Free amp software for linux

orange amp xmms is rebuilt from scratch orange version of my black amp theme.

Now "xmms" button opens preferences for xmms {I meant to do that for black amp but forgot}.

The Asterisk@Home Linux Distribution project enables the home user to quickly set up a VOIP Asterisk PBX. A web GUI makes configuration and operation easy. We also provide an xPL (home automation) interface for easy interaction with other devices in the home.
Main features:
- Asterisk 1.0.9
- Flash Operator Panel 0.21
- Festival Speech Engine version 1.95
- weather agi scripts
- wakeup calls
- Integrated WebMeetMe GUI
- AMP-1.10.008
- CentOS 3.5
- SugarCRM with Cisco XML Services interface + Click to Dial
- Music On Hold (mpg123)
- Fax support (spanDSP)
- xPL support
- Digium card auto-config
AMP (Asterisk Management Portal)
- AMP is a web GUI that allows the configuration of Asterisk without the need for editing config files. If you use AMP be careful not to make manual changes to the config files that will affect AMP?s operation.
Flash Operator Panel
- Flash Operator Panel is a real-time web interface to Asterisk. You can see what all of your extensions, trunks, and conferences are doing. We have extension 200 set up for testing. This is a live extension and will work as soon as a phone is registered to ext 200. The rest of the buttons are dummy objects for examples.
- The op_buttons.conf file has all of the configs for each button in Flash Operator Panel you can edit this using the Config Edit button on the maint menu. You must reboot the Asterisk or restart the panel service after you make changes. The normal config files for Flash Operator Panel end in .cfg we changed them to .conf so they would match the other Asterisk files.
Cisco XML Services
- Asterisk@Home has a Cisco XML Services that can be used to look up contacts in a database. Click on Web Address Book from the Asterisk@Home main menu. (user: admin password: password) and enter all of your contacts and their phone numbers.
Music On Hold (mpg123)
- Asterisk@Home has now uses mpg123 for music on hold. Put a call on hold and they hear music not dead air!
Fax support (SpanDSP)
- SpanDSP allows Asterisk to receive faxes. You can set this up in AMP.

Nosefart is a player for NES Sound Format (NSF) files (which consist of audio data ripped from Nintendo Entertainment System games).
It runs as a standalone player in Linux, with both command line and GTK interfaces, as well has having plugins for Winamp, XMMS, and CL-amp.
Enhancements:
- The file selection dialog now remembers what directory you used in the last session.
- If esd (or something else) is blocking the sound card, it now pops up an error rather than just hanging.

CAMP is a complete interface for MP3 players such as mpg123, xaudio, and amp. It is designed for the console using ANSI graphics, and also supports skins.
To compile and install the camp-source, you should type the following commands:
./configure
make
make install
cp skins.tar ~/.camp
cd ~/.camp
tar xvf skins.tar
Make sure to edit "/.camp/camprc" in your homedirectory to point the player variable to mpg123 or any other command-line mp3 player. CAMP is known to compile and run under any linux distribution. older versions did also compile under SunOS, but I havent cared to test newer version. It does also compile under *BSD, but i think the keys are fucked-up, and I will use ncurses for reading keys, as soon as I feel like rewriting that.
Enhancements:
- Fixed a stupid bug wich fucked up the playlist-search (my bad)
- Fixed a bug (segfault) with filenames w/o id3, whith a lenght >100 chars
- Fixed IR (lircd) remote support in background mode.
- Fixed lots of small segfault bugs.
- mpg123 jukebox mode now added and complete.
- Added some more skin stuff, once again, and a new skin leetamp

Audio::Data is a module for representing audio data to perl.

SYNOPSIS

use Audio::Data;
my $audio = Audio::Data->new(rate => , ...);

$audio->method(...)

$audio OP ...

Audio::Data represents audio data to perl in a fairly compact and efficient manner using C via XS to hold data as a C array of float values. The use of float avoids many issues with dynamic range, and typical float has 24-bit mantissa so quantization noise should be acceptable. Many machines have floating point hardware these days, and in such cases operations on float should be as fast or faster than some kind of "scaled integer".

Nominally data is expected to be between +1.0 and -1.0 - although only code which interacts with outside world (reading/writing files or devices) really cares.
It can also represent elements (samples) which are "complex numbers" which simplifies many Digital Signal Processing methods.

Methods

The interface is object-oriented, and provides the methods below.
$audio = Audio::Data->new([method => value ...])
The "constructor" - takes a list of method/value pairs and calls $audio->method(value) on the object in order. Typically first "method" will be rate to set sampling rate of the object.

$rate = $audio->rate
Get sampling rate of object.
$audio->rate($newrate)
Set sampling rate of the object. If object contains existing data it is re-sampled to the new rate. (Code to do this was derived from a now dated version of sox.)
$audio->comment($string)
Sets simple string comment associated with data.
$string = $audio->comment
Get the comment
$time = $audio->duration
Return duration of object (in seconds).
$time = $audio->samples
Return number of samples in the object.
@data = $audio->data
Return data as list of values - not recommended for large data.
$audio->data(@data)
Sets elements from @data.
$audio->length($N)
Set number of samples to $N - tuncating or padding with zeros (silence).
($max,$min) = $audio->bounds([$start_time[,$end_time]])
Returns a list of two values representing the limits of the values between the two times if $end_time isnt specified it defaults to the duration of the object, and if start time isnt specified it defaults to zero.
$copy = $audio->clone
Creates copy of data carrying over sample rate and complex-ness of data.
$slice = $audio->timerange($start_time,$end_time);
Returns a time-slice between specified times.
$audio->Load($fh)
Reads Sun/NeXT .au data from the perl file handle (which should have binmode() applied to it.)
This will eventually change - to allow it to load other formats and perhaps to return list of Audio::Data objects to represnt multiple channels (e.g. stereo).
$audio->Save($fh[,$comment])
Write a Sun/NeXT .au file to perl file handle. $comment if specified is used as the comment.
$audio->tone($freq,$dur,$amp);
Append a sinusoidal tone of specified freqency (in Hz) and duration (in seconds), and peak amplitude $amp.
$audio->silence($dur);
Append a period of 0 value of specified duration.
$audio->noise($dur,$amp);
Append burst of (white) noise of specified duration and peak amplitude.
$window = $audio->hamming($SIZE,$start_sample[,$k])
Returns a "raised cosine window" sample of $SIZE samples starting at specified sample. If $k is specified it overrides the default value of 0.46 (e.g. a value of 0.5 would give a Hanning window as opposed to a Hamming window.)
windowed = ((1.0-k)+k*cos(x*PI))
$freq = $audio->fft($SIZE)
$time = $freq->ifft($SIZE);
Perform a Fast Fourier Transform (or its inverse). (Note that in general result of these methods have complex numbers as the elements. $SIZE should be a power-of-two (if it isnt next larger power of two is used). Data is padded with zeros as necessary to get to $SIZE samples.
@values = $audio->amplitude([$N[,$count]])
Return values of amplitude for sample $N..$N+$count inclusive. if $N is not specified it defaults to zero. If $count is not specified it defaults to 1 for scalar context and rest-of-data in array context.
@values = $audio->dB([$N[,$count]])
Return amplitude - in deci-Bells. (0dB is 1/2**15 i.e. least detectable value to 16-bit device.) Defaults as for amplitude.
@values = $audio->phase([$N [,$count]])
Return Phase - (if data are real returns 0). Defaults as for amplitude.
$diff = $audio->difference
Returns the first difference between successive elements of the data - so result is one sample shorter. This is a simple high-pass filter and is much used to remove DC offsets.
$Avalues = $audio->lpc($NUM_POLES,[$auto [,$refl]])
Perform Linear Predictive Coding analysis of $audio and return coefficents of resulting All-Pole filter. 0th Element is not a filter coefficent (there is no A[0] in such a filter) - but is a measure of the "error" in the matching process. $auto is an output argument and returns computed autocorrelation. $refl is also output and are so-called reflection coefficents used in "lattice" realization of the filter. (Code for this lifted from "Festival" speech systems speech_tools.)
$auto = $audio->autocorrelation($LENGTH)
Returns an (unscaled) autocorrelation function - can be used to cause peaks when data is periodic - and is used as a precursor to LPC analysis.

GlobeCom Jukebox is a jukebox with integrated CDDB aware ripping and groupware functionality. Youre probably familiar with MP3 files and perhaps already have a collection of music in MP3 format that you play on a desktop computer using Winamp, MusicMatch or similar. So why consider an MP3 jukebox ?

1. A Jukebox will place all your MP3 files in a central location. Its easier to catalog, locate and play the files from a single point of access if you use a dedicated Jukebox.
2. You can put the Jukebox in a closet and run it without a monitor. You can listen to your music even when your PC is turned off but can still control is from your PC when you want to.
3. You can have a single Jukebox serving music to the entire office or house.
4. You can put the Jukebox near the stereo and take the soundcard output direct to your high quality stereo amp.
5. The Jukebox runs on Linux. After your PC running Winamp crashes for the third time that day, you will appreciate a music system that runs for months or years without interruption.

If you would like to encode ripped CDs directly to OGG instead of MP3, Once you have the Jukebox up and running log in as an administrator.

Go to Administrate and then to Conversion Programs
1.) change the priority of wav to mp3 to -100
2.) change the priority of wav to ogg to 30