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mp3plot 0.4.0 Alpha
mp3plot project prints out a plot of the bitrate distribution of a VBR MP3 file. more>>
mp3plot project prints out a plot of the bitrate distribution of a VBR MP3 file (it will also do it for CBR files although it isnt very meaningful).
It should be architecture independent but I havent tested beyond PCs.
Theres a much more mature tool that does the same and more: mp3stat at signal-lost.homeip.net/projects. mp3stat refuses to work on my system(s) and having an interest in mp3s internal structure I gave a shot at it with mp3plot.
<<lessIt should be architecture independent but I havent tested beyond PCs.
Theres a much more mature tool that does the same and more: mp3stat at signal-lost.homeip.net/projects. mp3stat refuses to work on my system(s) and having an interest in mp3s internal structure I gave a shot at it with mp3plot.
Download (0.030MB)
Added: 2007-05-29 License: GPL (GNU General Public License) Price:
878 downloads
xmpgedit 0.72 Beta3
xmpgedit is an MP3 editor for VBR/CBR encoded files. more>>
xmpgedit project is an MP3 editor for VBR and CBR encoded files.
xmpgedit is an MPEG 1 layer 1/2/3 (MP3), MPEG 2, and MPEG 2.5 audio file editor that is capable of processing both Constant Bit Rate (CBR) and Variable Bit Rate (VBR) encoded files.
xmpgedit can cut an input MPEG file into one or more output files, as well as join one or more input MPEG files into a single output file. Since no decoding or encoding occurs during editing, there is no audio quality loss when editing with xmpgedit.
Enhancements:
- This release fixes an occasional crash in xmpgedit when adjusting the volume after playback has stopped.
- xmpgedit playto start time is fixed, which would play beyond the specified edit time in some circumstances.
- Added the command "Copy start time" to the xmpgedit edit menu.
<<lessxmpgedit is an MPEG 1 layer 1/2/3 (MP3), MPEG 2, and MPEG 2.5 audio file editor that is capable of processing both Constant Bit Rate (CBR) and Variable Bit Rate (VBR) encoded files.
xmpgedit can cut an input MPEG file into one or more output files, as well as join one or more input MPEG files into a single output file. Since no decoding or encoding occurs during editing, there is no audio quality loss when editing with xmpgedit.
Enhancements:
- This release fixes an occasional crash in xmpgedit when adjusting the volume after playback has stopped.
- xmpgedit playto start time is fixed, which would play beyond the specified edit time in some circumstances.
- Added the command "Copy start time" to the xmpgedit edit menu.
Download (0.18MB)
Added: 2006-04-20 License: GPL (GNU General Public License) Price:
1282 downloads
Audio::TagLib::MPEG::XingHeader 1.42
Audio::TagLib::MPEG::XingHeader is an implementation of the Xing VBR headers. more>>
Audio::TagLib::MPEG::XingHeader is an implementation of the Xing VBR headers.
SYNOPSIS
use Audio::TagLib::MPEG::XingHeader;
my $i = Audio::TagLib::MPEG::XingHeader->new($data);
print $i->isValid() ? "valid" : "invalid", "n";
This is a minimalistic implementation of the Xing VBR headers. Xing headers are often added to VBR (variable bit rate) MP3 streams to make it easy to compute the length and quality of a VBR stream. Our implementation is only concerned with the total size of the stream (so that we can calculate the total playing time and the average bitrate). It uses http://home.pcisys.net/~melanson/codecs/mp3extensions.txt and the XMMS sources as references.
new(ByteVector $data)
Parses a Xing header based on $data. The data must be at least 16 bytes long (anything longer than this is discarded).
DESTROY()
Destroy this XingHeader instance
BOOL isValid()
Returns true if the data was parsed properly and if there is a vaild Xing header present.
UV totalFrames()
Returns the total number of frames.
UV totalSize()
Returns the total size of stream in bytes.
IV xingHeaderOffset(PV $version, PV $channelMode) [static]
Returns the offset for the start of this Xing header, given the version and channels of the frame
see Audio::TagLib::MPEG::Header
<<lessSYNOPSIS
use Audio::TagLib::MPEG::XingHeader;
my $i = Audio::TagLib::MPEG::XingHeader->new($data);
print $i->isValid() ? "valid" : "invalid", "n";
This is a minimalistic implementation of the Xing VBR headers. Xing headers are often added to VBR (variable bit rate) MP3 streams to make it easy to compute the length and quality of a VBR stream. Our implementation is only concerned with the total size of the stream (so that we can calculate the total playing time and the average bitrate). It uses http://home.pcisys.net/~melanson/codecs/mp3extensions.txt and the XMMS sources as references.
new(ByteVector $data)
Parses a Xing header based on $data. The data must be at least 16 bytes long (anything longer than this is discarded).
DESTROY()
Destroy this XingHeader instance
BOOL isValid()
Returns true if the data was parsed properly and if there is a vaild Xing header present.
UV totalFrames()
Returns the total number of frames.
UV totalSize()
Returns the total size of stream in bytes.
IV xingHeaderOffset(PV $version, PV $channelMode) [static]
Returns the offset for the start of this Xing header, given the version and channels of the frame
see Audio::TagLib::MPEG::Header
Download (1.4MB)
Added: 2006-11-14 License: Perl Artistic License Price:
1076 downloads
Punit 0.2
Punit is an attempt to create a Python CD ripper using cdparanoia and lame for Linux. more>>
Punit is an attempt to create a Python CD ripper using cdparanoia and lame for Linux.
I couldnt get Sound Juicer to rip mp3, GooBox encodes mp3 using VBR which screws up the mp3 times in Foobar2000. I wanted a simple mp3 ripper and i wanted to learn Python so I made my own.
Right now, its a straight forward ripper to mp3 with the format 128 kbps, 44100Hz, stereo. Comments, suggestions, improvements are all welcomed.
Thanks to CDDB.py for the CDDB module.
<<lessI couldnt get Sound Juicer to rip mp3, GooBox encodes mp3 using VBR which screws up the mp3 times in Foobar2000. I wanted a simple mp3 ripper and i wanted to learn Python so I made my own.
Right now, its a straight forward ripper to mp3 with the format 128 kbps, 44100Hz, stereo. Comments, suggestions, improvements are all welcomed.
Thanks to CDDB.py for the CDDB module.
Download (0.032MB)
Added: 2007-08-11 License: GPL (GNU General Public License) Price:
804 downloads

LAME MP3 Encoder 3.98.2
Today, LAME is considered the best MP3 encoder at mid-high bitrates and at VBR. more>> LAME development started around mid-1998. Mike Cheng started it as a patch against the 8hz-MP3 encoder sources. After some quality concerns raised by others, he decided to start from scratch based on the dist10 sources. His goal was only to speed up the dist10 sources, and leave its quality untouched. That branch (a patch against the reference sources) became Lame 2.0, and only on Lame 3.81 did we replaced of all dist10 code, making LAME no more only a patch.
The project quickly became a team project. Mike Cheng eventually left leadership and started working on tooLame, an MP2 encoder. Mark Taylor became leader and started pursuing increased quality in addition to better speed. He can be considered the initiator of the LAME project in its current form. He released version 3.0 featuring gpsycho, a new psychoacoustic model he developed.
In early 2003 Mark left project leadership, and since then the project has been lead through the cooperation of the active developers (currently 4 individuals).
Today, LAME is considered the best MP3 encoder at mid-high bitrates and at VBR, mostly thanks to the dedicated work of its developers and the open source licensing model that allowed the project to tap into engineering resources from all around the world. Both quality and speed improvements are still happening, probably making LAME the only MP3 encoder still being actively developed.<<less
Download (1.27MB)
Added: 2009-04-08 License: Freeware Price:
198 downloads
Other version of LAME MP3 Encoder
License:Freeware
MP3SPI for Linux 1.9.4
MP3SPI is a Java Service Provider Interface that adds MP3 more>> MP3SPI is a Java Service Provider Interface that adds MP3 (MPEG 1/2/2.5 Layer 1/2/3) audio format support for Java Platform. It supports streaming, ID3v2 frames, Equalizer, .... It is based on JLayer and Tritonus Java libraries.
MP3 support (MPEG 1/2/2.5 Layer 1/2/3). VBR support, ID3v2 frames support.Skip support.Equalizer support.This release targets J2SE 1.3 and 1.4 but it provides audio properties that will be available in J2SE 1.5 :
MpegAudioFormat (bitrate, vbr).
MpegAudioFileFormat (duration, title, author, album, date, copyright, comments).It also provides custom properties :
MpegAudioFileFormat (mp3.version.mpeg, mp3.version.layer, mp3.framerate.fps, mp3.id3tagv2, ...).
DecodedMpegAudioInputStream (mp3.frame.bitrate, mp3.equalizer, ...) jUnit tests included.CPU usage : ~12% under PIII 1Ghz/Win2K+J2SE 1.4.1
FootPrint : ~10MB under WinNT4/Win2K + J2SE 1.4.1
RIFF/MP3 header support added.
FLAC and MAC headers denied.
Skip bug fixed for 320kbps files.
ID3v2.x support improved :
size computation bug fixed.
"mp3.id3tag.publisher" (TPUB/TPB) added.
"mp3.id3tag.orchestra" (TPE2/TP2) added.
"mp3.id3tag.length" (TLEN/TLE) added.
Mark limit increased.<<less
Download (289KB)
Added: 2009-04-04 License: Freeware Price: Free
202 downloads
cutmp3 1.9.2
cutmp3 is a small and fast command line MP3 editor. more>>
cutmp3 is a small and fast command line MP3 editor. cutmp3 lets you select sections of an MP3 interactively or via a timetable and save them to separate files without quality loss.
It uses mpg123 for playback and works with VBR files and even with files bigger than 2GB.
Other features are configurable silence seeking and ID3 tag seeking, which are useful for concatenated mp3s.
Playback is realized via mpg123, so be sure to have it installed or at least have a symlink to your favorite mp3 playing program.
I recommend a symlink to mpg321 which works better!
You can mark beginning and end of a segment with a and b and save the segment with s.
Using a timetable or direct times with VBR files delivers exact(!) results at the cost of slightly lower speed. cutmp3 even works with files bigger than 2 GB!
If you want a working graphical software you can try mp3directcut,
which runs fairly well in WINE after I asked the author about a WINEd version.
Installation:
make (you will need readline-devel or similar!)
install it to /usr/bin with
make install
Enhancements:
- A small fix for timetable cutting with negative values has been made.
- Another option for silence seeking has been added.
- The documentation has been updated.
<<lessIt uses mpg123 for playback and works with VBR files and even with files bigger than 2GB.
Other features are configurable silence seeking and ID3 tag seeking, which are useful for concatenated mp3s.
Playback is realized via mpg123, so be sure to have it installed or at least have a symlink to your favorite mp3 playing program.
I recommend a symlink to mpg321 which works better!
You can mark beginning and end of a segment with a and b and save the segment with s.
Using a timetable or direct times with VBR files delivers exact(!) results at the cost of slightly lower speed. cutmp3 even works with files bigger than 2 GB!
If you want a working graphical software you can try mp3directcut,
which runs fairly well in WINE after I asked the author about a WINEd version.
Installation:
make (you will need readline-devel or similar!)
install it to /usr/bin with
make install
Enhancements:
- A small fix for timetable cutting with negative values has been made.
- Another option for silence seeking has been added.
- The documentation has been updated.
Download (0.033MB)
Added: 2007-04-06 License: GPL (GNU General Public License) Price:
931 downloads
Aften 0.05
Aften is a simple, open-source, A/52 (AC-3) audio encoder. more>>
Aften project is a simple, open-source, A/52 (AC-3) audio encoder.
Main features:
- Implemented my own wav reader
- Converted the fixed-point algorithms to floating-point
- Rearranged the methods and structures
- Added stereo rematrixing (mid/side)
- Added short block MDCT and block switching
- Added VBR encoding mode
- Added variable bandwidth
- Added more complete WAV format support
- Added support for using the alternate bit stream syntax
- Created separate library and frontend
- Added input filters
Enhancements:
- Bit allocation speedups, a compile-time choice of using floats or doubles internally, an internal restructuring of MDCT functions, and bugfixes. quality=0 is now a valid setting.
<<lessMain features:
- Implemented my own wav reader
- Converted the fixed-point algorithms to floating-point
- Rearranged the methods and structures
- Added stereo rematrixing (mid/side)
- Added short block MDCT and block switching
- Added VBR encoding mode
- Added variable bandwidth
- Added more complete WAV format support
- Added support for using the alternate bit stream syntax
- Created separate library and frontend
- Added input filters
Enhancements:
- Bit allocation speedups, a compile-time choice of using floats or doubles internally, an internal restructuring of MDCT functions, and bugfixes. quality=0 is now a valid setting.
Download (0.046MB)
Added: 2006-08-22 License: GPL (GNU General Public License) Price:
1165 downloads
mpgedit 0.72 Beta3
mpgedit is an MPEG 1 layer 1/2/3 (mp3), MPEG 2, and MPEG 2.5 audio file editor. more>>
mpgedit is an MPEG 1 layer 1/2/3 (mp3), MPEG 2, and MPEG 2.5 audio file editor that is capable of processing both Constant Bit Rate (CBR) and Variable Bit Rate (VBR) encoded files.
Since no file decoding / encoding occurs during editing, there is no audio quality loss when editing with mpgedit.
When editing VBR files that have a XING header, mpgedit updates the output files XING header information to reflect the new file size and average bit rate.
mpgedit operates in two modes: a command line and a curses-based, full-screen interactive shell. All editing functionality is available in both modes. The chief advantage of the interactive mode comes when using the playback feature for selection and verification of edit begin and end times, which is more rapidly performed interactively.
xmpgedit provides most of the same editing capabilities as the curses-mode mpgedit, but adds a graphical representation of the volume levels of mp3 file being being edited.
Enhancements:
- This release fixes an occasional crash in xmpgedit when adjusting the volume after playback has stopped.
- mpgedit playto start time is fixed, which would play beyond the specified edit time in some circumstances.
- The command "Copy start time" has been added to the xmpgedit edit menu.
- mpgedit playback and editing with the -I option is fixed, which would frequently repeat up to a second of material at the start of the edit.
<<lessSince no file decoding / encoding occurs during editing, there is no audio quality loss when editing with mpgedit.
When editing VBR files that have a XING header, mpgedit updates the output files XING header information to reflect the new file size and average bit rate.
mpgedit operates in two modes: a command line and a curses-based, full-screen interactive shell. All editing functionality is available in both modes. The chief advantage of the interactive mode comes when using the playback feature for selection and verification of edit begin and end times, which is more rapidly performed interactively.
xmpgedit provides most of the same editing capabilities as the curses-mode mpgedit, but adds a graphical representation of the volume levels of mp3 file being being edited.
Enhancements:
- This release fixes an occasional crash in xmpgedit when adjusting the volume after playback has stopped.
- mpgedit playto start time is fixed, which would play beyond the specified edit time in some circumstances.
- The command "Copy start time" has been added to the xmpgedit edit menu.
- mpgedit playback and editing with the -I option is fixed, which would frequently repeat up to a second of material at the start of the edit.
Download (0.20MB)
Added: 2006-04-20 License: GPL (GNU General Public License) Price:
1283 downloads
dvdspanky 1.0.10
dvdspanky is a CLI tool to convert video files into DVD compatible MPEG streams. more>>
dvdspanky is a CLI tool to convert video files into DVD compatible MPEG streams. It is designed to be easy to use no matter the input source, to automate common transcoding tasks and provide powerful features. It is written in C and provides a front-end to transcode, MJPEG tools, mplayer and feh. The output can be used in programs like dvdauthor.
Main features:
- Consistent options no matter the input source
- Clean and clear output
- VBR and CBR encoding
- Output preview
- Destination file size specification
- PAL and NTSC export
- Post-processing
- Automatic volume adjustment
- Automatic cropping and letter-boxing
- Automatic mplayer fallback decoding
- Automatic aspect calculation
- Automatic DVD compatibility adjustments
<<lessMain features:
- Consistent options no matter the input source
- Clean and clear output
- VBR and CBR encoding
- Output preview
- Destination file size specification
- PAL and NTSC export
- Post-processing
- Automatic volume adjustment
- Automatic cropping and letter-boxing
- Automatic mplayer fallback decoding
- Automatic aspect calculation
- Automatic DVD compatibility adjustments
Download (0.099MB)
Added: 2007-06-22 License: GPL (GNU General Public License) Price:
854 downloads
tooLAME 0.2i
tooLAME is an optimized Mpeg Audio 1/2 Layer 2 encoder. more>>
tooLAME is an optimized Mpeg Audio 1/2 Layer 2 encoder.
It is based heavily on:
- the ISO dist10 code
- improvement to algorithms as part of the LAME project
Installation:
1. edit Makefile
at least change the architecture type (ARCH) to suit your machine.
2. make
Usage:
./toolame [options] < input > < output >
Input File
tooLAME parses AIFF and WAV files for file info
raw PCM is assumed if no header is found
for stdin use a -
Output File
file is automatically renamed from *.* to *.mp2
for stdout use a -
Input Options
-s [int]
if inputting raw PCM sound, you must specify the sample rate
default sample rate is 44.1khz.
-a
downmix from stereo to mono
if the incoming file is stereo, combine the audio into
a single channel
-x
force byte-swapping of the input. (current endian detection is dodgy,
so if toolame produces only noise, use -x )
-g
swap the LR channels of a stereo file
Output Options
-m [char]
the encoding mode (default j)
s stereo
d dual channel
j joint stereo
m mono
-p [int]
which psy model to use (default 1)
Different models for the psychoacoustics
Models: -1 to 4
-b [int]
the total bitrate
For 48/44.1/32kHz default = 192
For 24/22.05/16kHz default = 96
-v [int]
Switch on VBR mode.
The higher the number the better the quality.
Useful range -10 to 10.
See README.VBR for details.
Operation
-f
fast mode turns off calculation of the psychoacoustic model.
Instead a set of default values are assumed
-q [int]
quick mode calculates the psy model every num frames.
Misc
-d emp
de-emphasis (default n)
-c
mark as copyright
-o
mark as original
-e
add error protection
-r
force padding bits off
-D
add DAB extensions
-t [int]
talkativity setting. 0 = no message. 3 = too much information
<<lessIt is based heavily on:
- the ISO dist10 code
- improvement to algorithms as part of the LAME project
Installation:
1. edit Makefile
at least change the architecture type (ARCH) to suit your machine.
2. make
Usage:
./toolame [options] < input > < output >
Input File
tooLAME parses AIFF and WAV files for file info
raw PCM is assumed if no header is found
for stdin use a -
Output File
file is automatically renamed from *.* to *.mp2
for stdout use a -
Input Options
-s [int]
if inputting raw PCM sound, you must specify the sample rate
default sample rate is 44.1khz.
-a
downmix from stereo to mono
if the incoming file is stereo, combine the audio into
a single channel
-x
force byte-swapping of the input. (current endian detection is dodgy,
so if toolame produces only noise, use -x )
-g
swap the LR channels of a stereo file
Output Options
-m [char]
the encoding mode (default j)
s stereo
d dual channel
j joint stereo
m mono
-p [int]
which psy model to use (default 1)
Different models for the psychoacoustics
Models: -1 to 4
-b [int]
the total bitrate
For 48/44.1/32kHz default = 192
For 24/22.05/16kHz default = 96
-v [int]
Switch on VBR mode.
The higher the number the better the quality.
Useful range -10 to 10.
See README.VBR for details.
Operation
-f
fast mode turns off calculation of the psychoacoustic model.
Instead a set of default values are assumed
-q [int]
quick mode calculates the psy model every num frames.
Misc
-d emp
de-emphasis (default n)
-c
mark as copyright
-o
mark as original
-e
add error protection
-r
force padding bits off
-D
add DAB extensions
-t [int]
talkativity setting. 0 = no message. 3 = too much information
Download (0.12MB)
Added: 2005-12-20 License: GPL (GNU General Public License) Price:
1403 downloads
mp3asm 0.1.3
mp3asm is an mpeg 1/2/2.5 audio layer 1/2/3 (cbr/vbr) frame level editor. more>>
mp3asm is an mpeg 1/2/2.5 audio layer 1/2/3 (cbr/vbr) frame level editor. The current textmode version allows cutting, copying, and pasting of frames while never violating the mpeg audio standard.
Well, most ppl probably prefer decoding, wav editing, reencoding, but that has consequences. Encoders use advanced filtering to create audio data thats easy to compress so that ppl can hardly tell the difference (depending on BR offcourse) and then they compress that filtered audiodata. Now filtering once from original full range unfiltered audio will provide the nicest result.
And offcourse, u dont have to waste precious cycles on the whole lengthy process.
Yes indeed, with wavediting programs u can edit right up to the millisecond, or even better. But to be honest, an 128kbps mpeg 1 layer 3file has 25 frames per second, which means u can cut right down to 40ms. If u dont mind the quality of a 128kbps file, how can u mind a few millisecs?Every subsequent filtering will result in more loss of data, because the only real difference between encoders is the filterbanks.
So when this app is up and running theres no more reason to waste a perfectly good rip cos of a few bad starting frames or a long lead out. At this point i even believe it should be possible to provide for fading in/out, i aint certain yet actually. I have no clue whatsoever on why Olli fromme halted development back in 97.Maybe cos mp3s were a rarity back then? Maybe some personal reason. I will prolly never know. What I do know is that he had an unbelievable idea and he wrote a program that did the job very well.
Enhancements:
- fixed parsing sideinfo (off by 1 bit) for mpeg 1 layer 3 mono (found by Alain Daurat (alain.daurat@libertysurf.fr))
<<lessWell, most ppl probably prefer decoding, wav editing, reencoding, but that has consequences. Encoders use advanced filtering to create audio data thats easy to compress so that ppl can hardly tell the difference (depending on BR offcourse) and then they compress that filtered audiodata. Now filtering once from original full range unfiltered audio will provide the nicest result.
And offcourse, u dont have to waste precious cycles on the whole lengthy process.
Yes indeed, with wavediting programs u can edit right up to the millisecond, or even better. But to be honest, an 128kbps mpeg 1 layer 3file has 25 frames per second, which means u can cut right down to 40ms. If u dont mind the quality of a 128kbps file, how can u mind a few millisecs?Every subsequent filtering will result in more loss of data, because the only real difference between encoders is the filterbanks.
So when this app is up and running theres no more reason to waste a perfectly good rip cos of a few bad starting frames or a long lead out. At this point i even believe it should be possible to provide for fading in/out, i aint certain yet actually. I have no clue whatsoever on why Olli fromme halted development back in 97.Maybe cos mp3s were a rarity back then? Maybe some personal reason. I will prolly never know. What I do know is that he had an unbelievable idea and he wrote a program that did the job very well.
Enhancements:
- fixed parsing sideinfo (off by 1 bit) for mpeg 1 layer 3 mono (found by Alain Daurat (alain.daurat@libertysurf.fr))
Download (0.045MB)
Added: 2006-07-21 License: GPL (GNU General Public License) Price:
1192 downloads
mpeglib 0.4.1
mpeglib is a mp3 and mpeg I video/audio library for linux. more>>
mpeglib is a mp3 and mpeg I video/audio library for linux.
The library includeds three command line players, for mp3,wav,mpeg video.
Main features:
MPEG I library.
This library contains:
- mpeg I audio player (layer I,II,III (mp3))
- mpeg I video player
- mpeg I system layer player
- wav player
Supported features:
- direct/fast seek in all players
- length detection
- video synchronisation, based on timestamps
- mmx Support where necessary
- VBR/ID3 support for mp3
- shoutcast/icecast support
- CDI/Video CD support
- plugin architecture for input,decoder,output
Supported Outputs:
Audio
- Support for OSS/Linux, Sun
Video
- X11 standard calls (fallback)
- X11 Shared mem
- X11 XFree86 4.0 DGA 2.0 (needs root)
- X11 XFree86 4.0 XVideo Extension (hardware yuv->rgb renderer)
Supported Inputs
- file,http.
- Supports on Linux Video CDs (vcd,cdi)
<<lessThe library includeds three command line players, for mp3,wav,mpeg video.
Main features:
MPEG I library.
This library contains:
- mpeg I audio player (layer I,II,III (mp3))
- mpeg I video player
- mpeg I system layer player
- wav player
Supported features:
- direct/fast seek in all players
- length detection
- video synchronisation, based on timestamps
- mmx Support where necessary
- VBR/ID3 support for mp3
- shoutcast/icecast support
- CDI/Video CD support
- plugin architecture for input,decoder,output
Supported Outputs:
Audio
- Support for OSS/Linux, Sun
Video
- X11 standard calls (fallback)
- X11 Shared mem
- X11 XFree86 4.0 DGA 2.0 (needs root)
- X11 XFree86 4.0 XVideo Extension (hardware yuv->rgb renderer)
Supported Inputs
- file,http.
- Supports on Linux Video CDs (vcd,cdi)
Download (0.70MB)
Added: 2005-07-07 License: GPL (GNU General Public License) Price:
1585 downloads
CELT 0.6.0
CELT is an experimental audio codec for use in low-delay communication. more>>
CELT 0.6.0 is created to be an experimental audio codec for use in low-delay communication. CELT stands for "Code-Excited Lapped Transform". It applies some of the CELP principles, but does everything in the frequency domain, which removes some of the limitations of CELP.
Major Features:
- Ultra-low latency (typically from 3 to 9 ms)
- Full audio bandwidth (44.1 kHz and 48 kHz)
- Stereo support
- Packet loss concealment
- Constant bit-rates from 32 kbps to 128 kbps and above
- A fixed-point version of the encoder and decoder
- The CELT codec is meant to close the gap between Vorbis and Speex for applications where both high quality audio and low delay are desired.
Enhancements:
- Has just been released, with many quality improvements, including better stereo coupling, better handling of transients, and better handling of highly tonal signals.
- Packet loss robustness has been improved through the optional use of independent (intra) frames.
- Supports a larger dynamic range, suitable for encoding 24-bit audio (float version only).
- There is also a very early VBR implementation.
Added: 2009-07-07 License: BSD License Price: FREE
13 downloads
JlayerME for Linux 0.13
improves VBRI VBR support more>> JLayerME is a JAVA library that decodes MP3 files in real-time. It supports MPEG 1/2 Layer 3 audio format only. It is oriented for J2ME platforms.
JLayerME is the J2ME-oriented version of JLayer. It only supports MPEG 1 Layer 3 (i.e. MP3). Release 0.1 is not J2ME full compliant. It runs under J2SE. The decoder is 10% faster and needs less memory than the JLayer classic one. Its the first step to make it runs under CDC/CVM (future) devices. Final goal is to make it runs under CLDC/KVM devices. This project might seem crazy ... yes it is
Unexpected background sound bug fixed.
BitReserve initialization bug fixed.
Changes.txt file added.
Two last seconds drop bug fixed.
JAR size improved : 46 KB (instead of 53KB).
MPEG 2 frames support added.
JVM 1.1 (IE + NS) support added.<<less
Download (148KB)
Added: 2009-04-01 License: Freeware Price: Free
205 downloads
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