thanks, you & EmSiV cause work for moderators to split these 3 posts from the sticky topic to the discussion topic.
What have you thought replying here, when nobody replies here, and there are 2 big red links "discussion has been moved here and here" ?!

As Tag&Rename isn't freeware, it costs around 24 bucks, I am reluctant to add it (though I'd do it, when somebody comments knowledgeful on my following thoughts), as there are excellent free alternatives, Speeks Front end etc., and:
the more uncommented alternatives, the more the newbie gets irritated. This should be a straight forward List with best filtered settings/software. Eg., we don't mention or recommend MP3-Fraunhofer, which might (have been) be at certain settings competitive to comparable MP3-Lame, as Lame-Mp3 covers it.

Another good, powerful (and free!) tagging & renaming tool: The Godfather ( http://users.otenet.gr/~jtcliper/tgf/ )

hello,
when i read all the material, in the initial post, i got the impression that the insane preset - 320 kbps CBR matches quality with my current custom preset for Lame encoder: 3.97b2, VBR - OLD, V0, Q=0, kbps range: 192 - 320, but produces larger file.

ABX, ABX!! The two settings you mention are completely different.

That's not the case. One of them is a (quite braindead) VBR setting, the other CBR.
It's not recommended to limit the range of bitrates when using VBR.

to: audiomars
it was ABX test belive me. it was quite a test.
when i made the tests, i was listening to the files lot of times each, on various of players and speakers. i even made different combinations with different players and different speakers.
They are different yes but i'm interested of filesize and quality. for me it is not important is the algorithm CBR or VBR. Yes theoritically CBR 320 should produce the best quality that mp3 format can achive. But if you put it to practice conditions, 320 CBR makes some improovements that you cannot hear. if you cannot hear it than why it must be there to increase file size without hearable benefit?

to:
Dev0
First i made tests with wider range 112 - 320, but while i was monitoring the kbps i saw that only once kbps fall to 192, on one single file only, and on all the other files the kpbs NEVER fall under 224. this is the reason to repeat the tests with 192 as lowest value. usually i use 112 - 320 as range.

That doesn't sound like an ABX to me. More like comparison.

Edit: typo

actually Digital, i couldn't find the word, that why i wrote abx. my bad. You are right i compared them to see the quality loss and file size. it was quite a comparison. different players, different speakers and on each player i tested all speakers, so i can eliminate the differencies of players and speakers.

DJDiabolik wrote
>it was ABX test belive me. it was quite a test.

Without the ABX logs and links to the samples it sounds more like quite a bunch of BS.

A new version of LAME has been released. 3.97b3... is this the new recommended version?

EDIT: I actually wanted to post to this thread...

3.97b3? Are you sure? Where did you find that one?

Edit: quote problems:-)

On sourceforge.net, offcourse...

http://sourceforge.net/project/showfiles.php?group_id=290

psycho's right:

http://sourceforge.net/project/showfiles.p...lease_id=440772

Changes: Just a small fix over beta2 regarding a potential problem with some specific signals (like trumpet)

john33 :-] ?

EDIT: Paging john33 ;-]

Cool, so now it's up to beta 3 status. Maybe sometime this decade, ver. 3.97 will become an official release (i.e. lose the "beta").

I'll probably wait at least a month or so before upgrading. Or better yet, wait until Hydrogenaudio officially recommends 3.97b3 over 3.97b2.

Thanks for the heads-up, psycho.

Can I bring up a very old issue.

The recommended settings page makes a very bold claim...



I needed to demonstrate the quality of mp3 @ 96kbps last week.

I grabbed the edit of various material which I used years back for demonstrating ReplayGain. I edited it down to 1 minute, and resampled to 32kHz in Cool Edit Pro.

I encoded it using Lame 3.97b2 @ 96kbps CBR. It didn't sound very good (shock!). I tried a low pass filter @ 14.5kHz - it helped a bit.

I dug out an old FhG mp3 encoder for Cool Edit, and tried that. It sounded better. (ABX 8/8, though it wasn't really worth it at 96kbps!). I used the FhG file to demonstrate what mp3 could do at 96kbps.

I can upload the files, but I don't think the content matters that much. It's the bitrate, and using CBR. (IIRC FhG has the advantage of using intensity stereo, while lame doesn't?).


Just a small point. It makes us sound rather arrogant (and possibly incorrect, judging from what I found) to claim that Lame is better for everything always.

YMMV. I'd encourage someone else to try it. I'm sure there are plenty of releases of FhG's FastEnc out there to play with!

Cheers,
David.

Very good point: about two 1.5 years ago I was converting FLAC's (decompressing and recompressing, not transcoding) to LAME 3.90.3 --alt-preset cbr 96 for my Nokia mobile (a 6230 if I remember correctly) and on (quite) a few occasions files produced with MusicMatch Jukebox sounded slightly better (mainly alternative and metal), switched to EAC, WavPack and HE-AAC now, don't have those files anymore and can't therefore provide any ABX info, sorry.

Lame does not have any specific code to deal with low bitrates. So on 'mid low" (70-100kbps) it can sometimes sound good, but it is likely to fail on some samples. At very low bitrates, it will fail badly in most cases.
FhG encoders have some specific handling for low bitrates.

Yup, further proof of 2Bdecided's point's correctness.
Not a knock on LAME to any extent anyway: my (decently trained ears) simply love --preset fast extreme :]] !

I'm going slightly off topic here. When you said "about two 1.5 years ago", is that a fancy way of saying "about 3 years ago"?

Getting back on topic, I haven't tried experimenting with lower bitrate songs. I guess the conclusion is to use a LAME alternative MP3 encoder for those audio sources.

Yeah. Nice to see free softwares leaving out those "beta"s

Like for instance, Aoyumi recently produced "aoTuV Release 1". Granted, it is identical to "beta 4.51", with the sole difference being the vendor tag. But it's an improvement anyways

Hopefully with 3.98alpha will somehow graduates to 4.0RC ... or even full-fledged 4.0

4.0 is already being worked on.

4.0 if i remember correctly is a completely different encoder. It's a different branch with mostly rewritten code all over the place. Thus, it has nothing to do with 3.9x.

- Lyx

So 4.x will eventually be superior to 3.x?

For now, I'm just waiting for Hydrogenaudio to give the thumbs up to 3.97b3.

This is an honest (and definitive, "from the horse's mouth") answer.

Can someone with access update the "recommended settings" thread please?

Cheers,
David.

Guruboolez tried a lot of samples with various MP3 encoders at ~96 kbps here:
http://forum.hardware.fr/hardwarefr/VideoS...jet-84950-1.htm

LAME was clearly the best MP3 encoder and he chose it to be used in the test.


Personally, I have tested WMP10 FhG "CBR 48 kbps", Helix "CBR 48 kbps" and LAME "--preset 48" with a few samples. On average all were equally bad. Only this sample that has extreme stereo separation showed LAME to be clearly worse than the two others:



The used settings were these:

FhG (WMP10): "default 48 kbps CBR"
Helix MP3 (the latest binary from Rarewares): "default 48 kbps CBR" (If I recall correctly I couldn't make Helix to encode 48 kbps VBR)
LAME 3.97b2: --preset 48

The tested samples are from Gabriel's recent 48 kbps test.


The five other samples that I tried didn't show as clear differences:







I tested these because a web site needed to publish some audio files at about 48 kbps MP3. We ended up using LAME ABR in mono mode because the stereo files were simply too bad.


Edit: typo

The FhG codec was tested in CBR mode, but lame was tested in ABR mode.

It is an interesting test though. I don't think guru translated it for publication on HA, but it makes enough sense in the original French even if you don't speak it!

Your ABX results are interesting because some of the tracks where I heard the biggest difference were nearly mono.

This is probably a difficult topic to generate interest in here on HA, because none of us are encoding at 96kbps CBR!

Cheers,
David.

Not exactly. Two Fraunhofer encoders were tested (WMP and Adobe Audition's one). WMP was set to CBR 96 whereas Audition allowed VBR encodings.
I used two different settings for Fhg VBR (-q20 & -q30) because average bitrate isn't the same with "classical" music and "various" music.

It may be interesting to note that all VBR encodings were finally coded in stereo (and not "joint stereo"): it's likely to be a bug, because I explicitely check the option "joint stereo" in the GUI. I discovered it much later. In other words, Fhg VBR with a proper stereo coding mode may sound better than what I tested last year (and can therefore be closer to LAME or simply better than it).

Thanks guru,

I saw that bit, but haven't been keeping track with FhG codecs to know which one ships with Audition now.

The last one I have still has the "bad" HQ mode, but the "good" FastEnc. I assumed the WMP 96kbps CBR was nearly equivalent to "FastEnc", but could be wrong.


Interesting that they messed up joint stereo with VBR. At that bitrate, you'd think even intensity stereo would be an option.

Cheers,
David.

please confirm - there is NO ANY DIFFERENCE IN QUALITY between --vbr-new and non vbr-new settings:
lame -V 2 file.wav file.mp3
lame -V 2 --vbr-new file.wav file.mp3
produce same quality, but on different speed?

"In terms of quality, --vbr-new appears to be better than the old model ... --vbr-new is currently recommended over the default VBR mode due to both the speed and quality increases afforded by the new algorithm."
See wiki.

Edit: formatting.

Would someone please modify the quoted sentence to somehing "like 3.97 final-> .... " as 97 has gone final?

[NOTE: This is from the thread " Lame versions and alphas - READ THIS!" - the Pinned Thread

For V2 I would say:
V2 will likely be transparent to standard people with not standard equipment

Is there anywhere a short summary why these are recommended LAME settings?
Actually, there are several recommended settings.
I opted for this one:
-V 0 --vbr-new --id3v2-only --pad-id3v2 --ta "%a" --tt "%t" --tl "%g" --ty "%y" --tn "%n" %s %d
While there is a lot of explanation and description of quality settings, I don't see explanation why e.g. --pad-id3v2 parameter is recommended, or why other parameters
--ta "%a" --tt "%t" --tl "%g" --ty "%y" --tn "%n" %s %d
are included and some others are not (like --tg for genre).

The lame.exe "--pad-id3v2" switch is recommended because it avoids the whole file needing to be re-written during tag-updates later on(unless the update adds more than 128 bytes to the ID3v2 tag). The lame.exe tagging switches are recommended instead of EAC's own tagging implementation because some people have reported problems with EAC's tagging implementation. The lame.exe "--tg" switch isn't recommended because if the genre isn't one of 148 genres that lame.exe supports, then those files won't be encoded to MP3 and lame.exe shows an error message. I would personally recommend to use the "--tg" switch in addition to the "--ignore-tag-errors" switch, which will make lame.exe still encode files with un-supported genres and just set the genre tag to "Other".

Hmmm,
I planned to use some unsuported genres. Does that mean that if I put in EAC as a genre e.g. "HR Rock" that it will actually be converted into "Other" by Lame?
Is there any way to use custom genres in mp3 tags?
I know that for flac tags, genres are not predefined, I just need to figure out how to put them there.

Yes, if you use lame.exe's "--ignore-tag-errors" switch. If you don't, then the file(s) will not even be encoded and an error message displayed in the console.

Yes, but not with lame.exe's ID3v2 implementation. You can either use EAC's own tagging implementation which does support custom genres, or you could use the REACT EAC plugin, which uses metamp3.exe for MP3 tagging and which does support custom genres.

It seems that there's no point doing that. Why specifying custom genre, when it will be tagged as Other.
I can then specify it as Other in the first place.

How do I use EAC's tagging implementation? Does that mean that I cannot use Lame as an external encoder in that case?

No, you can still use lame.exe for encoding. Just remove the lame.exe tagging switches from your command-line and then enable "Add ID3 tag" under "Compression Options > External Compression". You can adjust the ID3 settings under "Compression Options > ID3 Tag".

Please read an EAC guide to get to learn the different settings of EAC.

SatCP's EAC Quickstart Manual :
http://users.pandora.be/satcp/eac-qs-en.htm#quickstart

When using a command line does it matter if there is a space or not between the -V and the number because on the Wiki's LAME page it references it both ways, e.g. -V4 --vbr-new and -V 4 --vbr-new?

Thanks,
but I'm using flacattack to convert to both flac and mp3, so it's a bit more complicated. I'm still trying to figure out how parameters are passed between EAC->flacattack->lame and EAC->flacattack->flac.
Any instructions/explanation about that?

I have no experience with Flacattack, but i have just checked and it uses lame.exe for tagging, so you're out of luck with that, i'm affraid.

Hi
Can any1 explain whats EAC command line option,
%l--alt-preset 128%l %h--alt-preset standard%h %s%d
Bit rate 192
High quality
Its totally different from the recomended settings

EAC allows you to toggle between two encoding settings for an external compressor: Low and High.

The settings between %l %l are the "low" and the %h %h the "high". They can be any two settings you choose. The radio buttons let you choose which settings will be passed to the LAME encoder.

--alt preset 128 and --alt preset standard are outdated LAME parameters.
--alt preset 128 = Average Bit Rate (ABR) of 128 = --abr 128 (in 3.95+)
--alt preset standard = generally accepted as transparent (equal to CD) = -V3 (in 3.95+)

See HA LAME Wiki, the top of this thread, or the quick start cited earlier in this thread for the current equivalents. There are a whole host of variations you can use.
Mine is: %l-V3 --vbr-new%l%h-V0 -q0 -b32 --vbr-new%h %s %d
Low Setting: -V3 --vbr-new
High Setting: -V0 -q0 -b32 --vbr-new

Note: I do not recommend using my settings, I just place them here as an example.

With these, and most other command line options in the external compressor screen, the bit rate pull down is ignored.

%s and %d are EAC variables for the "source" (wav) and "destination" (mp3) paths of the encoded file. They complete the LAME command line.

All of this can be confusing, since different sources each have their own set of recommended and best settings, some more current than others.

I found a list of Lame supported genres,
but number 42 is missing.
Do you know what is it?

I found it. It's 42 - Soul

It's : Soul

I have just tested it out and lame.exe does support the "Soul" genre also, so it is just a mistake in the LAME documentation.

Edit: Not fast enough

HI GUYS
Great threat, old though haha.

Until now I was encoding with nero, at 128 kbps.
But that's not good enough, so I checked LAME.

Everyone say that using -V2 its good enough. -V2 it's between 170 kbps and 210 kbps.
I want stereo, NOT Joint stereo. And the ability to encode FLAC to LAME.
I run some tests using LameXp.

With Quality 2 (V2) in Winamp says 271 kbps (VBR). Isn't it supposed to be between 170 and 210? 271 is way up.

So I tried restricting the bitrate. I want 192 VBR, so I used 2 above and 2 bellow. I mean, encode between min: 160 and max: 224 kbps. This got a song with 223kbps (VBR). In the range.

Am I doing something wrong? I think the quality bar in LameXP is wrong.
What do you think about encoding between 160 and 224?

best luck, LUCAS

You need to go back and read through this thread. You also need to search the forums and read through the documentation located here at the Lame wiki page. In short, joint stereo is a lossless process that actually allows the encoder to perform better. Many other lossy codecs (I believe Nero AAC and even lossless FLAC) use processes similar to this (or maybe even the same procedure, I am not an expert). Additionally, you don't want to restrict Lame when it comes to bitrates. Pick a proper -V value and go with it. -V 2 is often recommended but many people now feel that -V 3 is a better choice it results in overall lower bitrates while having transparent (ie "CD quality") results.

The best thing that you can do is download foobar2000 and conduct some blind ABX tests to determine what bitrate setting is right for you. A blind ABX test will force you to ignore the bitrates of the files and hear the quality. That is what you really want to do. There is nothing out there that states files encoded between 160 and 224 kbps have the best quality (any place/person that actually says this should be ignored as everyone's preferred quality setting is a little different). foobar2000 can also convert your FLAC track to mp3 files using Lame while preserving the track tag information.

KORNCHILD

I know what's joint stereo, and that it allows better bitrates, I just don't like the idea of not having two clearly separated channels. I'd prefer to use simple stereo and end up with songs bigger in size, I don't care about the difference in size.

As you said I'll conduct some ABX blind tests.
I knew about the lame information here. There's a list that says that lame -V2 is between 170 and 210 kbps, but I ended up with a 270 kbps song.
So, if I'm right, the information in that table is wrong, 270 is way more than 210 kbps.
Is there something I don't know? Or it is supposed to be out of the 170-210kbps range?

Changing the subject, but still with mp3.
FLAC to LAME V4 is the same as FLAC to LAME V2 to LAME V4 ?
LAME 320 CBR to 192 CBR is the same as LAME 320 CBR to LAME 256 CBR to LAME 192 CBR ?

Thanks, LUCAS

If you're encoding rock or metal, you're bitrates will most likely be higher. See this.