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: Broadcast 101 – Audio in an IP Infrastructure

Uncompressed audio has been in the IP game a lot longer than uncompressed video. Because of its long history, it’s had chance to create a fair number of formats ahead of the current standard AES67. Since many people were trying to achieve the same thing, we find that some formats are compatible with AES67 – in part, whilst we that others are not compatible.

To navigate this difficult world of compatibility, Axon CTO Peter Schut continues the Broadcast 101 webinar series with this video recorded this month.

Peter starts by explaining the different audio formats available today including Dante, RAVENNA and others and outlines the ways in which they do and don’t interoperate. After spending a couple of minutes summarising each format individually, including the two SMPTE audio formats -30 and -31, he shows a helpful table comparing the,

Timing is next on the list discussing PTP and the way that SMPTE ST 2059 is used then packet time is covered explaining how the RTP payload fits into the equation. This payload directly affects the duration of audio you can fit into a packet. The duration is important in terms of keeping a low latency and is restricted to either 1ms or 125 microseconds by SMPTE ST 2110-30.

Peter finishes up this webinar talking about some further details about the interoperability problems between the formats.

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Speaker

Peter Schut Peter Schut
CTO,
Axon

Video: Real World IP – PTP

PTP, Precision Time Protocol, underpins the recent uncompressed video and audio over IP standards. It takes over the role of facility-wide synchronisation from black and burst signals. So it’s no surprise that The Broadcast Bridge invited Meinberg to speak at their ‘Real World IP’ event exploring all aspects of video over IP.

David Boldt, head of software engineering at Meinberg, explains how you can accurately transmit time over a network. He summarises the way that PTP accounts for the time taken for messages to move from A to B. David covers different types of clock explaining the often-heard terms ‘boundary clock’ and ‘transparent clock’ exploring their pros and cons.

Unlike black and burst which is a distributed signal, PTP is a system with bi-directional communication which makes redundancy all the more critical and, in some ways, complicated. David talks about different ways to attack the main/reserve problem.

PTP is a cross-industry standard which needs to be interpreted by devices to map the PTP time into an understanding of how the signal should look in order for everything to be in time. SMPTE 2059 does this task which David cover.

PTP-over-WAN: David looks at a case study of delivering PTP over a WAN. Commonly assumed not practical by many, David shows how this ways done without using a GPS antenna at the destination. To finish off the talk, there’s a teaser of the new features coming up in the backwards-compatible PTP Version 2.1 before a Q&A.

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Speakers

Daniel Boldt

Daniel Boldt
Head of Software Engineering
Meinberg