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Streaming audio over IP networks is being increasingly used by broadcasting companies, among others, to provide high-quality audio feeds (for example to provide audio links between a sports venue or concert hall and the broadcaster's studios). Audio quality and delay (on duplex transmissions) are key issues for contribution links, and in the past these links have made widespread use of ISDN services but these are becoming increasingly difficult or expensive to obtain in some parts of Europe and are being phased out in others. Many proprietary systems exist for transporting high quality audio over IP based on TCP or UDP for contribution purposes. Within a single building or music venue, audio over Ethernet is more likely to be used instead, avoiding audio data compression and IP encapsulation.
[edit] Examples of useThe BBC has begun using Audio Contribution over IP in Scotland as part of the Pacific Quay development in Glasgow. The platforms used are the WorldNet Oslo for multiple channel contribution and distribution with the WorldCast Horizon deployed in stereo drop-off locations. The aptX algorithm is used with 24 bit depth and 48 kHz sampling. This is a higher specification than normal CDs, and is used to avoid concatenation effects. There have been problems found in the past when NICAM and MPEG audio paths have been used in sequence in the broadcast chain. The audio heard in the presentation studio has been acceptable, but listeners at home using DAB have complained about artifacts. To avoid this, aptX is used at high bit rates. This is a differential PCM system that does not use psychoacoustic processing. It is able to resynchronise after momentary loss of the IP path without noticeable clicking, and is licensed to most Codec manufactures. A similar system is being installed in the English Regions and will be installed in Wales and Northern Ireland. The audio packets are sent using UDP over the BBC’s Layer3 network. To ensure the audio is not corrupted, Quality of service (QoS) is set to ensure that the packets are given priority over other network traffic. [edit] EBU N/ACIPThe European Broadcasting Union (EBU) together with many equipment manufacturers has defined a common framework for Audio Contribution over IP in order to achieve interoperability between products. The frameworks define RTP as a common protocol and media payload type formats according to IETF definitions. SIP is used as signalling for call setup and control. The recommendation is currently published as an EBU Tech3326 document.[1] Even if open source tools are used for reference testing, codecs such as ogg vorbis, speex and flac are excluded, even if the development is payed by tax payers and broadcasting fees. [edit] References
[edit] External links
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