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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
LAME 3.97
LAME is an MP3 encoder and graphical frame analyzer. more>>
LAME is short from LAME Aint an MP3 Encoder and is a research project for learning about and improving MP3 encoding technology. LAME includes an MP3 encoding library, simple frontend application, a much-improved psycho-acoustic model (GPSYCHO), and a graphical frame analyzer (MP3x).
Please note that any commercial use (including distributing the LAME encoding engine in a free encoder) may require a patent license from Thomson Multimedia.
Main features:
- Many improvements in quality in speed over ISO reference software. See history.
- MPEG1,2 and 2.5 layer III encoding.
- CBR (constant bitrate) and two types of variable bitrate, VBR and ABR.
- Encoding engine can be compiled as a shared library (Linux/UNIX), DLL or ACM codec (Windows)
- free format encoding and decoding
- GPSYCHO: a GPLd psycho acoustic and noise shaping model.
- Powerfull and easy to use presets
- Quality is comparable to FhG encoding engines and substantially better than most other encoders.
- Fast! Encodes faster than real time on a PII 266 at highest quality mode.
- MP3x: a GTK/X-Window MP3 frame analyzer for both .mp3 and unencoded audio files.
Software which uses "LAME":
- andromeda (PHP and ASP) Dynamically presents collections of mp3s as streaming web sites.
- rip (Perl) Script for ripping and encoding.
- avifile AVI/DIVX encoder and decoder for Linux.
- Grip (Linux) gtk-based cd-player, ripper and encoder. Supports cddb, cdparanoia and LAME.
- jbm2 (Linux) A KDE jukebox style application for public places (bars, pubs,...)
- Krabber (Linux) A KDE ripper & encoder, can use LAME.
- Mp3Maker (Linux) A WindowMaker enhanced front end to cdda2wav/cdparanoia and lame/bladeenc.
- dekagenc (Linux) Bourne shell script for ripping, encoding and CDDB naming.
- ripperX (Linux) GTK frontend for rippers and several encoders featuring CDDB support.
- T.E.A.R. (Linux) frontend to LAME, cdparanoia and CDDB.
- Xmcd. (Linux) CD Player with CDDB and ripping to MP3 and OGG.
- xtunes (Linux) GTK frontend for LAME, MAD, cdparanoia, cdrecord and more.
- DropMP3 (Mac) includes LAME binaries.
- CDex (Windows) Ripper & encoder, includes LAME binaries (the Blade compatible dll)
- Lamedrop (Windows) OggDrop style frontend.
- LAMEX (Windows) An activex control for LAME, and a GUI. Source code only, includes LAME.
- m3w (Windows) A live mp3 streamer for the WWW. Works with LAME, icecast, soundcard input
- out_lame (Windows) Winamp output plug-in. Create MP3 files directly from Winamp!
- RazorLame (Windows) The RazorBlade front end now supports LAME.
- winLAME (Windows) The only *nice* windows UI for LAME, according to the author :-)
- DarkIce Live streamer for IceCast.
- LiveIce Real time streaming of mp3s. Works with IceCast
- MuSE A mixing, encoding and streaming engine.
- Flash Forth a Flash-like development library
Enhancements:
- This version is identical to 3.97b3, which was promoted to release.
<<lessPlease note that any commercial use (including distributing the LAME encoding engine in a free encoder) may require a patent license from Thomson Multimedia.
Main features:
- Many improvements in quality in speed over ISO reference software. See history.
- MPEG1,2 and 2.5 layer III encoding.
- CBR (constant bitrate) and two types of variable bitrate, VBR and ABR.
- Encoding engine can be compiled as a shared library (Linux/UNIX), DLL or ACM codec (Windows)
- free format encoding and decoding
- GPSYCHO: a GPLd psycho acoustic and noise shaping model.
- Powerfull and easy to use presets
- Quality is comparable to FhG encoding engines and substantially better than most other encoders.
- Fast! Encodes faster than real time on a PII 266 at highest quality mode.
- MP3x: a GTK/X-Window MP3 frame analyzer for both .mp3 and unencoded audio files.
Software which uses "LAME":
- andromeda (PHP and ASP) Dynamically presents collections of mp3s as streaming web sites.
- rip (Perl) Script for ripping and encoding.
- avifile AVI/DIVX encoder and decoder for Linux.
- Grip (Linux) gtk-based cd-player, ripper and encoder. Supports cddb, cdparanoia and LAME.
- jbm2 (Linux) A KDE jukebox style application for public places (bars, pubs,...)
- Krabber (Linux) A KDE ripper & encoder, can use LAME.
- Mp3Maker (Linux) A WindowMaker enhanced front end to cdda2wav/cdparanoia and lame/bladeenc.
- dekagenc (Linux) Bourne shell script for ripping, encoding and CDDB naming.
- ripperX (Linux) GTK frontend for rippers and several encoders featuring CDDB support.
- T.E.A.R. (Linux) frontend to LAME, cdparanoia and CDDB.
- Xmcd. (Linux) CD Player with CDDB and ripping to MP3 and OGG.
- xtunes (Linux) GTK frontend for LAME, MAD, cdparanoia, cdrecord and more.
- DropMP3 (Mac) includes LAME binaries.
- CDex (Windows) Ripper & encoder, includes LAME binaries (the Blade compatible dll)
- Lamedrop (Windows) OggDrop style frontend.
- LAMEX (Windows) An activex control for LAME, and a GUI. Source code only, includes LAME.
- m3w (Windows) A live mp3 streamer for the WWW. Works with LAME, icecast, soundcard input
- out_lame (Windows) Winamp output plug-in. Create MP3 files directly from Winamp!
- RazorLame (Windows) The RazorBlade front end now supports LAME.
- winLAME (Windows) The only *nice* windows UI for LAME, according to the author :-)
- DarkIce Live streamer for IceCast.
- LiveIce Real time streaming of mp3s. Works with IceCast
- MuSE A mixing, encoding and streaming engine.
- Flash Forth a Flash-like development library
Enhancements:
- This version is identical to 3.97b3, which was promoted to release.
Download (1.3MB)
Added: 2006-09-24 License: GPL (GNU General Public License) Price:
1205 downloads
lame-rtp 2
This is a unified diff (you probably need GNU patch to apply it) to lame 3.58. more>>
This is a unified diff (you probably need GNU patch to apply it) to lame 3.58. Please note that this diff is now obsolete as recent beta versions of lame include a version of this code. Just type "make mp3rtp". The code was broken, but the CVS code is reported to work (and interoperate with playRTPMPEG) as of Feb 21 2000.
Please note that the output stream will only have the correct speed if the input is live recorded stream from your sound card or lame encodes exactly in real-time on your CPU!
RTP is the Realtime Transport Protocol as defined in RFC 1889. It is the transport mechanism of your choice to multicast an mp3 stream.
<<lessPlease note that the output stream will only have the correct speed if the input is live recorded stream from your sound card or lame encodes exactly in real-time on your CPU!
RTP is the Realtime Transport Protocol as defined in RFC 1889. It is the transport mechanism of your choice to multicast an mp3 stream.
Download (0.012MB)
Added: 2006-07-27 License: GPL (GNU General Public License) Price:
1189 downloads
Dynamic MP3 Lister
Dynamic MP3 Lister is a PHP script for downloading/streaming MP3s from a Web server. more>>
Dynamic MP3 Lister was a project I started a long time ago to create dynamic lists of MP3s quickly and easily.
Features MP3 Information extraction for things like bitrate, channels, playtime and more.
Please note that this script is discontinued, and is only shown here now as an example of my work.
<<lessFeatures MP3 Information extraction for things like bitrate, channels, playtime and more.
Please note that this script is discontinued, and is only shown here now as an example of my work.
Download (0.009MB)
Added: 2005-05-05 License: Free for non-commercial use Price:
1643 downloads
Bundle::MP3 1.00
Bundle::MP3 is a bundle to install all MP3-related modules. more>>
Bundle::MP3 is a bundle to install all MP3-related modules.
SYNOPSIS
perl -MCPAN -e install Bundle::MP3
CONTENTS
MP3::Info - For MP3::Info by Chris Sandor
Digest::MD5 - Used by MP3::Napster
MP3::Napster - For MP3::Napster by Lincoln Stein
Bundle::Xmms - For Xmms by Doug MacEachern
This bundle installs a bunch of modules related to sharing and playing MP3 files, including MP3::Info by Chris Sandor, MP3::Napster, by Lincoln Stein, and the Xmms bundle by Doug MacEachern.
<<lessSYNOPSIS
perl -MCPAN -e install Bundle::MP3
CONTENTS
MP3::Info - For MP3::Info by Chris Sandor
Digest::MD5 - Used by MP3::Napster
MP3::Napster - For MP3::Napster by Lincoln Stein
Bundle::Xmms - For Xmms by Doug MacEachern
This bundle installs a bunch of modules related to sharing and playing MP3 files, including MP3::Info by Chris Sandor, MP3::Napster, by Lincoln Stein, and the Xmms bundle by Doug MacEachern.
Download (MB)
Added: 2006-11-09 License: Perl Artistic License Price:
1082 downloads
Template::Plugin::MP3 1.02
Template::Plugin::MP3 is a Perl interface to the MP3::Info Module. more>>
Template::Plugin::MP3 is a Perl interface to the MP3::Info Module.
SYNOPSIS
[% USE mp3 = MP3("Montana.mp3") %]
[% mp3.title %]
[% mp3.album %]
# perldoc MP3::Info for more ideas
Template::Plugin::MP3 provides a simple wrapper for using MP3::Info in object oriented mode; see MP3::Info for more details.
<<lessSYNOPSIS
[% USE mp3 = MP3("Montana.mp3") %]
[% mp3.title %]
[% mp3.album %]
# perldoc MP3::Info for more ideas
Template::Plugin::MP3 provides a simple wrapper for using MP3::Info in object oriented mode; see MP3::Info for more details.
Download (0.018MB)
Added: 2006-11-08 License: Perl Artistic License Price:
1087 downloads
Stickloader 0.5
Stickloader is a LAME front end for quick filling of mp3 sticks. more>>
Stickloader is an easy solution for copying music files from your hard disk to your USB stick and re-encoding them at a lower bitrate for more efficient usage of your MP3 player disk space.
Stickloader is a LAME front end for quick filling of mp3 sticks.
MP3 files and whole directories can be easily dragged on the Stickloader window (which always stays on top) and they are automatically re-encoded using LAME and copied to your USB stick by using a temporary directory to avoid blocking the encoding process.
The programm is written with Java 5.0 using the Standard Widget Toolkit SWT and is using LAME for encoding the data. Therefore it should be system-independent and run on every system where Java, SWT and LAME are available.
<<lessStickloader is a LAME front end for quick filling of mp3 sticks.
MP3 files and whole directories can be easily dragged on the Stickloader window (which always stays on top) and they are automatically re-encoded using LAME and copied to your USB stick by using a temporary directory to avoid blocking the encoding process.
The programm is written with Java 5.0 using the Standard Widget Toolkit SWT and is using LAME for encoding the data. Therefore it should be system-independent and run on every system where Java, SWT and LAME are available.
Download (0.82MB)
Added: 2006-07-21 License: Freely Distributable Price:
1190 downloads
Data::ENAML 0.03
Data::ENAML is a Perl extension for ENAML data representation. more>>
Data::ENAML is a Perl extension for ENAML data representation.
SYNOPSIS
use Data::ENAML qw (serialize deserialize);
print serialize(login => {nick => Schop,
email => ariel@atheist.org.il,
tagline => If I had no modem I would not lose Regina});
$struct = deserialize(bad-nick: {nick: "c00l dewd" text: "spaces not allowed"});
ENAML stands for ENAML is Not A Markup Language. (And as we all know, Gnu is Not UNIX, Pine Is Not Email, Wine Is Not Emulator, Lame Aint Mp3 Encoder and so on).
ENAML was defined by Robey Pointer for use in Say2, check http://www.lag.net/say2.
<<lessSYNOPSIS
use Data::ENAML qw (serialize deserialize);
print serialize(login => {nick => Schop,
email => ariel@atheist.org.il,
tagline => If I had no modem I would not lose Regina});
$struct = deserialize(bad-nick: {nick: "c00l dewd" text: "spaces not allowed"});
ENAML stands for ENAML is Not A Markup Language. (And as we all know, Gnu is Not UNIX, Pine Is Not Email, Wine Is Not Emulator, Lame Aint Mp3 Encoder and so on).
ENAML was defined by Robey Pointer for use in Say2, check http://www.lag.net/say2.
Download (0.004MB)
Added: 2006-11-15 License: Perl Artistic License Price:
1073 downloads
BladeEnc 0.9.4.2
BladeEnc is a cross-platform MP3 encoder. more>>
BladeEnc is a freeware MP3 encoder. It is based on the same ISO compression routines as mpegEnc, so you can expect roughly the same, or better, quality . The main difference is the appearance and speed.
BladeEnc doesnt have a nice, user-friendly interface like mpegEnc, but it is more than three times faster, and it works with several popular front-end graphical user interfaces.
BladeEncs output quality is one of those rare subjects that completely divides the world in two parts. Either you love it or you hate it, there never seems to be an opinion inbetween. Different audiophiles and mp3 experts tends to come to completely different conclusions depending on their methods and testsamples.
The reason for this is of course that BladeEnc is a very different mp3 encoder (compared to Fraunhofer, LAME etc) with a very unique approach to mp3 encoding.
In order to compress sound to an mp3 file, you need to make certain sacrifices in quality. Taking into account how we percieve sound, the mp3 encoder tries to remove the details that it believes us to be least likely to notice. How much that needs to be removed depends on the bitrate and the encoder often has the choice of doing different kinds of sacrifices.
It can remove low volume tones that are "shadowed" by high volume tones of similar frequencies, remove the high frequency part of the sound spectrum, cut down the stereo effect (so called joint stereo) and simply decrease the samplerate. What approach is the best depends on a lot of things, like the style of music and the selected bitrate.
Main features:
- Sourcecode available under the LGPL-license!
- Stereo or Mono output. Can downmix to Mono on the fly.
- Supports the following bitrates: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kBit/s. However, for bitrates lower than 128 kBit we seriously recommend you to use another encoder.
- Flags like Private, Original and Copyright can be set.
- Input samples can be in either 32, 44.1 or 48 kHz.
- Both 8 and 16-bit samples are supported.
- Working CRC checksum generation (since 0.80). The ISO reference code had broken CRC calculations, which has been inherited into every ISO based encoder that havent added a fix for it.
- Can be plugged directly into many popular 3rd party products, giving them integrated mp3 encoding abilities.
- Encodes chunks of data from memory to memory, no need to use files or pipes.
- Can be compiled for nearly any operating system still in use.
- Commandline based, makes it easy to include BladeEnc into BAT files and shell scripts.
- Only mp3 encoder that supports gapless encoding.
- Reads standard uncompressed WAV- and AIFF-files as well as well as RAW PCM-data.
- Batch encoding. Can encode any number of samples in a row.
- Wildcards supported. You can for example encode all WAV-files in a directory by typing *.WAV".
- Input samples can be automatically deleted after encoding.
- Large selection of graphical frontends available.
- Task priority can be set from the commandline and is by default set to LOWEST so that you still can use your computer effectively while encoding (Windows & OS/2 only).
- Full support for pipes and redirection (stdin and stdout).
- Textbased configuration file where you can change default settings.
<<lessBladeEnc doesnt have a nice, user-friendly interface like mpegEnc, but it is more than three times faster, and it works with several popular front-end graphical user interfaces.
BladeEncs output quality is one of those rare subjects that completely divides the world in two parts. Either you love it or you hate it, there never seems to be an opinion inbetween. Different audiophiles and mp3 experts tends to come to completely different conclusions depending on their methods and testsamples.
The reason for this is of course that BladeEnc is a very different mp3 encoder (compared to Fraunhofer, LAME etc) with a very unique approach to mp3 encoding.
In order to compress sound to an mp3 file, you need to make certain sacrifices in quality. Taking into account how we percieve sound, the mp3 encoder tries to remove the details that it believes us to be least likely to notice. How much that needs to be removed depends on the bitrate and the encoder often has the choice of doing different kinds of sacrifices.
It can remove low volume tones that are "shadowed" by high volume tones of similar frequencies, remove the high frequency part of the sound spectrum, cut down the stereo effect (so called joint stereo) and simply decrease the samplerate. What approach is the best depends on a lot of things, like the style of music and the selected bitrate.
Main features:
- Sourcecode available under the LGPL-license!
- Stereo or Mono output. Can downmix to Mono on the fly.
- Supports the following bitrates: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kBit/s. However, for bitrates lower than 128 kBit we seriously recommend you to use another encoder.
- Flags like Private, Original and Copyright can be set.
- Input samples can be in either 32, 44.1 or 48 kHz.
- Both 8 and 16-bit samples are supported.
- Working CRC checksum generation (since 0.80). The ISO reference code had broken CRC calculations, which has been inherited into every ISO based encoder that havent added a fix for it.
- Can be plugged directly into many popular 3rd party products, giving them integrated mp3 encoding abilities.
- Encodes chunks of data from memory to memory, no need to use files or pipes.
- Can be compiled for nearly any operating system still in use.
- Commandline based, makes it easy to include BladeEnc into BAT files and shell scripts.
- Only mp3 encoder that supports gapless encoding.
- Reads standard uncompressed WAV- and AIFF-files as well as well as RAW PCM-data.
- Batch encoding. Can encode any number of samples in a row.
- Wildcards supported. You can for example encode all WAV-files in a directory by typing *.WAV".
- Input samples can be automatically deleted after encoding.
- Large selection of graphical frontends available.
- Task priority can be set from the commandline and is by default set to LOWEST so that you still can use your computer effectively while encoding (Windows & OS/2 only).
- Full support for pipes and redirection (stdin and stdout).
- Textbased configuration file where you can change default settings.
Download (0.05MB)
Added: 2005-05-10 License: LGPL (GNU Lesser General Public License) Price:
2371 downloads
FLAC 1.1.4
FLAC stands for Free Lossless Audio Codec. more>>
FLAC is Free Lossless Audio Codec. Grossly oversimplified, FLAC is similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality.
This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see supported devices) just like you would an MP3 file.
FLAC is freely available and supported on most operating systems, including Windows, "unix" (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.
The FLAC project consists of:
the stream format
reference encoders and decoders in library form
flac, a command-line program to encode and decode FLAC files
metaflac, a command-line metadata editor for FLAC files
input plugins for various music players
When we say that FLAC is "Free" it means more than just that it is available at no cost. It means that the specification of the format is fully open to the public to be used for any purpose (the FLAC project reserves the right to set the FLAC specification and certify compliance), and that neither the FLAC format nor any of the implemented encoding/decoding methods are covered by any known patent. It also means that all the source code is available under open-source licenses. It is the first truly open and free lossless audio format.
Main features:
- Lossless: The encoding of audio (PCM) data incurs no loss of information, and the decoded audio is bit-for-bit identical to what went into the encoder. Each frame contains a 16-bit CRC of the frame data for detecting transmission errors. The integrity of the audio data is further insured by storing an MD5 signature of the original unencoded audio data in the file header, which can be compared against later during decoding or testing.
- Fast: FLAC is asymmetric in favor of decode speed. Decoding requires only integer arithmetic, and is much less compute-intensive than for most perceptual codecs. Real-time decode performance is easily achievable on even modest hardware.
- Hardware support: Because of FLACs free reference implementation and low decoding complexity, FLAC is currently the only lossless codec that has any kind of hardware support.
- Streamable: Each FLAC frame contains enough data to decode that frame. FLAC does not even rely on previous or following frames. FLAC uses sync codes and CRCs (similar to MPEG and other formats), which, along with framing, allow decoders to pick up in the middle of a stream with a minimum of delay.
- Seekable: FLAC supports fast sample-accurate seeking. Not only is this useful for playback, it makes FLAC files suitable for use in editing applications.
- Flexible metadata: New metadata blocks can be defined and implemented in future versions of FLAC without breaking older streams or decoders. Currently there are metadata types for tags, cue sheets, and seek tables. Applications can write their own APPLICATION metadata once they register an ID
- Suitable for archiving: FLAC is an open format, and there is no generation loss if you need to convert your data to another format in the future. In addition to the frame CRCs and MD5 signature, flac has a verify option that decodes the encoded stream in parallel with the encoding process and compares the result to the original, aborting with an error if there is a mismatch.
- Convenient CD archiving: FLAC has a "cue sheet" metadata block for storing a CD table of contents and all track and index points. For instance, you can rip a CD to a single file, then import the CDs extracted cue sheet while encoding to yield a single file representation of the entire CD. If your original CD is damaged, the cue sheet can be exported later in order to burn an exact copy.
- Error resistant: Because of FLACs framing, stream errors limit the damage to the frame in which the error occurred, typically a small fraction of a second worth of data. Contrast this with some other lossless codecs, in which a single error destroys the remainder of the stream.
What FLAC is not:
- Lossy. FLAC is intended for lossless compression only, as there are many good lossy formats already, such as Vorbis, MPC, and MP3 (see LAME for an excellent open-source implementation).
- SDMI compliant, et cetera. There is no intention to support any methods of copy protection, which are, for all practical purposes, a complete waste of bits. (Another way to look at it is that since copy protection is futile, it really carries no information, so you might say FLAC already losslessly compresses all possible copy protection information down to zero bits!) Of course, we cant stop what some misguided person does with proprietary metadata blocks, but then again, non-proprietary decoders will skip them anyway.
<<lessThis is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see supported devices) just like you would an MP3 file.
FLAC is freely available and supported on most operating systems, including Windows, "unix" (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.
The FLAC project consists of:
the stream format
reference encoders and decoders in library form
flac, a command-line program to encode and decode FLAC files
metaflac, a command-line metadata editor for FLAC files
input plugins for various music players
When we say that FLAC is "Free" it means more than just that it is available at no cost. It means that the specification of the format is fully open to the public to be used for any purpose (the FLAC project reserves the right to set the FLAC specification and certify compliance), and that neither the FLAC format nor any of the implemented encoding/decoding methods are covered by any known patent. It also means that all the source code is available under open-source licenses. It is the first truly open and free lossless audio format.
Main features:
- Lossless: The encoding of audio (PCM) data incurs no loss of information, and the decoded audio is bit-for-bit identical to what went into the encoder. Each frame contains a 16-bit CRC of the frame data for detecting transmission errors. The integrity of the audio data is further insured by storing an MD5 signature of the original unencoded audio data in the file header, which can be compared against later during decoding or testing.
- Fast: FLAC is asymmetric in favor of decode speed. Decoding requires only integer arithmetic, and is much less compute-intensive than for most perceptual codecs. Real-time decode performance is easily achievable on even modest hardware.
- Hardware support: Because of FLACs free reference implementation and low decoding complexity, FLAC is currently the only lossless codec that has any kind of hardware support.
- Streamable: Each FLAC frame contains enough data to decode that frame. FLAC does not even rely on previous or following frames. FLAC uses sync codes and CRCs (similar to MPEG and other formats), which, along with framing, allow decoders to pick up in the middle of a stream with a minimum of delay.
- Seekable: FLAC supports fast sample-accurate seeking. Not only is this useful for playback, it makes FLAC files suitable for use in editing applications.
- Flexible metadata: New metadata blocks can be defined and implemented in future versions of FLAC without breaking older streams or decoders. Currently there are metadata types for tags, cue sheets, and seek tables. Applications can write their own APPLICATION metadata once they register an ID
- Suitable for archiving: FLAC is an open format, and there is no generation loss if you need to convert your data to another format in the future. In addition to the frame CRCs and MD5 signature, flac has a verify option that decodes the encoded stream in parallel with the encoding process and compares the result to the original, aborting with an error if there is a mismatch.
- Convenient CD archiving: FLAC has a "cue sheet" metadata block for storing a CD table of contents and all track and index points. For instance, you can rip a CD to a single file, then import the CDs extracted cue sheet while encoding to yield a single file representation of the entire CD. If your original CD is damaged, the cue sheet can be exported later in order to burn an exact copy.
- Error resistant: Because of FLACs framing, stream errors limit the damage to the frame in which the error occurred, typically a small fraction of a second worth of data. Contrast this with some other lossless codecs, in which a single error destroys the remainder of the stream.
What FLAC is not:
- Lossy. FLAC is intended for lossless compression only, as there are many good lossy formats already, such as Vorbis, MPC, and MP3 (see LAME for an excellent open-source implementation).
- SDMI compliant, et cetera. There is no intention to support any methods of copy protection, which are, for all practical purposes, a complete waste of bits. (Another way to look at it is that since copy protection is futile, it really carries no information, so you might say FLAC already losslessly compresses all possible copy protection information down to zero bits!) Of course, we cant stop what some misguided person does with proprietary metadata blocks, but then again, non-proprietary decoders will skip them anyway.
Download (1.9MB)
Added: 2007-02-14 License: GPL (GNU General Public License) Price:
994 downloads
Apache::MP3::Skin 0.91
Apache::MP3::Skin is a subclass of Apache::MP3::Playlist with the ability to skin the output using HTML::Template. more>>
Apache::MP3::Skin is a subclass of Apache::MP3::Playlist with the ability to "skin" the output using HTML::Template.
SYNOPSIS
# httpd.conf or srm.conf
AddType audio/mpeg .mp3 .MP3
# httpd.conf or access.conf
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Skin
PerlSetVar HomePath /songs # optional
PerlSetVar DefaultSkin default.tmpl # required
# Without DefaultSkin being set to a valid file
# Apache::MP3::Skin will be the same as Apache::MP3
< /Location >
Apache::MP3::Skin subclasses Apache::MP3::Playlist enabling the use of skin files which are html files with special tags enabled by HTML::Template. See Apache::MP3 for details on installing and using.
<<lessSYNOPSIS
# httpd.conf or srm.conf
AddType audio/mpeg .mp3 .MP3
# httpd.conf or access.conf
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Skin
PerlSetVar HomePath /songs # optional
PerlSetVar DefaultSkin default.tmpl # required
# Without DefaultSkin being set to a valid file
# Apache::MP3::Skin will be the same as Apache::MP3
< /Location >
Apache::MP3::Skin subclasses Apache::MP3::Playlist enabling the use of skin files which are html files with special tags enabled by HTML::Template. See Apache::MP3 for details on installing and using.
Download (0.018MB)
Added: 2006-10-12 License: Perl Artistic License Price:
1110 downloads
MP3::ID3Lib::Frame 0.12
MP3::ID3Lib::Frame is an ID3v1/ID3v2 Tagging Frames. more>>
MP3::ID3Lib::Frame is an ID3v1/ID3v2 Tagging Frames.
SYNOPSIS
use MP3::ID3Lib;
my $id3 = MP3::ID3Lib->new($filename);
foreach my $frame (@{$id3->frames}) {
my $code = $frame->code;
my $description = $frame->description;
my $value = $frame->value;
$frame->set("Orange") if $code eq COMM;
print "$description: $valuen";
}
$id3->commit;
See MP3::ID3Lib for more information.
<<lessSYNOPSIS
use MP3::ID3Lib;
my $id3 = MP3::ID3Lib->new($filename);
foreach my $frame (@{$id3->frames}) {
my $code = $frame->code;
my $description = $frame->description;
my $value = $frame->value;
$frame->set("Orange") if $code eq COMM;
print "$description: $valuen";
}
$id3->commit;
See MP3::ID3Lib for more information.
Download (0.037MB)
Added: 2006-11-09 License: Perl Artistic License Price:
1079 downloads
Osalp 0.7.3
Osalp is a project designed to implement a world class set of classes in C++ that will handle all of the audio functions. more>>
Osalp is a project designed to implement a world class set of classes in C++ that will handle all of the audio functions one would like. It is designed to be multi-platform with UNIX based platforms as the base.
This project is still in the beta code phase and a beta version that will illustrate the power and flexibility is now available. This version supports the Linux (OSS) audio device, Solaris Sparc audio device, FreeBSD (OSS) audio device, wav, au, aiff, aifc, mp3, and numerousother formats.
It is important to note that this is not an application but a C++ library that others can use to create an audio application or to easly add audio capabilities to an existing application. OSALP was originally designed and developed by Bruce Forsberg out of need to manage and edit large sound files in a simple manner. Currently the project is being maintained by Darrick Servis.
The library is built on a set of core classes that provide the basic functionality. New classes to operate on data are derived from these classes. These classes provide a powerful chaining process. This allows one to build an audio chain much like one would build with building blocks. Audio data is encapsulated into a single class. This allows one to handle data conversions in one place.
There is a file base class (aflibFile) that defines the API for any device or file classes that are to be developed. They are implemented as dynamically loaded shared objects so that new file types can be added without recompiling the base library or needing to link them to an application. This will allow third parties to support their proprietary formats as a binary "plugable modules".
Linux & FreeBSD Device (OSS) -- aflibDevFile
Solaris Sparc Device -- aflibSolarisSparcDevFile
WAV (linear, mu-law, a-law) -- aflibWavFile
AU (linear, mu-law, a-law) -- aflibAuFile
AIFC -- aflibAifcFile
AIFF -- aflibAiffFile
MP3 using Lame encoder -- aflibLameFile
MP3 using Blade encoder -- aflibBladeFile
MP3 reader using splay library -- aflibMpgFile
MP3 reader using mpg123 executable -- aflibMpg123File
Sox library interface (supports most formats supported by the sox sound tools library -- aflibSoxFile
Currently there are several worker classes. These are the classes that actually do the work. These classes are not tied to any GUI but are GUI neutral. This allows developers to write code using the GUI of their choice.
Audio Sample Rate Converter -- aflibAudioSampleRateCvt
Audio Pitch Change -- aflibAudioPitch
Audio Test Source -- aflibAudioConstantSrc
Audio Editing -- aflibAudioEdit
Audio Timer Recording -- aflibAudioRecorder
Audio VU Meter and Spectrum Display -- aflibAudioSpectrum
Audio Mixing -- aflibAudioMixer
Butterworth Filter -- aflibAudioBWFilter
Reading Audio Data from Memory -- aflibAudioMemoryInput
Reading and Writing Audio Data to Devices or Files -- aflibAudioFile
There are also utility classes. These are not part of the main audio chain but are probably needed by most audio applications or are used indirectly by the worker classes.
FFT -- aflibFFT
User Environment Storage and Retrieval -- aflibEnvFile
Audio sample data -- aflibSampleData
Sample rate conversion -- aflibConverter
<<lessThis project is still in the beta code phase and a beta version that will illustrate the power and flexibility is now available. This version supports the Linux (OSS) audio device, Solaris Sparc audio device, FreeBSD (OSS) audio device, wav, au, aiff, aifc, mp3, and numerousother formats.
It is important to note that this is not an application but a C++ library that others can use to create an audio application or to easly add audio capabilities to an existing application. OSALP was originally designed and developed by Bruce Forsberg out of need to manage and edit large sound files in a simple manner. Currently the project is being maintained by Darrick Servis.
The library is built on a set of core classes that provide the basic functionality. New classes to operate on data are derived from these classes. These classes provide a powerful chaining process. This allows one to build an audio chain much like one would build with building blocks. Audio data is encapsulated into a single class. This allows one to handle data conversions in one place.
There is a file base class (aflibFile) that defines the API for any device or file classes that are to be developed. They are implemented as dynamically loaded shared objects so that new file types can be added without recompiling the base library or needing to link them to an application. This will allow third parties to support their proprietary formats as a binary "plugable modules".
Linux & FreeBSD Device (OSS) -- aflibDevFile
Solaris Sparc Device -- aflibSolarisSparcDevFile
WAV (linear, mu-law, a-law) -- aflibWavFile
AU (linear, mu-law, a-law) -- aflibAuFile
AIFC -- aflibAifcFile
AIFF -- aflibAiffFile
MP3 using Lame encoder -- aflibLameFile
MP3 using Blade encoder -- aflibBladeFile
MP3 reader using splay library -- aflibMpgFile
MP3 reader using mpg123 executable -- aflibMpg123File
Sox library interface (supports most formats supported by the sox sound tools library -- aflibSoxFile
Currently there are several worker classes. These are the classes that actually do the work. These classes are not tied to any GUI but are GUI neutral. This allows developers to write code using the GUI of their choice.
Audio Sample Rate Converter -- aflibAudioSampleRateCvt
Audio Pitch Change -- aflibAudioPitch
Audio Test Source -- aflibAudioConstantSrc
Audio Editing -- aflibAudioEdit
Audio Timer Recording -- aflibAudioRecorder
Audio VU Meter and Spectrum Display -- aflibAudioSpectrum
Audio Mixing -- aflibAudioMixer
Butterworth Filter -- aflibAudioBWFilter
Reading Audio Data from Memory -- aflibAudioMemoryInput
Reading and Writing Audio Data to Devices or Files -- aflibAudioFile
There are also utility classes. These are not part of the main audio chain but are probably needed by most audio applications or are used indirectly by the worker classes.
FFT -- aflibFFT
User Environment Storage and Retrieval -- aflibEnvFile
Audio sample data -- aflibSampleData
Sample rate conversion -- aflibConverter
Download (1.0MB)
Added: 2006-02-15 License: LGPL (GNU Lesser General Public License) Price:
1348 downloads
cdmp3 0.5.0
cdmp3 uses cdparanoia or cdda2wav to extract single tracks or even a whole CD and converts them on-the-fly into MP3 or Ogg Vorbi more>>
cdmp3 uses cdparanoia or cdda2wav to extract single tracks or even a whole CD and converts them on-the-fly into MP3 or Ogg Vorbis files. Using the CDDB database it is possible to include information like the artists name or the tracks title into the file name or as meta tags into the file content.
cdmp3 consists of a single perl script. Copy it to where ever you want it to reside, /usr/local/bin might be a good place. You might also want to create a symlink, named "cdogg", which points to cdmp3, so that by running cdogg, you preset cdmp3 to creating Ogg Vorbis files instead of MP3.
Additionally you will need the next packages, but this depends on the command line arguments you give to cdmp3:
cdparanoia http://www.xiph.org/paranoia/
cdda2wav ftp://ftp.berlios.de/pub/cdrecord/ (package cdrtools)
lame http://www.mp3dev.org/
oggenc http://www.vorbis.com (package vorbis-tools)
CDDB_get perl module http://armin.emx.at/cddb/
bfr http://www.glines.org:8000/software/buffer.html
If you are running Debian, all these packages are provided by the Debian
project itself or by third parties.
Enhancements:
- converted cdmp3 to perl
- merged get_cddb into cdmp3 main script
- added "--print-cddb": fetch and print cddb entry, exit
- added %y and %g filename format patterns
- support two-digit track numbers (%0n and %0u)
- patterns are really handled printf(1)-like now
- more flexible command line arguments
<<lesscdmp3 consists of a single perl script. Copy it to where ever you want it to reside, /usr/local/bin might be a good place. You might also want to create a symlink, named "cdogg", which points to cdmp3, so that by running cdogg, you preset cdmp3 to creating Ogg Vorbis files instead of MP3.
Additionally you will need the next packages, but this depends on the command line arguments you give to cdmp3:
cdparanoia http://www.xiph.org/paranoia/
cdda2wav ftp://ftp.berlios.de/pub/cdrecord/ (package cdrtools)
lame http://www.mp3dev.org/
oggenc http://www.vorbis.com (package vorbis-tools)
CDDB_get perl module http://armin.emx.at/cddb/
bfr http://www.glines.org:8000/software/buffer.html
If you are running Debian, all these packages are provided by the Debian
project itself or by third parties.
Enhancements:
- converted cdmp3 to perl
- merged get_cddb into cdmp3 main script
- added "--print-cddb": fetch and print cddb entry, exit
- added %y and %g filename format patterns
- support two-digit track numbers (%0n and %0u)
- patterns are really handled printf(1)-like now
- more flexible command line arguments
Download (0.013MB)
Added: 2006-07-26 License: GPL (GNU General Public License) Price:
1185 downloads
Apache::MP3::Sorted 4.00
Apache::MP3::Sorted is a Perl module to generate sorted streamable directories of MP3 files. more>>
Apache::MP3::Sorted is a Perl module to generate sorted streamable directories of MP3 files.
SYNOPSIS
# httpd.conf or srm.conf
AddType audio/mpeg mp3 MP3
# httpd.conf or access.conf
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Sorted
PerlSetVar SortFields Album,Title,-Duration
PerlSetVar Fields Title,Artist,Album,Duration
< /Location >
Apache::MP3::Sorted subclasses Apache::MP3 to allow for sorting of MP3 listings by various criteria. See Apache::MP3 for details on installing and using.
<<lessSYNOPSIS
# httpd.conf or srm.conf
AddType audio/mpeg mp3 MP3
# httpd.conf or access.conf
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Sorted
PerlSetVar SortFields Album,Title,-Duration
PerlSetVar Fields Title,Artist,Album,Duration
< /Location >
Apache::MP3::Sorted subclasses Apache::MP3 to allow for sorting of MP3 listings by various criteria. See Apache::MP3 for details on installing and using.
Download (0.32MB)
Added: 2006-11-07 License: Perl Artistic License Price:
1082 downloads
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