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- 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
-