Audio Over IP Networking In Broadcast

The transition from analog to digital is nearly complete. The use of computer networking as a cost-effective and technically superior means of connecting, routing and managing AV systems is a reality that is changing equipment, workflows and possibilities today, writes Brad Price, Product Marketing Manager, Audinate.

IP comes to broadcast
Today's broadcasters are seeing a steady increase in the number of audio and video channels in use. Controlling and routing all of these signals using legacy non-network transports can be a daunting task, involving expensive specialized routers and complex workflows that vary from one manufacturer to another. Adding and incorporating new equipment can mean upgrading and replacing many other pieces of gear to maintain compatibility and provide bandwidth. Simple tasks such as maintaining lip sync can require format conversions and expert use of costly equipment.

In contrast, an IP-based solution can handle hundreds or thousands of channels of audio, connecting dozens of devices using inexpensive Cat5E cabling and a few inexpensive gigabit network switches. There are no specialized routers needed to provide conversion and distribution; all changes are made quickly and easily in software running on ordinary computers, with no disruption of production activities. Metadata delivery and synchronization is a natural fit for IP networking, which provides an open platform for development of delivery systems that non-networked systems cannot match.

Gigabit (and faster) network speeds have made IP networking the best way to transmit bit-perfect audio between as many devices as needed, with low latency and tight synchronization. IP technology offers a best-of-all-worlds combination of increasing performance with decreasing costs, a trend that is sustained by the growing use of IP networks across nearly every type of industry.

Available products
The growth of IP systems for media is reflected in the hundreds of new products in all key audio categories. Today there are over 800 networked audio products available on the market, with Audinate's Dante audio networking used in more than 60%. Complete end-to-end systems from microphones to loudspeakers may be easily specified across multiple vendors, and new products are constantly being released.

Readily available I/O options add audio networking to existing systems, so older equipment can remain a vital part of modern broadcast facilities. Many mixers, routers, intercom systems and amplifiers support multiple interface cards, allowing older digital protocols such as MADI or CobraNet to bridge to a Dante network, and dedicated convertors are available for nearly any format. Audio flows to remain entirely in the digital domain, avoiding signal-degradation from A/D and D/A conversions.

Building Bridges with AES67
Realistic, functional interoperability is required to allow facilities to use products they prefer, even if they employ different audio over IP technology.

AES67 targets this need. AES67 is a networked audio interoperability specification developed by the Audio Engineering Society that describes methods for exchanging digital audio on a TCP/IP network using RTP, or Real-time Transport Protocol.

It is important to note that AES67 is not a feature complete audio networking solution and does not include all the components required for that role. Fully operable audio networking requires tools for discovery, routing, diagnostics and configuration. A system using AES67 to connect multiple network solutions will still require separate management tools for each solution in order to control devices, making setup more complex and error-prone than a single-solution system.

AES67 delivers basic interconnectivity, and focuses upon the audio network transport element. It specifies 48kHz, 24-bit streams with one-millisecond latency as a lowest common denominator. AES67 allows support for higher sample rates and bit depths, which means that audio formats may vary between devices in the ecosystem.

AES67 represents a pragmatic evolution in audio networking. Unlike previous specifications (e.g. AVB), AES67 offers a standards-based way to deliver multichannel audio between devices across a network without requiring specialized network equipment, and is the first specification to achieve this goal. Like Dante, the AES67 transport operates with common off-the-shelf switches in a Layer 3 architecture, and so we anticipate that AES67 deployments will not face the adoption delays that have challenged AVB rollouts.

A Long Term Solution
AES67 enhances complete audio networking solutions by providing a standards-driven approach for useful, low-level interconnection with others. By specifying only the baseline connectivity of audio streams, it resembles an Audio over IP version of MADI or AES3.

Leading audio networking solutions will continue to provide the features necessary for reliable, complete systems that are easy to use and understand, including sophisticated signal routing software, virtual soundcards, network health and clock monitoring, user naming for devices and channels, and much more.

It's important that companies in this area continue to be involved in the development and implementation of new standards and protocols that further the possibilities of audio networking for TV and radio broadcasting. Interconnectivity standards like AES67 provide a common denominator for manufacturers that will help drive the adoption of IP technologies by removing barriers and doubts for users.

www.audinate.com

This article is also available to read at BFV online.

(JP/LM)