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IP audio codec : ウィキペディア英語版
IP audio codec
IP audio codecs are used to send broadcast-quality audio over IP from remote locations to radio and television studios around the globe. IP codecs are ideal for use in remote broadcasts, as studio/transmitter links (STLs) or for studio-to-studio audio distribution.
IP audio codecs use audio compression algorithms to send high fidelity audio over both wired broadband IP networks and wireless 3G, 3.5G and 4G broadband networks.
Audio-over-IP has proved itself to be the broadcast network infrastructure of the future, and as a consequence, increasing numbers of broadcasters are migrating to low-cost wired and wireless IP networks instead of older and more costly fixed line technologies such as ISDN, X.21 and POTS/PSTN. ISDN and POTS/PSTN leased lines are also being phased out in Europe and Australia, increasing the push into IP technologies for audio broadcasting.
IP networks are more flexible, cheaper to upgrade and just as reliable as older network technologies. As a result, broadcasters using IP codecs are able to design and operate more adaptable audio networks with streamlined workflows and reduced operating costs.
==Examples==
Tieline Technology is one company focused on both wired and wireless broadband IP codec solutions. The latest IP audio codecs can send broadcast audio over stereo unicast, multicast and multiple unicast connections. Using multicast and multiple unicast connections, audio can be sent over IP networks from one IP audio codec to several destination audio codecs.
IP codecs generally use session initiation protocol in order to connect to a variety of different codecs designed by different manufacturers.
Another example would be APT Codecs who provide IP Audio Codecs for use in professional studio transmitter links (STLs) and studio networking. IP STLs and Studio-to-Studio Links are absolutely vital for broadcasters. Traditionally these links have been implemented with synchronous (Tx, Rx and Clock) data circuits. For wired circuits, broadcasters usually contract Telephone Companies to provide fixed bandwidth T1, E1 or X.21 / V.35 terminated circuits (or the Telco transports the audio for the radio station using synchronous networks as the preferred carrier).
While this arrangement has worked well for many years, it requires broadcasters to allocate considerable amounts from their budget for the ongoing operational costs of the Telco services. With the advent of IP technology, broadcasters have been looking at minimizing these operational costs by replacing their existing synchronous networks with packetized ones.
However, as a transport mechanism, IP presents a number of challenges that need to be addressed by the Chief Engineer before migrating away from synchronous networks. These issues are not insurmountable, and, if correctly and intelligently implemented; the use of IP will greatly assist a radio station’s efficiency.
Primary issues of concern for a Chief Engineer are as follows:
*Latency
*Lost Packets
*Packet Size
*UDP versus TCP
*Jitter
*Error Correction / Concealment
*Clock Skew
*Audio Algorithms
*IPv6 Compatibility
IP as a transport mechanism for broadcasters is here to stay because it allows Radio broadcast networks to bundle their audio with data, reduce operational costs and amalgamate IT and audio into a single resource. However, these massive and well-defined advantages come with some safety warnings - networks should be well managed, packets should be prioritized and correct choices should be made with regard to audio compression. Assuming all these boxes are ticked, then broadcasters will enjoy the benefits of the transition away from synchronous networks without running into serious problems

抄文引用元・出典: フリー百科事典『 ウィキペディア(Wikipedia)
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