Free realvideo vbr software for linux

RealPlayer plays streaming audio and video over the Internet in real-time. RealPlayer plays RealAudio, RealVideo, MP3, 3GPP Video, Flash, SMIL 2.0, JPEG, GIF, PNG, RealText, Ogg Vorbis, RealPix and Ogg Theora.
It is available for Windows, Macintosh, Pocket PC, Nokia 9200 Series, Nokia 7650, Palm OS 5-based handhelds, Linux, Solaris, and many Unix variants.
Main features:
Play popular datatypes
- RealPlayer@ 10 supports RealAudio, RealVideo 10, MP3, Ogg Vorbis and Theora, H263, AAC and more. Get ready for accelerated video, full screen playback, and a lot more to play.
Mozilla-compatible plug-in
- You can now watch and listen to embedded video right in your Web browser without opening RealPlayer. Enjoy media from your favorite music and news sites with just one click.
New UI adopts your theme
- The elegant UI design is based on GTK technology. That means the player adopts the theme youve chosen, and blends itself accordingly into your desktop environment.
Enhancements:
- fixes

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.

1. Ultra-low latency (typically from 3 to 9 ms)
2. Full audio bandwidth (44.1 kHz and 48 kHz)
3. Stereo support
4. Packet loss concealment
5. Constant bit-rates from 32 kbps to 128 kbps and above
6. A fixed-point version of the encoder and decoder
7. The CELT codec is meant to close the gap between Vorbis and Speex for applications where both high quality audio and low delay are desired.

Enhancements:

• Has just been released, with many quality improvements, including better stereo coupling, better handling of transients, and better handling of highly tonal signals.
• Packet loss robustness has been improved through the optional use of independent (intra) frames.
• Supports a larger dynamic range, suitable for encoding 24-bit audio (float version only).
• There is also a very early VBR implementation.
  

xmpgedit project is an MP3 editor for VBR and CBR encoded files.
xmpgedit 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.
xmpgedit can cut an input MPEG file into one or more output files, as well as join one or more input MPEG files into a single output file. Since no decoding or encoding occurs during editing, there is no audio quality loss when editing with xmpgedit.
Enhancements:
- This release fixes an occasional crash in xmpgedit when adjusting the volume after playback has stopped.
- xmpgedit playto start time is fixed, which would play beyond the specified edit time in some circumstances.
- Added the command "Copy start time" to the xmpgedit edit menu.

MP3::Info is a Perl module that can manipulate / fetch info from MP3 audio files.

SYNOPSIS

#!perl -w
use MP3::Info;
my $file = Pearls_Before_Swine.mp3;
set_mp3tag($file, Pearls Before Swine, q"77s",
Sticks and Stones, 1990,
q"(c) 1990 77s LTD.", rock & roll);

my $tag = get_mp3tag($file) or die "No TAG info";
$tag->{GENRE} = rock;
set_mp3tag($file, $tag);

my $info = get_mp3info($file);
printf "$file length is %d:%dn", $info->{MM}, $info->{SS};
my $mp3 = new MP3::Info $file;
$mp3->title(Perls Before Swine);
printf "$file length is %s, title is %sn",
$mp3->time, $mp3->title;

$mp3 = MP3::Info->new(FILE)

OOP interface to the rest of the module. The same keys available via get_mp3info and get_mp3tag are available via the returned object (using upper case or lower case; but note that all-caps "VERSION" will return the module version, not the MP3 version).

Passing a value to one of the methods will set the value for that tag in the MP3 file, if applicable.

use_mp3_utf8([STATUS])

Tells MP3::Info to (or not) return TAG info in UTF-8. TRUE is 1, FALSE is 0. Default is TRUE, if available.

Will only be able to turn it on if Encode is available. ID3v2 tags will be converted to UTF-8 according to the encoding specified in each tag; ID3v1 tags will be assumed Latin-1 and converted to UTF-8.

Function returns status (TRUE/FALSE). If no argument is supplied, or an unaccepted argument is supplied, function merely returns status.

This function is not exported by default, but may be exported with the :utf8 or :all export tag.

use_winamp_genres()

Puts WinAmp genres into @mp3_genres and %mp3_genres (adds 68 additional genres to the default list of 80). This is a separate function because these are non-standard genres, but they are included because they are widely used.

You can import the data structures with one of:
use MP3::Info qw(:genres);
use MP3::Info qw(:DEFAULT :genres);
use MP3::Info qw(:all);

remove_mp3tag (FILE [, VERSION, BUFFER])

Can remove ID3v1 or ID3v2 tags. VERSION should be 1 for ID3v1 (the default), 2 for ID3v2, and ALL for both.

For ID3v1, removes last 128 bytes from file if those last 128 bytes begin with the text TAG. File will be 128 bytes shorter.

For ID3v2, removes ID3v2 tag. Because an ID3v2 tag is at the beginning of the file, we rewrite the file after removing the tag data. The buffer for rewriting the file is 4MB. BUFFER (in bytes) ca change the buffer size.

Returns the number of bytes removed, or -1 if no tag removed, or undef if there is an error.

set_mp3tag (FILE, TITLE, ARTIST, ALBUM, YEAR, COMMENT, GENRE [, TRACKNUM])

set_mp3tag (FILE, $HASHREF)

Adds/changes tag information in an MP3 audio file. Will clobber any existing information in file.

Fields are TITLE, ARTIST, ALBUM, YEAR, COMMENT, GENRE. All fields have a 30-byte limit, except for YEAR, which has a four-byte limit, and GENRE, which is one byte in the file. The GENRE passed in the function is a case-insensitive text string representing a genre found in @mp3_genres.

Will accept either a list of values, or a hashref of the type returned by get_mp3tag.
If TRACKNUM is present (for ID3v1.1), then the COMMENT field can only be 28 bytes.

ID3v2 support may come eventually. Note that if you set a tag on a file with ID3v2, the set tag will be for ID3v1[.1] only, and if you call get_mp3tag on the file, it will show you the (unchanged) ID3v2 tags, unless you specify ID3v1.

get_mp3tag (FILE [, VERSION, RAW_V2])

Returns hash reference containing tag information in MP3 file. The keys returned are the same as those supplied for set_mp3tag, except in the case of RAW_V2 being set.
If VERSION is 1, the information is taken from the ID3v1 tag (if present). If VERSION is 2, the information is taken from the ID3v2 tag (if present). If VERSION is not supplied, or is false, the ID3v1 tag is read if present, and then, if present, the ID3v2 tag information will override any existing ID3v1 tag info.

If RAW_V2 is 1, the raw ID3v2 tag data is returned, without any manipulation of text encoding. The key name is the same as the frame ID (ID to name mappings are in the global %v2_tag_names).

If RAW_V2 is 2, the ID3v2 tag data is returned, manipulating for Unicode if necessary, etc. It also takes multiple values for a given key (such as comments) and puts them in an arrayref.

If the ID3v2 version is older than ID3v2.2.0 or newer than ID3v2.4.0, it will not be read.

Strings returned will be in Latin-1, unless UTF-8 is specified (use_mp3_utf8), (unless RAW_V2 is 1).

Also returns a TAGVERSION key, containing the ID3 version used for the returned data (if TAGVERSION argument is 0, may contain two versions).
get_mp3info (FILE)

Returns hash reference containing file information for MP3 file. This data cannot be changed. Returned data:

VERSION MPEG audio version (1, 2, 2.5)
LAYER MPEG layer description (1, 2, 3)
STEREO boolean for audio is in stereo

VBR boolean for variable bitrate
BITRATE bitrate in kbps (average for VBR files)
FREQUENCY frequency in kHz
SIZE bytes in audio stream
OFFSET bytes offset that stream begins

SECS total seconds
MM minutes
SS leftover seconds
MS leftover milliseconds
TIME time in MM:SS

COPYRIGHT boolean for audio is copyrighted
PADDING boolean for MP3 frames are padded
MODE channel mode (0 = stereo, 1 = joint stereo,
2 = dual channel, 3 = single channel)
FRAMES approximate number of frames
FRAME_LENGTH approximate length of a frame
VBR_SCALE VBR scale from VBR header
On error, returns nothing and sets $@.

mkvtoolnix is a set of tools that allow users to display information about, extract streams from, merge several streams into, and split Matroska media files. Supported video stream types include streams from AVIs, Ogg/OGM, and RealVideo files.
Supported audio formats include Vorbis, RealAudio, MP3, MP2, FLAC, AC3, and AAC. Several subtitle formats are supported, including SRT, SSA, ASS, VobSub, and many others.
The resulting files from mkvtoolnix can be played back with various media players on Linux/Unix or the Matroska Direct Show filters under Windows.
Enhancements:
- This release fixes a couple of issues with the AVC/h.264 support, changes the defaults so that the annoying "NALU size length" error message does not appear anymore, and features a few usability enhancements.
- It is now possible to create files in which no track has its "default track" flag set.
- Support for 14-bit coded DTS files was added.

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