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  1. 2006/12/11 iSAC 2.0 release (0) _ beckgom

iSAC 2.0 release

Coding fabula | 2006/12/11 20:26 | beckgom
Global IP Sound (GIPS) today announced the availability of version 2.0 of its iSAC codec.  iSAC 2.0 is now optimized for use in hardware devices. This solution allows the makers of IP phones, gateways and chipsets to deliver products that offer the same high quality voice experience that users of software-based VoIP softphones have achieves using GIPS' voice platform. (i.e. GoogleTalk, Skype, AOL are all GIPS customers.)

The latest fixed-point version of the GIPS iSAC adaptive VoIP codec, part of the GIPS Soundware voice processing software suite, is designed to deliver wideband quality in both low- and high-bit rate applications. Several IP-chipset manufacturers including Texas Instruments will incorporate GIPS's iSAC 2.0 codec into their VoIP solutions.

"As a leading enabler of wideband telephony solutions, we realize the importance of technologies that facilitate high quality voice communications," said Fred Zimmerman, executive director, VoIP customer premises solutions, Texas Instruments. "The addition of GIPS' wideband codec to our portfolio further enables us to offer our customers VoIP solutions that deliver an unparalleled user experience that surpasses what callers have grown accustomed to with their traditional phone service."

iSAC was previously only available in floating point code for use in softphones and other applications running on PCs and less resource dependent platforms. iSAC 2.0 is optimized to run on chips embedded within VoIP phones and other hardware devices, guaranteeing high voice quality.  This release will also be available in a low complexity version to ease the integration in resource intensive devices like mobile phones. In addition, it facilitates the interoperability of IP phones, gateways, and chipsets with the hundreds of millions of iSAC-enabled softphones and Internet-based voice services that are currently deployed through GIPS' major ISP customers such as Yahoo!, Skype and AOL.

"The market has not only come to appreciate the benefits of wideband telephony, but prefer it over traditional PSTN" said Roar Hagen, chief technology officer, Global IP Sound. "As manufacturers begin to deploy VoIP enabled hardware, users are going to expect the same full, robust sound they are accustomed to from their softphones.  GIPS is proud to be able to deliver this high quality audio on all platforms and devices."

The GIPS iSAC technology automatically adjusts transmission rates, supporting real-time multimedia, conferencing, distance learning, and multi-user gaming applications in a VoIP environment. Due to the nature of wideband audio, iSAC can deliver sound quality that exceeds that of PSTN calls by utilizing a greater range of the speech signal.  The codec also handles other challenges, including non-speech audio such as music and background noise.

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오늘 과제 미팅때문에 급하게 찾아보다가 재미있는 코덱을 하나 찾았습니다.
이름하여 iSAC...
이름에서 유추할 있듯이 Global IP Sound 라는 회사에서 (iLBC를 만든 회사죠)
release한 코덱입니다.
다만 오픈 코덱은 아닌것 같네요. 기사가 2006년 9월에 난 건데, 아직 홈페이지에 코드 오픈이라든지 이런 것은 찾을 수 없습니다.
패킷 사이즈는 iLBC가 그랬듯이 가변적이고 (30ms-60ms)
비트레이트는 EVRC처럼 채널 상태에 따라 바뀌는 것이 가능하다고 하네요. (10-32 kbit/s)
아직 대역폭이 8kHz로 광대역 처리만 가능하고, AMR-WB보다 좋은 성능을 나타낸다고 합니다. (들어봤음 좋겠네요. AMR-WB도 훌륭한 코덱인데..)
그리고 야후, 스카이페, AOL 등과 작업을 한다고..조금만 지나면 품질은 느낄 수 있을 것 같습니다.
(그래도 소스코드를 공개하라!! 공부좀 하게;;)