Free ladspa software for linux

Implements a LADSPA 1.1 host.

SYNOPSIS

use Audio::LADSPA;

for my $class (Audio::LADSPA->plugins) {
print "t",$class->name," (",$class->id,"/",$class->label,")";
}

LADSPA plugins are objects in shared libraries that can generate or transform audio signals (like VST or Direct-X plugins on Mac and Win32 systems). Most of the existing LADSPA plugins are pretty low-level compared to VST plugins (you get seperate oscilator, ADSR and delay plugins instead of "complete" virtual synthesizers etc). See also http://www.ladspa.org/

With these modules you can create a LADSPA host, which can load the plugins, query their capabilities, connect them together in a network, and run audio streams through them.

The LADSPA API was developed for linux but should be platform independent, so you might be able to compile these modules and the nessisary plugins on win32 systems (please let me know if it works or not).

Audio::LADSPA::Buffer is a Perl module for LADSPA buffer.

SYNOPSIS

use Audio::LADSPA;

my $buffer = Audio::LADSPA::Buffer->new($size);

$plugin->connect(Port name => $buffer);

$buffer->set( @values );

# or get a buffer from a plugin..

$buffer2 = $plugin->get_buffer(Other port);

# make audio buffer louder

$buffer *= 2;

Audio::LADSPA::Buffer objects implement the connections between Audio::LADSA::Plugin objects. You can set up the buffer, and let some plugin write to it, while others read from it, or read from or write to it yourself.

There is no real difference between audio and control buffers, except that control buffers have a size of 1 sample *) and audio buffers are usually bigger.
*) Samples in LADSPA are implemented as native floats.

FreeADSP stands for Free Audio Digital Signals Processor and is a free audio-oriented real-time DSP software. What it does is taking a signal as input, transform it through arbitrary-disposed modules, and then putting the result back to some output (that is, for example, the same thing that regular guitar stomp boxes do).

What are these modules?

Unlike similar applications which are designed to use only one or two kinds of audio computational plugins, FreeADSP is capable of putting to work everything that could be used to do a computation (for example a circuital representation, a mathematical formula or a neural net). This is possible because FreeADSP uses external module loader plugins which know how to make things work properly.

So, what can this program do?

The main purpose of this software is processing sounds at real-time. While there are already a lot of packages around that can do it, often with some sensible limitation, FreeADSP aims at being THE program of real-time sound processing. To achive this goal its design is extremely modular, and this is what makes it capable of doing unbelivable things, at least theorically.

As said before, the possibility of using external plugins to load other stuff to process data means the possibility of using potentially everything that can do a computation, from well-known LADSPA, DSSI or VST/VSTi plugins to neural nets, also simultaneously, and in an extremely clean fashion.

The whole I/O and user interface parts are handled by external plugins too. This means again a lot more flexibility and also easier portability and integration on different OSes and/or hardware architectures (also thanks to FreeADSPs high configurability)! Furthermore an user could easily switch user interface without having to put the program and his work into the trash can.

The I/O is handled in such a generic way that one can use every kind of input or signal to interact either with modules (for example, one could use a breath control to manipulate the output volume of a fretless bass, obtaining a wind instrument-like feel) or with FreeADSP itself (start/stop processing, etc.)

Last but no least, FreeADSP also lets you manage collections of presets, eventually organised into banks and record what you play at each stage of processing.

CLAM is a multiplatform software framework for research and application development for the Audio and Music domains.
It offers a conceptual model for building systems as well as tools for the synthesis, analysis, and transformation of audio signals using high level representation (notes, spectral peaks, etc.).
It also provides multiplatform system abstraction and tools for audio and MIDI file handling, device handling, GUI and XML support, and more. It has proper Linux audio architecture integration (Ladspa, Alsa, OSC, Jack, VST, etc.) and rapid graphical prototyping both for the processing core and the user interface (Qt designer integration).
Enhancements:
- More appealing audio widgets were provided for use in user applications.
- GenderChange processing is now working in real-time.
- Chord extraction and visualization has been greatly optimized.
- Multiple NetworkEditor processings can be moved at once using multiple selection, and control senders can be created by double clicking the receiver control connector.
- Initial Faust support and better Ladspa hosting were provided.

Glame is a multi-track wave-editor with synthesis capabilities.

Glame is targeted to be the GIMP for audio processing. Glame supports non destructive multitrack editing, recording and mixing; realtime effects using native or LADSPA plugins; OSS, ALSA or SGI audio and WAV, Mp3 and Ogg file formats.

LDRUM is an open-source drummachine that offers ten channels, realtime control, a simple pattern sequencer and a graphical user-interface.
LDrum currently runs under Linux only, it is developed in C++ and uses JACK, Qt, ALSA and LADSPA (its a JAQL app;)
LDRUM uses two homemade libraries called libjackpp and libladspapp. These are included in the LDRUM tarball but might be of interest standalone for somebody. So use the following links below to get some infos and the source.
Main features:
10 stereo channels
- Each channel can play a stereo sample. The sample can be changed while playing. A channel can be triggered by the internal sequencer, the keyboard, MIDI or the GUI simultaneous. You can use your favorite sequencer to trigger LDRUM.
16 voices
- LDRUM is limited to 16 channels to be played simultaneous.
MIDI support
- The LDRUM uses the ALSA sequencer api for triggering channels and controlling parameter via MIDI.
Realtime controlable parameters
- Each channel offers a set of parameters which can be controlled in realtime via MIDI or the graphical user interface. You can assign individual MIDI controllers to each parameter via an easy to use MIDI learn function (just right-click over the parameter).
The following parameters are available:
- mute
- solo
- level
- level velocity modulation
- panorama
- equalizer low freq
- equalizer mid freq
- equalizer high freq
- equalizer velocity modulation
- length
- length velocity modulation
- pitch
Sample import
- LDRUM can import all audio file types libsndfile supports.
Load/save drum patches
- A LDRUM patch is stored in an xml file with the extension ".ldp". A patch stores all channel parameters, the channels samples and the MIDI note assignments. Patches can be loaded and saved via the GUI.
Load/save sequences
- A LDRUM sequence is stored in a xml file with the extension ".lds". A sequence stores the sequencer banks which contain the patterns of the channels. Sequences can be loaded and saved via the GUI.
Color schemes
- The LDRUM can be colored the way you like it. It offers an graphical color scheme editor to create you own personal look.

Ecawave is a simple graphical audio file editor. The user-interface is based on Qt libraries, while almost all audio functionality is taken directly from Ecasound libraries. As ecawave is designed for editing large audio files, all processing is done direct-to-disk. Simple waveform caching is used to speed-up file operations.

Ecawave supports all audio file formats and effect algorithms provided by Ecasound libraries. This includes JACK, ALSA, OSS, aRts, over 20 file formats, over 30 effect types, LADSPA plugins and multi-operator effect presets.

First of all, you should first try to use the binary-rpm release. Its probably the easiest and the most straigthforward way to install this software.

If you need to compile ecawave, the file INSTTALL will give you the instructions.

JACK Rack is an effects "rack" for the JACK low latency audio API.

The rack can be filled with LADSPA effects plugins and can be controlled using the ALSA sequencer. JACK Rack is phat; it turns your computer into an effects box.