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.

Video: The Basics of SMPTE ST 2110 in 60 Minutes

SMPTE ST 2110 is a growing suite of standards detailing uncompressed media transport over networks. Now at 8 documents, it’s far more than just ‘video over IP’. This talk looks at the new ways that video can be transported, dealing with PTP timing, creating ‘SDPs’ and is a thorough look at all the documents.

Building on this talk from Ed Calverly which explains how we can use networks to carry uncompressed video, Wes Simpson goes through all the parts of the ST 2110 suite explaining how they work and interoperate as part of the IP Showcase at NAB 2019.

Wes starts by highlighting the new parts of 2110, namely the overview document which gives a high level overview of all the standard docs, the addition of compressed bit-rate video carriage and the recommended practice document for splitting a single video and sending it over multiple links; both of which are detailed later in the talk.

SMPTE ST 2110 is fundamentally different, as highlighted next, in that it splits up all the separate parts of the signal (i.e. video, audio and metadata) so they can be transferred and processed separately. This is a great advantage in terms of reading metadata without having to ingest large amounts of video meaning that the networking and processing requirements are much lighter than they would otherwise be. However, when essences are separated, putting them back together without any synchronisation issues is tricky.

ST 2110-10 deals with timing and knowing which packets of one essence are associated with packets of another essence at any particular point in time. It does this with PTP, which is detailed in IEEE 1588 and also in SMPTE ST 2059-2. Two standards are needed to make this work because the IEEE defined how to derive and carry timing over the network, SMPTE then detailed how to match the PTP times to phases of media. Wes highlights that care needs to be used when using PTP and AES67 as the audio standard requires specific timing parameters.

The next section moves into the video portion of 2110 dealing with video encapsulation on the networks pixel grouping and the headers needed for the packets. Wes then spends some time walking us through calculating the bitrate of a stream. Whilst for most people using a look-up table of standard formats would suffice, understanding how to calculate the throughput helps develop a very good understanding of the way 2110 is carried on the wire as you have to take note not only of the video itself (4:2:2 10 bit, for instance) but also the pixel groupings, UDP, RTP and IP headers.

Timing of packets on the wire isn’t anything new as it is also important for compressed applications, but it is of similar importance to ensure that packets are sent properly paced on wire. This is to say that if you need to send 10 packets, you send them one at a time with equal time between them, not all at once right next to each other. Such ‘micro bursting’ can cause problems not only for the receiver which then needs to use more buffers, but also when mixed with other streams on the network it can affect the efficiency of the routers and switches leading to jitter and possibly dropped packets. 2110-21 sets standards to govern the timing of network pacing for all of the 2110 suite.

Referring back to his warning earlier regarding timing and AES67, Wes now goes into detail on the 2110-30 standard which describes the use of audio for these uncompressed workflows. He explains how the sample rates and packet times relate to the ability to carry multiple audios with some configurations allowing 64 audios in one stream rather than the typical 8.

‘Essences’, rather than media, is a word often heard when talking about 2110. This is an acknowledgement that metadata is just as important as the media described in 2110. It’s sent separately as described by 2110-40. Wes explains the way captions/subtitles, ad triggers, timecode and more can be encapsulated in the stream as ancillary ‘ANC’ packets.

2110-22 is an exciting new addition as this enables the use of compressed video such as VC-2 and JPEG-XS which are ultra low latency codecs allowing the video stream to be reduced by half, a quarter or more. As described in this talk the ability to create workflows on a single IP infrastructure seamlessly moving into and out of compressed video is allowing remote production across countries allowing for equipment to be centralised with people and control surfaces elsewhere.

Noted as ‘forthcoming’ by Wes, but having since been published, is RP 2110-23 which adds back in a feature that was lost when migrating from 2022-6 into 2110 – the ability to send a UHD feed as 4x HD feeds. This can be useful to allow for UHD to be used as a production format but for multiviewers to only need to work in HD mode for monitoring. Wes explains the different modes available. The talk finishes by looking at RTP timestamps and SDPs.

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The slides for this talk are available here
Speakers

Wes Simpson Wes Simpson
President,
Telecom Product Consulting

Video: SMPTE Technical Primers

The Broadcast Knowledge exists to help individuals up-skill whatever your starting point. Videos like this are far too rare giving an introduction to a large number of topics. For those starting out or who need to revise a topic, this really hits the mark particularly as there are many new topics.

John Mailhot takes the lead on SMPTE 2110 explaining that it’s built on separate media (essence) flows. He covers how synchronisation is maintained and also gives an overview of the many parts of the SMPTE ST 2110 suite. He talks in more detail about the audio and metadata parts of the standard suite.

Eric Gsell discusses digital archiving and the considerations which come with deciding what formats to use. He explains colour space, the CIE model and the colour spaces we use such as 709, 2100 and P3 before turning to file formats. With the advent of HDR video and displays which can show bright video, Eric takes some time to explain why this could represent a problem for visual health as we don’t fully understand how the displays and the eye interact with this type of material. He finishes off by explaining the different ways of measuring the light output of displays and their standardisation.

Yvonne Thomas talks about the cloud starting by explaining the different between platform as a service (PaaS), infrastructure as a service (IaaS) and similar cloud terms. As cloud migrations are forecast to grow significantly, Yvonne looks at the drivers behind this and the benefits that it can bring when used in the right way. Using the cloud, Yvonne shows, can be an opportunity for improving workflows and adding more feedback and iterative refinement into your products and infrastructure.

Looking at video deployments in the cloud, Yvonne introduces video codecs AV1 and VVC both, in their own way, successors to HEVC/h.265 as well as the two transport protocols SRT and RIST which exist to reliably send video with low latency over lossy networks such as the internet. To learn more about these protocols, check out this popular talk on RIST by Merrick Ackermans and this SRT Overview.

Rounding off the primer is Linda Gedemer from Source Sound VR who introduces immersive audio, measuring sound output (SPL) from speakers and looking at the interesting problem of forward speakers in cinemas. The have long been behind the screen which has meant the screens have to be perforated to let the sound through which interferes with the sound itself. Now that cinema screens are changing to be solid screens, not completely dissimilar to large outdoor video displays, the speakers are having to move but now with them out of the line of sight, how can we keep the sound in the right place for the audience?

This video is a great summary of many of the key challenges in the industry and works well for beginners and those who just need to keep up.

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Speakers

John Mailhot John Mailhot
Systems Architect for IP Convergence,
Imagine Communications
Eric Gsell Eric Gsell
Staff Engineer,
Dolby Laboratories
Linda Gedemer, PhD Linda Gedemer, PhD
Technical Director, VR Audio Evangelist
Source Sound VR
Yvonne Thomas Yvonne Thomas
Strategic Technologist
Digital TV Group

Video: JPEG XS in Action for IP Production

JPEG XS is a new intra-frame compression standard delivering JPEG 2000 quality with 1000x lower latency – microseconds instead of milliseconds. This codec provides relatively low bandwidth (visually lossless compression at ratio of 10:1) with very-low and fixed latency, which makes it ideal for remote production of live events.

In this video Andy Rayner from Nevion shows how JPEG XS fits in all-IP broadcast technology with SMPTE ST 2110-22 standard. Then he presents the world’s first full JPEG-XS deployment for live IP production created for a large sports broadcaster. It was designed for pan-European WAN operation and based on ST 2110 standard with ST 2022-7 protection.

Andy discusses challenges of IP to IP processing (ST 2110-20 to ST 2110-22 conversion) and shows how to keep video and audio in sync through the whole processing chain.

This presentation proves that JPEG-XS is working, low latency distributed production is possible and the value of the ST2110-22 addition to the 2110 suite.

You can see the slides here.

Watch now!

Speaker

Andy Rayner Andy Rayner
Chief Technologist
Nevion Ltd.