internet Speech Audio Codec(iSAC)

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From Wikipedia, the free encyclopedia  
internet Speech Audio Codec (iSAC)
Internet media type audio/isac[1]
Developed by Global IP Solutions, now
Google Inc
Type of format Audio compression format

 

iSAC Codec

Developer(s) Global IP Solutions, now
Google Inc
Written in C
Operating system Cross-platform
Type Audio codec,
reference implementation
License formerly proprietary, now 3-clause BSD
Website [1]

 

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, eg.

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 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

 

 

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