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Lossy Audio Compression => MP3 => MP3 - Tech => Topic started by: Vietwoojagig on 29 January, 2004, 05:38:35 AM

Title: -V n (in 3.95.1)
Post by: Vietwoojagig on 29 January, 2004, 05:38:35 AM
Hi,
Code: [Select]
Switch            equals               target  Y  b  lowpass resample
-V 0            = --preset extreme       240     128  19500
-V 0 --vbr-new  = --preset fast extreme  240     128  19500
-V 1                                     210     128  19000
-V 1 --vbr-new                           210     128  19000
-V 2            = --preset standard      190     128  19000
-V 2 --vbr-new  = --preset fast standard    190     128  19000
-V 3                                     175   1      18000
-V 3 --vbr-new                           175   1      18000
-V 4            = --preset medium        165   1      18000
-V 4 --vbr-new  = --preset fast medium   165   1      18000
-V 5                                     130   1      17000
-V 5 --vbr-new                           130   1      17000
-V 6                                     115   1      16000
-V 6 --vbr-new                           115   1      16000
-V 7                                     100   1      14900   32000
-V 7 --vbr-new                           100   1      14900   32000
-V 8                                      85   1      12500   32000
-V 8 - vbr-new                            85   1      12500   32000
-V 9                                      65   1      10000   24000
-V 9 - vbr-new                            65   1      10000   24000


Questions:
1. Is it save to use the -V switches, with no corresponding preset (especially V 1 / V3)?
2. When is it more usefull, to use abr rather than -V n (e.g. --preset 120 better than -V 5)?
3. Are the displayed target average bitrates ok (+/- 10 kbps)?

Thanks

Edit: changed with Gabriel's suggestions. Added:-Y, -b, --lowpass, --resample
Edit: modified -b to fit with 3.96
Title: -V n (in 3.95.1)
Post by: Gabriel on 29 January, 2004, 06:11:13 AM
Nice layout.

Quote
3. Are the displayed target average bitrates ok (+/- 10 kbps)?

I'd say:
V2 - 190
V3 - 175
V4 - 165
V5 - 130
Title: -V n (in 3.95.1)
Post by: Dologan on 04 May, 2004, 07:35:11 PM
Is --r3mix still remapped to -V3 --vbr-new in 3.96 final?
Title: -V n (in 3.95.1)
Post by: Gabriel on 05 May, 2004, 03:53:34 AM
Quote
Is --r3mix still remapped to -V3 --vbr-new in 3.96 final?

Yes
Title: -V n (in 3.95.1)
Post by: Schinkentoni on 05 May, 2004, 05:05:15 AM
I just want to pick up the first question of Vietwoojagig:

Is it save to use the -V switches, with no corresponding preset (especially V 1 / V3)?
Title: -V n (in 3.95.1)
Post by: 2Bdecided on 05 May, 2004, 06:02:48 AM
The word you're looking for is safe, not save!

safe=not dangerous
save=keep or store

Don't flame me - I'm used to people with English as a second language at work getting irritated if I don't help them. Accepting that, with any language other than English I'm beyond help myself, so any help I give certainly isn't meant as criticism!

Sorry I can't answer the actual question! I guess that V1 can't be worse than V2, and V3 can't be worse than V4, so they're worth a try.

Cheers,
David.
Title: -V n (in 3.95.1)
Post by: Vietwoojagig on 05 May, 2004, 06:42:10 AM
Quote
The word you're looking for is safe, not save!

safe=not dangerous
save=keep or store

Don't flame me - I'm used to people with English as a second language at work getting irritated if I don't help them. Accepting that, with any language other than English I'm beyond help myself, so any help I give certainly isn't meant as criticism!

Oh dear, shame on me. Of course I know the difference between save and safe. But sometimes I don't see the mistakes, even if you would nail them on my head. 
Title: -V n (in 3.95.1)
Post by: Halcyon on 05 May, 2004, 09:15:23 AM
Vietwoojagig,

don't worry. It's just a sign of your lingual mind starting to think phonetically. It's just transitioning from one set of phonemic transcription to another.

I make mistakes like that all the time. I wrote much better English when I knew and used it much less.

These days I'm bi-lingually handicapped. Sometimes I can't express myself properly in my native tongue (or my second native tongue), but English works out ok. Then again, often I think in Finnish and my English comes out really broken.

So, you're actually just advancing

And compared to the average American...  don't even get me started

As for your questions, I don't think there is enough data to conclude 1) yet (at least I haven't found the answer). Well, I'm using -V switches on 3.96, based on some preliminary findings, but no conclusive data is available yet (?).

As for 2) and 3) I'm also interested. Perhaps the developers are the best ones to answer this, if they have time/interest.

cheers,
halcyon

PS It's an odd thing with English speaking natives: they complain when you try to speak their language and they complain when you don't. Nothing's ever good enough. And my God, what happens if you err on the side of correcting their English? A war breaks out, that's what happens ;-D
Title: -V n (in 3.95.1)
Post by: Dologan on 05 May, 2004, 10:22:16 AM
Quote
Quote
Is --r3mix still remapped to -V3 --vbr-new in 3.96 final?

Yes

Quote
Sorry I can't answer the actual question! I guess that V1 can't be worse than V2, and V3 can't be worse than V4, so they're worth a try.

Well, considering the outdated nature of --r3mix and --preset medium being more recent and based off the proven --aps, I wouldn't be too surprised if -V4 (--vbr-any) could actually be better than -V3 --vbr-new, despite the lower bitrate. The question would be whether -V3 (--vbr-old) has more to do with --r3mix or with --aps -Y?
Title: -V n (in 3.95.1)
Post by: Vietwoojagig on 05 May, 2004, 10:41:16 AM
Quote
Well, considering the outdated nature of --r3mix and --preset medium being more recent and based off the proven --aps, I wouldn't be too surprised if -V4 (--vbr-any) could actually be better than -V3 --vbr-new, despite the lower bitrate. The question would be whether -V3 (--vbr-old) has more to do with --r3mix or with --aps -Y?

Only the name "--r3mix" is mapped to"-V3" and not "-V3" with the outdated code. So I'm quite sure that that  V0 > V1 > V2 > V3 > V4 > ... > V9.

As I understand it, the names "r3mix", "preset medium", "preset standard", "preset extreme" are only mapped to V-switches for compatibilty reasons.

In future, I will only use the V-switches, knowing, that V2 is the setting for transparency.
Title: -V n (in 3.95.1)
Post by: Dologan on 05 May, 2004, 11:19:52 AM
Quote
Only the name "--r3mix" is mapped to"-V3" and not "-V3" with the outdated code. So I'm quite sure that that  V0 > V1 > V2 > V3 > V4 > ... > V9.

As I understand it, the names "r3mix", "preset medium", "preset standard", "preset extreme" are only mapped to V-switches for compatibilty reasons.

From what I had understood, the -Vx now equal the --preset they are mapped with. So -V2 is the same as --preset standard and, under the same logic, -V3 --vbr-new would be --r3mix. Now you are telling me then that --r3mix is no longer --r3mix, but -V3 --vbr-new (whatever that is). Is this right? Has --r3mix been reduced to merely a symbolic link pointing to something entirely different?
Title: -V n (in 3.95.1)
Post by: Vietwoojagig on 05 May, 2004, 11:24:21 AM
Quote
From what I had understood, the -Vx now equal the --preset they are mapped with. So -V2 is the same as --preset standard and, under the same logic, -V3 --vbr-new would be --r3mix. Now you are telling me then that --r3mix is no longer --r3mix, but -V3 --vbr-new (whatever that is). Is this right? Has --r3mix been reduced to merely a symbolic link pointing to something entirely different?

Yes
Title: -V n (in 3.95.1)
Post by: Tomb on 05 May, 2004, 11:33:49 AM
Quote
PS It's an odd thing with English speaking natives: they complain when you try to speak their language and they complain when you don't. Nothing's ever good enough. And my God, what happens if you err on the side of correcting their English? A war breaks out, that's what happens ;-D

I say by jove. Steady on old boy, what. Correcting the Queens - now that would be damn foolish move old bean. Spiffing what and pass the Pimms and Gordons.

PS. Yes I am English and even worse, a Cockney 

Talk Cockney here  (http://www.cockneyrhymingslang.co.uk/rabbit_1.asp) or here! (http://www.whoohoo.co.uk/main.asp)
Title: -V n (in 3.95.1)
Post by: Dologan on 05 May, 2004, 12:07:35 PM
Quote
Quote
From what I had understood, the -Vx now equal the --preset they are mapped with. So -V2 is the same as --preset standard and, under the same logic, -V3 --vbr-new would be --r3mix. Now you are telling me then that --r3mix is no longer --r3mix, but -V3 --vbr-new (whatever that is). Is this right? Has --r3mix been reduced to merely a symbolic link pointing to something entirely different?

Yes

Hmm... Interesting.
Gabriel, could you please confirm this? Or Vietwoojagig, could you show me where you know this from? Thank you.
Title: -V n (in 3.95.1)
Post by: Gabriel on 05 May, 2004, 12:38:23 PM
Quote
Gabriel, could you please confirm this?

Yes
Title: -V n (in 3.95.1)
Post by: Jebus on 05 May, 2004, 02:16:28 PM
Wow, that is really interesting. So the --r3mix tunings are really gone now (finally!) and all that remains is the name. This is truely significant!
Title: -V n (in 3.95.1)
Post by: Squeller on 24 May, 2004, 07:02:15 AM
Quote
2. When is it more usefull, to use abr rather than -V n (e.g. --preset 120 better than -V 5)?

I'd be interested in the answer to this question, too. No one has an idea?
Title: -V n (in 3.95.1)
Post by: shafff on 24 June, 2004, 01:56:01 PM
-V2 (lame 3.95.1) gives me 141kbps on film track (pianist, russian translation). Double speed on 2.4 ghz
i'll do -V4 (lame 3.97alpha) this week-end on english track (in hope that picture quality will be better; using xvid).
Title: -V n (in 3.95.1)
Post by: westgroveg on 03 July, 2004, 09:20:55 PM
Does LAME 3.96, -V2/--preset standard still use -b 128?
Title: -V n (in 3.95.1)
Post by: Vietwoojagig on 07 July, 2004, 05:29:55 AM
Quote
Does LAME 3.96, -V2/--preset standard still use -b 128?

Yes. -b 96 did not work on some samples as desired.
Title: -V n (in 3.95.1)
Post by: Gabriel on 04 August, 2004, 03:43:01 AM
Regarding "target bitrates" related to -V x, I'd like to point to this:
http://lame.sourceforge.net/lame_ui_example.html (http://lame.sourceforge.net/lame_ui_example.html)

In this sample UI, you can visually see the usual bitrate range of the different -V x levels
Title: -V n (in 3.95.1)
Post by: 384kbps on 25 August, 2004, 02:58:33 PM
Quote
I just want to pick up the first question of Vietwoojagig:

Is it save to use the -V switches, with no corresponding preset (especially V 1 / V3)?[a href="index.php?act=findpost&pid=208813"][{POST_SNAPBACK}][/a]

I have learned till now the difference between 'save' and 'safe' , but i still can't see here any clear answer on the question above, sorry.

So if I use "Lame.exe -V1 input.wav output.mp3"...
- ...will that produce smaller preset extreme mp3 files by using the 'extreme' maskings/shapings/modells but only a bit more aggressive?
- ...or will that produce bigger preset standard mp3 files by using the maskings/shapings/modells of preset standard but only a bit more gentle?
- ...or will '-V1' and '-V3' work beyond every well-tested preset?

Thanks for my enlightenment!
Title: -V n (in 3.95.1)
Post by: Gabriel on 26 August, 2004, 03:30:03 AM
-V1 is between -V0 and -V2.
I do not know what to say more: it is not -V0 neither -V2, just -V1
Title: -V n (in 3.95.1)
Post by: Pio2001 on 26 August, 2004, 06:11:57 AM
...so -V1 includes the alt-preset code level tunings, but with a bitrate wetween standard and extreme.

I've removed the --r3mix line, since it doesn't mean --r3mix anyway.
Title: -V n (in 3.95.1)
Post by: GeSomeone on 26 August, 2004, 07:26:31 AM
Quote
So the --r3mix tunings are really gone now (finally!) and all that remains is the name. This is truly significant!

No, there were no r3mix tunings. It was just a shortcut for a long string of command-line options that was advocated by some  until it was beaten by (alt) preset standard.

[span style='font-size:8pt;line-height:100%']sorry to reply to such old message[/span]
Title: -V n (in 3.95.1)
Post by: 384kbps on 01 September, 2004, 01:45:57 AM
Quote
...so -V1 includes the alt-preset code level tunings,...
[a href="index.php?act=findpost&pid=237101"][{POST_SNAPBACK}][/a]

Thanks, that's just all i wanted to know!
Title: -V n (in 3.95.1)
Post by: LoFiYo on 29 September, 2004, 11:15:54 PM
Quote
...so -V1 includes the alt-preset code level tunings, but with a bitrate wetween standard and extreme.
I thought Gabriel was trying to move away from Dibrom's tuning (or "hack") by tuning in non-hack ways. In other words, hasn't he been removing more and more of Dibrom's tuning and trying to make it sound good by working from other angles? I don't understand source codes, so I can't be sure, but could someone (or Gabriel himself) comment on this? Sorry if I'm totally mistaken 
Title: -V n (in 3.95.1)
Post by: Gabriel on 30 September, 2004, 06:22:38 AM
Quote
I thought Gabriel was trying to move away from Dibrom's tuning (or "hack") by tuning in non-hack ways. In other words, hasn't he been removing more and more of Dibrom's tuning and trying to make it sound good by working from other angles?
[a href="index.php?act=findpost&pid=245369"][{POST_SNAPBACK}][/a]


No. My goal was to change the way those "tunings" are enabled. I wanted to be able to set each of those by separate parameters instead of a few "multi-purpose" parameters.
Title: -V n (in 3.95.1)
Post by: detokaal on 18 October, 2004, 04:30:33 PM
Would it be possible to add the -q switch to the tables?  I believe --preset standard  is -q 2 whereas --preset medium is -q 3.

Also, I am assuming there is no harm in say, --preset medium -q 0, since it theoretically may increase quality, even though it is much slower?
Title: -V n (in 3.95.1)
Post by: Gabriel on 19 October, 2004, 04:56:00 AM
Quote
I believe --preset standard is -q 2 whereas --preset medium is -q 3

No
Title: -V n (in 3.95.1)
Post by: detokaal on 19 October, 2004, 05:02:28 PM
Rather than ask again - I just checked myself.  --preset medium is qual=3, --preset standard is qual=3, and --preset extreme is qual=3 also according to --verbose command line when using 3.96.1.  Interesting that even --preset insane is qual=3.  It appears that -q 0/1/2 isn't used in any of the presets, unless I have something in EAC set wrong and it is adding other command line switches I don't know about.
Title: -V n (in 3.95.1)
Post by: freakngoat on 19 October, 2004, 05:19:36 PM
Quote
Rather than ask again - I just checked myself.  --preset medium is qual=3, --preset standard is qual=3, and --preset extreme is qual=3 also according to --verbose command line when using 3.96.1.  Interesting that even --preset insane is qual=3.  It appears that -q 0/1/2 isn't used in any of the presets, unless I have something in EAC set wrong and it is adding other command line switches I don't know about.
[a href="index.php?act=findpost&pid=248755"][{POST_SNAPBACK}][/a]


What's with the q-value obsession? I'm not picking on you detokaal, I've been seeing it creeping up in other discussions. AFAIK, the -V settings were tuned using a given q-value and changing this isn't a reliable way to adjust quality. Rather just change the -V value. It seems the -V values have been distributed to cover the full bitrate spectrum, so it seems that anyone could find the ideal quality/space tradeoff for their particular ears and application.

Though I guess if you really want to be experimental, go for it. Might as well start tweaking the code as well. Perhaps Gabriel can either validate or invalidate this point.
Title: -V n (in 3.95.1)
Post by: Gabriel on 20 October, 2004, 03:45:18 AM
Yes, changing -V is (generally) a better idea than changing -q.

Btw, I think that using the DEFAULT setting in current versions is fine. That mean that for vbr, you only adjust -V according to your needs and do not change anything else.
Title: -V n (in 3.95.1)
Post by: ezra2323 on 14 December, 2004, 09:51:15 PM
After reading through this thread, I'm still unclear (forgive me if I just don't GET IT). Is 3.96.1 -v5 a superior alternative to 3.90.3 --alt preset 128?

By superior, I mean superior sound quality for the targeted file size. Also, will the extra lowpass command used in Roberto's listening test improve -V5?

Since 3.96.1 -v5 was selected over 3.90.3 --alt preset 128 for Roberto's multi-format listening test, I ASSUME it is superior but I hate to assume anything in this forum. 
Title: -V n (in 3.95.1)
Post by: dev0 on 15 December, 2004, 12:12:06 AM
Quote
After reading through this thread, I'm still unclear (forgive me if I just don't GET IT). Is 3.96.1 -v5 a superior alternative to 3.90.3 --alt preset 128?[a href="index.php?act=findpost&pid=259777"][{POST_SNAPBACK}][/a]

Yes.
Title: -V n (in 3.95.1)
Post by: k.eight.a on 15 December, 2004, 12:32:30 PM
Quote
2. When is it more usefull, to use abr rather than -V n (e.g. --preset 120 better than -V 5)?
[a href="index.php?act=findpost&pid=177935"][{POST_SNAPBACK}][/a]

From the very start of this thread I'm still interested...
Quote
Quote
After reading through this thread, I'm still unclear (forgive me if I just don't GET IT). Is 3.96.1 -v5 a superior alternative to 3.90.3 --alt preset 128?[a href="index.php?act=findpost&pid=259777"][{POST_SNAPBACK}][/a]

Yes.
[a href="index.php?act=findpost&pid=259803"][{POST_SNAPBACK}][/a]

This is only a small part of the answer.
Title: -V n (in 3.95.1)
Post by: whcanilang on 22 December, 2004, 03:16:17 PM
Quote
After reading through this thread, I'm still unclear (forgive me if I just don't GET IT). Is 3.96.1 -v5 a superior alternative to 3.90.3 --alt preset 128?

By superior, I mean superior sound quality for the targeted file size. Also, will the extra lowpass command used in Roberto's listening test improve -V5?

Since 3.96.1 -v5 was selected over 3.90.3 --alt preset 128 for Roberto's multi-format listening test, I ASSUME it is superior but I hate to assume anything in this forum. 
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=259777")


I just noticed that [a href="http://lame.sourceforge.net/USAGE]http://lame.sourceforge.net/USAGE[/url] now discourages the use of "-V 5" in favor of ABR 128.  Where is Roberto's listening comparison test?  I missed it and I'd like to read it.  Thanks!
Title: -V n (in 3.95.1)
Post by: music_man_mpc on 22 December, 2004, 03:42:30 PM
Quote
Where is Roberto's listening comparison test?  I missed it and I'd like to read it.  Thanks!
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=261259")

[a href="http://www.hydrogenaudio.org/forums/index.php?showtopic=21904]http://www.hydrogenaudio.org/forums/index....showtopic=21904[/url]
Title: -V n (in 3.95.1)
Post by: Gabriel on 23 December, 2004, 03:16:39 AM
Quote
I just noticed that http://lame.sourceforge.net/USAGE (http://lame.sourceforge.net/USAGE) now discourages the use of "-V 5" in favor of ABR 128.

It is not "now". It's just that it hasn't been updated for a while.
note: I'd welcome an updated version
Title: -V n (in 3.95.1)
Post by: esa372 on 19 January, 2005, 12:23:50 PM
Quote
Is 3.96.1 -v5 a superior alternative to 3.90.3 --alt preset 128?


Quote
For ~ 128kbps VBR, the general consensus has been to use "-V5 --athaa-sensitivity 1" with LAME 3.95.1 or higher. The ATH adjustment was found to result in less warbling/phasing, but may raise the bitrate a little bit.


[span style='font-size:8pt;line-height:100%'](From the best settings for 128kbps VBR (http://www.hydrogenaudio.org/forums/index.php?showtopic=30667) thread.)[/span]



:edit: link
Title: -V n (in 3.95.1)
Post by: veikko on 09 February, 2005, 05:25:55 PM
Hi.

I just registered myself for the first time ever to a forum of anykind

But this just seemed like a place of people who know the stuff they talk about,
and that's always good. Also this place has had by far the best attitude towards
"the new guy" making his first post, so bear with me, since this truly is my first post

So, here's my thing, I'm in a situation now to encode all of my audio cds
to mp3 format.

By judging form the discussions here, i really should use eac or plextools
for the ripping and lame (3.90.3 or 3.96.1 if i'm correct?) for the encoding.

Even more reading of this forum has led me to the conclusion to use vbr ape
or as you would put it in the format of -V0 while using lame 3.96.1.

So the thing I would like to know is about the q value,
is there any point in adjusting it upwards trough the commandline myself?

I mean seriously, I thought about this long time, that do I even dare to ask such a question,
because I respect what Gabriel has stated about the defaults,
but I just want to know that what are the potential effects of changing
the qval to 2 or even 1 or 0. What does it really change?
Does it affect the quality of the audio in any way?
I even read about it potentially lessening the outcoming quality of the finished "product",
is that still true?

well nothing more at this point, hope this didn't  strike you guys as a totally stupid question.

thanks.
Title: -V n (in 3.95.1)
Post by: Gabriel on 09 February, 2005, 05:34:39 PM
A bug regarding -q0 and -q1 has been recently identified and corrected in 3.97 alphas.
3.94-3.96.1 were affected by this problem. Regarding 3.90.3, I do not remember as it is too old.
Title: -V n (in 3.95.1)
Post by: kwanbis on 09 February, 2005, 06:30:47 PM
Garbiel would this fix be backported to 3.96.x?
Title: -V n (in 3.95.1)
Post by: Gabriel on 10 February, 2005, 03:51:03 AM
Probably not as we are not working on 3.96 anymore.
Title: -V n (in 3.95.1)
Post by: moi on 26 February, 2005, 03:06:26 PM
Quote
Nice layout.

Quote
3. Are the displayed target average bitrates ok (+/- 10 kbps)?

I'd say:
V2 - 190
V3 - 175
V4 - 165
V5 - 130
[a href="index.php?act=findpost&pid=177942"][{POST_SNAPBACK}][/a]


To get the average bit rate closer to 128 kbps, would you recommend"

V5

or--

with the switches used in the listening test--

V5 --athaa-sensitiivity 1  ??

What exactly does that --athaa-sensitiivity 1 mean, and what difference does it make over just V5?
Title: -V n (in 3.95.1)
Post by: Jojo on 26 February, 2005, 06:40:22 PM
Quote
To get the average bit rate closer to 128 kbps, would you recommend"

V5

or--

with the switches used in the listening test--

V5 --athaa-sensitiivity 1   ??
[a href="index.php?act=findpost&pid=277345"][{POST_SNAPBACK}][/a]

use LAME 3.96.1 and -V5 --athaa-sensitivity 1. It results better qualits than -V5 (applies to -V5 only!) I've encoded 5 files or so and in average they were 128kbps. However, it is possible that some files have 111kbps and others 134kbps...Even if you get a bit over 128kbps in average, let's say 132kbps...who cares...it's not that much of a difference (in my example 0,5 kilobyte per second = 30 kilobyte per minute)...not a big deal, but the quality is the best mp3 has to offer in the 128kbps range...
Title: -V n (in 3.95.1)
Post by: JEN on 06 March, 2005, 02:02:07 PM
my hearing is either getting really bad (i'm 26 years old!) or lame 3.96.1 is amazing because i cant abx -v6 --vbr-new 

So I guess that means lame 3.96.1 @ -v6 --vbr-new will be my new portable choice.
Title: -V n (in 3.95.1)
Post by: Dologan on 06 March, 2005, 02:19:22 PM
Quote
use LAME 3.96.1 and -V5 --athaa-sensitivity 1. It results better qualits than -V5 (applies to -V5 only!) I've encoded 5 files or so and in average they were 128kbps. However, it is possible that some files have 111kbps and others 134kbps...Even if you get a bit over 128kbps in average, let's say 132kbps...who cares...it's not that much of a difference (in my example 0,5 kilobyte per second = 30 kilobyte per minute)...not a big deal, but the quality is the best mp3 has to offer in the 128kbps range...
[a href="index.php?act=findpost&pid=277385"][{POST_SNAPBACK}][/a]

Actually, the --athaa-sensitivity 1 switch was also recommended for -V4 on some thread I can't seem to find right now. I use it for my -V4 encodes and I haven't noticed any problems at all. (Actually, I can't ABX the difference, but if the golden ears say it's better, I trust them )
Title: -V n (in 3.95.1)
Post by: schonenberg on 05 April, 2005, 03:38:56 AM
Quote
Hi.

I just registered myself for the first time ever to a forum of anykind

But this just seemed like a place of people who know the stuff they talk about,
and that's always good. Also this place has had by far the best attitude towards
"the new guy" making his first post, so bear with me, since this truly is my first post

So, here's my thing, I'm in a situation now to encode all of my audio cds
to mp3 format.

By judging form the discussions here, i really should use eac or plextools
for the ripping and lame (3.90.3 or 3.96.1 if i'm correct?) for the encoding.

Even more reading of this forum has led me to the conclusion to use vbr ape
or as you would put it in the format of -V0 while using lame 3.96.1.

So the thing I would like to know is about the q value,
is there any point in adjusting it upwards trough the commandline myself?

I mean seriously, I thought about this long time, that do I even dare to ask such a question,
because I respect what Gabriel has stated about the defaults,
but I just want to know that what are the potential effects of changing
the qval to 2 or even 1 or 0. What does it really change?
Does it affect the quality of the audio in any way?
I even read about it potentially lessening the outcoming quality of the finished "product",
is that still true?

well nothing more at this point, hope this didn't  strike you guys as a totally stupid question.

thanks.
[a href="index.php?act=findpost&pid=272169"][{POST_SNAPBACK}][/a]



That is a valid question, one I bothered people with myself. -q 0 increases the quality of the psychoacoustic algorithm, it does not touch the bitrate at all.
In simple terms Lame will spend much more time examining each sample when encoding.

Lame 3.90.3 at --alt-preset standard is the recommended mp3 settings. Lame 3.97 final might replace 3.90.3 as the best encoder.


I use lame 3.96.1 -V 5 --athaa-sensitivity 1 -q 0, which might not improve the quality much over not using -q, but encodes fast enough for me and sounds good on my apex dvd player and my pc/stereo setup.

For cd's that I have to archive and won't have access to in the future, I use Musepack/MPC at --standard. MPC has fewer problem samples at higher bitrates and is tuned only for high transparent bitrates, but is not supported by any portables  . It is very good for computer use.
Title: -V n (in 3.95.1)
Post by: guruboolez on 05 April, 2005, 03:48:37 AM
Quote
Lame 3.90.3 at --alt-preset standard is the absolute best mp3, except in certain samples.
[a href="index.php?act=findpost&pid=288389"][{POST_SNAPBACK}][/a]

preset standard is not the "absolute best" mp3. It's a compromize between transparency (I'd rather say 'robustness') and bitrate. For many people, --preset standard is not a good compromize: they could obtain the same level of transparency at lower bitrate with lower presets. For other people, --preset extreme improves the quality, and not only on killer samples.
For absolute best mp3, use --preset insane (or even freeformat at 640 kbps).
Title: -V n (in 3.95.1)
Post by: Madrigal on 05 April, 2005, 08:46:17 AM
Quote
Hi.

I just registered myself for the first time ever to a forum of anykind

But this just seemed like a place of people who know the stuff they talk about,
and that's always good. Also this place has had by far the best attitude towards
"the new guy" making his first post, so bear with me, since this truly is my first post [a href="index.php?act=findpost&pid=272169"][{POST_SNAPBACK}][/a]
Hi.

Welcome to HA. I sincerely hope your experience here is as enlightening and meaningful as mine has been.

Regards,
Madrigal
Title: -V n (in 3.95.1)
Post by: TheStonepedo on 19 May, 2005, 11:42:58 PM
I fail to understand the b value of 128 on -V 0.  Is there a simple explanation for why the bitrate should not vary in the full 32-320 kbps range?  Also, can -b 32 override -V 0?
For example would running:
-V 0 -b 32 -B 320
make any difference at all, or would the preset-style V switch override the b and B?
Title: -V n (in 3.95.1)
Post by: tycho on 20 May, 2005, 02:36:47 AM
I believe the -b 32 will override the default (128 in this case).

/Add: I think the simple explaination why 32-112 range should not be used at all is that more or less any samples encoded with these bitrates will not be transparent (except digital silence - which is btw encoded as 32kbps anyway).

On a resembling matter, I've been using Audiograbber (ugh!) with the lame_enc.dll (3.96.1), with -V 2 equvalent, and it appears that it uses -b 96 as default? Should be 128. It could be that Audiograbber sets -b 96 internally, but it seems unlikely. I ask because I wonder if the lame dll has the same mappings as the exe.
Title: -V n (in 3.95.1)
Post by: Jojo on 20 May, 2005, 12:20:30 PM
Quote
I fail to understand the b value of 128 on -V 0.  Is there a simple explanation for why the bitrate should not vary in the full 32-320 kbps range?  Also, can -b 32 override -V 0?
For example would running:
-V 0 -b 32 -B 320
make any difference at all, or would the preset-style V switch override the b and B?
[a href="index.php?act=findpost&pid=298963"][{POST_SNAPBACK}][/a]

you shouldn't mess with the --presets at all. So just use -V0 and that's it. No -b or -B unless you like to waste space or worse sound quality
Anyway, LAME 3.96.1 uses -b 128 as default, however the newest LAME 3.97 alpha builts use -b 32. That's why it is the best to just stick with the presets. Note: LAME Alpha and Beta release shouldn't be used for anything else than testing!
Title: -V n (in 3.95.1)
Post by: =trott= on 23 June, 2005, 07:02:48 AM
Quote
I fail to understand the b value of 128 on -V 0. 
[a href="index.php?act=findpost&pid=298963"][{POST_SNAPBACK}][/a]


The way I understand it is this: lame has different variable bitrate modes. you have ABR and a true VBR mode. (the ABR is what was used with the old r3mix.net settings)
However, apparantly the VBR mode is somewhat unreliable below 112 Kbits/sec. It will mostly produce good encodes but may output some low-quality parts. Therefore, lame in VBR mode (which is what the presets use) has a bitrate floor of 112 Kbits/sec.

Though I may be wrong
Title: -V n (in 3.95.1)
Post by: guruboolez on 23 June, 2005, 07:22:59 AM
Quote
you have ABR and a true VBR mode. (the ABR is what was used with the old r3mix.net settings)


No, --r3mix was VBR, not ABR.

Quote
Therefore, lame in VBR mode (which is what the presets use) has a bitrate floor of 112 Kbits/sec.
.
Not anymore. At least --vbr-new mode in the current 3.97 alpha.
Title: -V n (in 3.95.1)
Post by: latuman on 16 November, 2005, 06:20:55 AM
Hi!

I'm new to this forum. I registered because I needed some encoding information for my new-born Creative MuVo TX SE. Since it's only 256MB, I'd like to fill it up with proper encodings.

Sorry if this is the wrong section or something. I tried searching for topics about mp3 player encodings but failed.

So. I use dBPowerAmp which has command line access to Lame, and Lame is the latest 3.97b I believe. The one recommended here anyway.

Since I'm probably going to invest to Sennheiser PX-100 or something similar, I figured -V 6 might be appropriate.

-V6 --vbr-new was in mind actually. Now I have read about this -Y parameter, which cuts everything above 16kHz. I tried it, but it just produced the same results as in normal -V6. Also, I checked the full command line reference, and it was not there. Is this somehow obsolete and/or replaced by some other parameter?

I put my money on --highpass 16 , but I dont think it did anything. Probably not the switch I'm searching for or completely wrong anyway.

I'm not an Audiophile. I was considering -V8 but thought that new headphones might require the -V6 so I was sticking with that. I guess its pretty good.

I can now fit 3 -V2 albums on this player. More is required and that quality is an overkill for me anyway (although I do encoded all my CD's with that).

Thank you for your time!

And the final question is this: The whole thing with --alt preset being ABR, was this true in some versions? I always wanted the recommended --alt preset standard but ended up with bitrates never topping 224kbps. With the latest Lame though, no problems, just curious about why did I had this experience in the past.
Title: -V n (in 3.95.1)
Post by: Alex B on 16 November, 2005, 07:05:57 AM
Quote
-V6 --vbr-new was in mind actually. Now I have read about this -Y parameter, which cuts everything above 16kHz. I tried it, but it just produced the same results as in normal -V6. Also, I checked the full command line reference, and it was not there. Is this somehow obsolete and/or replaced by some other parameter?

I put my money on --highpass 16 , but I dont think it did anything. Probably not the switch I'm searching for or completely wrong anyway.

A quote from http://lame.sourceforge.net/doc/html/switchs.html (http://lame.sourceforge.net/doc/html/switchs.html):
"--highpass    highpass filtering frequency in kHz
Set an highpass filtering frequency. Frequencies below the specified one will be cutoff."

-V6 --vbr-new uses a lowpass of 15600 Hz. You should not add any other switches. In my opinion -V6 --vbr-new is a good and balanced option to use with portables. I have found that it produces bitrates from about 100 to 150 kbps with various music genres (http://www.hydrogenaudio.org/forums/index....ndpost&p=335491 (http://www.hydrogenaudio.org/forums/index.php?showtopic=38020&view=findpost&p=335491)).


Quote
I'm not an Audiophile. I was considering -V8 but thought that new headphones might require the -V6 so I was sticking with that. I guess its pretty good.

In my experience -V8 is inferior. It resamples to 32 kHz and uses a lowpass of 12500 Hz, but that doesn't even solve the quality problems -V8 has below 12500 Hz. However, just the lowpass setting clearly changes the perceived quality with many music types if the listener can hear frequencies over 12500 Hz. On the other hand, hearing a lowpass of 15.600 Hz is much harder. You can try that for example by encoding a test sample with -b 320 and -b 320 --lowpass 15.6. (I have done personal ABX tests and I suppose other listening tests that can confirm this have been published at HA.)


Edit: There seems to be a small typo in the LAME documentation: "Set an highpass filtering frequency."
Title: -V n (in 3.95.1)
Post by: latuman on 16 November, 2005, 11:27:58 AM
I might be a bit thick, but whats the point using --lowpass ?

I tried -b 320 --lowpass 12.5 just for kicks, and the filesize was the same, however the sound was clearly inferior.

I need to mentiont that I never understood these kHz terms. Sometimes they represent how high a sound is and sometimes... not? Tried also --lowpass 15.6 and it sure was transparent to the original.
Title: -V n (in 3.95.1)
Post by: Gambit on 16 November, 2005, 06:22:26 PM
Quote
I might be a bit thick, but whats the point using --lowpass ?

I tried -b 320 --lowpass 12.5 just for kicks, and the filesize was the same, however the sound was clearly inferior.

I need to mentiont that I never understood these kHz terms. Sometimes they represent how high a sound is and sometimes... not? Tried also --lowpass 15.6 and it sure was transparent to the original.
[a href="index.php?act=findpost&pid=342318"][{POST_SNAPBACK}][/a]

That's a CBR encoding, obviously lowpassing has no effect on the filesize...
Title: -V n (in 3.95.1)
Post by: Alex B on 16 November, 2005, 07:16:53 PM
Quote
I might be a bit thick, but whats the point using --lowpass ?

I tried -b 320 --lowpass 12.5 just for kicks, and the filesize was the same, however the sound was clearly inferior.

I need to mentiont that I never understood these kHz terms. Sometimes they represent how high a sound is and sometimes... not? Tried also --lowpass 15.6 and it sure was transparent to the original.[a href="index.php?act=findpost&pid=342318"][{POST_SNAPBACK}][/a]

Cutting some more or less inaudible high frequencies off can make the overall audible quality better. The space is always limited with MP3 files. LAME adjusts this automatically so you don't have to worry about manual tweaking.

I mentioned -b 320 just because it is the highest quality setting LAME 3.97b1 has. It is always constant 320 kbps so the resulting file size is also constant like Gambit said. Using it for testing the audibility of a lowpass setting would minimize the possible effect of other factors.
Title: -V n (in 3.95.1)
Post by: latuman on 17 November, 2005, 04:54:07 AM
So I shall leave that setting alone for portable use. -V6 then is the way to go
Title: -V n (in 3.95.1)
Post by: DickxLaurent on 18 June, 2006, 10:44:59 PM
Is there an updated copy of the chart in the first post that would coincide with LAME 3.97b?  For example, in my own observations I've found that -V3 now uses a lowpass around 16000.

Please let me know if and where I can find a current reference.  Thanks!
Title: -V n (in 3.95.1)
Post by: Alex B on 18 June, 2006, 11:42:58 PM
I have posted an updated table here: http://www.hydrogenaudio.org/forums/index....ndpost&p=335491 (http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=38020&view=findpost&p=335491). The table was for b1, but I think b2 uses the same lowpass values.
Title: -V n (in 3.95.1)
Post by: DickxLaurent on 19 June, 2006, 12:19:10 AM
I have posted an updated table here: http://www.hydrogenaudio.org/forums/index....ndpost&p=335491 (http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=38020&view=findpost&p=335491). The table was for b1, but I think b2 uses the same lowpass values.

Thanks. I appreciate the update.
Title: -V n (in 3.95.1)
Post by: pika2000 on 19 June, 2006, 01:59:49 AM
For example, in my own observations I've found that -V3 now uses a lowpass around 16000.

That's because -V 3 and greater use -Y switch.
Title: -V n (in 3.95.1)
Post by: Jojo on 20 June, 2006, 03:21:28 PM
the minimum bitrate should be removed from the chart since it is no longer being used
Title: -V n (in 3.95.1)
Post by: ThyBzi on 29 November, 2006, 02:36:48 PM
Sorry for possible offtopic. I just found this topic as closest to my problem, but it can equally well be a new topic.

So, what's the reason for me to be confused. When I've changed my SB Live! 5.1 to Audigy 2 ZS, I just started to hear some higher frequencies of sound (including my huge collection of self-grabbed mp3s). And I used (and use  LAME to encode them.

I started with YAMP encoder which was using LAME 3.80, then I moved to EAC & LAME 3.92 (newest release for that time). I've been hearing and analyzing the sound spectre of output mp3s, and arrived at a conclusion that the best (in size/quality) variant for me is:
Code: [Select]
-b 32 -m j -h -V 2 -B 320


Then there was 3.93 and 3.93.1. I've been grabbing and compressing and expanding my mp3 collection. Grabbing, and compressing, and extending...

Then here came the 3.95.
Right away I noticed that mp3s I'm getting with it (using same parameters) are oftenly slightly smaller. That was the reason to REGRAB and RECOMPRESS all my collection (about 40 or 50 GB for that time).

Then there were 3.95.1, 3.96 and 3.96.1... Regrabbing, recompressing, extending collection...

And then "here came a new challenger" (© Toshiden Battle Arena . I've bought Audigy 2 ZS.

First time there was some kind of euphoria - I started to hear HIGHER sounds on my computer! All that hi-hats' and cymbals' high harmonies, and so on...

Then it turned to some kind of depression. I started to hear mp3-specific distortions in MY MP3s! Which were made with my LAME parameters (I also recommended using them to all my friends... and so on). It was a real shock for me, because I used to consider my mp3s ideal.

So, I started to re-examine various mp3 codecs, incl. FhG, GoGo, SCMPX, Blade, Xing and recently appeared LAME 3.97.

The results I got for LAME shocked my once more. Awful, just AWFUL distortions all above 16-17 KHz. Even 320 at Stereo (not Joint Stereo) mode, and even with filter disabling (-k). I tried lot of different options and their combinations, all of them resulted much worse quality then trivial 3.93.1 at 320, Stereo mode (with -k), or at VBR2 Stereo (with -k also), and even worse then LAME 3.93.1 of GoGo gives me at 224 kbps or even 192 kbps (no matter Stereo or Joint Stereo mode)!

I tried using q0, and V0, and so on - it didn't improve anything enough.

I also tried various mp3 decoders: Winamp 5.1, Adobe Audition (in which I analyze sound frequency spectre) and LAME by itself. In Audiotion results were even more awful, then in Winamp and LAME, but in them both spectre was still too bad at look (at freqs above), and I heared that all in Winamp.

Even more - those distortions (looking like some kind of collapse of frequency spectre graph) grew more and  more for 3.95.1 -> 3.96.1 -> 3.97. I just did't beleive my eyes that LAME produced that collapse.

So I temporary turned back to 3.93.1, which produces an ideal graph allover the sound spectre (even on freqs above 20 KHz!) when using the following parameters:
Code: [Select]
 -b 128 -m s -h -V 2 -B 320 -k

The resulting mp3 freq graph doesn't drift in any significant way from the from original WAV graph. 320 kbps' graph (@3.93.1) also doesn't differ from the original in any way, but average filesize increases significantly.

Could your please explain, what happened with LAME starting from 3.95? (or 3.94, which I didn't test)
Could it be some problem of my decoder (even despite I used LAME to decode mp3s)? Or what?

Thank your in advance.

P.S. I can post, or send, or attach here (if it's allowed), or put at your disposal in any other way the graphical results of spectral analysis I carried out.
Title: -V n (in 3.95.1)
Post by: jmartis on 29 November, 2006, 03:25:30 PM
Spertrograms are no judge of sound quality. Use an ABX comparsion tool to make sure you hear differencies between the original file and the compressed mp3.

(edit- spelling)
Title: -V n (in 3.95.1)
Post by: Firon on 29 November, 2006, 09:27:29 PM
-k will absolutely destroy the quality of the files. Also, M/S stereo will make things sound WORSE. Leave it as joint stereo (it is NOT lossy in LAME). -h does nothing for VBR, -q0 does nothing for VBR. Setting min/max bitrate is not needed in 3.97 when using VBR.
Spectral analysis is virtually useless for measuring sound quality.


Stop screwing with the parameters and just use -V2 --vbr-new
Nothing more, nothing less. Try LAME 3.97 or 3.98a7 if you prefer going a little on the wild side. Then ABX it if you still think you hear "distortions".
Title: -V n (in 3.95.1)
Post by: greynol on 29 November, 2006, 09:29:18 PM
-m s isn't such a bright idea either. 

Never mind, you caught it in your edit.
Title: -V n (in 3.95.1)
Post by: ThyBzi on 30 November, 2006, 01:22:10 PM
Thanx for your support, people

But, as far as I understood, ABX test is based on "blind" listening to audio, but my acoustic system (and perhaps my ears ))) is not ideal, so in future (when I purchase a better one) I could regret my choise made now. Are there any less subjective, MORE objective test? Which can reveal the real (no only heared) differences between two soundtracks?

My father offered my to "subtract" original signal from the encoded->decoded one, and then to estimate the "difference signal" remained. Is this method good, how do you think? Are there any software to do that? AFAIK, Sound Forge and CoolEdit can't...

Or may be some graphic tests?

And one more question. You say spectral analysis couldn't be objective in this way. But if it shows some evident distortions, collapses etc - could they NOT be real distortions of sound and its quality? Especially if they aren't above 20 KHz, but within 16-20 KHz interval?

(Of course, I agree that even identical graphs do not obligatory mean identical sound, or sound of identical quality.)

P.S. By the way, I did't understand from the FAQ, what does -vbr-new key?

P.P.S. Joint Stereo "pushes together" channels, when they spead stronger than in "regular song"... Am I wrong? Does Stereo mode produce better quality? Does it increase filesize considerably?
Title: -V n (in 3.95.1)
Post by: halb27 on 30 November, 2006, 01:45:49 PM
To me the main problem is that you seem to think it's absolutely necessary to get everything until 20 kHz.
However our ears are less and less insensitive the more you go beyond 16 kHz. This is true even for young people though they are able to hear stuff beyond 16 kHz.

If you care so much about that you should absolutely do an abx test. You don't need good equipment to find out whether or not you need HF beyond say 18 kHz.
Try Lame -V3 which has an 18 kHz lowpass and report us about whether or not you're really missing HF or find things distorted (which of course is possible only if you can abx something).

You should be a lot more afraid of -k which might give you more HF but as well a much higher chance that you get audible distortions.
Title: -V n (in 3.95.1)
Post by: greynol on 30 November, 2006, 01:49:48 PM
Are there any less subjective, MORE objective test? Which can reveal the real (no only heared) differences between two soundtracks?
Unlike your ears, your eyes aren't meant for hearing.

My father offered my to "subtract" original signal from the encoded->decoded one, and then to estimate the "difference signal" remained. Is that method good, how do you think?
No.

CoolEdit can't do that...
Acutally CoolEdit can, but it tells you absolutely nothing about how your brain interprets what your ears are hearing.

Or may be some graphic tests?
NO!  Listening tests are all that matter.

And one more question. You say spectral analysis couldn't be objective in this way. But if it shows some evident distortions, collapses etc - could they NOT be real distortions of sound and its quality?
Of course they represent real "distortions", but you need to ask yourself if you can hear these distortions in a blind listening test.

Joint Stereo "pushes together" channels, when they spead stronger than in "regular song"... Am I wrong?
Read this (http://wiki.hydrogenaudio.org/index.php?title=Joint_stereo).

Does Stereo mode produce better quality?
No, it usually produces worse quality at any given bitrate.

Does it increase filesize considerably?
For the same quality?  Yes!
Title: -V n (in 3.95.1)
Post by: pepoluan on 30 November, 2006, 02:28:21 PM
Thanx for your support, people

But, as far as I understood, ABX test is based on "blind" listening to audio, but my acoustic system (and perhaps my ears ))) is not ideal, so in future (when I purchase a better one) I could regret my choise made now. Are there any less subjective, MORE objective test? Which can reveal the real (no only heared) differences between two soundtracks?

My father offered my to "subtract" original signal from the encoded->decoded one, and then to estimate the "difference signal" remained. Is this method good, how do you think? Are there any software to do that? AFAIK, Sound Forge and CoolEdit can't...
CoolEdit can do that easily, but...

Our ear is extremely complex. "Visual" differences, even "extremely evident" visual differences... may not be audible. Why? Simply because our hearing, whcih comprises the ears, the inner ear system, the auditory nerves, and the brain, will perform some extremely complex post-processing and filtering.

Case in point: Try encoding with the highest quality of LAME ( -V 0), or Vorbis (-q 10). Decode. Subtract from the original wave. You will see "extremely evident" visual differences. But even if you pump the decoded wave (i.e. not original) through a 1-million-dollar universal-audiophile-grade equipment... the chances that you will hear any difference is less than 1 ppm (part per million).

Or may be some graphic tests?
Like I said above, what's visible to your eye may never, ever be audible to your ear.

And one more question. You say spectral analysis couldn't be objective in this way. But if it shows some evident distortions, collapses etc - could they NOT be real distortions of sound and its quality? Especially if they aren't above 20 KHz, but within 16-20 KHz interval?
Spectral analysis is not un-objective, it is simply useless. "Visible" distortion, even "extremely evident" visible distortion, like I wrote above, may not be audible to your ear even the slightest.

P.P.S. Joint Stereo "pushes together" channels, when they spead stronger than in "regular song"... Am I wrong? Does Stereo mode produce better quality? Does it increase filesize considerably?
You are partially wrong.

In low-bitrate, Joint Stereo is really Intensity Stereo. *That* will collapse stereo separation.

On higher bitrates, Joint Stereo is actually Mid-Side (MS) Coding. Instead of coding L & R channels separately, it encodes M=(L+R)/2 and S=(L-R)/2. The L & R channels can be recovered very easily: L=M+S and R=M-S. The reason for using MS is that since the S channel usually requires far less bitdepth, encoding using MS will be smaller than encoding using LR, provided that the difference between L & R are not too big. LAME is capable of determining when to use MS and when to use LR.

See this page: http://harmsy.freeuk.com/mostync/ (http://harmsy.freeuk.com/mostync/) (also contained in Greynol's link to "Joint Stereo" on HA wiki).

Edit: Rats. Greynol is faster
Title: -V n (in 3.95.1)
Post by: ThyBzi on 30 November, 2006, 02:31:27 PM
"(The mere fact that Joint Stereo is used in lossless compression ought to be enough to destroy - in one stroke - the myth that JS "destroys stereo separation")" (cite from http://harmsy.freeuk.com/mostync/) (http://harmsy.freeuk.com/mostync/)) - I like this argument! It's quite cogent  But then I don't understand for what reason most CBR mp3s are made using S mode (even on LAME). And for what reason all other Stereo modes are supported by LAME?

Quote
Acutally CoolEdit can, but it tells you absolutely nothing about how your brain interprets what your ears are hearing.

CoolEdit can subtact noise, but it's not the same. It uses the noise sample not "as is", but as a kind of "spectre", am I wrong? Or did I miss some function of CoolEdit, or Adobe Audition (for now)?

Quote
Of course they represent real "distortions"

But LAME 3.93 and GoGo didn't produce them... Is this "feature" (new to 3.95 and the following versions) just an improvement - i.e. to give more bits to the "more useful" freqs?

...So, as I understood, the only way for me to get a better quality is to get a better soundsystem and to carry out the ABX test.

By the way, can you suggest how to choose most appropiate sample for tests? I like metal music, and oftenly there are lot of live cymbals & hihats, which sound good, and have much high harmonies...
...Or it just may be any sample where I can hear distortions (or maybe just invent them ?
Title: -V n (in 3.95.1)
Post by: pepoluan on 30 November, 2006, 02:40:18 PM
"(The mere fact that Joint Stereo is used in lossless compression ought to be enough to destroy - in one stroke - the myth that JS "destroys stereo separation")" (cite from http://harmsy.freeuk.com/mostync/) (http://harmsy.freeuk.com/mostync/)) - I like this argument! It's quite cogent  But then I don't understand for what reason most CBR mp3s are made using S mode (even on LAME).
Because of the fallacy that S mode (or shall we say, LR encoding) is better than JS mode (or shall we say, MS encoding).

And for what reason all other Stereo modes are supported by LAME?
Because in cases where L & R channels are extremely different, MS encoding is not better than LR encoding. But because the S channel is usually allocated less bitdepth, it may cause degradation.

Quote
Acutally CoolEdit can, but it tells you absolutely nothing about how your brain interprets what your ears are hearing.
CoolEdit can subtact noise, but it's not the same. It uses the noise sample not "as is", but as a kind of "spectre", am I wrong? Or did I miss some function of CoolEdit, or Adobe Audition (for now)?
Here's how (IIRC - I don't have Audition on my office computer):
1. Open first wave.
2. Invert.
3. Select all.
4. Copy.
5. Open second wave.
6. Paste mix.
Title: -V n (in 3.95.1)
Post by: greynol on 30 November, 2006, 02:58:54 PM
Or did I miss some function of CoolEdit, or Adobe Audition (for now)?
You simply use the Mix Paste function (Ctrl+Shift+V), but again, it tells you nothing about what a lossy file sounds like.  Pepoluan already mentioned this, but you also need to make sure your files are synchronized since lossy files often introduce delay.

Quote
Of course they represent real "distortions"
But LAME 3.93 and GoGo didn't produce them... Is this "feature" (new to 3.95 and the following versions) just an improvement - i.e. to give more bits to the "more useful" freqs?
I'm not sure what you're seeing or what you're not seeing but it is utterly pointless to judge how something sounds based on what it looks like.

The more bits used trying to reproduce high frequencies which you will not be able to hear mean fewer bits that are available to reproduce what it is that you can hear.  Encode something with Blade @ 192 and the same thing with Lame @ 192, I don't care what versions you use; look at them and then listen to them.

...So, as I understood, the only way for me to get a better quality is to get a better soundsystem and to carry out the ABX test.
The ABX test in this case is supposed to help you determine whether you can actually tell the difference between the original source and a lossy version of it.  I think it's commonly accepted that a high-end sound system is not necessary to accomplish this.

By the way, can you suggest how to choose most appropiate sample for tests? I like metal music, and oftenly there are lot of live cymbals & hihats, which sound good, and have much high harmonies...
...Or it just may be any sample where I can hear distortions (or maybe just invent them ?
Just use the music that you ordinarily listen to.  Samples with lots of cymbals and hi-hats are a good way to reveal some of the more easily identifiable artifacts.

Do you have foobar2000 installed on your system with the ABX component?
Title: -V n (in 3.95.1)
Post by: ThyBzi on 01 December, 2006, 12:49:38 PM
Quote
Do you have foobar2000 installed on your system with the ABX component?
No, I downloaded foobar2000, but didn't find any ABX component for it (using search on its site). Is abchr I downloaded a little earlier worse for some reason?

By the way, using abchr, I couldn't hear those artifacts I see... It was only one test as now, but maybe the sample I used wasn't suitable. I'll try another samples and/or another sound system (and/or someone else's ears ).

I also tried to mix paste (thanx Pepoluan and Greynol for that idea ) samples using Adobe Audition, but it was too difficult to guess how great is the "lossy offset", mentioned by Greynol. Simple comparison of files' lenght did not help me. Is there any bright idea how to find out that lossy offset?

As I see, EAC can determine the offset generated by specific codec, but this operation returns an error for some reason (I tried to determine offset for GoGo and two different LAMEs). What can be a reason? I didn't find anything about it at EAC website's FAQ.

And some more questions about notorious spectral analysis. )

1) for two same-sounding (for me ) samples, is the one with "nicer" spectre (i.e. most bordering upon original) a better one (concerning quality)? Or it is no judge of quality even for same-sounding samples?

2) For two samples encoded using the same original, do same-looking graphs signify the same quality? (of course, accurate within my eyes )

And more questions... (It would be great grateful (edited ) if the LAME developer will answer )
1) Why qval=3 is now default (at least, when using -V2 --vbr-new)? As I remember it was qval=2 in older LAME versions...
2) What is --vbr-new?
3) Are there any differences between two spellings: -V 2 (readme file provided with LAME 3.97) and -V2 (found here, on hydrogenaudio.org)?
4) Please tell me, does -k really "destroy the quality of files" (like Firon said)? Almost any LAMEs mp3 graph remains same-looking when using this key, and the only difference is lowpass-filter-collapse dissapperance. Or that means nothing again?..

Also, M/S stereo will make things sound WORSE. Leave it as joint stereo (it is NOT lossy in LAME).
But isn't JS (when using at LAME, on high bitrates) mostly M/S stereo? Or your mean forced M/S, without using L/R Stereo when it's needed?
Title: -V n (in 3.95.1)
Post by: greynol on 01 December, 2006, 02:03:09 PM
I downloaded foobar2000, but didn't find any ABX component for it (using search on its site).
It is included as part of the installation.  I don't remember if it gets selected by default.

Is abchr I downloaded a little earlier worse for some reason?
From what I understand, this test is designed so that people could rank different files rather than determine whether they could tell the difference between two files.

By the way, using abchr, I couldn't hear those artifacts I see... It was only one test as now, but maybe the sample I used wasn't suitable.
It sounds like abchr was able to accomplish the same thing as abx from the point of view that you realized that you couldn't tell the difference.

I also tried to mix paste (thanx Pepoluan and Greynol for that idea ) samples using Adobe Audition, but it was too difficult to guess how great is the "lossy offset", mentioned by Greynol. Simple comparison of files' lenght did not help me. Is there any bright idea how to find out that lossy offset?
You need to switch to Multitrack View and load the two files one under the other, zoom in to a portion of the audio that you can easily distinguish.  Drag one of the files so they line up to the precise sample and then lock them.  Right click on one file and choose edit.  Then select the entire track.  Switch back to multitrack view again.  Right click on the other file and choose edit.  Copy what is currently selected.  Switch back to multitrack view.  Right click on the first file again.  Use the mix-paste function and check that you want to invert both channels.

As I see, EAC can determine the offset generated by specific codec, but this operation returns an error for some reason (I tried to determine offset for GoGo and two different LAMEs). What can be a reason?
It's a pretty outdated and IMO useless feature.  Anyway, I wouldn't bother with this at the moment.

1) for two same-sounding (for me ) samples, is the one with "nicer" spectre (i.e. most bordering upon original) a better one (concerning quality)? Or it is no judge of quality even for same-sounding samples?
What you hear is all that matters.  It is nice to see what lossy codecs throw out but please give up on this method of determining quality.

2) For two samples encoded using the same original, do same-looking graphs signify the same quality? (of course, accurate within my eyes )
Please give up on this method of determing quality.
Title: -V n (in 3.95.1)
Post by: ThyBzi on 02 December, 2006, 06:09:14 AM
Thanx a lot, you all just broke down my hasty and wrong conclusions.

But I still would be gratefull if someone answers my remaining questions.

Quote
1) Why qval=3 is now default (at least, when using -V2 --vbr-new)? As I remember it was qval=2 in older LAME versions...
2) What is --vbr-new?
3) Are there any differences between two spellings: -V 2 (readme file provided with LAME 3.97) and -V2 (found here, on hydrogenaudio.org)?
4) Please tell me, does -k really "destroy the quality of files" (like Firon said)? Almost any LAMEs mp3 graph remains same-looking when using this key, and the only difference is lowpass-filter-collapse dissapperance. Or that means nothing again?..
Title: -V n (in 3.95.1)
Post by: db1989 on 02 December, 2006, 06:23:34 AM
2) The new VBR method, faster and reputed several times to be of equal or better quality than the old.
3) No.
4) Well, you're choosing to keep additional frequencies which you probably cannot hear and which the MP3 format has some difficulties encoding. And yes, sight graphs mean very little in the context of audio!
Title: -V n (in 3.95.1)
Post by: robert on 02 December, 2006, 06:33:33 AM
1 - because it makes no difference for VBR NEW, note: LAME 3.98 qval will default to 0, but it still makes no difference!
2 - it's a different approach for VBR encoding, we started development some years ago, it will be the default VBR mode for LAME 3.98
3 - no, with single letter options taking parameters you can leave the blank out
4 - -k is a shortcut for setting high-/low-pass filters to minimum/maximum. LAME 3.98 will not have this switch anymore, but you can still change lowpass as before.
Title: -V n (in 3.95.1)
Post by: db1989 on 02 December, 2006, 06:36:27 AM
Wow, LAME 3.98 should be interesting!  Thanks for the information, robert!
Title: -V n (in 3.95.1)
Post by: ThyBzi on 02 December, 2006, 07:04:39 AM
Thanks Dv1989 and thanks Robert (an answer from LAMEs developer is a nice thing)!

2 - it's a different approach for VBR encoding, we started development some years ago, it will be the default VBR mode for LAME 3.98

But why an old algorythm is default for now? Is the new one still a "flaw design"?

3 - no, with single letter options taking parameters you can leave the blank out

What's about single letter? -V 2 and -V2 is what I'm interested about.  Dv1989 said that's all the same...
Title: -V n (in 3.95.1)
Post by: halb27 on 02 December, 2006, 07:49:22 AM
Before 3.97 development started --vbr-new wasn't widely used though it was available before (called 'fast' mode then).
With 3.97 (where I think has been work done with --vbr-new) it was found --vbr-new is of comparable quality to --vbr-old. Sometimes it's better, sometimes it's worse, and in the overall view there seems to be a tendency that is't slightly better. AFAIK we don't have a very solid basis to prefer one over the other qualitywise, but as --vbr-new is faster it is preferable to use --vbr-new.

Within the same version I think it's wise not to change the defaults, and you can use whatever you like by own settings.

3.98 will be a major step ahead, vbr behavior is significantly improving, and --vbr-new is default with 3.98.
Title: -V n (in 3.95.1)
Post by: db1989 on 02 December, 2006, 08:38:08 AM
But why an old algorythm is default for now? Is the new one still a "flaw design"?

The developers were obviously not confident enough, or did feel that it made sense, to have LAME default to the new mode. As we have just discovered, this has changed by now! Ooh, it's exciting stuff.

Quote

3 - no, with single letter options taking parameters you can leave the blank out

What's about single letter? -V 2 and -V2 is what I'm interested about.  Dv1989 said that's all the same...

Robert meant the same as I: you can leave out the space between the switch (in this case -V) and the parameter (i.e. 2) - and the end result will be the same.
Title: -V n (in 3.95.1)
Post by: ThyBzi on 02 December, 2006, 11:42:10 AM
Thanx for all! I'll try using -V2 --vbr-new, and I think it will be good for me.
Title: -V n (in 3.95.1)
Post by: ThyBzi on 04 December, 2006, 02:44:00 AM
Setting min/max bitrate is not needed in 3.97 when using VBR.
Spectral analysis is virtually useless for measuring sound quality.

Hmm... But when not setting minimum frame bitrate, LAME uses bitrate lower than 128 (on some songs - oftenly)... is it good?
Title: -V n (in 3.95.1)
Post by: greynol on 04 December, 2006, 02:49:02 AM
Hmm... But when not setting minimum frame bitrate, LAME uses bitrate lower than 128... is it good?

Between the reservoir and complexity of the frames to be encoded, less than 128 kbits can still provide transparency.
Title: -V n (in 3.95.1)
Post by: Firon on 04 December, 2006, 03:49:24 AM
Quote
You are partially wrong.

In low-bitrate, Joint Stereo is really Intensity Stereo. *That* will collapse stereo separation.

Not in LAME though. LAME doesn't have Intensity Stereo (unless it got it recently...), but FhG and probably others do.


Also, M/S stereo will make things sound WORSE. Leave it as joint stereo (it is NOT lossy in LAME).

But isn't JS (when using at LAME, on high bitrates) mostly M/S stereo? Or your mean forced M/S, without using L/R Stereo when it's needed?


I'm sorry. I meant forcing L/R stereo (-m s) for all frames.

Hmm... But when not setting minimum frame bitrate, LAME uses bitrate lower than 128 (on some songs - oftenly)... is it good?

Yes, because it determined that it didn't need 128 for that particular frame to reach the quality level you specified.
Title: -V n (in 3.95.1)
Post by: ThyBzi on 04 December, 2006, 11:12:10 AM
Thanks again  Now I see, LAME is even more clever than I thought a few days ago.
I like it, and I like this forum and its members who helped me so much.