Free audio moosic software for linux

Audio Daemon is a perl module to daemonize various players and support a single UDP interace for different players. Currently you can set up a server for MPG123, Xmms and an icecast stream (using libshout).

Audio::Daemon is made of two parts, the client and server.

The Client (Audio::Daemon::Client) remains the same across the board no matter what server you are communicating with, with a few exceptions noted in the pods.

The Server currently can be configured to be MPG123, Xmms or an Icecast (libshout) stream. Neither the Client nor the Servers need to be run as root and I strongly advise against it.

Audio::Daemon::MPG132:
Requires Audio::Play::MPG123 an optionally Audio::Mixer for volume control. It automatically spawns mpg123 in the background so there is no need to prestart it.

Audio::Daemon::Xmms:
Requires the Xmms and MP3::Info perl module and xmms installed. Id suggest getting the latest version of the Xmms perl module. It uses the volume and random feature built into xmms and if it doesnt find an xmms running it will try to spawn its own copy. Bear in mind that
xmms MUST HAVE an X display to export to, and that xmms must be running as the same user Audio::Daemon::Xmms is.

Audio::Daemon::Shout:
This wasnt easy to do. It requires MP3::Info and libshout to be installed (http://developer.icecast.org/libshout/) as well as a functioning icecast server. The timing here is pretty critical so I recomend being careful not to be polling the server too much. Be sure to read the icecast docs as Im still confused on how I got this part to work.

Moosic is a music player for Unix systems. It focuses on convenient and powerful playlist management. This project consists of a server process that maintains a queue of songs to be played and one or more client programs which sends commands to the server.

The server continually runs through its song queue, popping songs off the top of the list and playing each with an external program. The client is a simple command-line utility which allows you to easily perform powerful operations upon the servers queue, including the addition of whole directory trees, automatic shuffling, and song removal according to regular expressions.

The server comes configured to play MP3, Ogg, MIDI, MOD, and WAV files, but can be configured to also play any other file format you want.

Audio::OSS is a pure-perl interface to OSS (open sound system) audio devices.

SYNOPSIS

use Audio::OSS qw(:funcs :formats :mixer);

my $dsp = IO::Handle->new(" dsp_reset($dsp) or die "reset failed: $!";

my $mask = get_supported_formats($dsp);
if ($mask & AFMT_S16_LE) {
set_fmt($dsp, AFMT_S16_LE) or die set format failed: $!";
}
my $current_format = set_fmt($dsp, AFMT_QUERY);

my $sps_actual = set_sps($dsp, 16000);

set_fragment($dsp, $fragshift, $nfrags);
my ($frags_avail, $frags_total, $fragsize, $bytes_avail)
= get_outbuf_info($dsp);
my ($bytes, $blocks, $dma_ptr) = get_outbuf_ptr($dsp);

my $mixer = IO::Handle->new(" my $miclevel = mixer_read($mixer, SOUND_MIXER_MIC);

Audio::OSS is a pure Perl interface to the Open Sound System, as used on Linux, FreeBSD, and other Unix systems.

It provides a procedural interface based around filehandles opened on the audio device (usually /dev/dsp* for PCM audio).

It also defines constants for various ioctl calls and other things based on the OSS system header files, so you dont have to rely on .ph files that may or may be correct or even present on your system.

Currently, only the PCM audio input and output functions are supported. Mixer support is likely in the future, sequencer support less likely.

Audio::MFCC is a Perl module for computing mel-frequency cepstral coefficients.

SYNOPSIS

use Audio::MFCC;
my $fe = Audio::MFCC->init(%params)
$fe->start_utt;
my @ceps = $fe->process_utt($rawdata, $nsamps);
my $leftover = $fe->end_utt;

This module provides an interface to the Sphinx-II feature extraction library which can be used to extract mel-frequency cepstral coefficients from data. These coefficients can then be passed to the Speech::Recognizer::SPX::uttproc_cepdata function.
You might find this useful if, for example, you wish to do the actual recognition on a different machine from the audio capture, and dont have the bandwidth to send a full stream of audio data over the network.

Currently, Sphinx-II also uses delta and double-delta cepstral vectors as input to its vector quantization module, but the calculation of these values is done inside the recognizers utterance processing module.. In the future it may be possible to move the extraction of these features into the feature extraction library, or to use entirely different features as input (for example, LPC coefficients, though currently, mel-scale cepstra give the best recognition performance).

Audio::Ao is a Perl wrapper for the Ao audio library.

SYNOPSIS

use Audio::Ao qw(:all);

initialize_ao;
my $device = open_live(default_driver_id(), 16, $rate, $channels,
is_big_endian(), {});
while (#have data) {
play($device, $data_buffer, $len_of_buffer);
}
close_ao($device($device));
shutdown_ao;

Provides access to Libao, "a cross-platform library that allows programs to output PCM audio data to the native audio devices on a wide variety of platforms." Libao currently supports OSS, ESD, ALSA, Sun audio, and aRts.

Audio::DB are tools for generating relational databases of MP3s.

SYNOPSIS

use Audio::DB;
my $mp3 = Audio::DB->new(-user =>user,
-pass =>password,
-host =>db_host,
-dsn =>music_db,
-adaptor => mysql);

$mp3->initialize(1);

$mp3->load_database(-dirs =>[/path/to/MP3s/],
-tmp =>/tmp);

Audio::DB is a module for creating relational databases of MP3 files directly from data stored in ID3 tags or from flatfiles of information of track information. Once created, Audio::DB provides various methods for creating reports and web pages of your collection.

Although its nutritious and delicious on its own, Audio::DB was created for use with Apache::Audio::DB, a subclass of Apache::MP3. This module makes it easy to make your collection web-accessible, complete with browsing, searching, streaming, multiple users, playlists, ratings, and more!

Audio::ESD is a Perl extension for talking to the Enlightened Sound Daemon.

SYNOPSIS

use Audio::ESD;
my $stream = Audio::ESD->play_stream({ # these are the defaults
sample_rate => 16000,
channels => 1,
fallback => 0,
bits_sample => 16,
encoding => linear })
or die "Failed to open ESD stream: $!n";
print $stream $data; # etcetera

This module provides a Perl wrapper around the Enlightened Sound Daemons client library. Input, output, and monitoring streams are supported, as well as some (but not all) of the control functions. Samples are supported but untested.