I think I've found a minor bug in lame.
I was trying to encode some files at very low bitrate and I found this bug :
the raw pcm part can't encode mono files at low bitrates
the encoded files is in mono but the data are doubled (or something like tha,
resulting a file of the double size), so when it's played back, it like the sound
played slowly (very funny, indeed).
I was using some .au files which were taken for raw pcm files (which works fine with
j-stereo at higher bitrates). So I converted them to wav and the same parameters (lame
-m m --resample 16 -b 16) worked fine, the resulting file was what I expected. So
there might be a bug on handling raw pcm stereo.
also when I was doing some tests, I noticed this :
-------------------------------
# lame -m j --resample 16 -b 16 fbs.au fbs10.mp3
Assuming raw pcm input file
LAME version 3.85 (www.sulaco.org/mp3)
Resampling: input=44.1kHz output=16.0kHz
Using polyphase lowpass filter, transition band: 581 Hz - 774 Hz
Encoding fbs.au to fbs10.mp3
Encoding as 16.0 kHz 16 kbps j-stereo MPEG2 LayerIII (32.0x) qval=5
-------------------------------
while this one was better
-------------------------------
# lame -m m --resample 16 -B 16 -V 9 --no-hist fbs.au fbs8.mp3
lame: unrec option --no-hist
Assuming raw pcm input file
LAME version 3.85 (www.sulaco.org/mp3)
Resampling: input=44.1kHz output=16.0kHz
Using polyphase lowpass filter, transition band: 4452 Hz - 4645 Hz
Encoding fbs.au to fbs8.mp3
Encoding as 16.0 kHz VBR(q=9) single-ch MPEG2 LayerIII (14.0x estimated) qval=2
-------------------------------
Look at the lowpass filter of the first one !!!
I've tested the raw pcm problem both on Linux and Win32.
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