Search Results for: PTP

Video: Introduction To AES67 & SMPTE ST 2110

Posted on by Russell Trafford-Jones

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 Calverley’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: ATSC 3.0 Part II – Cutting Edge OFDM with IP

Posted on by Russell Trafford-Jones

RF, modulation, Single Frequency Networks (SFNs) – there’s a lot to love about this talk which is the second in a series of ATSC seminars however much is transferable to DVB. Today we’re focussed on transmission showing how ATSC 3.0 improves on DVB-T, how it simultaneously delivers feeds with different levels of robustness, the benefits of SFNs and much more.

In the second in this series of ATSC 3.0 talks, GatesAir’s Joe Seccia leads the proceedings starting by explaining why ATSC 3.0 didn’t simply adopt DVB-T2’s modulation scheme. The answer, explained in detail by Joe, is that by putting in further work, they got closer to the Shannon limit than DVB-T2 does. He continues to highlight the relevant standards which comprise the ATSC 3.0 standard which define the RF physical layer.

After showing how the different processes such as convolutional encoding and multiplexing fit together in the transmission chain, Joe focuses in on Layered Division Multiplexing (LDM) where a high robustness signal can be carefully combined with a lower robustness signal such that where one interferes with the other, there is enough separation to allow it to be decoded.

Next we are introduced to PLPs – Physical Layer Pipes. These can also be found in DVB-T2 and DVB-S2 and are logical channels carrying one or more services, with a modulation scheme and robustness particular to that individual pipe. Within those lie Frames and Subframes and Joe gives a good breakdown of the difference in meaning of the three, the Frame being at the top of the pile containing the other two. We look at how the bootstrap signal at a known modulation scheme and symbol rate details what’s coming next such which allow this very dynamic working with streams being sent with different modulation settings. The bootstrap is also important as it contains Early Alert System (EAS) signalling.

Layered Division Multiplexing is the next hot topic we hit and this elicits questions from the audience. LDM is important because it allows two streams to be sent at the same time with independent or related broadcasts. For instance this could deliver UHD content with HD underneath with the HD modulated to give much better robustness.

Another way of maintaining robustness is to establish an SFN which is now possible with ATSC 3.0. Joe explains how this is possible and how the RF from different antennae can help with reception. Importantly he also outlines how toward out the maximum separation between antennae and talks through different deployment techniques. He then works through some specific cases to understand the transmission power needed.

As the end of the video nears, Joe talks about MIMO transmission explaining how this, among other benefits, can allow channel bonding where two 6Mhz channels can be treated as a single 12Mhz channel. He talks about how PTP can complement GPS in maintaining timing if diverse systems are linked with ethernet and he then finishes with a walkthrough of configuring a system.

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Speakers

Joe Seccia Joe Seccia
Manager, TV Transmission Market and Product Development Strategy
GatesAir

Video: Benefits of IP Systems for Sporting Venues

Posted on by Russell Trafford-Jones

As you walk around any exhibitions there seems to be a myriad of ‘benefits’ of IP working, many of which don’t resonate for particular use cases. Only the most extraordinary businesses need all of the benefits, so in this talk, Imagine Communication’s John Mailhot discusses how IP helps sports venues.

John sets the scene by separating out the function of OB trucks and the ‘inside production’ facilities which have a whole host of non-TV production to do including driving scoreboards, displays inside the venue, replays and importantly has to deal with over 250 events a year, not all of which will have an OB truck.

We see that the scale that IP can work at is a great benefit as many signals can fit down one fibre and 2022-7 seamless switching can easily provide full redundancy for every fibre and SFP. This is a level of redundancy which is simply not seen in SDI systems. With stadia being very large, necessitating cable runs of over 500m, the fact that IP needs fewer cables overall is a great benefit.

John shows an example of an Arista switch only 7U in height which provides 144x 100G ports meaning it could support over 4000 inputs and 4000 outputs. Such density is unprecedented and for OB trucks can be a dealbreaker. For sports venues, this can also be a big motivator but also allow more flexibility in distributing the solution rather than relying on a massive central interconnect with a 1100×1100 SDI router in a central CTA.

TV is nothing without audio and the benefits to audio in 2110 are non trivial since with the audio being split off from the video, we are no longer limited to dealing with just 16 channels per video and de-embedding from a video frame any time we want to touch it.

Timing is an interesting benefit. I say this because, whilst PTP can end up being quite complex compared to black and burst, it has some big benefits. First off, it can live in the same cables as your data where as black and burst requires a whole separate cable infrastructure. PTP also allows you to timestamp all essences which helps with lip-sync throughout your workflow.

John leads us through some examples of how this works for different areas finishing by summing up the relevant benefits such as scalability, multi-format, space efficient, and timing amongst others.

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Speakers

John Mailhot John Mailhot
CTO, Networking & Infrastructure,
Imagine Communications

Video: Timing Tails & Buffers

Posted on by Russell Trafford-Jones

Timing and synchronisation have always been a fundamental aspect of TV and as we move to IP, we see that timing is just as important. Whilst there are digital workflows that don’t need to be synchronised against each other, many do such as studio productions. However, as we see in this talk from The Broadcast Bridge’s Tony Orme, IP networks make timing all the more variable and accounting for this is key to success.

To start with Tony looks at the way the OBs, also known as REMIs, are moving to IP and need a timing plane across all of the different parts of production. We see how traditionally synchronisation is needed and the effect of timing problems not only in missed data but also with all essences being sent separately synchronisation problems between them can easily creep in.

When it comes to IP timing itself, Tony explains how PTP is used to record the capture time of the media/essences and distribute through the system. Looking at the data on the wire and the interval between that and the last will show a distribution of, hopefully, a few microseconds variation. This variation gives rise to jitter which is a varying delay in data arrival. The larger the spread, the more difficult it will be to recover data. To examine this more closely, Tony looks at the reasons for and the impacts of congestion, jitter, reordering of data.

Bursting, to make one of these as an example, is a much overlooked issue on networks. While it can occur in many scenarios without any undue problems, microbusting can be a major issue and one that you need to look for to find. This surrounds the issue of how you decide that a data flow is, say, 500Mbps. If you had an encoder which sent data at 1Gbps for 5 minutes and no data for 5 minutes, then over the 10 minute window, the average bitrate would have been 500Mbps. This clearly isn’t a 500Mbps encoder, but how narrow do you need to have your measurement window to be happy it is, indeed, 500Mbps by all reasonable definitions? Do you need to measure it over 1 second, 1 millisecond? Behind microbursting is the tendency of computers to send whatever data they have as quickly as possible; if a computer has a 10Gbe NIC, then it will send at 10Gbps. What video receivers actually need is well spaced packets which always come a set time apart.

Buffers a necessary for IP transmission, in fact within a computer there are many buffers. So using and understanding buffers is very important. Tony takes us through the thought process of considering what buffers are and why we need them. With this groundwork laid, understanding their use and potential problems is easier and well illustrated in this talk. For instance, since there are buffers in many parts of the chain to send data from an application to a NIC and have it arrive at the destination, the best way to maximise the chances of having a deterministic delay in the Tx path is to insert PTP information almost at the point of egress in the NIC rather than in the application itself.

The talk concludes by looking at buffer fill models and the problems that come with streaming using TCP/IP rather then UDP/IP (or RTP). The latter being the most common.

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Speakers

Tony Orme Tony Orme
Editor,
The Broadcast Bridge