Video: AES67 & ST 2110 Deeper Dive – The Audio Files

A deeper dive here, in the continuing series of videos looking at AES67, SMPTE ST 2110 and Ravenna. Andreas Hildebrand from ALC Networx is back to investigate the next level down on how AES67 and ST 2110 operate and how they can be configured. The talk, however, remains accessible throughout and starts with an reminder of what AES67 is and why it exists. This is was also covered in his first talk.

After explaining the AES67 was created as a way for multiple audio-over-IP standards to interoperate, Andreas looks at the stack, stepping through it to explain each element. The first topic is timing. He explains that every device on the AES67 network is not only governed by PTP, but it’s also runs its own clock which is called the Local Clock. From the Local Clock, the device then also creates a Media Clock which is based on the Local Clock time but is used to crate any frequency needed for the media (48KHz, for instance). Finally an RTP clock is kept for transmission over the network.

The next item featured on the stack is encoding. AES67 is baed on linear audio, also known as PCM. AES67 ensures that 48KHz, 16 & 24-bit audio is supported on all devices and allows up to 8 channels per stream. Importantly, Andreas explains the different versions of packet time which are supported, 1ms being mandatory which allows 48 samples of 48 KHz audio into teach IP packet.

SDP – Session Description Protocol is next which describes in a simple text file what’s in the AES67 stream giving its configuration. Then Andreas looks at what Link Offset is and examines its role in determining latency and the types of latency it’s been made to compensate for. He then talks you through working out what latency setting you need to use including taking into account the number of switches in a network and our frame size.

SMPTE ST 2110 is the focus for the last part of the talk. This, Andreas explains, is a way of moving, typically uncompressed, professional media (also known as essences) around a network for live production with very low latency. It sends audio separately to the video and uses AES67 to do so. This is defined in standard ST 2110-30. However, there are some important configurations for AES67 which are mandated in order to be compatible which Andreas explains. One example is forcing all devices to be slave only, another is setting the RTP clock offset to zero. Andreas finishes the talk by summarising what parts of ST 2110 and AES67 overlap including discussing the frame sizes supported.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX Gmbh.

Video: Introduction To AES67 & SMPTE ST 2110

While standardisation of video and audio over IP is welcome, this does leave us with a plethora of standards numbers to keep track of along with interoperability edge cases to keep track of. Audio-over-IP standard AES67 is part of the SMPTE ST-2110 standards suite and was born largely from RAVENNA which is still in use in it’s own right. It’s with this backdrop that Andreas Hildebrand from ALC NetworX who have been developing RAVENNA for 10 years now, takes the mic to explain how this all fits together. Whilst there are many technologies at play, this webinar focusses on AES67 and 2110.

Andreas explains how AES67 started out of a plan to unite the many proprietary audio-over-IP formats. For instance, synchronisation – like ST 2110 as we’ll see later – was based on PTP. Andreas gives an overview of this synchronisation and then we shows how they looked at each of the OSI layers and defined a technology that could service everyone. RTP, the Real-time Transport Protocol has been in use for a long time for transport of video and audio so made a perfect option for the transport layer. Andreas highlights the important timing information in the headers and how it can be delivered by unicast or IGMP multicast.

As for the audio, standard PCM is the audio of choice here. Andreas details the different format options available such as 24-bit with 8 channels and 48 samples per packet. By varying the format permutations, we can increase the sample rate to 96kHz or modify the number of audio tracks. To signal all of this format information, Session Description Protocol messages are sent which are small text files outlining the format of the upcoming audio. These are defined in RFC 4566. For a deeper introduction to IP basics and these topics, have a look at Ed Calverly’s talk.

The second half of the video is an introduction to ST-2110. A deeper dive can be found elsewhere on the site from Wes Simpson.
Andreas starts from the basis of ST 2022-6 showing how that was an SDI-based format where all the audio, video and metadata were combined together. ST 2110 brings the splitting of media, known as ‘essences’, which allows them to follow separate workflows without requiring lots of de-embedding and embedding processes.

Like most modern standards, ATSC 3.0 is another example, SMPTE ST 2110 is a suite of many standards documents. Andreas takes the time to explain each one and the ones currently being worked on. The first standard is ST 2110-10 which defines the use of PTP for timing and synchronisation. This uses SMPTE ST 2059 to relate PTP time to the phase of media essences.

2110-20 is up next and is the main standard that defines use of uncompressed video with headline features such as being raster/resolution agnostic, colour sampling and more. 2110-21 defines traffic shaping. Andreas takes time to explain why traffic shaping is necessary and what Narrow, Narrow-Linear, Wide mean in terms of packet timing. Finishing the video theme, 2110-22 defines the carriage of mezzanine-compressed video. Intended for compression like TICO and JPEG XS which have light, fast compression, this is the first time that compressed media has entered the 2110 suite.

2110-30 marks the beginning of the audio standards describing how AES67 can be used. As Andreas demonstrates, AES67 has some modes which are not compatible, so he spends time explaining the constraints and how to implement this. For more detail on this topic, check out his previous talk on the matter. 2110-31 introduces AES3 audio which, like in SDI, provides both the ability to have PCM audio, but also non-PCM audio like Dolby E and D.

Finishing up the talk, we hear about 2110-40 which governs transport of ancillary metadata and a look to the standards still being written, 2110-23 Single Video essence over multiple 2110-20 streams, 2110-24 for transport of SD signals and 2110-41 Transport of extensible, dynamic metadata.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX Gmbh.

Webinar: RAVENNA and its Relationship to AES67 and SMPTE ST 2110


This webinar is now available on-demand

This first in a series of webinars, this will have a broad scope covering the history of audio networking in, the development of RAVENNA the the consequent developments of AES67 and ST2110. Whether you’re new to or already familiar with RAVENNA and/or AES67 & ST2110 you’ll benefit from this webinar either as revision or as an excellent starting point for understanding the landscape of Audio-over-IP standards and technologies.

This webinar is presented Andreas Hildebrand who has previously appeared on The Broadcast Knowledge giving insight into The Audio Parts of ST 2110, ST 2110-30 and NMOS IS-08 — Audio Transport and Routing amongst others.

This talk looks at how audio IP works and the benefits of using an IP system. Since the invention of RAVENNA, the AES and SMPTE have moved to using audio over IP so the walk will examine how RAVENNA and SMPTE place the audio data on the network and get that to the decoders. Interoperability between systems is important but can only happen if certain parameters are correct, something that Andreas will mention but will also be a subject for future webinars.

Whether you want to revise the basics from an expert or learn them for the first time, now’s the time to register.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX GmbH

Video: ST 2110-30 and NMOS IS-08 — Audio Transport and Routing

Andreas Hildebrand starts by introducing 2110 and how it works in terms of sending the essences separately using multicast IP. This talk focusses on the ability of audio-only devices to subscribe to the audio streams without needing the video streams. Andreas then goes on to introduce AES67 which is a standard defining interoperability for audio defining timing, session description, encoding, QOS, transport and much more. Of all the things which are defined in AES67, discovery was deliberately not included and Andreas explains why.

Within SMPTE 2110, there are constraints added to AES67 under the sub-standard 2110-30. The different categories A, B and C (and their X counterparts) are explained in terms how how many audios are defined and the sample lengths with their implications detailed.

As for discovery and other aspects of creating a working system, Andreas looks towards AMWA’s NMOS suite summarising the specifications for Discovery & Registration, Connection Management, Network Control, Event & Tally, Audio Channel Mapping. It’s the latter which is the focus of the last part of this talk.

IS-08 defines a way of defining input and output blocks allowing a channel mapping to be defined. Using IS-05, we can determine which source stream should connect to which destination device. Then IS-08 gives the capability to determine which of the audios within this stream can be mapped to the output(s) of the receiving device and on top of this allows mapping from multiple received streams into the output(s) of one device. The talk then finishes with a deeper look at this process including where example code can be found.

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Speaker

Andreas Hildebrand Andreas Hildebrand
Senior Product Manager,
ALC NetworX