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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
Subsonic 3.1
Subsonic is a Web-based media streamer. more>>
Subsonic is a free, web-based media streamer. Although primarily designed for MP3, it works for any audio or video format that can stream over HTTP (for instance WMA and OGG).
With Subsonic you have access to your entire music collection wherever you are. Use Subsonic application to share your music with friends, or to listen to your own music while at work. You can stream to multiple players simultaneously, for instance to one player in your kitchen and another in your living room.
Subsonic is designed to handle very large music collections (many thousand albums). It uses a combination of directory structure and tag parsing to organize the music.
If you have constrained bandwidth, you may set an upper limit for the bitrate of the music streams. Subsonic will then automatically resample the music to a suitable bitrate. (Supported for MP3 only).
In addition to being a streaming media server, Subsonic works very well as a local jukebox. The intuitive web interface, as well as search and index facilities, are optimized for efficient browsing through large media libraries.
Subsonic is free software distributed under the LGPL open-source licence.
Main features:
- Supports MP3, OGG, AAC, WMA and any other music or video format that can be streamed over HTTP.
- Works with any network-enabled media player, such as Winamp, iTunes, XMMS, MusicMatch and Windows Media Player.
- Displays cover art, dynamically scaling it to fit nicely on the screen.
- Finds cover art and album info using Amazon Web Services.
- On-the-fly resampling to lower bitrates using the high-quality LAME encoder (MP3 only).
- Common playlist features (add, remove, rearrange, repeat, shuffle, undo).
- Playlists can be saved and restored. Both M3U and PLS formats are supported.
- Configurable index.
- Google-like search functionality.
- Automatic update of search index at specified intervals.
- Supports multiple simultaneous players. The playlist for any player can be managed from any location.
- Secure. Users must log in using a username and password.
- Lean web interface optimized for constrained bandwidth environments and efficient browsing through large music collections.
- WAP interface allows you to control Subsonic from any mobile phone or PDA.
- In addition to streaming, single files or entire directories may be downloaded from Subsonic.
- Files can be uploaded to Subsonic. Zip-files can be automatically unpacked.
- Implements the SHOUTcast protocol. Players which support this (including Winamp, iTunes and XMMS) will display the current artist and song name.
- Provides links to album reviews and more at allmusic.com.
- Supports internationalization. Currently, English and Norwegian translations are provided.
Enhancements:
- New: Optimized scanning of music folders. Dramatic improvement for network disks.
- Bugfix: Fixed minor concurrency bug in search index creation.
- Bugfix: Make cache work if clocks on remote disks are out of sync.
- Bugfix: Ensure that cover art images are ordered alphabetically.
- Bugfix: Remove nag message for licensed users.
<<lessWith Subsonic you have access to your entire music collection wherever you are. Use Subsonic application to share your music with friends, or to listen to your own music while at work. You can stream to multiple players simultaneously, for instance to one player in your kitchen and another in your living room.
Subsonic is designed to handle very large music collections (many thousand albums). It uses a combination of directory structure and tag parsing to organize the music.
If you have constrained bandwidth, you may set an upper limit for the bitrate of the music streams. Subsonic will then automatically resample the music to a suitable bitrate. (Supported for MP3 only).
In addition to being a streaming media server, Subsonic works very well as a local jukebox. The intuitive web interface, as well as search and index facilities, are optimized for efficient browsing through large media libraries.
Subsonic is free software distributed under the LGPL open-source licence.
Main features:
- Supports MP3, OGG, AAC, WMA and any other music or video format that can be streamed over HTTP.
- Works with any network-enabled media player, such as Winamp, iTunes, XMMS, MusicMatch and Windows Media Player.
- Displays cover art, dynamically scaling it to fit nicely on the screen.
- Finds cover art and album info using Amazon Web Services.
- On-the-fly resampling to lower bitrates using the high-quality LAME encoder (MP3 only).
- Common playlist features (add, remove, rearrange, repeat, shuffle, undo).
- Playlists can be saved and restored. Both M3U and PLS formats are supported.
- Configurable index.
- Google-like search functionality.
- Automatic update of search index at specified intervals.
- Supports multiple simultaneous players. The playlist for any player can be managed from any location.
- Secure. Users must log in using a username and password.
- Lean web interface optimized for constrained bandwidth environments and efficient browsing through large music collections.
- WAP interface allows you to control Subsonic from any mobile phone or PDA.
- In addition to streaming, single files or entire directories may be downloaded from Subsonic.
- Files can be uploaded to Subsonic. Zip-files can be automatically unpacked.
- Implements the SHOUTcast protocol. Players which support this (including Winamp, iTunes and XMMS) will display the current artist and song name.
- Provides links to album reviews and more at allmusic.com.
- Supports internationalization. Currently, English and Norwegian translations are provided.
Enhancements:
- New: Optimized scanning of music folders. Dramatic improvement for network disks.
- Bugfix: Fixed minor concurrency bug in search index creation.
- Bugfix: Make cache work if clocks on remote disks are out of sync.
- Bugfix: Ensure that cover art images are ordered alphabetically.
- Bugfix: Remove nag message for licensed users.
Download (11MB)
Added: 2007-08-03 License: LGPL (GNU Lesser General Public License) Price:
841 downloads
iTunes 0.86
iTunes is a Music Library File Format. more>>
iTunes is a Music Library File Format.
SYNOPSIS
This file describes the iTunes Music Library file format for application versions 1.1.2 and 3.0.
The iTunes Music Library file contains a series of variable length blocks. The first part of the file refers to iTunes in general and to the number of playlists the file contains.
The middle part holds all of the MP3 data, which is only stored once per item. Each item has a unique identifier used later in the file. The last part of the file contains a list of playlists, and within each playlist is a list of the item identifiers from the middle of the file.
Markers
Each block of data begins with a four letter marker which denotes the type of block. Some blocks contain sub-blocks, which means you cannot simply read to the next marker.
The blocks show up roughly in this order:
hdfm -- file header
hdsm -- part header ?
htlm -- master items list header ?
htim -- master item record
hohm -- data sublock, multi-purpose (in htim, hpim blocks)
hplm -- playlist list header ?
hpim -- start of playlist block
hptm -- item indentifier sub-block for hpim
<<lessSYNOPSIS
This file describes the iTunes Music Library file format for application versions 1.1.2 and 3.0.
The iTunes Music Library file contains a series of variable length blocks. The first part of the file refers to iTunes in general and to the number of playlists the file contains.
The middle part holds all of the MP3 data, which is only stored once per item. Each item has a unique identifier used later in the file. The last part of the file contains a list of playlists, and within each playlist is a list of the item identifiers from the middle of the file.
Markers
Each block of data begins with a four letter marker which denotes the type of block. Some blocks contain sub-blocks, which means you cannot simply read to the next marker.
The blocks show up roughly in this order:
hdfm -- file header
hdsm -- part header ?
htlm -- master items list header ?
htim -- master item record
hohm -- data sublock, multi-purpose (in htim, hpim blocks)
hplm -- playlist list header ?
hpim -- start of playlist block
hptm -- item indentifier sub-block for hpim
Download (0.11MB)
Added: 2006-11-17 License: Perl Artistic License Price:
597729 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
Today, LAME is considered the best MP3 encoder at mid-high bitrates and at ... Today, LAME is considered the best MP3 encoder at mid-high bitrates and at VBR, mostly thanksLicense:Freeware
POE::Component::Enc::Mp3 1.2
POE::Component::Enc::Mp3 is a mp3 encoder wrapper. more>>
POE::Component::Enc::Mp3 is a mp3 encoder wrapper.
SYNOPSIS
use POE qw(Component::Enc::Mp3);
$mp3 = POE::Component::Enc::Mp3->new($bitrate => 160);
$mp3->enc("/tmp/tst.wav");
POE::Kernel->run();
This POE component encodes raw audio files into mp3 format. It is merely a wrapper for the notlame program.
METHODS
The module provides an object oriented interface as follows:
new
Used to initialise the system and create a module instance. The following parameters are available:
alias
Indicates the name of a session to which module callbacks are posted. Default: main.
bitrate
Should be self-evident. If left unspecified, defaults to 160.
enc < file-name > [del-orig]
Encodes the given file, naming it with a .mp3 extension. An optional true value for the second parameter indicates that the original file should be deleted.
e.g. $mp3->enc("/tmp/tst.wav");
<<lessSYNOPSIS
use POE qw(Component::Enc::Mp3);
$mp3 = POE::Component::Enc::Mp3->new($bitrate => 160);
$mp3->enc("/tmp/tst.wav");
POE::Kernel->run();
This POE component encodes raw audio files into mp3 format. It is merely a wrapper for the notlame program.
METHODS
The module provides an object oriented interface as follows:
new
Used to initialise the system and create a module instance. The following parameters are available:
alias
Indicates the name of a session to which module callbacks are posted. Default: main.
bitrate
Should be self-evident. If left unspecified, defaults to 160.
enc < file-name > [del-orig]
Encodes the given file, naming it with a .mp3 extension. An optional true value for the second parameter indicates that the original file should be deleted.
e.g. $mp3->enc("/tmp/tst.wav");
Download (0.003MB)
Added: 2006-11-07 License: Perl Artistic License Price:
1081 downloads
WideMP3 1.0-beta1
WideMP3 is a relatively simple distributed MP3 encoder. more>>
WideMP3 is a relatively simple distributed MP3 encoder. It works in a client-server fashion, with multiple servers and a client that feeds the PCM audio over a network connection to the server. The server sends back the encoded audio, which the client then writes to a file. The actual encoding is done via an external encoder, such as BladeEnc.
Installation is very simple simple.
Extract the source distribution, and compile and install as follows:
./configure
make
su
make install
Current supported platforms are:
Debian GNU/Linux 2.2 on Intel x86 and Alpha.
Debian GNU/Linux 2.1 on Intel x86
OpenBSD 2.7 on Intel x86
It should also work on FreeBSD, although this has not yet been tested. Positive and negative reports on different platforms will be appreciated, especially if accompanied by a patch to make it work.
If you want to contribute a patch, you must be willing for the patch to be released under the GNU General Public License. Patches should be in context diff format (diff -c) or unified diff format (diff -u).
Currently supported MP3 encoders:
Bladeenc 0.9x - http://bladeenc.mp3.no/
Positive and negative reports for different encoders will be appreciated, especially if accompanied by a patch to make it work.
<<lessInstallation is very simple simple.
Extract the source distribution, and compile and install as follows:
./configure
make
su
make install
Current supported platforms are:
Debian GNU/Linux 2.2 on Intel x86 and Alpha.
Debian GNU/Linux 2.1 on Intel x86
OpenBSD 2.7 on Intel x86
It should also work on FreeBSD, although this has not yet been tested. Positive and negative reports on different platforms will be appreciated, especially if accompanied by a patch to make it work.
If you want to contribute a patch, you must be willing for the patch to be released under the GNU General Public License. Patches should be in context diff format (diff -c) or unified diff format (diff -u).
Currently supported MP3 encoders:
Bladeenc 0.9x - http://bladeenc.mp3.no/
Positive and negative reports for different encoders will be appreciated, especially if accompanied by a patch to make it work.
Download (0.062MB)
Added: 2006-07-20 License: GPL (GNU General Public License) Price:
1191 downloads
Apache::MP3::Resample 4.00
Apache::MP3::Resample can downsample MP3/FLAC/Shorten files during streaming. more>>
Apache::MP3::Resample can downsample MP3/FLAC/Shorten files during streaming.
SYNOPSIS
# httpd.conf or access.conf
AddType audio/shorten .shn .SHN
AddType audio/flac .flac
Alias /apache_mp3 /usr/share/libapache-mp3-perl
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Resample
PerlSetVar CacheDir /var/cache/Apache::MP3
PerlSetVar AllowDownload no
PerlSetVar SortFields Album,Track,Title,-Duration
PerlSetVar Fields Track,Title,Artist,Album,Duration,Bitrate
< /Location >
Apache::MP3::Resample subclasses Apache::MP3::Playlist to allow the user to downsample audio files before streaming them. This allows users on slower connections to stream songs. When this module is installed, a menu of bitrates is presented in the upper right-hand corner of the screen. The user can choose from one of the bitrates, or select a mode that performs no resampling. The selected bitrate is maintained in a persistent cookie so that resampling is performed whenever the user returns to the site.
This module requires a command-line MP3 encoder to resample and reencode the audio data. If not otherwise specified, Apache::MP3::Resample will try to use the Open Source Lame MP3 encoder. This utility is available at http://www.sulaco.org/mp3. Version 3.90 was used during the development of this module. Your results with other versions may vary.
When you install Lame (or the encoder of your choice), be sure to place it in a directory located in Apaches PATH so that the module can find them at run time. You may need to set the PATH environment variable during Apaches launch, or by explicitly adding a SetEnv directive to the Apache configuration file.
You should be aware that the decoding/reencoding process is CPU-intensive, and server performance may degrade as the number of simultaneous users increases.
<<lessSYNOPSIS
# httpd.conf or access.conf
AddType audio/shorten .shn .SHN
AddType audio/flac .flac
Alias /apache_mp3 /usr/share/libapache-mp3-perl
< Location /songs >
SetHandler perl-script
PerlHandler Apache::MP3::Resample
PerlSetVar CacheDir /var/cache/Apache::MP3
PerlSetVar AllowDownload no
PerlSetVar SortFields Album,Track,Title,-Duration
PerlSetVar Fields Track,Title,Artist,Album,Duration,Bitrate
< /Location >
Apache::MP3::Resample subclasses Apache::MP3::Playlist to allow the user to downsample audio files before streaming them. This allows users on slower connections to stream songs. When this module is installed, a menu of bitrates is presented in the upper right-hand corner of the screen. The user can choose from one of the bitrates, or select a mode that performs no resampling. The selected bitrate is maintained in a persistent cookie so that resampling is performed whenever the user returns to the site.
This module requires a command-line MP3 encoder to resample and reencode the audio data. If not otherwise specified, Apache::MP3::Resample will try to use the Open Source Lame MP3 encoder. This utility is available at http://www.sulaco.org/mp3. Version 3.90 was used during the development of this module. Your results with other versions may vary.
When you install Lame (or the encoder of your choice), be sure to place it in a directory located in Apaches PATH so that the module can find them at run time. You may need to set the PATH environment variable during Apaches launch, or by explicitly adding a SetEnv directive to the Apache configuration file.
You should be aware that the decoding/reencoding process is CPU-intensive, and server performance may degrade as the number of simultaneous users increases.
Download (0.32MB)
Added: 2006-06-22 License: Perl Artistic License Price:
1221 downloads
Grip 3.3.1
Grip is a CD player and CD ripper/MP3-encoder for the GNOME desktop. more>>
Grip is a CD player and CD ripper/MP3-encoder for the GNOME desktop. It has the ripping capabilities of cdparanoia built in, but can also use external rippers (such as cdda2wav).
It also provides an automated frontend for MP3 encoders (presets for lame, bladeenc, l3enc, xingmp3enc, mp3encode, and gogo), letting you take a disc and transform it easily straight into MP3s.
The Ogg Vorbis format is also supported. Internet disc lookups are supported for retrieving track information from disc database servers.
Grip works with DigitalDJ to provide a unified, "computerized" version of your music collection.
Main features:
- Full-featured CD player with a small screen footprint in "condensed" mode
- Database lookup/submission to share track information over the net
- HTTP proxy support for those behind firewalls
- Loop, shuffle, and playlist modes
- Ripping of single, multiple, or partial tracks
- Encoding of ripped .wav files into MP3 files (as well support for OGG and FLAC)
- Simultaneous rip and encode
- Support for multiple encode processes on SMP machines
- Adding ID3v1/v2 tags to MP3 files
- Cooperating with DigitalDJ, my SQL-based MP3 jukebox
Enhancements:
- de.po: updated (G?tz Waschk)
- it.po: updated (Ceoldo Costantino)
- fr.po: updated (Eric Lassauge)
- pl_PL.po: added (Piotr Adamocha)
- id3.c: put a zero byte before the id3v1 track number (Vladimir Petrov)
- discdb.c: string parsing fixes to support i18n (Vladimir Petrov)
- discdb.c: better handling of non-UTF-8 local discdb files (Vladimir Petrov)
- various: tweaks to filesystem-safe character escaping (Vladimir Petrov)
- cdplay.c: allow retrieving of non-UTF-8 discdb entries (Vladimir Petrov)
- discdb.c: fixed a possible buffer overflow crash (Dean Brettle)
- grip.spec.in: added some missing BuildRequires (Stephen E. Dudek)
- configure.in: upped version to 3.3.1
<<lessIt also provides an automated frontend for MP3 encoders (presets for lame, bladeenc, l3enc, xingmp3enc, mp3encode, and gogo), letting you take a disc and transform it easily straight into MP3s.
The Ogg Vorbis format is also supported. Internet disc lookups are supported for retrieving track information from disc database servers.
Grip works with DigitalDJ to provide a unified, "computerized" version of your music collection.
Main features:
- Full-featured CD player with a small screen footprint in "condensed" mode
- Database lookup/submission to share track information over the net
- HTTP proxy support for those behind firewalls
- Loop, shuffle, and playlist modes
- Ripping of single, multiple, or partial tracks
- Encoding of ripped .wav files into MP3 files (as well support for OGG and FLAC)
- Simultaneous rip and encode
- Support for multiple encode processes on SMP machines
- Adding ID3v1/v2 tags to MP3 files
- Cooperating with DigitalDJ, my SQL-based MP3 jukebox
Enhancements:
- de.po: updated (G?tz Waschk)
- it.po: updated (Ceoldo Costantino)
- fr.po: updated (Eric Lassauge)
- pl_PL.po: added (Piotr Adamocha)
- id3.c: put a zero byte before the id3v1 track number (Vladimir Petrov)
- discdb.c: string parsing fixes to support i18n (Vladimir Petrov)
- discdb.c: better handling of non-UTF-8 local discdb files (Vladimir Petrov)
- various: tweaks to filesystem-safe character escaping (Vladimir Petrov)
- cdplay.c: allow retrieving of non-UTF-8 discdb entries (Vladimir Petrov)
- discdb.c: fixed a possible buffer overflow crash (Dean Brettle)
- grip.spec.in: added some missing BuildRequires (Stephen E. Dudek)
- configure.in: upped version to 3.3.1
Download (0.57MB)
Added: 2005-06-29 License: GPL (GNU General Public License) Price:
1578 downloads
MP3FS 0.07
MP3FS is A read-only FUSE filesystem which transcodes audio formats (currently FLAC) to MP3 on the fly when opened and read. more>>
&MP3FS is A read-only FUSE filesystem which transcodes audio formats (currently FLAC) to MP3 on the fly when opened and read. This was written to enable me to use my FLAC collection with software and/or hardware which only understands MP3. e.g. gmediaserver to a netgear MP101 mp3 player.
MP3FS is also a novel alternative to traditional mp3 encoder applications. Just use your favorite file browser to select the files you want encoded and copy them somewhere!
How it Works:
When a file is opened, the decoder and encoder are initialised and the file metadata is read. At this time the final filesize can be determined as we only support constant bitrate mp3s.
ID3v2 tags are created from the vorbis comments in the flac file. They are located at the start of the file. As such, an application scanning a directory to read tags should not cause too much of a performance hit as the actual encoder does not need to be invoked (depending on how much data the read asks for).
A special exception to this is when an application tries to read the very last block first. Many applications do this to look for an id3v1 tag (stored in the last 128 bytes of the file). When this is detected, the filesystem simply return zeros (I dont support id3v1 tags). This dramatically speeds up applications, however it could potentially lead to corrupt mp3 files if the zeros are still in kernel cache when the application comes back to read the actual audio sequentially. In my experimentation this has not happened, I always get another read for the final block.
As the file is read, it is transcoded into an internal per-file buffer. This buffer continues to grow while the file is being read until the whole file is transcoded in memory. The memory is freed only when the file is closed. This simplifies the implementation.
Seeking within a file will cause the file to be transcoded up to the seek point (if not already done). This is not usually a problem since most programs will read a file from start to finish. Future enhancements may provide true random seeking.
For build instructions see INSTALL
Usage:
mount your filesystem like this: (you will probably have to be root)
mp3fs musicdir bitrate mountpoint [-o fuse_options]
e.g.:
mp3fs /mnt/music 256 /mnt/mp3 -o allow_other,ro
Here are the original files:
dave@bender:~/mp3fs$ ls -l /mnt/music/Smashing Pumpkins/Pisces Iscariot/
total 345732
-rw-r--r-- 1 mythtv mythtv 10267876 2005-06-19 18:36 01 - Soothe.flac
-rw-r--r-- 1 mythtv mythtv 23512276 2005-06-19 18:36 02 - Frail And Bedazzled.flac
-rw-r--r-- 1 mythtv mythtv 23332187 2005-06-19 18:36 03 - Plum.flac
-rw-r--r-- 1 mythtv mythtv 26402936 2005-06-19 18:36 04 - Whir.flac
-rw-r--r-- 1 mythtv mythtv 21591252 2005-06-19 18:36 05 - Blew Away.flac
-rw-r--r-- 1 mythtv mythtv 16719855 2005-06-19 18:36 06 - Pissant.flac
-rw-r--r-- 1 mythtv mythtv 33454889 2005-06-19 18:36 07 - Hello Kitty Kat.flac
-rw-r--r-- 1 mythtv mythtv 32073747 2005-06-19 18:36 08 - Obscured.flac
-rw-r--r-- 1 mythtv mythtv 17614217 2005-06-19 18:36 09 - Landslide.flac
-rw-r--r-- 1 mythtv mythtv 65406696 2005-06-19 18:36 10 - Starla.flac
-rw-r--r-- 1 mythtv mythtv 18651734 2005-06-19 18:36 11 - Blue.flac
-rw-r--r-- 1 mythtv mythtv 25055200 2005-06-19 18:36 12 - Girl Named Sandoz.flac
-rw-r--r-- 1 mythtv mythtv 28060023 2005-06-19 18:36 13 - La Dolly Vita.flac
-rw-r--r-- 1 mythtv mythtv 11432008 2005-06-19 18:36 14 - Spaced.flac
And now you can use the (virtual) mp3 files from the MP3FS mountpoint:
dave@bender:~/mp3fs$ ls -l /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/
total 345732
-rw-r--r-- 1 mythtv mythtv 2446849 2005-06-19 18:36 01 - Soothe.mp3
-rw-r--r-- 1 mythtv mythtv 3197934 2005-06-19 18:36 02 - Frail And Bedazzled.mp3
-rw-r--r-- 1 mythtv mythtv 3467503 2005-06-19 18:36 03 - Plum.mp3
-rw-r--r-- 1 mythtv mythtv 4003745 2005-06-19 18:36 04 - Whir.mp3
-rw-r--r-- 1 mythtv mythtv 3414845 2005-06-19 18:36 05 - Blew Away.mp3
-rw-r--r-- 1 mythtv mythtv 2413413 2005-06-19 18:36 06 - Pissant.mp3
-rw-r--r-- 1 mythtv mythtv 4348572 2005-06-19 18:36 07 - Hello Kitty Kat.mp3
-rw-r--r-- 1 mythtv mythtv 5132656 2005-06-19 18:36 08 - Obscured.mp3
-rw-r--r-- 1 mythtv mythtv 3099704 2005-06-19 18:36 09 - Landslide.mp3
-rw-r--r-- 1 mythtv mythtv 10542719 2005-06-19 18:36 10 - Starla.mp3
-rw-r--r-- 1 mythtv mythtv 3210041 2005-06-19 18:36 11 - Blue.mp3
-rw-r--r-- 1 mythtv mythtv 3449127 2005-06-19 18:36 12 - Girl Named Sandoz.mp3
-rw-r--r-- 1 mythtv mythtv 4098213 2005-06-19 18:36 13 - La Dolly Vita.mp3
-rw-r--r-- 1 mythtv mythtv 2337344 2005-06-19 18:36 14 - Spaced.mp3
dave@bender:~/mp3fs$ id3info /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/01 - Soothe.mp3
*** Tag information for /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/01 - Soothe.mp3
=== TSSE (Software/Hardware and settings used for encoding): LAME v3.96.1
=== TIT2 (Title/songname/content description): Soothe
=== TPE1 (Lead performer(s)/Soloist(s)): Smashing Pumpkins
=== TALB (Album/Movie/Show title): Pisces Iscariot
=== TRCK (Track number/Position in set): 1
*** mp3 info
MPEG1/layer III
Bitrate: 128KBps
Frequency: 44KHz
dave@bender:~/mp3fs$ time cp /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/01 - Soothe.mp3 /tmp/
real 0m12.917s
user 0m0.004s
sys 0m0.020s
dave@bender:~/mp3fs$ xmms /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/*
Enhancements:
- A segfault and several memory leaks were fixed.
<<lessMP3FS is also a novel alternative to traditional mp3 encoder applications. Just use your favorite file browser to select the files you want encoded and copy them somewhere!
How it Works:
When a file is opened, the decoder and encoder are initialised and the file metadata is read. At this time the final filesize can be determined as we only support constant bitrate mp3s.
ID3v2 tags are created from the vorbis comments in the flac file. They are located at the start of the file. As such, an application scanning a directory to read tags should not cause too much of a performance hit as the actual encoder does not need to be invoked (depending on how much data the read asks for).
A special exception to this is when an application tries to read the very last block first. Many applications do this to look for an id3v1 tag (stored in the last 128 bytes of the file). When this is detected, the filesystem simply return zeros (I dont support id3v1 tags). This dramatically speeds up applications, however it could potentially lead to corrupt mp3 files if the zeros are still in kernel cache when the application comes back to read the actual audio sequentially. In my experimentation this has not happened, I always get another read for the final block.
As the file is read, it is transcoded into an internal per-file buffer. This buffer continues to grow while the file is being read until the whole file is transcoded in memory. The memory is freed only when the file is closed. This simplifies the implementation.
Seeking within a file will cause the file to be transcoded up to the seek point (if not already done). This is not usually a problem since most programs will read a file from start to finish. Future enhancements may provide true random seeking.
For build instructions see INSTALL
Usage:
mount your filesystem like this: (you will probably have to be root)
mp3fs musicdir bitrate mountpoint [-o fuse_options]
e.g.:
mp3fs /mnt/music 256 /mnt/mp3 -o allow_other,ro
Here are the original files:
dave@bender:~/mp3fs$ ls -l /mnt/music/Smashing Pumpkins/Pisces Iscariot/
total 345732
-rw-r--r-- 1 mythtv mythtv 10267876 2005-06-19 18:36 01 - Soothe.flac
-rw-r--r-- 1 mythtv mythtv 23512276 2005-06-19 18:36 02 - Frail And Bedazzled.flac
-rw-r--r-- 1 mythtv mythtv 23332187 2005-06-19 18:36 03 - Plum.flac
-rw-r--r-- 1 mythtv mythtv 26402936 2005-06-19 18:36 04 - Whir.flac
-rw-r--r-- 1 mythtv mythtv 21591252 2005-06-19 18:36 05 - Blew Away.flac
-rw-r--r-- 1 mythtv mythtv 16719855 2005-06-19 18:36 06 - Pissant.flac
-rw-r--r-- 1 mythtv mythtv 33454889 2005-06-19 18:36 07 - Hello Kitty Kat.flac
-rw-r--r-- 1 mythtv mythtv 32073747 2005-06-19 18:36 08 - Obscured.flac
-rw-r--r-- 1 mythtv mythtv 17614217 2005-06-19 18:36 09 - Landslide.flac
-rw-r--r-- 1 mythtv mythtv 65406696 2005-06-19 18:36 10 - Starla.flac
-rw-r--r-- 1 mythtv mythtv 18651734 2005-06-19 18:36 11 - Blue.flac
-rw-r--r-- 1 mythtv mythtv 25055200 2005-06-19 18:36 12 - Girl Named Sandoz.flac
-rw-r--r-- 1 mythtv mythtv 28060023 2005-06-19 18:36 13 - La Dolly Vita.flac
-rw-r--r-- 1 mythtv mythtv 11432008 2005-06-19 18:36 14 - Spaced.flac
And now you can use the (virtual) mp3 files from the MP3FS mountpoint:
dave@bender:~/mp3fs$ ls -l /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/
total 345732
-rw-r--r-- 1 mythtv mythtv 2446849 2005-06-19 18:36 01 - Soothe.mp3
-rw-r--r-- 1 mythtv mythtv 3197934 2005-06-19 18:36 02 - Frail And Bedazzled.mp3
-rw-r--r-- 1 mythtv mythtv 3467503 2005-06-19 18:36 03 - Plum.mp3
-rw-r--r-- 1 mythtv mythtv 4003745 2005-06-19 18:36 04 - Whir.mp3
-rw-r--r-- 1 mythtv mythtv 3414845 2005-06-19 18:36 05 - Blew Away.mp3
-rw-r--r-- 1 mythtv mythtv 2413413 2005-06-19 18:36 06 - Pissant.mp3
-rw-r--r-- 1 mythtv mythtv 4348572 2005-06-19 18:36 07 - Hello Kitty Kat.mp3
-rw-r--r-- 1 mythtv mythtv 5132656 2005-06-19 18:36 08 - Obscured.mp3
-rw-r--r-- 1 mythtv mythtv 3099704 2005-06-19 18:36 09 - Landslide.mp3
-rw-r--r-- 1 mythtv mythtv 10542719 2005-06-19 18:36 10 - Starla.mp3
-rw-r--r-- 1 mythtv mythtv 3210041 2005-06-19 18:36 11 - Blue.mp3
-rw-r--r-- 1 mythtv mythtv 3449127 2005-06-19 18:36 12 - Girl Named Sandoz.mp3
-rw-r--r-- 1 mythtv mythtv 4098213 2005-06-19 18:36 13 - La Dolly Vita.mp3
-rw-r--r-- 1 mythtv mythtv 2337344 2005-06-19 18:36 14 - Spaced.mp3
dave@bender:~/mp3fs$ id3info /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/01 - Soothe.mp3
*** Tag information for /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/01 - Soothe.mp3
=== TSSE (Software/Hardware and settings used for encoding): LAME v3.96.1
=== TIT2 (Title/songname/content description): Soothe
=== TPE1 (Lead performer(s)/Soloist(s)): Smashing Pumpkins
=== TALB (Album/Movie/Show title): Pisces Iscariot
=== TRCK (Track number/Position in set): 1
*** mp3 info
MPEG1/layer III
Bitrate: 128KBps
Frequency: 44KHz
dave@bender:~/mp3fs$ time cp /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/01 - Soothe.mp3 /tmp/
real 0m12.917s
user 0m0.004s
sys 0m0.020s
dave@bender:~/mp3fs$ xmms /mnt/mp3/Smashing Pumpkins/Pisces Iscariot/*
Enhancements:
- A segfault and several memory leaks were fixed.
Download (0.32MB)
Added: 2007-04-09 License: GPL (GNU General Public License) Price:
932 downloads
MP3c 0.31
MP3c is a special program for converting audio-cds to mp3/ogg-format. more>>
MP3c is a special program for converting audio-cds to ogg/mp3-format. It is a frontend for every grab-program and mp3-encoder. Nearly every program should be supported. MP3c has two different run-modes: An interactive mode with a menu-enviroment (based on curses-lib), which can used on textconsoles as well under X11, and a batchmode, which allows automatic encoding of cds inside shellscripts (eg at night within cronjobs).
There are two different possibilities for encoding (depends on your ripper and encoder): Non-Fly encoding, where an audio-track is first grabbed from cds and then encoded, and On-Fly encoding, where grabbing and encoding is one step. Normally you should refer to Non-Fly encoding, since your cdrom-drive hasnt to work all the time, and some encoder produces a "click" at the beginning of mp3/ogg-files if On-Fly encoding is used. The only disadvantage of Non-Fly encoding is, that there is a temp-file, which can grow up to 50-100 MB sometimes.
Information about songtitle on your cds are received with help of CDDB database. In contrast to other programs, an CDDB-entry is detailed analyzed to find every information even for sampler-cds.
Creation of filenames is userdefined. You can use pattern to constitute the appearance of your mp3-files individually. Furthermore you can decide how spaces within filenames should be handled (you can allow them, convert them to underscores "_" or just kill them).
Interessting is the possibility of shellscript creation, which contains all functionality for encoding your cds. You can start these scripts as daemon-processes or run them at night inside a cronjob.
Recommended programs for the use of MP3c are: CDparanoia as audio-cd grabber, Encoder as MP3-encoder and MP3info for creating MP3-ID-tags. For OGG use oggenc. If you need, you can download these programs here, or a complete package with these programs inside. Furthermore you can download runable binaries, but only for glibc 2.1 (linux i386) at the moment. If you like to create RPMs and/or binaries for other systems and libraries, please send these to me, so that I can put it on my page.
Enhancements:
- Switched to CDDB protocol level 5 to support detection and saving of year and genre information.
<<lessThere are two different possibilities for encoding (depends on your ripper and encoder): Non-Fly encoding, where an audio-track is first grabbed from cds and then encoded, and On-Fly encoding, where grabbing and encoding is one step. Normally you should refer to Non-Fly encoding, since your cdrom-drive hasnt to work all the time, and some encoder produces a "click" at the beginning of mp3/ogg-files if On-Fly encoding is used. The only disadvantage of Non-Fly encoding is, that there is a temp-file, which can grow up to 50-100 MB sometimes.
Information about songtitle on your cds are received with help of CDDB database. In contrast to other programs, an CDDB-entry is detailed analyzed to find every information even for sampler-cds.
Creation of filenames is userdefined. You can use pattern to constitute the appearance of your mp3-files individually. Furthermore you can decide how spaces within filenames should be handled (you can allow them, convert them to underscores "_" or just kill them).
Interessting is the possibility of shellscript creation, which contains all functionality for encoding your cds. You can start these scripts as daemon-processes or run them at night inside a cronjob.
Recommended programs for the use of MP3c are: CDparanoia as audio-cd grabber, Encoder as MP3-encoder and MP3info for creating MP3-ID-tags. For OGG use oggenc. If you need, you can download these programs here, or a complete package with these programs inside. Furthermore you can download runable binaries, but only for glibc 2.1 (linux i386) at the moment. If you like to create RPMs and/or binaries for other systems and libraries, please send these to me, so that I can put it on my page.
Enhancements:
- Switched to CDDB protocol level 5 to support detection and saving of year and genre information.
Download (0.20MB)
Added: 2006-07-30 License: GPL (GNU General Public License) Price:
1184 downloads
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
RipperX 2.7.0
ripperX is an audio CD ripper + encoder GUI including CDDB support and the ability to encode as OGG, FLAC, MP3 and MP2 (toolame) more>>
RipperX is a GTK program to rip CD audio and encode it into MP3, OGG, or FLAC files.
It can rip and encode in parallel, and has plugins for cdparanoia, BladeEnc, Lame, GoGo, FHG (l3enc and mp3enc), XingMp3enc, 8hz-mp3, and the ISO encoder.
RipperX also has support for CDDB and ID3v1 tags.
Enhancements:
- bug fixing release
<<lessIt can rip and encode in parallel, and has plugins for cdparanoia, BladeEnc, Lame, GoGo, FHG (l3enc and mp3enc), XingMp3enc, 8hz-mp3, and the ISO encoder.
RipperX also has support for CDDB and ID3v1 tags.
Enhancements:
- bug fixing release
Download (0.16MB)
Added: 2006-07-16 License: GPL (GNU General Public License) Price:
1198 downloads
Versatile Encoder 0.8p2
Versatile Encoder allows you to encode your own videos into good quality mpeg4 or h264. more>>
Versatile Encoder allows you to encode your own videos into good quality mpeg4 or h264. The input can be a sequence of files (as generated by blender), a file (if you want to recompress something) or a DVD/VCD (if you need to backup some movie).
This piece of software tries to guess good defaults and to help the user in his choices. It works at least under Windows and Linux, the coding platform is Linux.
It should work without a hassle under all BSD variants and probably under MacOSX too, report me if you get it working somewhere else.
<<lessThis piece of software tries to guess good defaults and to help the user in his choices. It works at least under Windows and Linux, the coding platform is Linux.
It should work without a hassle under all BSD variants and probably under MacOSX too, report me if you get it working somewhere else.
Download (0.092MB)
Added: 2006-07-10 License: BSD License Price:
1207 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
gnormalize 0.52
gnormalize is an audio converter, a front end to normalize, an encoder/decoder, a ripper, a tag editor and a cd player. more>>
gnormalize is an audio converter, a front end to normalize, an encoder/decoder, a ripper, a metadata (tag) editor and a cd player. gnormalize uses gtk2-perl under GNU/Linux.
gnormalize decodes the MP3/MP4/MPC/OGG/APE/FLAC file to WAV, then normalizes the WAV to a targeted volume level and re-encodes it. Moreover, gnormalize can extract Audio CD track and output as various popular audio formats (MP3, MP4, MPC, OGG, APE, FLAC, WAV) with fast speed and high quality. gnormalize can also convert audio format between MP3, MP4, MPC, OGG, APE and FLAC with high fidelity, which meets your need to play and collect audio files. It can change the encoding and Metadata (tag) properties of final normalized files.
gnormalize can be used to adjust the volume of audio files to a standard volume level, where different recording levels on different albums can cause the volume to vary greatly from song to song.
gnormalize is an alternative for replaygain, because "there is no consistent standard by which to define the appropriate replay gain which mp3 encoders and players agree on, and no automatic way to set the volume adjustment for each track".
Main features:
- Convert from/to MP3, MP4, MPC, OGG, APE, FLAC and WAV;
- Support for LAME, FAAC, OGGENC, MPPENC, FLAC and MAC encoders;
- Support for LAME, FAAD, OGGDEC, MPPDEC, FLAC and MAC decoders;
- Rip audio CDs utilizing cdparanoia or cdda2wav;
- Play audio CDs utilizing cdcd or Audio::CD or cdplay;
- Support normalization with adjustable sensibility;
- Can normalize/convert files recursively into the directory;
- Edit and get MetaData like artist, title, album, ... ;
- Get MetaInfo like bitrate, time, frequency, ... ;
- Includes mppenc, mppdec;
- Read the CDDB entry for an audio CD with CDDB_get;
- Translation to English and Portuguese (Brasil).
<<lessgnormalize decodes the MP3/MP4/MPC/OGG/APE/FLAC file to WAV, then normalizes the WAV to a targeted volume level and re-encodes it. Moreover, gnormalize can extract Audio CD track and output as various popular audio formats (MP3, MP4, MPC, OGG, APE, FLAC, WAV) with fast speed and high quality. gnormalize can also convert audio format between MP3, MP4, MPC, OGG, APE and FLAC with high fidelity, which meets your need to play and collect audio files. It can change the encoding and Metadata (tag) properties of final normalized files.
gnormalize can be used to adjust the volume of audio files to a standard volume level, where different recording levels on different albums can cause the volume to vary greatly from song to song.
gnormalize is an alternative for replaygain, because "there is no consistent standard by which to define the appropriate replay gain which mp3 encoders and players agree on, and no automatic way to set the volume adjustment for each track".
Main features:
- Convert from/to MP3, MP4, MPC, OGG, APE, FLAC and WAV;
- Support for LAME, FAAC, OGGENC, MPPENC, FLAC and MAC encoders;
- Support for LAME, FAAD, OGGDEC, MPPDEC, FLAC and MAC decoders;
- Rip audio CDs utilizing cdparanoia or cdda2wav;
- Play audio CDs utilizing cdcd or Audio::CD or cdplay;
- Support normalization with adjustable sensibility;
- Can normalize/convert files recursively into the directory;
- Edit and get MetaData like artist, title, album, ... ;
- Get MetaInfo like bitrate, time, frequency, ... ;
- Includes mppenc, mppdec;
- Read the CDDB entry for an audio CD with CDDB_get;
- Translation to English and Portuguese (Brasil).
Download (0.53MB)
Added: 2006-12-03 License: GPL (GNU General Public License) Price:
1064 downloads
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