internet Speech Audio Codec(iSAC)
2011-08-31 09:18
841 查看
From Wikipedia, the free encyclopedia
internet Speech Audio Codec (iSAC)
iSAC Codec
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
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] |
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
相关文章推荐
- ISAC(internet Speech Audio Codec):
- Senior Software Engineer - Audio/Speech Codec(职位推荐)
- RTP Payload Format for Opus Speech and Audio Codec
- rk3288 audio 驱动分析 (https://github.com/54shady/kernel_drivers_examples/tree/master/debug/codec)
- Audio codec linux driver 之 ALSA 架构的 DAPM 学习
- 音频基础 DAI:Digital Audio Interfaces(音频设备的硬件接口 codec android)
- 常用的ITU Speech Codec大全(G.711,G.722.1,G.722.2,G.723.1,G.726,G.728,G.729,G.729.1)
- 常用的ITU Speech Codec大全(G.711,G.722.1,G.722.2,G.723.1,G.726,G.728,G.729,G.729.1)
- audio:audio codec 分类小结
- Audio Codec介绍-1
- Audio Codec介绍.docx
- Audio Codec介绍-2
- 瑞典皇家理工学院工程实例:An Advanced Speech Codec for a Voice over IP Transmission System
- Audio Codec : MPEG2 AAC 系统描述
- DAPM之七:文档《AUDIO CODEC DAPM》放出
- speech codec (G.711, G.723, G.726, G.729, iLBC)
- 语音音频压缩格式 speex vs nellymoser:The CELT ultra-low delay audio codec
- 英国 Scalable Audio Codec
- Audio Codec : MPEG2 AAC -- TNS
- CentOS 5.8 asterisk-1.8.10.1 安装之一:安装,添加蓝牙支持,添加AMR-NB audio codec