Audio over IP, re-thinking audio delivery



Since the early 1980s, complex audio signal routing has most often been done with the help of Time Division Multiplexing (TDM), a method of transmitting and receiving independent digital signals through a single cable that can carry more than one channel.

The broadcast industry first adopted a standard developed by the Audio Engineering Society (AES) for the exchange of digital audio signals between professional audio devices called AES3, and later Multichannel Audio Digital Interface (MADI), in order to transport signals. The growing number of audio channels required in production and broadcast called for digital audio routers that could handle a larger capacity of digital signals. However, TDM connections were not very efficient as the traffic grew, and the industry needed a re-vamp to address the situation. Today, network technologies finally allow the industry to fully realise real-time applications: networking standards have evolved to an extent that turns interactive audio applications into a realistic possibility.

Throughout this year we have seen a huge push in technologies to facilitate IP-based infrastructures in the media industry. With terminologies like High Frame Rate (HFR) video, High Dynamic Range (HDR) and Ultra High Definition (UHD) being bandied about in the broadcast arena, which all reference video production and broadcasting, we have started to see a new term being thrown around in the audio world: that of Next Generation Audio (NGA).

NGA is a term being used not only to describe the emerging object or channel-based immersive sound formats (Dolby Atmos is a good example) that go together with the new broadcast and media distribution video formats, but also to describe new practices that are now acknowledged to be dependent on IP-based workflows and distribution platforms, as recommended by both the Society of Motion Picture and Television Engineers (SMPTE) and the Institute of Electrical and Electronics Engineers (IEEE).

Riding on the back of all the exciting changes that are currently happening in the media business, the topic of audio-over-IP (AoIP) and the ongoing transition to fully IP-enabled operations in all platforms has become big news.

Internet Protocol (IP) technology is revolutionising the broadcast industry, pushing aside traditional approaches that relied on a mass of cable connections running from one point to another, each restricted to transporting a given type of media or data between the two locations and, in audio terms, limited to a relatively low channel count. These are being replaced with network connections that allow for higher audio channel counts and the ability to pass different types of media and data over the same connection.

In an IP-based operation, a device only needs one connection to a network to be able to send and receive audio and data to and from any other device on the network, rather than needing a direct connection with every other device. Not only is the number of cables massively reduced, but the cables themselves are shorter, since they only need to connect with the nearest switch, rather than running all the way to the piece of kit they connect with. As is well known, network-based systems rely on conventional and inexpensive off-the-shelf IT technology originally designed to forward a large number of data packets without having to deliver them in real-time. It is precisely for this reason that IP technology has been at a disadvantage with regard to audio transportation.

Yet the technological progress that has been made in this field makes it increasingly attractive for broadcasters, post production houses, concert venues and sound studios to implement AoIP technology. Besides the tremendous cost-saving effect on equipment, network-based solutions for audio delivery hold the potential of enabling broadcasters, recording studios and live venues to leverage their existing infrastructure and to achieve greater flexibility in terms of set-up, network configuration and content sharing.

While established AoIP standards like Dante, Ravenna Network and AVB have been designed with the fields of broadcasting and live performance events in mind, they have also paved the way for multi-channel setups and the distribution of real-time high resolution audio streams in recording and mix-down studios.

It is, for instance, now possible to work with multiple loudspeaker setups like 5.1, Dolby Atmos, or Auro 3D simply by using a single Ethernet router to distribute audio material among several devices. AoIP has expanded to become one of the most progressive audio technologies today, so much so that even home-based studio musicians are ditching their USB interfaces for networked solutions.

The audio industry as a whole, in collaboration with the Audio Engineering Society (AES), has developed an interoperability standard called AES67 which allows existing protocols to connect to each other and interchange audio. The standard is not intended to be a solution on its own, but rather one that provides a means for exchanging audio streams between areas with different networking solutions or technologies. AES67 defines a minimum feature set: if two devices are AES67 compliant, then they can exchange audio streams with each other.

Last September, the approval of the first broadcast IP standards (SMPTE ST-2110) moved the potential of IP for broadcasting to a completely new level. Whilst audio can be streamed as part of a video signal, ST-2110 looks towards “elemental” streaming, in which audio, video and metadata is streamed separately, but over the same network and in perfect alignment. The ST-2110 standard defines the use of AES67 for audio as part of its elemental streaming. This means the audio component and its metadata are easy to access and don’t require unpacking a video stream, therefore removing the latency issues introduced by encoding and decoding.

Audio professionals, whether they are working in post-production facilities, recording studios, live productions or the broadcast field, are now conscious of the advantages of working with networked systems, and they seem to be confident that they are able to adopt flexible solutions based on any of the existing popular protocols (such as Dante and Ravenna), knowing they will be able to connect and interchange audio whenever needed thanks to the AES67 interoperability standard.

For the audio engineers out there, the emergence of these key standards will smooth the path – but it’s critical that you gain a keen understanding of AES67 and SMPTE ST-2110, as well as the nuts and bolts of configuring and managing IP switches. As if you don’t already do enough!


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