There are many ways of achieving a hybrid of OTT-delivered and broadcast-delivered content, but they are not necessarily interoperable. DVB aims to solve the interoperability issue, along with the problem of service discovery with DVB-I. This specification was developed to bring linear TV over the internet up to the standard of traditional broadcast in terms of both video quality and user experience.
DVB-I supports any device with a suitable internet connection and media player, including TV sets, smartphones, tablets and media streaming devices. The medium itself can still be satellite, cable or DTT, but services are encapsulated in IP. Where both broadband and broadcast connections are available, devices can present an integrated list of services and content, combining both streamed and broadcast services.
DVB-I standard relies on three components developed separately within DVB: the low latency operation, multicast streaming and advanced service discovery. In this webinar, Rufael Mekuria from Unified Streaming focuses on low latency distributed workflow for encoding and packaging.
The process starts with an ABR (adaptive bit rate) encoder responsible for producing streams with multiple bit rates and clear segmentation – this allows clients to automatically choose the best video quality depending on available bandwidth. Next step is packaging where streaming manifests are added and content encryption is applied, then data is distributed through origin servers and CDNs.
Rufael explains that low latency mode is based on an enhancement to the DVB-DASH streaming specification known as DVB Bluebook A168. This incorporates the chunked transfer encoding of the MPEG CMAF (Common Media Application Format), developed to enable co-existence between the two principle flavors of adaptive bit rate streaming: HLS and DASH. Chunked transfer encoding is a compromise between segment size and encoding efficiency (shorter segments make it harder for encoders to work efficiently). The encoder splits the segments into groups of frames none of which requires a frame from a later group to enable decoding. The DASH packager then puts each group of frames into a CMAF chunk and pushes it to the CDN. DVB claims this approach can cut end-to-end stream latency from a typical 20-30 seconds down to 3-4 seconds.
The other topics covered are: encryption (exhanging key parameters using CPIX), content insertion, metadata, supplemental descriptors, TTML subitles and MPD proxy.
Intercom systems form the backbone of any broadcast production environment. There have been great strides made in the advancement of these systems, and matrix intercoms are very mature solution now, with partylines, IFBs and groups, wide range of connectivity options and easy signal monitoring. However, they have flaws as well. Initial cost is high and there’s lack of flexibility as system size is limited by the matrix port count. It is possible to trunk multiple frames, but it is difficult, expensive and takes rack space. Moreover, everything cables back to a central matrix which might be a single point of failure.
In this presentation, Martin Dyster from The Telos Alliance looks at the parallels between the emergence of Audio over IP (AoIP) standards and the development of products in the intercom market. First a short history of Audio over IP protocols is shown, including Telos Livewire (2003), Audinate Dante (2006), Wheatstone WheatNet (2008) and ALC Networks Ravenna (2010). With all these protocols available a question of interoperability has arisen – if you try to connect equipment using two different AoIP protocols it simply won’t work.
In 2010 The Audio Engineering Society formed the x192 Working Group which was the driving force behind the AES67. This standard was ratified in 2013 and allowed interconnecting audio equipment from different vendors. In 2017 SMPTE adapted AES67 as the audio format for ST 2110 standard.
Audio over IP replaces the idea of connecting all devices “point-to-point” with multicast IP flows – all devices are connected via a common fabric and all audio routes are simply messages that go from one device to another. Martin explains how Telos were inspired by this approach to move away from the matrix based intercoms and create a distributed system, in which there is no central core and DSP processing is built in intercom panels. Each panel contains audio mix engines and a set of AES67 receivers and transmitters which use multicast IP flows. Any ST 2110-30 / AES67 compatible devices present on the network can connect with intercom panels without an external interface. Analog and other baseband audio needs to be converted to ST 2110-30 / AES67.
Martin finishes his presentation by highlighting advantages of AoIP intercom systems, including lower entry and maintenance cost, easy expansion (multi studio or even multi site) and resilient operation (no single point of failure). Moreover, adaptation of multicast IP audio flows removes the need for DAs, patch bays and centralised routers, which reduces cabling and saves rack space.
For a long time now, broadcasters have been using dark fibre and CWDM (Coarse Wavelength Division Multiplexing) for transmission of multiple SDI feeds to and from remote sites. As an analogue process, WDM is based on a concept called Frequency Division Multiplexing (FDM). The bandwidth of a fibre is divided into multiple channels and each channel occupies a part of the large frequency spectrum. Each channel operates at a different frequency and at a different optical wavelength. All these wavelengths (i.e., colours) of laser light are combined and de-combined using a passive prism and optical filters.
In this presentation Roy Folkman from Embrionix shows what advantages can be achieved by moving from CWDM technology to real-time media-over-IP system. The recent project for CPAC (Cable Public Affairs Channel) in Canada has been used as an example. The scope of this project was to replace an aging CWDM system connecting government buildings and CPAC Studios which could carry 8 SDI signals in each direction with a single dark fibre pair. The first idea was to use a newer CWDM system which would allow up to 18 SDI signals, but quite quickly it became apparent that an IP system could be implemented at similar cost.
As this was an SDI replacement, SMPTE ST 2022-6 was used in this project with a upgrade path to ST 2110 possible. Roy explains that, from CPAC point of view, using ST 2022-6 was a comfortable first step into real-time media-over-IP which allowed for cost reduction and simplification (no PTP generation and distribution required, re-use of existing SDI frame syncs and routing with audio breakaway capability). The benefits of using IP were: increased capacity, integrated routing (in-band control) and ease of future expansion.
A single 1RU 48-port switch on each side and a single dark fibre pair gave the system a capacity of 48 HD SDI signals in each direction. SFP gateways with small Embronix enclosures have been used to convert SDI outs of cameras to IP fibre – that also allowed to extend the distance between the cameras and the switch above SDI cabling limit of 100 meters. SFP gateway modules converting IP to SDI have been installed directly in the switches in both sites.
Roy finishes his presentation with possible future expansion of the system, such as migration to ST 2110 (firmware upgrade for SFP modules), increased capacity (by adding additional dark fibres ands switches), SDI and IP routing integration with unified control system (NMOS), remote camera control and addition of processing functions to SFP modules (Multiviewers, Up/Down/CrossConversion, Compression).
The SMPTE ST 2110-40 standard specifies the real-time, RTP transport of SMPTE ST 291-1 Ancillary Data packets. It allows creation of IP essence flows carrying the VANC data familiar to us from SDI (like AFD, closed captions or ad triggering), complementing the existing video and audio portions of the SMPTE ST 2110 suite.
This presentation, by Bill McLaughlin from EEG, is an updated tutorial on subtitling, closed captioning, and other ancillary data workflows using the ST 2110-40 standard. Topics include synchronization, merging of data from different sources and standards conversion.
Building on Bill’s previous presentation at the IP Showcase), this talk at NAB 2019 demonstrates a big increase in the number of vendors supporting ST 2110-40 standard. Previously a generic packet analyser like Wireshark with dissector was recommended for troubleshooting IP ancillary data. But now most leading multiviewer / analyser products can display captioning, subtitling and timecode from 2110-40 streams. At the recent “JT-NM Tested Program” event 29 products passed 2110-40 Reception Validation. Moreover, 27 products passed 2110-40 Transmitter Validation which mean that their output can be reconstructed into SDI video signals with appropriate timing and then decoded correctly.
Bill points out that ST 2110-40 is not really a new standard at this point, it only defines how to carry ancillary data from the traditional payloads over IP. Special care needs to be taken when different VANC data packets are concatenated in the IP domain. A lot of existing devices are simple ST 2110-40 receivers which would require a kind of VANC funnel to create a combined stream of all the relevant ancillary data, making sure that line numbers and packet types don’t conflict, especially when signals need to be converted back to SDI.
There is a new ST 2110-41 standard being developed for additional ancilary data which do not match up with ancillary data standardised in ST 291-1. Another idea discussed is to move away from SDI VANC data format and use a TTML track (Timed Text Markup Language – textual information associated with timing information) to carry ancillary information.