Understanding the parameters in the dynamic range database
Feb 2, 2017 at 7:01 PM Thread Starter Post #1 of 27

bwanaaa

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I understand the concept of dynamic range and its usefulness in comparing different masterings of an album. But why do they report DR min and DR max. What is the significance of these measurements and how are they more useful than just the average DR?
 
Feb 2, 2017 at 11:15 PM Post #2 of 27
The DR reading is the peak decibel reading minus the average loudness level.  I don't know if that is an RMS average or not.  The min and max refer to individual tracks.  So let us say they list an album at DR14 with min of DR9 and max of DR17. 
 
The album gets a DR14 rating while the min means one of the tracks is only DR9 and the best track is a DR17.   The composite number of 14 is the average track DR for all tracks on the album. 
 
Feb 3, 2017 at 3:12 AM Post #3 of 27
  I understand the concept of dynamic range and its usefulness in comparing different masterings of an album.

 
Rather ironically, the DR database does not actually list/measure dynamic range! It measures/lists peak to average levels rather than peak to noise floor, which in effect is the Crest Factor, not the Dynamic Range. There are various fairly serious flaws/shortcomings with the DR database, although I strongly support it's fundamental aim; to provide consumers with some indication/comparison, in the hope that it will influence sales and in turn drive the labels/artists away from the loudness war.
 
G
 
Feb 3, 2017 at 9:49 PM Post #4 of 27
   
Rather ironically, the DR database does not actually list/measure dynamic range! It measures/lists peak to average levels rather than peak to noise floor, which in effect is the Crest Factor, not the Dynamic Range. There are various fairly serious flaws/shortcomings with the DR database, although I strongly support it's fundamental aim; to provide consumers with some indication/comparison, in the hope that it will influence sales and in turn drive the labels/artists away from the loudness war.
 
G

In the context of the loudness war his measurements make sense, though I agree, probably not named quite right.  Crest factor is a good single-number way to rate how much a recording has been (over)processed to play in the loudness war.  Not total DR, of course.  
 
I too hope it will influence sales and turn the loudness war tables, but I seriously doubt it will have any impact at all.  The war isn't fought on the battlefield of integrity and shame, it's totally ego driven and fought on the great plains of  "mine's bigger than yours" under the fog of "creative license".  The database won't even be seen by those people.
 
Feb 4, 2017 at 1:12 AM Post #5 of 27
Determining personal dynamic range, and personal audio resolution capabilities are covered in an AES presentation in the first post of the following thread.  Probably would get a lot less magical thinking from subjectivists if they would do such a thing for themselves.
 
http://audiosciencereview.com/forum/index.php?threads/judging-individual-high-resolution-audio-perception-capabilities.117/
 
Yes, I agree the DR ratings are closer to crest factor.  But that doesn't apply quite so cleanly to music in people's minds.
 
For anyone not sure how crest factor works this is a reasonably simple explanation.
 
http://www.programmablepower.com/blog/what-is-crest-factor-and-why-is-it-important/
 
Feb 4, 2017 at 12:51 PM Post #6 of 27
  [1] Crest factor is a good single-number way to rate how much a recording has been (over)processed to play in the loudness war.  Not total DR, of course.  
 
[2] I too hope it will influence sales and turn the loudness war tables, but I seriously doubt it will have any impact at all.

 
1. I'm sure you probably know this but for others: Crest factor does not really tell us how much a recording has been processed. Different instruments or even the same instrument played differently, different pieces of music and different genres all give different crest factors even with no processing. It's entirely possible to have one piece of music with light processing/compression, another with much heavier compression and for the first piece to have a lower crest factor (or DR reading) than the second. I wouldn't say crest factor is a "good" single-number way to rate how much a recording has been (over)processed, it's a vague, inconclusive and poor way IHMO. However, as there is no accurate way/measurement, I would say that crest factor is the least bad way rather than a "good" way.
 
2. Agreed. Loudness normalisation appears to be the only way forward.
 
G
 
Feb 4, 2017 at 1:09 PM Post #7 of 27
 [1] Crest factor is a good single-number way to rate how much a recording has been (over)processed to play in the loudness war.  Not total DR, of course.  
 
[2] I too hope it will influence sales and turn the loudness war tables, but I seriously doubt it will have any impact at all.

 
1. I'm sure you probably know this but for others: Crest factor does not really tell us how much a recording has been processed. Different instruments or even the same instrument played differently, different pieces of music and different genres all give different crest factors even with no processing. It's entirely possible to have one piece of music with light processing/compression, another with much heavier compression and for the first piece to have a lower crest factor (or DR reading) than the second. I wouldn't say crest factor is a "good" single-number way to rate how much a recording has been (over)processed, it's a vague, inconclusive and poor way IHMO. However, as there is no accurate way/measurement, I would say that crest factor is the least bad way rather than a "good" way.
 
2. Agreed. Loudness normalisation appears to be the only way forward.
 
G

Crest factor along with Leq seems to flesh out the extremes reliably, the really big loudness war players seem to be easy to flag. So not quite what the DR meter does, but related. It gets vague when looking at less extremely processed stuff where at least some moderation in recovery times has been selected.

Absent any other simple means to quantify abused processing and develop a database, the DR meter fills a need.

I find the idea of remediation of the loudness war like pushing a train uphill.
 
Feb 4, 2017 at 3:47 PM Post #8 of 27
The max and min values are for single tracks within the album. That is, a DR10 album might have some tracks at 12, some at 8, so they list the best/worst cases. Not very meaningful unless you have some weird chimera album of tracks not mastered together. For a (supposedly) detailed explanation of the DR rating, see here. The philosophy behind the rating is that signals like square waves and sine waves are really loud with full-scale peaks, so they came up with a rating where those kind of signals get low scores. This means that actual music whose waveform looks like a sausage (and thus "square-wave-ish") will also get a low score.
 
Feb 4, 2017 at 6:55 PM Post #9 of 27
thank you to all who answered. If there is wide variation in DR scores in one album compared to another version of the same album (for example, flac, vs DSD vs something else), can one assume that the tracks have been 'over-processed'? The analogy I might make is to photography. There are some people who like the look of 'overdone' high dynamic range photography - but it is not real.  
 
Feb 4, 2017 at 8:17 PM Post #10 of 27
thank you to all who answered. If there is wide variation in DR scores in one album compared to another version of the same album (for example, flac, vs DSD vs something else), can one assume that the tracks have been 'over-processed'? The analogy I might make is to photography. There are some people who like the look of 'overdone' high dynamic range photography - but it is not real.  

Generally audio processing results in less dynamic range. "Over processed" is subjective. The people doing it don't think it's over processed, others find the result annoying, uninteresting, or just plain distorted. The question of what's good pretty much lies in the hands of the one writing the check.
 
Feb 4, 2017 at 9:46 PM Post #11 of 27
  thank you to all who answered. If there is wide variation in DR scores in one album compared to another version of the same album (for example, flac, vs DSD vs something else), can one assume that the tracks have been 'over-processed'? The analogy I might make is to photography. There are some people who like the look of 'overdone' high dynamic range photography - but it is not real.  


it's the opposite of photography. on a pic you increase the dynamic/contrast/etc to show "more", in audio to make a sound noticeable you have to make it loud. and when everything is loud, there is no dynamic anymore.
but just like post processing in photo, you have those who suck at it, and you have some who really improve the product and become the last artist in the creative process. compression is but one of so many tools, it's how people use it that makes a difference.
personally, I can't relax with very dynamic songs, it's good to have sometimes when you want it, a good symphony wouldn't be the same at DR5. but when I just have music in the background trying to relax, getting 5mn I can barely hear, and then explosions everywhere for the next 2mn like you get in some really good classical music, or a song that I need to turn up to notice more the all song, that's not what I need/enjoy. and again it's a personal opinion, but voices without any compression tend to annoy me real fast.
so while I cry about loudness war like anybody else, I absolutely do not think that more dynamic systematically means better music.
 
Feb 4, 2017 at 10:05 PM Post #12 of 27
The problem is, you can't undo processing. You can only apply it. It would be easy to post-process anything that is widely dynamic originally and squish it down to something that meets your needs. I do this all the time with background music. However once it's been crunched to within an inch of its life, there's nothing you can do the re-expand it. The damage is done, partly because you have no idea what they did to process it so there is no way to apply an inverse, and partly because some of the processing is so severe that there is no opportunity to re-expand it.
 
Feb 5, 2017 at 4:58 AM Post #13 of 27
  [1] If there is wide variation in DR scores in one album compared to another version of the same album (for example, flac, vs DSD vs something else), can one assume that the tracks have been 'over-processed'?
[2] There are some people who like the look of 'overdone' high dynamic range photography - but it is not real.  

 
1. Probably but not necessarily, it depends. If we compare say a vinyl version of an album to say a CD/flac, then we can't assume that, as the meter used to make the DR reading gives highly inaccurate results for vinyl (see here). Plus, it's almost impossible to accurately define what "over" processed actually is. In a relatively few, very extreme cases virtually everyone would agree but the further from that level of extremity we go, the less agreement we would get.
 
2. Again, we have to be careful here. In the vast majority of cases, as far as music is concerned, there is no "real" to begin with, it's just an illusion which has been created with some artificial/manufactured reality to make it more relate-able. A photography analogy would be if we were to take a still from say a forest scene on Pandora (the planet featured in film Avatar). We could certainly over-process that still to the point of degrading the image but we can't really say the result is "not real" (or less real) because of course there is no "real" to begin with, or rather, the "real" is a artificial/manufactured illusion, created from the subjective judgement of the artists, colourist and director (respectively in music; the musicians, mastering engineer and producer).
 
[1] Crest factor along with Leq seems to flesh out the extremes reliably, the really big loudness war players seem to be easy to flag. So not quite what the DR meter does, but related.

[2] Absent any other simple means to quantify abused processing and develop a database, the DR meter fills a need.

[3] I find the idea of remediation of the loudness war like pushing a train uphill.

 
1. We still need to be careful with what we compare though. What maybe moderate, perfectly acceptable processing for one piece of music, could easily be extremely over-processed for another. ...
2. Agreed, providing we take the previous point into consideration. If we don't (and certainly some don't), we might easily quantify processing on a particular track as "abused" when in fact it's moderate and perfectly acceptable, even desirable/better.
3. Yes, although it has been successfully dealt with in the TV broadcast world (in many countries) and there does seem to be "light at the end of the tunnel" in the music industry.
 
It would be easy to post-process anything that is widely dynamic originally and squish it down to something that meets your needs.

 
"Easy" yes, not necessarily good though! Certainly nowhere near as good as an experienced pro with a collection of high-end tools.
 
G
 
Feb 5, 2017 at 12:00 PM Post #14 of 27
 
"Easy" yes, not necessarily good though! Certainly nowhere near as good as an experienced pro with a collection of high-end tools.
 

I guess I'm the wrong person to comment.  Too many years of using and designing audio processors.
 
"Good enough" depends on your goal.  I find the tools built into Audition fine for most purposes, even destruction-level processing, but I come to them with a lot of experience.  I have developed several processing "stacks" (several cascaded processor functions) that accomplish several different goals.  I've also demo'd some of the high-end tools.  I don't find their capabilities much better, but the controls are often easier to comprehend when relating to an expected result.
 
For many years I've felt that processing in broadcast should be reverseable some extent.  If the entire broadcast processing chain could output a metadata stream that worked as a "Rosetta Stone", receivers could take that data and "un-do" portions or all of the damage done.  Automotive receivers could have true road-noise modifiers that include DR control and spectral modification to maximize audibility.  Home receivers could have simplified activity-targeted presets (low-background, party background, concert, etc.), and have a pseudo "expert" mode that allowed for the complete application of an inverse algorithm.  The the technology could easily be encoded into downloaded music files too, though I doubt the music industry would warm up to that idea.  But it would allow for a completely unprocessed version to become available to the listener, at his option, or anything between that and the producer's version.  I
 
t might take a new file type to pull it off.  I'm thinking of calling it Maximum Audio Quality, you know, MAQ...er...um....ok, something else. 
 
Feb 6, 2017 at 8:27 AM Post #15 of 27
  [1] "Good enough" depends on your goal.  I find the tools built into Audition fine for most purposes, even destruction-level processing, but I come to them with a lot of experience.  I have developed several processing "stacks" (several cascaded processor functions) that accomplish several different goals.  
 
[2] I've also demo'd some of the high-end tools.  I don't find their capabilities much better, but the controls are often easier to comprehend when relating to an expected result.
 
[3] Home receivers could have simplified activity-targeted presets (low-background, party background, concert, etc.), [3a] and have a pseudo "expert" mode that allowed for the complete application of an inverse algorithm. [3b] The the technology could easily be encoded into downloaded music files too, though I doubt the music industry would warm up to that idea.  [3c] But it would allow for a completely unprocessed version to become available to the listener, at his option, or anything between that and the producer's version.

 
1. I have a couple of problems with this: Firstly, I don't know Audition well but it has rather basic controllability/functionality compared to what I'm used to (and what I need). Secondly, even if I could some how get around it's limitations with a "stack" of processors, this is hardly a consumer friendly solution. I'm guessing a fraction of one percent of consumers would even attempt such a stack and only a small fraction of those would get decent results.
 
2. That's strange, I've found the exact opposite! Much greater capabilities and far more complex controls which are relatively difficult to comprehend. At this stage it's probably worth covering a bit about compression, as that's mainly what we're talking about here as far as dynamic range and processing is concerned: This 7min vid, is a basic primer on compression; what compression is, the basic controls, how they're used and what they sound like. I recommend this vid for anyone who doesn't really know what compression is, only has a vague idea, just wants a refresher or wants to be sure they've understood the basics.
 

 
In response to pinnahertz, here is a brief (just under 7mins) tutorial of a compressor which I commonly use. I strongly recommend watching this vid, especially for those who think they already have a decent understanding of compression basics. A couple of notes: It's quite advanced but all the controls this vid covers are constantly being demonstrated, so you can hear what's happening even if you don't fully understand the controls themselves. Secondly, this is just part one of the tutorial and even both parts together only cover some of the controls available in this compressor.
 

 
Some further observations/points: A. The videos demonstrate compression on individual channels or on a sub-group/submix (the drums submix in the second vid), rather than master-buss compression. Obviously the same techniques apply to the entire mix (master-buss) as to a submix, although with different settings. It should be noted that in the vast majority of cases most of the compression applied to a mix is applied during mixing rather than during mastering, IE. To individual channels and submixes rather than on the master-buss. B. Both vids were demonstrating different popular music genres. In classical music, compression is still quite commonly employed on individual tracks, sub-groups and on the master-buss but not as ubiquitously and far more subtly than in the popular genres. C. There are some "highend" compressor processors which only have the basic compressor controls, these tend to be the modelling compressors: Software plugins with algorithms modelled to emulate a vintage analogue compressor. These software compressors are non-linear, they introduce various distortions (including IMD commonly) which provide the "character" of the original unit. Generally there is no control over any of the attack or release curves, other parameters or even the amount of distortion, beyond how hard it is driven (as with the original units). So it's a case of using one of these compressors for it's particular character or using a completely different compressor if that "character" is not appropriate for a particular channel/submix/mix.
 
3. Yes, a simple set of presets to cover some basic situations could be useful but what you suggest already exists. Dolby Digital already includes 6 presets (DRC profiles) which are set in the DD metadata and many AV Receivers allow that setting to be overridden. Also, the "Loudness" control on some amps (particularly car sound systems) is effectively a simple compression preset. How these presets interact with the content is variable though, depending on the content. Sometimes it's relatively benign other times it's very annoyingly not so. "Pumping" is a common problem for example, due to an inappropriate release time/curve of the compressor, especially if it fights with pumping already deliberately/artistically applied or with content which changes dynamics quite rapidly. And, this is just one of several quite common issues with simple consumer compression and presets.
 
3a. This is simply impossible and impractical, for several reasons, the main two being: 1. Typically several different compressors are used in a mix, on individual channels and sub-groups and then another compressor used on the master-buss (in mastering). It's simply impossible to un-pick (un-mix) a mix, let alone un-mix the mix, identify which compressors have been used, where and with what settings and then apply inverse compression algorithms to each channel, sub-group and the entire mix. 2. Even if this were possible in theory, it would still be impossible in practice because although basic compression algorithms are freely available (and therefore an inverse algo could be freely designed) this is definitely not true of the higher end compressors whose algorithms are trade secrets (or covered by exclusive copyright licenses in the case of some modelling compressors). I can't see how you would get ALL of the companies/software developers to effectively donate the algos upon which their company relies!
 
3b. How could it be "easily encoded" into the files themselves? Not only would you need metadata fields to cover every possible parameter of every existing (and future) compressor but you'd need to somehow embed data to change any/all of those parameters in time with the music, as automation of compressor parameters during mixing/mastering is not uncommon. This doesn't sound to me like something which could be "easily encoded"! And of course, that's assuming the impossibilities/impracticalities in 3a have all been overcome and some chip developed (for inclusion in DACs/AVRs) upon which all this metadata and embedded data can act.
 
3c. I've seen similar requests several times here on head-fi. I can only assume those making the request simply don't understand how/why compression is applied. If you've watched both videos above, you'd realise that compression changes the volume, tonality and "presence", and as separate instances (with different settings) and/or different compressors are used on different individual channels and sub-groups, this in turn changes how each of those channels are balanced and positioned against each other and how one applies other processing such as EQ, reverb, etc., to all the channels. Removing all the compression does NOT result in the same mix just with a bigger DR, it results in a complete dogs-dinner of a mix, where the elements in the mix no longer balance with each other (dramatically so!) and virtually all the other processing is also now wrong/inappropriate. In the majority of cases, the result would be an un-listenable mix! And just to reiterate what I mentioned above, while there maybe a few exceptions, it's largely a myth that a highly dynamic mix is delivered to the mastering engineer who then applies massive amounts of additional compression to kill the track in the name of making it louder. I've been handed mixes (even entire albums) to master where I couldn't have added a single dB more compression even if I'd wanted to. While this is rare, it's also rare that I could apply an additional level of compression equal to or greater than that already applied (during mixing).
 
G
 

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