lossyWAV is a new free lossy pre-processor for PCM audio contained in the WAV file format. It reduces bit depth of the input signal, which, when used in conjunction with certain lossless codecs, reduces the bitrate of the encoded file significantly compared to unpreprocessed compression. lossyWAV's primary goal is to maintain transparency with a high degree of confidence when processing any audio data.

Lame MP3 encoder (command line tool) and ACM MP3 encoder codec that you can use in most applications for example in Virtualdub, to install the ACM codec just extract all files and right click on the LameACM.inf and select Install.

Fraunhofer IIS mp3surround - Сотрудники Института Фраунгофера предложили набор кодеков и программного обеспечения который в будущем сменит формат MP3, для создания и воспроизведения "многоканальных" МР3-файлов.

MPEG-4 ALS (audio lossless coding – кодирование звука без потерь), разработанные корпорацией NTT (Nippon Telegraph and Telephone). Спецификации, разработка которых велась с 2002 года, были приняты ISO (International Organization for Standardization) и IEC (International Electro technical Commission).

Directshow Filters for Ogg Vorbis, Speex, Theora and FLAC - набор фильтров, позволяющий воспроизводить файлы формата Ogg и OGM в любом DirectShow-плеере ( Windows Media Player и RadLight 4). В комплекте также поставляются декодеры для Ogg Vorbis & Speex (audio) , Theora (video) и FLAC.

A graphical interface for MP3 decoder/encoder
Lame Front-End is a graphical interface for MP3 decoder/encoder.

Lame Front-End is a graphical interface for the popular and very fast MP3 encoder and decoder - LAME. It offers easy access to all LAME parameters without the need for command line usage.

The program offers a simple mode (for beginners), which was designed according to the LAME authors guidelines (http://lame.sourceforge.net/lame_ui_example.php), and an advanced mode which reveals additional expert options.

Lame Front-End can work in batch mode, so it can be used by external applications to encoding/decoding files, e.g. by Exact Audio Copy, CDex.

- Converting WAV, PCM, AIFF, RAW files to MP3 (encoding).
- Converting MP3, MP2, MP1 files to WAV (decoding).
- Two modes: Simple for beginners, Expert for advanced users.
- Saving ID3v2 tags in the MP3 files.
- Integration with the Windows Explorer. Program adds a command Open with Lame Front-End to the context menu for supported files.
- Saving and loading LAME parameters.
- Small, easy to use and don't need any installation.

WavPack is a completely open audio compression format providing lossless, high-quality lossy, and a unique hybrid compression mode. Although the technology is loosely based on previous versions of WavPack, the new version 4 format has been designed from the ground up to offer unparalleled performance and functionality.

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.

AC3Filter is a high quality free DirectShow filter designed for real time audio decoding and processing. It has a priority on wide functionality and convenient settings. Filter can decode following audio formats: AC3/DTS/MPEG Audio. It also supports multi-channel and digital (SPDIF) outputs.

Fraunhofer MPEG Layer-3 Audio Decoder - Этот кодек позволяет воспроизводить файлы, закодированные в очень популярном формате музыки MP3

CoreAAC, DirectShow-Decoder для AAC аудио кодека, использует механизм FAAD2 , включая поддержку SBR (HE-AAC)