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Old 20 June 2005, 01:31 pm   #1
davidm
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Default low level clicks and noise

I have been using MPC for some time and have been very satisfied, but recently I have noticed click and distortion when copying some classical cd,s

This proble occures when when either foobar or winamp are used as players.

I have done some tests and have determined that using a test cd static tones below -80db are replaced by clicks and noise.

Is there a command line to introduce the correct dither or alternativly are ther better plugins for the players.

This is a major problem as the artifacts are clearly audible on low noise classical music.
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Old 20 June 2005, 04:53 pm   #2
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Please provide us with a small sample (lossless file) that portrays this issue. You could either post a link to a file or send a file to us via our IRC network.
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Old 20 June 2005, 08:08 pm   #3
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Default low level clicks

I have undertaken a few more tests and have gained more detail

On tone signals below -70 a number of spurious noises occur, these are made many times worse when dither is applied to the recording. You can imagine on a track which has a high replay gain this can be obvious.

I have prepared a short 5 second dithered tone at -80dB if this wave file is encoded in mpc, then has replay gain applied you will hear many unusual noises.

I have placed this wave file on our web site at
http://www.wavecor.co.uk/minus80.wav

I hope that this helps.

Thanks for the support

David McGhee
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Old 20 June 2005, 08:45 pm   #4
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First of all, this is an artificial tone with background hiss, nothing which can be found in classical music. Such artificial samples are no indication of anything regarding quality of codecs. Secondly, -80db is very, very low volume, and no audio track in the world comes anywhere near such a volume, so your claim that replaygaining a track could make such artifacts noticable is false. In any normal case you would not notice artifacts that Musepack, Vorbis or LAME produce on your sample. If you extremely amplify it, you can't expect much. All lossy codecs would produce artifacts on such a sample, which are often not noticable unless you highly amplify it.
Here is is an archive containing flac files converted from Musepack @ q7, Vorbis @ q7 and LAME @ extreme amplified by 48db: minus80.zip
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Old 20 June 2005, 09:31 pm   #5
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Default low level noises

I find your reply most unhelpful, there are many instruments includig piano which will produce the condition which I have demonstrated.

My tests confirm that these artifacts do not occur when using ogg. or mp3 or wavpack or aac/mp4.

I was trying to be helpful in high-lighting this problem with the mpc encoder, these very low level artifacts are only a problem if you intend mpc to be a quality coder.

To suggest that they are unimportant does not help

David McGhee
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Old 20 June 2005, 09:39 pm   #6
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I have edited my post some seconds before you replied, please reread.

If you find my reply unhelpful, there's not much I can do. My "tests" confirm that other codecs produce artifacts as well, so I don't know what you're trying to say.

You have provided no real world example that portrays a similar artifacting.

And as I said before, only in very extreme, unnatural cases of amplification, such artifacts are noticable. The very design of lossy audio codecs is meant to change the audio signal without causing a change in perceptual quality. This is what Musepack does excellently. What you call a "problem" is not a real-world, natural condition problem.
Until you provide any real world example of such artifacting, there's not much left to be said.
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Old 21 June 2005, 01:31 pm   #7
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The extreme low gain appear to trip the tonality estimation of mpc and mp3. Both artifact badly at standard and extreme. Higher settings sound fine.

I ran wavegain on the WAV sample and now mpc / mp3 encode correct even at standard settings. Shy is right about inducing artificial gain levels, but this is still interesting to me.
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Old 22 June 2005, 05:56 am   #8
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Problems occurring at -80 dB shouldnt be audible in any situation. Even with a dynamic compressor which would bring lowest part of the signal to a much higher one, I seriously doubt that audible problems might be revealed to the listeners. I have more than 1000 classical music CDs, and even ultra-quiet parts of wide-dynamic compositions (piano, full orchestra) arent at -80 dB. Its typically -65...-50 dB. There is of course some information at -80 dB, but its usually noise, with very little information coming from instruments.

Nevertheless, musepack has poor performances at mid/high profile (--standard, --extreme and sometimes --insane) with low volume tracks. This problem really becomes audible, and sometimes dramatic, after replaygaining. I have many tracks that are measured by RG at > +20 dB; some of them also reach the +30 dB floor: organ, orchestra and chamber music (especially contemporary one) are periodically implicated. Therefore, if you play these tracks with RG track mode enabled, there will be a terrible ringing. I suppose it could be highly improved, because other encoders are performing *much* better even with low setting: vorbis at q4, faac at q100 and also lame MP3 3.97a at 128 kbps offer all (much) better encoding than mpc --standard and sometimes extreme.

Ive uploaded some encodings to illustrate this problem. Reference file is an organ piece, coming from complete Liszt organ music (5CD) played by Olivier Vernet. The RG value for this single track is +29.47 dB; the RG value for the short sample Ive selected is +40 dB. But Ive manually edited the RG value to match the reference one (+30 dB).

http://guruboolez.free.fr/MPC/low_volume_encodings.zip

PCM file could be downloaded in the same folder:
http://guruboolez.free.fr/MPC/Liszt_Choral.wv


The ringing, absolutely terrible at q5 after RGing, is still annoying at q6, but clearly less with q7. With q8 profile, ABXing wasnt a real problem, but the annoyance is near zero (reference sound quality is more questionable). Ive ABXed q9, but failed at q10.

Code:
ABC/HR for Java, Version 0.5a, 21 juin 2005
Testname: 

Tester: 

1R = C:\tests\Liszt\Liszt - Choral MPC Q6.wav
2R = C:\tests\Liszt\Liszt - Choral MPC Q10.wav
3L = C:\tests\Liszt\Liszt - Choral MPC Q5.wav
4L = C:\tests\Liszt\Liszt - Choral MPC Q7.wav
5R = C:\tests\Liszt\Liszt - Choral MPC Q8.wav
6R = C:\tests\Liszt\Liszt - Choral MPC Q9.wav

---------------------------------------
General Comments: 
---------------------------------------
1R File: C:\tests\Liszt\Liszt - Choral MPC Q6.wav
1R Rating: 1.5
1R Comment: awful ringing
---------------------------------------
2R File: C:\tests\Liszt\Liszt - Choral MPC Q10.wav
2R Rating: 4.8
2R Comment: Transparent this time. I've tried to focused on background details (organ mechanic), but obtained questionable pval on ABX (11% chance for guessing).
---------------------------------------
3L File: C:\tests\Liszt\Liszt - Choral MPC Q5.wav
3L Rating: 1.0
3L Comment:  below telephone . 0/5 would be more realistic.
---------------------------------------
4L File: C:\tests\Liszt\Liszt - Choral MPC Q7.wav
4L Rating: 3.0
4L Comment: ringing, obvious but much less annoying than 1R and 3L
---------------------------------------
5R File: C:\tests\Liszt\Liszt - Choral MPC Q8.wav
5R Rating: 4.0
5R Comment: unconstant, fluctuant noise
---------------------------------------
6R File: C:\tests\Liszt\Liszt - Choral MPC Q9.wav
6R Rating: 4.5
6R Comment: ABXable, but variations of noise are very subtle
---------------------------------------

ABX Results:
Original vs C:\tests\Liszt\Liszt - Choral MPC Q10.wav
    11 out of 16, pval = 0.105
Original vs C:\tests\Liszt\Liszt - Choral MPC Q7.wav
    8 out of 8, pval = 0.0030
Original vs C:\tests\Liszt\Liszt - Choral MPC Q5.wav
    8 out of 8, pval = 0.0030
Original vs C:\tests\Liszt\Liszt - Choral MPC Q8.wav
    7 out of 8, pval = 0.035
Original vs C:\tests\Liszt\Liszt - Choral MPC Q6.wav
    8 out of 8, pval = 0.0030
Original vs C:\tests\Liszt\Liszt - Choral MPC Q9.wav
    8 out of 8, pval = 0.0030


---- Detailed ABX results ----
Original vs C:\tests\Liszt\Liszt - Choral MPC Q10.wav
Playback Range: 01.041 to 21.638
    9:22:16 PM p 1/1 pval = 0.5
    9:22:24 PM p 2/2 pval = 0.25
    9:22:30 PM f 2/3 pval = 0.5
Playback Range: 06.556 to 08.678
    9:22:46 PM p 3/4 pval = 0.312
    9:22:52 PM p 4/5 pval = 0.187
    9:23:01 PM p 5/6 pval = 0.109
    9:23:05 PM p 6/7 pval = 0.062
    9:23:09 PM p 7/8 pval = 0.035
    9:23:13 PM f 7/9 pval = 0.089
    9:23:18 PM p 8/10 pval = 0.054
    9:23:22 PM f 8/11 pval = 0.113
    9:23:26 PM p 9/12 pval = 0.072
    9:23:30 PM p 10/13 pval = 0.046
    9:23:34 PM p 11/14 pval = 0.028
    9:23:39 PM f 11/15 pval = 0.059
    9:23:47 PM f 11/16 pval = 0.105

Original vs C:\tests\Liszt\Liszt - Choral MPC Q7.wav
Playback Range: 01.041 to 21.638
    9:18:36 PM p 1/1 pval = 0.5
    9:18:39 PM p 2/2 pval = 0.25
    9:18:41 PM p 3/3 pval = 0.125
    9:18:43 PM p 4/4 pval = 0.062
    9:18:45 PM p 5/5 pval = 0.031
    9:18:48 PM p 6/6 pval = 0.015
    9:18:50 PM p 7/7 pval = 0.0070
    9:18:53 PM p 8/8 pval = 0.0030

Original vs C:\tests\Liszt\Liszt - Choral MPC Q5.wav
Playback Range: 01.041 to 21.638
    9:17:54 PM p 1/1 pval = 0.5
    9:17:55 PM p 2/2 pval = 0.25
    9:17:56 PM p 3/3 pval = 0.125
    9:17:58 PM p 4/4 pval = 0.062
    9:17:59 PM p 5/5 pval = 0.031
    9:18:00 PM p 6/6 pval = 0.015
    9:18:07 PM p 7/7 pval = 0.0070
    9:18:09 PM p 8/8 pval = 0.0030

Original vs C:\tests\Liszt\Liszt - Choral MPC Q8.wav
Playback Range: 01.041 to 21.638
    9:19:28 PM f 0/1 pval = 1.0
    9:19:31 PM p 1/2 pval = 0.75
    9:19:33 PM p 2/3 pval = 0.5
    9:19:37 PM p 3/4 pval = 0.312
    9:19:39 PM p 4/5 pval = 0.187
    9:19:42 PM p 5/6 pval = 0.109
    9:19:46 PM p 6/7 pval = 0.062
    9:19:50 PM p 7/8 pval = 0.035

Original vs C:\tests\Liszt\Liszt - Choral MPC Q6.wav
Playback Range: 01.041 to 21.638
    9:17:26 PM p 1/1 pval = 0.5
    9:17:27 PM p 2/2 pval = 0.25
    9:17:29 PM p 3/3 pval = 0.125
    9:17:30 PM p 4/4 pval = 0.062
    9:17:31 PM p 5/5 pval = 0.031
    9:17:33 PM p 6/6 pval = 0.015
    9:17:35 PM p 7/7 pval = 0.0070
    9:17:37 PM p 8/8 pval = 0.0030

Original vs C:\tests\Liszt\Liszt - Choral MPC Q9.wav
Playback Range: 01.041 to 21.638
    9:20:21 PM p 1/1 pval = 0.5
    9:20:31 PM p 2/2 pval = 0.25
    9:20:38 PM p 3/3 pval = 0.125
    9:20:49 PM p 4/4 pval = 0.062
    9:21:00 PM p 5/5 pval = 0.031
    9:21:21 PM p 6/6 pval = 0.015
    9:21:40 PM p 7/7 pval = 0.0070
    9:21:49 PM p 8/8 pval = 0.0030
http://guruboolez.free.fr/MPC/Liszt_...playgained.txt




Of course, this situation is not very common, and the big artefact only appears with RG enabled, and with trackgain only. But performance is definitively poor (compared to other tools), and the bad surprises comes from an exceptional low performance of musepack at -q5, used to be a champion with this encoding profile (it was ranked first after a blind listening test involving 18 classical music samples Ive performed last summer), and which introduces here terrible artefacts that even competitors at 100 kbps could avoid.

Working in this issue (IMO the biggest quality flaw of the format at q5/-q6) might be a direction for further tunings of the current encoder.
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Old 22 June 2005, 07:36 pm   #9
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Default Low Level Cilcks

I'm glad that someone else has noticed this problem.

Since starting this thred I have gained more background info.

A typical test CD will have a range of fixed test tone levels from 0dB down to -120dB, this low level(well below the 16 bit minimun code level) is possible because of dither noise applied at the recording stage.

My recent tests show that the current mpc coder starts to produce noise artifacts at -70dB, inpractice this means that a low level classical recording of say piano or organ will produce audible errors. this is in clear comparisson to other coders which will reproduce the entire test CD rangewithout trouble.

Though I have not come to a firm conclusion I suspect that the mpc coder does better without record dither.

I have an mpc library of over 4k. files and have built a good confidence level in mpc. The problem only came to light when I play some recent cd copys which were not as good as they should be.
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Old 22 June 2005, 07:59 pm   #10
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I have forwarded the URL to Frank Klemm through CiTay. I don't know what he'll say. I just know the current developers are incapable of handling this.
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Old 23 June 2005, 09:26 am   #11
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I've uploaded additional samples, and make a report on HA.org:

http://www.hydrogenaudio.org/forums/...howtopic=35030
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Old 23 June 2005, 10:28 am   #12
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Guruboolez,

Could you try

--standard --ltq 250
--xtreme --ltq 260
--insane --ltq 270
...
and so on? This uses the very sensitive "Filburt" scale in quiet. Musepack will label them as "Below Telephone"-profile encodings, but this is obviously a bug, because --ltq 2xx is higher quality then the default --ltq 5xx.

Cheers,

Tim
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Old 23 June 2005, 10:38 am   #13
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It's not a bug, but a feature introduced by Frank Klemm to discourage the use of personal command line (they're used to lower the quality).
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Old 23 June 2005, 01:52 pm   #14
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Default LowlLevel Clicks

I have just tried --xtreme --ltq 260 and it seems to work ok.

This leaves me very cunfused as I took the avilable preset lines as the best option, it now appears that my library will need recoding to get back to the expecte quality level.

Are there other command line options that will allow me to choose the best coding level fo a particular music type. I also have many mono files is there a downmix to mono switch?

Many thanks for your help.
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Old 23 June 2005, 02:19 pm   #15
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In my experience, mono files don't need extra swithes for extra bitrate-savings, i.e. adding --mx xx (ex. --ms 10) will not save some kbps on _mono_ files, but you could possibly save a few kbps by downmixing a 2-channel mono wav, to a 1-channel mono wav, before encoding to musepack. Musepack itself has no switch for downmixing to mono.

As for using --ltq 2xx, expect about 5% larger filesizes in general, IIRC.

Cheers,

Tim
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Old 24 June 2005, 12:17 am   #16
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I have the same results here with the Liszt sample. That is to say, without being the slightest familiar with the Listz_choral sample, --quality 6 --ltq 260 seemed at first hearing quite transparent to me, where --qualty 5, --quality 5 --ltq 250 and --quality 6, obviously were not transparent.

Code:
ABC/HR for Java, Version 0.5a, 24 juni 2005
Testname: mpc --ltq 2xx test

Tester: 

1R = C:\Test\003 Liszt_Choral.q5ltq250.wav
2R = C:\Test\002 Liszt_Choral.q5.wav
3L = C:\Test\004 Liszt_Choral.q6.wav
4L = C:\Test\005 Liszt_Choral.q6ltq260.wav

---------------------------------------
General Comments: 
---------------------------------------
1R File: C:\Test\003 Liszt_Choral.q5ltq250.wav
1R Rating: 2.1
1R Comment: 
---------------------------------------
2R File: C:\Test\002 Liszt_Choral.q5.wav
2R Rating: 1.0
2R Comment: 
---------------------------------------
3L File: C:\Test\004 Liszt_Choral.q6.wav
3L Rating: 3.0
3L Comment: 
---------------------------------------

ABX Results:
Original vs C:\Test\005 Liszt_Choral.q6ltq260.wav
    2 out of 8, pval = 0.964
Original vs C:\Test\003 Liszt_Choral.q5ltq250.wav
    8 out of 8, pval = 0.0030
Original vs C:\Test\004 Liszt_Choral.q6.wav
    8 out of 8, pval = 0.0030


---- Detailed ABX results ----
Original vs C:\Test\005 Liszt_Choral.q6ltq260.wav
Playback Range: 00.000 to 21.638
    1:56:04 AM f 0/1 pval = 1.0
    1:56:14 AM f 0/2 pval = 1.0
    1:56:32 AM f 0/3 pval = 1.0
    1:56:38 AM p 1/4 pval = 0.937
    1:56:53 AM p 2/5 pval = 0.812
    1:57:00 AM f 2/6 pval = 0.89
    1:57:17 AM f 2/7 pval = 0.937
    1:57:24 AM f 2/8 pval = 0.964

Original vs C:\Test\003 Liszt_Choral.q5ltq250.wav
Playback Range: 00.000 to 21.638
    1:51:13 AM p 1/1 pval = 0.5
    1:51:19 AM p 2/2 pval = 0.25
    1:51:22 AM p 3/3 pval = 0.125
    1:51:26 AM p 4/4 pval = 0.062
    1:51:29 AM p 5/5 pval = 0.031
    1:51:34 AM p 6/6 pval = 0.015
    1:51:38 AM p 7/7 pval = 0.0070
    1:51:47 AM p 8/8 pval = 0.0030

Original vs C:\Test\004 Liszt_Choral.q6.wav
Playback Range: 00.000 to 21.638
    1:52:57 AM p 1/1 pval = 0.5
    1:53:04 AM p 2/2 pval = 0.25
    1:53:23 AM p 3/3 pval = 0.125
    1:53:32 AM p 4/4 pval = 0.062
    1:53:40 AM p 5/5 pval = 0.031
    1:53:46 AM p 6/6 pval = 0.015
    1:53:51 AM p 7/7 pval = 0.0070
    1:53:54 AM p 8/8 pval = 0.0030
Obviously, Guruboolez, you can do better, and I hope you forgive me my crippled ears. ops:

As for the suggestion as posted on HA.org to use --ath_gain -14, or something in that style, for normal classical music with the occasional pianissimo passage, I don't think that is a wise swith to use, because it would lower the ath_scale with 14 db in all subbands equally, even at 20 khz, which would not lead to audible gain, but to bitrate bloat in frequency-regions where it makes no audible difference, imho.


Cheers,

Tim (Tim Mervielde on HA.org)
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Old 27 June 2005, 09:49 am   #17
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I've quickly tried both switches, and --ath_gain -14 offers better improvments than --ltq 250 with the samples I've posted.
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Old 27 June 2005, 07:04 pm   #18
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mmm i recall ath_gain -14 being used back in the day for soft/silent/quiet/whatever sources

is that right?
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Old 28 June 2005, 07:31 am   #19
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I've also tried --ath_gain with --quality 4 profile. Bitrate jumps from ~140 to ~160 kbps and reach a comparable value than --quality 4.5 (I obtained these numbers by encoding 160 classical music samples; RG from -10 dB to +30 dB).

--quality 4 --ath_gain -8 [I've also tried -11] considerably reduces the level of distortion with various samples compared to --quality 4.5. The progress is audible even on tracks at normal volume (89 dB).

It may be interesting to compare --quality 5 --ath_gain -14 to --quality 5.x with tracks at normal volume to check possible progress. The regression I've noticed between 1.01j and 1.15 is maybe a consequence of different ATH threshold or something directly linked to ATH. It' just a possibility, tests are needed to confirm or infirm that.
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Old 28 June 2005, 07:09 pm   #20
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Here is what Frank Klemm said about the issue and several other related points. Translated by CiTay:


Quote:
The calculated masked threshold is indeed depending on the level. It changes if lower levels
are approached. This modification was made sometime between encoder version 1.06 and 1.1.

With high levels, the NMR (noise-to-mask ratio) was raised by 0.5 dB, with low levels,
it was lowered. The masked threshold (ATH) was lowered by 6 dB in total.

The original behavior was that up to a certain threshold, things were coded with full NMR,
and after that it would suddenly get muted. A signal around that switching threshold
produced audible artifacts, despite the fact that many bits were used for coding.

The current behavior is that the coding gradually gets worse with very low levels and there's
almost no usable signal in the end. Only when this point is reached, the coding is stopped.

When you're looking at the error signal over the signal strength, there's a slowly declining
function that approaches the ATH from above. The old behavior first caused the error signal to
fall ca. 20 dB below the ATH and only raise to ATH-level again when the coding was stopped.

Extensive listening tests with headphones were conducted (headphones because of the high
listening level). For listening material, among others, the Bolero by M. Ravel was used.
Volume was adjusted to ca. 114 dB SPL at -0 dB signal strength.

At this volume, noise in the recording and quantization artefacts are already an issue with
many 16 bit recordings. As long as this level is not (clearly) exceeded, the quality of the
coding was clearly better, despite the lower bitrate (even though the NMR was raised by 0.5 dB
and the ATH lowered by 6 dB, there are spare bits with almost every kind of music).
The fluctuation in the coding - which was caused by activation and deactivation of subbands -
disappears.

But if you turn up the volume clearly above this level (ca. from 120 dB SPL at -0 dB signal
strength), you hear the coding errors which are then pretty different from the older versions.

Now, if you disregard the question "what good are replaygains above +10 dB?" (with classical
music, only album-based replaygain should be used anyway), the problem can be solved by
lowering the ATH. It will result in a slightly higher bitrate.

If this problem is relevant for daily use in any kind of way, i dare say "no".
For most pop titles, you can increase the ATH by 30 dB and still not notice anything.
Even with classical music, 10 dB are often possible.

A clean solution is not possible with a 1-pass-coder; you would first need a rough
volume estimation of the whole song to estimate the maximum position of the volume knob -
and even then, you could still re-adjust during the title.

Furthermore, i would recommend corrections within Replaygain. A "quick-to-hack" solution
would be that the title-based replaygain of neighboring tracks in an album must not
differentiate by more than 6 dB.

From these (calculated) values:

- 7,81 dB
- 6,41 dB
- 7,61 dB
+4,81 dB
- 8,11 dB
- 6,12 dB
+1,12 dB
- 9,12 dB

you will then get:

- 7,81 dB
- 6,41 dB
- 7,61 dB
- 2,11 dB // raised to -8,11 + 6
- 8,11 dB
- 6,12 dB
- 3,12 dB // raised to -9,12 + 6
- 9,12 dB


Then, short voice tracks/interludes/preludes etc. don't get boosted to +40 dB anymore.
Because this is currently the only limit: Replaygain values of more than +40 dB are
simply reduced to 0 dB (not really that clean either). This limit should also be
reduced to +12 dB (corresponds to K-26).

If this proposal is taken up, i could send some reasonably tuned example code.
Somewhere in the depths of my hard disk there should be something.
In that code, the increase of these "holes" is also depending on the Album-replaygain,
the title length and sometimes from more distant neighboring tracks.
A "1 second digital null" before the first title approximately gets the value of the
first track, a "2 second digital null" in between two tracks gets the mean value
of both tracks.



static const Profile_Setting_t Profiles [16] = {
{ 0 },
{ 0 },
{ 0 },
{ 0 },
{ 0 },
/* Short MinVal EarModel Ltq_ min Ltq_ Band- tmpMask CVD_ varLtq MS Comb NS_ Trans */
/* Thr Choice Flag offset TMN NMT SMR max Width _used used channel Penal used PNS Det */
{ 1.e9f, 1, 300, 30, 3.0, -1.0, 0, 106, 4820, 1, 1, 1., 3, 24, 6, 1.09f, 200 }, // 0: pre-Telephone
{ 1.e9f, 1, 300, 24, 6.0, 0.5, 0, 100, 7570, 1, 1, 1., 3, 20, 6, 0.77f, 180 }, // 1: pre-Telephone
{ 1.e9f, 1, 400, 18, 9.0, 2.0, 0, 94, 10300, 1, 1, 1., 4, 18, 6, 0.55f, 160 }, // 2: Telephone
{ 50.0f, 2, 430, 12, 12.0, 3.5, 0, 88, 13090, 1, 1, 1., 5, 15, 6, 0.39f, 140 }, // 3: Thumb
{ 15.0f, 2, 440, 6, 15.0, 5.0, 0, 82, 15800, 1, 1, 1., 6, 10, 6, 0.27f, 120 }, // 4: Radio
{ 5.0f, 2, 550, 0, 18.0, 6.5, 1, 76, 19980, 1, 2, 1., 11, 9, 6, 0.00f, 100 }, // 5: Standard
{ 4.0f, 2, 560, -6, 21.0, 8.0, 2, 70, 22000, 1, 2, 1., 12, 7, 6, 0.00f, 80 }, // 6: Xtreme
{ 3.0f, 2, 570, -12, 24.0, 9.5, 3, 64, 24000, 1, 2, 2., 13, 5, 6, 0.00f, 60 }, // 7: Insane
{ 2.8f, 2, 580, -18, 27.0, 11.0, 4, 58, 26000, 1, 2, 4., 13, 4, 6, 0.00f, 40 }, // 8: BrainDead
{ 2.6f, 2, 590, -24, 30.0, 12.5, 5, 52, 28000, 1, 2, 8., 13, 4, 6, 0.00f, 20 }, // 9: post-BrainDead
{ 2.4f, 2, 599, -30, 33.0, 14.0, 6, 46, 30000, 1, 2, 16., 15, 2, 6, 0.00f, 10 }, //10: post-BrainDead
};


The Ltq_offset entry is the alteration of the masked threshold against the standard model.
A reduction by 6 dB decreases the ATH by 6 dB in the whole frequency range.

The value left of that (EarModel) can be used for ATH fine-tuning for higher frequencies.
An increasing by 20 results in a ATH decrease by 1.5 dB at 10 KHz and 6 dB at 20 KHz.

--quality 6 against --quality 5 has the following differences in the ATH with this:

- 6,0 dB for low frequencies
- 6,5 dB for 8 kHz
- 7,0 dB for 11 kHz
- 8,0 dB for 16,3 kHz
- 9,0 dB for 20 kHz
-10,0 dB for 23 kHz

If there are further questions or if something was unintelligible, just keep asking.
I still have no time, but when i have 15 minutes silence, i can answer to such things.

Motto of the day: The ingeniousness of a construction lies within its simplicity.
Everyone can build something complicated. (Sergeij P. Koroljow)
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