frontend: a bunch of static variables (input_format, swapbytes,
          silent, brhist, update_inteval) are set in parse_args and used in
          get_audio.  should be cleaned up.




NOGAP encoding:

-nogap:  more testing, fix options, test id3 tags?
Can we change id3 tags without reseting the encoder??
At the end of encoding 1.wav, call lame_get_mf_samples_to_encode()
to find the number of non encoded buffered PCM samples.  Then
encode samples from 2.wav until these PCM samples have been
encoded, *THEN* call lame_encode_flush_nogap() and close
out file 1.mp3.


NOGAP decoding:  
lame --decode --nogap file1.mp3 file2.mp3 file3.mp3
should also work.  What needs to be done:
get_audio.c:  We need a way to open a second mp3 file, without
              calling lame_decode_init() and reinitializing mpglib.
              And the mpglib needs to know to look for new Xing
              tags at the beginning of file2.mp3 and file3.mp3.



1.
Does stdin work when LAME is compiled to use libsndfile? 
(new version of libsndfile will support this - try this out)


2. 
LAME has problems with pure DC input.  i.e. a square wave with
a frequency well below 20 Hz.  Not very important, but it should
be fixed.


3. 
modify mpglib to output
floating point and have finaly quantization step a
easy-to-change module so it can output other than 16bit.
(replace mpglib with MAD?  MAD has agreed to write a call back
which will return all data needed by the frame analyzer)

3.1  mgplib has bugs with i-stereo.  flag denoting invalid
i-stereo value (= frame is m/s stereo) is not correct.  

4.
Clipping detection.  Compute statistics for input near 2^15, 
(should be done in fill_buffer() in util.c).  Have command
line encoder print statistics about % of samples which might
end up clipped.

Better: do a full decode - but this seems too expensive. 


5.
lowpass filter: for M/S stereo, use more filtering for the side
channel, less filtering for mid channel.  We need to first replace
the polyphase filter with an FIR lowpass filter with finer frequency
resolution before implementing this. 

6. 
LAME has a 31 point FIR filter used for resampling, which
can also be used as a lowpass.  When resampling is done, 
use that filter to also lowpass instead of the polyphase filter.

7.
Even when resampling is not needed, should we use an FIR filter
for the lowpass?  If it is not too much slower, yes.  If it
is slower, then it should be an option since it will produce
higher quality.  

8.
We should consider moving the experts options from the *long
help* text into an *experts only* help text. The average Joe gets
knocked down by the huge number of possibilities to setup lame.
 





997..
ACM wrapper for LAME.  Note: ACM is obsolete and was replaced
with DirectShow and/or a "DMO".  We do have a beta directshow filter.

Contact "Florian Bruckner" <h9651030@miraculix.wu-wien.ac.at> for info - 
he is working on the VorbisACM.



998.
Merge GOGO's fast assembler routines.




999.
It would be nice to save some information whilst encoding
a: wave -> mp3
	a RIFF/wave can contain LIST chunks with information
	about author, title, etc. 
	==> could go into TAG fields of resulting mp3
b: mp3 -> mp3
	==> we could copy the TAG directly
c: mp3 <-> ogg
	==> copy TAG info into oggs info field and vice versa
d: mp3 -> wave
	==> copy TAG into LIST chunk
