At 11:26 2002-11-28 -0800, Shel Ritter wrote:
> I'm, assuming that some bit of audio quality is going to be lost, and
> at least a small bit of transient distortion is going to be introduced
> when resampling one mp3 to another. Or are you not referrring (...)
The discussion was about stereo wav files with speech and which output
settings to use (number of channels, frequency etc). Resampling one mp3 to
another is the same as doing
lame --decode prim.mp3 prim.wav
lame prim.wav bis.mp3
I'm guessing, in your case, the mp3 encoding of the wav file to 320 kbps
adds much more distortion and quality loss than ever the 320 kpbs -> 128(?)
kbps re-encoding, because the sound has already been put through the
psycho-acoustic filters etc. Also, since 320 kbps in most audiophile
listening tests is considered indistinguishable (right?) from the original
recording, and these tests do involve the decoding process (step 1 above),
I don't see how step 2 could worsen things. (Then again, going from
uncompressed audio directly to 128 kbps would be better because there's
then more original information from which to shape the sound into 128 kbps.
But that's not what you asked, I guess.)
Also, resampling mp3 files with lame is very easy; you don't need Easy
CD-DA extractor. Just
lame <settings> *.mp3
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