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Old 6th May 2014, 04:34 PM   #41
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"Loudness" compensation in playback equipment has always puzzled me since I think the ear/brain adjusts for this automatically. When I go to the Seattle Symphony does the orchestra need to "equalize" their sound when playing softly vice full force? Everything we hear in nature is not equalized depending upon absolute SPL.

Dave.
Yup. Many funny issues I've pondered over the years... without a good conclusion. For the moment, I can't think of an analogy from visual perception.

But in audiophile practice, I think it is fair to say that music at home sounds better when there's some boosting of the bass as the loudness goes down and the olde Fletcher-Munson curves (as realized by 4-pot "loundess" volume controls) fits nicely (assuming you've done the gain management needed).

Long, long ago, there was a false interpretation of the FM curves that suggested treble be boosted too. I am not sure how many people advocate a bit of treble tweaking too, regardless of the FM curves.

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Old 9th May 2014, 04:26 AM   #42
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Originally Posted by Davey View Post
"Loudness" compensation in playback equipment has always puzzled me since I think the ear/brain adjusts for this automatically. When I go to the Seattle Symphony does the orchestra need to "equalize" their sound when playing softly vice full force? Everything we hear in nature is not equalized depending upon absolute SPL.
The orchestra sounds very different at mp than they do at fff, though.

The idea of loudness correction is to get the listener to perceive the mp as mp and the fff as fff, even if the overall level is lower.
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Old 11th May 2014, 10:55 PM   #43
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bass and treble sound totally different at low levels , mids are not that afected by level
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Old 22nd May 2014, 05:41 PM   #44
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So, is this idea going anywhere?

The reason I ask is because I had the exact same idea just the other day and then I was quite happy to find that someone had already suggested it to the miniDSP team.

What I'd like is an automatic continuously variable loudness control (ACVLC!). I have no interest in using the miniDSP volume pot for this. Getting up off my butt to change a manual loudness control dial every time I change my main volume would render my pre-amp remote control useless.

If the miniDSP could simply apply a user-configurable amount of F-M curve based on the input signal level, it would be perfect. Set it and forget it. Great "loudness" response at all listening levels, without leaving your chair.

In the meantime, I simply configured EQ settings that sound excellent at my "average" listening levels. Bass is lacking at very low volume, but if the volume is that low then usually it is my wife listening and she would never notice. When I crank up the volume to much louder than normal, I just let the bass shake the house. Not ideal, but much better than tweaking the EQ every time I change the volume.

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Old 22nd May 2014, 06:07 PM   #45
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What I'd like is an automatic continuously variable loudness control (ACVLC!). I have no interest in using the miniDSP volume pot for this. Getting up off my butt to change a manual loudness control dial every time I change my main volume would render my pre-amp remote control useless.
Granted, nobody I've ever heard from seems to understand how Fletcher-Munson works in real life, myself certainly included. But for sure, what you are talking about, if I actually understand what you are suggesting, seems inapplicable.

It is easy to utter the words,".... and when the music is soft, use the 80dB curve....". But it is quite another thing to have a transistor triggered for "soft" since even the loudest moments of music have silent instances.

It is possible to have a time-weighted averaging and many circuits do things like that. But that's not part of the Fletcher-Munson curve family and I have never seen that kind of "3D or "4D" data, even though it is in-theory possible.

To make a long story short, just as "dB flat" is a non-starter for tuning your music system, likely for loudness compensation from Fletcher-Munson curves too. In both cases, we can each choose "personal corrections" that make the music sound right to us. But that's personal taste. It is comforting to be able to "quote" curves to support our actions. But it is rare to be able to rest-your-case at that point.

If it is a matter of personal taste, no need to wait for somebody else to create a Fletcher-Munson-tuned device.

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Last edited by bentoronto; 22nd May 2014 at 06:09 PM.
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Old 1st April 2015, 12:39 PM   #46
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Quote:
Originally Posted by cogitech View Post
So, is this idea going anywhere?

The reason I ask is because I had the exact same idea just the other day and then I was quite happy to find that someone had already suggested it to the miniDSP team.

What I'd like is an automatic continuously variable loudness control (ACVLC!). I have no interest in using the miniDSP volume pot for this. Getting up off my butt to change a manual loudness control dial every time I change my main volume would render my pre-amp remote control useless.
One year later, yet someone else investigating the idea (me). I think it is a really good one.

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Originally Posted by cogitech View Post
If the miniDSP could simply apply a user-configurable amount of F-M curve based on the input signal level, it would be perfect. Set it and forget it. Great "loudness" response at all listening levels, without leaving your chair.
If you mean miniDSP should track the "live" input signal level - that is not what you want. It would be totally messing your dynamics (low level passages of the same track being played with different correction than loud ones).

What you want (as pointed out before in the thread) is a way to calibrate miniDSP to your SPL level at the listening position wrt. a given volume knob reference (obviously the one from the miniDSP being the straightforward way). Calibration only needs to be done once.

Yet better - but harder - the recording level should be also taken into account, but that's only "nice to have".

Like you, I would not use miniDSP volume control, which makes things significantly more difficult (how to tell miniDSP which position the volume control knob is).
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Old 1st April 2015, 01:25 PM   #47
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Originally Posted by Davey View Post
"Loudness" compensation in playback equipment has always puzzled me since I think the ear/brain adjusts for this automatically. When I go to the Seattle Symphony does the orchestra need to "equalize" their sound when playing softly vice full force? Everything we hear in nature is not equalized depending upon absolute SPL.

Dave.
Of course the orchestra does not have to equalize soft passages vs loud ones, because they play at the reference level.

Loudness compensation is not about equalizing low vs loud passages according to an absolute SPL, but about (statically) equalizing the sound when played back at a given lower relative level compared to the original so that the relative loudness of sounds at different frequencies is preserved.

It is inherently flawed - cannot be done 100% accurately, because this would require that:

1. the original (average or reference) loudness is known, so we can pick a reference equal loudness curve and do the correction towards the lower listening equal-loudness curve. Unfortunately, this information is not readily available.

2. all equal-loudness curves be equi-distant at all frequencies - this would ensure that the correction necessary when going from say 80phon to 60 phon is the same as going from 90 phon to 70 phon.

Without No 2, one cannot statically equalize one equal loudness level to another - it would be necassary to perform level-dependent EQ, a daunting task.

However, if we look at the newer ISO curves, No.2 seems to be remarkably well fulfilled: Fletcher?Munson curves - Wikipedia, the free encyclopedia

No.1 is a tough one and does not seem solvable automatically.
However, a DSP solution can still be significantly better than no correction at all.

EDIT: there's also a No.3 difficulty and that is the recording EQ.
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Old 1st April 2015, 02:25 PM   #48
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I have abandoned the idea of using a miniDSP and I am instead going to build a passive variable loudness control circuit and integrate it into my pre-amp. The goal will be to integrate it in such a way that the volume buttons on my remote actually adjusts the loudness control. Volume will be preset manually to an appropriate level, just below "too loud".
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Old 1st April 2015, 02:59 PM   #49
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I've been contemplating "loudness compensation" for 50 years and still don't understand it. For sure, not a "real time" thing ("live") which would be as aurally offensive as compression. But then what is the time course averaging of the operational level compared to a "reference level" or when an orchestra is playing a loud passage and you are playing it back not so loud?

In practice, I find loudness compensation (of the old, 4-gang volume control sort) to be pretty nice for late-night listening. But a distortion any other time even with the best of gain management. In as much as it is helpful only when music is just background wallpaper, it just doesn't matter getting it any too precise so long as it is about right at your usual low-level loudness. Like with a lot of otherwise valid acoustic theory, Fletcher-Munson curves (and speaker enclosure sims) aren't nearly as predictive of what sounds good as you might think.

Ben
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Old 1st April 2015, 06:40 PM   #50
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Originally Posted by bentoronto View Post
I've been contemplating "loudness compensation" for 50 years and still don't understand it. For sure, not a "real time" thing ("live") which would be as aurally offensive as compression. But then what is the time course averaging of the operational level compared to a "reference level" or when an orchestra is playing a loud passage and you are playing it back not so loud?
You don't need time average, just the loudness ratio between live/playback at a reference frequency. Think about it this way:

- the original source (say a piano) plays a 1000Hz note at say loudness 60phon. According to the 60phon equal-loudness curve in the FM model, this requires 60dB SPL.

- same orchestra now plays a lower 100Hz note, equally loud: according to the 60 phon curve, at 100Hz it takes a higher SPL to generate the same loudness sensation, around 75 dB.

- now you are playing the recording of the above such that the 1000Hz note plays at 20dBSPL, 40dB lower that "live". The 1000Hz note is now 20 phon loud (corresponding to 20dB SPL at 1000Hz)

- this means that now the 100Hz note plays at 75-40 = 35dB SPL.

- however, in order to preserve the same 20 phon loudness of the 100Hz compared to 1000Hz you would need to follow the 20phon loudness curve and see that it would actually require ~45 dB for the same loudness. That is 10 dB higher than uncorrected !

Because the phon scale at ~1000 Hz roughly corresponds to the SPL scale, the correction you need when lowering the SPL from "live" levels to playback can be obtained by superimposing the 2 equal-loudness curves at 1000Hz and computing the SPL difference elswhere (see picture).

Now comes the part where you need to "scale" sounds of different loudnesses this way. Generally, as you pointed out, that would require a different correction for each loudness. But: if, in addition, the equal-loudness curves are equi-distant at all frequencies, then the absolute loudness level is no longer relevant and the difference between 2 curves which are say 10 phon apart at 1000 hz is the same no matter their absolute level. This is key for implementing a "static" equalization and, according to the ISO model, - but not to the original FM model ! - a reasonable assumption.

Unfortunately, the live vs. playback loudness ratio is content and recording dependent and cannot be trivially computed out of the recording. Secondly, the recording contais some equalization which is difficult to factor in. If you solve this problems, you can, in theory, implement accurate loudness correction.


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Like with a lot of otherwise valid acoustic theory, Fletcher-Munson curves (and speaker enclosure sims) aren't nearly as predictive of what sounds good as you might think.
Ben
That is precisely because IMHO nobody did it right so far, for the reasons I outline in the previous posts. You need to know the the original-to-target ratio and assume that the curves are equi-distant (they are in the ISO model, which is pure-tone based, but who knows how accurate that is for music ?)
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