Yes , I think a variable loudness software control would be something good , something that can be engineered inside the active loudspeaker in a dsp , offering a good balance in the sound regardless of volume settings.
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Some comments from Floyd Toole about room curve targets, room EQ and more
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..very poorly executed in many cases, with unacceptable detoriation in soundquality if used. But it would be a good idea, if it could be done without soundquality issues.
I haven't seen any proper analysis of the different loudness systems in use today...
The prominent ones are:
Audyssey Dynamic EQ (Denon and Marantz only)
Dolby Loudness (Bundled wherever you find Dolby Surround decoding)
THX Loudness (Most THX licenced AVR's)
Other?
The thing with loudness, is that the first prerequisite is to know what the SPL is at the MLP - ie: the system has to be calibrated - which currently is almost exclusive to AVR's!
tuga
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It would have to be very clever.
Image, for example in a symphony, a brash and loud Allegro con gusto first movement followed by a very soft and dynamically-flat Adagio.
How would the variable-loudness code recognise that no change in gain or QE should be applied to the Adagio?
Maybe it should go in the gimmicks bin...
Yes, but I think it would be an improvement in highfidelity if done right - maybe as important as room correction programes ?
But I guess that a variable loudness control is not as good salespoint than a room correction program.
Or go back to old school loudness dials - like many integrated's had in the 80's!
tuga
Major Contributor
Yes, but I think it would be an improvement in highfidelity if done right - maybe as important as room correction programes ?
But I guess that a variable loudness control is not as good salespoint than a room correction program.
I would rather have a dynamic compression circuit in all digital playback equipment including car stereos, it could even come "on" as default.
This would be an easy way to end the "loudness wars".
More than frequency response "countor" I find that a lot of gear could do with lower distortion / increased "clarity" / less "veiling" at lower listening levels.
tuga
Major Contributor
Whatever it is it needs to be defeatable by the user.
From your lips to God's ear.
I've expressed this idea many times.
Personally I don't find Audyssey's DEQ "unacceptable" and normally have it operational.
To me any negative effect is worth the tradeoff of having a "more correct" sounding bass level at low listening levels.
My main complaint is the decision they've made to change channel balance and raise rear channel levels at low volumes.
Since I rarely listen at high levels anymore, I tame the post calibration levels in the rear by about 2db.
If I do go loud, I can always just click DEQ Off.
No it's not perfect but the best I've found in 50 some years.
Human hearing tests justifies it's use.
JMHO
YMMV
This hasn't been my experience. Audyssey DEQ is a godsend.
Loudness controls don't attempt to make every track equally loud.
So I[m not sure what you are worried about.
Let' make believe Fletcher-Munson and its updates don't measure a real thing
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That is what most complain about. It too is based on psychoacoustics...though perhaps not applicable in all cases. If you don't like, it you do what you did...bump the surrounds down a little.
If you go loud, you *are* turning DEQ increasingly off. At reference level, it is off.
Whether or not DEQ is the best way, human hearing absolutely justifies some means of addressing the equal loudness curve. There's simply no question about it.
I just find that psychoacoustic curve a slippery slope. LOL
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I don’t fully grasp the meaning of “minimum -phase” behavior of rooms and why transducers are intrinsic minimum-phase devices. All in the light of the ability of room EQ to improve the room response.Some more very helpful comments of him
In think if you go back and review my - extensive - contributions to this thread you will realize that "neutral" is not "only in anechoic chamber". What has been found over several decades of conscientious investigation and publication is:
(a) in double blind tests in normally reflective rooms (different ones over the years) listeners give the highest ratings to loudspeakers that measure essentially flat and smooth on axis, and at least smooth off axis in an anechoic chamber or functional equivalent. What they are recognizing and responding favorably to is the absence of resonances - i.e. neutrality.
(b) Room curves do not correlate as well with listener preferences, except at bass frequencies, below the about 300 400 Hz transition frequency. Adjusting loudspeakers having different flaws to match full-bandwidth room curves of highly rated loudspeakers cannot yield the same high quality sound. This is especially true if narrow-band equalization is used above the transition frequency. This fact is not to be found in the advertising literature of "room EQ" products. Guess why?
(c) Although different rooms add their own "signature" sounds, when evaluated in different rooms listeners unerringly zero in on the most neutral - resonance-free - loudspeakers. It is merely that loudspeakers in different rooms are identifiable, just as in live sound we hear the same voices and musical instruments, but in different rooms. Humans have a significant ability to separate the two - except at bass frequencies in small rooms, where the room is the dominant factor and intervention is necessary to restore broadband neutrality - Chapter 8 in the 3rd edition.
Audio professionals, in my experience, are generally not aware of these facts. However, they substantially determine what we get to listen to.
From https://www.avsforum...57.html#post58521736
As for "voicing", what I wrote early in Chapter 3 says it all:
LOUDSPEAKER “VOICING”. Music composition and arrangement involves voicing to combine various instruments, notes and chords to achieve specific timbres. Musical instruments are voiced to produce timbres that distinguish the makers. Pianos and organs are voiced in the process of tuning, to achieve a tonal quality that appeals to the tuner or that is more appropriate to the musical repertoire. This is all very well, but what has it to do with loudspeakers that are expected to accurately reproduce those tones and timbres?
It shouldn’t be necessary if the circle of confusion did not exist, and all monitor and reproducing loudspeakers were “neutral” in their timbres. However that is not the case, and so the final stage in loudspeaker development often involves a “voicing” session in which the tonal balance is manipulated to achieve what is hoped to be a satisfactory compromise for a selection of recordings expected to be played by the target audience. There are the “everybody loves (too much) bass” voices, the time-tested boom and tizz “happy-face” voices, the “slightly depressed upper-midrange voices” (compensating for overly bright close-miked recordings, and strident string tone in some classical recordings), the daringly honest “tell it as it is” neutral voices, and so on. It is a guessing game, and some people are better at it than others. It is these spectral/timbral tendencies that, consciously or unconsciously, become the signature sounds of certain brands. Until the circle of confusion is eliminated, the guessing game will continue, to the everlasting gratitude of product reviewers, and to the frustration of critical listeners. It is important for consumers to realize that it is not a crime to use tone controls. Instead, it is an intelligent and practical way to compensate for inevitable variations in recordings, i.e. to “revoice” the reproduction if and when necessary. At the present time no loudspeaker can sound perfectly balanced for all recordings.
From https://www.avsforum...57.html#post58522906
To Resonate or not to Resonate, That is the Question?
Apologies to Shakespeare . . .
This discussion has drifted into an area of literal interpretations of classical definitions with some semantics thrown in. If there is a shallow hump in a frequency response, in literal terms it is a very low-Q resonance, implying a mechanical, electrical or acoustical system with a "favored" frequency range. In a physical system as complex as a loudspeaker it may sometimes be difficult to decide what is happening. Crossovers are equalizers, by any other name, that interact with transducers having inherently non-flat tendencies - the result is a combination of both electrical and mechanical elements. Equalizers can be resonators just as surely as acoustical cavities, enclosure panels and cone breakup. So a frequency response feature may be partly mechanical and partly electrical , but the end result can be that of a resonance having Q. Achieving a desirable flat on-axis sound using passive or active networks can result in non-flat off-axis behavior because transducers have frequency-dependent directivity. In a room the result is that even with flat direct sound, the early reflected and later reflected sounds may exhibit emphasis over a range of frequencies that could forgivably be interpreted as a low-Q resonance.
As discussed many times in this thread, transducers are inherently minimum-phase devices, so electrical EQ can modify the performance of mechanical resonances - a huge advantage for active loudspeakers or those for which accurate anechoic data are available.
In the crossover between a 6- to 8-inch woofer and a 1-inch tweeter, a directivity mismatch at crossover is unavoidable. Above crossover, the tweeter has much wider dispersion than the woofer, so there is an energy rise over a wide frequency range. Is this a resonance? Technically not, in the dictionary definition sense. However, there is a broad hump in radiated energy, so perceptually it may appear to be so. Figure 4.13 shows such an example where even crude room curves were adequate to recognize the energy excess in an above-crossover energy excess and attenuate it. Because wide bandwidth (low-Q) phenomena are detected at very small deviations there was a clear improvement in perceived sound quality even though medium and higher-Q "real" resonances were essentially unchanged. Addressing all of the "resonances" was not surprisingly the best.
So, don't get hung up on semantics. Deviations from a linear frequency response are all describable as "resonances" if one chooses to. Broadband trends are very low-Q, narrower trends, medium Q, and so on. Even a bass tone control is an opportunity to manipulate a "resonance" - in this case the hump that develops above the low cutoff frequency which, depending on the system design will have a Q.
Narrow dips are usually the result of destructive acoustical interference and are usually audibly innocuous because they change with direction/position. Broader dips can be interpreted as anti-resonances if one chooses to, whether there is an associated frequency selective absorption process or not. Mostly not.
From https://www.avsforum...61.html#post58530612
As I said, because loudspeaker transducers are minimum-phase devices one can use electrical parametric EQ to attenuate the mechanical resonances in transducers - using anechoic data of course. So, if you add a hump to an otherwise neutral/resonance free speaker you have added a resonance. This is why it is crucial to pay attention to what "room equalizers" are doing. If they "see" a ripple in a measured curve caused by acoustical interference of direct and reflected sound, and try to flatten it, they may be adding a resonance and degrading a good loudspeaker.
From https://www.avsforum...61.html#post58530898
Yes spatial averaging smooths curves; they look better - that is one reason why it is favored. However, because these curves are typically steady-state curves little of value is learned about the speaker, and nothing is specific to the prime listening location. Spectral smoothing is another "feature" that smooths curves. Tradeoffs are not always advantages.
Spatial averages reduce the ability to be analytical about room modes/standing waves.
So, it comes down to "how much do you know about the loudspeaker - in anechoic data?" If none, the usual case, such room curve data cannot be trusted. If one knows a lot, e.g. a spinorma, one can predict the room curve with reasonable precision. The room curve, by itself, is not reliably associated with sound quality.
From https://www.avsforum...61.html#post58534080
I have one of the most powerful and expensive multichannel processors on the market, widely praised for its processing power and complicated signal processing. It is enormously flexible, clearly designed by smart people in the math and DSP categories, but equally clearly these people did not understand the acoustics and psychoacoustics of loudspeakers and rooms - my speciality. The result is that, in its self-calibration mode it does things that should not be done.
I won't go into the details of my history with this unit, but it began with a setup procedure, using their proprietary microphone, spatial averaging with weighted mic locations, and allusions to combined IIR and FIR processing promising a very special result.
It was indeed special, because the superb sound of my Revel Salon2s was clearly degraded. Measurements I made with REW disagreed with the unit's displayed result, but agreed with my ears. With help from a product specialist, manual EQ overrides were able to restore the essence of good sound.
Some subsequent fiddles have taken it to the point that I can enjoy programs, but only by overriding or disabling some of the internal processes.
I know that there are other digital equalizers with problems. All originate with clever math/DSP engineers doing things that may make academic sense, but that pay insufficient attention to the peculiarities of human perception. At professional audio gatherings I have had extended discussions/arguments with some of their engineers. It has always come down to opinion, not fact, and the opinions are inclined to enhance the customers' perceived value in the product. It is part of a mighty struggle to be different or distinctive in a product that delivers something that nowadays many people can do for themselves with off-the-shelf DSP, free measurement software and a $100 mic.
None of these processes are supported by published double-blind subjective evaluations. Tell me if I am wrong.
The universal availability of "room EQ" is now a kind of "disease" in audio. People place trust in these devices that is misplaced. As I have stated several times, when the operating manual of the "calibration" device basically states that if the customer does not like the sound from the default target curve, then change the target curve. At this point it becomes a subjectively guided tone-control exercise, not a calibration. The "circle of confusion" for whatever program used during the tweaking is now permanently installed in the system.
All that said, equalization is part of the necessary treatment of room modes in bass. There is no escape from that, but even there, something that should be simple is sometimes compromised. Chapter 8 in the 3rd edition.
Toole, F. E. (2015). “The Measurement and Calibration of Sound Reproducing Systems”, J. Audio Eng. Soc., vol. 63, pp.512-541. This is an open-access paper available to non-members atwww.aes.org
http://www.aes.org/e-lib/browse.cfm?elib=17839
From https://www.avsforum...62.html#post58538632
E-r-r-r-r, have we not been saying that the small peaks and dips in room curves are likely to be caused by non-minimum phase phenomena, most likely caused by reflections and cannot be equalized. To two ears and a brain they are innocuous spaciousness, not coloration. It is attempts to "correct" such fluctuations that lead to degradation of well designed loudspeakers. So, above the transition frequency, small details in steady-state room curves should be ignored because unless you have comprehensive anechoic data on the loudspeakers you don't know what caused them.
Setting up a system according to personal preferences in spectral balance includes the circle of confusion and therefore generalization to all programs is not possible. Depending on the shortcomings in your loudspeakers and room results can vary. Better to have easily accessible tone controls that can be instantly adapted to your personal preference - for any program.
I don't have them, and miss them, so I, like you, arrived at a compromise setting that suits some programs better than others. Funny that so many elaborate, expensive, high-end audio products have no easily accessible tone controls. I can only
From https://www.avsforum...62.html#post58544284
For perspective it is worth remembering that about 30% of the factors influencing our judgement of sound quality relate to bass performance - and this is dominated by the room, only correctable in-situ. So, ANY loudspeaker can sound better after room EQ, so long as it competently addresses the bass frequencies - this is not a guarantee, but really is not difficult for at least the prime listener.
Above the transition frequency the M2 is sufficiently good that it should not require anything beyond broadband spectral tilts to fit with preferred program material.
From https://www.avsforum...60.html#post58544590
Are there any good articles or books on this topic?
Multicore
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I would like to have a loudness eq that's adjusted according to the amplification gain, i.e. the volume knob. For room eq we need a set-up procedure anyway. So let's incorporate into that the determination of the volume setting suitable for the quiet tea and gossip with the vicar's wife and that of the maximum (the social context of which I won't describe).
So long as I don't touch the volume knob there's simply a static eq curve applied and no processing based on or applied to the signal dynamics.
So long as I don't touch the volume knob there's simply a static eq curve applied and no processing based on or applied to the signal dynamics.
Those do exist, "loudness EQ", but thinking it through that would have to be based upon you telling it where your "zero" level was which would be the point where you've based your optimal headphone (speaker) EQ. My EQ's are based around my comfortable listening levels, but I'm sure that I listen to some headphones louder than others because I've probably optimised them for somewhat different playback volumes unintentionally. I tend to turn them up & down until they sound right.
EDIT: other people will have to tell you how to implement loudness EQ as it's not something I do, but it does exist in some forms on different platforms.
Multicore
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These things don't exist in a convenient hi-fi product. That's why I described the measurement-based set up to established the user's desired range of gains. I can implement loudness normalization using my MiniDSP toys but it's not remotely convenient.
Yes, it should be implemented, and in the best possible way, in more products. In my mind the problem is that they have to decide at which point the loudness normalisation does nothing, and I'm not sure where that point is. I'm trying to imagine if there are any pitfalls in this implementation. It would play into your EQ decisions for sure. The problem is the device doesn't know what SPL you're listening to at your listening position which is the main crucial point of information it needs in order to take a perfect anechoic flat speaker to a loudness EQ'd speaker......and to be brutally honest anechoic flat speaker sounds pretty damn good at any reasonable SPL level, so it starts to become a bit moot, but I'm also sure there's room for optimisation.....but given all that, I think it really comes down to the system knowing what your SPL is at your listening position otherwise it can't hope to tune the loudness curve, that's what I'm thinking, and maybe that's why it's not that much of a common feature in hardware & software. And if you care that much about "sound", you would have already optimised your headphone & speakers at your normal listening levels anyway. Praps loudness EQ is a bit of a non starter in the real world.
Multicore
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A cheap mic like you get with an AVR would suffice. And software using voice prompts and the remote control can be used to determine the loudness range that the end user wants. Incorporate this with the setup procedure that we usually in any case need for bass management.
Then during operation there's a 0 to 10 control for how much loudness eq normalization to apply, 0 being flat no matter what and 10 being fully normalized across the loudness range determined in the setup.
Something like that could help, but it's a minefield if your speakers aren't anechoic flat to begin with & you don't know what level you're listening at. I'm kind of assuming anechoic flat sounds best at a reference 80 odd dB or something like that - so if you really did have anechoic flat speakers (a big if!), then the loudness compensation would happen around that, but then you have things like the room adding more bass if you haven't done a roomEQ, so that messes with the tonality.......so loudness compensation is realistically the absolute last thing you would do to optimise your listening, so praps that's why it's not a common feature, because people generally don't have perfectly anechoic flat speakers & generally haven't done roomEQ, so loudness optimisation is totally moot (an added complication perhaps) if you haven't done all that (now I'm thinking it through).
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