internet Speech Audio Codec
Appearance
Internet media type |
audio/isac[1] |
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Developed by | Global IP Solutions, now Google Inc |
Type of format | Audio compression format |
Developer(s) | Global IP Solutions, now Google Inc |
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Written in | C |
Operating system | Cross-platform |
Type | Audio codec, reference implementation |
License | formerly proprietary, now 3-clause BSD |
Website | webrtc |
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
- ^ a b c "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.
- ^ a b "WebRTC FAQ - What are the parameters of iSAC?". Retrieved 2011-06-23.
- ^ a b "WebRTC components". Retrieved 2011-06-23.