Hello,
I have been following this forum for some months now and finally having enough time decided to put my music collection into alt-preset standard as it seemed the easiest chioce. Most of my music went off without a hitch except for the very rare file that would come back as somewhere between 0 and say 150k after encoding. The question is though I was encoding a lot of Buckedhead mp3's a friend gave me on disc and LAME had this message while encoding them:

11.025 kHz VBR(q=2) single-ch MPEG-2.5 Layer III (ca. 7.4x) qval=2

I am encoding with LAME 2.90.2 and Razorlame 1.1.5 on an Athlon 1.33 Ghz with plenty of ram and other goodies. Even when i tried to decode to wav i got 0k file sizes. It's not a life or death thing really but it would be nice to be able to up their quality along with the rest, especially since Buckethead shreds so well.
Any advice is appreciated, even if it's just "hey you idiot switch this" or what have you. Thanks much for your time and advice. If more info is needed just let me know.

Viral

LAME 2.90.2? Maybe you mean 3.90.2?

Also, are you sure (in RazorLAME) you got the "Use only custom options" ticked under the "Expert" tab (in "LAME Options" under "Edit")? Sounds like you've enabled some messed up setting in RazorLAME; downsample to 11.025 kHz and mono setting and MPEG 2.5 Layer III enabled (shouldn't it be MPEG 1 Layer III)? Try looking over your switches again. Or it could be that rare bug in LAME that someone talked about which would make LAME have problems with certain files on certain computers, and the LAME binary having been compiled with a certain compiler(?!) or something like that. (But I thought that made LAME crash without encoding at all?) Try with 3.93.1 also, and see if it happens using that one too (in 3.93.1 the rare LAME bug is supposed to have been fixed).

Anyway, I would much rather recommend you try (instead of RazorLAME) madah's excellent multi-format-front-end "Frontah" (where only the quality options are available), which also tags your files for you (if you want). It's not so easy to mess up any setting in there, and it's much more userfriendly (in my opinion).

Thanks for the advice, I'll try frontah and see if thats works out. And yes I meant 3.90.2, and from what I understand I would prefer to stick with it rather than go to 3.93.1. I doublechecked all the settings and they are all where they should be, nothing changed between the rest of the files and the Buckethead ones except for the strange output. Thats why I was freaked, the output was so far from normal I had no idead what to address first. Again thanks for the help.

Viral

It seems to me like you've been transcoding... haven't you?

Anyway: encoding: CDDA-WAV-(MP3 or whatever) : lossless->lossy format

transcoding: in the language used on these forums is (mp3->mp3) or (mpc->mp3) or whatever... lossy->lossy

Do a search on transcoding first.

But to convince you, I think you're doing something equivalent to:

Open a bitmap in a photo editor. Save it as Jpeg 20% quality (photo_20.jpg). Then open it and save it as Jpeg Quality 80% (photo_80.jpg). Now, based on god-knows-what-logic, it seems like the 80 file should look better, but if you look more carefully at it it looks worse.

To resume, crap in -> crap out. even crappier

So then if I have been given an mp3 disc full of cbr 128 Buckethead mp3's because my friend thinks that's good quality should I just leave them at that or tell him to airmail me another in better quality from Japan? It seems there should be a happy medium. And if I were to go from what I have now to one of the semi-obscure expert lossless routes won't they be flawed because of the source also? By the way I figured out that the files were just strange-it's like he got them from a streaming source at 11 kHz mono. Go figure. Anyway thanks again for the help and suggestions.

11khz mono will sound pretty damn bad no matter what format you choose... ugh

Yes it definately sounds like it to me that you are transcoding your files... A common misconception is the belief that re-encoding files with higher quality settings creates better sounding files, but insted it makes files worse, and even more annoying is that the resulting files are bigger.

In audio coding, there is no way to make a file sound better by re-encoding. At all! With mp3s, whatever goes into the encoder comes out slightly worse in audible quality, so if the source is a 128kbps mp3, re-encoding to --alt-preset standard will result in a file around 129kbps*, but will probably sound like a 96kbps* mp3 file (ie worse).

My advice to you is keep the files as they are, or if your friend has the original uncompressed cds, get him to encode it in --alt-preset standard or even a lossless file format such as .ape or .flac and send them to you.

Hope this helps!

(*) estimations, not to be taken as definitive figures!