On the limits of "Hi-Fi" intent

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http://www.tonmeister.ca/wordpress/2015/06/30/bo-tech-what-is-beam-width-control/

FIR correction and tools like RePhase, Acourate, Audiolense, DRC-FIR among others give us the ability to try this. Yet I see a lot of doubt on the merits on this forum. Is this really audible?
I am convinced it is, by testing it. Should I put on a flame suit now?

Is that deviating from the art or trying to get closer to the neutral start point? I approve of anything that tries to get the system closer to ideal.

As an aside, has anyone ever tried to characterise the faults of classic studio monitors from yesteryear to get that 'in the control room' sound on pop music. When the led zep remixed were done last year it was reported they heard the squeak of John Bonham's kick drum pedal which they had not heard before, which suggests one of two things
1) they were off their gourds on drugs when they mixed it orignally
2) Monitoring has improved in the last 40 years
 
I'd be of the opinion it brings us closer. Though as with everything, handle these tools with care. Pre-ringing can be an audible side effect when care is not taken. In my opinion getting the timing right gives a more powerful midrange as well. Where bass is supporting the mid tones. The bass is felt more than heard. Clean and immediate.

As an avid Led Zeppelin fan I know that squeak. I loved the fact that I could hear it on the new remasters. I hadn't read about it, it was something I noticed when listening. Fun to see it confirmed.
 
Another question that might fit this discussion: what about time coherency?

Yet I see a lot of doubt on the merits on this forum. Is this really audible?
I am convinced it is, by testing it. Should I put on a flame suit now?

I used to believe that phase was not relevant, there were tests that showed it wasn't, and the ear itself is not phase correct above about 500 Hz., so how could it detect phase.

Then I read a paper by Dr. Griesinger on how the ear could detect phase through amplitude modulation detection and he concluded that this effect was important from about 700 - 7000 Hz. After that it made sense to me, although I have to say that the "jury is still out".

Curiously I checked my own speakers for phase in this critical region and found that in getting the response right through the crossover the phase change over the bandwidth of 700 - 7000 Hz was very small < 90 degrees total. I have not looked at phase for most of my life so I am not sure how many speakers do this.

My point is that maybe it is significant, maybe not. People like Toole discount it as he does with nonlinear distortion. Like I say, the jury is still out on the perception of phase and group delay.
 
This is a real effect that you can verify for yourself. You might also read the paper I linked in the thread and some of the references in that paper. It's well documented. More important with dialog than music, but real none the less.


You did, but chose to ignore it. I even posted some examples.

Pano - I never denied that the effect was there, that would be pointless. My point is that I questioned why it wouldn't be corrected in the mix, to which you answered "I don't know." This is not strong support for making this change and maybe correcting what has already been corrected.
 
My point is that it takes more than just "It sounds better to me!" to be a valid support for altering the art.

Ah, but that's not the point I was making. I'm saying as designers we have to allow latitude that less "accurate" (technically) on playback may actually sound more "accurate" (not pleasing, accurate) to individuals and that the differences are individualized

One example. A classical lover year in and out may have season tickets to the symphony in the balcony front row. That typically has much more robust upper/mid bass spectral balance than listening 12 rows back main floor. For that listener, a playback system that provides that will be more "accurate", even if it's not what the producer on any particular disc intended.

This premise has nothing to do with what is more "pleasing" or not.
 
Then I read a paper by Dr. Griesinger on how the ear could detect phase through amplitude modulation detection and he concluded that this effect was important from about 700 - 7000 Hz. After that it made sense to me, although I have to say that the "jury is still out".

I've posted many many times over many years (here and elsewhere) that phase does matter and affects minimum detectable distortion levels. This comes directly from studies in the JASA and other audition journals, completely unrelated to the "audio" field.
 
I've been on this quest for a while and posted some research I found on it a few years ago:
http://www.diyaudio.com/forums/multi-way/243885-speaker-clarity-darkness-please-chime-7.html#post3666873
With example pulses of Dunlavy speakers:
99DAL4fig3.jpg

As Dunlavy was a proponent to time coherency. He achieved it with well thought out speakers and crossover.

Fast forward to this year gives me a DSP corrected impulse from my own speakers like this:
impulse25.jpg

Recorded at the listening position. In a living room with a few damping panels placed at early reflection points.
Group delay plot is almost flat to 30 Hz (2.5 ms group delay at 30 Hz).
Speakers response goes to 17 Hz -3 dB point.
 
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The whole market is built up on a mix of hopefully correct graphs, figures and peoples worries not to be able to correctly be presented what the artist wished to share. Now, If that would be something to strive for we would need standards that if chosen would automatically configure our rigs to conform. Sure, I would vote for an override button. In the digital world of today it would be easy to envelope the audiostream into an package with metadata that describe to your gear about the artists intentions which would be the default sound of the art. The override would be that you can define your preferences when it comes to room settings and tonal charasteristics. There will probably be much more of this when economics allowes companies to search for new frontiers to explote in some years. Now everyone tries to make us believe that 24/96 is more than we will ever need...

Regards
 
The whole market is built up on a mix of hopefully correct graphs, figures and peoples worries not to be able to correctly be presented what the artist wished to share.

I would like to see more graphs actually. In the high end world of HiFi that part seems to be missing a lot. Plenty of claims, not a lot of info to back it up.

Even many on this forum still wave away the graphs and measurements and claim it doesn't capture our listening experience. It might or it might not to a full extend but we can still learn a lot from it. It's just not that easy to interpret sometimes. We need to know/learn where to look first. The measurements are more linear than we are :D.
 
Ah, but that's not the point I was making. I'm saying as designers we have to allow latitude that less "accurate" (technically) on playback may actually sound more "accurate" (not pleasing, accurate) to individuals and that the differences are individualized

One example. A classical lover year in and out may have season tickets to the symphony in the balcony front row. That typically has much more robust upper/mid bass spectral balance than listening 12 rows back main floor. For that listener, a playback system that provides that will be more "accurate", even if it's not what the producer on any particular disc intended.

This premise has nothing to do with what is more "pleasing" or not.

As a designer, I need quantification of effects to be of any use and quantifying what you say in the first paragraph is not easy, certainly never done to my knowledge. The fact is that Toole and Olive find that accuracy in the speakers is the key to accuracy of the illusion. I find that as well. Your contradiction to this does not allow for a conscious decision about design changes.

I agree that with the "you are there" allusion for a concert hall that some deviation might make for a more "realistic" specific implementation, but one cannot generalize that to say that all speakers should be designed differently as a result. There is always going to be a tradeoff between "imaging" and "spaciousness" and the implementer must be conscious of this. I, personally, will never do something that degrades imaging, even if it does enhance spaciousness. Others may have the opposite belief. I also find that accuracy is the best overall choice for a playback system even if there are a few specific issues that could benefit from a different approach.
 
I've posted many many times over many years (here and elsewhere) that phase does matter and affects minimum detectable distortion levels. This comes directly from studies in the JASA and other audition journals, completely unrelated to the "audio" field.

The prevailing belief about this when I was starting out was that phase was not audible. Paul Klipsch made a big issue of this. I have only recently come around to the believe that it could be important because prior to reading the Griesinger paper I could not see a mechanism in the ear that could detect phase. Even now, Toole and Olive discount it, meaning that it cannot be a dominate effect, but significance is TBD.
 
Just my own experience: I have made numerous “field” recordings of the small orchestra I used to work with, typically two channel ORTF placed about 6 ft. above and 6-10 ft. behind the conductor. I also have a number of “house mic” recordings of the same orchestra in different halls, some spaced pair somewhere between the proscenium and first row, or in one case a Decca Tree just above the conductor. These placements in all cases are over-bright, and impart a harsher sound to the strings than what is heard back in the hall . . . very much that early “DG sound”, but also (with a relatively moderate high end roll off) reasonably accurate to the sound in “front row” seating (as is the “stereo perspective”). Those recordings, while “objectively accurate”, are when played in my listening room (16x24, with relatively “normal” living room acoustics”) way too “bright”, and I correct with a “house curve” to get to something approximating the “hall sound” of the original performance venue. “Objectively flat” from beginning to end just sounds wrong.

“Spatial perspective” is also “wrong” with these recordings, giving more “separation” and easier “localization” than one actually hears out in the hall. I find that full-range dipoles provide some degree of correction for this in my listening room, and produce a more “lifelike” acoustic image, especially with the ORTF recordings. It’s not “perfect”, but it’s pretty good.

What one finds in “commercial” recordings is a mixed bag. In general what I hear is that the less “processing” is done in production the better the recording sounds in my listening room . . . rather obviously because my system is “tuned” for “objectively flat” source material. A reasonable amount of correction for microphone placement (tempering those shrill strings) is best done at the source. Any “correction” for the acoustics of the “average listening room” is bound to be wrong, and to mess with the room corrections that I apply. “Studio mix” recordings are all over the ballpark, but the well done ones can be excellent. Movies are (almost) uniformly good (there’s the benefit of “standards”), since they assume that “house correction” will be done “at the house” (where it should be).

My takeaway is that recordings should be “objectively accurate”, but that includes appropriate compensation for microphone response and placement. Speakers, on the other hand, should be “voiced” to suit the listening room (almost certainly NOT “objectively flat”) so as to produce a “subjectively accurate” replication of the original performance when presented with an “objectively accurate” recording. “Studio produced” recordings should take their que from the movie industry and be mixed to sound the same over a range of properly set up reproduction environments.

And that’s where I take (some) exception to Earl’s position (as I understand it). My listening room is not voiced “flat”. It is not “objectively accurate”. It is, however, voiced to sound “right” with “objectively accurate” recordings . . .
 
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J Those recordings, while “objectively accurate”, are when played in my listening room (16x24, with relatively “normal” living room acoustics”) way too “bright”,.

In the case where the recording has not been correctly balanced to the FR that you would here if there, then I don't see correction being an issue, but would want to understand more WHY the producer didn't apply any EQ in the production process.

This is where I always used to like record reviews from people who knew the hall it was recorded in as they could comment on how accurately the balance and ambience was maintained. Reviewers like JGH in US and John Crabbe in UK were particularly good for that. Gramophone has completely given up on that aspect these days :(
 
The prevailing belief about this when I was starting out was that phase was not audible. Paul Klipsch made a big issue of this. I have only recently come around to the believe that it could be important because prior to reading the Griesinger paper I could not see a mechanism in the ear that could detect phase. Even now, Toole and Olive discount it, meaning that it cannot be a dominate effect, but significance is TBD.
If it can affect the spectrum at the innear ear, it might be audible if not masked. From the audition studies, phase affects the envelope which can cause harmonics to become temporarily unmasked

I started my academic journey in audio (formal schooling) investigating group delay audibility in the mid 80s and have read a great deal on the auditory science of this
 
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Thanks Dewardh - a very good post the distills a lot of the judgement calls that go into producing a recording.

And that’s where I take (some) exception to Earl’s position (as I understand it). My listening room is not voiced “flat”. It is not “objectively accurate”. It is, however, voiced to sound “right” with “objectively accurate” recordings . . .
Well put. That's what I've been trying to get across on this forum for years.

For me, and many others, the ultimate goal isn't a strict scientific reproduction of the recorded signal, the goal is for it to sound "right" or true to life. That's my goal, because that's what I enjoy and what gives me the most pleasure listening to music at home. I want playback to sound real and natural, the means to achieve that are secondary.

This goal is too often poo-pooed as silly and derided as "Sounds right to me" nonsense. It's implied that the goal of making playback sound right or realistic is somehow bad, sloppy, disrespectful to the art or somehow morally reprehensible.. Why? Isn't the point of it all an illusion of reality?

We are told over and over again that any deviation from a narrow definition of the signal is some sort of intellectual failure. To me that seems to ignore both the goal of illusion and the complexities of the real world. I see that - the denial of the complexities and goals of the real world - as an intellectual shortcut. And ultimately unsatisfying for the enjoyment of art.
 
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