Free vbr. software for linux

mpeglib is a mp3 and mpeg I video/audio library for linux.
The library includeds three command line players, for mp3,wav,mpeg video.
Main features:
MPEG I library.
This library contains:
- mpeg I audio player (layer I,II,III (mp3))
- mpeg I video player
- mpeg I system layer player
- wav player
Supported features:
- direct/fast seek in all players
- length detection
- video synchronisation, based on timestamps
- mmx Support where necessary
- VBR/ID3 support for mp3
- shoutcast/icecast support
- CDI/Video CD support
- plugin architecture for input,decoder,output
Supported Outputs:
Audio
- Support for OSS/Linux, Sun
Video
- X11 standard calls (fallback)
- X11 Shared mem
- X11 XFree86 4.0 DGA 2.0 (needs root)
- X11 XFree86 4.0 XVideo Extension (hardware yuv->rgb renderer)
Supported Inputs
- file,http.
- Supports on Linux Video CDs (vcd,cdi)

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.

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 $@.

mp3asm is an mpeg 1/2/2.5 audio layer 1/2/3 (cbr/vbr) frame level editor. The current textmode version allows cutting, copying, and pasting of frames while never violating the mpeg audio standard.
Well, most ppl probably prefer decoding, wav editing, reencoding, but that has consequences. Encoders use advanced filtering to create audio data thats easy to compress so that ppl can hardly tell the difference (depending on BR offcourse) and then they compress that filtered audiodata. Now filtering once from original full range unfiltered audio will provide the nicest result.
And offcourse, u dont have to waste precious cycles on the whole lengthy process.
Yes indeed, with wavediting programs u can edit right up to the millisecond, or even better. But to be honest, an 128kbps mpeg 1 layer 3file has 25 frames per second, which means u can cut right down to 40ms. If u dont mind the quality of a 128kbps file, how can u mind a few millisecs?Every subsequent filtering will result in more loss of data, because the only real difference between encoders is the filterbanks.
So when this app is up and running theres no more reason to waste a perfectly good rip cos of a few bad starting frames or a long lead out. At this point i even believe it should be possible to provide for fading in/out, i aint certain yet actually. I have no clue whatsoever on why Olli fromme halted development back in 97.Maybe cos mp3s were a rarity back then? Maybe some personal reason. I will prolly never know. What I do know is that he had an unbelievable idea and he wrote a program that did the job very well.
Enhancements:
- fixed parsing sideinfo (off by 1 bit) for mpeg 1 layer 3 mono (found by Alain Daurat (alain.daurat@libertysurf.fr))

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

PyMP3Cut is a Python command line tool designed to cut huge (> 100MB) MP3 files at high speed without requiring the extra disk space and processing time usually needed by visual audio editing tools, which convert the MP3 format to more easily manageable formats like WAV before doing anything.
It cuts and reads simultaneously according to the autodetected MP3 frame rate and a timeline passed as a command line argument. It doesnt currently deal with Variable Bit Rate (VBR) MP3 files, though.
Main features:
- PyMP3Cut is a Python command line tool designed to cut very huge MP3 files at a blazzingly fast rate without requiring the extra disk space and processing time usually needed by Audacity or other similar visual audio editing tools, which convert the MP3 format to more easily manageable formats like WAV before doing anything. The WAV conversion usually requires 10 times more disk space !
- PyMP3Cut doesnt convert anything : it reads and cuts simultaneously, according to the autodetected audio frame rate and a timeline passed as a command line argument. You can think of PyMP3Cut as being some sort of very careful chainsaw ;-) Since theres no back and forth MP3 conversion, theres no quality loss either !
- PyMP3Cut was designed to slice high quality MP3 recordings of day-long congresses into smaller per-speaker MP3 files. It only needs the exact same amount of disk space as the original file to slice, even less if you plan to skip some parts, which PyMP3Cut can do automatically if you use a specially formatted *SKIP* entry in your timeline. It was successfully used many times against several hundredths megabytes MP3 files.
Enhancements:
- Bill Eldridge added the --segment command line option.

getID3() is a PHP4 script that extracts useful information from MP3s & other multimedia file formats:
Tag formats:
- ID3v1 (v1.0 & v1.1)
- ID3v2 (v2.2, v2.3 & v2.4)
- APE tags (v1 & v2)
- (Ogg) VorbisComment
- Lyrics3 (v1 & v2)
Lossy Audio-only formats:
- MP3, MP2, MP1 (MPEG-1, layer III/II/I audio, including Fraunhofer, Xing and LAME VBR/CBR headers)
- Ogg Vorbis
- Musepack / MPEGplus
- AAC & MP4
- AC-3
- RealAudio
- VQF
- Speex
Lossless Audio-only formats:
- WAV (including extended chunks such as BWF and CART)
- AIFF (Audio Interchange File Format)
- Monkeys Audio
- FLAC & OggFLAC
- LA (Lossless Audio)
- OptimFROG
- WavPack
- TTA
- LPAC (Lossless Predictive Audio Compressor)
- Bonk
- LiteWave
- Shorten
- RKAU
- Apple Lossless Audio Codec
- RealAudio Lossless
- CD-audio (*.cda)
- NeXT/Sun .au
- Creative .voc
- AVR (Audio Visual Research)
- MIDI
Audio-Video formats:
- AVI
- ASF (ASF, Windows Media Audio, Windows Media Video)
- MPEG-1 & MPEG-2
- Quicktime
- RealVideo
- NSV (Nullsoft Streaming Video)
Graphic formats:
- JPG
- PNG
- GIF
- BMP (Windows & OS/2)
- TIFF
- SWF (Flash)
- PhotoCD
Data formats:
- ZIP
- TAR
- GZIP
- ISO 9660 (CD-ROM image)
- SZIP
getID3() can write:
- ID3v1 (v1 & v1.1)
- ID3v2 (v2.3, v2.4)
- APE (v2)
- Ogg Vorbis comments
- FLAC comments
Whats New in 1.7.7 Stable Release:
- All 1.x bugfixes have been ported from getID3() 1.7.2 to 1.7.7

mp3plot project prints out a plot of the bitrate distribution of a VBR MP3 file (it will also do it for CBR files although it isnt very meaningful).

It should be architecture independent but I havent tested beyond PCs.
Theres a much more mature tool that does the same and more: mp3stat at signal-lost.homeip.net/projects. mp3stat refuses to work on my system(s) and having an interest in mp3s internal structure I gave a shot at it with mp3plot.