Video: Edge Compute and the CDN

Posted on by Russell Trafford-Jones

Requests to servers are returned only tens of milliseconds later which is hardly any time to wait. But they quickly add up to users waiting seconds for their player to find out what video it wants, get it and finally start showing it. We all know that time is money when it comes to people ‘pressing play’ so reducing this startup time.

Anime streaming service, CrunchyRoll went to task to reduce their startup time. Michael Dale, VP of Engineering there, sits with Heather Chamberlin Mellish from AWS to describe how they used AWS to optimise the communications needed to establish a streaming session,

Named Katana, the project looked at the 12+ requests involved between third parties and the player itself which were all needed to start the session. Advertising companies need to be consulted, streaming manifest files need to represent chunks from multiple CDNs, SSAI and metrics were done with third-party vendors and the service is protected with DRM. These are just some of the factors which led to so many return trips needing to be accomplished before shipping.

This talk provides an overview and a little bit of a ‘behind the scenes’ of a blog post which also covers this project.

Key to success was deploying on AWS Lambda@Edge which is a server which allows you to run code within AWS’s CloudFront. If you have Python or Javascript, this allows you to run it at the edge server closest to the user. For Crunchyroll’s global audience, this is particularly useful as it avoids having to set up infrastructure in every one of the AWS regions but still reduces much of the return trip time. Michael explains that, although Lambda is often viewed as an ephemeral service, when it’s not in use it can be suspended and used again in the future allowing it to maintain state for a player.

Michael explains the ways in which Katana has achieved success. Many of the third-party services have been brought into Lambda@Edge and AWS. DRM and Advertising are still third-party, but doing most things within the edge and also pre-emptively returning information such as manifests has removed many requests. The video breaks down their use of GraphQL and how Multi-CDN and SSAI workflows have been implemented.

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Speakers

Michael Dale Michael Dale
VP Engineering,
Crunchyroll
Heather Mellish Heather Chamberlin Mellish
Principal Edge Go To Market Specialist,
AWS Amazon Web Services

Video: Timing Requirements in Broadcast Applications

Posted on by Russell Trafford-Jones

How does timing for AES67 and SMPTE ST 2110-30 work? All is revealed in this short video by Andrea Hildebrand who explains why we need PTP timing and how we relate the absolute time to the signals themselves.

In a network for audio streams, Andreas starts, we want all the streams to run on their native sample rate, use the same clock, but also want to have the possibility of multiple concurrent streams using different sample rates. Also, it’s important to have a deterministic end-to-end latency and that, when streams arrive, they should be suitably aligned. We achieve all of this by distributing time around the system. Audio has very high accuracy requirements of down to within 10 microseconds for typical 48KHz broadcast signals, but AES11 requires within 1 microsecond which is why the Precision Time Protocol, PTP is used which is defined by the standard IEEE 1588. For more information on PTP, check out our PTP back library

End devices run their own local clocks, synchronised to the PTP on the network. In charge of it all, there is a grandmaster locked to GPS which can then distribute to other secondary clocks which feed the end devices. The end device can generate a media clock from the PTP and by using PTP, different facilities can be kept in time with each other. All media is then timestamped with the time when they were generated. For advice on architecting PTP, have a listen to this talk from Arista’s Gerard Phillips.

RTP is used to carry professional media streams like AES. RTP builds on top of UDP to add the critical timing information we need. Namely, the timestamp but also the sequence number. Andreas looks at the structure of the RTP packet header to see where the timestamp and identifiers go. To follow up on the IT basics underpinning AES67 and SMPTE ST 2110, check out Ed Calverley’s presentation on the topic.

‘Profiles’ are required to link the time of day to media flows – to give the time some meaning in terms of the expected signal. The AES67 Media Profile does this for AES67 as an annexe in the standard. SMPTE use ST 2059 to define how to use AES67 as well as all the other essences it supports and relate them all back to an originating epoch time in 1970.

The talk finishes by looking at the overlap in timing specs for AES67 and ST 2110-30 (AES67 for 2110). For more information on how AES67 and ST 2110 work (and don’t work) together, watch Andreas’s ‘Deeper dive’ on the topic.

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Speakers

Andreas Hildebrand Andreas Hildebrand
RAVENNA Evangelist
ALC NetworX

Video: Next-generation audio in the European market – The state of play

Posted on by Russell Trafford-Jones

Next-generation audio refers to a range of new technologies which allow for immersive audio like 3D sound, for increased accessibility, for better personalisation and anything which delivers a step-change in the lister experience. NGA technologies can stand on their own but are often part of next-generation broadcast technologies like ATSC 3.0 or UHD/8K transmissions.

This talk from the Sports Video Group and Dolby presents one case study from a few that have happened in 2020 which delivered NGA over the air to homes. First, though, Dolby’s Jason Power brings us up to date on how NGA has been deployed to date and looks at what it is.

Whilst ‘3D sound’ is an easy to understand feature, ‘increased personalisation’ is less so. Jason introduces ideas for personalisation such as choosing which team you’re interested in and getting a different crowd mix dependant on that. The possibilities are vast and we’re only just starting to experiment with what’s possible and determine what people actually want or to change where your mics are, on the pitch or in the stands.

What can I do if I want to hear next-generation audio? Jason explains that four out of five TVs are now shipping with NGA audio and all of the five top manufacturers have support for at least one NGA technology. Such technologies are Dolby’s AC-4 and sADM. AC-4 allows delivery of Dolby Atmos which is an object-based audio format which allows the receiver much more freedom to render the sound correctly based on the current speaker set up. Should you change how many speakers you have, the decoder can render the sound differently to ensure the ‘stereo’ image remains correct.

To find out more about the technologies behind NGA, take a look at this talk from the Telos Alliance.

Next, Matthieu Parmentier talks about the Roland Garros event in 2020 which was delivered using sADM plus Dolby AC-4. sADM is an open specification for metadata interchange, the aim of which is to help interoperability between vendors. The S-ADM metadata is embedded in the SDI and then transported uncompressed as SMPTE 302M.

ATEME’s Mickaël Raulet completes the picture by explaining their approach which included setting up a full end-to-end system for testing and diagnosis. The event itself, we see, had three transmission paths. An SDR satellite backup and two feeds into the DVB-T2 transmitter at the Eiffel Tower.

The session ends with an extensive Q&A session where they discuss the challenges they faced and how they overcame them as well as how their businesses are changing.

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Speakers

Jason Power Jason Power
Senior Director of Commercial Partnerships & Standards,
Dolby
Mickaël Raulet Mickaël Raulet
Vice President of Innovation,
ATEME
Matthieu Parmentier Matthieu Parmentier
Head of Data & Artificial Intelligence
France Television
Roger Charlesworth Moderator:Roger Charlesworth
Charlesworth Media

Video: JPEG XS Interoperability Activity Group Update

Posted on by Russell Trafford-Jones


JPEG XS is a low-latency, light-compression codec often called a ‘mezzanine’ codec. Encoding within milliseconds, JPEG XS can compress full-bandwidth signals by 4x or more allowing scope for several generations of compression without significant degradation. The low-latency and resilience to de-generation make it ideal for enabling remote production.

John Dale from Media Links joins us to look at what’s being done within the Video Services Forum (VSF) to ensure interoperability. As a new standard, JPEG XS is yet to be or is still being implemented in many companies’ products. Therefore this is the perfect time to be looking at how to standardise interconnects,

Running JPEG XS over MPEG TS is one approach which is being written up in ‘VSF TR-07’ (Technical Reference 7) which will be imminently completed. It defines capabilities for 2K, 4K and 8K video with and without HDR. They have split the video formats into capability sets meaning that a vendor can comply with the specification by stating which subset(s) it can cope with. All formats up to 1080p60 are under capability set ‘A’ with ‘B’ covering UHD resolutions. After this work, they will look at JPEG XS over ST 2110-22 instead of MPEG TS. This is yet to start and will share much of the work from previous work.

Watch now!
Speaker

John Dale John Dale
Company Director and CMO,
Media Links.