History

Fast mode (-f) disables psycho-acoustic model for real time encoding on older machines.
New bit reservoir usage scheme to accommodate the new pre-echo detection formulas.
Tuning of AWS and ENER_AWS pre-echo formulas by Gabriel Bouvigne and Mark Taylor. They work great! Now on by default.
In jstereo, force blocktypes for left & right channels to be identical. FhG does this, so we will too.
Patches to compile MP3x under win32 (Thanks to Albert Faber).
bin_serach_stepsize limited to -210 through 45.
Mid/side quantization now varies scalefactors independently. Can lead to better quantizations, but it is slower (twice as many quantizations to look at). Running with "-m f" does not need this and will at the old speed

Preliminary documentation from Gabriel Bouvigne.
Even more speed improvements from Chris Matrakidis! Removed one FFT when using joint stereo, and many improvements in loop.c.
"Fake" ms_stereo mode renamed "Force" ms_stereo since it forces mid/side stereo on all frames. For some music this is said to be a problem, but for most music mode is probably better than the default jstereo because it uses specialized mid/side channel masking thresholds.
Small bugs in Force ms_stereo mode fixed.
Compaq Alpha fixes from Nathan Slingerland.
Some new experimental pre-echo detection formulas in l3psy.c (#ifdef AWS and #ifdef ENER_AWS, both off by default. Thanks to Gabriel Bouvigne and Andre Osterhues)
Several bugs in the syncing of data displayed by mp3x (the frame analyzer) were fixed.
Highq (-h) option added. This turns on things (just one so far) that should sound better but slow down LAME.

3.03

Faster (20%) & cleaner FFT (Thanks to Chris Matrakidis http://www.geocities.com/ResearchTriangle/8869/fft_summary.html)
Mods so it works with VC++ (Thanks to Gabriel Bouvigne, www.mp3tech.org)
MP3s marked "original" by default (Thanks to Gabriel Bouvigne, www.mp3tech.org)
Can now be compiled into a BladeEnc compatible .DLL (Thanks to Albert Faber, CDex author)
Patches for "silent mode" and stdin/stdout (Thanks to Lars Magne Ingebrigtsen)
Fixed rare bug: if a long_block is sandwiched between two short_blocks, it must be changed to a short_block, but the short_block ratios have not been computed in l3psy.c. Now always compute short_block ratios just in case.
Fixed bug with initial quantize step size when many coefficients are zero. (Thanks to Martin Weghofer).
Bug fixed in MP3x display of audible distortion.
Iimproved status display (Thanks to Lauri Ahonen).

Encoder could use ms_stereo even if channel 0 and 1 block types were different. (Thanks to Jan Rafaj)
Added -k option to disable the 16kHz cutoff at 128kbs. This cutoff is never used at higher bitrates. (Thanks to Jan Rafaj)
Modified pe bit allocation formula to make sense at bit rates other than 128kbs.
Fixed l3_xmin initialization problem which showed up under FreeBSD. (Thanks to Warren Toomey)

3.01

max_name_size increased to 300 (Thanks to Mike Oliphant)
Patch to allow seeks on input file (Thanks to Scott Manley)
Fixes for mono modes (Thanks to everyone who pointed this out)
Overflow in calc_noise2 fixed
Bit reservoir overflow when encoding lots of frames with all zeros (Thanks to Jani Frilander)

3.0

2.1f

2.1e

New command line switch -a auto-resamples a stereo input file to mono.
New command line switch -r resamples from 44.1khz to 32khz [this switch doesn't work really well. Very tinny sounding output files. Has to do with the way I do the resampling probably]
Both of these were put into the ISO code in the encode.c file, and are simply different ways of filling the input buffers from a file.

2.1d

Fixed memory alloc in musicin.c (for l3_sb_sample)
Added new command line switch (-x) to force swapping of byte order
Cleaned up memory routines in l3psy.c. All the mem_alloc() and free() routines where changed so that it was only done once and not every single time the routine was called.
Added a compile time switch -DTIMER that includes all timing info. It's a switch for the time being until some other people have tested on their system. Timing code has a tendency to do different things on different platforms.

2.1b

Fixed up bug: all PCM files were being read as WAV.
Played with the mem_alloc routine to fix crash under amigaos (just allocating twice as much memory as needed). Might see if we can totally do without this routine. Individual malloc()s where they are needed instead
Put Jan Peman's quality switch back in. This reduces quality via the '-q ' switch. Fun speedup which is mostly harmless if you're not concerned with quality.
Compiling with amiga-gcc works fine

2.1a

Version 2.1a released. User input/output has been cleaned up a bit. WAV file reading is there in a very rudimentary sense ie the program will recognize the header and skip it, but not read it. The WAV file is assumed to be 16bit stereo 44.1khz.

2.1

All tables now incorporated into the exe. Thanks to Lars Magne Ingebrigtseni(larsi@ifi.uio.no)

2.0

LAME is built from the ISO source code (dist10), and incorporates modifications from Mike Cheng and the 8Hz effort. The output file from LAME v2.0 is identical to the output of the ISO encoder.

1.0