Did anyone else find this http://www.soundonsound.com/sos/sep11/articles/loudness.htm as disturbing as I did? I feel there are several distortions and misrepresentations in this article which may reflect badly on Sound On Sound's reputation for accuracy. The main article appears to be accurate enough and well researched, (though I'll flatly disagree with some of the author's conclusions without apology, as I think the methodology unsound) but the sidebars contain some real headscratchers.

Quote:

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"Limiters reduce loudness ranges, don’t they? Well, yes — and no. In fact, this issue is much more complex than it seems. Imagine you’ve got an audio file that is normalised: you can’t add any more gain without getting distortion. Using a limiter or a compressor on such a file will nevertheless add gain to its content: the RMS levels will be increased. This adds dynamic range to the medium: instead of being, in the case of a 16-bit file, 96dB, it will increase to perhaps 100 or 105 dB. On the diagram to the right, this additional available dynamic range is illustrated by the grey rectangle. From that point of view, limiters don’t decrease the loudness range, they increase it."

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Firstly the author is referring to level not gain; the two are not interchangeable. Second, there's no distinction between clipping distortion of the kind commonly exhibited by a limiter pushed too hard and distortion caused by digital overs. Perhaps he should have simply said "overs". Next, the phrase, "adds dynamic range to the medium" cannot possibly be true, as digital PCM audio encoding has a fixed dynamic range based on bit depth and sample rate that no limiter can affect. By the author's own nomenclature, dynamic range and loudness range are not the same thing. He appears to be trying to imply that a limiter can increase the loudness range of a normalized audio file, when this is patently not the case. (technically, dynamic range increases, as the waveform gets taller, on average, but only in the sense of the difference between RMS average and digital black, which the author went out of his way in the main article to note was an incorrect usage of the term, which should properly be reserved for the absolute dynamic range of the digital system as I described, above) Furthermore, the decibels in this paragraph are not referenced to anything; this makes them essentially meaningless, though I presume dBSPL is what's meant, in context. Is this author trying to pull a fast one, or am I completely misreading this?

Edited by Jennifer Jones (30/09/11 10:45 AM)

Sounds like he's mixing up perceived values and actual values. I can see how a limited file can give the perception of greater dynamic range - but it's at the expense of clipping and distortion.

Just read it - laughably poor logic. Good grief SOS - what are you letting in these days!!!

Measuring dynamic range with variations in RMS - that's an idiotic argument in musical terms. The whole article suffers from confirmation bias. I'd love to have a chat with the author and understand why he has offered such flawed reasoning.

Edited by Jennifer Jones (30/09/11 11:01 AM)

That was a point I found quite hard to understand while editing the article, and it's entirely possibly I've introduced some confusions in the process of editing! At the time I asked Emmanuel to clarify it and he sent me the following explanation -- I hope he won't mind me reproducing it here:

Suppose you've got a medium whose dynamic range is 50dB.
Suppose you're recording a signal on it, for instance vocals from a mike.

50dB is not much. Let's say the singer is yelling at some point of the recording: you've got to set a low input gain to avoid distortion. But then the singer whispers: this whisper is lost in the medium's background noise.
Conversely, you can increase the input gain to get a clear recording of the whispers, but then, when the singer's yelling, the medium generates distortion.

Now insert a compressor/limiter between the mike and the medium. Set the recording device's input gain high. The singer's whispering: the compressor is transparent because this whisper is below the threshold. The whisper's recording is above the medium's background noise because the recording device's input gain is high enough. The singer's yelling: the compressor springs into action, the peak levels are attenuated, and the yelling is recorded correctly onto the medium, without distortion. A source with a higher dynamic range than the medium's is recorded without distortion.

At this point, one might argue that the dynamic range of the medium was not increased, but the dynamic range of the source was decreased to be "crammed" into the medium's dynamic range. Well, yes and no.

Yes, because the dynamic range of the signal (dynamic range of the RMS, highly correlated to loudness range, which is what eventually matters) was indeed reduced.

No, because the RMS dynamic range / loudness range of what's actually recorded onto the medium is now greater than 50dB. Having reduced the signal's crest factor at high levels, the compressor/limiter did actually add 5 or 10dB of additional RMS dynamic range / loudness range to the original 50dB. The RMS dynamic range of what's recorded onto the medium is now 55 or 60dB, while it would have been 50dB max without any dynamic processing.

Let's put that in numbers (arbitrary values just for the sake of the example).
Dynamic range of the medium: 50dB
Dynamic range of the original vocals: 70dB
Dynamic range of what's recorded with a compressor/limiter: 60dB.

Better?

There are some definition issues here. The dynamic range of a medium is generally given as the maximum level minus the minimum level that can be recorded. If expressed in peak levels in the digital domain, it's a clear definition. If expressed in peak levels in the analog domain, it gets more complicated: why would you consider the background noise's peak levels? It's the RMS level that counts. But then, you have to reason with RMS levels, and consider the maximum RMS level you can record. But for a given peak value (generally the max. instantaneous level the medium can handle while still behaving linearly), the RMS level depends on the signal's crest factor, so there is no absolute "RMS dynamic range" for any medium. It depends on the signal.

Also, when I write "Yes, because the dynamic range of the signal (dynamic range of the RMS, highly correlated to loudness range, which is what eventually matters) was indeed reduced.", it's not systematically true. As I show in the article, some signals show a resilience to compression: their RMS dynamic range / loudness range doesn't decrease at all when compression ratios are reasonable. In numbers:
- dynamic range of the medium: 50dB
- dynamic range of the source: 60dB
- dynamic range of what's recorded: 58dB.
The source dynamic range's reduction is negligible compared to the increase of the medium's.

Let's put the increase of the medium's dynamic range in other words.
Low level signal => no compression => signal is not processed => signal takes advantage of the medium's dynamic range
High level signal => compression => signal is processed, its crest factor is decreased => signal's RMS is increased whereas its peak doesn't increase => additional perceptual / RMS / dynamic / loudness range on "top" of the medium's dynamic range.

I hope it gets clearer now.... this is not easy to understand, and not easy to explain....

Edited by Jennifer Jones (30/09/11 11:02 AM)

Quote Mojobone:

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Furthermore, the decibels in this paragraph are not referenced to anything; this makes them essentially meaningless, though I presume dBSPL is what's meant, in context.

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Surely the dynamic range of a system can be expressed simply in dB without a reference?

Edited by Jennifer Jones (30/09/11 11:02 AM)

So is the upshot more to do with noise floor and distortion within a given recording medium?...the argument being that we are recording things with compression and limiters that otherwise would not/could not be recorded as the medium distorts or adds too much noise?..thereby we are artificially increasing dynamic range?...D'oh!

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My head hurts!

Edited by Jennifer Jones (30/09/11 11:02 AM)

Quote Sam Inglis:

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...because the RMS dynamic range / loudness range of what's actually recorded onto the medium is now greater than 50dB. Having reduced the signal's crest factor at high levels, the compressor/limiter did actually add 5 or 10dB of additional RMS dynamic range / loudness range to the original 50dB. The RMS dynamic range of what's recorded onto the medium is now 55 or 60dB, while it would have been 50dB max without any dynamic processing.

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Sorry Sam, but I just can't go along with this...

In the example above dynamic range of the medium has in no way been increased. All material still exists within the 50dB window. All that has happened is what in fact we all quite simply understand, a previously greater dynamic range has been reduced to fit within the operating parameters of a recording system.

How 'resistant' material may be to reduction in dynamic range under compression is simply a function of how it is measured - in time, frequency and amplitude domains.

For me Emmanuelle is simply expressing what we all appreciate but in an opposite way. It seems completely counterintuitive (as shown by some of the feedback I think) and potentially extremely confusing to beginners. If looking at dynamic range in this way led us to new areas of understanding , then it would to my mind have purpose, but as it is, it just seems pointlessly contrary.

Loopy

Edited by Jennifer Jones (30/09/11 11:02 AM)

then he's completely misunderstood what everyone is talking about. He's squashing 60dB of dynamic range into a smaller space - fine. That's what compressions and limiters are for. But the dynamic range is not preserved. The PERCEIVED dynamic range may be - but the actual dynamic range has gone.

He claims that it is "without distortion". Untrue - by using a compressor you have no choice but to introduce distortion. Any change in the waveform is a distortion. Moreover - any harsh use of the compressor will introduce harmonic distortion - which will be audible. IU'm afraid he's fallen for the fools game of "louder is better".


I'll make it simple. If you have a signal whose lowest to highest levels run a range of 60dB and it is then recorded into a medium playing back 58dB of dynamic range - then you have lost dynamic range. The whole article is fun, nice and pretty much an explanation of how to get more performance dBs into a limited dynamic range.

It's called compression and it reduces dynamic range. SOS - you've been had. This is an entropic device - you cannot recover the information once it is lost - you MIGHT like to try an expander to increase the dynamic range of a dynamically limited signal - but the information of the real dynamics are forever lost. It may be IMPLIED by certain mix tricks or compression etc - but the dynamics are GONE. It's a bunk article and if it ever does get posted as a paper to the AES..... get ready to duck.

I mean - c'mon - duh!!!




Oh - and RMS dynamic range doesn't really mean anything.


- just read the definitions in EBU 3342. This does not refer to dynamic range but "loudness range". Not the same thing.

Edited by Jennifer Jones (30/09/11 11:02 AM)

My head hurts. Too many ranges, levels and gains.

My idiot understanding is as follows.

Most recording media has lots of space for very loud and very quiet signals, as long as the input signal is calibrated appropriately so that it's loud enough to be detected (above the noise floor) and not too loud to saturate the medium (clipping, absolute distortion, whatever).

Compressors can help make quiet bits of the input signal sound louder, by (paradoxically) making the loudest bits a bit quieter. This makes a bit of space at the 'loud' end of the recording medium, so we can shift the signal along a bit (increase the input gain on the recording medium) without saturating it.

If the loudest chunks of the input signal are still too loud despite compression, a limiter will absolutely stop the loudest signals from going above a level which will saturate the recording medium. If we limit more and more of the loud end of the signal, we can again shift the signal along a bit (increase the input gain on the recording medium).

Eventually, the 'really quiet' chunks of the input signal, sound much louder on the recording medium. However this comes at a cost: the 'louder' and 'really loud' chunks of the input signal will have been squeezed (compressor) or smashed (limiter), changing their musical waveforms and making them sound different.

Ergo, the whole thing sounds louder, but with less variation between loud and quiet. As the compressors and limiters have done their work and deliberately distorted the recorded waveforms, it's impossible to recreate the original 'really loud' and 'really quiet' signals as that information is no longer there.


As it happens, most tin-pot radios, TV speakers, or car stereos, and a lot of cheap earbuds, don't want really quiet signals because their day-to-day noise floor is much higher than the device (or medium) used to record the signal on and so consumers can't hear it over the car noise or whatever. Therefore, commercially this might not be a problem.

When I'm buying music to listen to in a quiet space with a half-decent stereo (because perversely I like to listen to the artist rather than have them blaring as background music), my system has a low perceived noise floor, and so the lack of variation between quiet and loud, and the distortion on the loud bits, is noticeable, annoying, and gives me a headache after four tracks.

Is that about the nub of it?

Edited by Jennifer Jones (30/09/11 11:02 AM)

Quote turtles:

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My head hurts.

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turtles is not alone. What a thread! My head is wrecked!

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.. is this thing on?

Edited by Jennifer Jones (30/09/11 11:03 AM)

Sounds like the classic case of a person with a little knowledge - its absolutely technically wrong, and its misleading - but in a classic case of Bad Science it has lots of technical words and measurements in there so people without a clear understanding of the subject will be pulled in. I fairly sure Hugh and Paul will be swearing in a back office somewhere.

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Daniel Davis
Edinburgh Recording Studio Windmill Sound

Edited by Jennifer Jones (30/09/11 11:03 AM)

I've read that article three times trying to understand it.
I had problems understanding some of the terminology as I've never heard of most of the measurment terms "gated RMS variability" "EBU 3342 lounness range" "crest factor" etc and they weren't really explained very well.

I normally treat information in SOS as gospel,so it will be interesting re-reading the article, with the benifit of the comments above.

Whether the article is accurate or not, I really enjoyed reading such an in depth feature with its interesting (if slightly obtuse) graphs.
As an SOS reader for over 20 years, I'd welcome more SOS features aimed at a slightly higher level of understanding.

Edited by Jennifer Jones (30/09/11 11:03 AM)

I read the article whilst sitting next to a pool with a beer in the midst of a Spanish heat-wave, and thoroughly enjoyed it. I particulary liked the concept that I can brick-wall limit my audio to post Death Magnetic levels (incidentally - I love the sound of that CD) yet still maintain great dynamic range so long as I leave some silent bits in it. It all seemed thought provoking but not awfully useful. I'm not sure that "Bollocks" is quite fair so much as "from an alternative perspective" and unfortunately the use of very specifically defined language to describe alternative views is always going to be fraught.

A

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Andi, www.thedustbowl.net Mixing, Mastering, Audio Editing at The Dustbowl Audio

Edited by Jennifer Jones (30/09/11 11:03 AM)

When things seem bad, I recommend reading Wittgenstein's Tractatus Logico-Philosophicus. Twice.

Edited by Jennifer Jones (30/09/11 11:03 AM)

Following is my answer to criticism expressed by Mojobone, Narcoman, Urthlupe, and Daniel Davis.

To the others, I thank you for trying to understand

Btw my name is Emmanuel, not Emmanuele or Emmanuelle.

Quote Mojobone:

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Firstly the author is referring to level not gain; the two are not interchangeable.

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No - reread the sentence. I'm referring to gain.

Quote Mojobone:

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Second, there's no distinction between clipping distortion of the kind commonly exhibited by a limiter pushed too hard and distortion caused by digital overs.

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That's simply wrong. Digital overs cause harmonic distortion (harmonics 1, 3, 5 etc.). Well-adjusted limiters don't, and I'm not even sure that limiters pushed too hard do.

Quote Mojobone:

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He appears to be trying to imply that a limiter can increase the loudness range of a normalized audio file, when this is patently not the case

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I'm not trying to imply that - at all. Reread the article and the explanation given by Sam above.

Quote Mojobone:

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technically, dynamic range increases, as the waveform gets taller, on average, but only in the sense of the difference between RMS average and digital black,

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That's perfectly true. It means that if you put a limiter, you can achieve a higher RMS level from a given signal, and the DR of the medium will have seemed to increase - at the price of the loss of "crest factor". I didn't invent the method, it's been used for a good 50 years.

Quote Mojobone:

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Furthermore, the decibels in this paragraph are not referenced to anything; this makes them essentially meaningless, though I presume dBSPL is what's meant, in context. Is this author trying to pull a fast one, or am I completely misreading this?

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As Sam confirms it:
- a logarithmic level measure needs a reference, thus the "FS" in "dB FS" for instance, or the "SPL" in "dB SPL."
- a logarithmic range measure doesn't need a reference. It's essentially a "dimensionless" unit, a ratio. Same thing for the loudness range: EBU3341 loudness is expressed in "LU FS", and EBU3342 loudness range is expressed simply in "LU".

Quote Narcoman:

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Sounds like he's mixing up perceived values and actual values

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Read the article again! I give the translation between the two according to ITU1770 and EBU3341, and explain that RMS and EBU3341 loudness are highly correlated in the case of the corpus. Thus, the possibility to consider one or the other indifferently, in the case of this corpus, and in the case of the 3341 norm. You can think the EBU3341 standard is badly defined if you want, that's another story.

Quote Narcoman:

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Measuring dynamic range with variations in RMS - that's an idiotic argument in musical terms

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No - if anything, it's a first approximation that may be improved. The approximation is justified by the high correlation between EBU3341 and RMS, and, I repeat, in the case of the corpus I used.

Quote Urthlupe:

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In the example above dynamic range of the medium has in no way been increased. All material still exists within the 50dB window. All that has happened is what in fact we all quite simply understand, a previously greater dynamic range has been reduced to fit within the operating parameters of a recording system.

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I agree that this is difficult to understand - refer to Sam's explanation, I can't do much better than that.
The central issue is that very roughly, "instantaneous signal variability" = "RMS variability" + "crest factor". The medium conditions the "instantaneous signal variability", so if you reduce the "crest factor", you increase the "RMS variability".

The lack of common vocabulary does open the door to confusion.

Quote Urthlupe:

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If looking at dynamic range in this way led us to new areas of understanding , then it would to my mind have purpose, but as it is, it just seems pointlessly

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It's not quite "new". This method of reducing the "crest factor" in order to increase "RMS variability" for a given "dynamic range of a medium" is not as useful as it was, due to the high dynamic ranges achieved by digital media. In the 50's / 60's, it was another story.
But - consider temporarily the musical accompaniment as "background noise" and the lead vocals as "signal" - that makes you understand why lead vocals are compressed. And next time you compress lead vocals, you know why. Is that so "pointless"?

Quote narcoman:

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then he's completely misunderstood what everyone is talking about. He's squashing 60dB of dynamic range into a smaller space - fine. That's what compressions and limiters are for. But the dynamic range is not preserved. The PERCEIVED dynamic range may be - but the actual dynamic range has gone.

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Please re-read the article. And, can you give me a definition of "dynamic range" of a signal while you're at it?
The real question might very well be: why are you so angry?

Quote narcoman:

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He claims that it is "without distortion". Untrue - by using a compressor you have no choice but to introduce distortion. Any change in the waveform is a distortion. Moreover - any harsh use of the compressor will introduce harmonic distortion - which will be audible. IU'm afraid he's fallen for the fools game of "louder is better".

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Listen to "The Prettiest Thing" by Norah Jones: heavily limited, perfectly pristine.

"Louder is better"? Me? Did it occur to you that if I wasn't concerned about over limiting, I wouldn't have gone to all the trouble to investigate the issue to such lengths? The basic purpose of the article is to make people think about the actual reasons why recent music is sometimes tiring, to make them think about the loudness war, etc. "Less DR" doesn't cut it, the matter is much more complex. Know your enemy - if you want to fight loudness war, know what it does...

Read this review:
http://productionadvice.co.uk/loudness-war-dynamic-range/
Ian Shepherd is the organizer of the dynamic range day, I suppose he fell for "louder is better", too?

Quote narcoman:

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I'll make it simple. If you have a signal whose lowest to highest levels run a range of 60dB and it is then recorded into a medium playing back 58dB of dynamic range - then you have lost dynamic range. The whole article is fun, nice and pretty much an explanation of how to get more performance dBs into a limited dynamic range.

It's called compression and it reduces dynamic range. SOS - you've been had. This is an entropic device - you cannot recover the information once it is lost - you MIGHT like to try an expander to increase the dynamic range of a dynamically limited signal - but the information of the real dynamics are forever lost. It may be IMPLIED by certain mix tricks or compression etc - but the dynamics are GONE. It's a bunk article and if it ever does get posted as a paper to the AES..... get ready to duck.

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Cool down. Read the explanation again. I also state in another paragraph that limiters reduce the RMS variability.
The RMS variability of the original content will be reduced, OK, but the eventual RMS variability will be increased - and that was the point.
The issue is not so simple - I draw your attention again to "instantaneous signal variability" = "RMS variability" + "crest factor". Think it over.

What you're doing is that you read parts from the article, forget about the other parts, draw bad conclusions, and then for some reason think that I'm trying to fool the journal. Isn't that weird? Really, why would you think that I would try to trick SOS? What would I get from it?

Quote Daniel Davis:

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Sounds like the classic case of a person with a little knowledge - its absolutely technically wrong, and its misleading - but in a classic case of Bad Science it has lots of technical words and measurements in there so people without a clear understanding of the subject will be pulled in. I fairly sure Hugh and Paul will be swearing in a back office somewhere.

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Sorry to disappoint you, but Hugh's reaction was "It was in every way a superb article."
(See http://www.soundonsound.com/forum/showflat.php?Cat=&Number=937023& page=2&view=collapsed&sb=5&o=&fpart=1#937023.)

If you think that the article is technically wrong, then I guess both the ITU and the EBU are wrong too, since I take a lot of notions from them.

@Narcoman & Daniel Davis, I really wonder why some people are so aggressive? If you disagree, just say you do… don't tell me that I have little knowledge, or that this is Bad Science. You can even feel free to prove that I'm wrong. But opinions like that, where does that lead anyone?


Emmanuel

Edited by Jennifer Jones (30/09/11 11:03 AM)

This is clearly a very complicated topic, and some of Emmanuel's claims and arguments are 'challenging' -- but there's nothing wrong in that. It's only by exploring different theories and propositions that we can arrive at a definitive answer. I suspect there is also some confusion in parts because of different interpretations being placed on apparently familiar terms.

I'll let Emmanuel defend his own article, but would remind everyone to play nice please

There is one small point that I must challenge, though:

Quote Emmanuel D.:

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That's simply wrong. Digital overs cause harmonic distortion (harmonics 1, 3, 5 etc.).

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Digital overs don't result in harmonic distortion. They actually produce anharmonic distortion. The process is that digital overs initially generate harmonic distortion within the quantiser because, having run out of numbers, the waveform tops are clipped. However, these harmonics naturally extend beyond half the sampling rate, and they are therefore aliased within the frequency domain to appear at frequencies below their fundamental. The distortion products appear at frequencies which are the mathematical product of the harmonics and the sample rate, not a normal linear harmonic series.

hugh

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Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 11:03 AM)

Quote Emmanuel D.:

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Please re-read the article. And, can you give me a definition of "dynamic range" of a signal while you're at it?
The real question might very well be: why are you so angry?

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Yo big guy. Thanks for responding. No anger here at all (aside from the frustration of your article giving credence to the loudness 'tards in the world). Don't read between the lines.....

Dynamic range - can only be - loudest signal to quietest signal (not silence). Anything else is trying to fit a square peg into a round hole.

Quote Emmanuel D.:

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Listen to "The Prettiest Thing" by Norah Jones: heavily limited, perfectly pristine.

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Well this is becoming clearer now. That certainly isn't a pristine recording at all. It's distorted. It SOUNDS distorted and it reports as distorted via any simple DAW analysis. Pristine? It's very hard to listen to outside of a car or a laptop. I thought that the day it came out an I think it even more now. It's still a good record though!

Listen to the Berlin Phil. Something none classical ? What about Tron soundtrack mixed by Alan Meyerson - big distorted sounds in there, but all in the mix and not in the master presentation. OR Crooked Vultures on Vinyl. Great sound. Sounds terrible on CD (although still hugely gratifying on a rawk level). Dirtbombs - Ultraglide in Black. Limited to hell - works in the medium but i'd never say it was a "beautiful recording". I'd say it was a loud and rawks recording. Sometimes it's what we want!

I mixed the album of a well known two piece back in '04. The masters were nothing like my mixes in terms of clarity although they DID sound more powerful. In my job I get to hear a lot of mixes before they're mastered, well known ones too (Crazy by Gnarls Barkley still sticks in my mind as an amazing mix but poor and brittle master). Now - I haven't heard the mixes from Norah Jones' album but if my (not inconsiderable) experience with myriad other releases and projects is anything to go by .... well - you know where I'm going.

Quote Emmanuel D.:

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Read this review:
http://productionadvice.co.uk/loudness-war-dynamic-range/
Ian Shepherd is the organizer of the dynamic range day, I suppose he fell for "louder is better", too?

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No - he hasn't given your article a negative or positive response. He claimed it to be an excellent article. Not that he agrees with you. Hes even outlined where your article is wrong and where people will get confused.

Quote Emmanuel D.:

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Quote narcoman:

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I'll make it simple. If you have a signal whose lowest to highest levels run a range of 60dB and it is then recorded into a medium playing back 58dB of dynamic range - then you have lost dynamic range. The whole article is fun, nice and pretty much an explanation of how to get more performance dBs into a limited dynamic range.

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Cool down. Read the explanation again. I also state in another paragraph that limiters reduce the RMS variability.
The RMS variability of the original content will be reduced, OK, but the eventual RMS variability will be increased - and that was the point.
The issue is not so simple - I draw your attention again to "instantaneous signal variability" = "RMS variability" + "crest factor". Think it over.

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Is there something wrong with your thermostat? You've written a contentious article (mostly the title which is NOT supported by the standard you are looking at - there is a big difference between loudness range and dynamic range. Even the EBU paper says to be CAREFUL in that analysis) and it'll be given strong credence by the loudness loons. It IS that simple. You cannot claim greater dynamic range when you've raised the gain and limited the top. No anger here at all.

You've separated out RMS variability (which can only ever been experienced as a rough guide to perceived volume) from Crest factor. When dealing with an absolute of dynamic range (which ISN'T an RMS factor - it's very mechanism denotes time domain) you can't infer a time domain based event showing a correlation with a static difference measurement. It doesn't make sense. The mathematics of it just doesn't hold water! When dealing with the issues in limiters (that of removed transients) the RMS variability doesn't come into play. It just doesn't matter and has no effect on what the concerns over the loudness wars are about - that of loss of detail through removed transients.

Now - one COULD argue that quieter events are brought forward and may be more hearable - but I'd argue that those details were always there and could have been revealed by a simple turn of the gain knob. Okay - you could counter argue and say the larger transient events would be intolerable if unlimited but that's not what we're debating.

Quote Emmanuel D.:

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@Narcoman & Daniel Davis, I really wonder why some people are so aggressive? If you disagree, just say you do… don't tell me that I have little knowledge, or that this is Bad Science. You can even feel free to prove that I'm wrong. But opinions like that, where does that lead anyone?
Emmanuel

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It is (unintentional) bad science is is written like a marketing document. One doesn't have to prove you're wrong when you've not proven you're right! The points raised by your opponents have picked holes in your reasoning. I understand the POINT of your article but your reasoning is flawed. You've also mixed up loudness range with dynamic range (intentionally? Probably not) which EBU3341 goes at great pains to warn the user away from. You've also tried to intimate that your 50 dB range medium correctly encodes a 60dB range piece of work. THAT is bad reasoning. It doesn't - it's an entropic process!! Information is lost.

The problem we have is that SOS has seen fit to publish an article which the loudness loons will run with as proof that it's okay to smash the [ ****** ] out of stuff! Have you seen the number of people who have already jumped on your article as a holy grail!!

The intention of the article may be great (as I can see from your responses it is) but it's poorly written (it mixes up established definitions within the biz) and the title itself, unwisely chosen. I spent many years in academia as a published research scientist in maths for signal communication - one has to be VERY careful about title and wording in any paper. I wrote two papers for the Virtuosi project; the second was a rethink about my broadly panned paper on the energy at the turning points of waveforms through physical media after I was brutally shot down by none other than Carl Feynmann. I wasn't wrong but my argument hadn't been thought through well enough (something my idiot of a supervisor should have spotted).

I recommend a re-think about how you've presented this; your responses here do show you care about this subject but the article plays into the loudness loons hands.

Edited by Jennifer Jones (30/09/11 11:04 AM)

Quote Emmanuel D.:

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To the others, I thank you for trying to understand

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I am also trying to understand Emmanuel. Apolgies for mis-spelling your name.

Quote Emmanuel D.:

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Quote Urthlupe:

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In the example above dynamic range of the medium has in no way been increased. All material still exists within the 50dB window. All that has happened is what in fact we all quite simply understand, a previously greater dynamic range has been reduced to fit within the operating parameters of a recording system.

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I agree that this is difficult to understand - refer to Sam's explanation, I can't do much better than that.
The central issue is that very roughly, "instantaneous signal variability" = "RMS variability" + "crest factor". The medium conditions the "instantaneous signal variability", so if you reduce the "crest factor", you increase the "RMS variability".

The lack of common vocabulary does open the door to confusion.

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In that explanation Emmanuel you wrote - 'The RMS dynamic range of what's recorded onto the medium is now 55 or 60dB, while it would have been 50dB max without any dynamic processing. '

This is simply not correct, what is now recorded onto the medium does not exceed the dynamic range of that medium - 50dB in your example. A fundamentally inaccurate and confusing way to express your point, regardless of a lack of common vocabulary.

Quote Emmanuel D:

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Quote Urthlupe:

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If looking at dynamic range in this way led us to new areas of understanding , then it would to my mind have purpose, but as it is, it just seems pointlessly

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It's not quite "new". This method of reducing the "crest factor" in order to increase "RMS variability" for a given "dynamic range of a medium" is not as useful as it was, due to the high dynamic ranges achieved by digital media. In the 50's / 60's, it was another story.
But - consider temporarily the musical accompaniment as "background noise" and the lead vocals as "signal" - that makes you understand why lead vocals are compressed. And next time you compress lead vocals, you know why. Is that so "pointless"?

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I don't think you understand what I said. You might re-read it. I am also (unfortunately) 'not quite new', I have been using methods of controlling dynamic range in various audio recording, playback and restoration systems since the late 70's and hope to god I will understand my reasons the next time I compress lead vocals (and within the context of your article).

For what its worth Emmanuel I don't think you should take some of the comments here so personally - If nothing else your writing has stimulated thought and discussion which is a very positive thing.

Loopy

Edited by Jennifer Jones (30/09/11 11:04 AM)

...those of us who "compress vocals" do it knowing full well we are reducing it's dynamic range to fit in in the artistic or technical parameters of our mix!! we're not fooling ourselves that we've kept the same dynamics in a more limited space.

Edited by Jennifer Jones (30/09/11 11:04 AM)

Quote narcoman:

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...those of us who "compress vocals" do it knowing full well we are reducing it's dynamic range to fit in in the artistic or technical parameters of our mix!! we're not fooling ourselves that we've kept the same dynamics in a more limited space.

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... then please, be as kind as to explain what is the "dynamic range" of a vocal part, or for that matter of any piece of music. So that people can know what is "reducing its dynamic range to fit in".

Edited by Jennifer Jones (30/09/11 11:04 AM)

Quote Emmanuel D.:

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... then please, be as kind as to explain what is the "dynamic range" of a vocal part, or for that matter of any piece of music. So that people can know what is "reducing its dynamic range to fit in".

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That's a very obtuse request as it's fundamental to your argument....

The dynamic range of a signal is NOT it's perceived dynamics. At all. Ever. It's is purely measured and quantifiable. Now - sometimes there are different WAYS of measuring things - but in dynamic range it is only the quietest part of a signal to the loudest. Windowing a signal in the time domain can lead to skewed results.

The dynamic range of a part, or piece of music, can only EVER be it's quietest signal to it's loudest. That is it. ANY other measure is not dynamic range. You appear more and more to be referring perceptual coding - which is what SOME limiting does, gives the impression of louder music (where simple gain may have sufficed) within a limited dynamic range application.

If the minimum part of a vocal is down at -40dB(FS) and you're highest at -15dB(FS) (both very unlikely!!) then your dynamic range is bounded by those two values. The dynamic range of your signal is not (not now not ever):

I have a piece with 72 dB dynamic range , I record it through a mic and compressor bringing it down to 25dB dynamic range and that recording now has a 72dB dynamic range. The information is lost. It may have the appearance (or sonics) of maintaining all of it's dynamics since we as humans know what a quiet voice kind of sounds like - we'd hear it as intimate and subjectively "quieter". But it ISN'T quieter. The dynamic range is measurably what is present in the recording no matter what the original was.

You put a limiter at -6dB(FS) on a signal peaking at 0dB(FS) and then raise it's level by 6dB - you have LOST 6dB of dynamic range. You've brought the noise floor up. You may even perceptually reveal details that you didn't hear before - but your dynamic range is reduced. By 6dB as it happens.

Did you not read my prior lengthy response?

Sorry to be potentially a little rude here: is English not your first language and are we having subtle language issues that you may not be aware of? Perhaps that is at the root of the issue? Sorry if that isn't true - just wondering

Edited by Jennifer Jones (30/09/11 11:04 AM)

Quote Urthlupe:

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This is simply not correct, what is now recorded onto the medium does not exceed the dynamic range of that medium - 50dB in your example. A fundamentally inaccurate and confusing way to express your point, regardless of a lack of common vocabulary.

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I assure you the problem lies in the lack of a common vocabulary, and to the arbitrary acceptation of notions that don't stand a closer examination.

Let's start with the dynamic range of a medium.

The top of the range is well-defined, at least in the case of digital media: a sample should not exceed 0dB FS.
The bottom of the range is less well-defined. Assuming for a moment that the background noise is white noise, should we take the bottom value as the highest sample belonging to this noise? It wouldn't make sense. We should rather consider the RMS of the background noise, or better yet, its loudness.
But there is an immediate problem: the top of the range is an instantaneous value, and the bottom an averaged one.
Evaluating the dynamic range, you'd be subtracting an RMS level (or something not far from it) to a peak level. Now that doesn't make sense.
To make sense, we should be subtracting either an RMS value to another RMS value, or a peak value to a peak value.
But the peak value of a white noise is meaningless as far as its RMS and loudness are concerned.
So let's consider subtracting an RMS value to another RMS value.

The RMS value of the background noise (white or otherwise) is known.
But the RMS value depends on the signal that's to be stored.

Let's focus on this RMS value. We agree on the fact that if we subtract one RMS from another one, then the dynamic range of the medium will be the maximum value of the signal's RMS minus the background noise RMS value.
So, what's the maximum RMS value the signal can take?…. it depends on the signal itself.
It means that the SNR of a medium expressed in RMS values depends as much on the signal as to the medium
(I know it sounds strange, but since we can't express the SNR otherwise, how to go around this?)

One thing is sure, the signal's peak cannot go above 0dB FS. So the SNR of the medium is, still from that RMS perspective, related to the ratio between the signal's peak and its RMS. In other words, to the crest factor. If the crest factor decreases, the "RMS SNR" increases. And, any processing that reduces the crest factor will increase the medium's SNR when expressed in terms of RMS. Conclusion, using a limiter will increase the medium's SNR.

I know it sounds shocking. The core of the problem is: how to define the dynamic range of a medium in the first place? Peak minus peak makes no sense, since peak is almost not related to loudness. Peak minus RMS makes no sense, since those are two different notions. It only leaves RMS minus RMS or loudness minus loudness… and it leads to that conclusion that you apparently find unacceptable.

But really, provide an alternative solution, I'd be happy to hear it, and it's not a figure of speech.

Quote Urthlupe:

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For what its worth Emmanuel I don't think you should take some of the comments here so personally - If nothing else your writing has stimulated thought and discussion which is a very positive thing.

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That's good news

Edited by Jennifer Jones (30/09/11 11:05 AM)

Quote Hugh Robjohns:

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Digital overs don't result in harmonic distortion. They actually produce anharmonic distortion. The process is that digital overs initially generate harmonic distortion within the quantiser because, having run out of numbers, the waveform tops are clipped. However, these harmonics naturally extend beyond half the sampling rate, and they are therefore aliased within the frequency domain to appear at frequencies below their fundamental. The distortion products appear at frequencies which are the mathematical product of the harmonics and the sample rate, not a normal linear harmonic series.

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I'm fine with that.
So, the reformulation of my argument would be: limiters don't result in anharmonic distortion.
If I'm wrong, correct me, but the kind of distortion caused by limiters is usually not a variant of harmonic distortion.

Edited by Jennifer Jones (30/09/11 11:05 AM)

Quote Emmanuel D.:

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I assure you the problem lies in the lack of a common vocabulary, and to the arbitrary acceptation of notions that don't stand a closer examination.

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Possibly

Quote Emmanuel D.:

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Let's start with the dynamic range of a medium.

The top of the range is well-defined, at least in the case of digital media: a sample should not exceed 0dB FS.
The bottom of the range is less well-defined. Assuming for a moment that the background noise is white noise, should we take the bottom value as the highest sample belonging to this noise? It wouldn't make sense. We should rather consider the RMS of the background noise, or better yet, its loudness.

---

That's the problem. You should be using an instant measured point in the signal (noise included).

Quote Emmanuel D.:

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But there is an immediate problem: the top of the range is an instantaneous value, and the bottom an averaged one.
Evaluating the dynamic range, you'd be subtracting an RMS level (or something not far from it) to a peak level. Now that doesn't make sense.

---

of course.

Quote Emmanuel D.:

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To make sense, we should be subtracting either an RMS value to another RMS value, or a peak value to a peak value.
But the peak value of a white noise is meaningless as far as its RMS and loudness are concerned.

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That is where you're reasoning breaks down. You should average the PEAKS of the noise over time to use a working value. Second, if your signal is always OVER the noise floor then the noise floor is largely not relevant (in practise that gets more difficult if you are using a recording with sections of silence - a voice recording for example. Where you measure matters and is indeed a point for argument. BUT this becomes moot in terms of "noisy" music). You must make a decision to the minimum point.

Quote Emmanuel D.:

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It only leaves RMS minus RMS or loudness minus loudness… and it leads to that conclusion that you apparently find unacceptable.

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Yes. It's is unacceptable as a measure of dynamic range. It can only lead to an evaluation of the RMS of dynamic range. The only way you could make this acceptable would be to form an calculable expression of the two RMS values and look at the difference between the differentials of the two values. Bearing in mind that would mean forming a calculus expression for Bryan Adams I think we can dismiss that as a little bit "out there".

Quote Emmanuel D.:

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But really, provide an alternative solution, I'd be happy to hear it, and it's not a figure of speech.

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Elective measurement between peaks and troughs OR averaged measurement of troughs against peak (although that makes it somewhat "averaged". I recognise the difficulty in deciding a minimum but it isn't a valid solution to look at differences between RMS values as a measure of storable and worked dynamic range.

Edited by Jennifer Jones (30/09/11 11:05 AM)

Quote narcoman:

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You've separated out RMS variability (which can only ever been experienced as a rough guide to perceived volume) from Crest factor. When dealing with an absolute of dynamic range (which ISN'T an RMS factor - it's very mechanism denotes time domain) you can't infer a time domain based event showing a correlation with a static difference measurement. It doesn't make sense. The mathematics of it just doesn't hold water!

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Please notice the quotes I put:
"instantaneous signal variability" = "RMS variability" + "crest factor"
As I explain in a previous answer, I know well enough that you can't write an equation that contains both windowed and instantaneous signal descriptors.
But then what can I do? If I'm too precise, things get apparently too austere for some.
So I sum up, but then I'm not precise enough.

I'm sure you understand
"instantaneous signal variability" = "RMS variability" + "crest factor"
well enough to see through the necessary simplifications.

Quote narcoman:

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When dealing with the issues in limiters (that of removed transients) the RMS variability doesn't come into play. It just doesn't matter and has no effect on what the concerns over the loudness wars are about - that of loss of detail through removed transients.

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You just re-phrased the whole point of the article.
Indeed, the loudness war is about removed transients, not dynamic range. I couldn't agree more.

Edited by Jennifer Jones (30/09/11 11:05 AM)

Quote Emmanuel D.:

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As I explain in a previous answer, I know well enough that you can't write an equation that contains both windowed and instantaneous signal descriptors.
But then what can I do? If I'm too precise, things get apparently too austere for some.
So I sum up, but then I'm not precise enough.

---

Feel free to get as precise as you like - Maths PhD here!! I'll get it

Quote Emmanuel D.:

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You just re-phrased the whole point of the article.
Indeed, the loudness war is about removed transients, not dynamic range. I couldn't agree more.

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This is the fundamental disagreement - removal of transients IS the removal of dynamic range based on the correct dynamic range calculation. If you want a mathematical idiom to refer to - when you deal in two RMS values from the same source you are including and RMS of the included noise TWICE. As you limit the signal you merely bring the noise UP along with the signal. The dynamic range lost is precisely the amount of limiting done (of course, if no limiting take place then you're just moving the window).

To quantity your proposal as proof no dynamic range is lost (and it's in the definition of dynamic range that your argument doesn't work - it DOES work as an assessment of RMS values to information) you'll need to formulate a set of mathematical statements. Since including a root mean square in any set means using differential calculus for any instantaneous measurement you've already shown why you cannot compare the min and max values of the RMS as a valid form of dynamic range. What you really have is a measure of the average variability but not of any maxima or minima. Therefore NOT dynamic range. The two are not well correlated.


I can see where you might get the notion that SNR is increased - but you are only talking about the inherent system noise within the gain system POST limiting. However all that has happened in a "limited by 6dB" file (i.e. one in which 6dB has been removed from the top end) is the difference maxima and minima have been reduced. The crest factor is reduced, the RMS is raised (if we apply the 6dB gain to get the limited signal louder and the noise floor WITHIN the signal recorded is raised the same amount. Clipping transients is the SAME as removing dynamic range.


Edited by Jennifer Jones (30/09/11 11:05 AM)

Quote narcoman:

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That is where you're reasoning breaks down. You should average the PEAKS of the noise over time to use a working value. Second, if your signal is always OVER the noise floor then the noise floor is largely not relevant (in practise that gets more difficult if you are using a recording with sections of silence - a voice recording for example. Where you measure matters and is indeed a point for argument. BUT this becomes moot in terms of "noisy" music). You must make a decision to the minimum point.

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I will rephrase your first objection. The SNR would be a windowed measure of the signal's peaks minus a windowed measure of the noise's peaks? If so, how does that contradict what I said? On the contrary, it seems to be homogenous to my the conclusions I draw.

Your second objection I'm afraid I'm not getting. How can the noise floor be not relevant in the process of evaluating the SNR?

Quote narcoman:

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Yes. It's is unacceptable as a measure of dynamic range. It can only lead to an evaluation of the RMS of dynamic range. The only way you could make this acceptable would be to form an calculable expression of the two RMS values and look at the difference between the differentials of the two values. Bearing in mind that would mean forming a calculus expression for Bryan Adams I think we can dismiss that as a little bit "out there".

---

"the RMS of dynamic range"? we seem to be having a vocabulary problem once more
I begin to suspect we eventually agree, though we don't use the same approach exactly
can you rephrase that so that I can be sure?

"calculus"??? we're dealing with discrete measures, evaluating the differential is only subtracting the signal with itself, with a delay of one sample
not so "out there"
what would the derivative bring? can you be more explicit?

Quote narcoman:

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Elective measurement between peaks and troughs OR averaged measurement of troughs against peak (although that makes it somewhat "averaged". I recognise the difficulty in deciding a minimum

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Sorry, you'll have to rephrase, it seems that this is a very long piece of reasoning compacted in 3 lines, but it sounds interesting.

Edited by Jennifer Jones (30/09/11 11:05 AM)

Quote narcoman:

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This is the fundamental disagreement - removal of transients IS the removal of dynamic range based on the correct dynamic range calculation. If you want a mathematical idiom to refer to - when you deal in two RMS values from the same source you are including and RMS of the included noise TWICE. As you limit the signal you merely bring the noise UP along with the signal. The dynamic range lost is precisely the amount of limiting done (of course, if no limiting take place then you're just moving the window).

To quantity your proposal as proof no dynamic range is lost (and it's in the definition of dynamic range that your argument doesn't work - it DOES work as an assessment of RMS values to information) you'll need to formulate a set of mathematical statements. Since including a root mean square in any set means using differential calculus for any instantaneous measurement you've already shown why you cannot compare the min and max values of the RMS as a valid form of dynamic range. What you really have is a measure of the average variability but not of any maxima or minima. Therefore NOT dynamic range. The two are not well correlated.

---

OK now we (almost) understand each other.

In the article, I was careful of never using the word "dynamic range" as far as audio content is concerned:

There remains the question of whether one should use such a term as ‘dynamic range’ at all: there is no official definition for it, and it may be confused with the dynamic range of a recording medium, which is basically the difference between the highest and lowest level it can handle. During the course of this article, therefore, I won’t talk about ‘dynamic range’ in relation to a piece of music. Instead, I will be using ‘RMS variability’, or more generally ‘dynamic variability’. The term ‘dynamic range’ will be reserved for the measure of signal-to-noise ratio of a recording medium. I will use the term ‘loudness range’ in strict reference to the EBU 3342 document, and the term ‘loudness variability’ in other cases involving loudness instead of RMS.

I based my evaluation of "ranges" on the EBU3342 document, as I state. This particular quantity doesn't decrease with the loudness war. Apparently, you already knew that, but many people assume the contrary (just look at Wikipedia). Therefore, it was worth it to propose an experiment that would clearly show that the actual problem is crest factor, not a "range" in the like of which is defined by 3342.

I agree that the box about how limiters increase the dynamic range of the medium can be easily contradicted when considering other approaches of the dynamic range of a medium. However, you will agree that signals with a lower crest factor may allowed to show a higher RMS variability, given that the medium is the same. So that the use of limiters may result in higher "musical dynamics" (Forte.... Mezzoforte... Piano...)... and that was an interesting and useful point.

May I suggest that as far as mathematical concerns go, you PM me? Your approach is intriguing and I want to know more, but I fear that it may scare off many readers?

Edited by Jennifer Jones (30/09/11 11:05 AM)

Quote Emmanuel D.:

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I will rephrase your first objection. The SNR would be a windowed measure of the signal's peaks minus a windowed measure of the noise's peaks? If so, how does that contradict what I said? On the contrary, it seems to be homogenous to my the conclusions I draw.

---

That's the very problem. It may we windowed but you're asking for a time domain value to be a fair comparison for deducing dynamic range. By it's very nature a max and min can only be from an instant. As I say - I recognise the problem with deciding a minima when noise is present. But REMOVE noise from this as a mathematical concept - the noise is merely a problem of implementation not of reasoning. The noise can be used as valid signal. It makes no difference. Signal is signal.

Quote Emmanuel D.:

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Your second objection I'm afraid I'm not getting. How can the noise floor be not relevant in the process of evaluating the SNR?

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Why are we concerned with the SNR when looking at dynamic range? If the lowest part of your signal is under the noise floor then it IS noise. It's under our windowed function for a minima. Again - I recognise deciding the minima as the issue here.

Quote Emmanuel D.:

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"the RMS of dynamic range"? we seem to be having a vocabulary problem once more
I begin to suspect we eventually agree, though we don't use the same approach exactly
can you rephrase that so that I can be sure?

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You can't use a time domain collation as a result for deciding a max and min.

Quote Emmanuel D.:

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"calculus"??? we're dealing with discrete measures, evaluating the differential is only subtracting the signal with itself, with a delay of one sample

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EXACTLY!!!! You CAN'T use a time domain response for discrete measurement. You must search and find (in digital terms) the biggest and smallest sample value. No RMS.

Quote Emmanuel D.:

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not so "out there"
what would the derivative bring? can you be more explicit?

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It would bring a point value between the two RMS values. It wouldn't bring us a max or min for dynamic range!! I'm using it as the math basis for rejecting the hypothesis that RMS can be used as a meaningful basis for asserting dynamic range. It can't.

Quote Emmanuel D.:

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Quote narcoman:

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Elective measurement between peaks and troughs OR averaged measurement of troughs against peak (although that makes it somewhat "averaged". I recognise the difficulty in deciding a minimum

---

Sorry, you'll have to rephrase, it seems that this is a very long piece of reasoning compacted in 3 lines, but it sounds interesting.

---

Simple stuff. Nothing complex. If you want to take account of the noise floor (and I'm saying you shouldn't because once recorded the noise IS the signal) you would need to reason a minimum value. Personally as the noise is now considered signal then the minimum value all in is the one to choose. However - really you should look to decide you signal value in some kind of amplitude "window". What IS the minim of the voice as it tails off? Where DO we decide the sound has ended?

But basically - the dynamic range of a signal is only it's loudest amplitude minus it's quietest amplitude. Including noise!! The dynamic range of something peaking at 0dB(FS) that has lots of digital silence is the full 16bit or 24bit range. Now - if those peaks were limited then the dynamic raneg used to be MORE than 16 or 24bit would hold but it has been reduced by limiting. The peaks (notably the transients although not always) have been curtailed.

Edited by Jennifer Jones (30/09/11 11:06 AM)

Quote narcoman:

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I can see where you might get the notion that SNR is increased - but you are only talking about the inherent system noise within the gain system POST limiting. However all that has happened in a "limited by 6dB" file (i.e. one in which 6dB has been removed from the top end) is the difference maxima and minima have been reduced. The crest factor is reduced, the RMS is raised (if we apply the 6dB gain to get the limited signal louder and the noise floor WITHIN the signal recorded is raised the same amount. Clipping transients is the SAME as removing dynamic range.

---

I agree with you --- but where did you read that I was limiting a file? I'm limiting a signal in order to put it in a file.

When you say: "but you are only talking about the inherent system noise within the gain system POST limiting"
Why yes, that's what I do... that's exactly what I'm doing. Never stated otherwise, or maybe I wasn't clear enough.

Edited by Jennifer Jones (30/09/11 11:06 AM)

Quote Emmanuel D.:

---

I agree that the box about how limiters increase the dynamic range of the medium can be easily contradicted when considering other approaches of the dynamic range of a medium. However, you will agree that signals with a lower crest factor may allowed to show a higher RMS variability, given that the medium is the same. So that the use of limiters may result in higher "musical dynamics" (Forte.... Mezzoforte... Piano...)... and that was an interesting and useful point.

---

Absolutely - musical dynamics can easily be inferred by use AND abuse of limiters. But they are not a scientific measure! It's perfectly simple to produce a mix that elicits no excitement in the user (and they refer to it is flat or boring) yet could easily be shown that it is, in fact, very dynamic. The difference between the artistic term "dynamic" an the quantifiable one is a brutal line! The issue of limiters reducing dynamic range isn't a musical one but a technical one with all the associated wearying distortions.

Quote Emmanuel D.:

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May I suggest that as far as mathematical concerns go, you PM me? Your approach is intriguing and I want to know more, but I fear that it may scare off many readers?

---

It was merely a flight of fancy - but I don't think producing a minima from averaging of peak values in noise carries any more weight than seeking it's RMS. Suffice to say - the signal and noise become the signal without reference to SNR once recorded (not withstanding any new noise introduced through re-quantisation of signal in the gain process).

Edited by Jennifer Jones (30/09/11 11:06 AM)

Quote Emmanuel D.:

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I agree with you --- but where did you read that I was limiting a file? I'm limiting a signal in order to put it in a file.

When you say: "but you are only talking about the inherent system noise within the gain system POST limiting"
Why yes, that's what I do... that's exactly what I'm doing. Never stated otherwise, or maybe I wasn't clear enough.

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yeah - that's what I think after reading your posts on here. It's the article that isn't clear - not your thought process.

When limiting a signal at source (a real voice, or a synth etc) you're still raising the noise floor and reducing the dynamics of the original signal. Probably in a pleasing way - but it's a reduction none the less. The "file" bit was an assumption on my behalf. The problem with modern limited masters is the pre-recorded digital source being squished into CD for some willy waving contest on CD changers

Edited by Jennifer Jones (30/09/11 11:06 AM)

Quote narcoman:

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Absolutely - musical dynamics can easily be inferred by use AND abuse of limiters. But they are not a scientific measure! It's perfectly simple to produce a mix that elicits no excitement in the user (and they refer to it is flat or boring) yet could easily be shown that it is, in fact, very dynamic. The difference between the artistic term "dynamic" an the quantifiable one is a brutal line! The issue of limiters reducing dynamic range isn't a musical one but a technical one with all the associated wearying distortions.

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Indeed we completely agree with each other. That's the main point of the article, nicely phrased.
Still, many people have been saying otherwise. And that's was why I wrote the article....

Edited by Jennifer Jones (30/09/11 11:06 AM)

ha!!

Reckon that'd be a useful sig strip for both of us : "beware the distinction between dynamic and dynamic".

Edited by Jennifer Jones (30/09/11 11:06 AM)

This reminds me of the two wizards fighting in The Raven.

Anyway a good thread that seems to have resolved itself.

--------------------
http://soundcloud.com/richie-royale
http://www.mixcrate.com/richieroyale

Edited by Jennifer Jones (30/09/11 11:06 AM)

Quote Emmanuel D.:

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Let's start with the dynamic range of a medium.

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I think this is one area where there might be confusion in the terminology. Audio can have a dynamic range, but as far as I'm aware a medium can't. A medium has a signal-to-noise ratio, and we try to fit the audio signal's dynamic range inside the available rage described by the signal-to-noise ratio.

That signal-to-noise ratio is a measure of the averaged rms noise floor relative to either the peak signal level or the nominal operating signal level, depending on the device in question, expressed in decibels (with no reference suffix).

Quote:

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It means that the SNR of a medium expressed in RMS values depends as much on the signal as to the medium

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Er... no, I'm not convinced (see above). SNR and dynamic range are not interchangeable terms. SNR is a clearly defined measurement.

Quote:

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One thing is sure, the signal's peak cannot go above 0dB FS.

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Actually the 'reconstructed' signal can, quite legitimately -- read up on intersample peaks. Although I do understand the point you're trying to make here.

Quote:

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Conclusion, using a limiter will increase the medium's SNR.

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I have to say I'm not at all convinced by the logic of your argument, not least because I'm unclear on your base assumptions.

hugh

--------------------
Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 11:07 AM)

..but having waded through the discussion between Emmanuel and Narcoman the mists are lifting!

hugh

--------------------
Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 11:07 AM)

Hugh:

I will try to proceed step by step, and also to avoid terminology issues.

We can agree on the fact that for any recording medium, there is a limit above which the signal gets unreasonably degraded. Or degraded above a certain "normative limit" deemed as unacceptable. The degradation might be harmonic distortion, digital clipping, or anything else. This limit can be well-defined, or the phenomenon can be progressive, it depends. In this case, the "normative limit" will be used as a reference.

Also, for any recording medium, there is background noise. It can appear under the form of white noise, or vinyl clicks, or anything else. The background noise is often locally unpredictable, but globally predictable.

Let's say I've got an audio source. This audio source emits a signal, which also features a background noise. I will suppose that the source's signal's background noise is always softer than the recording's medium background noise, no matter what. It's negligible. This is seldom the case, but that aspect can be taken into account afterwards if necessary. All equipment noise except which of the recording medium is also ignored.

First case: let's imagine that the signal features a very low crest factor. There are no peaks, and the signal is very stable. I record it onto the medium. I increase the input level until the medium's upper limit as described is reached. I write down the input level setting. Let's write it as U1 dB. Now I decrease the input gain, until the signal is barely audible. At this point, it gets progressively masked by the medium's background noise. This input setting will be written as D1 dB. For this medium, for this signal, the available range, or whatever else you want to name it, is U1-D1 dB.

Second case: let's imagine that the signal features a very high crest factor, with very salient, short peaks in an otherwise stable context. The maximum input level, before the medium's upper limit is reached, is U2 dB. The minimum input level, where the sound begins to be masked by the medium's background noise, is D2 dB. The available range is U2-D2 dB.

Doing that, I will find that U1-D1 > U2-D2. Why is that?

Signal 2's peaks will reach the medium's upper limit while the majority of signal 2's content, or waveform, as you wish, will be much below. If signal 1's crest factor is C1 dB, and signal 2's is C2 dB, when the medium's upper limit is reached, then the RMS of signal 1 will be roughly C1 dB below this limit. As for signal 2's RMS, it will be roughly C2 dB below the upper limit. I said "roughly". This means that, approximately, U1+C1 = U2+C2. I suppose someone will rightly point out that I don't have the right to write this from a mathematical perspective, because a gain and a crest factor are not the same notion, but if you do the experiment yourself, you will indeed find that U1+C1 = U2+C2.

At the other end of the range, we have approximately D1 = D2. While signal 2's peaks will not be masked, the rest of it will be, and the majority of signal 2 will begin to be masked by the background noise near the same input gain as in the case of signal 1.

So, on the one hand (signal 1), I was able to use U1-D1 dB, with U1 = U2+C2-C1, which means I was able to use U2+C2-C1-D1 dB, and, since, approximately, D1 = D2, to use (U2-D2) + (C2-C1) dB.
On the other hand (signal 2), I was able to use U2-D2dB.
Since C2 > C1, C2 - C1 > 0. It means U1-D1 > U2-D2.

Conclusion, with signal 1, I was able to use more "range" or whatever people call it, than with signal 2.

This is the first part of it - Hugh, are you OK with this part at least? If so, I proceed to the second part.

Edited by Jennifer Jones (30/09/11 11:07 AM)

okay - yes. But the uninteresting fact here is that the dynamic range of U2-D2 is greater than U1-D1.
U1-D1 > U2-D2 isn't some great revelation, it's obvious .

It would be an issue if the dynamic range of the recording medium was 10dB but you're piece of audio to be recorded had a wholly normal 30dB dynamic range. If it was you'd compress, limit or otherwise distort the 30dB range to get it into the 10dB useable audio range. However even the most basic recording medium performs far better than that!

But okay, let's agree you've set up the basics of your premise.

Edited by Jennifer Jones (30/09/11 11:07 AM)

ah come on, I'm moving cautiously, because I want to be sure to be understood
it is a delicate topic, and even you told me that I should be extra careful about formulation
that's what I do

Edited by Jennifer Jones (30/09/11 11:07 AM)

okay. yes. I'll shut up until you've finished!

Edited by Jennifer Jones (30/09/11 11:07 AM)

I'll wait for Hugh's feedback before I go on. Tomorrow, probably...

Edited by Jennifer Jones (30/09/11 10:56 AM)

Best. Thread. Ever.

There is no going back to 'how to I get that dubstep wobble bass?' after this!

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Edited by Jennifer Jones (30/09/11 10:56 AM)

I guess the reason this has caused so much heated debate, and some very intelligent discussion is because it cannot be argued that really great engineering and songwriting has ever been ruined by too soft an approach to master limiting or bus compression.

It is very easy to argue that horrific brick wall limiting has caused and is very easily misused to cause square wave distortion and other horrible artifacts that have ruined records.

Now the problem is no matter how much devils advocate you want to play, is that it just kinda seems like your defending this trait that is only really applicable to the last 20 years or so. Sometimes it's not how good your argument is or how true your facts are. If it looks like your encouraging ants to the picnic people will think you actually are.

Remember music is such a subjective thing, and people LOVE it. Absolutely incomprehensibly love it, and when it's ruined by something. And that something is misused to cause it. And you come and say there's nothing wrong with loud records, and in fact music is actually no "louder" depending on what sort of timescale you use, then a backlash will most likely occur. Now I have to track down a vinyl version of Them Crooked Vultures.

--------------------
Squarehead Jam Jar Facebook Jam Jar

Edited by Jennifer Jones (30/09/11 10:56 AM)

Quote Emmanuel D.:

---

This is the first part of it - Hugh, are you OK with this part at least? If so, I proceed to the second part.

---

Yes, I'm with you, although I don't think you have specified the nature of the input signals adequately to define them clearly and the conclusion is therefore potentially misleading.

Quote:

---

Conclusion, with signal 1, I was able to use more "range" or whatever people call it, than with signal 2.

---

The signal-to-noise ratio of the medium obviously hasn't changed, and neither has the dynamic range of the two signals, which are, I would contest, exactly the same. They must be because the highest elements of both signals sit at the normative maximum limit of the system, and their lowest elements are just masked by the system noise floor.

Let's denote the normative system level limit as 'Lmax':

Then Lmax = U1+C1 = U2+C2

In your example C1 is close to zero, and D1=D2

Therefore, surely, the two signals actually have exactly the same total dynamic range as far as I can see, but C2's 'dynamic range' will appear to be lower that C1's if you use an rms measurement and ignore the crest factor differences.

If the two test signals have been generated such that their peak amplitudes are the same, the same input level must be required also.

hugh

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Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 10:57 AM)

Quote Hugh Robjohns:

---

Yes, I'm with you, although I don't think you have specified the nature of the input signals adequately to define them clearly and the conclusion is therefore potentially misleading.

---

OK so let's suppose signal 1 is a triangle wave. Its crest factor is approximately 5dB.

Signal 2 is the same triangle wave with 0.05s bursts of white noise every 1s. The white noise burst peaks are approximately 10dB over the peaks of the triangle wave, which gives us a crest factor of 13 or 14dB (10+5-epsilon), epsilon being the small influence of the bursts on the RMS.

At this point, this value of 14dB may be discussed. With a stationary waveform (a sine wave, a square wave, a triangle wave...), the crest factor is easily defined. With a non stationary waveform such as signal 2, it's more difficult: you have the choice between different crest factor measurement methods. The output may depend on the length of the window that you'll use to measure the RMS, and with both the length of the window and the model you will choose to measure the peaks. One possible method would be to measure the RMS of the complete signal (no windowing), and the absolute peak of the signal (still no windowing), it may be acceptable in this case since signal 2 is very periodic.

In any case, there is a value that we will call "crest factor", which is much higher in the case of signal 2 than in the case of signal 1. Let's suppose that for signal 2, it's 14dB.

We can also suppose that the RMS of the background noise of the medium (Rmin) is -40dB given a reference that was previously decided, and LMax is 0dB.

I will also suppose that the triangle wave (signal 1) will be masked by the white noise when its RMS is 10dB below which of the white noise. It may be another value. Signal 2 will be masked roughly at the same level.

In case of signal 1, I may use a "range" of the input gain of: LMax - Rmin - C1 + 10 = 0 +40 -5 +10 = 45dB.
With signal 2, I may use a "range" of the input gain of: LMax - Rmin - C2 +10 = 0 +40 -14 + 10 = 36dB.

To answer your second comment, I'm not saying anything here is "dynamic range" or "loudness range". I'm just observing what I can do with the input gain knob before I get into trouble (too loud, or not loud enough).

I suppose we're still OK?

Edited by Jennifer Jones (30/09/11 10:57 AM)

My word this is starting to turn into the PFJ meeting in the Life of Brian

see here if you don't know what i mean... http://www.youtube.com/watch?v=YawagQ6lLrA

As "interesting" a subject as this is, when do any of you guys actually get anytime to write, produce, master or make any actual music? (Answers really not necessary)



Edited by Jennifer Jones (30/09/11 10:57 AM)

Quote Emmanuel D.:

---

To answer your second comment, I'm not saying anything here is "dynamic range" or "loudness range". I'm just observing what I can do with the input gain knob before I get into trouble (too loud, or not loud enough).

---

Yes. All you are saying is that to accommodate the 10dB louder noise bursts you added to the original triangle signal you've had to turn the input sensitivity of the system down by 10dB.

Not rocket science... I'm still with you!

hugh

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Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 10:57 AM)

Quote gazola11:

---

My word this is starting to turn into the PFJ meeting in the Life of Brian

see here if you don't know what i mean... http://www.youtube.com/watch?v=YawagQ6lLrA

As "interesting" a subject as this is, when do any of you guys actually get anytime to write, produce, master or make any actual music? (Answers really not necessary)

---

Of the four people or so left in the world who even understand this nonsense I wonder how many are thinking of packing in music production for an easier life?

Edited by Jennifer Jones (30/09/11 10:57 AM)

Quote Hugh Robjohns:

---

Not rocket science... I'm still with you!

hugh

---

You may be the only one Hugh...

Edited by Jennifer Jones (30/09/11 10:57 AM)

i was initially reminded of this: error

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ohm's where the art is

Edited by Jennifer Jones (30/09/11 10:57 AM)

Quote Hugh Robjohns:

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Yes. All you are saying is that to accommodate the 10dB louder noise bursts you added to the original triangle signal you've had to turn the input sensitivity of the system down by 10dB.

---

Not exactly: if that was the case, if I was to turn the input sensitivity of the system down by 10dB, then the "lower limit" would also be 10dB below.

As it is, only the "upper limit" has to be changed.

Edited by Jennifer Jones (30/09/11 10:58 AM)

I'm enjoying this thread, although I am lagging behind the curve by a couple of days.

I think that at least some of the general confusion on this subject comes from listeners blaming lack of "dynamic range" (whatever your definition thereof) for all audible distortion. I'm sure that there are a few people who hear nasty cases of limiting, dislike the way it sounds at the higher amplitudes and then name the monster "dynamic range", whether or not this is actually a factor.

--------------------
'These are my principles and if you don't like them...well, I have others' (Groucho Marx) www.ownlittleworld.net/tunes.html

Edited by Jennifer Jones (30/09/11 10:57 AM)

Quote Pangloss:

---

I'm enjoying this thread, although I am lagging behind the curve by a couple of days.

I think that at least some of the general confusion on this subject comes from listeners blaming lack of "dynamic range" (whatever your definition thereof) for all audible distortion. I'm sure that there are a few people who hear nasty cases of limiting, dislike the way it sounds at the higher amplitudes and then name the monster "dynamic range", whether or not this is actually a factor.

---

Well, at least IMO, that's it exactly... at least an important part of it.

Edited by Jennifer Jones (30/09/11 10:58 AM)

A general comment about the thread, if I may.

I am aware that some aspects of the issue may seem austere, and I apologize to the readers that may feel lost when it gets too technical. Such "technicality" is definitely not required to actually be able to perform well in production - although it may be at time interesting if one wants to understand precisely what's going on.

There is indeed a certain amount of confusion as far as the notions of "dynamic range", "loudness range", "crest factor" are concerned. The only way to try to set the issue straight is to get to the bottom of things, and use very technical language.

Paradoxically, the actual issues behind it are pretty simple. The "proof" I'm currently in the process of cautiously rephrasing, concerning the fact that the actual "dynamic range" of a medium (mind the quotes, I don't want to get stuck in another vocabulary dispute yet again) is dependent on the signal that's stored on the medium, and that the use of limiters - limiters, not compressors, when set right, may result in an increase of "dynamic range" as far as the audio content on the medium is concerned - this "proof" may sound complicated enough, but it corresponds to a very simple reality, that can be found in many aspects of production. Similarly, it's reasonably easy to implement in ProTools.

The problem is, I don't know how I can at the same time be thorough and not use a technical vocabulary. I hope that once the "proof", and especially what it means in production, appears to be clearly understandable to more people, it will be easier to rephrase in less obscure ways.

It is a real bother: the issue is very simple, but at the same time it's very delicate to put it into words or symbols. It doesn't make it easier that the conclusion is, at first glance, completely counter-intuitive. And that the available vocabulary concerning dynamics is less than approximative.

So - forgive me if it's this austere, we're doing what we can here...

Edited by Jennifer Jones (30/09/11 10:58 AM)

I remain skeptical and unconvinced, i'm pretty sure there's something amiss, but i'm missing it in the convoluted linguistics and misuse of terms.

perhaps reading all this at 2.30 am isn't the best idea i've ever had.


I have issues with your use of terminology, and semantic gymnastics.

and despite the wish not to get bogged down in arguments about linguistics , the fact is you're going to get an awful lot of engineers scratching their heads and disagreeing, (even if you might be right about what it is you were trying to show) simply because some of the ways you have chosen to express things, are basically wrong in terms of the actual use of terminology.




that said, some of the things you are trying to express are not always commonly thought about, and many may be out of practice in actively using and comprehending the significant differences between otherwise similar concepts, that are expressed in similar reference terms...

like SNR and dynamic range.


I'm sitting back and waiting for Hugh and Narco and you to thrash out the semantics before i start kicking the tyres on the math, core concept, and reality.

but the debate IS worth having.... and is interesting to read....

Edited by Jennifer Jones (30/09/11 10:58 AM)

Well, don't hold your breath!! As much as I enjoyed talking with E - clearly a lucid and in depth thinker - there is nothing scientific about this at all! Now - the musical apparent dynamics are recorded, but they are recorded distorted. Either clipped or compressed, whichever takes your fancy. Certainly manages to keep a recorded representation of the performance in place but the dynamics themselves, the quietest to loudest signal, have been compromised to record a distorted version on the medium. there is no proof so far but there is an elegant summing up of why we compress and limit when we record. We do it to preserve some semblance of the dynamics in the performance in an otherwise dynamically inadequate recording medium.

In other words - there is nothing to prove as this is an aesthetic debate; not a mathematical one.


Couple of things:

A limiter IS a compressor; in practical terms one with a very high ratio (although in analogue terms actually not that high at all!!). You COULD clip instead of limit.

If you do limit you lop off transients. The apparent power of that transient may still appear in the recorded medium (you'll still hear the snap of the drum) but it's an illusion. The information is lost on the distorting nature of all limiters.

Emmanuel, I suspect that in order to illustrate your point some examples will have to come from your end. Without a mathematical basis I'm also afraid you can't have "proof". You might have empirical evidence supporting your aesthetic but you'll need expressions to show it as a proof.


Oh - and again, you cannot use an RMS value as a unit for unit variable against a peak or trough (a max or min) value since RMS is an abstract time domain summation. As I pointed out earlier - you would need to form a valid expression and compare the differential WRT time as your max or min values are instantaneous measures. You're making a similar error as to comparing momentum with impulse. In other words you can't use the RMS of the noise value as a variable of the lower bound for the "dynamic range" (hoho). You must use a measured value (or an average of measured minimums for safety) as a representative instant measure. An RMS value is, for all intents and purposes, an integral of dynamic values over time;

Although we take sample values and find the root of the sum of the squares/number of square this is a discrete version of :


let Fn{T} = the integral WRT time of all f[t]^2 (cus I can't figure out how to form an expression here!!)

Then Frms = limt(T->∞) √((1/2T) Fn{T})


Now - that's just one part of your problemo - the rest of it is semantics and definition. But this part - you cant use RMS of noise to be your noise floor. It's not representative of the noise floor or the average of the noise floor. It is the magnitude of the variation of the noise floor. We need a static value. None of that should change the basis of your argument though, but I'm afraid it's an argument of aesthetic viewpoint than revelation!!

Edited by Jennifer Jones (30/09/11 10:58 AM)

nicely put mate.... but all that aside.... I know why I'm up at this ungodly hour, but what the hell's your excuse????




Edited by Jennifer Jones (30/09/11 10:58 AM)

couldn't sleep.

Edited by Jennifer Jones (30/09/11 10:58 AM)

on nights like this we should go down the studio and ~Jam...


(or at least if i was physically in any shape to..... Neck grief is keeping me awake, amongst other things.... )

Edited by Jennifer Jones (30/09/11 10:59 AM)

Narcoman:

1. you promised to wait until I was done with the explanation, and here you go, a long contradicting post right in the middle of it
2. you always forget about the quotes I put around the words so as to avoid fights over terminology, and try to confront me right away on a terminology issue, the last example being about a "proof"; I was pretty sure you would jump on this one, and here we go. And yes english is not my first language so be tolerant.
3. why do so many mathematicians always try to ridicule the efforts of people who use mathematical concepts to describe some part of "reality" for the ground that it's not as thorough as they would like? In short, what's your problem with physics?

Don't you see the problem here? I'm currently pointing at some apparently very simple phenomenon: if you got a sample with a very salient attack, you can't make it as loud as if the attack wasn't there, because the attack will peak . Is that so unacceptable to you? And because I measure it by comparing two heterogeneous measures, you tell me that it doesn't exist??

I don't mean to be rude, but you remind me a bit of the guy who said that if an UFO landed in his garden, he would just close the shutters and look elsewhere, because the UFO itself would go against his principles and his logic.

* And can you please stop repeating that I can't subtract an RMS level from a peak one?

If I take a window of 1s, and measure for this window the peak value, and then the RMS (which you'll probably say doesn't exist in discrete spaces as such), I have the right to subtract one to the other. It's a perfectly legal descriptor.

If I measure the signal with a pseudo-peak meter on one hand, and with a VU meter on the other hand, I have the right to subtract one to the other.

Also, please stop challenging the definition of RMS of a signal, or do it later. I'm using the definition that each and every person in music information retrieval uses, which is the one implemented in this particular Matlab toolbox: https://www.jyu.fi/hum/laitokset/musiikki/en/research/coe/materials/mirtoo lbox

If you want to challenge that, do it after the subject is closed because it's another issue.

Quote:

---

In other words - there is nothing to prove as this is an aesthetic debate; not a mathematical one.

---

There is no such dichotomy, as you state, as "mathematics" on one end and "aesthetic" on the other. You've got a whole world of different domains in between. The domain I wish to address here is akin to both signal processing and music information retrieval. These domains allow to introduce approximations and shortcuts. Please stop answering in terms of mathematical paradigms, because it's not pertinent. If you disagree with the very principle of physics, I can't help it. I really wonder how you can deal with psychoacoustics - at all. Do you even find it acceptable? Because in terms of mathematical formalism, psychoacoustics implies quite a lot of absurdities. Yet, it's being used, and with great efficiency (mp3 coding for instance).

And for f***'s sake, Narcoman, understand that I'm not trying to fool people, and I know what I'm doing in the field I'm speaking of. For the moment, I'm trying to show and explain a method that enable mixers to recreate a certain amount of musical dynamics (FF, MF,PP....) when the environment in which the music is played is hostile, with a very high background noise. Narcoman, I really don't see what's wrong with that, and I'm a bit frustrated that you tell me that this method doesn't exist because it does not show the qualities you would like it to show.

Now can you let me finish, please!!!

Edited by Jennifer Jones (30/09/11 10:59 AM)

1. I'm not having a go. And your quotes and careful nature have been noted. Physics and Maths are inextricably linked and it's an area I'm very comfortable in. I hold a fairly unique position of mixing for a living (on AAA product) and coming from a maths background. I'm exactly who you need to discuss this with!!!:). Get some of this ironed out and you might have a great paper.

2. I'm not telling you RMS doesn't exist - I'm giving you a reason why you can't use RMS as a value for the minimum of the range as it's a measure of variability. You CANNOT say it is valid to take a peak value and subtract the RMS value and maintain that is a measure of the dynamic range. It's like taking the peak of a value and subtracting the standard deviation, it doesn't give you a useful range. However, this is only part of what you were referring to and doesn't impact on your point anyway..... I quote "If I measure the signal with a pseudo-peak meter on one hand, and with a VU meter on the other hand, I have the right to subtract one to the other. ". You have the right. But it doesn't give you any meaningful information.

3. Re: the UFO. Ahem - I'm afraid that's you. You've become blinkered by your own research. This isn't new. This isn't research. We already do this. But WE KNOW we lose information. You're speaking about perceptual information - perceptual information, phsycho-acoustics etc isn't real. It's the brain filling in the gaps.

4. I'm not fighting you on terminology. I'm explaining some loopholes in your method; in particular your persistence to using RMS as a valid minimum. If this is a paper for a journal be thankful I'm not one of your reviewers!! . You're conclusions on psycho acoustics still carry water although not any great revelation. the problem with your article was one of presentation - as are the argument here. You are presenting this to engineers, mathematicians and generally very scientific people. I have no issue with the "perceived information" element.

5. Psycho-acoustic phenomena explains THOROUGHLY what you are referring to. And it's what I initially brought into the debate; that of it being perceptual rather than actual. Psycho acoustics (including MP3) carries solid maths. As will yours when we get some of the difficulties ironed out

6. The only insult (and it's a minor one) is you think you're telling us some new method! We've all been limiting or compressing signals to give the illusion of more information for many many years! This is really a semantics debate though isn't it?

7. I thought you'd finished. All you had to do was finish your point. I'm not accusing you of trying to fool anyone. I'm accusing your methods of being wrong so your conclusions are based on some shakey ground. I've also given you ground on your point still being observable despite this.

(I'll tell you what's wrong with it - you're trying to teach people something we do on a daily basis. And I'm not trying to insult you - you're trying to show us a method for keep F and PP passages in the same recording so they are intelligible - but what the hell else do you think we USE limiters for? )

Edited by Jennifer Jones (30/09/11 10:59 AM)

I think also , if we discuss perceptions , then it must be said that as well as my rigorously thought through "hoover effect", another aspect is the effect that limiting and compression has on the sound of the audio. I would suspect that despite more and more devices becoming transparent , that to many trained ears, it is detected, in a similar manner to a mild background/foreground hum. Too many modern productions suffer from this "sheen effect" and the overwhelming sensation is that we are losing a sense of distance in recordings, and by that I mean that in potentially quieter sections of music it feels as if the performer has been shoved up my nose.

Now as an effect for a genre that has integrated these recording techniques into the sound of said genre , I have no problem...just don't expect me to like it! Dave

--------------------
My head hurts!

Edited by Jennifer Jones (30/09/11 10:59 AM)

Quote:

---

6. The only insult (and it's a minor one) is you think you're telling us some new method! We've all been limiting or compressing signals to give the illusion of more information for many many years! This is really a semantics debate though isn't it?

---

Look, you don't read what I write. I've been telling again and again that this method is not new. It's been used for a good 60+ years. It goes significantly further than only "cramming the signal's dynamics into the available range" or whatever you name it. If you understand it, fine. Many people don't. Maybe they want to know, or are at least curious if it can stand. Sometimes when you understand a process very well, you perform better than when you do it intuitively.

Quote:

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You haven't told us anything other than "with care - a little limiting can give the appearance of maintaining the dynamics".... Whoopy do!! We all know this. It's not new! It's not research!

---

OK so you don't wait for me to finish the explanation and then you complain that it's not finished? WTF?

This is becoming frustrating, mainly because of the constant obstacles you keep putting in the path of what I'm trying to say. I don't get "blinkered" by my research or anything like it. I'm not forgetting what I'm speaking about. I'm not telling anyone it's new. I like to look at things from a certain point of view, and I like to explain things. Limiting / compressing is a complex issue that's generally not fully understood, so it seemed like a good idea to write about it. That was all there is.

Edited by Jennifer Jones (30/09/11 10:59 AM)

Why are you splitting the explanation into multiple posts then complaining when people comment on the unfinished portions...

Edited by Jennifer Jones (30/09/11 10:59 AM)

I do read what you write. I enjoy it and, most of it, I nod here silently agreeing!

You'll notice I've edited out a couple of my stupid comments. Sorry about that !!:)

Let's just deal in the issue yeah?

Edited by Jennifer Jones (30/09/11 10:59 AM)

Quote:

---

you're trying to teach people something we do on a daily basis. And I'm not trying to insult you - you're trying to show us a method for keep F and PP passages in the same recording so they are intelligible - but what the hell else do you think we USE limiters for? )

---

There is really a huge misunderstanding here. I'm not trying to explain why to use a limiter. I'm explaining how it works. And as far as I can tell, the limiter works in two ways so that F and PP passages are more intelligible. One way is "compressing" the signal's "variability" in terms of "windowed measures". Another way is reducing the "crest factor". Those are two different things.

And the issue is difficult to instantiate. Take a Flux Compressor II, ratio 1:3, variable threshold, other settings at default. Lowering the threshold has no influence on the crest factor whatsoever, only on the "windowed measure variability". Take a Waves L2, with some signals, you can lower the threshold down to -10dB, it has no influence on the "windowed measure variability", but reduces only the "crest factor".

Both processes will keep FF and PP intelligible, but each process has a distinct influence of how the signal sounds after being processed. This is one of the things I'm trying to explain. It's not original research per se, the concept is not new. However, I'm pretty sure many SOS readers are not aware of this distinction, which may be very important in production.

Edited by Jennifer Jones (30/09/11 11:00 AM)

Quote:

---

Why are you splitting the explanation into multiple posts then complaining when people comment on the unfinished portions...

---

Why? Because there is already a post that explained that in one single bit, but it was too condensed. I'm breaking up things so it appears to be more gradual. And I'm complaining about Narcoman's comments because he explicitly agreed in a previous post to wait until the thing was finished before looking for the weak parts

Edited by Jennifer Jones (30/09/11 11:00 AM)

Quote narcoman:

---

5. Psycho-acoustic phenomena explains THOROUGHLY what you are referring to. And it's what I initially brought into the debate; that of it being perceptual rather than actual. As this is about perceived information rather than actual information then it's of very little importance. You haven't told us anything other than "with care - a little limiting can give the appearance of maintaining the dynamics".... Whoopy do!! We all know this. It's not new! It's not research!

---

I strongly get the impression that both of you are wandering about the battlefield, trying to explain issues involving the psychology of aural perception in purely mathematical and/or physical terms. (Though E seems more guilty of this than N.)

Edited by Jennifer Jones (30/09/11 11:00 AM)

WITH a blunderbuss for accuracy.

It's a lot of miscommunication really. Terminology can be a bitch but you've got to get these things right.... it's the only way to communicate concepts.

Edited by Jennifer Jones (30/09/11 11:00 AM)

This is beginning to sound like some of the debates we used to have in my college days when we all crowded into the female common room to discuss things we had not really understood yet!

I can't even remember what they were about any more!

P.S. But don't let me stop either of you silly people - I'm enjoying the new ideas of wandering RMS levels and crest factors!

Quote The Red Bladder:

---

This is beginning to sound like some of the debates we used to have in my college days when we all crowded into the female common room to discuss things we had not really understood yet!

I can't even remember what they were about any more!

P.S. But don't let me stop either of you silly people - I'm enjoying the new ideas of wandering RMS levels and crest factors!

---

So.
You saying I don't understand signals aye?

Despite, y'know, the lil ol' scrap of paper....


(hate to say it though - you can and do have wandering RMS values. It's a facet of windowing any time domain function)


Edited by Jennifer Jones (30/09/11 11:00 AM)

I enjoy the image

And that's basically the problem isn't it?

You've got those plug-ins or units, they're called "compressors", "limiters", "multi-band compressors", "multi-band limiters", they do spectacular things to your signal, they are used for a number of purposes. The question is, then, "what do they do?, how they do it?".

On the other hand, you've got some vocabulary: "attack time", "release time", "crest factor", "root mean square", "peak level", loudness range", "loudness", "dynamic range". Supposedly, those words relate to something precise, but often they don't. For instance, there are almost as many explanations of "Loudness" as there are papers about it.

Plus, you've got a field called psychoacoustics that ought to bring some solutions, but apparently it pays little attention to limiting / compressing. As a consequence, the only way people describe what a limiter does in terms of perception is by using impressions or feelings.

And then, you've got signal processing theory. It gives a lot of information... but it's specific information. Very difficult to relate to anything perceptual.

What does that leave us, in front of our compressor / limiters? We know how to use them to some extent, we know intuitively what certain settings will bring, we have good recipes that come in handy when being confronted to a particular problem... but that's pretty much it.

The right thing to do might possibly be to take each term, "Loudness", "Loudness Range", "Dynamic Range".... and see what they might mean. That's one of the things I was trying to do with the article.

The particular explanation that takes so much time, brings so much conflict, and that would , IMO, bring a little bit of progress in the understanding of dynamic processing, involves at least the following notions: "dynamic range of a medium", "loudness of a signa", "loudness range of a signal", "peak level", "RMS", "Crest Factor". The only term that I thought was well-defined is "RMS". Apparently not
So that leaves us with a discussion involving at least 6 terms that are not well-defined.

Honestly: what would you think are
- dynamic range of a medium
- loudness range of a signal
- loudness of a signal
- peak level
- RMS
- crest factor
?

But I mean honestly - not copying a particular definition found somewhere in the literature.
What do the words mean, really, in terms of production, composition, hearing.... ?
I'm sure there will be almost as many definitions as there will be answers.

In any case, that would be an interesting place to start again from. What do you readers think?

Edited by Jennifer Jones (30/09/11 11:00 AM)

Quote Emmanuel D.:

---

Not exactly: if that was the case, if I was to turn the input sensitivity of the system down by 10dB, then the "lower limit" would also be 10dB below.

---

You can't have it both ways. Either the peak amplitude of the second signal is 10dB higher and therefore you need to turn down the input sensitivity to accomodate it within the upper limit of the medium (and yes, that means the lowest level still disappears into the system noise); OR the two signals have the same peak-peak amplitude but the triangle wave element of the second signal is 10dB smaller than that of the first signal (the upper 10dB accommodating the loud noise burst). In which case, there is no requirement to turn down the input sensitivity at all.

Which is it?

...but I'm not sure we're going to get to thrash this out very effectively here because of the irregular times we each respond and the natural interruptions from others along the way.

I'm happy to continue our discussion via email if you prefer.

Hugh

--------------------
Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 11:00 AM)

@Narcoman

Sorry, man, I realize I was rude with you in an earlier post
I was a bit frustrated... I apologize. Publicly

The argument you propose that the RMS of the background noise isn't a good measure really intrigues me
I understand some of the points you make.
Though - and it's not to contradict you but to try to understand:

(1) Our aural perception is, essentially, windowed. Loudness models, are, quite adequately, windowed as well - or at least integrated. Given certain restrictions, the RMS of a signal is reasonably correlated to its loudness, whatever the loudness model. It is the case with loudness evaluation of a white noise: raise its RMS, its loudness will be raised as well, following a quasi-linear curve. So, if my issue is to select a representative descriptor of the background noise in terms of level and/or loudness, why couldn't I choose the RMS?

(2) The only constraint I would have, if I want to properly compare the background noise level to the highest level a medium can handle, is to select another windowed descriptor for the highest level.

(3) The particularity of this method would be that my measure of the highest level the medium can handle would be dependent on the analysis window size, and on the signal. This is, apparently, one major point of disagreement.

On the other hand, would I choose an instant measure of both values, I would find myself with measures that are highly de-correlated from any notion of loudness. This use of an instant descriptor would defeat the original purpose: I wouldn't be able to evaluate anything related to musical dynamics.

Also, I don't understand your sentence about the RMS being the magnitude of the variation of the background noise. RMS is a particular average of a sequence of values. It's not a variance. Did I miss something here?

So - Narcoman, what's your point of view here?
Thanks!


[Added later]
No wait, the RMS can be a variance if you consider both negative and positive values of the signal. But if you consider the absolute value of the signal, then it's not a variance anymore. And, I think it ought to be considered in regards to the absolute value of the signal, since this absolute value is the basis on which the ear evaluates loudness. I'm pretty sure that must be the key to solve one of the disagreements

Edited by Jennifer Jones (30/09/11 11:01 AM)

@Hugh

Yes, maybe it would be a good idea to go on this particular conversation via email - in a few days, though: I have to think about how to present that in a clearer way, be it with a couple of diagrams or better yet, with audio examples....

e

Edited by Jennifer Jones (30/09/11 11:01 AM)

Diagrams in addition to the ones you did would only help to illustrate your point. They'd be very welcome.

I'm beginning to get the purpose of your argument though - and I don't disagree with the perceptual stuff. It's only some of the reasoning that I'm questioning.

Re RMS : even as a static value it's still a value based on an integral. It doesn't really stand for any particular value per se.


Oh - apology not necessary I'd assumed you were just frustrated with my continual counter point !!

I'm going to be out of it a couple of days (taking a break on the seaside and leaving in five mins).
I'd certainly welcome further off line discussion!! Feel free to PM me and we can set up some kind of less"forum based" chat!

Edited by Jennifer Jones (30/09/11 11:01 AM)

How about taking this to the pro audio mailing list?
Lots of serious audio types there that may have some insight.

Please do NOT take this thing to private email, there is interesting content here on all sides of the discussion, and I am sure I am not the only one that is finding the discussion interesting.

Regards, Dan.

--------------------
Audiophiles use phono leads because they are unbalanced people!

Edited by Jennifer Jones (30/09/11 11:01 AM)

Indeed so. I've bookmarked this for re-reading as many times as it takes until I understand the arguments properly.

CC

--------------------
Put the fun back into dysfunctional.
Mr Punch's Studio

Edited by Jennifer Jones (30/09/11 11:01 AM)

Sorry to be so late in returning, it took a while to read all the intervening posts, and I'll admit I skimmed a few that seemed to break no new ground, but will catch up shortly.

Wow. I feel like the boy who lit the match in the outhouse. There's more going on here than an argument over semantics, there's also the one over nomenclature, but this is all mere obfuscation, when the over-arching problem with the article is that it purports to reveal new information using a revolutionary new process, and it most decidedly does not. More on that, later.

Quote:

---

ED:"But there is an immediate problem: the top of the range is an instantaneous value, and the bottom an averaged one.
Evaluating the dynamic range, you'd be subtracting an RMS level (or something not far from it) to a peak level. Now that doesn't make sense."

---

But it makes perfect sense, as mastering engineers do this all day, every day; it's called adjusting the crest factor. To an audio engineer, as opposed to an electrical engineer, there are two common meanings for the term dynamic range; when referring to a digital system it describes the difference between the smallest and largest sound which can be recorded, but when referring to a signal within such a system, it refers to the difference between the smallest and largest signals that have been recorded, a slightly different meaning than crest factor, which is the ratio of instantaneous peaks, AKA transients, and average level integrated over a specific range of time according to a specific formula, AKA RMS average. I believe this is the source of most of the confusion, here.

Mr. Deruty, I find your initial characterization of your explanation as being provided by your editor in support of your arguments deceptive, perhaps intentionally so. Matters of potential intellectual dishonesty aside, here's the rub:

Quote:

---

ED:"One thing is sure, the signal's peak cannot go above 0dB FS. So the SNR of the medium is, still from that RMS perspective, related to the ratio between the signal's peak and its RMS. In other words, to the crest factor. If the crest factor decreases, the "RMS SNR" increases. And, any processing that reduces the crest factor will increase the medium's SNR when expressed in terms of RMS. Conclusion, using a limiter will increase the medium's SNR."

---

This is exactly backwards, as Narcoman, and I are attempting to express; when you reduce crest factor you decrease the signal to noise ratio, because you've increased the noise relative to the peaks, along with the average. Bringing up signal to noise ratio is completely irrelevant and simply muddies the waters; I begin to suspect an agenda.

Quote:

---

Narcoman:"This is the fundamental disagreement - removal of transients IS the removal of dynamic range based on the correct dynamic range calculation. If you want a mathematical idiom to refer to - when you deal in two RMS values from the same source you are including and RMS of the included noise TWICE. As you limit the signal you merely bring the noise UP along with the signal. The dynamic range lost is precisely the amount of limiting done (of course, if no limiting take place then you're just moving the window)."

---

Furthermore, a limiter acts only on the peaks, not the RMS average, until you add the makeup gain, at which point you have decreased the crest factor, and most likely clipped, flattened or squashed the transient peaks, resulting in several kinds of distortion, (harmonic, intermodulation and smearing of transient information in the time domain) by which I mean the signal will no longer be linear in just about every way it can be, so Narcoman is correct:

Quote:

---

N:"When dealing with the issues in limiters (that of removed transients) the RMS variability doesn't come into play. It just doesn't matter and has no effect on what the concerns over the loudness wars are about - that of loss of detail through removed transients."

---

Also, I believe Emmanuel means "dense" when he writes "austere", not that it's germane to the discussion.

The larger problem is Mr. Deruty's use of the new-and-improved EBU specification to produce a different result than the other methods, as detailed in the graphic illustrations accompanying said article. This is precisely what such a specification is designed to do, to fool legislators into allowing broadcasters to favor commercial advertisements over their content in terms of loudness, but they can then say, "No we didn't, let me prove it with this new math." THAT is what I meant by "bollocks". A similar spec was foisted on the American public some years back, with the result that I must enjoy my television entertainment with the remote control firmly in my grasp to prevent being blown out my front window by the next blaring advert, the alternative being that I can't hear the dialogue at all.

I'm going to say this very politely, ... just once.


pick as many holes in his argument as you like, I'm broadly in the Narcoman camp on this, but accusations of intellectual dishonesty and hidden agendas are a personal attack, and have no place in this discussion....


furthermore, I'm delighted that there is robustly healthy and erudite debate on the matter , and that it is pretty civilized.

if we can keep it that way, it stands to benefit everyone in the long run.


so....


please ,


play nice.






(note, i am not known for being diplomatic, or polite.... i'm usually described as blunt and about as subtle as a house brick in the face. )

Quote Emmanuel D.:

---

Quote Hugh Robjohns:

---

Yes, I'm with you, although I don't think you have specified the nature of the input signals adequately to define them clearly and the conclusion is therefore potentially misleading.

---

OK so let's suppose signal 1 is a triangle wave. Its crest factor is approximately 5dB.

Signal 2 is the same triangle wave with 0.05s bursts of white noise every 1s. The white noise burst peaks are approximately 10dB over the peaks of the triangle wave, which gives us a crest factor of 13 or 14dB (10+5-epsilon), epsilon being the small influence of the bursts on the RMS.

At this point, this value of 14dB may be discussed. With a stationary waveform (a sine wave, a square wave, a triangle wave...), the crest factor is easily defined. With a non stationary waveform such as signal 2, it's more difficult: you have the choice between different crest factor measurement methods. The output may depend on the length of the window that you'll use to measure the RMS, and with both the length of the window and the model you will choose to measure the peaks. One possible method would be to measure the RMS of the complete signal (no windowing), and the absolute peak of the signal (still no windowing), it may be acceptable in this case since signal 2 is very periodic.

In any case, there is a value that we will call "crest factor", which is much higher in the case of signal 2 than in the case of signal 1. Let's suppose that for signal 2, it's 14dB.

We can also suppose that the RMS of the background noise of the medium (Rmin) is -40dB given a reference that was previously decided, and LMax is 0dB.

I will also suppose that the triangle wave (signal 1) will be masked by the white noise when its RMS is 10dB below which of the white noise. It may be another value. Signal 2 will be masked roughly at the same level.

In case of signal 1, I may use a "range" of the input gain of: LMax - Rmin - C1 + 10 = 0 +40 -5 +10 = 45dB.
With signal 2, I may use a "range" of the input gain of: LMax - Rmin - C2 +10 = 0 +40 -14 + 10 = 36dB.

To answer your second comment, I'm not saying anything here is "dynamic range" or "loudness range". I'm just observing what I can do with the input gain knob before I get into trouble (too loud, or not loud enough).

I suppose we're still OK?

---

Not at all. A steady-state triangle wave test tone has no crest factor. Never did, because such a signal is flat as a ruler and has zero dynamics, unless you've modulated its level in some way. I gather that this is theoretical, but it still doesn't wash, because the noise floor (your background noise) has bugger all to do with determining the loudness, nor the crest factor, nor anything you purport to determine. Simply put, you're using a different method of measuring crest factor, one that appears purposely designed to give an incorrect result.


*edit* Sorry, about misspelling your name in the title, the software tells me it's too late to correct it. Perhaps the moderators could be persuaded?

Edited by Jennifer Jones (30/09/11 10:50 AM)

Quote Sam Inglis:

---

Quote Mojobone:

---

Furthermore, the decibels in this paragraph are not referenced to anything; this makes them essentially meaningless, though I presume dBSPL is what's meant, in context.

---

Surely the dynamic range of a system can be expressed simply in dB without a reference?

---

I presumed the author meant, "upon playback" in the paragraph I quoted, as the sentence makes no sense to me otherwise, though it's still incorrect, as 'dynamic range' is in no way enhanced by limiting, in any sense of the term's meaning.

Edited by Jennifer Jones (30/09/11 10:50 AM)

Mr "Mojobone", we have a lot of problems. To cite only a few:

1) If I understand you well, both the EBU and I are dishonest, and in full knowledge of it.
What about all the scientific papers that prelude to the EBU3342 norm? I suppose they're dishonest too?
What is it then, a conspiracy? LOL

2) Look at what you wrote!
"A steady-state triangle wave test tone has no crest factor. Never did, because such a signal is flat as a ruler and has zero dynamics"
Who do you think you're kidding? Just read one definition of the crest factor, even on Wikipedia: http://en.wikipedia.org/wiki/Crest_factor

3) And this:
"when you reduce crest factor you decrease the signal to noise ratio, because you've increased the noise relative to the peaks, along with the average."
You're confusing the SNR of the signal and which of the medium! I was talking about the SNR of the medium.

With all due respect, I have no time to lose with such nonsense.
I'm sorry for the people who enjoy reading the thread, but as far as I'm concerned, the rest of the conversations will be by email.

I enjoy contradiction, even from Narcoman, and that's saying something At the very least the guy has a point of view, and I respect him for that, even though I disagree with him.

I understand that we may disagree. I understand that you may make mistakes, the issue is difficult. Or I could make mistakes as well. I probably did make mistakes here and there. Mistakes don't necessarily invalidate the point of view. I understand that you may be confused by some particular aspects - after all the crest factor of a stationary waveform is well defined, much less so when it comes to "real" signals. But this.... telling first that the article might be "bollocks", and then telling that you "begin to suspect an agenda".... no.

Edited by Jennifer Jones (30/09/11 10:50 AM)

Well, now we're getting somewhere; you and I don't define "crest factor" in the same way. Or we, do, but we're talking about completely different kinds of signals, as I see no earthly reason for compressing nor limiting a generated sine nor square wave, since the perceived loudness varies not at all. If I need it louder or less loud I turn a knob, no need for fancy levels management, and sustain is never a problem. The reason we're at odds is perhaps because you are limiting signals and I am compressing and sometimes limiting music. The period of your hypothetical triangle wave is too fast for me to hear how a limiter would affect the shape of its envelope, because my ear's integration time is too slow; does this increase your comprehension of my point of view?

It does appear that I missed a thing or two that you posted, and I'll get to that in a moment, but I'd like you to explain please, if possible, why EBU 3342 was promulgated, and also please note that the agenda I suspect is not yours. If there's any conspiracy, you may have uncovered it, congratulations. [*edit* never mind, I'll just go read the dang thing, but I'll hate myself in the morning]

In audio engineering, we look at diagrams that show only the portion of the wave that's on the plus side of zero, so that we can better visualize how our processes are re-shaping the waveform, and perhaps this form of visual training blunts our perception of what's going on conceptually, but I doubt it. (you're probably aware that absolute peaks are preferred over RMS for this purpose, as it has to do with shaping transients, more than managing levels)

In my initial reading of this thread, I couldn't understand why you kept bringing up the noise levels, and signal to noise ratios which I consider irrelevant when measuring loudness and crest factor in my digital audio environment; after all, I work damn hard to keep noise out of my recordings, and therefore negligible to the point of irrelevance, I dearly hope. We're also taught that our DAW systems themselves will introduce very little unwanted noise or distortion, unless we put it there. Finally, I twigged that you were talking about the noise floor in the listening environment and not in my damn tracks. (I hereby agree that it might be relevant) I think we also disagree on the definition of the medium; I'm talking about digital PCM encoding at 24 bits, and you're speaking of um, a car audio system? I'm gonna need a nudge, I'm afraid.

I have to make the gig, just now; I'll have more, later. Please don't shove off just yet, I might be about to learn something and as you might imagine, that hardly ever happens.

Edited by Jennifer Jones (30/09/11 10:50 AM)

http://tech.ebu.ch/docs/tech/tech3342.pdf was a pleasant surprise; AES papers are written so that no one but an electrical engineer can understand them. It clearly explains what the gating and thresholds are for, but they're not really all that relevant in how the "loudness" is actually determined, at least where music is concerned, as opposed to speech. (other than for leaving fades and gaps out of the equation.(by the way, American audio engineers are trained to never use the term "loudness", which is partly what's so upsetting about this whole "Loudness Units" business)


Mr. Deruty, I apologize for impugning your motives and for the rather provocative title of this thread, but you are indeed in error when you suggest that dynamics may be in any way increased by brickwall limiting. (dynamism, maybe, dynamics, not) A great deal of time and money has gone into this new specification, and I believe your graph shows that it won't work; that this new metering system, particularly the windowing in the time domain, has the effect of statistically hiding huge losses in dynamic range. (if we define "dynamic range" as the difference in decibels between the peaks and the valleys in the music) In other words, the windowing scheme may be less suited for metering music than speech, although I'm sure it must have been taken into consideration.Or it could be that this meter so loosely tracks RMS average that it's nearly useless, from a mastering perspective. (I doubt that this is the case, because TC Electronic came up with the algorithm and they know their biz) I'll dig through the papers some more to see if I can figure out why. I'd like to see the results of your computation with a sampling of only top forty singles over the same period, because album content has a dynamic profile more similar to a TV program and singles tend to have a a dynamic profile more similar to that of a television commercial.

Edited by Jennifer Jones (30/09/11 10:50 AM)

Much against my better judgement (and only because Narcoman and Hugh have been discussing this piece) I actually sat down and read the 'offending' article and immediately was struck by the repeated use of muddled and undefined terms, such as 'loudness' and 'dynamic range' - and it became very apparent that the author just does not understand most of the concepts he is trying to discuss.

For example -

Quote Emmanuel D.:

---

Imagine you’ve got an audio file that is normalised: you can’t add any more gain without getting distortion. Using a limiter or a compressor on such a file will nevertheless add gain to its content: the RMS levels will be increased. This adds dynamic range to the medium: instead of being, in the case of a 16-bit file, 96dB, it will increase to perhaps 100 or 105 dB. On the diagram to the right, this additional available dynamic range is illustrated by the grey rectangle. From that point of view, limiters don’t decrease the loudness range, they increase it.

---

With a statement like that, you just don't know where to start! Gain? Gain is the ratio of an input to an output. It is measured in volts and amps and expressed in dB.

Dynamic range? S/N ratio? Musical (quietest note to loudest note)? What?

Quote Emmanuel D.:

---

Many decades ago, engineers would compress the signal between the microphone and the recorder in order to increase the available dynamic range of the recording medium, so that its then low signal-to-noise ratio was less of a problem.

---

Well, I was recording music many decades ago and I don't remember defying the laws of physics then - and I very much doubt that I would be able to defy the laws of physics to today!

We can only speculate, as to what on Earth the author is alluding to! Perhaps he read somewhere about the use of companders (that's a compressor-expander, such as the old dbx boxes and Dolby A etc.) and only managed to understand half the story.

The article starts with the modest claim of being based on 'ground breaking research' and then goes on to compare Lady Gaga with the Beatles and Mike Oldfield with Nine Inch Nails. One is tempted to ask if comparing a rock with a ham sandwich would not be more meaningful, or comparing a poodle called Fred with the Watford bypass illustrate your point better!

All sorts of terms only half understood, or just not understood at all, are chucked about in gay abandon. 'Density' and 'levels' are strewn happily throughout - but right at the top of terms that the author either has not understood (or hopes that we shall gloss over, without inspection) comes loudness.

As Mojobone so rightly points out, loudness is a completely subjective term.

You can no more measure loudness using electronic instruments, than you can measure how good food tastes, or how inspiring a piece of music might be.

Loudness depends on whether you like what you are hearing, the harmonic and disharmonic content, distortion content, variation in perceived volume, stereo image, distribution of frequencies within the signal, what you have heard before, what you are seeing, your state of health and hearing, what else you are doing, the list just goes on and on!

The author seeks to hang the term loudness on the EBU document 'EBU – TECH 3341' which is a description of how broadcast material may be measured for perceived loudness, in order to solve a series of legal wrangles that have been rattling around the broadcast industry for some considerable time. To take a measurement protocol for advertising material and attempt to apply a legal definition onto a subjective term 'loudness' can only be summed up with the one word 'bogus!'

For those of you who are still puzzled as to the subjective nature of loudness, here's a simple experiment to illustrate just one aspect of loudness -

Take a monophonic recording of anything and convert it to a stereo recording by placing the same data on both left and right channels on your DAW. Play that to a test group of people and then take the same signal and time-slip one side by a few milliseconds. The group will almost every time state that the second recording (which is of precisely the same signal, the same amplification, speakers and energy, but with a minor time difference L to R) is somewhat louder.

You could go on, experimenting with distortion, mid-frequencies, adding and taking away the bass, using cheaper speakers and discover all sorts of interesting facts and aspects about the phenomenon that we call loudness!

There is, to be honest, nothing 'ground breaking' about not knowing enough about a subject!

Edited by Jennifer Jones (30/09/11 10:51 AM)

Quote The Red Bladder:

---

...I actually sat down and read the 'offending' article and immediately was struck by the repeated use of muddled and undefined terms, such as 'loudness' and 'dynamic range' - and it became very apparent that the author just does not understand most of the concepts he is trying to discuss.

---

At least you read the article, which is a good start It has been clearly established here already that a lot of the problems in comprehending Emmanuel's arguments and conclusions are down to confusion over the use of terminology and language. However, english is not Emmanuel's first language and I think it extremely unfair and impolite to attack his work as aggressively as many contributors to this thread have.

The article represents a serious body of research, and the comprehensive measurements he has made are interesting and significant in their own right. I believe we owe the author the respect of at least trying to understand his arguments, rather than simply trying to shoot them down becuase we don't fully understand his intended meaning. There may be weaknesses or errors in Emmanuel's claims and arguments, or perhaps he has some interesting new ways of considering the subject that we are struggling to comprehend. But unless we communicate in a friendly and constructive way we'll never get to know...

Emmanuel is clearly confident of his work and he was making considerable efforts to try to bridge the language barrier before being trampled to death under a barrage of negative interruptions. I am not surprised he is withdrawing from the debate here, but it is to the forum's collective loss.

I will certainly continue to converse with him privately, as I hope will Narcoman, and hopefully we will be able to unravel things and report back here at a later date.

hugh

--------------------
Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 10:51 AM)

Quote The Red Bladder:

---

We can only speculate, as to what on Earth the author is alluding to! Perhaps he read somewhere about the use of companders (that's a compressor-expander, such as the old dbx boxes and Dolby A etc.) and only managed to understand half the story.

---

That's similar to my conclusion. I suspect that he's trying to solve the sort of issue that radio broadcasters came up against years ago as the AM wavebands became more crowded. Someone like Bob Orban is going to be one of the experts in this field - he certainly knows how to create a clean sounding but loud signal, even if the users of his gear don't!

A compander is a double ended process so the final signal is close to the original - even though the transmission system is compromised. I guess we are talking about something that cannot be reversed here.

James.

--------------------
JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net

Edited by Jennifer Jones (30/09/11 10:51 AM)

Quote dmills:

---

How about taking this to the pro audio mailing list?
Lots of serious audio types there that may have some insight.

Please do NOT take this thing to private email, there is interesting content here on all sides of the discussion, and I am sure I am not the only one that is finding the discussion interesting.

---

Good idea Dan. They love this kind of discussion on there and you may even prompt a comment from Bob Orban as mentioned above. Go to

http://bach.pgm.com/mailman/listinfo/proaudio

for more information or to access the archives.

James.

--------------------
JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net

Edited by Jennifer Jones (30/09/11 10:51 AM)

Quote Hugh Robjohns:

---

... It has been clearly established here already that a lot of the problems in comprehending Emmanuel's arguments and conclusions are down to confusion over the use of terminology and language. However, english is not Emmanuel's first language ...

---

With all respect as a newbie who tried reading the article and couldn't understand it: I assumed my difficulties were only because it was technically above my head.

As a paying subscriber to SOS, I would be disappointed if the SOS editorial process, before publishing, didn't "fix" problems in the article that might arise because the author's first language may not be English.

--------------------
Alexis -Cubase 6.5.0/SX3.1.1.944, XP SP2, 4GB RAM (1GB not accessible, but used just to balance the computer so it doesn't tip over); Delta 66 in Omni i/O Studio; Motif8; UAD-1

Edited by Jennifer Jones (30/09/11 10:51 AM)

Quote Hugh Robjohns:

---

The article represents a serious body of research, and the comprehensive measurements he has made are interesting and significant in their own right. I believe we owe the author the respect of at least trying to understand his arguments, rather than simply trying to shoot them down becuase we don't fully understand his intended meaning. There may be weaknesses or errors in Emmanuel's claims and arguments, or perhaps he has some interesting new ways of considering the subject that we are struggling to comprehend. But unless we communicate in a friendly and constructive way we'll never get to know...

Emmanuel is clearly confident of his work and he was making considerable efforts to try to bridge the language barrier before being trampled to death under a barrage of negative interruptions. I am not surprised he is withdrawing from the debate here, but it is to the forum's collective loss.

---

Oh come on Hugh! Dear God! Are we now to give the author Brownie Points for trying?

If I'd written that article, you would have shot me down in flames, jumped on each and every inaccuracy and generally picked the poor thing to pieces and then posted a afterwards!

And you know what? I would have applauded you for doing so! Bad science is bad science and if we allow half truths and speculation, purported by those who do not even understand the fundamentals, then The Flat Earth Society should be running our school science curricula.

As for the silly idea of a behind the scenes discussion - the author should have done that before putting pen to paper, preferably with someone with a formal knowledge of acoustics and psychoacoustics. To do so now is pointless. The damage of publication has been done!

There may or may not be a discussion to be had over a statistical analysis of the relative volume levels of recordings over the years and there is definitely room for taking a fresh and less dogmatic look at the bête noire of the so-called loudness wars and perceived volume in general - but publishing articles that are based on half-baked truths and misunderstandings is not that discussion!

Acoustics and psychoacoustics is a field of study that is largely not understood, even by professionals such as musicians and architects, that deal with the manifestations of sound on a daily basis. Of the various senses, it is possibly the one we know the least about - for example, it is just about 12 months since an Italian seismologist was able to show that we do indeed hear infrasounds (i.e. below 20Hz) and that we (and other animals) do so, using our balance organs.

We need discussion and we need illumination in these and all things, but lead by those who know what they are talking about!

Edited by Jennifer Jones (30/09/11 10:51 AM)

Quote The Red Bladder:

---

Are we now to give the author Brownie Points for trying?

---

No. I'm asking for a little respect and appreciation for someone who is genuinely trying to explain his arguments despite clearly struggling with the language.

Quote:

---

If I'd written that article, you would have shot me down in flames

---

Had I received the article, I would have certainly attempted to clarify the meaning before deciding on whether or not to publish. I didn't see Emmanuel's article before publication, which is why I'm attempting to help clarify various aspects of it now, after the fact. I had hoped we could achieve that constructively here, but that approach is clearly not practical, so it will have to be done privately.

Quote:

---

Bad science is bad science

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I quite agree, and I'm as intolerant of it as you are. But before I start skewering someone who clearly has a strong belief in his arguments, I prefer to make sure where the lack of understanding really is first.

It may be that Emmanuel has completely misunderstood some basic concept of dynamic range control and that his arguments and claims are therefore completely negated. Or it may be that we are spectacularly failing to appreciate his arguments because his use of terminology is very different to our own understanding. At the moment, I am erring on the side of the latter, given the evident confusion we've already seen in this thread over such fundamentals as signal-noise ratio, dynamic range, gain, level and crest factor.

However, I will endeavour to get to the bottom of this, for my own curiosity as much as anything else.

Hugh

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Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 10:51 AM)

Quote Hugh Robjohns:

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...

It may be that Emmanuel has completely misunderstood some basic concept of dynamic range control and that his arguments and claims are therefore completely negated ...

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Shouldn't determining whether the "claims and arguments [of a submitted paper] are ... completely negated" occur before publication?

If SOS publishes an article entitled, "Advanced Compression Techniques for the Home Studio", do I need to ignore it until its accuracy has been vetted, in these forums or otherwise?

Obviously that last question is a load of bollocks. But really, how obvious is it, in light of the quoted sentence above?

I would be interested in hearing from the SOS staff who actually did proofread/edit the manuscript in question.

Respectfully,

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Alexis -Cubase 6.5.0/SX3.1.1.944, XP SP2, 4GB RAM (1GB not accessible, but used just to balance the computer so it doesn't tip over); Delta 66 in Omni i/O Studio; Motif8; UAD-1

Edited by Jennifer Jones (30/09/11 10:52 AM)

Quote alexis:

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Shouldn't determining whether the "claims and arguments [of a submitted paper] are ... completely negated" occur before publication?

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The answer to that is obvious... er.. obviously and the vast majority of SOS articles are thoroughly vetted before publication, as I'm sure a moment's thought will confirm. But I'm afraid I can't comment further as to what happened in this specific case as I wasn't involved at any stage in the production process for this article.

However, I think it is worth remembering that SOS is not a peer-reviewed academic journal. Some of our articles offer information or advice on techniques and technology, and we endeavour very hard to ensure these are accurate and reliable. On the other hand some offer personal opinions on techniques and technology, and these are inherently more contentious. This article is of the latter form.

Emmanuel has gathered a considerable body of data and has used it to support his personal observations and views on mastering techniques and the perceived loudness of material. For what it's worth, I think his views are interesting and worth further consideration, but there are undoubtedly some flaws in the way he has presented it (which may not be entirely his fault -- clearly language and the interpretation of terms is a significant issue) and that is unfortunate.

It seems to me that Emmanuel is using some terms in ways which are unfamiliar to many of us, and some of his comments on some technical details appear to be confused -- or we are misunderstanding his meaning. On the other hand, his data do seem to show some interesting things that are surprising to me, and which do support many of his claims. I enjoyed reading the article and found it both stimulating and frustrating, in roughly equal parts. It seems rather foolish and simplistic, therefore, to dismiss the entire work without attempting first to clarify the areas of doubt.

I also think it is good that his article has stirred up sufficient interest to debate the subject further. It's only by challenging the accepted that we expand the boundaries of our knowledge and understanding.

hugh

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Technical Editor, Sound On Sound

Edited by Jennifer Jones (30/09/11 10:52 AM)

While I think the article really needed some peer review prior to publication (and this thread is really a poster child for why peer review of technical articles is a good thing), and I could not really follow the argument, I am also not prepared to completely write off the author without making some effort to understand what his argument really is.

SOS is not a peer reviewed journal, you want that get JAES, but maybe having a group of people who can do some technical review of the more technical manuscripts would be a good thing, I can think of half a dozen people around here who have the ability and may be willing to look at the occasional manuscript.
SOS does not publish many such so it would be an occasional thing, but it might be worth setting up such a group?
Far better to pull an article due to review before publication then to have to issue a retraction after publication.

The accusation of academic malpractice was IMHO well out of order, being (quite possibly) wrong, is NOT the same as being dishonest.

Regards, Dan.

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Audiophiles use phono leads because they are unbalanced people!

Edited by Jennifer Jones (30/09/11 10:52 AM)

Quote Hugh Robjohns:

---

Quote alexis:

---

Shouldn't determining whether the "claims and arguments [of a submitted paper] are ... completely negated" occur before publication?

---

The answer to that is obvious... er.. obviously and the vast majority of SOS articles are thoroughly vetted before publication, as I'm sure a moment's thought will confirm. But I'm afraid I can't comment further as to what happened in this specific case as I wasn't involved at any stage in the production process for this article.

However, I think it is worth remembering that SOS is not a peer-reviewed academic journal. Some of our articles offer information or advice on techniques and technology, and we endeavour very hard to ensure these are accurate and reliable. On the other hand some offer personal opinions on techniques and technology, and these are inherently more contentious. This article is of the latter form.

Emmanuel has gathered a considerable body of data and has used it to support his personal observations and views on mastering techniques and the perceived loudness of material. For what it's worth, I think his views are interesting and worth further consideration, but there are undoubtedly some flaws in the way he has presented it (which may not be entirely his fault -- clearly language and the interpretation of terms is a significant issue) and that is unfortunate.

It seems to me that Emmanuel is using some terms in ways which are unfamiliar to many of us, and some of his comments on some technical details appear to be confused -- or we are misunderstanding his meaning. On the other hand, his data do seem to show some interesting things that are surprising to me, and which do support many of his claims. I enjoyed reading the article and found it both stimulating and frustrating, in roughly equal parts. It seems rather foolish and simplistic, therefore, to dismiss the entire work without attempting first to clarify the areas of doubt.

I also think it is good that his article has stirred up sufficient interest to debate the subject further. It's only by challenging the accepted that we expand the boundaries of our knowledge and understanding.

hugh

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But Hugh, that can be very disconcerting to the average reader who doesn't have a strong background in acoustics, etc., and who gets their technical knowledge in large part from SOS (not the Journal of Applied Physics and Acoustics).

I don't think I'm unusual in depending on SOS to only publish articles they feel are technically sound, unless of course there is a disclaimer. Obviously comments from commercial producers like "Here's how I make a hit record" are not SOS vetted, but this one had the "feel" and presentation of one of SOS's standard "Dear Reader, here is some SOS guidance ..." articles.

I don't know, maybe flagging articles in the future as "non-SOS opinion" would help. Otherwise, without the ability to distinguish which technical articles SOS stands behind and which they don't, - at least for this reader, the confidence in the accuracy of what is in SOS would drop significantly. And, again speaking only for myself, that confidence is what makes SOS stand head and shoulders above the competition.

Respectfully,

--------------------
Alexis -Cubase 6.5.0/SX3.1.1.944, XP SP2, 4GB RAM (1GB not accessible, but used just to balance the computer so it doesn't tip over); Delta 66 in Omni i/O Studio; Motif8; UAD-1

Edited by Jennifer Jones (30/09/11 10:52 AM)