internet Speech Audio Codec
| Internet media type |
audio/isac Google Inc |
|---|---|
| Type of format | Audio compression format |
reference implementation | |
|---|---|
| License | formerly proprietary, now 3-clause BSD |
| Website | webrtc |
internet Speech Audio Codec (iSAC) is a
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
- 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.722.2at comparable bit-rates
- Algorithmic delay of frame size plus 3 ms
See also
- iLBC
- Opus (audio codec)
- SILK
- G.722
- Speex
- List of codecs
References
- ^ a b c Grand, Tina le; Jones, Paul; Huart, Pascal; Shabestary, Turaj Zakizadeh; Alvestrand, Harald T. (2013). "RTP Payload Format for the iSAC Codec". Retrieved 2016-04-30.
- ZDNet. Retrieved 2011-06-23.
- ^ "iLBC Freeware". Archived from the original on 2011-07-05. Retrieved 2011-06-23.
- ^ webrtc.org/faq/#what-is-the-isac-audio-codec Archived 2011-06-07 at the Wayback Machine.
- ^ webrtc.org/license/additional-ip-grant/.
- ^ a b "WebRTC FAQ - What are the parameters of iSAC?". Archived from the original on 2016-11-17. Retrieved 2011-06-23.
- ^ a b "WebRTC components". Archived from the original on 2011-06-28. Retrieved 2011-06-23.
External links
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