> Those MP3s were created from 16 bit data, but once that data has been
> transformed to the frequency domain as part of the encoding process, that
> bit depth ceases to matter. In mpd's case, the MP3s are decoded as 28 bits
> by libmad, and truncated to 24 bits. This results in theoretically lower
> distortion than truncating or dithering that output to 16 bits. In your
> case, by forcing 16 bit output with MP3 you are getting more distortion.
That reminds me: when the signal needs to be resampled because the
output's sample rate doesn't match the file's sample rate (typical case
for me: 44.1KHz CD input and an audio card that only supports 48KHz),
would it be possible to do the resampling directly in the frequency
domain as part of the MP3/Ogg decoding? It seems like it would be
cheaper (CPU wise) and result in higher quality than using a separate
resampling phase after decoding.
Stefan
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