Aften is an audio encoder which generates compressed audio streams based on ATSC A/52 specification. This type of audio is also known as AC-3 or Dolby® Digital and is one of the audio codecs used in DVD-Video content.
- Converts PCM source audio into AC-3 audio
- Allows for setting of production metadata such as dialogue level, downmix levels, and extended bitstream information
- Supports RAW, WAV, AIFF, and CAFF input file formats
- Supports single multi-channel or multiple mono source files
- Includes several optional pre-processing filters
- Uses multi-threading and assembly optimizations to speed up encoding
FLAC Frontend is a convenient way for Windows users not used to working with command lines to use the official FLAC tools. It accepts WAVE, W64, AIFF and RAW files for encoding and outputs FLAC or OGG-FLAC files. It is able to decode FLAC files, test them, fingerprint them and re-encode them. It has drag-and-drop support too. It is tested on Windows XP SP3 and Windows 7, but should work with Windows XP SP2 or newer. It requires .NET 2.0 or later.
A Mac OSX GUI frontend for open source audio codecs. There is an older Windows version available as well.
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng. While there are many MP2 encoders, TooLAME is well-known and widely used for its particularly high audio quality. It has been unmaintained since 2003, but is directly succeeded by the TwoLAME code fork (the latest version, TwoLAME 0.3.13, was released January 21, 2011). The name TooLAME is a play on LAME and Layer II.
FAAC is an Advanced Audio Coder (MPEG2-AAC, MPEG4-AAC). The goal of FAAC is to explore the possibilities of AAC and exceed the quality of the currently best MP3 encoders.
- High quality audio
- High-speed encoding
- LC, Main, LTP support
- DRM support through DreaM
WavPack is a completely open audio compression format providing lossless, high-quality lossy, and a unique hybrid compression mode. For version 5.0.0, several new file formats and lossless DSD audio compression were added, making WavPack a universal audio archiving solution.
In the default lossless mode WavPack acts just like a WinZip compressor for audio files. However, unlike MP3 or WMA encoding which can affect the sound quality, not a single bit of the original information is lost, so there’s no chance of degradation. This makes lossless mode ideal for archiving audio material or any other situation where quality is paramount. The compression ratio depends on the source material, but generally is between 30% and 70%.
The hybrid mode provides all the advantages of lossless compression with an additional bonus. Instead of creating a single file, this mode creates both a relatively small, high-quality lossy file that can be used all by itself, and a “correction” file that (when combined with the lossy file) provides full lossless restoration. For some users this means never having to choose between lossless and lossy compression!
WavPack employs only well known, public domain techniques (i.e., linear prediction with LMS adaptation, Elias and Golomb codes) in its implementation. Methods and algorithms that have ever been patented (e.g., arithmetic coding, LZW compression) are specifically avoided. This ensures that WavPack encoders and decoders will remain open and royalty-free.