Free flac encoder software for linux

DreamEncoder is a TwinVQ encoder working in XWindow environment based. It uses free encoding library taken from TwinVQ SDK by NTT. As always program is available on GNU GPL, so you get the source code. You can use it as you wish, but please note that the code is mine .
DreamEncoder has been created to fill the hole - there wasnt any batch encoder (like TwinVQ Encoder by Andre Kampert for Win32).

To install DreamEncoder do the following steps:
Untar the package by typing:
# tar -zxf dreamenc-v?.?.?.tar.gz
then go to the dreamenc directory:
# cd dreamenc
and type:
# make
In a while computer will compile the source code, when finishes type:
# make install and the DreamEncoder is installed on your system. To use it type in xterm or run anyway dreamenc. The interface is (I think) intuitive so you cant complain thats non-intuitive.

FLAC is a lossless compression method for audio files. Using it, one is able to reduce the filesize and keep the same audio quality as a regular WAV file (CD quality).

The FLAC format is open source, and can be read without any problem by such applications as Amarok or Kaffeine.

Wav and Flac conversion contains two service menus: one to convert WAV files into FLAC, and one to convert FLAC files into WAV.

You will need the FLAC application to be installed on your system to use these service menus.

FLAC is Free Lossless Audio Codec. Grossly oversimplified, FLAC is similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality.
This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, see supported devices) just like you would an MP3 file.
FLAC is freely available and supported on most operating systems, including Windows, "unix" (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.
The FLAC project consists of:
the stream format
reference encoders and decoders in library form
flac, a command-line program to encode and decode FLAC files
metaflac, a command-line metadata editor for FLAC files
input plugins for various music players
When we say that FLAC is "Free" it means more than just that it is available at no cost. It means that the specification of the format is fully open to the public to be used for any purpose (the FLAC project reserves the right to set the FLAC specification and certify compliance), and that neither the FLAC format nor any of the implemented encoding/decoding methods are covered by any known patent. It also means that all the source code is available under open-source licenses. It is the first truly open and free lossless audio format.
Main features:
- Lossless: The encoding of audio (PCM) data incurs no loss of information, and the decoded audio is bit-for-bit identical to what went into the encoder. Each frame contains a 16-bit CRC of the frame data for detecting transmission errors. The integrity of the audio data is further insured by storing an MD5 signature of the original unencoded audio data in the file header, which can be compared against later during decoding or testing.
- Fast: FLAC is asymmetric in favor of decode speed. Decoding requires only integer arithmetic, and is much less compute-intensive than for most perceptual codecs. Real-time decode performance is easily achievable on even modest hardware.
- Hardware support: Because of FLACs free reference implementation and low decoding complexity, FLAC is currently the only lossless codec that has any kind of hardware support.
- Streamable: Each FLAC frame contains enough data to decode that frame. FLAC does not even rely on previous or following frames. FLAC uses sync codes and CRCs (similar to MPEG and other formats), which, along with framing, allow decoders to pick up in the middle of a stream with a minimum of delay.
- Seekable: FLAC supports fast sample-accurate seeking. Not only is this useful for playback, it makes FLAC files suitable for use in editing applications.
- Flexible metadata: New metadata blocks can be defined and implemented in future versions of FLAC without breaking older streams or decoders. Currently there are metadata types for tags, cue sheets, and seek tables. Applications can write their own APPLICATION metadata once they register an ID
- Suitable for archiving: FLAC is an open format, and there is no generation loss if you need to convert your data to another format in the future. In addition to the frame CRCs and MD5 signature, flac has a verify option that decodes the encoded stream in parallel with the encoding process and compares the result to the original, aborting with an error if there is a mismatch.
- Convenient CD archiving: FLAC has a "cue sheet" metadata block for storing a CD table of contents and all track and index points. For instance, you can rip a CD to a single file, then import the CDs extracted cue sheet while encoding to yield a single file representation of the entire CD. If your original CD is damaged, the cue sheet can be exported later in order to burn an exact copy.
- Error resistant: Because of FLACs framing, stream errors limit the damage to the frame in which the error occurred, typically a small fraction of a second worth of data. Contrast this with some other lossless codecs, in which a single error destroys the remainder of the stream.
What FLAC is not:
- Lossy. FLAC is intended for lossless compression only, as there are many good lossy formats already, such as Vorbis, MPC, and MP3 (see LAME for an excellent open-source implementation).
- SDMI compliant, et cetera. There is no intention to support any methods of copy protection, which are, for all practical purposes, a complete waste of bits. (Another way to look at it is that since copy protection is futile, it really carries no information, so you might say FLAC already losslessly compresses all possible copy protection information down to zero bits!) Of course, we cant stop what some misguided person does with proprietary metadata blocks, but then again, non-proprietary decoders will skip them anyway.

KmPg2 is a very simple to use MPEG2 encoding wizard that allows the user to create high quality DVD compatible MPEG2 streams, without requiring any technical knowledge.
Creating high quality MPEG2 streams involves a lot more than simply feeding the input video to mpeg2enc/transcode/mencoder/etc.
The input video needs to be carefully prepared by pulling it through a chain of stream processing tools like low-pass filters, color correction filters, static and dynamic chroma and luma noise filters, scalers, and so on.
This processing chain needs to be tuned in such a way that all irrelevant data (like noise) is removed before the video enters the encoder. KmPg2 uses the video processing / encoding tools from the MJPEGTools project to achieve this.
KmPg2 features a profiler that enables the user to interactively create custom pre-processing pipelines that are optimised for a specific type of video material. These preprocessing pipelines (profiles) can be stored and used with the actual encoding wizard.
For example, you could create a profile that is designed especially for restoring a single aging VHS tape, or you could create a profile tuned for your digital camcorder and re-use it for every new recording that needs to be encoded.
At this moment KmPg2 accepts only one format of input video: Raw Sony DV video streams. This is the video format that most people use nowadays for high quality video processing and editing. Support for using AVI files for input is expected in future versions.
Main features:
- Produces DVD compliant MPEG2 streams
- Seperate chroma/luma noise filtering
- Low-Pass filtering support
- Contrast/Brightness/Saturation correction
- White balance correction
- Special black-and-white mode
- Crop image to Widescreen (16:9) or Panavision (2.35:1)
- Output conforms to Rec.601 specifications
- Both constant bitrate and variable bitrate encoding
- Unsharp masking
- Supports progressive and interlaced source material
- Support for removing overscan area
- Provides feedback to user about average/peak bitrates
- Show output on screen while encoding
- Generate encoding shell scripts
- Generate quick preview-encodings of parts of source video

flactagger is a simple command line tool to tag FLAC files. flactagger is a tagger, not a tag editor/viewer.

flactagger is free software licenced under the GNU GPL.

Asunder is a graphical CD ripper and encoder. It supports WAV, MP3, Ogg Vorbis, and FLAC. It has CDDB support and can create M3U playlists.
It uses GTK2 and is desktop environment independent. It is multithreaded, so it can rip and encode at the same time. It aims to make CD ripping as quick and easy as possible.
Main features:
- Support for WAV, MP3, Ogg Vorbis, and FLAC audio files
- Uses CDDB to name and tag each track
- Can encode to multiple formats in one session
- Creates M3U playlists
- Allows for each track to be by a different artist
- Does not require a specific desktop environment

ALAC comes from Apple Lossless Audio Codec, is a reference decoder for Apples lossless audio codec. The decoder has been written by reverse engineering Apples proprietory audio format.
Located here is a basic decoder for Apple Lossless Audio Codec files (ALAC). ALAC is a proprietary lossless audio compression scheme. Apple never released any documents on the format.
What I provide here is a C implementation of a decoder, written from reverse engineering the file format. It turns out that most of the algorithms in the codec are fairly well known. ALAC uses an adaptive FIR prediction algorithm and stores the error values using a modified rice or golumb algorithm. Further details are in alac.c.
Although an encoder is not provided, by using the decoder as a sort of specification it should be fairly trivial to write an encoder. By exploiting other lossless audio encoders, such as FLAC, the task will be much easier. Although one wouldnt be able to copy the compression algorithms verbatim, as adaptive compression is used in ALAC and not in FLAC. There are, however, a bunch of academic papers on the issue.
The program located here will not be able to handle all ALAC files, it can only handle mono or stereo files. ALAC allows up to 8 channels. It should be trivial to finish the implementation once I find files that I can test it with. Likewise the decoder only supports 16bit sample sizes. Again, it should be trivial to fix.
The decoder is fairly self explanatory, it can read an ALAC stream from either a file or from stdin, and write it as raw PCM data or as a WAV file to either stdout or a file. In theory one should be able to stream data to the decoder.
Enhancements:
- Supports ALAC files created with newer versions of quicktime, a number of speed optimisations that vastly improve performance, ability to decode m4a files with data before headers and windows improvements.