Free realmedia vbr software for linux

mpgedit is an MPEG 1 layer 1/2/3 (mp3), MPEG 2, and MPEG 2.5 audio file editor that is capable of processing both Constant Bit Rate (CBR) and Variable Bit Rate (VBR) encoded files.
Since no file decoding / encoding occurs during editing, there is no audio quality loss when editing with mpgedit.
When editing VBR files that have a XING header, mpgedit updates the output files XING header information to reflect the new file size and average bit rate.
mpgedit operates in two modes: a command line and a curses-based, full-screen interactive shell. All editing functionality is available in both modes. The chief advantage of the interactive mode comes when using the playback feature for selection and verification of edit begin and end times, which is more rapidly performed interactively.
xmpgedit provides most of the same editing capabilities as the curses-mode mpgedit, but adds a graphical representation of the volume levels of mp3 file being being edited.
Enhancements:
- This release fixes an occasional crash in xmpgedit when adjusting the volume after playback has stopped.
- mpgedit playto start time is fixed, which would play beyond the specified edit time in some circumstances.
- The command "Copy start time" has been added to the xmpgedit edit menu.
- mpgedit playback and editing with the -I option is fixed, which would frequently repeat up to a second of material at the start of the edit.

LAME is short from LAME Aint an MP3 Encoder and is a research project for learning about and improving MP3 encoding technology. LAME includes an MP3 encoding library, simple frontend application, a much-improved psycho-acoustic model (GPSYCHO), and a graphical frame analyzer (MP3x).
Please note that any commercial use (including distributing the LAME encoding engine in a free encoder) may require a patent license from Thomson Multimedia.
Main features:
- Many improvements in quality in speed over ISO reference software. See history.
- MPEG1,2 and 2.5 layer III encoding.
- CBR (constant bitrate) and two types of variable bitrate, VBR and ABR.
- Encoding engine can be compiled as a shared library (Linux/UNIX), DLL or ACM codec (Windows)
- free format encoding and decoding
- GPSYCHO: a GPLd psycho acoustic and noise shaping model.
- Powerfull and easy to use presets
- Quality is comparable to FhG encoding engines and substantially better than most other encoders.
- Fast! Encodes faster than real time on a PII 266 at highest quality mode.
- MP3x: a GTK/X-Window MP3 frame analyzer for both .mp3 and unencoded audio files.
Software which uses "LAME":
- andromeda (PHP and ASP) Dynamically presents collections of mp3s as streaming web sites.
- rip (Perl) Script for ripping and encoding.
- avifile AVI/DIVX encoder and decoder for Linux.
- Grip (Linux) gtk-based cd-player, ripper and encoder. Supports cddb, cdparanoia and LAME.
- jbm2 (Linux) A KDE jukebox style application for public places (bars, pubs,...)
- Krabber (Linux) A KDE ripper & encoder, can use LAME.
- Mp3Maker (Linux) A WindowMaker enhanced front end to cdda2wav/cdparanoia and lame/bladeenc.
- dekagenc (Linux) Bourne shell script for ripping, encoding and CDDB naming.
- ripperX (Linux) GTK frontend for rippers and several encoders featuring CDDB support.
- T.E.A.R. (Linux) frontend to LAME, cdparanoia and CDDB.
- Xmcd. (Linux) CD Player with CDDB and ripping to MP3 and OGG.
- xtunes (Linux) GTK frontend for LAME, MAD, cdparanoia, cdrecord and more.
- DropMP3 (Mac) includes LAME binaries.
- CDex (Windows) Ripper & encoder, includes LAME binaries (the Blade compatible dll)
- Lamedrop (Windows) OggDrop style frontend.
- LAMEX (Windows) An activex control for LAME, and a GUI. Source code only, includes LAME.
- m3w (Windows) A live mp3 streamer for the WWW. Works with LAME, icecast, soundcard input
- out_lame (Windows) Winamp output plug-in. Create MP3 files directly from Winamp!
- RazorLame (Windows) The RazorBlade front end now supports LAME.
- winLAME (Windows) The only *nice* windows UI for LAME, according to the author :-)
- DarkIce Live streamer for IceCast.
- LiveIce Real time streaming of mp3s. Works with IceCast
- MuSE A mixing, encoding and streaming engine.
- Flash Forth a Flash-like development library
Enhancements:
- This version is identical to 3.97b3, which was promoted to release.

tooLAME is an optimized Mpeg Audio 1/2 Layer 2 encoder.

It is based heavily on:

- the ISO dist10 code
- improvement to algorithms as part of the LAME project

Installation:

1. edit Makefile
at least change the architecture type (ARCH) to suit your machine.
2. make

Usage:

./toolame [options] < input > < output >

Input File
tooLAME parses AIFF and WAV files for file info
raw PCM is assumed if no header is found
for stdin use a -

Output File
file is automatically renamed from *.* to *.mp2
for stdout use a -

Input Options
-s [int]
if inputting raw PCM sound, you must specify the sample rate
default sample rate is 44.1khz.

-a
downmix from stereo to mono
if the incoming file is stereo, combine the audio into
a single channel

-x
force byte-swapping of the input. (current endian detection is dodgy,
so if toolame produces only noise, use -x )

-g
swap the LR channels of a stereo file

Output Options
-m [char]
the encoding mode (default j)
s stereo
d dual channel
j joint stereo
m mono

-p [int]
which psy model to use (default 1)
Different models for the psychoacoustics
Models: -1 to 4

-b [int]
the total bitrate
For 48/44.1/32kHz default = 192
For 24/22.05/16kHz default = 96

-v [int]
Switch on VBR mode.
The higher the number the better the quality.
Useful range -10 to 10.
See README.VBR for details.

Operation
-f
fast mode turns off calculation of the psychoacoustic model.
Instead a set of default values are assumed

-q [int]
quick mode calculates the psy model every num frames.

Misc
-d emp
de-emphasis (default n)
-c
mark as copyright
-o
mark as original
-e
add error protection
-r
force padding bits off
-D
add DAB extensions
-t [int]
talkativity setting. 0 = no message. 3 = too much information

The FAAC project includes the AAC encoder FAAC and decoder FAAD2.
FAAC supports several MPEG-4 object types (LC, LTP, HE AAC, Main, PS) and file formats (raw AAC, MP4, ADTS AAC), multichannel and gapless en/decoding as well as MP4 metadata tags.
The codecs are compatible with standard-compliant audio applications using one or more of these profiles.
General FAAC compiling instructions:
1. Make sure you have autoconf, automake and libtool installed. For MP4 support, you must have libmp4v2 (included in faad2) installed.
2. cd to FAAC source dir
3. Run:
./bootstrap
./configure
make
make install
Usage:
faac [options]
Options:
-a X Set average bitrate to approximately X kbps per channel (i.e. using -a 64 averages at 128 kbps/stereo).
-c < bandwidth > Set the bandwidth in Hz (default value depends on sample rate)
-q < quality > Set quantizer quality (default: 100, averages at approx. 128 kbps VBR for a normal stereo input file at 16 bit and 44.1 kHz sample rate).
--tns Enable TNS coding.
--notns Disable TNS coding.
-n Disable mid/side coding.
-m X AAC MPEG version, X can be 2 or 4 (default: MPEG-2, so for the sake of interoperability with non-standard compliant players like QuickTime 6 you should set it to "4").
-o X AAC object type, X can be LC, MAIN or LTP (default: LC, for the same reason as with the MPEG version dont use Main or LTP).
-r RAW AAC output file (i.e. without ADTS headers).
-P Raw PCM input mode.
-R Raw PCM input sample rate in Hz (default: 44100 Hz).
-B Raw PCM input bit depth (default: 16 bits, also possible 8 bits).
-C Raw PCM input channels (default: 2).
- < stdin > If you simply use a hyphen/minus sign instead of an input file name, FAAC can encode directly from stdin, thus enabling piping within other applications like foobar2000 or mp4live.
Note: VBR output bitrate depends on -q AND -c, so you should only vary the default setting -q 100 -c 16000 if you know what youre doing and/or want to experiment with other cutoff frequencies at a given quality setting.
The ABR setting with -a is an approximate average bitrate that does not use a bit reservoir, i.e -a 64 and -q 100 at 44.1 kHz will result in exactly the same output file.
The following list should give some orientation for useful -q and -c settings, based on FAAC v1.17. The resulting VBR bitrates are referring to an average sounding stereo file with 16bit, 44.1 kHz, i.e. ct_reference.wav in this case. Multiplexing these AAC files to MP4 with e.g. mp4creator will result in a ~3 kbps lower bitrate because of the stripped ADTS headers:
-q 130 -c 22000 -m 4 (~218 kbps)
-q 120 -c 20000 -m 4 (~194 kbps)
-q 110 -c 18000 -m 4 (~158 kbps)
-q 100 -c 16000 -m 4 (~129 kbps)
-q 90 -c 14000 -m 4 (~103 kbps)
-q 80 -c 12000 -m 4 (~79 kbps)
-q 70 -c 10000 -m 4 (~62 kbps)
The added -m 4 switch does not change the bitrate or sound of course, but is recommended for most AAC/MP4 players that use an updated FAAD2-based plugin from this year (Winamp 2.x, foobar2000 etc.) or cant decode MPEG-2 AAC LC files like QuickTime 6. Philips Expanium users should not use this switch, because their CD portable does not know MPEG-4 AAC files.
Enhancements:
- Small bug fixes since last version

Aften project is a simple, open-source, A/52 (AC-3) audio encoder.
Main features:
- Implemented my own wav reader
- Converted the fixed-point algorithms to floating-point
- Rearranged the methods and structures
- Added stereo rematrixing (mid/side)
- Added short block MDCT and block switching
- Added VBR encoding mode
- Added variable bandwidth
- Added more complete WAV format support
- Added support for using the alternate bit stream syntax
- Created separate library and frontend
- Added input filters
Enhancements:
- Bit allocation speedups, a compile-time choice of using floats or doubles internally, an internal restructuring of MDCT functions, and bugfixes. quality=0 is now a valid setting.

Audio::TagLib::MPEG::XingHeader is an implementation of the Xing VBR headers.

SYNOPSIS

use Audio::TagLib::MPEG::XingHeader;

my $i = Audio::TagLib::MPEG::XingHeader->new($data);
print $i->isValid() ? "valid" : "invalid", "n";

This is a minimalistic implementation of the Xing VBR headers. Xing headers are often added to VBR (variable bit rate) MP3 streams to make it easy to compute the length and quality of a VBR stream. Our implementation is only concerned with the total size of the stream (so that we can calculate the total playing time and the average bitrate). It uses http://home.pcisys.net/~melanson/codecs/mp3extensions.txt and the XMMS sources as references.

new(ByteVector $data)

Parses a Xing header based on $data. The data must be at least 16 bytes long (anything longer than this is discarded).

DESTROY()

Destroy this XingHeader instance

BOOL isValid()

Returns true if the data was parsed properly and if there is a vaild Xing header present.

UV totalFrames()

Returns the total number of frames.

UV totalSize()

Returns the total size of stream in bytes.

IV xingHeaderOffset(PV $version, PV $channelMode) [static]

Returns the offset for the start of this Xing header, given the version and channels of the frame

see Audio::TagLib::MPEG::Header