Free vbr. software for linux

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.
  

Punit is an attempt to create a Python CD ripper using cdparanoia and lame for Linux.

I couldnt get Sound Juicer to rip mp3, GooBox encodes mp3 using VBR which screws up the mp3 times in Foobar2000. I wanted a simple mp3 ripper and i wanted to learn Python so I made my own.

Right now, its a straight forward ripper to mp3 with the format 128 kbps, 44100Hz, stereo. Comments, suggestions, improvements are all welcomed.

Thanks to CDDB.py for the CDDB module.

dvdspanky is a CLI tool to convert video files into DVD compatible MPEG streams. It is designed to be easy to use no matter the input source, to automate common transcoding tasks and provide powerful features. It is written in C and provides a front-end to transcode, MJPEG tools, mplayer and feh. The output can be used in programs like dvdauthor.
Main features:
- Consistent options no matter the input source
- Clean and clear output
- VBR and CBR encoding
- Output preview
- Destination file size specification
- PAL and NTSC export
- Post-processing
- Automatic volume adjustment
- Automatic cropping and letter-boxing
- Automatic mplayer fallback decoding
- Automatic aspect calculation
- Automatic DVD compatibility adjustments

cutmp3 is a small and fast command line MP3 editor. cutmp3 lets you select sections of an MP3 interactively or via a timetable and save them to separate files without quality loss.
It uses mpg123 for playback and works with VBR files and even with files bigger than 2GB.
Other features are configurable silence seeking and ID3 tag seeking, which are useful for concatenated mp3s.
Playback is realized via mpg123, so be sure to have it installed or at least have a symlink to your favorite mp3 playing program.
I recommend a symlink to mpg321 which works better!
You can mark beginning and end of a segment with a and b and save the segment with s.
Using a timetable or direct times with VBR files delivers exact(!) results at the cost of slightly lower speed. cutmp3 even works with files bigger than 2 GB!
If you want a working graphical software you can try mp3directcut,
which runs fairly well in WINE after I asked the author about a WINEd version.
Installation:
make (you will need readline-devel or similar!)
install it to /usr/bin with
make install
Enhancements:
- A small fix for timetable cutting with negative values has been made.
- Another option for silence seeking has been added.
- The documentation has been updated.

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

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.

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