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.
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.
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.
When it comes to IP, audio has always been ahead of video. Whilst audio often makes up for it in scale, its relatively low bandwidth requirements meant computing was up to the task of audio-over-IP long before uncompressed video-over-IP. Despite the early lead, audio-over-IP isn’t necessarily trivial. However, this talk aims to give you a heads up to the main hurdles so you can address them right from the beginning.
Matt Ward, Head of Video for UK-based Jigsaw24, starts this talk revising the reasons to go audio over IP (AoIP). The benefits vary for each company. For some, reducing cabling is a benefit, many are hoping it will be cheaper, for others achievable scale is key. Matt’s quick to point out the drawbacks we should be cautious of, not least of which are complexity and skill gaps.
Matt fast-tracks us to better installations by hitting a list of easy wins some of which are basic, but a disproportionately important as the project continues i.e. naming paths and devices and having IP addresses in logical groups. Others are more nuanced like ensuring cable performance. For CAT6 cabling, it’s easy to get companies to test each of your cables to ensure the cable and all terminations are still working at peak performance.
Planning your timing system is highlighted as next on the road to success with smaller facilities more susceptible to problems if they only have one clock. But any facility has to be carefully considered and Matt points out that the Best Master Clock Algorithm (BMCA).
Network considerations are the final stop on the tour, underlining that audio doesn’t have to run in its own network as long as QoS is used to maintain performance. Matt details his reasons to keep Spanning Tree Protocol off, unless you explicitly know that you need it on. The talk finishes by discussing multicast distribution and IGMP snooping.