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.
===============================================
오늘 과제 미팅때문에
급하게 찾아보다가 재미있는 코덱을 하나 찾았습니다.
이름하여 iSAC...
이름에서 유추할 수
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;;)
'skype'에 해당되는 글 1건
- 2006/12/11 iSAC 2.0 release (0) _ beckgom








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