I have a sony NW-E107. There is a certain album on my computer and all music from it I try to play on my mp3 player says "CANNOT PLAY".

It plays fine on my computer, I am guessing it got a little corrupted. Not enough to stop foobar2000 from playing it, but enough to stop my mp3 player.

It was zipped into a zip file, then unzipped, which might have something to do with it.

(Also, I wasnt sure if I was supposed to put this in this section or in the MP3 section. If this is the wrong section, I am sorry)

Thanks alot to anyone who can help, or try. (By the way, the album is Tommy by The Who. Great album)

The zipped file is an MP3 file, right?
If it's not then that explains.

Not being offensive, it's just you didn't state the audio format clearly. =)

If it is an MP3, if there's any info about its bitrate and sample rate then it might help us rectify the problem.

It could very well be the tags in those file that the mp3 player doesn't understand. You can remove all tags in those mp3's using Tag Frontend w/Tag.

It doesn't play 48khz file nor MP2 renamed to MP3

That's too bad...


Are there other players that able to play renamed MP2 files?
I haven't tested mine (a Creative Jukebox Zen Xtra), but I'm just curious.

Sorry It is MP3

This is what foobar2000 says:

bitrate = 256
codec = MP3
channels = 2
samplerate = 48000
mp3_stereo_mode = joint stereo
----------
8698223 samples @ 48000Hz
File size: 5 800 358 bytes



That might be it... I checked other music that works on it and they are 44100.

Is there any way to change 48000 to 44100?

Directly changing it (w/o transcoding) is probably not possible.
But, what do I know...

Other, more enlightened people might know if there exists a tool for directly changing a sample rate. =)

It's not possible to do it right, but what you could do is simply change the samplerate that the MP3 reports to 44.1. This will make the song a bit lower (tone-wise) and last a bit longer, but it will be playable and completely reversible.

Doing some quick tests with Foobar's "Change samplerate" component results in most songs sounding a bit weird, but if I hadn't heard the originals I'm sure I'd think they were normal. If your song is a bit slow to begin with, it might be unbearable. However, it actually seems to improve some of my fast techno.

Other than that, you'd have to reëncode to get it to 44.1.

[Edit]
@xequence:
Out of curiosity, why is it 48000 to begin with? Was it a recording from vinyl or something? Or was the encoding "out of your hands" so to speak ?

Well, that'd make the song sounds a bit 'slower', wouldn't it? =)
AFAIK, the only way to 'properly' convert the sample rate is to transcode it.

scan the files with Mr Question man, see what mp3 codec was used

http://www.burrrn.net/?page_id=5

I have no idea why they were encoded like that. I downloaded them off bittorent

So, I just use razorLAME to encode it again, and it will be the right sample rate?



MPEG 1 Layer III LAME 3.97

was the source an DVD? Sometimes I have seen released at 48 done that way...

I dont know, I didnt rip it.

Any frontend is fine, as long as you're using the --resample switch (i.e --resample 44.1)

@bubka
Is using the 'wrong' MP3 encoder can break playability w/ MP3 players?

I don't know, I know iPods have problem with LAME -aps files, maybe it is something similar, there are some really crappy mp3 encoders out there...

Ok =)

I never accually found out how to use the command line LAME....

Wow. razorLAME doesent have an option to change the output folder, so you cant really transcode MP3 files. Odd. (As in it outputs the encoded file to the same folder. You can have a Track1.wav as the original and Track 1.mp3 as the encoded one, but you are out of luck if the original is Track 1.mp3 Heh, ill have to find another frontend)

I'd be careful with that sentence if I were you...
It seems you indicate that iPod's inability to properly decode LAME's aps files is LAME's fault for being a 'crappy mp3 encoder'.

But of course, pardon my misinterpretation if you didn't mean it that way. =)

Try winLAME and replace its nLAME.dll w/ the one from Rarewares.org to use LAME 3.97b2.

I'd decode the file to WAV, use SSRC to downsample it to 44.1khz and (sadly) re-encode to MP3...

Yikes, just realized current winLAME can't be set to resample things up.
Multi frontend then.

Isnt lame the best MP3 encoder?



Sadly that looses the ID3 tags. I might have to do it though...

Use FoobarTwoThousands to resample and convert your files on-the-fly. It will also preserve ID3 tags.

Of course.
But bubka seemed to think otherwise or maybe it's just me imagining things... =)

yeah, didn't mean it that way, LAME is not the "crappy mp3 encoder"

Certain sony players would not play mp3 files if the original bit is not set for each frame...

There are way more crappy "mp3"-players out there which are strictly spoken not mp3-compatible, than there are "crappy encoders".

- Lyx

Yech...
Seems like they're trying to impose some kind of a 'copy-protection' even after being ripped from the Audio CD.
(though, if I'm not mistaken, LAME by default sets the original bit, right?)

Looks like a good program, thanks =)



I really dont know what that means, except the sony players wont play certain mp3 files

In what insistance would the original bit not be set for each frame?

EDIT: Transcoded it with multi frontend, it worked and I have it on my mp3 player now. (Though I can hear the difference between the transcoded and non transcoded file, its alright )