From: [EMAIL PROTECTED] (Ken Arromdee)
Subject: [mp3encoder] lame problem
Date: Fri, 23 Sep 2005 10:04:21 -0700 (PDT)

> First, the ACM directory in the source distro is bad.  ACM is empty and acm
> (lower case) contains the contents, making the Linux compile not work until
> you rename it.

It is already fixed. It is repackaged/updaed 12 Sep 2005.

> Second, I've been having a problem decoding a particular file.  (It seems, 
> from
> searching the web, that someone else had the same problem, but they were using
> a freeformat file, which this is not).
> 
> % lame Track12.mp3 --decode -o Track12.wav
> ID3v2 found. Be aware that the ID3 tag is currently lost when transcoding.
> input:  Track12.mp3  (32 kHz, 1 channel, MPEG-1 Layer III)
> output: Track12.wav  (16 bit, Microsoft WAVE)
> skipping initial 1105 samples (encoder+decoder delay)
> big_values too large! 403
> Can't step back 197!
> bitstream problem: resyncing...
> big_values too large! 502
> Frame#     1/598    635 kbps         bitstream problem: resyncing...
> Error: number of channels has changed in MP3 file - not supported
> Error: sample frequency has changed in MP3 file - not supported
> Frame#     2/598     80 kbps         Error: number of channels has changed in 
> MP3 file - not supported
> 
> etc.  Despite appearances, the error isn't the big_values problem.  The error
> is that somehow Lame thinks the file is 32 kHz, 1 channel, when it's really
> 44 kHz, 2 channel (which mp3info confirms).

It may be because your mp3 file (Track12.mp3) is damaged.
The decoding part of LAME is not main function and poor to handle such files.

Anyway, it could be fixed. Can you send the file to me ? No need to send
whole of file, but 10kB from the beginning is sufficient I think.
-- 
Takehiro TOMINAGA // may the source be with you!
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