internet Speech Audio Codec
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
- Sampling frequency of 16 kHz (wideband) or 32 kHz (superwideband)[1][6][7]
- Adaptive and variable bit rate of 10 kbit/s to 32 kbit/s (wideband) or 10 kbit/s to 52 kbit/s (superwideband)[1][6][7]
- Adaptive packet size 30 to 60 ms
- Complexity comparable to G.722.2 at comparable bit-rates
- Algorithmic delay of frame size plus 3 ms
See also
References
- 1 2 3 "RTP Payload Format for the iSAC Codec". 2013. Retrieved 2016-04-30.
- ↑ Dana Blankenhorn (2010-05-18). "Why Google bought Global IP Solutions". Retrieved 2011-06-23.
- ↑ "iLBC Freeware". Retrieved 2011-06-23.
- ↑ sites.google.com/site/webrtc/faq#TOC-What-is-the-iSAC-audio-codec-.
- ↑ sites.google.com/site/webrtc/license-rights/additional-ip-grant.
- 1 2 "WebRTC FAQ - What are the parameters of iSAC?". Retrieved 2011-06-23.
- 1 2 "WebRTC components". Retrieved 2011-06-23.
External links