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MP3 320 kbs vs Redbook ripped to WAV, Positive ABX test result with foobar on classical material (Ravel)
greynol
post Sep 20 2013, 18:48
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I'm wondering whether this sample will have an influence on further mp3 development.

That halb27's fork does so well is encouraging.

I also wonder how well AAC handles this, though not enough to put myself through another ABX session.


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pdq
post Sep 20 2013, 19:06
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QUOTE (greynol @ Sep 20 2013, 13:48) *
I'm wondering whether this sample will have an influence on further mp3 development.

This seems to be a real corner case, and even the OP doesn't seem to feel that it needs to be fixed or improved in any way. Personally I would treat it as a curious anomaly and leave it at that.

What I am much more interested in is the OP's ability to hear the subtlest of details. He could possibly be a valuable asset in assessing codec differences.
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UltimateMusicSno...
post Sep 20 2013, 19:17
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QUOTE (2Bdecided @ Sep 20 2013, 09:51) *
QUOTE (UltimateMusicSnob @ Sep 20 2013, 15:40) *
On the tests above, what were you listening for? Is this listening for a noise floor? If noise floor is just white noise, then the only thing to listen for is levels of noise, yes? A difference in how much white noise you hear below the music, when the volume is cranked way up? And of course, if not for the noise floor, then what is the audible cue?

Yes, noise floor. The noise floor of the recording isn't quite white. With dither there's perfectly white noise added on top. Without dither (truncation) the added noise is quieter but fractionally less uniform. All fairly subtle though and very hard to detect without a reference.

Volume was cranked up, but I would happily listen to the full 30 second clip at that volume - it's not loud. I wouldn't dream of listening to anything else at anywhere near that volume though!

It's interesting that lossyWAV doesn't even touch the first 5 seconds of the file (it maintains the full 16-bits unchanged), and throughout the rest of the file it never knocks off more than 1-bit from the right channel.

Cheers,
David.
This is something that it appears will be a fundamental difference between classical and pop repertoire now indefinitely, thanks to the inherent nature of the material, but also the loudness wars. One reason good stereo equipment for classical becomes expensive is that the signal is 'only' normalized to -6db to begin with, plus you have these very soft passages. So the amplification stages had better be very quiet indeed, with plenty of headroom. On speakers, I prefer to listen to classical on 3 or 4 out of 10 on the dial, but on a big powerful amp stage with big speakers.
And then of course the pop music now normalizes to, well, often 0 db, and if not 0 then -0.1 db.

The consequence is that no single MP3 algorithm could ever be optimized for these dramatically disparate groups of program material. As a business proposition I'd figure all the hard work for MP3 development goes into pop music, since the difference in sales figures is what, 2 or 3 orders of magnitude?

This post has been edited by UltimateMusicSnob: Sep 20 2013, 19:33
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UltimateMusicSno...
post Sep 20 2013, 19:31
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QUOTE (pdq @ Sep 20 2013, 13:06) *
QUOTE (greynol @ Sep 20 2013, 13:48) *
I'm wondering whether this sample will have an influence on further mp3 development.

This seems to be a real corner case, and even the OP doesn't seem to feel that it needs to be fixed or improved in any way. Personally I would treat it as a curious anomaly and leave it at that.

What I am much more interested in is the OP's ability to hear the subtlest of details. He could possibly be a valuable asset in assessing codec differences.

Yes, it's definitely edge of the envelope. greynol did replicate, but that just means two outliers instead of one on the record for this clip.

As I said at the end of the OP, Alternative sample files are **welcome**. These research questions are of great interest to me. Also, details of protocol matter, as the research questions vary, so finding out about improvements to testing methodology is also of great interest.
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2Bdecided
post Sep 20 2013, 19:51
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QUOTE (greynol @ Sep 20 2013, 18:37) *
I think David often likes to use lossyWAV as a tool to determine how many LSBs can be tossed without affecting audibility, based on a setting that hasn't yet been shown not to be transparent?
Yes, exactly.

It's also a helpful indicator of where the 20Hz-15kHz noise floor lies - which in my mind is another way of saying the same thing. Others may disagree. It might not be true that you can always quantise at x dB below a noise floor calculated by some reasonable method and maintain transparency - but it seems to work well enough.

Cheers,
David.
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2Bdecided
post Sep 20 2013, 19:51
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QUOTE (UltimateMusicSnob @ Sep 20 2013, 19:31) *
Is that you, or just a link showing what you're interested in?
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UltimateMusicSno...
post Sep 20 2013, 19:55
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QUOTE (2Bdecided @ Sep 20 2013, 13:51) *
QUOTE (UltimateMusicSnob @ Sep 20 2013, 19:31) *
Is that you, or just a link showing what you're interested in?

That's me.
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halb27
post Sep 20 2013, 20:36
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QUOTE (pdq @ Sep 20 2013, 20:06) *
... Personally I would treat it as a curious anomaly ...

An anomaly?
We're used to considering issues in stuff like electronic music, harpsichord music, or other relatively special contents as an anomaly (as long as we are not very much into these genres), and we ignore them more or less, or use very high quality mp3 which brings even these issues pretty close to transparency.
But this is a sample without any special a priori characteristics for being an issue to mp3. And the issue has been confirmed.
The good news is that this can be considered not relevant in practical listening situations, in case I understand UltimateMusicSnob correctly (please tell us if I'm wrong).

Well, I think in the end it's nothing that essentially changes our attitude towards the usage of mp3. We have known before that there are spots in the music where we can get only near-transparency. What's new is that this doesn't happen only to special stuff, but also to pretty 'normal' music.

An interesting question is:
Claiming the issue being irrelevant is (hopefully) correct for people who use very high quality mp3.
But what about the people who use say -V2, a setting which is considered to yield a very good quality/filesize relation?
UltimateMusicSnob, would you mind trying -V2 and tell us what you think about quality in practical listening situations?
(sorry I can't contribute myself with listening here - I cannot hear any deviation from the original even with lower quality settings than -V2).

This post has been edited by halb27: Sep 20 2013, 20:45


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lame3100m -V1 --insane-factor 0.75
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Wombat
post Sep 20 2013, 22:42
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Since that short sample has a headroom of more as 22dB it may be worth to shift it losslessly 18dB louder so no dither can influence the noise. UMS may test if the encode suffers by its pure loudness or by other things when brought back to higher levels.

Edit: Anyone noticed the left channel only being 15bit in that sample from he beginning? No wonder lossywav can't throw away much. Very srange to have it only on one channel.

This post has been edited by Wombat: Sep 20 2013, 23:28
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UltimateMusicSno...
post Sep 20 2013, 22:46
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QUOTE (halb27 @ Sep 20 2013, 14:36) *
QUOTE (pdq @ Sep 20 2013, 20:06) *
... Personally I would treat it as a curious anomaly ...

An anomaly?
We're used to considering issues in stuff like electronic music, harpsichord music, or other relatively special contents as an anomaly (as long as we are not very much into these genres), and we ignore them more or less, or use very high quality mp3 which brings even these issues pretty close to transparency.
But this is a sample without any special a priori characteristics for being an issue to mp3. And the issue has been confirmed.
The good news is that this can be considered not relevant in practical listening situations, in case I understand UltimateMusicSnob correctly (please tell us if I'm wrong). UMS: This is correct,

Well, I think in the end it's nothing that essentially changes our attitude towards the usage of mp3. We have known before that there are spots in the music where we can get only near-transparency. What's new is that this doesn't happen only to special stuff, but also to pretty 'normal' music.

An interesting question is:
Claiming the issue being irrelevant is (hopefully) correct for people who use very high quality mp3.
But what about the people who use say -V2, a setting which is considered to yield a very good quality/filesize relation?
UltimateMusicSnob, would you mind trying -V2 and tell us what you think about quality in practical listening situations?
(sorry I can't contribute myself with listening here - I cannot hear any deviation from the original even with lower quality settings than -V2).
I would happy to. Unless you say otherwise I'll assume that -V2 is the only switch for the command line.
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halb27
post Sep 20 2013, 22:54
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Yes, just -V2.


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UltimateMusicSno...
post Sep 21 2013, 18:02
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Okay, Ravel clip with 3100L -V2. All planned sets of 12 rounds each.

Before the results, general comments:
Very good algorithm, holds up extremely well in comparison to the -V0. All differences reported here are also subtle. Shares the mid-scooped characteristics of all MP3 encodings, still very subtle here. Shares the slightly diffused bass of -V0, but the 'aura' imparted to the sound is still very musical, an aurally pleasing result. Trade-off appears to be accuracy in preserving exact timbre of the instruments' sounds, for the 'bloom' or 'aura' that results. Transients?

I chose three listening targets ahead of time, all non-artifact.

First target was 'quality of bass'. Listened at 3.0 to about 9 seconds for the crescendo.
CODE
foo_abx 1.3.4 report
foobar2000 v1.2.8
2013/09/20 21:05:44

File A: C:\Users\KiarkAudio\Documents\Ravel Listening Tests\Ravel_Test_File_2_short.wav
File B: C:\Users\KiarkAudio\Documents\Ravel Listening Tests\Ravel_Test_File_6.mp3

21:05:44 : Test started.
21:06:37 : 01/01 50.0%
21:06:53 : 02/02 25.0%
21:07:34 : 03/03 12.5%
21:08:04 : 03/04 31.3%
21:08:38 : 04/05 18.8%
21:09:12 : 05/06 10.9%
21:10:01 : 06/07 6.3%
21:10:32 : 07/08 3.5%
21:11:37 : 08/09 2.0%
21:12:19 : 09/10 1.1%
21:12:55 : 10/11 0.6%
21:14:30 : 10/12 1.9%
21:14:35 : Test finished.

----------
Total: 10/12 (1.9%)

Cues used:
WAV bass has smooth, focused crescendo.
MP3 bass crescendo seems to proceed by plateaus, diffuse.


Second target was 'quality of treble'. Listened at 0 seconds to about 6 seconds.
CODE
foo_abx 1.3.4 report
foobar2000 v1.2.8
2013/09/21 09:22:12

File A: C:\Users\KiarkAudio\Documents\Ravel Listening Tests\Ravel_Test_File_2_short.wav
File B: C:\Users\KiarkAudio\Documents\Ravel Listening Tests\Ravel_Test_File_6.mp3

09:22:12 : Test started.
09:22:36 : 00/01 100.0%
09:22:46 : 01/02 75.0%
09:23:02 : 02/03 50.0%
09:23:14 : 03/04 31.3%
09:23:29 : 04/05 18.8%
09:23:54 : 05/06 10.9%
09:24:15 : 06/07 6.3%
09:25:18 : 07/08 3.5%
09:27:23 : 08/09 2.0%
09:28:27 : 09/10 1.1%
09:29:55 : 10/11 0.6%
09:31:14 : 10/12 1.9%
09:31:25 : Test finished.

----------
Total: 10/12 (1.9%)

Cues used:
Treble is just a bit brighter in MP3, plus some room sound seems to be missing. Can't hear the same reverb reflections in the MP3.


Third target was "flutes" as an impression of the instrument sound, rather than a frequency band. Concentrated on the sforzando flute attack at 8.0 seconds and following.
CODE
foo_abx 1.3.4 report
foobar2000 v1.2.8
2013/09/21 11:25:15

File A: C:\Users\KiarkAudio\Documents\Ravel Listening Tests\Ravel_Test_File_2_short.wav
File B: C:\Users\KiarkAudio\Documents\Ravel Listening Tests\Ravel_Test_File_6.mp3

11:25:15 : Test started.
11:25:38 : 01/01 50.0%
11:25:52 : 01/02 75.0%
11:26:08 : 02/03 50.0%
11:26:23 : 03/04 31.3%
11:26:36 : 04/05 18.8%
11:28:11 : 05/06 10.9%
11:30:40 : 06/07 6.3%
11:32:42 : 07/08 3.5%
11:33:24 : 08/09 2.0%
11:35:39 : 09/10 1.1%
11:37:26 : 10/11 0.6%
11:39:20 : 10/12 1.9%
11:39:22 : Test finished.

----------
Total: 10/12 (1.9%)

Cues used:
MP3 Flute attack at 8.0 seconds is less crisp than WAV. Flute timbre a little blurred, compared to focused sound of WAV.
Bass is diffuse, like the other trials.


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halb27
post Sep 22 2013, 22:02
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Thanks a lot for your test.
So it looks like -V2 quality is also okay for this sample in practical listening situations.

I'd like to add that other than with --bCVBR 316 special version lame3100l has no influence on this result. I've looked it up, and because of the low volume lame3100l does not increase SNR (with the exception of the last two frames which contain short resp. mixed blocks). As was to be expected. So Lame 3.100a2 should work just as fine.

This post has been edited by halb27: Sep 22 2013, 22:02


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UltimateMusicSno...
post Sep 23 2013, 03:19
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QUOTE (halb27 @ Sep 22 2013, 16:02) *
Thanks a lot for your test.
So it looks like -V2 quality is also okay for this sample in practical listening situations.

I'd like to add that other than with --bCVBR 316 special version lame3100l has no influence on this result. I've looked it up, and because of the low volume lame3100l does not increase SNR (with the exception of the last two frames which contain short resp. mixed blocks). As was to be expected. So Lame 3.100a2 should work just as fine.

Do you already know where on the -V scale the outcome re SNR will begin to differ? That inflection point is probably something to know about.
Also, I was just thinking about the psychoacoustic phenomenon of masking--for these very soft passages, what masking could possibly be occurring? Little or none?--so there may not be much for the core theoretical idea of MP3 to grab hold of here.
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halb27
post Sep 23 2013, 11:34
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I've tried -V2 to -V0 here: lame3100l's SNR increasing feature doesn't come into play here. An SNR increase of 3..4 db starts with settings like --bCVBR 266.

The reason behind this is that standard Lame behaves so well. Average bitrate of Lame3.100a2 for your sample and -V2/-V1/-V0 is 173/201/226 kbps despite being low volume. It's 195/228/264 for my standard test set of various pop music which is rather loud overall compared to your sample. lame3100l does a lot for the quality of short and mixed blocks even for moderate quality settings. But this is not relevant here. At -V2 or better it also takes care of long blocks which are relevant here. It's done in a way which doesn't increase average bitrate considerably. (For a stronger bitrate increase using a more demanding -V level is expected to be more appropriate). lame3100l's constrained VBR mechanism works by not allowing audio data bitrate to go too low (not below 153 kbps in the case of -V2 resp. 226 kbps in the case of -V0).
This minimum audio data bitrate is energy dependent however, and the values given are used for a reference level of moderate loudness (which actually is ~10 times the energy level of your sample - so not very much higher; high volume pop music has an energy ~100 times higher than my reference level). So what essentially keeps lame3100l -Vx from improving the situation is that standard Lame does so well in the first place, and also by the way I do the energy dependent SNR increase. As for possible changes concerning the last point see my next post, please.

When there's little masking a good VBR method should take this into account and behave accordingly. Looks like Lame (standard, not just my variant) does a good job at this.


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halb27
post Sep 23 2013, 12:03
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QUOTE (greynol @ Sep 20 2013, 19:48) *
I'm wondering whether this sample will have an influence on further mp3 development.

As far as the development of my Lame variant is concerned the sample has an influence.
This sample shows that the way I do constrained VBR depending on energy level is pretty questionable.
So I guess that I'll create another version where CVBR will have only a very weak energy dependence relevant only for extremely low volume spots.
I do not expect this to significantly improve lame3100l's quality for this sample however, at least not when using -Vx. For using -Vx I will still take care that average bitrate increase will be negligible compared to standard Lame. Whenever standard Lame works adequately as with this sample I do not expect this modification will come into play with -Vx (maybe a bit when using -V0).
Improvement may come with settings like --bCVBR 275, but whether that's audible is another question.

This post has been edited by halb27: Sep 23 2013, 12:09


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Arnold B. Kruege...
post Sep 23 2013, 13:10
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QUOTE (Wombat @ Sep 20 2013, 17:42) *
Since that short sample has a headroom of more as 22dB it may be worth to shift it losslessly 18dB louder so no dither can influence the noise. UMS may test if the encode suffers by its pure loudness or by other things when brought back to higher levels.

Edit: Anyone noticed the left channel only being 15bit in that sample from he beginning? No wonder lossywav can't throw away much. Very srange to have it only on one channel.


+1 with a bullet. Based on the evidence before us, let's call this snippet what it is, which is a badly made recording, and waste no more time with it. If I had some assurance that the track achieved levels within a few dB of FS at some later point, then of course it would become interesting again.

By peaking at 22 dB below FS, it sacrifices more than 3 bits worth of resolution.

Killler track and poorly made recording should be two different categories.

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2Bdecided
post Sep 23 2013, 13:49
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You didn't download it, did you? It's obviously a quiet moment in a full range classical recording. Whether it peaks at full scale I couldn't say, but it obviously goes louder than this. Maybe UMS can confirm the track and album peaks.

The only "badly made" issue is the 15bits in the left channel. That's quite weird, and probably audible. Truncating or dithering the right channel to 15 bits is audible (see above), though that would be an insane listening level for the whole piece (I assume).

Cheers,
David.
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Wombat
post Sep 23 2013, 14:04
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I uploaded a 3 bit louder sample for UMS to test if the problems he hears are still the same. This may clarify if it is only because of its silence or also other factors.
http://www.hydrogenaudio.org/forums/index....t=0#entry845676
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Arnold B. Kruege...
post Sep 23 2013, 14:42
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QUOTE (2Bdecided @ Sep 23 2013, 08:49) *
You didn't download it, did you?


Of course I did! Both files.

QUOTE
It's obviously a quiet moment in a full range classical recording.


Of course it is!

QUOTE
Whether it peaks at full scale I couldn't say, but it obviously goes louder than this. Maybe UMS can confirm the track and album peaks.


That's my point.


QUOTE
The only "badly made" issue is the 15bits in the left channel. That's quite weird, and probably audible. Truncating or dithering the right channel to 15 bits is audible (see above), though that would be an insane listening level for the whole piece (I assume).


15 bits in the left channel? Which sample, what time offset in the track?
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greynol
post Sep 23 2013, 14:48
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Please read the entire discussion. We already had this conversation and the peak value was revealed.

FFS!


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Arnold B. Kruege...
post Sep 23 2013, 15:29
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QUOTE (greynol @ Sep 23 2013, 09:48) *
Please read the entire discussion. We already had this conversation and the peak value was revealed.

FFS!


Did that too. Seems like an obvious disqualifiying fact was being overlooked for some reason.

The clip is also pathological in that its background noise level is something like less than 30 dB below its peak level.
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UltimateMusicSno...
post Sep 23 2013, 16:18
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QUOTE (Arnold B. Krueger @ Sep 23 2013, 09:29) *
QUOTE (greynol @ Sep 23 2013, 09:48) *
Please read the entire discussion. We already had this conversation and the peak value was revealed.

FFS!


Did that too. Seems like an obvious disqualifiying fact was being overlooked for some reason.

The clip is also pathological in that its background noise level is something like less than 30 dB below its peak level.

Classical recordings often present problems not found in pop music. This is a live recording in a hall, so that background noise is ambient noise floor, not recording equipment absolute noise floor. Some halls are very bad in this respect. That said, there are a LOT of pieces with very extended very soft passages. I've never yet heard the beginning of the second (slow) movement of Bartok's Second Piano Concerto done in a way that translates to a home listening system. There's an extended passage before the last section of Firebird Suite by Stravinsky that is vanishingly soft--yet perfectly audible in the concert hall if you're sitting there.

It's conceivable that classical labels could choose to normalize to -1.0db instead of -6.0, but I would bet if that range were to become standard, classical recording engineers would respond by preserving even more of the original dynamic range, which is extraordinary in comparison to pop. Your soft passages would be just as soft.

In other news, I found a slight DC offset which I believe is actually on the disk, don't know how that might interact with the treatments.
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greynol
post Sep 23 2013, 16:30
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It's ironic how mp3 is capable of exceeding the dynamic range of CDDA.


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2Bdecided
post Sep 23 2013, 17:02
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QUOTE (greynol @ Sep 23 2013, 14:48) *
We already had this conversation and the peak value was revealed.
My ability to forget things exceeds my ability to read things wink.gif

QUOTE (UltimateMusicSnob @ Sep 23 2013, 16:18) *
It's conceivable that classical labels could choose to normalize to -1.0db instead of -6.0, but I would bet if that range were to become standard...
There is no standard. Most CDs, popular and classical, are peak normalised to nearly 0dB FS. Normalising to -6dB for a wide range classical CD is a strange choice, though some people worry about distortion in lesser equipment when getting near full scale with "pure" classical music...
http://www.audiomisc.co.uk/BBC/iPlayerRulesOK/Page2.html

Cheers,
David.
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