Eye Opening Experience

[ClaudeJ1]

1 hour ago, mikebse2a3 said:

 

Single 15” with vent similar to 1802. Roy said final design has same frequency response as 1802 with about 3db less max output 🙂

As one would expect from the Sd difference in the drivers with similar motor force and Xmax specs. Nice. Roy has been a busy boy when he's not Fishing!!

[parlophone1]

Funny, I was thinking of KEF LS50 for myself, but what kept me off them until now is the lack of bas.

OK, and their insensitivity too, but they sing with healthy 100W per channel and above.

Seeing LS50 with that huge horn woofer makes me rethink all over again.

LS 50s are much more common in Europe and the price is a bit lower than in USA. Very good speakers for small rooms but healthy amount of clean power is a must for them.

[Chris A]

On 11/30/2018 at 4:07 PM, Delicious2 said:

...Now if we could just discuss and reach consensus on which variables are foundational or "macro" variables:

• basic symmetrical setup of speakers in a room that can support a symmetrical sound field in relation to the primary listening position
• speaker radiation pattern control, 
• room treatment to minimize dominate bass mode excitation and provide even absorption and diffusion
• all components in the chain working together such that none are operating outside their design parameters (e.g. an amp that can't handle an impedance swing gracefully)
• other "macro" variables which we discuss and may agree are foundational such as EQing to flat response in room (at least as a starting point or home base)

then we might discuss and reach consensus on what are more "micro" variables - those that make a difference but not nearly as much and wouldn't be considered foundational.

1. It's been my experience that listening room shape and size is the basic determinant of the upper boundary of sound quality that is achievable from any setup.  The following article is the best I've seen in terms of relative room dimensions and size, figures 12-15: https://community.klipsch.com/applications/core/interface/file/attachment.php?id=79899.  Rooms without shoebox shape, such as those with vaulted or attic-style canted ceilings, or L-shaped/non-rectangular floor plans, etc., are a problem for smaller home-size listening rooms.  Non-vertical walls usually results in the complete covering of the non-vertical or horizontal surfaces with absorption to suppress the acoustic energy directed toward the listening positions (except if the canted surfaces are more than 15 feet away from the loudspeakers).  Not having consistent and smooth walls in the near field of the loudspeakers will typically create differences in timbre and apparent loudness between channels.  Much more can be written on this subject, and in fact books have been written on setting up loudspeakers in rooms and discussing room acoustics issues.
    
2. Loudspeaker pattern control (directivity horizontally and vertically) is solely a function of the loudspeaker's design.  If the manufacturer doesn't get it right, there's nothing you can do about it (other than perhaps making your own loudspeakers/horns). 
    
3. Getting the reverberation times, both early decay times (EDTs) and diffuse decay times (T20, T30 curves) to reasonable values is probably one or two most important tasks on the owner.  This means that the T20 or T30 values are typically less than 0.6 s at any point, except perhaps low frequencies below 70 Hz.
    
4. All components working together--well, besides selecting them for noise floor and sound quality (each piece), hooking up the power to the power components to the same outlet (to avoid common mode noise), the only other major concern I've found is gain structuring of the preamp/crossover/amplifiers. Issues with electrical impedance are usually tied to someone trying to use tube/valve electronics.
    
5. Other macro variables: Besides the room reverberation times mentioned above, the following I've found to be the difference between an "okay" environment and one that draws you in to listen to whole albums, etc.:
   • Getting the on-axis EQ flat:  I used to think that ±3 dB was sufficient.  I found out that the flatter you make the SPL  vs. frequency, the better and more natural the resulting sound.  Taking measurements at 1 m front the front of each loudspeaker for individual loudspeaker EQ corrections is the preferred method, in my experience.
   • Flatter phase and/or group delay response: this is mostly controlled by the crossover filters used by the loudspeaker drivers, and getting the time misalignments corrected via DSP crossover.  You won't believe the difference in sound once this is achieved.  The soundstage and impulse response (i.e., percussion) are significantly improved, and the listening involvement goes up by at least a factor of two, perhaps more.
   • Absorbing nearfield acoustic reflections close to the loudspeakers and listening positions (i.e., coffee tables, leather covered HT chairs, etc.): This opens up the soundstage and strengthens the phantom center image for stereo-only recordings.  This "quiets down" the confusing acoustic reflections so that the human hearing system can do its job of recreating the realness of the recordings without being strained or confused by the interfering nearfield reflections. 
   • The quality of the compression drivers themselves in reproducing the top-most octave (10-20 kHz).  Nothing more needs to be said on this subject.

Chris

[Chris A]

On 11/30/2018 at 4:07 PM, Delicious2 said:

...what I find myself wondering about from this discussion is :  Do our systems become closer and closer to the ideal of what is possible with reproduced sound in the home and so more alike to each other and changes in components make LESS of a difference?  OR do our systems become more and more transparent and so sensitive to smaller and smaller changes (better microscopes)?

Well, all I can say is that I forget about the loudspeakers and electronics once the above measures are controlled, and I find myself just listening to the music, not the effect of various pieces of the system.  I find that I have no need to do anything else to the setup or room, and find myself searching through my recordings--even the ones that I thought were difficult to listen to...(within reason, of course).  Most of my urge to do the "audiophilia" thing disappears.  Realizing of course that not everyone might react this way, I have to say that my background as a musician seems to take over.  The small stuff no longer seems interesting to me to pursue.  Just listening.

 

Chris

[Chris A]

On 11/30/2018 at 4:07 PM, Delicious2 said:

Maybe there is more than one ideal or a continuum what we're striving for?  A system that gets all the macro variables right may not be particularly "refined" may not be a very good "microscope".  Conversely a system that is very refined and excels in certain "audiophile" criteria may not sound particularly "live".  Then there's the whole thing already touched upon about the "wow" factor and whether a system is satisfying long term... 

I'll let you know once I stop listening to my recordings (old and new).  That might be a while...

 

Chris

[Tarheel TJ]

Chris,

    Can you expand a bit on point #3?  How does one measure reverberation times?  I have a feeling this is one area I may need to address.

[Chris A]

Sure...one of the calculations that REW does automatically when you take a measurement upsweep (among many other calculations) is that it calculates eight different reverberation time curves vs. frequency:

early decay time, (EDT),

reverberation time to -20 dB (T20),

the same thing for -30 dB (T30),

some sort of combined measure (Topt),

"Clarity" for speech (C50)

"Clarity" for music (C80)

"Definition" for speech (D50)

"Center Time" or time to the center of gravity of the squared impulse response (TS)

 

Here is a plot of five of those measurement calculations that I did in my listening room a while back:

 

I usually look at the T30 curve and the EDT curve (time to -30 dB and the decay time looking only at the -10 dB of decay--which is often steeper than the other portions of the decay vs. time.  The Topt curve seems to be a bit noisier for my room.)   As you can see, the T20, T30, and Topt curves basically lay on top of each other, even though, intuitively, the T30 curve should be higher than the T20 curve because it takes longer to get to -30 dB than -20.  Most of the reason for this is that there is estimation taking place based on slope of decay curves, therefore there is some error in measurement. 

 

T60 (otherwise called RT60) is the time it takes to get to -60 dB from the original sweep direct arrival amplitude.  Many small listening rooms don't ever get there due to ambient noise, so RT60 is usually reserved for large concert halls and auditoriums, and is not used for home-sized listening rooms. 

 

One of the characteristics that you see is a dip from 100-250 Hz, indicating that there is steeper decay rate at that frequency band.  Some of this estimated reverberation time is affected by the "room mode" region, where the acoustic energy is channeled into areas of the room that are accentuated with acoustic energy, and other regions of the room that are relatively devoid of acoustic energy.  The other reason for this  faster decay rate is the use of bass traps in the room.  It turns out that the "boominess" of any room is centered on 100-200 Hz, and if you adsorb that acoustic energy at a slightly greater rate, it sounds cleaner and more defined. A little of this is good, too much is not good.  So I tune my bass traps by sliding them left or right across the room corner to decrease the volume of air trapped between the corner walls and the flat bass trap to decrease its effectiveness to the right level.  (My bass traps are in the near field on top of my TH subs--which are just behind the Jubilee bass bins, so I need to moderate the bass trap effectiveness somewhat because of their close location.  If I had two corners in the opposite end of the room that I could use, I might have tried those locations for the bass traps, but that option is not available to me in my room.)

 

One of the other characteristics that you see is a relative rise in the reverberation times at 800-2500 Hz.  I see this in many other rooms, too, and this is the majority reason why I believe that people find a large difference using diffusion panels.  I may try adding even more diffusion to my room in the future, but currently I have relatively good T20 and T30 values in that frequency band that are generally below about 0.45 seconds (450 ms). This is "good enough" for my needs.  You may choose to control this up or down based on personal taste and the type of setup that you have.  My setup is a compromise between a 5.1 array and a stereo array, so I'm good with the current levels.  If you look at a picture of my listening room (under my profile, and click on the "About Me" tab), you'll see a lot of absorption pads almost casually thrown about the front of the room just next to the front three loudspeakers (L, C, R).  This is a choice that I made to quiet down the nearfield reflections in order to increase clarity and strengthen the phantom center image in stereo mode.  This is in opposition to the sense of envelopment of the very front of the room, and it is a choice that I made based on trial-and-error subjective listening trials over some time (months of listening).  Your choices may be different, especially if you're only using two loudspeakers instead of 7 (including subwoofers).

 

Another characteristic that you see ins the steeper downward trend in the EDT curve (early decay time).  This is evidence that I'm using those absorption pads at the front of my 40' long listening room.  It turns out that I don't need those early reflections from the front wall close to the Jubilees/K-402-MEH, so the absorption pads help me to clean up the stereo and multichannel imaging/decay times.  If your room is not as deep, then you'd probably need the front of room reflections more than I do, so you might use less absorption.  There are guys on this forum running stereo/HT setups that are absorbing down to 0.2-0.3 s, and they report that they love it because it reveals the source music quality even more clearly and with more definition.  That would be too dead for my tastes. 

 

Chris

[Delicious2]

2 hours ago, Chris A said:

Well, all I can say is that I forget about the loudspeakers and electronics once the above measures are controlled, and I find myself just listening to the music, not the effect of various pieces of the system.  I find that I have no need to do anything else to the setup or room, and find myself searching through my recordings--even the ones that I thought were difficult to listen to...(within reason, of course).  Most of my urge to do the "audiophilia" thing disappears.  Realizing of course that not everyone might react this way, I have to say that my background as a musician seems to take over.  The small stuff no longer seems interesting to me to pursue.  Just listening.

 

Chris

Thanks for expanding on the macro variables idea Chris.  Your detailed answer grounded in measurement and experience helps me realize both how far I've come in recent months and some areas ( "Absorbing nearfield acoustic reflections close to the loudspeakers and listening positions") where I could do better.

I'm already starting to experience that relaxation of audiophilia nervosa - digging into my music collection as never before!

 

https://www.urbandictionary.com/define.php?term=audiophilia nervosa

 

I'd like to have that eye opening/ear opening experience of the OP with a megadollar system I couldn't possibly afford just as a point of reference.  At the moment I'm really enjoying my recent break-through into a new dimension.

[Chris A]

There is a more advanced discussion that I would like to share, but it turns out that an in-depth discussion would likely incite more emotional responses (which is actually not good) and perhaps misunderstandings.  Generally, that discussion involves the difference in sound that many audiophiles are trying to create, but that sound is not realistic.  I base these observations on the 15K+ music tracks that I've demastered and have long since realized that most people don't know that what they're listening to on stereo recordings is already altered from a natural sound quality into something that I'd call "audiophilia sound".  This includes significantly boosted highs and attenuated lows below 50-100 Hz (among other "enhancements" to their frequency response and dynamics)--to the point that a double bass and kick drum no longer sound anything like the real thing, and even jazz electric bass [5- and 6-string] is robbed of its visceral impact and low bass presence (in fact, double basses can no longer be distinguished from cellos).  If you want to hear the difference, simply find good 5.1 recordings of jazz groups and listen to them vis-à-vis typical stereo-only album mastered tracks. 

 

Why do I bring this up, rather than the "micro-variables" that you're talking about?  Because once you realize that a very high percentage of those micro-variables are almost completely dependent on this altered stereo music gestalt ideal--and it isn't based on how the music sounds in real life...you realize that you have no standard at all to base these micro-variables as improvements or otherwise. 

 

I think that many people coming over for a listen to my system are subtly disappointed because I don't set up my system or play stereo music tracks this way (the "audiophilia" way which is VERY audible to my ears now). Instead I choose a much more natural sound (timbre) that doesn't overemphasize the highs and under-emphasize the very lowest frequencies, etc.  Most people that have spent most of their lives not hearing the real thing will usually want that overhyped "audiophile sound" instead.  That's where I agree to disagree.  When you use mastering EQ to change the audio signatures of the instruments and voices to boost highs and attenuate lows, the entire timbre of these instruments and voices are no longer representative of real life.  That's the antithesis of "hi-fi".

 

Chris

[ODS123]

1 hour ago, Chris A said:

 

Why do I bring this up, rather than the "micro-variables" that you're talking about?  Because once you realize that a very high percentage of those micro-variables are almost completely dependent on this altered stereo music gestalt ideal--and it isn't based on how the music sounds in real life...you realize that you have no standard at all to base these micro-variables as improvements or otherwise. 

 

A bit tangential to your point....

 

One thing that has always struck me as sort of silly is the notion that good speakers NEVER cause listener fatigue.  You'll read posts like "I can listen all night long to XXX speakers and never feel the urge to turn it down or turn it off."  Well, as someone who has heard a lot of life music, I can tell you that REAL music played in a proper venue can absolutely cause listening fatigue after a while.  Trumpets, flutes, violins, and violas, (just to name a few) even when played by exceptional musicians, can start to grate after awhile.  So I always wonder, if speakers NEVER causes listener fatigue, then what isn't the speaker recreating that was there during the performance?

[Delicious2]

26 minutes ago, ODS123 said:

 

A bit tangential to your point....

 

One thing that has always struck me as sort of silly is the notion that good speakers NEVER cause listener fatigue.  You'll read posts like "I can listen all night long to XXX speakers and never feel the urge to turn it down or turn it off."  Well, as someone who has heard a lot of life music, I can tell you that REAL music played in a proper venue can absolutely cause listening fatigue after a while.  Trumpets, flutes, violins, and violas, (just to name a few) even when played by exceptional musicians, can start to grate after awhile.  So I always wonder, if speakers NEVER causes listener fatigue, then what isn't the speaker recreating that was there during the performance?

It has sometimes occurred to me that the best system would be like a chameleon - true to the recording and the music whether it was soothing, grating, agitating, inspiring or (insert adjective here).  Not sure I have ever encountered such a system or whether I would want to listen to it for long never mind own it given the apparent dismal state of mastering Chris describes.  It seems I could do with some ear "retraining" and so no longer crave the audio "junk food" of music mastered for earbuds and car stereos.

[Chris A]

Well...this is an interesting discussion..."how to retrain your hearing to prefer more natural listening curve".  I've found that for most music tracks, you will prefer the more balanced tracks immediately.  They will sound good at low or high SPL and every loudness level in between.

 

Other tracks, particularly those that have (for lack of a better term) "screaming electric guitar solos", you will find it more difficult to inoculate your hearing for that genre in order to accommodate a more natural listening curve.  I usually add a couple of dB of loudness tolerance during demastering to the 1-6 kHz band overall in order for these types of music tracks to retain interest, but I find that a lot of people have imprinted on these type of tracks that are very unbalanced due to heavy-handed mastering EQ, ostensibly in order to generate listener interest.  I also find that there are a couple of genres that particularly suffer from this issue: hard rock/metal and 1990s grunge. (No wonder why I never really bonded with these genres--because it basically hurts my ears.)

 

That's but one of a few interesting finds that I encountered along the way.  There are, of course, many others--so don't feel bad if you thought that type of music was, well, "hi-fi" right out of the record sleeve or jewel case.  Here's one extreme example (and please don't take this personally if you identify with this music). Here's a reverse engineered mastering EQ curve from a Smashing Pumpkins album (Siamese Dream):

 

 

After demastering, I'm not sure that anyone would really like to hear that track, I found.  The reason why I picked that album/track (that I bought just to demaster by request) was due to a comment that I got from another forum member who stated that this album "made his head hurt after a little while" but that "he really liked this music".  I found out why his head was hurting.  I can't listen to the original tracks for more than a few seconds before my hearing defense mechanisms automatically kick in.  Understand that this is an average mastering curve for the entire track--so the statistics of cumulative spectral density show this to be the most likely EQ curve used by the mixing and/or mastering guys.  Note that this is an extreme example. 

 

Chris