Free lame mp3 encoder 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.

crip is a terminal-based ripper/encoder/tagger tool for creating Ogg Vorbis/FLAC/MP3 files under UNIX/Linux. It is well-suited for anyone (especially the perfectionist) who seeks to make a lot of files from CDs and have them all properly labeled and professional-quality with a minimum of hassle and yet still have flexibility and full control over everything. Current versions of crip only support Ogg Vorbis and FLAC. If you want to encode MP3 files you should use crip-1.0.
I am constantly refining the process of creating perfect music files to be as automated as possible while still leaving the user with control over as much as possible. To see for yourself how painless it is to make professional-grade music files on your UNIX/Linux machine, go through the crip tutorial.
This script is special because it is the only one that I know that is capable of doing group vorbisgain/replaygain and/or normalization (adjust the volume to be as loud as possible without clipping/distortion) and group labelling/tagging, which makes it easy to allow a group of tracks to be treated as one piece. It can also trim off the silence at the beginning and end of these tracks/groups.
First the script fetches the CDDB info off the internet. Then it prompts you for the grouping of the tracks. This is important because it will treat each group of tracks as one piece, label and vorbisgain/replaygain and/or normalize them (using the volume gain/peak of that group). Normalization is now obsolete with the creation of vorbisgain (replaygain) utilities, so I have that turned off by default and itll run vorbisgain instead.
You can, of course, have each track be a group by itself (such as what youd want to do with most pop CDs). But since Ive also ripped a lot of Classical music I found it necessary to group tracks differently fairly often.
Then it will prompt you for the Artist and Album info (which is already defaulted to what is pulled from CDDB). Afterwards it will prompt you for a filename for each track you selected. Again this field is defaulted to what it suspects that youd want. For most pop CDs all I have to do is hit enter here because the filename is almost always exactly what Id want.
It will then prompt you that its ready to rip. From here everything is automated, so hit return and it usually finishes in about an hour. The script calls cdparanoia to rip the tracks, and then oggenc/flac/lame to encode them. It also labels the files with info appropriately, including the CDDB CD DiscID number (so you will always have a CD reference hex-number inside your OGG/FLAC/MP3 file).
Ive looked into other scripts out there that do something similar to this script, but decided to write my own since I couldnt find one that groups tracks and trims silence. I used this script to rip and encode my collection of over 200 Bach CDs, as well as a bunch of other Classical and non-Classical CDs very easily.
Ive provided links below that include some of the prerequisites that you may need.
Main features:
- Track grouping for automated tagging and appropriate normalization/vorbisgain on multi-track pieces.
- CDDB fetching to populate default information.
- CDDB submit to update the CDDB database with your more accurate info.
- Automates as much of the tagging as possible for fully-labeled professional quality music files.
- Automatically trims digital silence at the beginning and end of a track/group (if desired).
- European character support. Also European character-mapping support.
- Flexibility and full user control in tagging and file naming.
Enhancements:
- Bugfix: -m flag on the command line was not being processed

SimpleCDR-X was born in mid-June of 2001. It was clear to me that SimpleCDRs interface had limitations that could only be overcome by going to a GUI interface. I then proceeded to look at the various toolkits and then I discovered Glade.
Glade is perhaps one of the best programming utilities that I have found for Linux to date. It the development of a 2200 line interface much easier than it would have been otherwise. If Glade wasnt around I still might just be playing with the various toolkits instead of having a finished product. Glade allowed me to focus on functionality rather than trying to get the interface to look right with straight C code.
SimpleCDR-X like its brother SimpleCDR is a hybrid of C and C++. Most of the external utilities are managed by C++ classes called from the hybrid callbacks.c. The reason that I didnt opt to use GTK-- instead of the hybird was because most everyone already has GTK+, however, not everyone has GTK-- and some dont want to download a 1.5 MB file to compile or dig up the installation CDs.
Main features:
- Disc-At-Once CD copying
- Audio CD copying via cdrecord and CDparanoia or cdda2wav
- Audio CD Mastering
- MP3/OGG import via MADplay, LAME, or OGG123
- Import from CD via CDparanoia or cdda2wav
- Data CD Mastering
- Multi-session CD writing
- Rip tracks to wav
- Rip tracks to MP3/OGG on the fly via Blade Encode, LAME, or oggenc
- GTK+ Interface

Fenice is a multimedia streaming server compliant with the IETFs standards for real-time streaming of multimedia contents over Internet. Fenice implements RTSP - Real-Time Streaming Protocol (RFC2326) and RTP/RTCP - Real-Time Transport Protocol/RTP Control Protocol (RFC3550) supporting the RTP Profile for Audio and Video Conferences with Minimal Control (RFC3551).
Fenice supports the following encoding standards:
Audio
- MP3 (MPEG-1 Layer III) (RFC3119)
- OGG/Vorbis (work in progress)
Video
- MPEG-1/2 (RFC2250)
- Preliminary support for MPEG-4 (RFC3016, RFC3640)
- OGG/Theora (work in progress)
The main characteristic of Fenice is that it is adaptable to the state of the network gotten through the technique of the dynamic coding change.
Fenice is also able to manage live streaming sessions using external real-time audio/video encoders such as lame, ffmpeg or mjpeg-tools, even capturing audio and video streams from live-recording remote hosts (with Felice - Fenice Live CEaseless).
Fenice is the worlds first streaming server supporting Creative Commons licensing meta-data for audio/video streaming.
Enhancements:
- Log support via file or sysconfig was added.
- MPEG-4 support was finished, and works best for stored contents.
- The annoying 100% CPU bug was fixed.
- Fenice should now run smoothly and reliably in almost every condition.
- Various bugs were fixed.
- Fenice is GCC4 compatible.

CD-Rchive2 is a complete revamp of the popular but now aged CD-Rchive program. It is intended to be used with cdrecord-ProDVD, cdda2wav, and mkisofs for the production of data and music CDs and data DVDs.
Built in tools allow cloning of a complete CD or DVD and writing copies. A Boot Wizard will guide you through steps required to make a bootable CD or DVD, write the code, and compile a binary menu program which runs with isolinux.
Enhancements:
- Functionality was added when ripping tracks from CD to automatically convert to .mp3 instead of .wav via a checkbox on the .wav tab page.
- This requires lame to be installed.
- A problem of occassional incorrect audio device specification on single device machines such as laptops was fixed.

iPod Video Encoder is a command line tool for convenient encoding of video files for use on iPod video. iPod Video Encoder project uses the ffmpeg tool for the actual encoding.
It features recursive encoding of all files in a directory, and generating a podcast.xml file for convenient use in iTunes.
It can be used in a cron job that periodically checks directories for new files and encodes them without user intervention.
Main features:
- Encoding of single files
- Optional specification of a/v bitrates
- Recursively encode all files in a directory
- Generating a podcast.xml for convenient use in iTunes
Options:
--version show programs version number and exit
-h, --help show this help message and exit
-R, --recursive Process directories recusively
-f, --force Force re-encode existing iPod videos
-t, --test Only encode first 10 seconds, to produce test videos
-e EXTENSIONS, --extensions=EXTENSIONS
Comma separated list of input extensions. Default: avi
-i IPODEXT, --ipodext=IPODEXT
iPod extension. Default: .ipod.mp4
-b VIDEO_RATE, --video-rate=VIDEO_RATE
Video bit rate, in kbps (default: 1024)
-a AUDIO_RATE, --audio-rate=AUDIO_RATE
Audio bit rate, in kbps (128)
-W WIDTH, --width=WIDTH
Video width, in pixels (default: 320)
-H HEIGHT, --height=HEIGHT
Video height, in pixels (default: 240)
-P, --pretend Do not really encode, just print out the ffmpeg
commands which would be executed
-v, --verbose Enable verbose output
Podcast options:
-p, --podcast Generate a podcast.xml file for each directory.
--podcast-documentroot=PODCAST_DOCUMENTROOT
Podcast document root; eg. /var/data/Movies/
--podcast-baseurl=PODCAST_BASEURL
Podcast base URL; eg. http://localhost/
Enhancements:
- Only uses the -e option for processing directories.
- The default ipod extension is now _ipod.mp4 for Windows compatibility.
- The README file has been extended with information on the .ipod-encoder settings file.
- The program no longer crashes when HOME is not set on Windows.
- --width and --height parameters have been added.

FAME project is a set of tools for encoding multimedia content.

It currently consists in the following tools:

libfame

libfame is a video encoding library.

It can currently encode MPEG-1 and MPEG-4 rectangular video, as well as MPEG-4 video with arbitrary shape.

Compliance

Provide bitstreams compliant to the MPEG-1, MPEG-2 and MPEG-4 video standards.

Speed

Provide a fast implementation of the techniques used in MPEG standards.

Flexibility

Allow the user to choose between different options for speed, compression ratio and quality.

Portability Support many different platforms and architectures.

fame

fame is a multimedia encoder, which captures video from a video4linux device, and optionnaly sound, for MPEG encoding.

It is based on libfame and lame and thus supports the same output formats as these two libraries.

MO3 is a file format, encapsulating the features of several "MOD" formats (IT, XM, S3M, MTM, MOD), but with one big difference - MP3 and OGG compressed samples. A lossless codec is also incorporated for any samples that dont like lossy encoding, and the whole file structure is compressed too.
All this makes it possible to have great reductions in file sizes with practically identical sound quality. Actually MO3 allows greater quality compositions because MP3/OGG encoding works best with 16-bit samples, so theres no reason to use low quality 8-bit samples anymore.
Main features:
MP3/OGG compression
- any command-line MP3 or OGG encoder can be used with the MO3 encoder
Lossless compression
- samples that dont compress well with MP3/OGG can still be reduced in size
Structure compression
- all the non-sample data is also compressed
Sample specific config
- the encoding sample/bit rates are individually adjustable for each sample
Text removal
- message/instrument/sample texts can optionally be removed
Decompression
- MO3 files can be decoded back to their original formats using UNMO3
Playable
- MO3s can be played by XMPlay and software that uses the BASS library or UNMO3