician/player and audio reviewer:


""" How far away from the performers do you have to be for the reverberant soundfield to be at least half the sound you hear (in sustained sound)? Not very far. The precise answer depends on the hall; but usually at anywhere beyond around 20 feet, the reverberant sound predominates in a typical hall. "This would mean that [in a usual concert hall) only for the musicians and the conductor (and the microphones placed in their vicinity) is the direct sound not overpowered by the statistical (reverberant) sound".

Records and Reality: How Music Sounds.


Right from the beginning, there is this difference between what is recorded and what you would hear if you were at the performance: Almost all records are made with the microphones closer to the performers than the audience would be. The sound very close to the performers is also an aspect of the absolute sound of live music. But the sound that the composer and the performers intend for us to hear is the sound at audience locations, and the sound the audience would hear is presumably what we should be trying to hear at home from our audio systems.

Close-up and distant sounds differ in the relative amounts of direct and reflected sound. There 'is also an important difference in the spectral balance-that is, the relative prominence of the various frequency ranges. At first sight, it may not be clear why an increase in the distance should be associated to changes in balance. But these changes do.occur and, in fact, are substantial. The reason for and extent of these shifts in spectral balance are what I want to explain here as well I can.

As long as we are restricting our attention to spectral balance specifically, there is useful information available from acoustical theory and measurements. The numerical data agree well with. the results of listening in this case (something that doesn't always happen!).

To set up our measurement picture, imagine a sound source on stage radiating sound with the same intensity at all the audible frequencies. The basic question is: What would be the intensity at various frequencies at audience locations in the hall? This amounts to asking how the sound heard by the audience differs from flat frequency response relative to sources on stage. If we can answer this basic question, then we will have a fairly good idea of what transformations will occur in musical sound from stage to audience, as far as frequency balance is concerned. The issue is complicated by the fact that most musical instruments beam the higher frequencies, but we shall take that up a bit later on.

The most natural and convincing approach to the basic question (other than just listening) is the empirical method of putting a known sound source on stage and applying a spectrum analyzer to the sound at various locations around the hall. The results of such measurements are given for a number of halls in Halls for Music Performance, Two Decades of Experience: 1962-1982 (R. Talaske, ef al. editors, published by American Institute of Physics for the Acoustical Society of America, 1982). The data given there consist of graphs of spec trum analyses from 125 Hz (or, on occasion, 8000 Hz) of the response to a sound source with a steady state, uniform dispersion standardized essentially flat frequency response. (The minor deviations from flat power response of the source will not be important to us, since we are only going to be considering the general picture.)

The graphs show considerable variety from hall to hall in bass and mid-bass response, with the halls that are regarded as desirable for orchestral performances having considerable bass to mid-bass warmth. A less desirable feature of many halls is a slight 250 Hz depression, apparently caused by absorption arising from the seating pattern. In the midrange above 250 Hz up to the 2-4 kHz region, most of the halls are essentially flat. But around 4000 Hz, and sometimes as low as 2000 Hz, virtually every hall begins a rapid roll-off at even quite close-up audience locations. By 8000 Hz, there is typically a 7 to 10 dB dropoff from midrange level. The graphs are not given beyond 8 kHz; but from theoretical considerations, the roll-off at higher frequencies would be expected to be even greater.

Frequency Response of Two Concert Halls:
Davies Hall, San Francisco and Orchestra Hall , Chicago
Note the high frequency roll-off, largely unaffected by changes in the hall acoustics, whether via movable devices ( Davies) or architectural modifications (Chicago).

Before you decide to disconnect your tweeters, we need to consider carefully what these data mean. The sound source used for these experiments is a steady state source, sustained like a held note or chord. Measured or heard response from a source in a hall is always a combination of the direct, unreflected sound straight from source to you and the sound from subsequent reflections off the room boundaries-the walls, floor, ceiling. For a steady state source, the direct sound has constant volume, of course. Moreover, soon after the source begins radiating, the reflected sound builds up to a volume that also remains constant. The explanation of the measurements I have described is that this reflected sound, the reverberant soundfield, as it is called, has very little high frequency content. As the reverberant soundfield accounts for much of the total sound, a high frequency rolloff is expected.

Transient sounds are much different. The steeply rising transient wavefront is received directly first, with the high frequency content unattenuated by reflection. A sharp transient always contains extensive high frequency content. You can verify this fact by covering your tweeters and noting how transients become dulled. A transient sound does not build up a true, constant reverberant soundfield. There is not time for it to do so. And the boundary reflections that do occur will not blur the feeling of sharpness, of hard attack, nor will they in general confuse the sense of where the transient came from. The brain notes where the first wavefront came from and does not let itself be distracted by the reflections (the Haas effect). This perception of transient location makes possible the precise sense of where instruments are even in a distant audience location, where the reverberant field of sustained sound predominates.

The situation with transients is one reason why putting in a fast roll-off high frequency filter will not produce true concert hall sound from a record that is bright because of being too closely miked. Transients that should be sharp and clean will be dulled; and the sound will become muddy, muffled, and diffuse, even if the steady state tonal balance has been made more or less correct.

There is some loss of highs with distance even in transients, because the air itself absorbs high frequencies more than it absorbs lower ones. Below 1000 Hz, air absorption is a negligible effect. But from 1000 Hz on up, the rate of air absorption increases steadily with increasing frequency. At 1000 Hz, the air absorption is less than .25 dB per 100 feet; at 4000 Hz it is 1.2 dB per 100 feet; and at 10,000 Hz it is 4.3 dB per 100 feet. So 50 feet back, say, 4 kHz is down about .5 dB, 10 kHz is down about 2 dB, relative to 1 kHz. These amounts depend considerably on relative humidity. The figures given are for 40 percent humidity. In the winter, when indoor humidity is very low because of heating, the differential air absorption is higher, with 10 kHz down 8 dB at 100 feet when the humidity is 20 percent. 

Air Absorption of Higher Frequencies

Direct sound loses high frequencies only a fairly small amount, but the loss in the reverberant soundfield is much greater. The reason has to do with both air absorption and room boundary absorption. The rate at which the air and the room boundaries together absorb sound is usually measured by the reverberation time, with short reverberation time corresponding to high absorption and long reverberation to less absorption. By definition, the reverberation time at a fixed frequency is the time it takes a uniform soundfield in the hall at that frequency to drop 60 dB, measured from the time the source of the soundfield stops radiating. It turns out that this time does not depend on the absolute loudness of the soundfield. The choice of 60 dB is arbitrary; but the 60 dB figure seems to have been motivated by the fact that it is roughly the decibel separation between medium loud music levels and the noise floor in a reasonably quiet room.

You can get an approximate idea of a hall's reverberation time by noting the time it takes for the music to become inaudible after the players stop playing. In truly resonant spaces, such as large stone churches, the sound remains audible for a surprisingly long time.

When a single reverberation time is specified for a hall, it is usually for 500 or 1000 Hz or some average over this range. For a hall regarded as good for symphonic music, this reverberation time will usually be around two seconds with the hall occupied, though other acoustic characteristics of the hall can make shorter or longer times acceptable. The reverberation times at other frequencies are also important, and these other reverberation times generally differ substantially from the midrange time.

We have noted already that air absorbs sound rather little at frequencies below 1000 Hz, so absorption by the room boundaries becomes the main factor in determining the reverberation time for frequencies below 1000 Hz. The more absorbent the materials of which the hall is constructed, the shorter the reverberation time. The volume of the hall also plays a role because, in a large hall, the sound takes longer to go from one boundary, and hence one absorption, to another. This is one of the reasons that good halls have high ceilings: For a fixed seating area, a higher ceiling makes the volume larger and the reverberation time suitably long. Of course, this process can be carried too far; the Gothic cathedrals, with their vaulted ceilings, have reverberation times that are too long for satisfactory symphonic music listening, though the reverberation is suitable for organ music, antiphonal brass, and the like.

Most of the materials used in concert hall construction absorb bass frequencies less than midrange frequencies, and the good symphonic halls often have bass reverberation times of more than three seconds. The resulting warmth is usually regarded as a virtue. This regard is not just a case of making a virtue of necessity. It is possible to make a hollow-walled hall that would have short bass reverberation time, but such halls usually sound awful.

The air absorption of sound at high frequencies is so large that the high frequency reverberation times are short even if there is no absorption whatever by the room boundaries. For instance, at 40 percent humidity (humidity matters again), the maximum possible reverberation time for 10 kHz sound is 1.2 seconds. At 20 percent humidity, the maximum possible at 10 kHz is only .6 seconds. The maximum possible values above 10 kHz decrease steadily with increasing frequency. In practice, concert halls are designed to be nearly as "live" as can be arranged, that is, to have walls and ceiling that do not absorb high frequencies too strongly, so that the reverberation times are not too far from the maximum possible values. But it remains an inevitable fact that air absorption makes the high frequency reverberation times lower than the midrange ones, if the midrange times resemble the commonly accepted ideal value of around two seconds.

Now you see why concert hall response is at least potentially flat across the midrange, but tends to roll off as soon as air absorption becomes a significant factor, for around 4 kHz up. There just is not much higher frequency energy around in the reverberant soundfield because, as the sound bounces around the hall, the air soaks up the highs even if the walls don't.

Reverberation Times versus Frequency for
Davies Hall, San Francisco and Orchestra Hall ,Chicago .
(from Halls for Music Performance )

Naturally, this does not mean that there are no highs in the concert hall. In a close seat with direct sound, over half the total, the highs would be down only a few decibels since the highs in the direct sound would have suffered little distance attenuation; and the reverberant field, with its low content in high frequencies, would be only the smaller part of the perceived sound. But in a more remote seat, where the reverberant soundfield predominates, the suppression of highs would be much greater, as we saw in the experimental data. This all applies only to sustained, not transient, sound, as discussed.

How far away from the performers do you have to be for the reverberant soundfield to be at least half the sound you hear (in sustained sound)? Not very far. The precise answer depends on the hall; but usually at anywhere beyond around 20 feet, the reverberant sound predominates in a typical hall. "This would mean that [in a usual concert hall) only for the musicians and the conductor (and the microphones placed in their vicinity) is the direct sound not overpowered by the statistical (reverberant) sound".

Many musical instruments beam their high frequencies to a considerable extent. Trumpets, for instance, are much brighter on axis than far off axis. The effect of such beaming is to increase the proportion of direct compared to reverberant sound at the more distant audience locations. Since direct sound contains more high frequencies, beaming brightens the sound at distant locations and prevents excessive dullness. The effect is only partial, however. Distance attenuates even directly radiated highs, and the room sound with its lack of highs also continues to account for much of what is heard at a distance.

The beaming is directed up and out, toward the balconies. At a close-in seat in the orchestra, the highs are mostly being beamed over your head. An approximate uniformity of brightness is obtained because the distant balcony seats receive the beaming, brightening what would otherwise be too dull because of the predominance of reverberant sound, while the close-in seats, which receive more of the direct sound, are off the beaming axis, reducing what would otherwise be too much brightness.

The most important thing to note, however, is that no audience location can possibly receive anything like as much high frequency energy as a microphone that is both close to the performer and on the beaming axis, i. e., close to, in front, and up fairly high where microphones are in fact typically positioned!

Because, for the bass-midrange, smaller room volume makes for shorter reverberation times, living rooms have short bass-midrange reverberation times, typically on the order of half a second. High frequency reverberation times are also short, because they are always short on account of air absorption. Small rooms do not have the large difference between bass, midrange, and high frequency reverberation times that is typical of concert halls, where bass is often over three seconds, midrange is about two seconds, and highs are one second or less.

In a usual living room, a non-beaming source produces room sound at least as loud as direct sound at any location more than about three feet from the source. The room sound, which thus plays a large role at normal listening positions, is tailored not only by the reverberation times at different frequencies, but also by the directivity of the speakers at different frequencies. Most loudspeakers, whether by accident or design, become more directional at higher frequencies. If such a speaker has flat on-axis response, it will produce a proportionately lower total energy level in the higher frequencies because of its narrower directivity. Since the reverberant sound field treats all directions the same way, the room sound will have rolled-off high frequencies, just as it does in a concert hall. But this compensation seldom works out exactly right.

Data on directivity and the actual room response of loudspeakers are often provided in British magazines and some owner's manuals as well. While such data have perhaps rather distant relevance to choosing speakers, it is useful in getting some general idea of the effects we are discussing here, and indeed a general pattern emerges. Most speakers roll off above 10 kHz in room response, as expected from directivity considerations. On the other hand, many are relatively flat in the 4-10 kHz region. Thus, the concert hall roll-off, in fact, starts much sooner than the room-speaker roll-off for these speakers. It follows unquestionably that close-miked records will be too bright tonally if concert hall sound is the standard.

The roll-off in the 10 kHz up region cannot repair the damage done by brightness in the 4-10 kHz region. In fact, the 10-20 kHz octave, which plays a large role in transient accuracy and texture, has an effect on tonal character that is smaller than, or at least different from, the 4-10 kHz range. Too much in the top octave makes things edgy, grainy, and over-etched. Too much in the 4-10 kHz region gives music a finger-nails-on-blackboard harshness. Neither frequency range can repair disaster in the other.

These considerations offer an explanation of several otherwise seemingly inexplicable situations. For instance, how is it possible that many musicians regard 78 RPM records as truer to the sound of music than many modern LPs? Strictly in terms of midrange/treble tonal balance (and in my experience tonal balance is what most musicians listen for first and [almost] only), it might actually be true that 78s are closer to the real thing than multimiked LPs, just because technical limitations prevented 78s from having a peaky top end. Similarly, how is it possible that an inexpensive AM radio, say, can provide enough musical information to make it possible to identify singers or, for that matter, violinists? Again, the crucial identifying information is contained in the midrange, because the wide variations with audience location of the higher frequencies make these frequencies less a part of the performer's identity. You could look at this the other way around, too: Those wide variations in high frequencies are acceptable precisely because the crucial tonal information lies in the midrange. None of this is to be taken as meaning that live music does not contain extreme highs, nor that 78s and AM radios actually sound like music! As noted, hard transients contain very high frequencies indeed, far beyond 20 kHz, even at audience locations.

The relative absence of higher frequencies in the reverberant soundfield, the consequences for multi-miked recordings, and the relationship with directivity and room sound have been considered carefully by speaker designers, or by some of them at least. The Quad 63s have controlled directivity (cf. Peter Walker interview, TAS, Issue 23), as do Spendor SP-1s in perhaps a less systematic way. The Celestion SL-600 and SL-6 use an even more radical approach: They are not even flat on axis, but rather have the treble shelved a few decibels down from 2 kHz on. The designers seem to have felt that this produced more realistic sound from most records. :" It can be argued that the shelved down treble effect, even with acceptably miked recordings, approximates simply moving back in the hall a bit, and as such is consistent with preserving the tonal balance of the real sound of music."

There are recordings with concert hall correct tonal balance and ambience, or something very near at least: the Reference Recordings and Waterlily Acoustics orchestral records come to mind. Purist miking of large ensembles effectively forces relatively distant miking: You just cannot get very close to the instruments of an orchestra all at once with only two or three microphones. So, de facto, minimal microphoning tends to produce natural (I.e., fairly distant) tonal balance in the recording of orchestras.

For smaller ensembles or solo instruments, however, minimal miking, even Blumlein one-point stereo, can be far too close for concert naturalness in tonal quality. Such close-up records can be and often are exciting, but providers of audience-location sound they are not. My feeling is that a certain distance is desirable. Modern instruments, and old ones rebuilt for modern use, are made to be brilliant enough for use in large spaces.

Even chamber music, as presently performed, is intended for halls holding hundreds of people. To put the players literally in your living room, to bring the players to you rather than you to the hall, will produce an over-bearing, too brilliant sound. (Even though for chamber music, I, in fact, do like to sit in the first row or two, that is still a long way from close-miked sound.)

Though opera singers certainly sing louder than untrained voices, the human voice is the one instrument that cannot be rebuilt for increased volume and brilliance, and closemiked vocal records can sound natural if the singer relaxes to unforced voice level. As for the rest: Back off, I say. Too many audiophiles and recording engineers seem to feel that the existence of details-keys clicking, fingers striking strings-is almost synonymous with realism. Of course, the system should reproduce these details if they are on the record. But an over-abundance of such detail in a recording or in a system is an immediate tip-off that the recording is too close or that the system is hyping up the highs or mid-highs. We have been wading in deep waters here, and it would not be appropriate to draw overly doctrinaire conclusions about such complex matters, in which personal preference (among other things) plays a role. But certain conclusions seem inevitable: First, the quality of the recording is crucial. As DAW remarks : "A correctly engineered recording will sound satisfying on virtually any reasonably good playback system. Yet a poorly engineered recording will not please the careful listener on any system, regardless of quality". To this I would add specifically that no equipment can truly repair the damage inflicted by unduly close miking, since you need to keep the highs for the transients but to get rid of most of them in the steady state sound. The controlled directivity approach to speaker design may help, but in the end what is really needed is a correctly balanced record. In frequency balance, as in soundstage, there ain't no cure for the multi-mike blues.

 Variations in the bass and mid-bass, as long as they are not seriously deficient, or in the higher frequencies, as long as they are not over-prominent, tend to be consistent with differences in hall acoustics; and such variations keep things in the realm of live music. But, because good halls have flat midrange response themselves, midrange irregularities in equipment or recordings will not in general be consistent with concert hall experience. 



Regards and enjoy the MUSIC NOT DISTORTIONS,

R.