CELT 0 votes

The CELT codec is a compression algorithm for audio. Like MP3, Vorbis, and AAC it is suitable for transmitting music with high quality. Unlike these formats CELT imposes very little delay on the signal, even less than is typical for speech centric formats like Speex, GSM, or G.729.

Using CELT application developers can build software that allows musicians to perform together across the Internet, or simply build great sounding telephony systems. Why shouldn't your telephone sound as good as your stereo?

Like other open and unencumbered technology from the Xiph.Org Foundation CELT requires no royalties and no complicated licensing.

CELT stands for “Constrained Energy Lapped Transform”. It applies some of the CELP principles, but does everything in the frequency domain, which removes some of the limitations of CELP. CELT is suitable for both speech and music and currently features:

1. Ultra-low latency (typically from 5 to 22.5 ms)
2. Full audio bandwidth (≥20kHz; sample rates from 8 kHz to 48 kHz)
3. Support for both speech and music
4. A quality/bitrate trade-off competitive with widely used high delay codecs
5. Stereo support
6. Packet loss concealment
7. Constant bit-rates from 32 kbps to 128 kbps and above
8. A fixed-point version of the encoder and decoder

The CELT codec is meant to bridge the gap between Vorbis and Speex for applications where both high quality audio and low delay are desired.

LAME 0 votes

LAME is a high quality MPEG Audio Layer III (MP3) encoder licensed under the LGPL.

Following the great history of GNU naming, LAME originally stood for LAME Ain't an Mp3 Encoder. LAME started life as a GPL'd patch against the dist10 ISO demonstration source, and thus was incapable of producing an mp3 stream or even being compiled by itself. But in May 2000, the last remnants of the ISO source code were replaced, and now LAME is the source code for a fully LGPL'd MP3 encoder, with speed and quality to rival and often surpass all commercial competitors.

LAME is an educational tool to be used for learning about MP3 encoding. The goal of the LAME project is to use the open source model to improve the psycho acoustics, noise shaping and speed of MP3. LAME is not for everyone - it is distributed as source code only and requires the ability to use a C compiler. However, many popular ripping and encoding programs include the LAME encoding engine, see: Software which uses LAME.

internet Speech Audio Codec(ISAC) 0 votes

internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS) (acquired by Google Inc in 2011). It is suitable for VoIP applications and streaming audio. The encoded blocks have to be encapsulated in a suitable protocol for transport, eg. RTP.

It is one of the codecs used by AIM Triton, the Gizmo5, QQ, and Google Talk. It was formerly a proprietary codec licensed by Global IP Solutions. As of June 2011, it is part of open source WebRTC project, which includes a royalty-free license for iSAC when using the WebRTC codebase.

WavPack 0 votes

Hybrid Lossless Audio Compression

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.