lame 3.29 is out.
It has most of Leonid's fixes. In addition to the bug in the count1
values, the bit counting routins in loop.c were overestimating
the number of bits needed to encode a frame. This wouldn't be so
bad if the unused bits were put in the reservoir, but because of
how the code is structured they were basically wasted.
some more modularization work was done, and all the hooks are in place
for resampling/downsampling, although there is no actual resampling code.
This work also made it easy to add the following feature: LAME can
now take a .mp3 file as input. i.e.
lame input.mp3 output.mp3
will decode input.mp3 (with mpglib/mpg123) and then re-encode it.
Kind of silly right now, but if we get some resampling code added,
it will allow low bitrate internet broadcasting directly from
your mp3 collection.
Mark
>From the web page:
Bug in bigv_bitcount fixed. Loop.c was overestimating the
number of bits needed, resulting in wasted bits every frame.
(Leonid A. Kulakov)
Bug in *_choose_table() fixed These routines would not sellect
the optimal Huffman table in some cases. (Leonid A. Kulakov)
Tuning of ATH normalization (macik)
Removed unused variables and fixed function prototypes (Sigbj�rn
Skj�ret)
Sami Farin sent a .wav file that LAME built in support choked
on. I added a slightly more sophisticated wav header parsing to
handle this, but if you have trouble, use libsndfile.
Resampling hooks and options added. Buffering and resampling
routines need to be written.
LAME will now take an mp3 file as input. When resampling code
is working, LAME will be able to (for example) convert a high
bitrate stereo mp3 to a low bitrate mono mp3 for streaming.
Bug in bigv_bitcount fixed. Loop.c was overestimating the number of bits
needed, resulting in wasted bits every
frame. (Leonid A. Kulakov)
Bug in *_choose_table() fixed These routines would not sellect the optimal
Huffman table in some cases.
(Leonid A. Kulakov)
Tuning of ATH normalization (macik)
Removed unused variables and fixed function prototypes (Sigbj�rn Skj�ret)
Sami Farin sent a .wav file that LAME built in support choked on. I added a
slightly more sophisticated wav
header parsing to handle this, but if you have trouble, use libsndfile.
Resampling hooks and options added. Buffering and resampling routines need to
be written.
LAME will now take an mp3 file as input. When resampling code is working, LAME
will be able to (for example)
convert a high bitrate stereo mp3 to a low bitrate mono mp3 for streaming.
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