Free streaming audio software for linux

FFmpeg::Stream::Audio is an audio stream from a (multi)media stream group.

SYNOPSIS

$ff = FFmpeg->new(); #see FFmpeg
#...
$sg = $ff->create_streamgroup(); #see FFmpeg
$st = ($sg->streams())[0]; #this is a FFmpeg::Stream

Objects of this class are not intended to be instantiated directly by the end user. Access FFmpeg::Stream::Audio objects using methods in FFmpeg::StreamGroup. See FFmpeg::StreamGroup for more information.

This class represents an audio stream in a multimedia stream group, and has audio-specific attributes. General stream attributes can be found in the FFmpeg::Stream class.

streamixer contains an audio stream mixer server, tools for monitoring and feeding the server, and a stream resampling tool with various different interpolation models.
mixer is a stream mixing server.
resample v1.20.0 - Copyright (C) 1992,2006 Bisqwit (http://iki.fi/bisqwit/)
Portions copyright (C) 2001 ImageMagick Studio.
This program is under GPL. streamixer-1.20.0.tar.{gz,bz2}
are available at the homepage of the author.
Usage: /usr/local/bin/resample [ ...]
Reads stdin, writes stdout.
input- and outputspec:
b=8-bit, m=mono, rxxx=xxx Hz, examples:
"" = 16-bit 44100 Hz stereo
m = 16-bit 44100 Hz mono
b = 8-bit 44100 Hz stereo
mb = 8-bit 44100 Hz mono
mr22050 = 16-bit 22050 Hz mono
br22050 = 8-bit 22050 Hz stereo
etc. - must not be used.
Options:
--help, -h Help
--version, -V Version information
--blur, -b Blur: 0=normal, >0=blur (cpu intensive)
--lflip, -L Flip left channel
--rflip, -R Flip right channel
--ldouble Multiply left channel amplitude by 2
--rdouble Multiply right channel amplitude by 2
--sharpen, -s Equivalent to giving negative number to --blur
--filter, -f Select filter (interpolation scheme)
Available filters:
none, linear, triangle, hermite, hanning, hamming,
blackman, gaussian, quadratic, cubic, catrom, mitchell,
lanczos, bessel, sinc
Recommended filters: linear, lanczos, none
Enhancements:
- This release fixes the programs to work on the x86_64 architecture.
- The latency of the resample program was decreased by decreasing the buffer sizes.

PicoStreamer is a complete server for live audio/video streaming. It can stream MP3, AAC, and video live using Flash FLV. PicoStreamer does not require any Flash server. It only needs a Web server with Perl and PHP. It requires neither expensive dedicated servers nor compilation on the target machine.
A source client (for Win32) is included. A management/playing framework written in PHP makes PicoStreamer simple to use. The PicoIcy component acts as a local shoutcast server emulator, so you can use your preferred shoutcast source client to stream or stream Flash format directly from the Web.
Enhancements:
- Added live flash audio/video streaming support.
- Added Flash encoder.
- Added Flash player also for mp3.
- Added Brasilian (PT) language.
- Added dynamic playlist chooser.
- Added content-length managing.
- PicoIcy now in tray-bar.
- Fixed list blocking on weird chars.
- Removed php ctype_digits dependence.
- Removed perl html::entitites dependence.
- Fixed many minor bugs.

MP3 Streaming DownSampler for PHP is written with cable modem/DSL users with low upstream caps in mind. Using LAME, it dynamically downsamples a requested MP3 to a bitrate which can be streamed.

It doesnt use temporary files, so you dont have to take the time to do any batching, and you just give it a directory with symlinks to your music in a place the Web server can reach. The interface is simplistic and fast, meant for immediate plug and play access.

Mediastreamer is library written in C that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications.

It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM), video codecs (MPEG4, H263, Theora), I/O from soundcards, wav files, webcams, echo-cancelation, conferencing, and various other utilities.

Mediastreamer has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.

Audiere is a high-level audio API. It can play Ogg Vorbis, MP3, FLAC, uncompressed WAV, AIFF, MOD, S3M, XM, and IT files. For audio output, Audiere supports DirectSound or WinMM in Windows, OSS on Linux and Cygwin, and SGI AL on IRIX.
Audiere is open source and licensed under the LGPL. This means that you may freely use Audiere in commercial products, as long as you do not modify the source code. If you do modify Audiere and release a product that uses your modifications, you must release your changes to the code under the LGPL as well.
Audiere is portable. It is tested on Windows, Linux-i386, Cygwin, and IRIX with at least three major compilers. Most of Audiere is endian-independent, so it would work with few modifications on other architectures.
Main features:
- Braindead easy API
- Supported file formats: Uncompressed WAV*, Uncompressed AIFF*, Ogg Vorbis*, FLAC*, MP3, MOD, S3M, IT, XM (* supports seeking)
- Streaming and buffered audio
- Volume, pan, and pitch shift modification
- Flat tone, square wave, white noise, and pink noise generation
- Runtime enumeration of audio devices and supported file formats
- Custom file streams
- Python, Delphi, Java, XPCOM (JavaScript in Mozilla) bindings
Enhancements:
- Replaced mpegsound with a stand-alone version of the MPEG audio decoder from ffmpegs libavcodec library. The result is an MP3 decoder that is more portable and works with more MP3 files. MP3 files are now seekable as well. (Matt Campbell)
- Added Speex support.
- Added support for reading metadata tags from sample sources. So far, Vorbis comments are supported, as are ID3v1 and ID3v1.1 tags in MP3 files. Interface designed with help from Brian Robb and Andy Friesen.
- Added callback system for stream stop events. (Richard Schaaf and Chad Austin)
- Added CD audio support, using the MCI subsystem on Win32/Cygwin and libcdaudio on Linux. (Chad Austin and Richard Schaaf)
- Added MIDI support through the MCI subsystem on Win32 and Cygwin. (Chad Austin)
- Dramatically reduced the latency of the OSS device. (Matt Campbell)
- Rewrote the Resampler class to use DUMBs cubic interpolation resampler, resulting in much better resampling for devices such as OSS that use Audieres own mixer. (Matt Campbell)
- Added bindings to wxWidgets. (Emanuel Dejanu)
- Fixed a bug in the DirectSound device implementation which significantly slowed down opening of devices and buffers. (Matt Campbell)
- Added pitchshift to the Python bindings. (Jason Chu)
- Split Doxygen documentation into one for users and one for developers.

JAMin is the JACK Audio Connection Kit (JACK) Audio Mastering interface. JAMin is an open source application designed to perform professional audio mastering of stereo input streams.
JAMin project uses LADSPA for digital signal processing (DSP).
Main features:
- Linear filters
- JACK I/O
- 30 band graphic EQ
- 1023 band hand drawn EQ with parametric controls
- Spectrum analyser
- 3 band peak compressor
- Lookahead brickwall limiter
- Multiband stereo processing
- Presets and scenes
- Loudness maximiser

MPEG::Audio::Frame is a class for weeding out MPEG audio frames out of a file handle.

SYNOPSIS

use MPEG::Audio::Frame;

open FILE,"file.mp3";

while(my $frame = MPEG::Audio::Frame->read(*FILE)){
print $frame->offset(), ": ", $frame->bitrate(), "Kbps/", $frame->sample()/1000, "KHzn"; # or something.
}

A very simple, pure Perl module which allows parsing out data from mp3 files, or streams, and chunking them up into different frames. You can use this to accurately determine the length of an mp3, filter nonaudio data, or chunk up the file for streaming via datagram. Virtually anything is possible.