LMMS aims to be a free alternative to popular (but commercial and
closed-source) programs like FruityLoops, Cubase and Logic giving you the
ability of producing music with your computer by creating cool loops,
synthesizing and mixing sounds, arranging samples, having more fun with your
MIDI keyboard and much more...
LMMS combines the features of a tracker/sequencer program (pattern/channel/
sample/song/effect management) and those of powerful synthesizers and samplers
in a modern, user-friendly and easy to use graphical user interface.
WWW: http://lmms.sourceforge.net/
2006-12-01 audio/xmms-rateplug: Project disappeared from the internet
2006-12-01 chinese/iiimf-le-chewing: fails to install (dependency problem)
2006-12-01 deskutils/mhc-xemacs21-mule: hangs during build
2006-12-01 devel/alleyoop: Does not compile
2006-12-01 devel/hs-crypto: is incompatible with current GHC, needs updating
2006-12-01 editors/gedit-autocomplete-plugin: Not compatible with gedit versions >= 2.14
2006-12-01 emulators/basiliskII: Does not compile
2006-12-01 emulators/vmware-tools2: Unfetchable
2006-12-01 emulators/vmware2: Unfetchable
2006-12-03 finance/ccard: Project disappeared from the internet
XAnalyser is a program to analyse a stereo audio signal. It has two displays:
Frequency Spectrum
Using Fast Fourier Transform, the time domain of the signal is transformed into
the frequency domain, i.e. the amplitude (in logarithmic scale) of the
audio signal is plotted versus the frequency. Either the sum of the
left and right channel of the audio signal can be shown or both
channels simultaneously.
XY Scope
Roughly speaking, the audio signal of left channel deflects a point
horizontally and the right channel vertically (just as the beam of a CRT
would do). Thus, an audio signal only present on the left channel produces
a horizontal line, whereas an audio signal only present on the right channel
produces a vertical line. A mono signal produces a 45 degree line.
A stereo signal creates a wilde pattern (if the phase is correct,
predominately in the same direction as a mono signal) or may even fill
the entire scope.
WWW: http://arvin.schnell-web.net/xanalyser/
PR: 105059
Submitted by: Diane Bruce <db@db.net>
Approved by: tmclaugh (mentor)
fdmf is portable perl/C software for finding pairs of music files in a
collection that are likely to contain the same music. It works on the
music itself, not on the filename, tags, or headers. It uses an audio
fingerprint, or perceptual hash to recognize the duplicate files. It is
currently under heavy development, so it might be buggy, broken, or
otherwise bad. But it works for me.
WWW: http://www.w140.com/audio/
Author: Kurt Rosenfeld <kurt at w140 dot com>
Aqualung is a music player. It plays audio files from your filesystem
and has the feature of inserting no gaps between adjacent tracks.
WWW: http://aqualung.sourceforge.net/
This is a library to make it easy to manipulate RDF files describing LADSPA
plugins.
It can also be used for general RDF manipulation.
It can read RDF/XLM and N3 files and export N3 files, it also has a light
taxonomic inference capablility.
WWW: http://sourceforge.net/projects/lrdf/
Oggz provides a simple programming interface for reading and writing
Ogg files and streams. Ogg is an interleaving data container developed
by Monty at Xiph.Org, originally to support the Ogg Vorbis audio
format.
liboggz supports the flexibility afforded by the Ogg file format while
presenting the following API niceties:
* Strict adherence to the formatting requirements of Ogg bitstreams,
to ensure that only valid bitstreams are generated
* A simple, callback based open/read/close or open/write/close interface
to raw Ogg files
* A customisable seeking abstraction for seeking on multitrack Ogg data
* A packet queue for feeding incoming packets for writing, with
callback based notification when this queue is empty
* A means of overriding the IO functions used by Oggz, for easier
integration with media frameworks and similar systems.
* A handy table structure for storing information on each logical
bitstream
WWW: http://www.annodex.net/software/liboggz/html/
channels with different rate codecs and several people on each channel.
Primarily aimed at team gamers but can be used as an IP phone as well.
WWW: http://www.ventrilo.com/
PR: ports/95071
Submitted by: Anish Mistry <amistry@am-productions.biz>
UModPlayer or Universal Module Player is a audio module "tool-chain",
providing you functions to work with modules like playing, exporting,
getting information, and more.
* You can play the supported formats and seek to any order in the
song. You have pause, timer, display, and other standard features.
* You can view the pattern notes while playing.
* Playlist support: you can create playlists, delete or move
individual items in a playlist, import a playlist from the current
directory contents, save a playlist and load a saved playlist...
* You can specify any of the ModPlug options: noise reduction,
megabass, surround, reverb sound options specifying the grade and
the delay of most of the options.
* You can export the audio data of a module to any of the supported
formats
* You can read and export to a file the song builtin message, the
song instrument names and the song sample names.
* Each user of your UNIX box can save all the sound options.
* And much more!
WWW: http://umodplayer.sourceforge.net/
LibAiff is a library for C applications, providing transparent read and
write operations for Audio Interchange File Format files.
With LibAiff your application can easily use the Audio IFF format to
interchange digital audio.
LibAiff wants to implement all the features of the AIFF 1.3 standard,
including markers, comments, etc.
This version of LibAiff supports the following features:
* Reading any valid Audio IFF file.
* Writing a valid Audio IFF file.
* Reading a compressed AIFF Compressed (AIFC) file with audio encoded
in Linear PCM, both big-endian and little-endian.
* Read & write samples in all formats supported by the Audio IFF standard.
* Convert any sample format to and from 32 bits.
* Getting and setting all the AIFF Attributes.
* Reading and writing markers to positions on the sound.
* Reading instrument data from AIFF files.
WWW: http://aifftools.sourceforge.net/libaiff/
GNUstep and Mac OS X. Similar in look and feel to XMMS, it can read the
most-known sound file formats: MP3, Ogg, FLAC, Mod, XM, AIFF, WAV and more.
Very easy to use, it integrates well with the GNUstep desktop environment
and shows a nice example of a cross-platform OpenStep application.
PR: 102901
Submitted by: Gürkan Sengün
(ALUT).
It is well suited to producing succinct demo programs and to help
new developers to get started with OpenAL without distractions
such as loading sound samples from disk.
WWW: http://www.openal.org/
PR: ports/102854
Submitted by: Jona Joachim <walkingshadow at grummel.net>
All you need to receive DRM transmissions is a PC with a sound card and a
modified analog short-wave (MW, LW) receiver.
WWW: http://drm.sourceforge.net/
PR: ports/102761
Submitted by: Soeren Straarup <xride(at)x12.dk>
the code from alsa-lib 1.0.8, necessary to support DSSI on non-ALSA
systems.
WWW: http://home.jps.net/~musound/
More information on DSSI can be found at:
WWW: http://dssi.sourceforge.net/
PR: ports/100498
Submitted by: rlazio <mahonmesr@googlemail.com>
JukeBox and Dell DJ digital audio players under BSD, Linux, Mac OS X and
Windows.
WWW: http://libnjb.sourceforge.net/
PR: ports/100462
Submitted by: adrianm <teksimian@gmail.com>
libnoise is a portable C++ library that is used to generate coherent
noise, a type of smoothly-changing noise. libnoise can generate
Perlin noise, ridged multifractal noise, and other types of
coherent-noise.
Coherent noise is often used by graphics programmers to generate
natural-looking textures, planetary terrain, and other things. The
mountain scene shown above was rendered in Terragen with a terrain
file generated by libnoise. You can also view some other examples of
what libnoise can do.
In libnoise, coherent-noise generators are encapsulated in classes
called noise modules. There are many different types of noise
modules. Some noise modules can combine or modify the outputs of
other noise modules in various ways; you can join these modules
together to generate very complex coherent noise.
WWW: http://libnoise.sourceforge.net/
you played to last.fm, formerly known as AudioScrobbler.
WWW: http://code-monkey.de/pages/xmms2-scrobbler
PR: ports/102511
Submitted by: Alexander Botero-Lowry <alex at foxybanana.com>
Features include:
* LCD for elapsed time,
* Nice display of song information
* Interfaces to the playlist and media library
PR: ports/102508
Submitted by: Alexander Botero-Lowry <alex at foxybanana.com>
and has some unique features, like chroot()'ing and dropping privileges.
This makes it an ideal application for low-end jukeboxes.
WWW: http://g-rave.nl/projects/herrie/
PR: ports/101159
Submitted by: Ed Schouten <ed at fxq.nl>
Perl Audio Converter (PAC) is a tool for converting multiple audio types
from one format to another. It supports MP2, MP3, Ogg Vorbis, FLAC,
Shorten, Monkey Audio, FAAC (AAC/M4A/MP4), Musepack (MPC), Wavpack (WV),
OptimFrog (OFR/OFS), TTA, LPAC, Kexis (KXS), AIFF, AC3, Lossless Audio
(LA), BONK, AU, SND, RAW, VOC, SMP, RealAudio (RA/RAM), WAV, and WMA. It
can also convert audio from the following video formats/extensions: RM,
RV, ASF, DivX, MPG, MKV, MPEG, AVI, MOV, OGM, QT, VCD, VOB, and WMV. A
CD ripping function with CDDB support, batch and playlist conversion,
tag preservation for most supported formats, independent tag reading/
writing, and extensions for Konqueror and Amarok are also provided.
WWW: http://pacpl.sourceforge.net/
Author: Philip Lyons <viiron@gmail.com>
OptimFROG is a lossless audio compression program. Its main goal is to
reduce at maximum the size of audio files, while permitting bit identical
restoration for all input. It is similar with the ZIP compression, but it
is highly specialized to compress audio data.
OptimFROG obtains asymptotically the best lossless audio compression
ratios. It has Windows, Linux, and Mac versions, fully featured input
plug-ins for the Windows Media Player, foobar2000, Winamp2/3/5, dBpowerAMP,
XMPlay, QCD, and XMMS audio players (with bitstream error resilience,
ID3v1.1 and APEv2 read tagging support, ID3v2 compatible), optimal support
for all integer PCM wave formats up to 32 bits and an extensible streamable
(error tolerant) compressed format. It is also fast, the default mode
encodes CD quality audio data at 12.4x real-time and decodes at 17.4x real-
time on AMD Athlon XP 1800+ (the fastest mode encodes at 28.1x real-time
and decodes at 24.7x real-time). Self-extracting (sfx) archives can also be
created with a small overhead of just 54 KB.
WWW: http://www.losslessaudio.org/
Author: Florin Ghido <FlorinGhido@yahoo.com>
LPAC is a codec (coder / decoder) for lossless compression of digital audio
files. "Lossless" means that any compressed file can be decompressed in a way
it will be bit-wise identical with the original. This is the main advantage
of LPAC compared to lossy formats like MP3, WMA or RealAudio. On the other
hand, lossy codecs can achieve higher compression ratios. For example, MP3 at
128 kbit/s achieves a (fixed) compression ratio of 11, whereas LPAC's
compression ratios range from 1.5 to 4, strongly depending on the audio
material. Typically they are around 2 for pop music and 2.5 for classical
music. This may not seem much, but remember you will get back every single
bit, no matter how often you subsequently compress and decompress a file. It
is true that general archivers (Zip, LZH, gzip) are lossless, too, but they
often achieve nearly no compression on audio files.
WWW: http://www.nue.tu-berlin.de/wer/liebchen/lpac.html
Kexis - A lossless WAV file compressor. Kexis' main goal is to develop
prediction and encoding schemes to minimize compressed file size. Kexis
strives to be the premier lossless sound encoder.
WWW: http://sourceforge.net/projects/kexis/
This module returns a hash containing basic information about a
Musepack file, as well as tag information contained in the Musepack
file's APE tags. See Audio::APETags for more information about the
tags.
WWW: http://search.cpan.org/dist/Audio-Musepack/
This module returns a hash containing basic information about a FLAC file,
a representation of the embedded cue sheet if one exists, as well as tag
information contained in the FLAC file's Vorbis tags. There is no complete
list of tag keys for Vorbis tags, as they can be defined by the user; the
basic set of tags used for FLAC files include:
* ALBUM
* ARTIST
* TITLE
* DATE
* GENRE
* TRACKNUMBER
* COMMENT
WWW: http://search.cpan.org/dist/Audio-FLAC-Header/
A Crossfading/Gapless Output Plugin featuring:
* Crossfading
* Fadein/Fadeout
* Continuous output
* Gap-Killer
* Automatic detection of live albums
* High quality
* Secondary effect plugin
* Compatibility with bmp and audacious
WWW: http://www.eisenlohr.org/xmms-crossfade/index.html
Author: Peter Eisenlohr <p.eisenlohr@gmx.net>
This is a slave port of audio/xmms-crossfade
submits information about tracks being played to audioscrobbler.
WWW: http://www.frob.nl/scribble.html
PR: ports/100195
Submitted by: Stepan Zastupov [RedChrom] <redchrom at gmail.com>
You can ask for the results to be sorted by one or more of those tags,
and return either the list of filenames (the deault), a printf-style
formatted string for each file using its ID3 tags, or the actual Perl
data structure representing the results.
WWW: http://search.cpan.org/dist/MP3-Find/
PR: ports/100149
Submitted by: Jin-Shan Tseng <tjs at cdpa.nsysu.edu.tw>
formats (M3U, PLS, HTML, etc).
It is very usefull if you have a large amount of audio files and you want to
quickly and frequently build a playlist.
WWW: http://royale.zerezo.com/fapg/
PR: ports/99300
Submitted by: chinsan <chinsan.tw@gmail.com>
It acts as a frontend to XMMS2.
Author: Tilman Sauerbeck <tilman@code-monkey.de>
WWW: http://www.enlightenment.org/
PR: ports/99318
Submitted by: Stanislav Sedov <ssedov@mbsd.msk.ru>
