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Diffstat (limited to 'lib/liblame/TODO')
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diff --git a/lib/liblame/TODO b/lib/liblame/TODO new file mode 100644 index 0000000000..33d95751e9 --- /dev/null +++ b/lib/liblame/TODO @@ -0,0 +1,162 @@ +1. bug in resample code: downsampling from 44101 to 44100 causes + a seg fault. Workaround in place for now: resampling disabled + if input/output samplerates agree to 4 digits. + + + +2. high bitrate encodings have trouble on some hardware players. +Track this down. Probably caused by --strictly-enforce-ISO and +IXMAX_VAL. Try setting IXMAX_VAL back to 8191 and/or +maxmp3buf=8*960 to see if there is a working combination. + +note: one of the decoder bugs was identified. It is caused by using +different block sizes on both channels. A parameter need to be +added to Lame to handle workarounds. + + +3 frontend: code is a complete mess. But it has so many debugged + features it will be a lot of work to re-write. + + +4. MSVC project files. It would be nice to create a working + MSVC6 workspace, which included all the projects as possible + targets: + lame.exe + mp3x.exe (require GTK libs) + lame_enc.dll + ACM codec + directshow codec + + I think the only MSVC5 project that we need to preserve is + for lame_enc.dll, since Albert Faber (still?) doesn't use VC6? + But no reason we cant have VC5 and VC6 project files for the dll. + + + + + + +5. +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. + + + +6. +Does stdin work when LAME is compiled to use libsndfile? +(new version of libsndfile will support this - try this out) + + +7. +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. + + +8. +mgplib has bugs with i-stereo. flag denoting invalid +i-stereo value (= frame is m/s stereo) is not correct. + +9. +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. + +10. +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. + +11. +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. + +12. +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. + + + + +50. +Better tonality estimation. +Gpsycho uses predictability, and so needs a delay to detect the tonality +of a sound. +Nspsytune seems to miss tonals when several of them are too narrow. +We would probably need the best of both. + + + + +60. +Different ATH handling for sfb21. We are using the minimum value of ath +in each whole sfb. in sfb21 this leads to very high bitrates. +We could perhaps use 2 or 3 ath partitions in sfb21 + +note: partially done + + + +70. +Use mixed blocks. + + + + +90. +Use intensity stereo. This is a must-have for low bitrates, but if the +algorythm is very good it could also be used in every case. +Note: mpg123 (and all derivatives, like xmms and lame/mpglib) +have bugs in the intensity stereo decoding. Bugs have been there +for years since there are very few intensity stereo mp3's out there. + + + + + +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 -> wave + ==> copy TAG into LIST chunk + + + +1500. +Integrate plusV extensions + + + +2000. +To be able to encode as fast as FastEnc
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