> Hi,
>
> I got some suggestions for LAME. They're not too complicated (probably are
> to implement, though), so here goes:
>
> - LAME VBR doesn't encode the LSB (Least Significant Bit) correctly (as
> described on http://privatewww.essex.ac.uk/~djmrob/mp3decoders/lsb.html) -
> what exactly is causing this and how could it be fixed?
Youri is refering to the fact that LAME VBR does not encode the
LSB correctly. This test is a 3.1khz sine wave, but when quantized
down to a 16bit number was just +/-1. This is right at the
ATH, so I'm sure VBR is very tempted to remove/alter the signal.
> - A good suggestion, IMHO: freeformat VBR encoding up to 640 kbs - this
> feature could be implemented through the -B switch (with additional values
> of 384/448/512/640), which would default to 320, but could be set higher to
> give a freeformat VBR. This would have the advantage that some parts which
> are too difficult to compress even with 320 kbs could use even higher
> bitrates which would improve the quality of those frames. I'm sure not many
> people would mind a freeformat VBR MP3 if it would mean a nice improvement
> in quality without having the file bloated like with freeformat CBR.
This is an easy one :-)
Freeformat is supported up to 550kbs. use "--freeformat -b X" where X
can be any number between 8 and 550. I will make the change to allow
bitrates up to 640. Only one problem: I know of no decoder which will
play a 640kbs free format stream! Only one decoder will handle 550kbs
free format (LAME's modified version of mpg123) and the vast majority
of players will not decode free format at all.
> - An engine which would analyze the source data and find out which mode
> for -X would be best to use for compression (I have to admit though, that I
> am not too familiar with the -Xx settings - if someone could please explain
> these modes (or point me to a document which explains them), I'd be very
> grateful)
>
Unfortunately, I think the only way to tell which is better is with
listening tests. The different -X specify different ways to measure
"quality" of a quantization, and thus determines how LAME picks the
best quantization amoung the many possibilities. The analysis program
you describe would also have to have some way to measure "quality".
It is tempting to just use the encoded - original RMS difference
as the quality measure. This probably works at very high bitrates,
but at 128kbs, it ignores the psycho acoustic masking which
MP3 is based on, and is not the right way to measure quality.
I claim that this problem, and better psycho acoustics,
are the two big unanswered questions in audio compression!
Mark
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