Internet Speech Audio Codec
Audio codec standard
From Wikipedia, the free encyclopedia
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.
Internet mediatype
audio/isac[1]
DevelopedbyGlobal IP Solutions, now Google Inc
Type of formatAudio compression format
| internet Speech Audio Codec | |
|---|---|
| Internet media type |
audio/isac[1] |
| Developed by | Global IP Solutions, now Google Inc |
| Type of format | Audio compression format |
Written inC
| Codec | |
|---|---|
| Developers | 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 |
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 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
