I'm new to the forum but a Stereophile (and other rags) reader/subscriber for almost 20 years.  I still have several Mobile Fidelity D2D LPs and they really do sound superb, even on my modest turntable set up.  It's too bad they seem to have gone away.

I like many/most of us am still working on continual evolution and improvement of my listening system.  Actually, Martin Logan SL3s probably were my first good loudspeakers and with help, I did learn to position them for superb sound, although different (some things better-some things worse) from conventional cone/dynamic loudspeakers (now playing around with a pair of Sapphires but kept the SL3s).

Mark, Colo Spgs

Accurate sound is a BS term.  A distraction.  What you should REALLY be doing with your system is reverse-engineering the sound which the microphones captured live at the actual session.  That will require considerable attention to speaker placement and acoustical reflection and room properties in general.

Success is when you can realize that objective.  When you can hear the room properties produced at the original session and they pop out quite clearly even in your artificial listening area.  You'll know you nailed it when you have a three dimensional sense of the real space in which the recording was made, not some "canned" playback which takes you away from "being there" and it "being here."

Good linear frequency response is also a must have.  You want the singers and instruments to sound authentic.  Not canned.  Showing (other than the microphone choice the producer made) the signature tonal qualities which the real voice and instrument possessed. 

It might be a good idea to purchase for reference the sound colorations introduced by studio microphones, a project which Stereophile editors compiled some years back.  It's on one of the "editors choice" CDs if I recall correctly.  An SM57 does not sound like a Neumann U47.

Honest, well captured natural stereo recordings should be used as source material in order to avoid wandering around lost in the wilderness of artificial phony modern music.  With multi track modern studio manipulated recordings you won't have much in the way of "reality" with which to judge your progress.  Listen to artificial stuff AFTER you have all the natural recordings sounding like 3D. 

Get real instruments involved in your use of reference by which to judge your progress.  Wait for multi track music until the end.  Not the other way around.

The word "accuracy" seems much too small a word to describe what you should be holding up as the holy grail.  I find I can make a holographic 3D playback system only after a lot of room tuning, sometimes adding sound panels and the like.  It ain't easy...  And it will all ways take good linear full range equipment to produce a quality result.  Pick the best most trusted stuff out there and you won't go wrong.

Your post only distils what has been said in numerous other posts down to one post.

It all starts with the recording. This is why I mentioned direct to disk LP's and Groove Note CD's. IME recordings done live with a minimal number of microphones, passive mixers and no studio "massaging" sound the best. The aforementioned direct to disk LP's along with Groove Note and Maple Shade CD's are examples.

How about this. "Failing - A very difficult piece for solo string bass".

https://www.youtube.com/watch?v=9P8C6-XqaNs

A near perfect recording.

I thought "accurate sounding" was a good term since it conveys, concisely, that

the source material is not altered, hence room and components excellent. Of course the public needs to be knowledgeable. I do see your points and agree, but how to impliment your points without the consumer losing interest may be a challenge.

Cheers

Steve

Reading through the list and comments tells me what makes the audiophile world separate from the music listening world. A little sad to be honest that the audiophile industry makes up rules and theories that don't really work, instead of looking at some of the problems and figuring out how to fix them.

Here's some important things to look at.

1) If you have a speaker or system that is only producing a small sweet spot something is wrong. You should be able to walk anywhere in your room and hear a soundstage.

2) If you have a system where any part of it is too constricted the music will also be constricted. Music is vibration and if your system or speaker doesn't vibrate it is not playing the source to it's full potential. Slow rises can be tuned without killing resonance structures and harmonics vital to recreating musical notes. This by itself is one of the biggest misconceptions and down falls of high end audio.

3) If your system can not play side to side (past the speakers footprint) and front to back (well behind the speakers and in front of past where your seat is) you have something in the audio system that is squeezing the signal.

4) If your system can not play a big soundstage no matter what kind of music is being played you have a problem with the systems ability to reproduce the recorded info. You should also understand that all recordings have their own sonic signatures, and the more locked in to one sound your system is the less music you will be able to play.

5) Dynamics come with a big stage that is in tune. When a system is over built it can not play the dynamics of a wide selection of music. If your system is not able to play all recordings you should take this as a sign that something is not right with the way your system is hosting and carrying the audio language.

6) Your room is your speaker and if out of tune with your speakers mechanically and acoustically it will not create an even balance of air pressure. We need to remember when in an enclosed space it's more about pressure than waves.

One statement that I thought was a little funny was the idea that we are not going to admit it when our systems are not doing what they should. I don't know about others but I'm in this to hear the music, all the music, and have no desire to play something that is not doing everything I want it to.  Some of the accepted theories in high end audio if you explore them as we and many others have are myths and we have made legends of these misguidings. Instead of making the corrections to our thinking we would rather hang on to them tightly even though they not only don't make sense in the music world but cause many to live with systems that can't play a wide playlist. I can't imagine having an audio system that only sounds good when I sit in one tiny sweetspot. If you have an audio system that gets thin or shrinks when you move out of your "sweetspot" something is drastically wrong. Your room (most rooms) is certainly able to play music throughout correct? If so and your system doesn't do the same there is something wrong. The something is this. Your system is not letting the dynamics of the entire signal to make it's way through the audio chain, plain and simple.

An enjoyable read.
Well done, Mr(?) BRuggles.

BRuggles wrote:

"Accuracy" is the right word for the job. It refers to how close a "shot" gets to its "target," as opposed to repeatability - which is precision. There are many elements to sound, but it all comes down to the original vibrations. They are converted to electrical voltage oscillations courtesy of a transducer - the microphone. The recording device generally has other signal-altering devices upstream like preamps, transformers, and perhaps effects like compression/limiting, reverb, etc. Digital must be converted via a analog-to-digital converter, and that relies heavily on clock accuracy, encoding, more components on the signal path, etc. Analog is GENERALLY sent through anothe preamp on its way to a tape head to store the voltage oscillations as magnetic polarity oscillations. Later, that will be "transduced" again back to electricity and sent to another amplification stage to drive a cutting head on a lathe. Then you start the record-making process.

Eventually, the recording format must be decoded from 1's and 0's or mechanical grooves or whatever else and made into an oscillating voltage again. Then that must be amplified and sent to a loudspeaker to convert that electrical energy to kinetic to oscillate/vibrate the air. Even with this incredibly simplified process, there are a BUNCH o opportunities for the original air vibrations to not "match" the listener side of the speaker. Slew rate of an amplifier, phase distortions across speaker cones, capacitive reactance between cables and carpet can cause minor distortions, non-linearities, and introduced harmonics. We talk about different elements of sound such as frequency reproduction, dynamics, and transient response/acceleration.

Good Lord, talk about a harangue. A complete description of HOW recordings are made only demonstrates my point.

What I said was that the emphasis on technobabble starts with the use of unhelpful words such as "accurate."

I said to focus on "reverse engineering the sound captured on the recording." Then and only then will you get close to hearing what happened at the session.

By focusing on technical parameters such as "how accurate is my stuff" you stop looking at the goal and get caught up in the pieces instead of the whole.

The goal is to hear the recording. For example I once heard an ancient mechanical victrola play Caruso as I stood at the end of a huge ten foot acoustic horn. Caruso recorded by yelling directly into a ten foot horn onto a needle cutting a disc.

I on the other end heard Caruso as the mechanical needle vibrated those exact same sounds backwards to my ear---no electronics at all.

The impression from this uber simple piece of reverse engineering was that I was standing directly in front of Caruso as he yelled at me (OK-sang...but it was OPERA and REAL LOUD).

And it was exactly like Caruso was "alive" AND STANDING RIGHT THERE AT THE OTHER END OF THE HORN.

THAT is what I mean by reverse engineering. Was Caruso "accurate" when I heard him through the horn??? Hell no.

The sound was missing bass and treble. And was scratchy. And not at all "hi-fi."

But dude---IT WAS ALIVE. And reverse engineering did it.

Get it?

The way to reverse engineer playback when using electronic gear instead of mechanical involves attention to the room, the phase of the signal, the timbre of sounds, the musicality and "flow" and the realism of the components in general.

Being "close" to the target by "being close" is what you are calling accuracy. As that (in logic) is called a tautology it may also be added that it does not ADVANCE a discussion of how to be successful in this hobby.

Tautologies have a habit of doing that.

I can't tell you how many times I have heard manufacturers systems on demo and thought the gear sounded exceptionally exquisite however it also sounded totally WRONG. And what was the manufacturer crowing about??

How "accurate" his sound was, of course...

If you can get the original signal to appear in three dimensional correct space as an inversely manufactured duplicate of the original except displayed on "YOUR" end of the chain---Wow.

THAT IS NOT ACCURATE---THAT IS ALIVE.

BRuggles wrote:

What I am wondering is what measurements are lacking?

Ahh but the measurements are handy---they just don't tell you how well the system is set up in the room...

You can measure a Stradivarius Violin and try to "copy" it. But good luck duplicating the way the builder chose to brace the top.

Some things have to do with how they are "built." The ingredients are great to measure (a "cup" of this, an "ounce" of that.)

But that does not make you a great "Cook."

I have been working for a while on the idea that normal stereo and mono recordings include timing information mixed into the product and then "lost" if played on a set that does not include left minus right signal information.

Anybody remember the passive Hafler decoding scheme proposed for stereo? (Hafler SQ).

The Hafler decoding scheme used two extra speakers mounted in the rear of the listening area and providing "lost" spatial information decoded from the recording using out of phase signals.

The Dahlquist DQ-10 speaker in its original iteration was famous for having two "out of phase" EXTRA midrange drivers running along with the "in-phase" speaker array. In this way the designer attempted to "out-do" the life like competition he had in mind---which was the outstanding liveliness of the original Quad stat ESL-57.

To this day some folks swear they had an out of body experience or something when listening to the Dahlquist DQ-10s. Ha. Ha.

Both ideas provide some of the "lost" spatial information that existed in the original signal.

A co-incident mike technique used for a "true stereo" recording, for example, will LOSE some of the left minus right transfer timing information if played back on a "correctly set up" stereo speaker system.

Why? Because the mics actually heard both left and right AND left MINUS right. Just like your head heard it "live."

So for years I have been playing around with extra speakers running the signal under the main mix and adding some of the lost spatial information.

The result is better playback of the sound the mics heard.

'There are more things in Heaven and Earth, Horatio, than are dreamt of in your philosophy.'