internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS) (acquired by Google Inc in 2011[2][3]). It is suitable for VoIP applications and streaming audio. The encoded blocks have to be encapsulated in a suitable protocol for transport, e.g. 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,[4] which includes a royalty-free license for iSAC when using the WebRTC codebase.[5]
Parameters and features [edit]
- Sampling frequency 16 kHz[1] (or 32 kHz according to WebRTC[6][7])
- Adaptive and variable bit rate (10 kbit/s to 32 kbit/s) (or 10 kbit/s to 52 kbit/s according to WebRTC[6][7])
- Adaptive packet size 30 to 60ms
- Complexity comparable to G.722.2 at comparable bit-rates
- Algorithmic delay of frame size plus 3ms
See also [edit]
References [edit]
- ^ a b "RTP Payload Format for the iSAC Codec - Internet Draft - draft-legrand-rtp-isac-02.txt". 2009. Retrieved 2011-06-23.
- ^ Dana Blankenhorn (2010-05-18). "Why Google bought Global IP Solutions". Retrieved 2011-06-23.
- ^ "iLBC Freeware". Retrieved 2011-06-23.
- ^ http://sites.google.com/site/webrtc/faq#TOC-What-is-the-iSAC-audio-codec-
- ^ http://sites.google.com/site/webrtc/license-rights/additional-ip-grant
- ^ a b "WebRTC FAQ - What are the parameters of iSAC?". Retrieved 2011-06-23.
- ^ a b "WebRTC components". Retrieved 2011-06-23.
External links [edit]
