internet Speech Audio Codec（iSAC）

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iSAC（Internet Speech Audio Codec）是一种宽带语音编解码器，由Global IP Solutions开发并被Google收购。它适用于VoIP应用和流媒体音频，在10到32 kbit/s的可变比特率下工作，支持16 kHz的采样频率。iSAC已作为专有编解码器发布，并于2011年成为开源WebRTC项目的一部分。

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