While the transition from analogue to digital is nearly complete in all realms of professional audio, the next wave of change is already here. 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.
Today’s broadcasters are seeing a steady increase in the number of audio and video channels in use. Controlling and routing all these signals using legacy non-network transports can be a daunting task, involving expensive specialised 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 many 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 specialised 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. Even metadata delivery and synchronisation is a natural fit for IP networking, which provides an open and well understood platform for development of delivery systems that non-networked systems cannot match.
Gigabit and faster network speeds have made IP networking an indispensable medium. In broadcast, it is simply the best way to transmit bit-perfect audio between as many devices as needed, with low latency and tight synchronisation. Today’s IP technology offers a best-of-all-worlds combination of increasing performance with decreasing costs, a trend sustained by the growing use of IP networks across nearly every type of industry.
The growth of IP systems for media has resulted in the rapid deployment of hundreds of new products in all key audio categories. Today, there are more than 800 networked audio products available. Complete end-to-end systems from microphones to loudspeakers may be specified across different brands, and new products are constantly being released.
Manufacturers have risen to the challenge of integrating digital audio networking with 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 them to bridge older digital protocols such as MADI or CobraNet to a modern Audio over IP network, and dedicated convertors are available for nearly any format. This permits audio flows to remain entirely in the digital domain, avoiding signal degradation from intermediate A/D and D/A conversions.
Building bridges – AES67
Realistic, functional interoperability is required to allow facilities to use products they prefer, even if they employ different audio over IP technology. AES67 seeks to accomplish this task. The AES67 standard is a networked audio interoperability specification developed by the Audio Engineering Society. It describes techniques for exchanging digital audio on a TCP/IP networking using RTP (Real-time Transport Protocol). Additionally, AES67 specifies particular implementation constraints to facilitate interoperability between implementations.
It is important to note that AES67 is not a complete audio networking solution and does not include all of the components required for that role. Technologies provide the layers of discovery, routing, diagnostics, auto-configuration, software and support needed to form a workable audio networking solution for both users and manufacturers. In contrast, a system using AES67 to connect multiple network solutions still requires separate management tools for each solution in order to control devices, making setup much more complex and error-prone than with a single-solution system.
AES67 promises basic interconnectivity at its core, and has fairly modest and achievable goals. Primarily focused on the audio networking transport element, AES67 does not specifically address system control, signal routing or channel labelling. For audio quality, it requires 48kHz, 24-bit stream with one-millisecond latency as the lowest common denominator. AES67 allows, but does not require, support for higher sample rates and different bit depths, which means that supported audio formats may vary between devices in the ecosystem.
Since AES67 is essentially a set of network standards around how audio channels move across an IP network, it 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 specialised network equipment. This is significant, as it is the first specification to achieve this goal. The AES67 transport operates with common off-the-shelf switches in a Layer 3 architecture. Therefore, we do not expect AES67 deployments to face the adoption delays that have challenged AVB rollouts.
A long-term solution
In creating the AES67 interoperability standard, the AES organisation used proven existing standards, which mitigates the risk of moving to networked audio.
AES67 does not replace complete audio networking solutions, but enhances them by providing a standards-driven approach for useful, low-level interconnection with others. Leading audio networking solutions will continue to provide the features necessary for reliable, complete systems that are easy to use and understand, including matrix-style signal routing software, virtual soundcards, network health and clock status monitoring, real-world naming for devices and channels, and much more.
In contrast, AES67 only specifies the baseline connectivity of audio streams, and more closely resembles an audio-over-IP version of MADI or AES3. It focuses on how audio channels move through the network between points, without defining how routing or switch-defined control may occur.
It’s important that companies in this area continue to be involved in the development and tracking of new standards and protocols that further the possibilities of audio networking for TV and radio broadcasting. One thing is certain: the industry will continue to develop new standards that require implementation in a sensible way, and it is important to have robust networking solutions that can evolve to incorporate the latest standards.
Brad Price is Product Marketing Manager at Audinate.