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Should you use Fletcher-Munson loudness compensation?

Mostly agree, The Audyssey DEQ is a pretty good implementation of the F-M curve idea.
My only complaint with it (as I mentioned before) is the way they decided to mess with the rear channels level.
AFAIK it has nothing to do with the F-M curve and their reason for doing it I've never found a straight answer on?

IIRC, when asked, Chris K or other spokesperson said that is from their in-house research, which told them that audio sources behind the listener suffer more.
 
With RME ADI-2 app you control the exact amount of Bass and Treble boost as well as the volume level where it kicks in. This allows each speaker configured to have customized settings that sound the best with them rather than being limited to four presets.

? Audyssey DEQ RLO isn't for accomodating different speakers. It's for accomodating different production or output styles (different levels of average loudness). If you're playing mostly hip hop you'll use a different RLO than if you mainly listen to chamber music.
 
IIRC, when asked, Chris K or other spokesperson said that is from their in-house research, which told them that audio sources behind the listener suffer more.
Yep, I've heard something along that line myself, IMHO the practice is BS and at the minimum should be switchable.
It noticeable upsets that F / R balance, is not part of the F-M research, and I know of no one else in the industry doing something similar with multich loudness compensation.
 
Indeed, I'm waiting for them to make it switchable on/off, which seems a no-brainer.
 
Indeed, I'm waiting for them to make it switchable on/off, which seems a no-brainer.
For sure, whats the big deal. One of the coolest features of the $20 Editor app was being able to switch off that midrange dip.
Another Audyssey idea that was simply a preference thing that should have been users choice from day 1.
 
I have a brand new 11" iPad Pro. It can run Divinity: Original Sin 2 without a hitch. The app is supposed to run on phones!

I'm considering buying an Audyssey:X licence.
Divinity runs on an ipad?
 
You are wrong it's frontal presentation how ever ISO 226 used compensated hedaphones for presentation.
Read part 4.
How ever there ware concerns among which some are expressed in
And there is ISO 226 2007 revision which again did address some.
I wrote two scripts (works on any browser, no installation needed) covering both papers. The first one generates Equal Loudness Contours from 40 to 90 in steps of 5 (customizable in any text editor) according to ISO226 2003 standard.

The second script applies Steven's proposed revisions in the second paper (to my best understanding of it) and generates directly the differences from reference 40 phons. Similar target curves can also be generated from the ISO226 curves by ¦A¦/¦B¦ trace arithmetic in REW (align all curves SPL to 0dB at 1000Hz, 0 octaves to get comparable division results).

You can drag and drop both zip contents directly onto REW.

rew.jpg


The 85dB Steven curve fits perfectly as a target curve to my speakers steady state response measured at 85dB. Standard ISO curves are a bit too steep.
 

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The FM curves are interesting in that they just show how futile the idea of sound fidelity is. FM compensation sounds good, that's why the Sonos systems sound so nice I think. But look at the range of tonal compensation! At 70db, it's something like a 20db compensation from whatever our reference was.

So not only are our recordings created with who knows what microphones, with their arbitrary spatial and tonal response, but they are mixed at some unknown level, potentially different at levels at different times, each having a huge effect on the tonal balance.

Old receivers had a 'loudness' button but in 2024 I'd love to see a DAC that has two knobs, a FM compensation knob and a volume knob. Throw in a stereo width knob while we're at it.

Of course we'll never see that, DAC manufacturers are too busy chasing that next dB of snr rather than anything that makes music sound good.
 
The FM curves are interesting in that they just show how futile the idea of sound fidelity is. FM compensation sounds good, that's why the Sonos systems sound so nice I think. But look at the range of tonal compensation! At 70db, it's something like a 20db compensation from whatever our reference was.

So not only are our recordings created with who knows what microphones, with their arbitrary spatial and tonal response, but they are mixed at some unknown level, potentially different at levels at different times, each having a huge effect on the tonal balance.

Old receivers had a 'loudness' button but in 2024 I'd love to see a DAC that has two knobs, a FM compensation knob and a volume knob. Throw in a stereo width knob while we're at it.

Of course we'll never see that, DAC manufacturers are too busy chasing that next dB of snr rather than anything that makes music sound good.
Well thers software which can do it (ISO 226) without knobs of course and you can do normalisation on material side hopefully EBU R128 and for free if you are not very picky and with PEQ and convolution (EQ-APO). Or you can do it with self filters pretty good. Now days we are starting to have dongle alike pocket DAC's with full PEQ support and more than fine and audible transparent performance. Future more you have simple apps that does it all like Wavelet (Android only and paid for ISO 226) and Web based (browser) one's for producing EQ's like Auto-EQ. So it's more reachable than ever but people's habits didn't exactly change nor the most of them goes that mile even if it doesn't costs them anything. Just to mention also wasted huge processing power (like server machine a decade ago) on the palm of your hand. Just don't expect from industry to resolve it, they live from telling lies.
 
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The FM curves are interesting in that they just show how futile the idea of sound fidelity is. FM compensation sounds good, that's why the Sonos systems sound so nice I think. But look at the range of tonal compensation! At 70db, it's something like a 20db compensation from whatever our reference was.

So not only are our recordings created with who knows what microphones, with their arbitrary spatial and tonal response, but they are mixed at some unknown level, potentially different at levels at different times, each having a huge effect on the tonal balance.

Old receivers had a 'loudness' button but in 2024 I'd love to see a DAC that has two knobs, a FM compensation knob and a volume knob. Throw in a stereo width knob while we're at it.

Of course we'll never see that, DAC manufacturers are too busy chasing that next dB of snr rather than anything that makes music sound good.
The best of the 80's receivers had a loudkness knob rather than a button - they allowed you to set loudness from "0" to "max" - and it wasn't a level, but a strength of compensation filter setting - worked very well indeed.
 
but just as with modern AVR solutions like Dynamic EQ, you're at the whim of no-standards music production... so the amount of compensation needed will vary from mix to mix, genre to genre, era to era...
 
but just as with modern AVR solutions like Dynamic EQ, you're at the whim of no-standards music production... so the amount of compensation needed will vary from mix to mix, genre to genre, era to era...
Absolutely - which is why a subjective, manual setting of the reference level is required....

It definitely shouldn't be just an on/off.
 
I wrote two scripts (works on any browser, no installation needed) covering both papers. The first one generates Equal Loudness Contours from 40 to 90 in steps of 5 (customizable in any text editor) according to ISO226 2003 standard.

The second script applies Steven's proposed revisions in the second paper (to my best understanding of it) and generates directly the differences from reference 40 phons. Similar target curves can also be generated from the ISO226 curves by ¦A¦/¦B¦ trace arithmetic in REW (align all curves SPL to 0dB at 1000Hz, 0 octaves to get comparable division results).

You can drag and drop both zip contents directly onto REW.

View attachment 390945

The 85dB Steven curve fits perfectly as a target curve to my speakers steady state response measured at 85dB. Standard ISO curves are a bit too steep.
Thank you for this. I have two rookie questions. Would it be difficult to update it to the ISO226 2023 standard? And even if yet, would you still recommend the Steven curve?

Also is there an obvious way to load these into the Wavelet android app? I have tried to convert it, am pretty sure I butchered it.
 
Thank you for this. I have two rookie questions. Would it be difficult to update it to the ISO226 2023 standard? And even if yet, would you still recommend the Steven curve?

Also is there an obvious way to load these into the Wavelet android app? I have tried to convert it, am pretty sure I butchered it.
The 2023 ISO paper is not available publicly but if anyone can share it here, I can happily implement. SInce I haven't seen it, I don't know if Steven's corrections are still better or not. I am on IOS and PC WIndows and haven't heard about the Wavelet app :(
 
As a big fan of loudness, my take on this subject:

It would be very wrong to tune speakers in any way to the shape of these curves. These curves represents how we listen to the sound, not how it's played back. If a flat microphone recorded the sound and it's played back on a flat speaker, these curves are baked into the recording, with the only variable being the recording/playback level.

In my opinion, the only meaningful info that can be detracted from the curves is, what everyone knows intuitively already: That we have a harder time hearing bass and treble at lower playback volume.

With the variability in HTRF etc., it would probably even be pseudo-science to even try to extract any loudness targets from the curves as well. I would not even recommend normalized curves as this is not how your brain expects to hear music a lower levels. Most of the time if you want to turn down the volume, it's because you want to concentrate on something else (such as writing a post on ASR). Your brain expects a reduction in certain frequencies.

The purpose of loudness is simply to create a pleasant listening experience at lower volumes, while still giving your brain the appropriate signals that the volume was indeed reduced. If you want do concentrate on the music, the simple answer is turning the volume up!

Thus the right way about it is to have a speaker that plays flat at louder volume levels, and for those with DSP available apply a loudness filter at lower volumes. For example CamillaDSP has great and configurable volume dependant loudness filter available.

The way I do it is to find the loudest level I would want to listen to the speaker and using gain to lock that as max volume in CamillaDSP. Then I turn the volume down to find the lowest level where I'm still satisfied with the bass and treble. Below that level I let CamillaDSP apply a loudness filter that gradually boosts the bass and treble over the subsequent 20dB reduction in volume (a 6dB boost works well for me on two different sets of speakers). Of course, I can't prove this is scientifically correct, but it makes the music very enjoyable to listen to at lower volume where it would otherwise sound dull.
 
The above post is a bit confusing. You start with the premise that using loudness compensation is wrong, but then end saying you use loudness compensation after all?
 
The above post is a bit confusing. You start with the premise that using loudness compensation is wrong, but then end saying you use loudness compensation after all?
Perhaps I didn't expres myself clearly. No, loudness compensation is great, and as I started saying I'm a big fan.

What I adviced against is using equal loudness contours (Fletcher-Munson curves) as basis for loudness compensation.
 
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What I adviced against is using equal loudness contours (Fletcher-Munson curves) as basis for loudness compensation.
I see. So if I'm reading it right now, you're saying you apply a gradually increasing boost to the bass and treble that maxes out at 6dB once you hit a certain volume level which you decide upon by ear?
 
I see. So if I'm reading it right now, you're saying you apply a gradually increasing boost to the bass and treble that maxes out at 6dB once you hit a certain volume level which you decide upon by ear?
That's what works for me, yes. But it's subjective. What I would advice is to simply adjust it to your liking. If you're "critically listening" (not a fan of that term), i.e. concentrating on the music and want an accurate representation, you're likely to listen at a louder level anyway.

You can try to make a filter like described by solderdude on page 2:
E.g. with the values in the pic attached and try to listen to it at lower volume.

It sounds really really bad. It's muddy, dark, and lacks treble. There is no clarity. Vocals are mumbled.

Compare that with the filter in the second screenshot. Sound is clear and airy. Vocals are crystal clear as they would be if listening a much louder level.

My own personal conclusion is that equal loudness contours only tell how a person perceives the level of one frequency vs another in isolation (which is how the tests were conducted). It's not really valid to how we perceive complex sounds such as music. Our brain is a powerful EQ on its own and has expectations how combinations of sounds should sound like.

Edited: As a sentence accidentally slipped out.
 

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I think it would make some sense that a more generic, gentle, low Q filter would be less invasive than following the loudness curves exactly. It sort of makes sense not to try to nail the wiggle in the treble region, for example.
 
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