NOTE: Automatically created during Factory devel project migration by admin.

NOTE: Automatically created during Factory devel project migration by admin.

Aubio is a library for real time audio labelling. Its features include
segmenting a sound file before each of its attacks, performing pitch
detection, tapping the beat and producing midi streams from live audio.
The name aubio comes from 'audio' with a typo: several transcription
errors are likely to be found in the results too.

Disposable Soft Synth Interface (DSSI, pronounced "dizzy") is a
proposal for a plug-in API for software instruments (soft synths) with
user interfaces, permitting them to be hosted in-process by Linux audio
applications. Think of it as LADSPA-for-instruments or something
comparable to a simpler version of VSTi.

FFTW is a C subroutine library for computing the Discrete Fourier
Transform (DFT) in one or more dimensions, of both real and complex
data, and of arbitrary input size.

FLAC is an open source lossless audio codec developed by Josh Coalson.

FluidSynth (formerly IIWU Synth) is based on the SoundFont(tm) 2
specifications. It is a real-time "software synthesizer". FluidSynth
can read MIDI events from the MIDI input device and render them to the
audio device. It can also play MIDI files.

FluidSynth-DSSI is the plug-in implementation of fluidsynth on DSSI
(Disposable Soft Synth Interface) with a GTK+ GUI.

JACK is a low-latency audio server written primarily for the Linux
operating system. It can connect a number of different applications to
an audio device, as well as allow them to share audio between
themselves. Its clients can run in their own processes (as a normal
application), or they can run within a JACK server (as a plug-in).

The Linux Audio Developer's Simple Plug-in API (LADSPA) provides the
ability to write simple plug-in audio processors in C/C++ and link them
dynamically. The package contains the LADSPA SDK, the CMT plug-in
libraries (including Freeverb), and the swh plug-ins.

LASH (formerly LADCCA) is a session management system for JACK and ALSA
audio applications on GNU/Linux. Its aim is to allow you to have many
different audio programs running at once and to save the setup, close
them down, then reload the setup at some other time.

This package is based on the package 'lcms' from project 'openSUSE:Factory'.

Little cms is a small speed optimized color management engine.

Libao is a cross-platform audio output library. It currently supports
ESD, OSS, Solaris, and IRIX.

This package is based on the package 'libcdaudio' from project 'openSUSE:Factory'.

libcdaudio is a library designed to provide functions to control
operation of a CD-ROM when playing audio CDs. It also contains
functions for CDDB and CD index lookup.

libid3tag is a library for reading and writing ID3 tags, both ID3v1 and
the various versions of ID3v2.

This is a lightweight, easy-to-use implementation of the OSC protocol
(see http://www.cnmat.berkeley.edu/OpenSoundControl/ for details).

This is a library to make it easy to manipulate RDF files describing
LADSPA plug-ins. It can also be used for general RDF manipulation.

It can read RDF, XLM, and N3 files and export N3 files. Ot also has a
light taxonomic inference capability.

Libogg is a library for manipulating ogg bitstreams. It handles both
making ogg bitstreams and getting packets from ogg bitstreams.

Ogg is the native bitstream format of the libvorbis (Ogg Vorbis audio
codec ) and the libtheora (Theora video codec)

liboggz is a library that provides simple parsing and seeking of files
and streams based on the Ogg file format. liboggz requires libogg to
work.

liboggz knows about Ogg speex, Ogg vorbis, Ogg theora, and the Ogg
based Annodex formats, thus allows parsing (though not decoding) of
these files.

Authors:
--------
Conrad Parker
Silvia Pfeiffer
Andre Pang
Zentaro Kavanagh
David Kuehling

Secret Rabbit Code (aka libsamplerate) is a Sample Rate Converter for
audio. One example of where such a thing would be useful is in
converting audio from the CD sample rate of 44.1kHz to the 48kHz sample
rate used by DAT players.

SRC is capable of arbitrary and time varying conversions; from
downsampling by a factor of 12 to upsampling by the same factor. The
conversion ratio can also vary with time for speeding up and slowing
down effects.