History
Starting with LAME 3.0:
red = features and bug fixes which effect quality
blue = features and bug fixes which effect speed
black = usability, portability, other
LAME 3.60beta (not yet released)
LAME 3.59beta January 4 1999
-
Takehiro Tominaga: --noath option. Disables ATH maskings.
-
Gabriel Bouvigne: updated HTML docs.
-
Iwasa Kazmi: makefile fixes
-
Mark Taylor: Fixed bug where first frame of data was always overwritten
with 0's. Thanks to 'gol'
-
Mark Taylor: bug fixes in mid/side masking
ratios (thanks to Menno Bakker)
-
Mark Taylor: replaced norm_l, norm_s table data with formulas.
LAME 3.58beta December 13 1999
-
Segher Boessenkool: More accurate quantization
procedure! Enabled with -h.
-
Mathew Hendry, Acy Stapp and Takehiro Tominaga: ASM
optimizations for quantize_xrpow and quantize_xrpow_ISO.
-
Chuck Zenkus: "encoder inside" logo on web page
-
Mark Taylor: a couple people have asked for this.
Allow LAME to overide VBR_min_bitrate if analog_silence detected.
Analog_silence defined a la Robert: energy < ATH.
-
An Van Lam: Valid bitrates were being printed for layer 2, not layer 3!
-
Ethan Yeo: Makefile.MSVC updated
-
Mark Stephens: updated all MSVC project files
-
Robert Hegemann: lowpass and highpass filters can be enabled with
--lowpass, --highpass
-
Mark Taylor: MS switching is now smoother:
ms_ratio average over 4 granules
-
Takehiro Tominaga: Scalefactor pre-emphasis
fixed (and now turned back on)
-
Takehiro Tominaga: Bug in M/S maskings:
switch to turn on stereo demasking code was buggy.
LAME 3.57beta November 22 1999
-
Sigbjxrn Skjfret: patch to allow encoding from 8bit input files when using
LIBSNDFILE
-
Automatic downsampling to nearest valid samplerate.
-
Scalefactor bands demarked on MDCT plot in frameanalyzer
-
Scalefactor preemphasis disabled for now. The algorithm was
often doing more harm than good.
LAME 3.56beta November 19 1999
-
Kimmo Mustonen: portabilty code cleanup.
-
Vladimir Marek: id3 genre patch.
-
Conrad Sanderson: new applypatch script.
-
Initial window type now "STOP_TYPE" to reduce initial attenuation.
This is needed because the new encoder delay is so short. With a
NORM_TYPE, the first 240 samples would be attenuated.
-
Padding at end of file now adjusted (hopefully!) to produce as little padding
as possible while still guarantee all input samples are encoded.
-
Takehiro Tominaga: Reduced shortblock extra
bit allocation formulas by 10% since new huffman coding is at least 10%
more efficient.
LAME 3.55beta November 11 1999
-
Albert Faber: updated BladeEnc.dll
-
Simple lowpass filter added to linear downsampling routine.
-
Nils Faerber: updated man page.
-
All floating point variables are delcared FLOAT or FLOAT8.
Change the definition of FLOAT8 in machine.h to run at 32bit preceision.
-
Bug (introduced in 3.54beta) in stereo->mono downsampling fixed.
LAME 3.54beta November 8 1999
-
Encoder delay is now 48 samples. Can be adjusted to 1160 to sync
with FhG (see ENCDELAY in encoder.h) This is kind of amazing, since
if Takehiro put his MDCT/filterbank routine in a decoder, we could have
a total delay of only 96 samples.
-
More inconstancies found and fixed in MPEG2 tables.
-
Resampling from an MP3 input file now works. But we still dont have
a lowpass filter so dont expect good results.
LAME 3.53beta November 8 1999
-
Takehiro Tominaga: Fixed MPEG2 problem in new
MDCT routines. Takehiro's combined filterbank/MDCT routine is now
the default. Removes all buffering from psymodel.c and the filterbanks/MDCT
routines.
LAME 3.52beta November 8 1999
-
By permission of copyright holders of all GPL code in LAME, all GPL
code is now released under a modified version of the LGPL (see the README
file)
-
By popular demand, all C++ comments changed to C style comments
-
Linear resampling now works. Use --resample to set an output samplerate
different from the input samplerate. (doesn't seem to work with mp3
input files, and there is no lowpass filter, so dont expect good results
just yet)
-
Takehiro Tominaga: Faster Huffman encoding
routines
The following changes are disabled because of MPEG2
problems. But to try them, set MDCTDELAY=48 in encoder.h, instead
of MDCTDELAY=528.:
-
Takehiro Tominaga: New MDCT routines with shorter
delay (48 samples instead of 528) and even faster than the old routines.
-
Takehiro Tominaga: Removed extra buffering
in psymodel.c
LAME 3.51 November 7 1999
-
Takehiro Tominaga: Bug in quantize.c absolute threshold of hearing calculation
for non-44.1kHz input files.
LAME 3.50 November 1 1999
-
LAME 3.37beta becomes official LAME 3.50 release
LAME 3.37beta November 1 1999
-
Lionel Bonnet: Found severe bug in MPEG2 Short
block SNR.
-
Sergey Sapelin: VBR Toc improvement.
-
Sergey Dubov: fskip() routine
-
Conrad Sanderson: replacement for filterbank.c. Not much faster
but amazingly simpler.
LAME 3.36beta October 25 1999
-
Albert Faber: more MSVC and BladeDLL updates
-
Kimmo Mustonen: Much code cleanup and Amiga updates
-
Anton Oleynikov: Borland C updates
-
More stdin fixes: For some reason, forward fseek()'s would fail when
used on pipes even though it is okay with redirection from "<". So I
changed all the forward fseek()'s to use fread(). This should improve
stdin support for wav/aiff files. If you know the input file is raw
pcm, you can still use the '-r' option to avoid *all* seeking of any kind.
LAME 3.35beta October 21 1999
-
Leonid Kulakov: Serious bug in MPEG2 scalefactor
band tables fixed.
-
Portability patches from: Anton Oleynikov, Sigbjørn Skjæret,
Mathew Hendry, Richard Gorton
-
Alfred Weyers: compiler options, updated timestatus.
-
Albert Faber: BladeDll and other updates (new machine.h).
-
Monty: updated Makefile to fix gcc inline math bug.
LAME 3.34beta October 12 1999
-
Bug fixed: minimum bitrate in VBR mode could
be ignored for a few frames.
-
New (minor) VBR tunings.
-
Tim Ruddick: New wav/aiff header parsing routines. Better parsing
and fewer fseek()'s.
-
Anton Oleynikov: patches to work with Borland C
-
Gabriel Bouvigne: Experimental voice option
enabled with --voice
LAME 3.33beta October 11 1999
-
Robert Hegemann: RH VBR mode now the default and
only VBR mode. The new code will always quantize to 0 distortion
and the quality is increased by reducing the masking from the psy-model.
-X0 is still the default for now.
-
Robert Hegemann: new -X5 mode
-
Mathew Hendry: New timing code, removes the need for HAVETIMES
-
Mathew Hendry: assembler quantize_xrpow for
Windows
-
Iwasa Kazmi: stdin/stdout patch for Windows
-
New option: "--athonly" will ignore the psy-model output and use only the
absolute threshold of hearing for the masking.
LAME 3.32beta October 8 1999
-
Takehiro Tominaga: faster long block spreading
function convolution for non 44.1kHz sampling frequencies, and faster short
block spreading function convolution for all sampling frequencies.
-
Takehiro Tominaga: Completly rewritten huffman
table selection and count_bits(). More efficient table selection
results in many more bits per frame.
-
Takehiro Tominaga: More efficient scalefac
compress setting.
-
Mike Cheng: new calc_noise2()
-
Alfred Weyers: patch for timestatus() seconds rollover
LAME 3.31beta September 28 1999
-
Albert Faber: updated his BladeDLL code. This allows LAME to
be compiled into a BladeEnc compatiable .dll.
-
Mike Cheng: faster l3psycho_ener() routine.
-
Sigbjørn Skjæret: more code cleanup.
LAME 3.30beta September 27 1999
-
Conrad Sanderson: ID3 tag code added (type 'lame' for instructions)
-
new mdct.c from Mike Cheng (no faster, but much cleaner code)
-
Mathew Hendry: Microsoft nmake makefile and a couple other changes for
MSVC
-
More modulization work: One input sound file interface handles mp3's,
uncompressed audio, with or without LIBSNDFILE. Fixes (hopefully)
a bunch of file I/O bugs introduced in 3.29
-
LAME will now print valid samplerate/bitrate combinations
-
stdin/stdout fix for OS/2 (Paul Hartman)
-
For mp3 input files, totalframes estimated based on filesize and first
frame bitrate.
-
Updated all functions with new style prototypes. (Sigbjørn
Skjæret)
LAME 3.29beta September 21 1999
-
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 (Sigbjørn
Skjæret)
-
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.
LAME 3.28beta September 15 1999
-
Serious bug fixed in high frequency MDCT coefficients.
Huffman coding was reversing the order of the count1 block quadruples.
(Leonid A. Kulakov)
-
nint() problems under Tru64 unix fixed and preprocessor variable HAVE_NINT
removed. (Bob Bell)
-
Compiler warning fixes and code cleanup (Sigbjørn Skjæret,
Lionel Bonnet)
-
USAGE file now includes suggestions for downsampling. For low bitrate
encodings, proper downsampling can give dramatically better results.
(John Hayward-Warburton)
LAME 3.27beta September 12 1999
-
Several bugs in encode.c and l3bitstream.c fixed by Lionel Bonnet.
-
Bugs in new VBR (#define RH) formula for mono input file and mid/side encoding
fixed.
LAME 3.26beta September 10 1999
-
The "-m m" option (mono .mp3 file) will automatically mix left and right
channels if the input file is stereo. (Alfred Weyers)
-
New quant_compare algorithm (method for deciding
which of two quantizations is better) enabled with -X4 (Greg Maxwell)
-
New mid/side VBR bit allocation formula. Mid
channel bits are set by the quality requirements, and then the side channel
uses a reduced number of bits (in a proportion coming from the fixed bitrate
code). This might not be optimal, but it should be pretty good and
no one knows what the optimal solution should be. (Greg Maxwell)
-
New VBR (#define RH) tunings based on detailed listening
tests by Macik and Greg Maxwell.
-
Sigbjørn Skjæret fixed several compiler warnings (which turned
out to be potential bugs)
-
Takehiro Tominaga fixed a low bitrate bug in reduce_side()
-
Alfred Weyers fixed some buffer overflows.
-
New ATH (absolute threshold of hearing) formula replaces
buggy ISO code, and adds analog silence treatment (removal
of coefficients below below ATH). These are turned on by default
but have not been fully tested. (Robert Hegemann)
-
Bug in short block spreading function fixed.
(Robert Hegemann)
LAME 3.25beta August 22 1999
-
Sigbjørn Skjæret fixed a zero byte malloc call. This
bug was introduced in 3.24 and causes problems on non Linux systems.
-
Bit allocation routines would sometimes allocate more than 4095 bits to
one channel of one granule. A couple of people reported problems
that might be caused by this, especially at higher bitrates.
-
Nils Faerber updated the man page and fixed many of the compiler warnings.
LAME 3.24beta August 15 1999
-
This release contains the following new code (for developers) which is
disabled by default:
-
Robert Hegemann: Completely overhauled VBR code. Now computes
exact number of bits required for the given qualty and then quantized with
the appropriate bitrate.
-
Several new quantization quality measures.
LAME 3.23beta August 8 1999
-
Very nice continuously updated VBR histogram display from Iwasa Kazmi.
(disabled with --nohist).
-
More modulerization work. The encoding engine can now be compiled
into libmp3lame, but the interface is awkward.
-
Bug fixed in FFT Hann window formula (Leonid A. Kulakov).
-
New LAME logo on the download page. Created by Chris Michalisles.
-
Several VBR algorithm improvements from Robert Hegemann.
New quantization noise metrics and VBR quality measure takes into account
mid/side encoding. Should produce smaller files with the same quality,
especially when using jstereo.
LAME 3.22beta July 27 1999
-
Downsampling (stereo to mono) bug with MPEG2 fixed. (Mike Oliphant)
-
Downsampling now merges L & R channels - before it only took the L
channel.
-
More modularization and code cleanup from Albert Faber and myself.
-
Input filesize limit removed for raw pcm input files. For other file
types, LAME will still only read the first 2^32 samples, (27 hours of playing
time at 44.1kHz).
LAME 3.21beta July 26 1999
-
Correct Mid/Side masking thresholds for JSTEREO mode!
This is enabled with -h. It makes LAME about 20% slower since
it computes psycho-acoustics for L,R Mid and Side channels.
-
"Analog silence" threshold added. Keeps VBR
from upping the bitrate during very quite passages. (Robert.Hegemann)
-
New VBR quality setting from Robert Hegemann.
It is based on the idea that distortion at lower bit rates sounds worse
than at higher bitrates, and so the allowed distortion (VBR quality setting)
is proportional to the bitrate. Because of this, default minimum
bitrate is now 32kbs.
-
Expermental subblock gain code enabled with -Z.
-
New "-r" option for raw pcm input files. With -r, LAME will not do
any fseek()'s or look for wav and aiff headers on the input file.
-
Bug fixes in mp3x (frame analyzer) for viewing frames near end of the file.
-
Bug fixed to allow setting the sampling rate of raw pcm input files.
LAME 3.20beta July 19 1999
-
Bug in get_audio.c fixed. Libsndfile wrappers would not compile (Miguel
Revilla Rodriguez)
-
Nils Faerber found some unitialized variables and some wierd extranous
computations in filter_subband, now fixed. This was causing seg faults
on some machines.
LAME 3.19beta July 18 1999
-
Oops! Robert Hegemann immediatly found a bug
in the new (old -Z option) quantization code. calc_noise1 was
not returning tot_noise, so non ms-stereo frames were buggy.
LAME 3.18beta July 17 1999
-
Many psycho-acoustic bug fixes. Dan Nelson
discovered a bug in MPEG2: For short blocks, the code assumes 42 partition
bands. MPEG1 sometimes has less, MPEG2 can have more. In MPEG1,
this bug would not have effected the output if your compiler initializes
static variables to 0 on creation. In MPEG2 it leads to array out-of-bounds
access errors. Finally, there was a related bug in MPEG1/MPEG2, short &
long blocks where the energy above 16kHz was all added to partition band
0. (the lowest frequeny partition band!)
-
The -Z option (Gabriel Bouvigne's idea of using total
quantization noise to choose between two quantizations with the same value
of "over") is now the default. I believe this helps remove the trilling
sound in Jan's testsignal4.wav. The quality of testsignal2.wav and
testsignal4.wav are now better than Xing and getting closer to FhG.
-
Bug fixes in frame & sample count for downsampling mode. (ben "jacobs")
-
Patches to improve modulization. (ben "jacobs")
LAME 3.17beta July 11 1999
-
substantial code cleanup towards goal of making LAME more modular.
LAME 3.16beta July 11 1999
-
New tunings of window switching, and better bit allocation
based on pe. (Jan Rafaj. improves both testsignal2.wav and
testsignal4.wav).
-
Bug in mid/side quantization when side channel was
zero fixed. (Albert Faber)
-
Removed some extranous computations in l3psy.c (Robert Hegemann)
-
More detailed timing status info, including hours display. (Sakari Ailus)
and percentage indicator (Conrad Sanderson).
-
Window_subband and calc_noise1,calc_noise2 speedups.
Quantize_xrpow speedup should be significant on non GNU/intel systems.
(Mike Cheng)
-
Better initial guess for VBR bitrate. Should
speed up VBR encoding. (Gabriel Bouvigne)
-
More advanced .wav header parsing. fixes bugs involving click in
first frame. (Robert.Hegemann)
-
Correct filesize and total frame computation when using LIBSNDFILE (ben
"jacobs")
-
Click in last frame (buffering problem) when using libsndfile fixed.
-
Audio I/O code overhauled. There is now a uniform audio i/o interface
to libsndfile or the LAME built in wav/aiff routines. All audio i/o
code localized to get_audio.c.
LAME 3.15beta
-
times()/clock() problem fixed for non-unix OS. (Ben "Jacobs")
-
Fixed uninitialized pe[] when using fast mode. (Ben "Jacobs")
LAME 3.13 June 24 1999
-
Patches for BeOS from Gertjan van Ratingen.
-
Makefile info for OS/2 Warp 4.0 (from dink.org).
-
Status display now based on wall clock time, not cpu time.
-
mem_alloc no longer allocates twice as much memory as needed (Jan Peman).
3.12pre9
-
Updated BLADEDLL code to handle recent changes (Albert Faber).
-
Bug fixed in parsing options when not using GTK (Albert Faber).
-
MPEG2 Layer III psycho acoustics now working.
-
Improved huffman encoding Chris Matrakidis. (10%
faster). I dont know how he finds these improvements! LAME
with full quality now encodes faster than real time on my PII 266.
-
Fixed time display when encoding takes more than 60 minutes.
3.12pre8
-
New mid/side stereo
criterion. LAME will use mid/side stereo only when the difference
between L & R masking thresholds (averaged over all scalefactors) is
less then 5db. In several test samples it does a very good job mimicking
the FhG encoder.
-
Bug in mid/side stereo fixed: independent variation
of mid & side channel scalefactors disabled. Because of the way
outer_loop is currently coded, when encoding mid/side coefficietns using
left/right thresholds, you have to vary the scalefactors simultaneously.
-
Bug in side/mid energy ratio calculation fixed. (Thanks
to Robert Hegemann)
-
Default mode is stereo (not jstereo) if bitrate is chosen as 192kbs or
higher. Tero Auvinen first pointed out that FhG seems to think at
160kbs, their encoder is so good it doesn't need jstereo tricks. Since
LAME is not as good as FhG, I am going to claim that 192kbs LAME is so
good it doens't need jstereo tricks, and thus it is disabled by default.
-
WAV header parsing for big-endian machines, and automatic detection of
big-endian machines. (Thanks to Sigbjørn Skjæret).
-
added 56 sample delay to sync LAME with FhG.
-
MP3x (frame analyzer) can now handle MPEG2 streams.
3.12pre7
-
MPEG2 layer III now working! lower bit rates (down to 8kbs) and 3
more sampling frequencies: 16000, 22050, 24000Hz. Quality is poor
- the psy-model does not yet work with these sampling frequencies.
-
Fixed "ERROR: outer_loop(): huff_bits < 0." bug when using VBR.
-
bash and sh scripts to run LAME on multiple files now included. (from
Robert Hegemann and Gerhard Wesp respectively)
-
bug fix in encoding times for longer files from (Alvaro Martinez
Echevarria)
-
yet another segfault in the frame analyzer fixed.
-
ISO psy-model/bit allocation routines removed. This allowed makeframe()
to be made much simpler, and most of the complicated buffering is now gone.
Eventually I would like the encoding engine to be a stand alone library.
3.12pre6
-
Better VBR tuning. Find minimum bitrate with
distortion less than the allows maximum. A minimum bit rate is imposed
on frames with short blocks (where the measured distortion can not be trusted).
A minimum frame bitrate can be specified with -b, default=64kbs.
-
LIBSNDFILE support.
With libsndfile, LAME can encode almost all sound formats. Albert
Faber did the work for this, including getting libsndfile running under
win32.
-
CRC checksum now working! (Thanks to Johannes Overmann )
-
frame analyzer will now work with mono .mp3 files
-
more code tweeks from Jan Peman.
-
Compaq-Alpha(Linux) fixes and speedups from Nils
Faerber.
-
Faster bin_search_StepSize from Juha Laukala.
-
Faster quantize() from Mike Cheng
-
Faster quantize_xrpow() from Chris Matrakidis.
xrpow_flag removed since this option is now on by default.
-
Fixed .wav header parsing from Nils Faerber.
-
Xing VBR frame info header code from Albert Faber. "Xing" and
"LAME 3.12" embedded in first frame.
-
Bug in VBR bit allocation based on "over" value fixed.
LAME 3.11 June 3 1999
Almost all warnings (-Wall) now fixed! (Thanks to Jan Peman)
More coding improvements from Gabriel Bouvigne and Warren Toomey.
VBR (variable bit rate). Increases
bit rate for short blocks and for frames where the number of bands containing
audible distortion is greater than a given value. Much tuning needs
to be done.
Patch to remove all atan() calls from James Droppo.
LAME 3.10 May 30 1999
-
Fast mode (-f) disables psycho-acoustic model
for real time encoding on older machines. Thanks to Lauri Ahonen
who first sent a patch for this.
-
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 myself. They work great! now on by default.
-
In jstereo, force blocktypes for left & right channels to be identical.
FhG seems to do this. It can be disabled with "-d".
-
Patches to compile MP3x under win32 (Thanks to Albert Faber).
-
bin_serach_stepsize limited to a quantizationStepSize
of -210 through 45.
-
outer_loop() will now vary Mid & Side 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 run at the old speed
-
Bug in inner_loop would allow quantizations larger
than allowed. (introduced in lame3.04, now fixed.)
-
Updated HTML documentation from Gabriel Bouvigne.
-
Unix man page from William Schelter.
-
numlines[] bug fixed. (Thanks to Rafael Luebbert,
MPecker author).
-
Quantization speed improvements from Chirs Matrakidis.
-
When comparing quantizations with the same number
of bands with audible distortion, use the one with the largest scalefactors,
not the first one outer_loop happened to find.
-
Improved defination of best quantization when using -f (fast mode).
-
subblock code now working. But no algorithm to choose subblock gains
yet.
-
Linux now segfaults on floating point exceptions. Should prevent
me from releasing binaries that crash on other operating systems.
May 22 1999
-
Version 3.04 released.
-
Preliminary documentation from Gabriel Bouvigne.
-
I wouldn't have thought it was possible, but now
there are 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.
May 18 1999
-
Version 3.03 released.
-
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.
-
improved status display (Thanks to Lauri Ahonen).
May 12 1999
-
Version 3.02 released.
-
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)
May 11 1999
-
Version 3.01 released
-
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)
May 10 1999
-
Version 3.0 released
-
added GPSYCHO (developed by Mark Taylor)
-
added MP3x (developed by Mark Taylor)
-
LAME now maintained by Mark Taylor
November 8 1998
-
Version 2.1f released
-
50% faster filter_subband() routine in encode.c contributed by James Droppo
November 2 1998
-
Version 2.1e released.
-
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.
October 31 1998
-
Version 2.1d released
-
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.
October 18 1998
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Version 2.1b released.
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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.
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Compiling with amiga-gcc works fine
October 16 1998
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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.
October 6 1998
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Version 2.1 released with all tables now incorporated into the exe. Thanks
to Lars Magne Ingebrigtseni(larsi@ifi.uio.no)
October 4 1998In response to some concerns about the quality of
the encoder, I have rebuilt the encoder from scratch and carefully compared
output at all stages with the output of the unmodified ISO encoder. Version2.0
of LAME is built from the ISO source code (dist10), and incorporates modifications
from myself and the 8hz effort. The output file from LAME v2.0 is identical
to the output of the ISO encoder for my test file.Since I do not have heaps
of time, I left the ISO AIFF file reader in the code, and did not incorporate
a WAV file reader.Added section on qualityOctober
1 1998
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Updated web page and released LAME v1.0
Up to September 1998
Working on the 8hz source code...
-
Patched the 8hz source code
-
45% faster than original source (on my freebsd p166).
-
m1 - sped up the mdct.c and quantize() functions [MDCTD, MDCTD2, LOOPD]
-
m2 - sped up the filter_subband routine using Stephane Tavenard's
work from musicin [FILTST]
-
m2 - minor cleanup of window_subband [WINDST2]
-
m2 - Cleaned up a few bits in l3psy.c. Replaced a sparse matrix multiply
with a hand configured unrolling [PSYD]
-
m3 - (amiga only) Added in the asm FFT for m68k (based on sources from
Henryk
Richter and Stephane Tavenard)
-
m4 - raw pcm support back in
-
m5 - put in a byte-ordering switch for raw PCM reading (just in case)
-
m6 - reworked the whole fft.c file. fft now 10-15% faster.
-
m7 - totally new fft routine. exploits fact that this is a real->complex
fft. About twice as fast as previous fastest fft (in m6). (C fft routine
is faster than the asm one on an m68k!)
-
m8
-
- Now encodes from stdin. Use '-' as the input filename. Thanks to Brad
Threatt
-
- Worked out that the 1024point FFT only ever uses the first 6 phi values,
and the first 465 energy values. Saves a bunch of calculations.
-
- Added a speed-up/quality switch. Speed is increased but quality is decreased
slightly.
My ears are bad enough not to be able to notice the difference in quality
at low settings :). Setting '-q 1' improves speed by about 10%. '-q 100'
improves speed by about 26%. Enoding of my test track goes from 111s (at
default '-q 0') to 82s (at -q 100). Thanks to Jan Peman for this
tip.
-
m9 - fixed an error in l3psy.c. numlines[] is overwritten with incorrect
data. Added a new variable numlines_s[] to fix this. Thanks again to Jan
Peman.
-
m10 - Down to 106 seconds by selecting a few more compiler options. Also
added a pow20() function in l3loop.c to speed up (ever so slightly) calls
to pow(2.0, x)
-
m11
-
No speedups. Just cleaned up some bits of the code.
-
Changed K&R prototyping to 'normal' format. Thanks to Steffan Haeuser
for his help here.
-
Changed some C++ style comments to normal C comments in huffman.c
-
Removed the #warning from psy_data.h (it was getting annoying!)
-
Removed reference in bitstream.c to malloc.h. Is there a system left where
malloc.h hasn't been superceded by stdlib.h?
-
In Progess:
-
my PSYD hack for the spreading functions is only valid for 44.1khz - Should
really put in a "if freq = 44.1khz" switch for it. Someone might want to
extend the speedup for 48 and 32khz.
-
Putting in Jan Peman's quantanf_init speedup.