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 media type | audio/isac[1] |
|---|---|
| Developed by | Global IP Solutions, now Google Inc |
| Type of format | Audio compression format |
| 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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