Has analog lost the battle?
In the realm of telephony, Audio over IP (AoIP) has become an accepted alternative to regular circuit-switched calls. Applications like Skype and Apple Messages are popular for general chat and international calls. Skype calls have become accepted for interviews during newscasts, but for ‘broadcast-quality’ calls it has taken longer for AoIP to become accepted. Synchronisation has been a big issue with audio-for-video, but now fixed with several protocols.
For the broadcaster, there must be a good reason to move from existing audio-specific protocols to AoIP. There needs to be a sound business case, will it reduces costs or provide new features.
Through the life of broadcast audio, the XLR connector has provided sterling service. It still remains the primary connector for analog and AES3 digital audio. The 1990s saw the introduction of the multi-channel interface MADI (AES10), which carries up to 64 digital audio channels over a copper coax or fibre links.
Audio is frequently embedded in the associated video channel, interleaved in the ancillary data space that dates back to the days of CRT horizontal and vertical flyback (ask a retired broadcast engineer about all that).
These three methods of interconnection add an element of complexity to systems. Audio can be routed a distributed as a copper pair, AES3 or analog. It can as MADI, or SDI embedded. This inevitably leads to many stages of embedding and de-embedding, multiplexing to MADI and demultiplexing.
Packet switching over IP presents an alternative which can radically simplify a system to a network switch connected in star fashion to the audio sources and processing gear.
Connecting audio over Ethernet is already commonplace for studio communications, but the migration of program audio has taken longer.
Digital audio has the huge advantage over video in the relatively small bandwidth required. An uncompressed audio stereo pair is around the same data rate as highly compressed standard-def video, circa 3Mb/s. So 1Gb Ethernet can easily handle many, many circuits, and 10Gb, well that’s a luxury.
Several formats are now available for AoIP—AVB, Dante, Ravenna et al. AES67 (the AES standard for audio applications of networks - High-performance streaming audio-over-IP interoperability), provides recommendations for interoperation between devices from different vendors.
These formats overcome the vagaries of IP networks, jitter, delays, which made broadcast quality circuits unreliable in the past. Technologies like forward error correction and quality of service management ensure low error rates.
Once audio data is formatted for transmission over IP networks, new possibilities emerge. Software defined networks allows reconfiguration that is just not possible with hard-wired AES3 and MADI circuits. Broadcasters are looking to achieve better use of facilities. There is no point in having an expensive sound control room lying idle while the studio floor is set. If sound control rooms can be assigned around a facility they can be better utilised. IP allows anything from commentary boxes to stage boxes to be connected over Ethernet without the need for special circuits and routing can use low-cost commodity IT Ethernet switches.
The associated switch to IP becomes compelling as more and more audio processing products move from hardware processing (DSP and FPGA) to software, which can run on commodity PC platforms. AoIP provides the same features and more, supports the high audio quality standards, yet promises lower costs. Operators are freed from worrying about physical interconnects and can connect sources to destination over virtual routes.
For audio system builders this is now all a reality. The video guys are still developing and trialling their video-over-IP systems, so the business case for migration is not so clear. Give it a few years, and all will be different.
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