As we saw yesterday, there’s an increasingly buoyant market for video codecs and whilst this is a breath of fresh air after AVC’s multi-decade dominance, we will likely never again see a market which isn’t fragmented with several dominant players, say AV1, AVC, VVC and VP9, each sharing 85% market share relatively equally and then ‘the rest’ bringing up the rear. So multi-codec distribution to home viewers is going to have to deal with delivering different codecs to different people.
fuboTV do this today and Nick Krzemienski is here to tell us how. Starting with an overview of fuboTV primarily streams both live and on VOD. Nick shows us the workflow they use and then explains how their AVC & HEVC combined workflow is set up. Starting with the ideal case where a single fmp4 is encoded into both AVD and HEVC, he proposes you would simply package both into an HLS and DASH manifest and let players work out the rest. Depending on your players, you may have to split out your manifests into single-codec files.
DRM’s very important for a sports broadcaster so Nick looks at how this might be achieved. CMAF allows you to deliver m3u8 and mpd files using CENC (Common ENCryption). This promises a single DRM process ahead of packaging, but the reality, we hear from Nick, is that you’ll need two sets of media for HLS and DASH if you’re going to use CENC.
When you’re delivering multiple manifest and, hence, multiple sources, how do you manage this? Nick outlines, and shows the code, of how he achieves this at the edge. Using Lamda, he’s able to look at the incoming requests and existing files at the CDN to deliver the right asset with the logic done close to the viewer. Nick closes by with his thoughts on the future for streaming and answering questions from the audience.
These next 12 months are going to see 3 new MPEG standards being released. What does this mean for the industry? How useful will they be and when can we start using them? MPEG’s coming to the market with a range of commercial models to show it’s learning from the mistakes of the past so it should be interesting to see the adoption levels in the year after their release. This is part of the second session of the Vienna Video Tech Meetup and delves into startup time for streaming services.
In the first talk, Dr. Christian Feldmann explains the current codec landscape highlighting the ubiquitous AVC (H.264), UHD’s friend, HEVC (H.265), and the newer VP9 & AV1. The latter two differentiate themselves by being free to used and are open, particularly AV1. Whilst slow, both the latter are seeing increasing adoption in streaming, but no one’s suggesting that AVC isn’t still the go-to codec for most online streaming.
Christian then introduces the three new codecs, EVC (Essential Video Coding), LCEVC (Low-Complexity Enhancement Video Coding) and VVC (Versatile Video Coding) all of which have different aims. We start by looking at EVC whose aim is too replicate the encoding efficiency of HEVC, but importantly to produce a royalty-free baseline profile as well as a main profile which improves efficiency further but with royalties. This will be the first time that you’ve been able to use an MPEG codec in this way to eliminate your liability for royalty payments. There is further protection in that if any of the tools is found to have patent problems, it can be individually turned off, the idea being that companies can have more confidence in deploying the new technology.
The next codec in the spotlight is LCEVC which uses an enhancement technique to encode video. The aim of this codec is to enable lower-end hardware to access high resolutions and/or lower bitrates. This can be useful in set-top boxes and for online streaming, but also for non-broadcast applications like small embedded recorders. It can achieve a light improvement in compression over HEVC, but it’s well known that HEVC is very computationally heavy.
LCEVC reduces computational needs by only encoding a lower resolution version (say, SD) of the video in a codec of your choice, whether that be AVC, HEVC or otherwise. The decoder will then decode this and upscale the video back to the original resolution, HD in this example. This would look soft, normally, but LCEVC also sends enhancement data to add back in the edges and detail that would have otherwise been lost. This can be done in CPU whilst the other decoding could be done by the dedicated AVC/HEVC hardware and naturally encoding/decoding a quarter-resolution image is much easier than the full resolution.
Lastly, VVC goes under the spotlight. This is the direct successor to HEVC and is also known as H.266. VVC naturally has the aim of improving compression over HEVC by the traditional 50% target but also has important optimisations for more types of content such as 360 degree video and screen content such as video games.
To finish this first Vienna Video Tech Meetup, Christoph Prager lays out the reasons he thinks that everyone involved in online streaming should obsess about Video Startup Time. After defining that he means the time between pressing play and seeing the first frame of video. The longer that delay, the assumption is that the longer the wait, the more users won’t bother watching. To understand what video streaming should be like, he examines Spotify’s example who have always had the goal of bringing the audio start time down to 200ms. Christophe points to this podcast for more details on what Spotify has done to optimise this metric which includes activating GUI elements before, strictly speaking, they can do anything because the audio still hasn’t loaded. This, however, has an impact of immediacy with perception being half the battle.
“for every additional second of startup delay, an additional 5.8% of your viewership leaves”
Christophe also draws on Akamai’s 2012 white paper which, among other things, investigated how startup time puts viewers off. Christophe also cites research from Snap who found that within 2 seconds, the entirety of the audience for that video would have gone. Snap, of course, to specialise in very short videos, but taken with the right caveats, this could indicate that Akamai’s numbers, if the research was repeated today, may be higher for 2020. Christophe finishes up by looking at the individual components which go towards adding latency to the user experience: Player startup time, DRM load time, Ad load time, Ad tag load time.
In a country where the weather can be life threatenin and where earthquakes and wild fires pose a real threat to life, an Early Alert System (EAS) is very important. This talk looks at the ‘Advanced Emergency Alerting’ system (AEA) that is available in ATSC 3.0 and the coalition behind it. It also talks about some of the interactive features possible.
Richard Chernock is back to dig deeper in to the set of standards which is known as ATSC 3.0. He starts by looking at the broadcaster’s role in being a public information provider both to first responders and to the public at large. ATSC 3.0 was seen as an opportunity to go much further than EAS available in ATSC 1.0. One improvement, as covered previously, allows for very robust transmission methods. AEA also provides rich media, version information and expiry information. Additionally it can be delivered to targeted areas.
The AWARN (Advance Warning and Response Network) is a project to look world-wide at the different EAS activities ongoing in order to bring learning into ATSC and represents both broadcasters and national agencies such as FEMA and homeland security. It provides practical advice on resilience (backup generator provision), how to maximise the verboseness of information, encryption and much more.
Finishing off this short talk, Richard highlights the OTT-style interactive services possible with ATSC 3.0. He shows a quiz format where the graphics are within the control of the broadcaster. Other examples discussed are interactive access to sports replays, purchasing merchandise, the ability to synchronise with a second screen and advert displays. Watch now! Please note this is a 30 minute video but the version on YouTube repeats hence lasting 1.5 hours Speakers
Whilst there are plenty of videos explaining the basics streaming, few of them talk you through the basics of actually implementing a video player on your website. The principles taught in this hands-on Bitmovin webinar are transferable to many players, but importantly at the end of this talk you’ll have your own implementation of a video player which you can make in real time using their remix project at glitch.com which allows you to edit code and run it immediately in the browser to see your changes.
Ahead of the tutorial, the talk both explains the basics of compression and OTT led by Kieran Farr, Bitmovin’s VP of marketing and Andrea Fassina, Developer Evangelist. Andrea outlines a simplified OTT architecture where he looks at the ‘ingest’ stage which, in this example, is getting the videos from Instagram either via the API or manually. It then looks at the encoding step which compresses the input further and creates a range of different bitrates. Andrea explains that MPEG standards such as H.264, H.265 are commonly used to do this making the point that MPEG standards typically require royalty payments. This year, we are expecting to see VVC released by MPEG (H.266).
Andrea then explains the relationship between resolution, frame rate and file sizes. Clearly smaller files are better as they require less time to download leading to quicker downloads so faster startup times. Andrea discusses how the resolutions match the display resolutions with TVs having 1920×1080 resolution or 2160×3840 resolution. Given that higher resolutions have more picture detail, there is more information to be sent leading to larger file sizes.
Source: Bitmovin https://bit.ly/2VwStwC
When you come to set up your transcoder and player, there are a number of options you need to set. These are determined by these basics, so before launching into the code, Andrea looks further into the fundamental concepts. He next looks at video compression to explain the ways in which compression is achieved and the compromises within. Andrea starts from the first MJPEG codecs where each frame was its own JPEG image and they simply animated from one JPEG to another to show the video – not unlike animated GIFs used on the internet. However by treating each frame on its own ignores a lot of compression opportunity. When looking at one frame to the next, there are a lot of parts of the image which are the same or very similar. This allowed MPEG to step up their efforts and look across a number of frames to spot the similarities. This is typically referred to as temporal compression as is it uses time as part of the process.
In order to achieve this, MPEG splits all frames into blocks, squares in AVC, which are called macro blocks which be compared between frames. They then have 3 types of frame called ‘I’, ‘P’ and ‘B’ frames. The I frames have a complete description of that frame, similar to a JPEG photograph. P frames don’t have a complete description of the frame, rather they some blocks which have new information and some information saying that ‘this block is the same as this block in this other frame. B frames have no complete new image parts, but create the frame purely out of frames from the recent future and recent past; the B stands for ‘bi-directional’.
Ahead of launching into the code, we then look at the different video codecs available. He talks about AVC (discussed in detail here), HEVC (detailed in this talk) and compares the two. One difference is HEVC uses much more flexible macro block sizes. Whilst this increases computational complexity, it reduces the need to send redundant information so is an important part of the achieving the 50% bitrate reduction that HEVC typically shows over AVC. VP9 and AV1 complete the line-up as Andrea gives an overview of which platforms can support these different codecs.
Source: Bitmovin https://bit.ly/2VwStwC
Andrea then introduces the topic of Adaptive bitrate, ABR. This is vital in the effective delivery of video to the home or mobile phones where bandwidth varies over time. It requires creating several different renditions of your content at various bitrates, resolutions and even frame rate. Whilst these multiple encodes put a computational burden on the transcode stage, it’s not acceptable to allow a viewer’s player to go black, so it’s important to keep the low bitrate version. However there is a lot of work which can go into optimising the number and range of bitrates you choose.
Lastly we look at container formats such as MP4 used in both HLS and MPEG-DASH and is based on the file format ISO BMFF. Streaming MP4 is usually called fragmented MP4 (fMP4) as it is split up into chunks. Similarly MPEG2 Transport Streams (TS files) can be used as a wrapper around video and audio codecs. Andrea explains how the TS file is built up and the video, audio and other data such as captions are multiplexed together.
The last half of the video is the hands-on section during which Andrea talks us through how to implement a video player in realtime on the glitch project allowing you to follow along and do the same edits, seeing the results in your browser as you go. He explains how to create a list of source files, get the player working and styled correctly.
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