internet Speech Audio Codec

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internet Speech Audio Codec
Internet media type audio/isac[1]
Developed by Global IP Solutions, now Google Inc
Type of format Audio compression format
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 webrtc.org

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 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[edit]

References[edit]

External links[edit]