Free lame mp3 encoder audacity software for linux

BladeEnc is a freeware MP3 encoder. It is based on the same ISO compression routines as mpegEnc, so you can expect roughly the same, or better, quality . The main difference is the appearance and speed.
BladeEnc doesnt have a nice, user-friendly interface like mpegEnc, but it is more than three times faster, and it works with several popular front-end graphical user interfaces.
BladeEncs output quality is one of those rare subjects that completely divides the world in two parts. Either you love it or you hate it, there never seems to be an opinion inbetween. Different audiophiles and mp3 experts tends to come to completely different conclusions depending on their methods and testsamples.
The reason for this is of course that BladeEnc is a very different mp3 encoder (compared to Fraunhofer, LAME etc) with a very unique approach to mp3 encoding.
In order to compress sound to an mp3 file, you need to make certain sacrifices in quality. Taking into account how we percieve sound, the mp3 encoder tries to remove the details that it believes us to be least likely to notice. How much that needs to be removed depends on the bitrate and the encoder often has the choice of doing different kinds of sacrifices.
It can remove low volume tones that are "shadowed" by high volume tones of similar frequencies, remove the high frequency part of the sound spectrum, cut down the stereo effect (so called joint stereo) and simply decrease the samplerate. What approach is the best depends on a lot of things, like the style of music and the selected bitrate.
Main features:
- Sourcecode available under the LGPL-license!
- Stereo or Mono output. Can downmix to Mono on the fly.
- Supports the following bitrates: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kBit/s. However, for bitrates lower than 128 kBit we seriously recommend you to use another encoder.
- Flags like Private, Original and Copyright can be set.
- Input samples can be in either 32, 44.1 or 48 kHz.
- Both 8 and 16-bit samples are supported.
- Working CRC checksum generation (since 0.80). The ISO reference code had broken CRC calculations, which has been inherited into every ISO based encoder that havent added a fix for it.
- Can be plugged directly into many popular 3rd party products, giving them integrated mp3 encoding abilities.
- Encodes chunks of data from memory to memory, no need to use files or pipes.
- Can be compiled for nearly any operating system still in use.
- Commandline based, makes it easy to include BladeEnc into BAT files and shell scripts.
- Only mp3 encoder that supports gapless encoding.
- Reads standard uncompressed WAV- and AIFF-files as well as well as RAW PCM-data.
- Batch encoding. Can encode any number of samples in a row.
- Wildcards supported. You can for example encode all WAV-files in a directory by typing *.WAV".
- Input samples can be automatically deleted after encoding.
- Large selection of graphical frontends available.
- Task priority can be set from the commandline and is by default set to LOWEST so that you still can use your computer effectively while encoding (Windows & OS/2 only).
- Full support for pipes and redirection (stdin and stdout).
- Textbased configuration file where you can change default settings.

mpTcv is a GUI for LAME to convert MP3 to other bit rates. mpTcv lets the user choose a directory to recode to another bit rate.

This is useful for cramming more data onto USB sticks, MP3 players, etc.

POE::Component::Enc::Mp3 is a mp3 encoder wrapper.

SYNOPSIS

use POE qw(Component::Enc::Mp3);

$mp3 = POE::Component::Enc::Mp3->new($bitrate => 160);

$mp3->enc("/tmp/tst.wav");

POE::Kernel->run();

This POE component encodes raw audio files into mp3 format. It is merely a wrapper for the notlame program.

METHODS

The module provides an object oriented interface as follows:

new

Used to initialise the system and create a module instance. The following parameters are available:

alias

Indicates the name of a session to which module callbacks are posted. Default: main.

bitrate

Should be self-evident. If left unspecified, defaults to 160.

enc < file-name > [del-orig]

Encodes the given file, naming it with a .mp3 extension. An optional true value for the second parameter indicates that the original file should be deleted.

e.g. $mp3->enc("/tmp/tst.wav");

WideMP3 is a relatively simple distributed MP3 encoder. It works in a client-server fashion, with multiple servers and a client that feeds the PCM audio over a network connection to the server. The server sends back the encoded audio, which the client then writes to a file. The actual encoding is done via an external encoder, such as BladeEnc.

Installation is very simple simple.

Extract the source distribution, and compile and install as follows:

./configure
make
su
make install

Current supported platforms are:

Debian GNU/Linux 2.2 on Intel x86 and Alpha.
Debian GNU/Linux 2.1 on Intel x86
OpenBSD 2.7 on Intel x86

It should also work on FreeBSD, although this has not yet been tested. Positive and negative reports on different platforms will be appreciated, especially if accompanied by a patch to make it work.

If you want to contribute a patch, you must be willing for the patch to be released under the GNU General Public License. Patches should be in context diff format (diff -c) or unified diff format (diff -u).

Currently supported MP3 encoders:
Bladeenc 0.9x - http://bladeenc.mp3.no/
Positive and negative reports for different encoders will be appreciated, especially if accompanied by a patch to make it work.

This is a unified diff (you probably need GNU patch to apply it) to lame 3.58. Please note that this diff is now obsolete as recent beta versions of lame include a version of this code. Just type "make mp3rtp". The code was broken, but the CVS code is reported to work (and interoperate with playRTPMPEG) as of Feb 21 2000.

Please note that the output stream will only have the correct speed if the input is live recorded stream from your sound card or lame encodes exactly in real-time on your CPU!

RTP is the Realtime Transport Protocol as defined in RFC 1889. It is the transport mechanism of your choice to multicast an mp3 stream.

gnormalize is an audio converter, a front end to normalize, an encoder/decoder, a ripper, a metadata (tag) editor and a cd player. gnormalize uses gtk2-perl under GNU/Linux.
gnormalize decodes the MP3/MP4/MPC/OGG/APE/FLAC file to WAV, then normalizes the WAV to a targeted volume level and re-encodes it. Moreover, gnormalize can extract Audio CD track and output as various popular audio formats (MP3, MP4, MPC, OGG, APE, FLAC, WAV) with fast speed and high quality. gnormalize can also convert audio format between MP3, MP4, MPC, OGG, APE and FLAC with high fidelity, which meets your need to play and collect audio files. It can change the encoding and Metadata (tag) properties of final normalized files.
gnormalize can be used to adjust the volume of audio files to a standard volume level, where different recording levels on different albums can cause the volume to vary greatly from song to song.
gnormalize is an alternative for replaygain, because "there is no consistent standard by which to define the appropriate replay gain which mp3 encoders and players agree on, and no automatic way to set the volume adjustment for each track".
Main features:
- Convert from/to MP3, MP4, MPC, OGG, APE, FLAC and WAV;
- Support for LAME, FAAC, OGGENC, MPPENC, FLAC and MAC encoders;
- Support for LAME, FAAD, OGGDEC, MPPDEC, FLAC and MAC decoders;
- Rip audio CDs utilizing cdparanoia or cdda2wav;
- Play audio CDs utilizing cdcd or Audio::CD or cdplay;
- Support normalization with adjustable sensibility;
- Can normalize/convert files recursively into the directory;
- Edit and get MetaData like artist, title, album, ... ;
- Get MetaInfo like bitrate, time, frequency, ... ;
- Includes mppenc, mppdec;
- Read the CDDB entry for an audio CD with CDDB_get;
- Translation to English and Portuguese (Brasil).