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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
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
Kwirc 0.1.9.3
Kwirc is an irc client for KDE. more>>
Kwirc is another irc client irc : it is designed to be sexy.
Main features:
- SSL Support
- Multiserver
- Lame mode with smileys
- Ruby scripting
- Sexy interface
- Random crashes
Enhancements:
- Initial release
<<lessMain features:
- SSL Support
- Multiserver
- Lame mode with smileys
- Ruby scripting
- Sexy interface
- Random crashes
Enhancements:
- Initial release
Download (0.23MB)
Added: 2005-06-03 License: GPL (GNU General Public License) Price:
1604 downloads
SimpleCDR-X 1.3.3
SimpleCDR-X is a GTK+ based tool for CD writing, mastering, and audio ripping/compression. more>>
SimpleCDR-X was born in mid-June of 2001. It was clear to me that SimpleCDRs interface had limitations that could only be overcome by going to a GUI interface. I then proceeded to look at the various toolkits and then I discovered Glade.
Glade is perhaps one of the best programming utilities that I have found for Linux to date. It the development of a 2200 line interface much easier than it would have been otherwise. If Glade wasnt around I still might just be playing with the various toolkits instead of having a finished product. Glade allowed me to focus on functionality rather than trying to get the interface to look right with straight C code.
SimpleCDR-X like its brother SimpleCDR is a hybrid of C and C++. Most of the external utilities are managed by C++ classes called from the hybrid callbacks.c. The reason that I didnt opt to use GTK-- instead of the hybird was because most everyone already has GTK+, however, not everyone has GTK-- and some dont want to download a 1.5 MB file to compile or dig up the installation CDs.
Main features:
- Disc-At-Once CD copying
- Audio CD copying via cdrecord and CDparanoia or cdda2wav
- Audio CD Mastering
- MP3/OGG import via MADplay, LAME, or OGG123
- Import from CD via CDparanoia or cdda2wav
- Data CD Mastering
- Multi-session CD writing
- Rip tracks to wav
- Rip tracks to MP3/OGG on the fly via Blade Encode, LAME, or oggenc
- GTK+ Interface
<<lessGlade is perhaps one of the best programming utilities that I have found for Linux to date. It the development of a 2200 line interface much easier than it would have been otherwise. If Glade wasnt around I still might just be playing with the various toolkits instead of having a finished product. Glade allowed me to focus on functionality rather than trying to get the interface to look right with straight C code.
SimpleCDR-X like its brother SimpleCDR is a hybrid of C and C++. Most of the external utilities are managed by C++ classes called from the hybrid callbacks.c. The reason that I didnt opt to use GTK-- instead of the hybird was because most everyone already has GTK+, however, not everyone has GTK-- and some dont want to download a 1.5 MB file to compile or dig up the installation CDs.
Main features:
- Disc-At-Once CD copying
- Audio CD copying via cdrecord and CDparanoia or cdda2wav
- Audio CD Mastering
- MP3/OGG import via MADplay, LAME, or OGG123
- Import from CD via CDparanoia or cdda2wav
- Data CD Mastering
- Multi-session CD writing
- Rip tracks to wav
- Rip tracks to MP3/OGG on the fly via Blade Encode, LAME, or oggenc
- GTK+ Interface
Download (0.40MB)
Added: 2005-06-06 License: GPL (GNU General Public License) Price:
1600 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
Fenice 1.10
Fenice is a standards-compliant multimedia streaming server. more>>
Fenice is a multimedia streaming server compliant with the IETFs standards for real-time streaming of multimedia contents over Internet. Fenice implements RTSP - Real-Time Streaming Protocol (RFC2326) and RTP/RTCP - Real-Time Transport Protocol/RTP Control Protocol (RFC3550) supporting the RTP Profile for Audio and Video Conferences with Minimal Control (RFC3551).
Fenice supports the following encoding standards:
Audio
- MP3 (MPEG-1 Layer III) (RFC3119)
- OGG/Vorbis (work in progress)
Video
- MPEG-1/2 (RFC2250)
- Preliminary support for MPEG-4 (RFC3016, RFC3640)
- OGG/Theora (work in progress)
The main characteristic of Fenice is that it is adaptable to the state of the network gotten through the technique of the dynamic coding change.
Fenice is also able to manage live streaming sessions using external real-time audio/video encoders such as lame, ffmpeg or mjpeg-tools, even capturing audio and video streams from live-recording remote hosts (with Felice - Fenice Live CEaseless).
Fenice is the worlds first streaming server supporting Creative Commons licensing meta-data for audio/video streaming.
Enhancements:
- Log support via file or sysconfig was added.
- MPEG-4 support was finished, and works best for stored contents.
- The annoying 100% CPU bug was fixed.
- Fenice should now run smoothly and reliably in almost every condition.
- Various bugs were fixed.
- Fenice is GCC4 compatible.
<<lessFenice supports the following encoding standards:
Audio
- MP3 (MPEG-1 Layer III) (RFC3119)
- OGG/Vorbis (work in progress)
Video
- MPEG-1/2 (RFC2250)
- Preliminary support for MPEG-4 (RFC3016, RFC3640)
- OGG/Theora (work in progress)
The main characteristic of Fenice is that it is adaptable to the state of the network gotten through the technique of the dynamic coding change.
Fenice is also able to manage live streaming sessions using external real-time audio/video encoders such as lame, ffmpeg or mjpeg-tools, even capturing audio and video streams from live-recording remote hosts (with Felice - Fenice Live CEaseless).
Fenice is the worlds first streaming server supporting Creative Commons licensing meta-data for audio/video streaming.
Enhancements:
- Log support via file or sysconfig was added.
- MPEG-4 support was finished, and works best for stored contents.
- The annoying 100% CPU bug was fixed.
- Fenice should now run smoothly and reliably in almost every condition.
- Various bugs were fixed.
- Fenice is GCC4 compatible.
Download (MB)
Added: 2005-07-27 License: GPL (GNU General Public License) Price:
1554 downloads
CD-Rchive2 2.0.3
CD-Rchive2 is a complete revamp of the popular but now aged CD-Rchive program. more>>
CD-Rchive2 is a complete revamp of the popular but now aged CD-Rchive program. It is intended to be used with cdrecord-ProDVD, cdda2wav, and mkisofs for the production of data and music CDs and data DVDs.
Built in tools allow cloning of a complete CD or DVD and writing copies. A Boot Wizard will guide you through steps required to make a bootable CD or DVD, write the code, and compile a binary menu program which runs with isolinux.
Enhancements:
- Functionality was added when ripping tracks from CD to automatically convert to .mp3 instead of .wav via a checkbox on the .wav tab page.
- This requires lame to be installed.
- A problem of occassional incorrect audio device specification on single device machines such as laptops was fixed.
<<lessBuilt in tools allow cloning of a complete CD or DVD and writing copies. A Boot Wizard will guide you through steps required to make a bootable CD or DVD, write the code, and compile a binary menu program which runs with isolinux.
Enhancements:
- Functionality was added when ripping tracks from CD to automatically convert to .mp3 instead of .wav via a checkbox on the .wav tab page.
- This requires lame to be installed.
- A problem of occassional incorrect audio device specification on single device machines such as laptops was fixed.
Download (3.0MB)
Added: 2005-09-26 License: GPL (GNU General Public License) Price:
1490 downloads
Cdwrite 2.2
Cdwrite is the shell for creation of data and audio disks, including compilations. more>>
Cdwrite is the shell for creation of data and audio disks, including compilations. It allows to use pregaps and recognizes indices.
The shell needs mkisofs and cdrecord for data and cdparanoia, cdda2wav, cdrdao, and -- optionally -- lame for audio.
Enhancements:
- This release allows you to manage not only CD but also DVD discs (writing, erasing, formatting, and blanking).
- It includes a bugfix for the invalid "create data image" option and a man page.
- The RPM package is now better suited to the valid standards.
<<lessThe shell needs mkisofs and cdrecord for data and cdparanoia, cdda2wav, cdrdao, and -- optionally -- lame for audio.
Enhancements:
- This release allows you to manage not only CD but also DVD discs (writing, erasing, formatting, and blanking).
- It includes a bugfix for the invalid "create data image" option and a man page.
- The RPM package is now better suited to the valid standards.
Download (0.017MB)
Added: 2005-11-04 License: GPL (GNU General Public License) Price:
1449 downloads
Arson 0.9.8 Beta2
Arson is a KDE frontend to various CD burning, and ripping tools. more>>
Arson is a KDE frontend to various CD burning, and ripping tools. Arson project was originally begun to burn audio CDs because i could find no other frontends which used cdrdao (in disk at once mode), which could decode various encoded audio formats (mp3, ogg), and displayed accurate track length as the playlist was created. But as usual once the initial plans were implemented I just kept going...
Arson has expanded to be a CD ripper (with many output formats), a VCD/Music/Data burner, a CD copier, a device unlocker, and a CDRW blanker. And thats in the current version!
Main features:
- Full progress displayed for all lengthy operations
- Drag and drop from Konqueror to create play lists
- Audio CD Burning
Accurate track length tally displayed as track list grows
Can burn using cdrdao or cdrecord (in Disk at Once, or Track at Once)
Supports various audio file types, currently:
- Wav
- Mp3 (decode using either mpg123 or LAME)
- Ogg Vorbis (optional)
- SHN
- FLAC (optional)
Can optionally normalize (in batch or mix mode) all tracks before burning to even out volumes
Supports sox to fix broken audio tracks (tracks not in 44100Khz, 16bit, stereo)
Can open, and optionally verify MD5 disk sets
Can load track lists from m3u files
Supports editing and burning CD-Text info with cdrdao
- VCD Burning
Can create and burn VCDs and SVCDs
- Data CD Burning
Existing ISO, and CUE/BIN files
Image creation/burning from single directory tree
Complete filesystem creation
ISO images burned with either cdrdao or cdrecord
- Audio CD ripping/encoding (rip tracks from CD to file)
Can rip audio tracks using cdda2wav OR cdparanoia
Encoding in various output file formats, currently:
- Wav
- Mp3 (bladeenc, and LAME supported)
- Ogg Vorbis (optional)
- FLAC (optional)
- AU
- CDR
- AIFF
- AIFC
CdIndex support (a free CDDB-like service)
HTTP retrieval
Freedb support (a free, open CDDB service)
HTTP, and local retrieval
HTTP, and local submit
CD-Text retrieval
Supports generic SCSI and cooked IOCTL interfaces
Auto tagging of MP3 files using id3v2
Configurable audio quality presets (bitrate, channels, etc)
- CD-to-CD copying
Direct copy
CD-to-file-to-CD copying
Using either readcd/cdrecord or cdrdao
- Data CD ripping
Rip to ISO file with readcd
Rip to CUE/BIN files with cdrdao
- Multisession burning
- Device unlock/reset
- CDRW Blanking
Whats New in 0.9.8 Beta2 Release:
- Fixed bug #829892 - Icon installation broken
- Fixed bug #815006 - Incorrect Freedb Choices. User is now presented with a choice if multiple albums returned
- Fixed bug #826786 - Arson crashes when D&Ding a directory
- Started removal of KDE2 dependancies, including configure.in.in, etc
- Fixed bug with rerunning of configure from a fresh CVS install
<<lessArson has expanded to be a CD ripper (with many output formats), a VCD/Music/Data burner, a CD copier, a device unlocker, and a CDRW blanker. And thats in the current version!
Main features:
- Full progress displayed for all lengthy operations
- Drag and drop from Konqueror to create play lists
- Audio CD Burning
Accurate track length tally displayed as track list grows
Can burn using cdrdao or cdrecord (in Disk at Once, or Track at Once)
Supports various audio file types, currently:
- Wav
- Mp3 (decode using either mpg123 or LAME)
- Ogg Vorbis (optional)
- SHN
- FLAC (optional)
Can optionally normalize (in batch or mix mode) all tracks before burning to even out volumes
Supports sox to fix broken audio tracks (tracks not in 44100Khz, 16bit, stereo)
Can open, and optionally verify MD5 disk sets
Can load track lists from m3u files
Supports editing and burning CD-Text info with cdrdao
- VCD Burning
Can create and burn VCDs and SVCDs
- Data CD Burning
Existing ISO, and CUE/BIN files
Image creation/burning from single directory tree
Complete filesystem creation
ISO images burned with either cdrdao or cdrecord
- Audio CD ripping/encoding (rip tracks from CD to file)
Can rip audio tracks using cdda2wav OR cdparanoia
Encoding in various output file formats, currently:
- Wav
- Mp3 (bladeenc, and LAME supported)
- Ogg Vorbis (optional)
- FLAC (optional)
- AU
- CDR
- AIFF
- AIFC
CdIndex support (a free CDDB-like service)
HTTP retrieval
Freedb support (a free, open CDDB service)
HTTP, and local retrieval
HTTP, and local submit
CD-Text retrieval
Supports generic SCSI and cooked IOCTL interfaces
Auto tagging of MP3 files using id3v2
Configurable audio quality presets (bitrate, channels, etc)
- CD-to-CD copying
Direct copy
CD-to-file-to-CD copying
Using either readcd/cdrecord or cdrdao
- Data CD ripping
Rip to ISO file with readcd
Rip to CUE/BIN files with cdrdao
- Multisession burning
- Device unlock/reset
- CDRW Blanking
Whats New in 0.9.8 Beta2 Release:
- Fixed bug #829892 - Icon installation broken
- Fixed bug #815006 - Incorrect Freedb Choices. User is now presented with a choice if multiple albums returned
- Fixed bug #826786 - Arson crashes when D&Ding a directory
- Started removal of KDE2 dependancies, including configure.in.in, etc
- Fixed bug with rerunning of configure from a fresh CVS install
Download (0.54MB)
Added: 2006-02-01 License: GPL (GNU General Public License) Price:
1361 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
AudConvert 0.52
AudConvert is an application that is designed to take any audio format and convert it to any other audio format. more>>
AudConvert is an application that is designed to take any audio format and convert it to any other audio format.
The idea for AudConvert came from my need to turn my Ogg Vorbis collection into MP3s for portable devices.
Yes, this process sometimes will result in lower quality, but sometimes it must be done.
Main features:
- Input any directory of files, get out the same directory structure (or flat directory) of newly encoded files.
- Multi-threaded: Encode up to 8 files simultaneously.
This is the first release of this software and it needs a lot of testing.
Supported Inputs:
- Ogg Vorbis (oggdec)
- MP3 (mpg123)
- FLAC (flac)
Supported Outputs:
- Ogg Vorbis (oggenc)
- MP3 (lame)
<<lessThe idea for AudConvert came from my need to turn my Ogg Vorbis collection into MP3s for portable devices.
Yes, this process sometimes will result in lower quality, but sometimes it must be done.
Main features:
- Input any directory of files, get out the same directory structure (or flat directory) of newly encoded files.
- Multi-threaded: Encode up to 8 files simultaneously.
This is the first release of this software and it needs a lot of testing.
Supported Inputs:
- Ogg Vorbis (oggdec)
- MP3 (mpg123)
- FLAC (flac)
Supported Outputs:
- Ogg Vorbis (oggenc)
- MP3 (lame)
Download (0.022MB)
Added: 2006-03-11 License: GPL (GNU General Public License) Price:
1322 downloads
MP3 Jukebox-Tk 09022001
MP3 Jukebox-Tk is an organizer/jukebox for local mp3 collections. more>>
MP3 Jukebox-Tk is an organizer/jukebox for local mp3 collections. MP3 Jukebox-Tk sorts your mp3 files by band and presents a jukebox style interface to let you choose which ones to play.
It then passes these off to XMMS for playing. It also keeps stats on what you listen to, and can use those stats to generate a weighted random playlist for you. Non-weighted (truly random) playlists are also available.
<<lessIt then passes these off to XMMS for playing. It also keeps stats on what you listen to, and can use those stats to generate a weighted random playlist for you. Non-weighted (truly random) playlists are also available.
Download (0.019MB)
Added: 2006-04-21 License: GPL (GNU General Public License) Price:
1296 downloads
gTVTimer 0.5
gTVTimer is an easy to use videorecorder and frontend to MEncoder. more>>
gTVTimer is an easy to use videorecorder and frontend to MEncoder. You can use gTVTimer to schedule recordings from your analog TV card.
There is still a lot to add/improve. Feel free to send bug reports and suggestions.
Main features:
- supported video codecs: DivX, MJPEG, MPEG2, XviD
- supported audio codecs: MP2 (lavc, toolame), MP3 (lavc, lame)
- supported filters: crop, resize, deinterlace, denoise
- simple TV application
- preview of recorded videos
- translations: English, German
<<lessThere is still a lot to add/improve. Feel free to send bug reports and suggestions.
Main features:
- supported video codecs: DivX, MJPEG, MPEG2, XviD
- supported audio codecs: MP2 (lavc, toolame), MP3 (lavc, lame)
- supported filters: crop, resize, deinterlace, denoise
- simple TV application
- preview of recorded videos
- translations: English, German
Download (0.085MB)
Added: 2006-05-23 License: GPL (GNU General Public License) Price:
1251 downloads
Audio::MPEG 0.04
Audio::MPEG is a Perl module for encoding and decoding of MPEG Audio (MP3). more>>
Audio::MPEG is a Perl module for encoding and decoding of MPEG Audio (MP3).
SYNOPSIS
use Audio::MPEG;
Audio::MPEG is a Perl interface to the LAME and MAD MPEG audio Layers I, II, and III encoding and decoding libraries.
Rationale
I have been building a fairly extensive MP3 library, and decided to write some software to help manage the collection. Its turned out to be a rather cool piece of software (incidentally, I will be releasing it under the GPL shortly), with both a web and command line interface, good searching, integrated ripping, archive statistics, etc.
However, I also wanted to be able to stream audio, and verify the integrity of files in the archive. It is certainly possible to stream audio (even with re-encoding at a different bitrate) without resorting to writing interface glue like this module, but verification of the files was clumsy at best (e.g. scanning stdout/err for strings), and useless at worst.
Thus, Audio::MPEG was born.
LAME
This is arguably the best quality MPEG encoder available (certainly the best GPL encoder). Portions of the code have been optimized to take advantage of some of the advanced features for Intel/AMD processors, but even on non-optimized machines, such as the PowerPC, it performs quite well (faster than real-time on late 90s (and later) machines).
MAD
This is a relatively new MPEG decoding library. I chose it after struggling to clean up the MPEG decoding library included with LAME (which is based on Michael Hipps mpg123(1) implementation). In the end, I was very pleased with the results. MAD performs its decoding with an internal precision of 24 bits (pro-level quality) with fixed-point arithmetic. The code is very clean, and seems rock-solid. Although it may seem that it should be faster than the mpg123(1) library due to the use of fixed-point arithmetic, it is in fact about 60% or so of the speed (due to the higher resolution audio). However, the ease of coding against MAD, and the higher precision of the output more than makes up for the slower decoding.
Audio::MPEG can export the data at its highest precision for programs that wish to manipulate the data at the higher resolution.
Operating System Environment
I have only tested this on a Linux 2.4.x system so far, but I see no reason why it should not work on any Un*x variant. In fact, it may actually even work on a Windoze box (the underlying LAME and MAD libraries apparently compile somehow on them). I am doing no special magic with the interface, so presumably it will work under Windows. As you can probably tell, I dont really care if it does (Ill may start caring if M$ releases the source code to Windows under GPL, BSD, or Artistic licenses...). But, for you poor, misguided souls that insist upon running Windows, I expect that there should be little problem getting it to work.
Performance
You would think that with encoding/decoding audio, which is quite a compute-intensive task, Perl would be much slower than the equivalent pure C programs. Surprise... it is only about 3% slower (!) Even with the mechanism I use here (Perl->C->Perl for every frame, Perl 5.6.1 and Linux 2.4.4 (PowerPC 7500) performs just fantastic. So, the moral of this paragraph is to run your own performance tests, but theres no need to think of your own Perl encoder/decoder will be inferior to a pure C/C++ implementation. The only drawback is that, depending upon how much buffer space you use for reading, memory usage will be at least 3 times as much (eh... RAM is cheap...)
<<lessSYNOPSIS
use Audio::MPEG;
Audio::MPEG is a Perl interface to the LAME and MAD MPEG audio Layers I, II, and III encoding and decoding libraries.
Rationale
I have been building a fairly extensive MP3 library, and decided to write some software to help manage the collection. Its turned out to be a rather cool piece of software (incidentally, I will be releasing it under the GPL shortly), with both a web and command line interface, good searching, integrated ripping, archive statistics, etc.
However, I also wanted to be able to stream audio, and verify the integrity of files in the archive. It is certainly possible to stream audio (even with re-encoding at a different bitrate) without resorting to writing interface glue like this module, but verification of the files was clumsy at best (e.g. scanning stdout/err for strings), and useless at worst.
Thus, Audio::MPEG was born.
LAME
This is arguably the best quality MPEG encoder available (certainly the best GPL encoder). Portions of the code have been optimized to take advantage of some of the advanced features for Intel/AMD processors, but even on non-optimized machines, such as the PowerPC, it performs quite well (faster than real-time on late 90s (and later) machines).
MAD
This is a relatively new MPEG decoding library. I chose it after struggling to clean up the MPEG decoding library included with LAME (which is based on Michael Hipps mpg123(1) implementation). In the end, I was very pleased with the results. MAD performs its decoding with an internal precision of 24 bits (pro-level quality) with fixed-point arithmetic. The code is very clean, and seems rock-solid. Although it may seem that it should be faster than the mpg123(1) library due to the use of fixed-point arithmetic, it is in fact about 60% or so of the speed (due to the higher resolution audio). However, the ease of coding against MAD, and the higher precision of the output more than makes up for the slower decoding.
Audio::MPEG can export the data at its highest precision for programs that wish to manipulate the data at the higher resolution.
Operating System Environment
I have only tested this on a Linux 2.4.x system so far, but I see no reason why it should not work on any Un*x variant. In fact, it may actually even work on a Windoze box (the underlying LAME and MAD libraries apparently compile somehow on them). I am doing no special magic with the interface, so presumably it will work under Windows. As you can probably tell, I dont really care if it does (Ill may start caring if M$ releases the source code to Windows under GPL, BSD, or Artistic licenses...). But, for you poor, misguided souls that insist upon running Windows, I expect that there should be little problem getting it to work.
Performance
You would think that with encoding/decoding audio, which is quite a compute-intensive task, Perl would be much slower than the equivalent pure C programs. Surprise... it is only about 3% slower (!) Even with the mechanism I use here (Perl->C->Perl for every frame, Perl 5.6.1 and Linux 2.4.4 (PowerPC 7500) performs just fantastic. So, the moral of this paragraph is to run your own performance tests, but theres no need to think of your own Perl encoder/decoder will be inferior to a pure C/C++ implementation. The only drawback is that, depending upon how much buffer space you use for reading, memory usage will be at least 3 times as much (eh... RAM is cheap...)
Download (00057MB)
Added: 2006-06-19 License: GPL (GNU General Public License) Price:
1225 downloads
Audio::FindChunks 0.03
Audio::FindChunks can breaks audio files into sound/silence parts. more>>
Audio::FindChunks can breaks audio files into sound/silence parts.
SYNOPSIS
use Audio::FindChunks;
# Duplicate input to output, caching RMS values to a file (as a side effect)
Audio::FindChunks->new(rms_filename => x.rms, filter => 1)->get(rms_data);
# Output human-readable info, using RMS cache file xxx.rms if present:
Audio::FindChunks->new(cache_rms => 1, filename => xxx.mp3,
stem_strip_extension => 1)->output_blocks();
# Remove start/end silence (if longer than 0.2sec):
Audio::FindChunks->new(cache_rms => 1, filename => xxx.mp3,
min_actual_silence_sec => 1e100)->split_file();
# Split a multiple-sides tape recording
Audio::FindChunks->new(filename => xxx.mp3, min_actual_silence_sec => 11
)->split_file({verbose => 1});
Audio sequence is broken into parts which contain only noise ("gaps"), and parts with usable signal ("tracks").
The following configuration settings (and defaults) are supported:
# For getting PCM flow (and if averaging data is read from cache)
frequency => 44100, # If raw_pcm or override_header_info only
bytes_per_sample => 4, # likewise
channels => 2, # likewise
sizedata => MY_INF, # likewise (how many bytes of PCM to read)
out_fh => *STDOUT, # mirror WAV/PCM to this FH if filter
# Process non-WAV data:
preprocess => {mp3 => [[qw(lame --silent --decode)], [], [-]]}, # Second contains extra args to read stdin
# RMS cache (used if valid_rms)
rms_extension => .rms, # Appended to the filestem
# Averaging to RMS info
sec_per_chunk => 0.1, # The window for taking mean square
# thresholds picking from the list of sorted 3-medians of RMS data
threshold_in_sorted_min_rel => 0, # relative position of threashold_min
threshold_in_sorted_min_sec => 1, # shifted by this amount in the list
threshold_factor_min => 1, # the list elt is multiplied by this
threshold_in_sorted_max_rel => 0.5, # likewise
threshold_in_sorted_max_sec => 0, # likewise
threshold_factor_max => 1, # likewise
threshold_ratio => 0.15, # relative position between min/max
# Chunkification: smoothification
above_thres_window => 11, # in units of chunks
above_thres_window_rel => 0.25, # fractions of chunks above threshold
# in a window to make chunk signal
# Splitting into runs of signal/noise
max_tracks => 9999, # fail if more signal/noise runs
min_signal_sec => 5, # such runs of signal are forced
min_silence_sec => 2, # likewise
ignore_signal_sec => 1, # short runs of signal are ignored
min_silence_chunks_merge (see below) # and long resulting runs of silence
# are forced
# Calculate average signal in an interval "deeply inside" silence runs
local_level_ignore_pre_sec => 0.3, # offset the start of this interval
local_level_ignore_pre_rel => 0.02, # additional relative offset
local_level_ignore_post_sec => 0.3, # likewise for end of the interval
local_level_ignore_post_rel => 0.02, # likewise
# Enlargement of signal runs: attach consequent chunks with signal this much
# above this average over the neighbour silence run
local_threshold_factor => 1.05,
# Final enlargement of runs of signal
extend_track_end_sec => 0.5, # Unconditional enlargement
extend_track_begin_sec => 0.3, # likewise
min_boundary_silence_sec => 0.2, # Ignore short silence at start/end
Note that above_thres_window is the only value specified directly in units of chunks; the other *_sec may be optionally specified in units of chunks by setting the corresponding *_chunks value. Note also that this window should better be decreased if minimal allowed silence length parameters are decreased.
These values are mirrored from other values if not explicitly specified:
min_actual_silence_sec<<less
SYNOPSIS
use Audio::FindChunks;
# Duplicate input to output, caching RMS values to a file (as a side effect)
Audio::FindChunks->new(rms_filename => x.rms, filter => 1)->get(rms_data);
# Output human-readable info, using RMS cache file xxx.rms if present:
Audio::FindChunks->new(cache_rms => 1, filename => xxx.mp3,
stem_strip_extension => 1)->output_blocks();
# Remove start/end silence (if longer than 0.2sec):
Audio::FindChunks->new(cache_rms => 1, filename => xxx.mp3,
min_actual_silence_sec => 1e100)->split_file();
# Split a multiple-sides tape recording
Audio::FindChunks->new(filename => xxx.mp3, min_actual_silence_sec => 11
)->split_file({verbose => 1});
Audio sequence is broken into parts which contain only noise ("gaps"), and parts with usable signal ("tracks").
The following configuration settings (and defaults) are supported:
# For getting PCM flow (and if averaging data is read from cache)
frequency => 44100, # If raw_pcm or override_header_info only
bytes_per_sample => 4, # likewise
channels => 2, # likewise
sizedata => MY_INF, # likewise (how many bytes of PCM to read)
out_fh => *STDOUT, # mirror WAV/PCM to this FH if filter
# Process non-WAV data:
preprocess => {mp3 => [[qw(lame --silent --decode)], [], [-]]}, # Second contains extra args to read stdin
# RMS cache (used if valid_rms)
rms_extension => .rms, # Appended to the filestem
# Averaging to RMS info
sec_per_chunk => 0.1, # The window for taking mean square
# thresholds picking from the list of sorted 3-medians of RMS data
threshold_in_sorted_min_rel => 0, # relative position of threashold_min
threshold_in_sorted_min_sec => 1, # shifted by this amount in the list
threshold_factor_min => 1, # the list elt is multiplied by this
threshold_in_sorted_max_rel => 0.5, # likewise
threshold_in_sorted_max_sec => 0, # likewise
threshold_factor_max => 1, # likewise
threshold_ratio => 0.15, # relative position between min/max
# Chunkification: smoothification
above_thres_window => 11, # in units of chunks
above_thres_window_rel => 0.25, # fractions of chunks above threshold
# in a window to make chunk signal
# Splitting into runs of signal/noise
max_tracks => 9999, # fail if more signal/noise runs
min_signal_sec => 5, # such runs of signal are forced
min_silence_sec => 2, # likewise
ignore_signal_sec => 1, # short runs of signal are ignored
min_silence_chunks_merge (see below) # and long resulting runs of silence
# are forced
# Calculate average signal in an interval "deeply inside" silence runs
local_level_ignore_pre_sec => 0.3, # offset the start of this interval
local_level_ignore_pre_rel => 0.02, # additional relative offset
local_level_ignore_post_sec => 0.3, # likewise for end of the interval
local_level_ignore_post_rel => 0.02, # likewise
# Enlargement of signal runs: attach consequent chunks with signal this much
# above this average over the neighbour silence run
local_threshold_factor => 1.05,
# Final enlargement of runs of signal
extend_track_end_sec => 0.5, # Unconditional enlargement
extend_track_begin_sec => 0.3, # likewise
min_boundary_silence_sec => 0.2, # Ignore short silence at start/end
Note that above_thres_window is the only value specified directly in units of chunks; the other *_sec may be optionally specified in units of chunks by setting the corresponding *_chunks value. Note also that this window should better be decreased if minimal allowed silence length parameters are decreased.
These values are mirrored from other values if not explicitly specified:
min_actual_silence_sec<<less
Download (0.024MB)
Added: 2006-06-19 License: Perl Artistic License Price:
1222 downloads
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