Skype publishes SILK audio codec source code

Skype has announced that it has published the source code for its SILK audio codec, introduced last year, which the company uses in its internet telephony applications for Windows and Mac OS X. Daniel Berg, Skype’s Chief Technology Officer said "This represents a key step in the development of an international standard for a wideband codec for use on the Internet,". The release of the source code comes as part of Skype's recent submission of an an Internet Draft to the Internet Engineering Task Force (IETF).

According to the announcement post on the Skype blog, the SILK audio codec is the result of three years of development and "enables super-wideband audio and optimises call quality, even in low network bandwidth environments". It also reportedly uses 50% less network bandwidth than previously required to transmit audio over the network.

Berg says that the developers focused on four things during the development of the codec, including improving audio bandwidth from 8 kHz to 12 kHz, real-time bandwidth scalability to deal with various network conditions and "balancing codec optimisation between voice, music and background noise". The company's fourth goal was to be able to deliver "a more consistent audio experience, regardless of network conditions and an individual’s voice signature".

The SILK source code is licensed for non-commercial use and is intended for "internal evaluation and testing purposes only". Companies wanting to use the codec for commercial products must request a SILK IP License (requires registration).

See also:

• Skype for Linux 2.1 Beta 2 adds screen sharing, a report from The H.
• Open source Linux Skype client development confirmed, a report from The H.

(crve)