Mobile Moves Ahead

What exactly does 5G mean? And what are the implications of this new mobile technology for media businesses?


[Reprinted from the September/October 2019 issue of The Financial Manager]

People like labels. Companies like labels. Industries like labels. In today’s complex world, they simplify things, and allow us to sort them into more easily understood categories. 

So how much simpler could the label “5G” be? One number and one letter – a thing of beauty. But as Einstein famously said, “Everything should be made as simple as possible, but no simpler.” And that’s where the problems with understanding 5G begin.

The label is applied to a very wide range of technologies, and it is understood to mean different things by different people, companies and industries. That kind of confusion certainly has occurred for other forms of emerging technology. But in the case of 5G, the term has been leveraged by marketers for their particular purposes, and already co-opted into substantial hype and resulting misconceptions.

The value of the label 5G has been thereby diluted, and its value diminished rather than increased in helping us understand what the technology is, and what it means to the businesses that may use or be affected by it in the near future.


The origin of the term comes from the world of wireless technology standards, where it implies (as you might guess) the fifth generation of cellular phone technology. But as in the passage of previous generations of technologies, the boundaries between them are not hard and fast. (See “The Wireless Solution” below)

5G will replace 4G in two distinct phases. First, the introduction of 5G will prompt the rollout of new mobile software, because the new technology allows somewhat more efficient use of existing wireless bandwidth than 4G does. This is an incremental change, not unlike previous transitions between wireless technology generations.

This transition has already begun in some markets, and will continue over the next few years, as new mobile devices that support the technology become available.

But the second phase of 5G will truly open up new horizons, as it will allow far faster connections. This is mostly due to its use of wholly new, higher frequency operations – so-called “millimeter wave” (mmWave) bands – which allow wider bandwidth connections per user. Don’t expect to see this broadly deployed until the mid2020s, however. And even then, it will only be found in densely populated urban areas. Even 5G’s strongest champions agree that it’s unlikely mmWave 5G will ever be seen in rural areas.

Until now (i.e., through the 4G era), wireless phones have operated in various bands between 600 MHz and 6 GHz regions, but with the full deployment of 5G, frequency bands above 24 GHz (mmWave) will be used. That will allow users to connect to the network on much wider-bandwidth channels. As a result, full 5G deployment will provide consumer devices with connectivity up to 20 times faster than with 4G, while also increasing the capacity of simultaneous users per cell.

If this seems too good to be true, bear in mind that these higher transmission frequencies will necessarily limit the coverage zones for individual cells to much smaller areas. They may be as small as a few tens of meters in diameter per cell.

This means that the 5G mmWave deployments will require vast numbers of transmitters and antennas spaced closely together for the system to work practically for customers, which is why the full impact of 5G will only be felt in urban areas.

Notwithstanding the geographically limited nature of mmWave deployments, though, other benefits promised by 5G generally include: 

  • Greater robustness, such as fewer dropped calls; 
  • Low latency, for faster data transfer; 
  • More sophisticated antenna design to allow more users to connect at higher speeds in a given area without interference; 
  • Network slicing, which allows multiple forms of usage to share the network simultaneously, such as smartphones and Internet-of-things (IoT) devices; 
  • Edge computing/virtualization, which puts cloud computing servers physically closer to customers, further reducing latency and reducing network congestion.


Another 5G feature, which worries some in the traditional media businesses, is its so-called “broadcast mode.” This allows a one to-many delivery format akin to broadcast radio or television service.

Such capability already exists in the current 4G LTE system, but 5G further expands it. Yet even 5G’s broadcast mode is not infinitely scalable like traditional broadcasting has always been. Even if it was, the economics of the system do not appear likely to entice mobile network operators into entering the radio or television business.

However, one media sector that does seem particularly vulnerable to new competition from 5G is the multichannel video program distributor (MVPD) business. Some wireless operators are considering the use of 5G as a “last meter” (or “last tens of-meters”) delivery method for IP-based service bundles to the customer without stringing cable into the home.

5G’s wideband, wireless connection from the antenna on the street pole to the homes on the block (or to the multiple units of an apartment building) are an appealing and potentially cost-effective alternative to traditional hard-wired fiber or coax – or even small-dish satellite – delivery of television and/or broadband Internet service to residential customers. So some 5G effects may be felt by traditional fixed (i.e., wired) rather than mobile operations. Or, put another way, 5G offers wireless operators new ways to compete with traditional wired telecom businesses.

Nevertheless, the net result of 5G’s impact on existing media businesses may be ultimately beneficial, as traditional operators find ways to use 5G services to improve operations. For example, content creators will benefit from enhanced connectivity for live backhaul of content from remote sites, while weather forecasters and other data-collectors can deploy massive numbers of sensors using IoT devices for accelerating their services.

Crowdsourcing of content or audience interaction also could be boosted by the increased capacity and speed of 5G. Yet another possibility is convergence with the similarly IP-based ATSC 3.0 system soon expected to be deployed by television broadcasters, by which broadcast transmission and wireless broadband service can be used together and simultaneously to provide rich and responsive media experiences to tomorrow’s audiences.

Yes, there’s a lot to like – and some to worry – about 5G among media businesses. The best advice now is to avoid the hype and continue to study the real prospects for the technology, to learn how it can best work with, or against, your existing services.

Most of us have already experienced transitions between generations of modern mobile telephony. It seems almost miraculous how the world of cellular phones has evolved so quickly over the mere four decades in which it has existed. But there’s no magic to the evolution – just good standards and practices by the industry.There are international standards bodies that continually work to allow wireless services to develop at a relatively fast pace. Among them is the International Telecommunications Union, a part of the United Nations that manages and resolves the use of different telecommunications methods used in different countries and regions. Another is the 3rd Generation Partnership Project (3GPP), which develops international standards for interoperable mobile broadband software and hardware.3GPP has developed an approach called long-term evolution, which expects new releases of backward-compatible mobile telephony standards to be issued on a regular basis. This means that wireless network operators can support multiple generations of consumer equipment simultaneously over periods of time.For example, when the industry moved from 3G to 4G, there was no mystical night during which everyone’s phones upgraded. People’s older phones continued to work like they used to, even as new technology was deployed by carriers. At the same time, customers who purchased new phones could enjoy the benefits of the latest services. Wireless operators maintain and continue to support the legacy generation of technology for a long period of time after they begin to deploy the next generation of technology. This also means that wireless operators require adequate spectrum to operate multiple, parallel systems during any transition between technology generations.And of course, the Internet – to which wireless phones allow mobile connectivity – also evolves in an elegant fashion, as managed by the standards of the Internet Engineering Task Force and the World Wide Web Consortium. Ideally, customers never notice these evolutions, since their old phones continue to work until they replace them with new phones, which then work better. What customers don’t recognize is that their new phones may be connecting to a different network than their old phones did, even though they are still using the same wireless provider’s services. This quiet evolution is designed to theoretically continue indefinitely.

Skip Pizzi is vice president of technology education and outreach at the National Association of Broadcasters. He can be reached at

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Quality Control & Monitoring in OTT Workflow


OTT and video streaming are here to stay. Millennials are increasingly watching content on their mobile devices and computers and a 71% growth in viewership has been observed since 2012. However, watching video on mobile devices is not limited to the youth. In the US, 86% of smartphone users watch video content on their phones. [1]  Broadcasters need to brace up for the changing viewing styles by embracing Over-The-Top (OTT) workflows, which threatens to take the center stage if the viewing patterns are any indication. Ensuring right delivery of technically sound content is critical for every broadcaster. The right set of Quality Control tools are a must to ensure that you stay ahead in the OTT race.

The Changing Face of Media Delivery

The traditional broadcast delivery is a linear flow with content being pushed downstream to set top boxes on the consumer side. The channel of delivery may be cable or satellite, DTH or IP network. The delivery format is singular and resolutions are SD, HD or UHD. Users view the content on their TV sets. But this mode of programming has been rapidly changing. Viewers are increasingly shifting to watch content when they want, where they want and on the device they want. Content needs to be streamed as per user requirements, on demand and as per the resolution of the playing device. The content is now being pulled by the consumers as per their needs. Broadcasters need to ready their content for this mode of playback – they don’t control the delivery, consumer does. Welcome OTT!

What is OTT?

OTT uses the Internet to bring audio-video content to the consumer. As opposed to traditional video distribution methods, which operate under a dedicated and controlled network, OTT video uses the Internet, which is an unmanaged network, used across the globe by millions of people.  OTT content from broadcasters and video service providers typically include streaming of content such as TV programs, movies, live sports, and other special events. YouTube videos are also a prime example of OTT video. Other OTT providers include Amazon, Netflix, Hulu, etc.

To enable OTT deliveries, broadcasters need to embrace multiple technologies, more complex than the traditional linear flow – a delivery where the content is repurposed based on the user device, the quality/bandwidth of the delivery is changed based on the network congestion, and the content is not broadcasted to multitudes, but pulled by individual consumers. This is achieved using adaptive bit-rate (ABR) technology.

What is ABR?

With ABR, multiple versions of a video are created – each version encoded at a different bit rate and profile. Each of these versions is further broken into short-duration segment, which is aligned with the same segment in other versions.  Depending on the network bandwidth available on the consumer device, an appropriate segment from a specific file is sent to the user. This assures that the user receives the best quality video in an uninterrupted manner.

Different wrapper formats have emerged for ABR technology, the most popular ones being HLS, DASH and HSS.  Different devices consume different streaming formats, for example, Android and iOS devices consume HLS (now a universal streaming format), Microsoft XBOX, Windows8 phones can consume HLS / HSS and so on. For a broadcaster, embracing these multiple OTT formats over and above the linear flow has suddenly made their life  a lot more complex. And amid this complexity, one is still struggling to cost effectively provide streaming option in addition to traditional deliveries.

OTT workflow – Video-on-Demand (VOD) vs. Live Streams

For the purpose of this document, we would like to distinguish OTT workflows based on whether we are talking about stored program content or live programming. Stored programs are managed using file-based workflows typically as VOD assets. Here, the broadcaster has the time and luxury to ensure quality of assets during the content preparation stage using file-based QC tools.  BATON® ABR, as discussed later, serves the need very well at this stage. However, in the case of live-streams, content is transcoded in real-time into the chosen ABR formats and made available for streaming. In both the cases, any delivery issues with real-time streaming are verified using real-time OTT content monitoring tools like ORION™-OTT, discussed later in this document. The next sections discuss the QC and monitoring needs for both the workflows in further details.

QC & MONITORING for VOD assets in OTT workflow

To be able to effectively monetize OTT for VOD assets, the media companies need a unified QC & monitoring solution, as shown in Fig. 1, for content preparation as well as content distribution to ensure good experience for viewers in the OTT/ABR world.

        Figure 1. Typical Broadcast Workflow Enhanced with the OTT Delivery Flows

Next, let us look at the QC needs during the content preparation stage, and the monitoring during the content distribution stage.


You need to ensure that the quality of original content is good. At the content preparation stage, file-based QC solutions like Interra Systems’ BATON®  helps you to address quality challenges, quite comfortably. From ingest to editing issues, compression artifacts introduced during transcoding, as well as file assembly issues – most are easily identified by the modern QC tools.

You must deploy the right QC tools that match your quality needs for the content preparation stage to mitigate your risks and ensure that technically sound content is ready for delivery. 

The QC checks can be broadly classified as follows:

Baseband Quality Checks

It is important to ensure that the content is checked on various quality parameters before the final delivery. A comprehensive QC tool needs to be used for a wide range of baseband quality checks, such as video signal levels, color bleeding, blotches, blur, defective pixels, black frames, color bars, RGB color gamut, mosquito noise, audio levels, audio noise and a host of other such artifacts. Good QC tool must ensure that these are detected with high level of accuracy and reliability and minimal false positives. 

Chroma Phase Error 

Compression Artifacts

When the content is compressed, several compression artifacts like blockiness, pixelation, Moire pattern,   ringing artifacts, and more can get introduced in the lossy compressed video. A good QC tool needs to ensure that the transcoded content is free from these artifacts.

File Integrity and Standards Compliance Checks

The file integrity and compliance checks ensure that the file or content being delivered is not corrupt and has been encoded as per the standard to ensure that the downstream tools are able to play it without issues. This becomes even more important in the OTT context, where there are a host of devices with different form factors and players from multitude of vendors – and the content is expected to play well on all of those devices. 

Multi-Segment ABR Transcoding

Once master/mezzanine content has been verified using a file-based QC solution, it can be subsequently submitted for ABR transcoding. ABR transcoding is a complex process involving creation of multiple renditions of the same content at different quality levels/bitrates. The transcoding process is not only time consuming but also needs to ensure proper alignment between different variants and rightful segmentation of each variant. Failure to achieve this would result in playback issues leading to revenue loss. That’s where different file-based QC & monitoring and systems come into the picture. These tools can check for ABR specific issues and alarm user before ABR package goes out for delivery.

OTT delivery, as discussed earlier, deploys the ABR technology. ABR requires content to be split into short segments of typically 10 seconds each. This ensures seamless and fast switching between different variants. Typically each ABR package is encoded at multiple bitrates (typically three or more). When the content is played, the streamed content switches between different bitrates as it moves to the next chunk, managed using the manifest file and depending on the network congestion and other factors influencing delivery quality.

When content is transcoded for ABR playback, several additional checks need to be done on the transcoded content to ensure that the content is ABR ready. Some of those checks are listed below:

Checks to ensure each segment starts with an independent frame. This is to ensure that any chunk does not have any decoding dependency on the previous one, so that during playback, a seamless switch can happen when moving from one chunk to another.

Checks to ensure that all variants of the content are properly aligned in terms of number of segments, segment duration, total duration and content structure. A client can choose to playback a particular variant depending on the download bandwidth available and device screen size, therefore, it is imperative to ensure that all the variants are consistent with each other and allow seamless switching across all of the available variants.
Ensure consistency between metadata and actual content properties.
A client uses the metadata in the manifest files to choose the best playback quality. If there is any inconsistency between the metadata and the actual media properties, it may lead to playback issues and hence a bad user experience.

Once the content is validated on these checks, it is ready for delivery, both for linear as well as OTT flows. The content is encrypted with one or more DRM technologies before it is moved to the origin server for OTT delivery.

A good file-based QC solution should have capability to perform all of the above ABR checks and also do a deep analysis to identify any baseband issue. Once the content moves to the distribution stage, the focus shifts to ensuring a smooth content delivery and the best possible user experience.  This creates a need for state-of-the-art monitoring solutions that can ensure a superior QoS as well as baseline QoE. The next section talks about the monitoring requirements in detail. 


At this stage, we need to ensure that no issues will be encountered during delivery of VOD content – in short ensure QoS as well as QoE. The monitoring requirement at this stage is to perform real-time streaming validations. There is some overlap with the File-QC done during the content preparation stage, and that is necessary to ensure the content sanity as it is replicated from the origin server to cache/replica servers in a typical distribution or Content Distribution Network (CDN) environment. However, the accuracy and details of file-based QC are not needed at this stage. It is sufficient to do limited QC which is significantly faster.  Monitoring tools at this stage need to ensure the following:

The content manifest should be accessible over HTTP/HTTPS and all the references to profile manifests and individual segments should be accessible
Ensure that the content is properly conditioned for ABR (refer to ABR checks in the previous section)
Server responds fast enough to ensure that content is downloaded within acceptable delays and buffering needs -this can have a major impact on the playback experience for a user
Content downloads are simulated in network congestion environment to observe how the distribution server behaves under stress conditions
The content may also be decrypted at this stage to ensure that no issues were introduced during encryption
Basic audio-video quality checks (for example, blockiness, black frame, audio loudness etc.)
Passive monitoring of all the requests/responses for actual clients accessing the content
All HTTP response codes – 3xx,4xx,5xx should be monitored and logged

Several OTT monitoring solutions have emerged in the market. The OTT technology is still evolving and the requirements for what is needed for a monitoring solution at this point are also evolving. The monitoring tools need to be architecturally versatile to accommodate this evolving environment, while broadcasters figure out which issues are the most critical ones to focus on. The ideal scenario for VOD based assets is the one where OTT monitoring leverages the file-based QC tool, working in tandem and seamlessly with it. The Fig 2 below illustrates how a complete QC & monitoring solution works from ingest to delivery in the OTT world.

Figure 2: QC & Monitoring – Adaptive Bitrate Streaming

ABR Transcoding & MONITORING for OTT Live Streams

The typical live stream workflow enhanced for OTT deliveries is shown below in Fig. 3. The live stream is split into segments, and as segments are received, they are transcoded in real-time to the desired ABR format. The segments are encrypted using any of the popular DRM technologies and placed for consumption real-time on the origin servers. More segments get added while the older ones get removed. The process continues through the duration of the live content.   

As evident, a full file-based QC does not play a role here. However, we still need to ensure the following:

Basic baseband checks are done. The segments that the live content is chopped into is free from basic baseband quality issues
ABR transcoding is happening properly. ABR segments  are “good”, complying with the ABR specs to ensure seamless delivery
Timing and the load is well managed. ABR segments are made available at the right time on the servers, and the servers are able to manage the load

Essentially, on the live content, we need to perform basic segment integrity and content quality checks, ABR transcoding checks and file download checks. In short, ensure QoS and baseline QoE checks on the growing content. These checks are performed using content monitoring tools like ORION-OTT, similar to the VOD flow.

Figure 3: Monitoring for Live Streams in OTT Workflow

QC & MONITORING solutions for ott Enhanced workflows

Interra Systems provides end-to-end seamless and versatile solutions for software-based content verification, monitoring, and analysis solutions for file-based and real-time workflows in the digital media industry.

To ensure the quality of original content for VOD assets, Interra Systems’ BATON® ABR identifies quality issues from ingest to editing, compression artifacts, as well as file assembly issues. Once the content is validated with BATON ABR, it is ready for OTT delivery. Real-time streaming validations, both for VOD assets and live streams, are done with Interra Systems’ ORION™ – OTT. 


OTT technology is still evolving, and the requirements for monitoring are also changing. Monitoring tools need to be architecturally versatile in order to accommodate this environment and allow broadcasters to figure out which issues are the most critical ones to focus on. Ultimately, broadcasters should choose an OTT monitoring solution for Live and VOD assets that works in tandem with a file-based QC tool. By deploying a complete QC and monitoring solution for ingest to delivery, broadcasters can deliver the best QoS and QoE to viewers in the OTT world.

[1] Ericsson: TV AND MEDIA 2015|The empowered TV and media consumer’s influence

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How to Effectively Measure, Analyze, and Optimize QC for File-Based Broadcast Workflows

Over the years, the broadcast industry has shifted from tape-based to file-based workflows in an effort to increase operational efficiencies and reduce overall expenditures. Since that transition, file-based automated content QC solutions have emerged as the ideal method for ensuring superior content quality, providing increased cost savings and reliability over the traditional approach of visual inspection.

Yet, over the last decade, file-based media workflows have become much more sophisticated, putting tremendous pressure on broadcasters when it comes to ensuring content quality. While broadcasters used to get away with supporting a simple QC model of checking a file, reviewing its verification report, taking necessary actions, and then forgetting about the issue, this approach is no longer effective for long-term planning of content QC processes and making strategic changes. Broadcasters need advanced tools that give a more holistic view and deeper insights about how the content QC has happened over a long period of time and across different departments and sites.

This article describes a Measure, Analyze, Optimize (MAO) framework for incorporating data analytics in content QC processes. In addition to describing the benefits of this framework, the article will examine different use cases that illustrate how to get the best out of content QC.

When content QC solutions are used continuously, a large amount of QC data is generated. By analyzing this data, broadcasters can identify trends and gain a deeper understanding of content quality across the organization. These insights can help broadcasters make strategic improvements in the content QC processes and achieve greater organizational efficiencies.

The best framework for analyzing QC data is the MAO framework. The MAO framework is a three-stage process that starts out by tracking important QC performance metrics over the long term. Next comes analysis. During this phase, the metrics are analyzed to identify common patterns and operational issues; the analysis sheds light on areas of improvement and suggests required changes. For the optimization stage, those changes are applied to the content QC processes.

The cycle is then repeated. Using a file-based QC solution, broadcasters will take a second measurement, verifying that the applied changes have led to improvements in different QC performance metrics. New measurements are analyzed for additional issues and required changes.

The following sections will look at concrete examples of ways the MAO framework can be applied to improve content QC processes.

Asset Categorization

Broadcasters today are handling a huge number of assets. With so many files at their disposal, most broadcasters only have a vague idea of what types of assets they’ve acquired or created over the years. Cataloging all of the assets can be a time-consuming and expensive process. Whether or not a content management system (CMS) is being utilized, all assets usually pass through some level of QC.

QC reports contain a wealth of metadata information as well as error information about the files. The database where QC reports are stored can be exploited to analyze and categorize the assets. For instance, broadcasters can determine how many hours of content have been processed, what bit rates are being used, and which assets are HD files.

When transitioning from SD to HD, categorizing assets can be extremely useful, as it will give broadcasters an understanding of how much legacy content still exists. Figure 1 shows that a total number of 38,864 files had undergone QC. The files contained 24 different values of resolution. The most common resolution was 625(SD). About 60 percent of files were encoded at this resolution. The second most common value was 1080(HD), which covered about another 34 percent of files. There was one odd file with a resolution of 528×480.  Since there is a significant amount of SD assets, the broadcast organization may decide in this case that maintaining an SD-to-HD upconversion workflow is necessary.

Figure 1. Resolution of different files

QC Results Summarization

Performing an in-depth analysis of QC results is crucial for understanding the content QC process and ultimately reducing the number of files that are failing. For example, broadcasters can look at what percentage of files are failing due to an error, the different kinds of errors present in various files, and which errors are more prominent than others.

By digging deeper in the data, broadcasters can see how the failure rate is changing over time. Figure 2 shows how the number of tasks is changing from month to month based on success, failure, or warnings. In this particular example, there isn’t much improvement in the failure rate with time.

Figure 2. Month wise variation in success and failures

Looking at this data, broadcasters can identify the reasons why the failure rate is not improving with time. Once the reasons have been identified, the next step is to carry out the operational improvements and achieve decreasing failure rates.

There are several ways to reduce the failure rate data. One option is to restrict failures to specific watch folders to identify folders that have more problems. Alternatively, broadcasters can look at the same data for different types of content separately (e.g., HD vs. SD content). If the broadcaster is operating stations in multiple geographic locations, gaining insight into which site tends to have more failures compared with others can be valuable.

Broadcasters can also decrease the failure rate data by analyzing certain parameters like test plans, watch folders, content locations, checkers, etc. (See Figure 3.)

Figure 3. Task results by different criteria

Taking a look at the watch folders in Figure 3, it’s clear that the “Stories” watch folder seems to have a higher failure rate than others. Furthermore, the “SD Open Stories Test Plan” tends to have more failures than other test plans. These data points give broadcasters a clear plan of action with regards to where to focus attention to improve content quality.

The checker wise distribution is more useful for seeing if QC tasks are evenly distributed across different checkers. In this particular example it seems that two checkers are overloaded and are handling most of the tasks.

Another way to approach QC results is to look at the most common errors found across all tasks. In this case, broadcasters are advised to look at the files with specific problems in detail and identify if there are common causes in the workflow causing the problem in so many files. It’s important to note that sometimes this data may not be sufficient. Specific errors may be happening in fewer files but the number of occurrences of the error in those files is very high.

Capacity Planning

In order to have an efficient QC process, broadcasters must ensure that the QC system is lean and mean. There are several ways to address this:

Checkers shouldn’t be sitting idle.
QC tasks should be completed in a reasonable amount of time.
Higher priority tasks should be given resources accordingly.
Create enough capacity to handle peak load situations.
In the case of multi-site QC systems, the resources should be equitably distributed.
A regular review of distribution of resources and their usage should take place.

Figure 4. Core Utilization and Task Queue

Broadcasters can increase the average performance index by allocating more cores per task, disabling non-essential checks from the test plan, and/or ensuring faster access to content from checkers. While using more CPUs for the QC task may seem appealing, it doesn’t improve performance proportionally. Furthermore, while increasing the performance index may reduce the QC time, the system may be sitting idle if there is not enough content to be processed all the time. Adding more checkers to a QC system is expensive. Broadcasters will want to find a good balance between the need to get QC done fast and ensuring that the QC system is reasonably utilized all the time.  In this context, it is useful to divide the performance data according to various parameters. For instance, broadcasters may want to look at performance index for SD/HD/4K/8K files separately. This helps to decide which files require more cores to achieve better overall performance

Each check in a test plan adds to overall QC time. In particular, video/audio quality checks add significantly to the QC time. Reviewing the test plan QC results, broadcasters can determine which checks never fail in a particular workflow. Some of these checks may be required from a regulatory compliance perspective and should never be switched off. But others can be disabled altogether.

Sometimes different departments of the same broadcast organization purchase independent copies of QC systems. In the case of multiple offices in different locations, having independent QC systems cannot be avoided. The analytics system can import data from these different installations and provide a combined top level overview for strategic planning.


The Holy Grail for broadcasters is to provide an immersive and seamless television experience to viewers on every screen. As the content lifecycle continues to grow in the future, with broadcasters contributing and distributing content in new ways, having a structured and automated content QC approach that leverages analytics will become an even more essential part of modern file-based media workflows, ensuring broadcasters can deliver high-quality content.

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Retail Differentiation versus Service Provider Walled Garden

Both Sides Are Unhappy with STELA Progress.

Reports have emerged (the best summary appears on Multichannel News, “Group to FCC: Avoid ‘Walled Garden’ Approach to Video”) that both sides are unhappy with the progress of STELA. The debate comes down to this:

  • Service providers want the ability to control the look and feel (menus, recommendations, integration of live, VOD and DVR, etc.) around their services.
  • CE manufacturers want the ability to integrate content from multiple services (Pay TV, Netflix, YouTube, etc.) in order to give consumers a unique and highly integrated experience.

As background, The FCC mandate limits the scope to “downloadable conditional access” to enable retail products to play content from service provider services. Of course, this narrow scope replaces the physical Cablecard standard, which broke open the gates of competition in the set-top box market, away from the duopoly of Cisco (formerly Scientific Atlanta) and Motorola (then General Instruments, and now ARRIS) to allow players including Humax, Samsung, and Pace into the market. It also paved the way for retail TiVo boxes (despite technical challenges around tuning adapters for switched digital video implementations), although the set-top free household never emerged, as only one or two models of TV ever included a CableCard slot.

In one sense, a replacement has already emerged for service provider delivery to retail devices, based on the FCC definition of an “IP output”. The FCC writes that boxes “shall comply with an open industry standard that provides for audiovisual communications including service discovery, video transport, and remote control command pass-through standards for home networking.” For background, please read an earlier ABI Research insight on CVP-2, now VidiPath. This standard, which is embodied by DLNA’s VidiPath technology (DirecTV’s RVU also likely meets the definition). Note that we conducted a webinar for DLNA on VidiPath. This standard delivers a remote UI from a set-top box to a retail device, as well as delivering video using DTCP-IP standards within the home.

Against a Backdrop of a Competitive Environment

Overall, it is clear that there is significant innovation and choice within the consumer video landscape. Significant, new, differentiated services are being launched on a nearly monthly basis, including Dish’s Sling TV, Sony’s Vue, HBO Now, a new Showtime service, etc. In addition, traditional Pay TV providers are developing thinner and thinner packages (litigating with content providers over them) for “cord cutters,” with better economics than have traditionally been available for Pay TV. Consumers now have excellent choices of subscription VOD, transactional (rental and purchases) movies and TV, and ad supported content.

Interestingly, a number of dynamics have emerged in the over the top (OTT) world of Netflix, Amazon Instant Video, and iTunes that, we believe, will favor the service provider positioning over the CE OEM positioning.  Specifically, it is clear that service providers have the capability to choose the devices, locations, and content they make available. Netflix has chosen a penetration strategy, aiming to be on every internet connected device with a screen, while Apple has been more judicious, favoring Apple products as well as PC’s (through iTunes). Apple generally allows competing subscription services through the iTunes App Store (Amazon Prime Instant Video, Hulu Plus), but unless the service provider will offer revenue share for the rentals, Apple politely declines the opportunity.

Finally, at times Netflix has made its catalog available via APIs – allowing other services to “index” the Netflix catalog. Currently, based on our understanding, those APIs are only available to companies with a specific licensing agreement with Netflix (i.e. Roku has an agreement to include Netflix results in universal search). Of course, this allows service providers to define their business intent (allow traffic to the service while not being devalued to a content provider as opposed to an experience provider), and evaluate each opportunity on its own merits. There are technical wheels at work here too – developments toward universal content IDs (which could point to a show, a season, or an episode) make content more universally accessible. However, companies that have that linked up in a database (such as Rovi and Gracenote / TMS) will typically only provide it under license with a service provider, such as Netflix. Further, encryption of URLs is one of many obfuscation techniques in use to make it harder to scrape service catalog details.

Statutory Revenue Share, Negotiated Revenue Share and Partnership

The minimal arrangement that the STELA group could do is basically to endorse a variation of the HTML5 EME / MSE extensions. This would essentially allow service providers to choose a DRM provider and deliver content securely to any device which met their criteria. Tuning of modern set-top boxes and modern Pay TV services can be done only within a closed system, or by leveraging a standard such as VidiPath. These services have a mix of live content (well indexed by metadata providers), DVR content, VOD content, and interactive ads, all of which are delivered over a mix of relatively fixed location yet encrypted channels, switched digital video channels which are delivered on demand, and IP delivered channels.

As we discussed in the Netflix-Roku case, as well as Amazon’s decision about what services to deliver on an iPad through the App Store, negotiated revenue sharing is a business relationship well understood and within control of the service providers and device manufacturers.

Based on our belief that service providers’ ability to control the integration and delivery of their experience will be held up by the FCC and STELA commission, another solution to the impasse would be to develop statutory licensing rates. This would allow, for instance, a new CE manufacturer to deliver a service composed of a mix of HBO, ESPN, and ABC content with highly personalized menus. It would do so based on a rate table that determines the rates at which a wholesaler could aggregate content. Of course, negotiated license rates would be better – but it could provide a starting point for new services. This would provide the entry into a new wholesale arrangement, but may occur at the expense of control of exclusively licensed content.

Copyright Allied Business Intelligence, Inc. All Rights Reserved. This document is protected by US and International Copyright Law. No part of this document may be republished or entered into an information storage / retrieval system or database of any kind without the expressed written permission of ABI Research.  This article was contributed to NAB Thought Gallery for posting by ABI Research.

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Tech Talk

The Cloud and the Evolution of Post-Production

For many years, the pace of change in mass market IT has dwarfed that of broadcast technology. For example, consumer camera phone resolution has increased over the last ten years by 30,000%. In comparison, broadcast resolution has barely tripled. With the development costs of consumer cameras amortised over hundreds of millions of units, there is no way that the broadcast industry can compete on the pace of progress. Nor should it try to.

But in post production, large parts of the existing IT infrastructures are exactly those which benefit most from the latest revolutions in mass market IT. Computer processing power, storage, and Internet access, all fuelled by mass-market consumer demand, are exactly the features offered by cloud solutions.

As new technology enables more and more of the broadcast workflow to move into the cloud, little more than a modest computer with an Internet connection is needed to be able to create there. Processor speed, file sizes and security, long regarded as significant barriers to cloud-based workflows, are no longer issues. Instead of transitioning to newer and newer generations of broadcast hardware, we’re moving from one IT solution to another, saving significant costs.

What Makes the Cloud Suited for Post-Production?

Just as large-scale suppliers provide electricity more economically, processing and storage requirements can often be met more economically by cloud services. And just as electricity comes to us directly via connection to a power grid, data comes directly to a computer via connection to the Internet. In addition, cloud services from different providers can easily interact through automated interfaces, providing a highly flexible way of using solutions and add-ons from different suppliers throughout a given workflow.

However, when it comes to video post-production, cloud services are not all equal. Generic cloud-computing providers, such as Amazon Web Services, are built for IT and consumer data rather than the voluminous data and throughput performance requirements associated with broadcast video. The requirements for editing video are formidable, with huge amounts of data and the need for real-time responses. The process of rendering effects and transitions in a generic cloud environment are expensive since these services charge according to CPU power and time used, and any unresponsiveness would be frustrating for users accustomed to dedicated desktop solutions. When all of this is taken into account, building a broadcast post-production operation on a generic cloud infrastructure simply doesn’t meet broadcast standards for performance, cost, or reliability.

Dedicated platforms optimised for broadcast applications offer lower cost and far greater performance. They also provide increased control over where the data is stored for legal, regulatory, and personal preference requirements. The ease of switching between cloud providers means that the availability of a single broadcast-ready cloud will, over time, eliminate poorly performing alternatives.

Capacity and Efficiency

The capacity of the Internet and cloud services is essentially unlimited, with supply expanding to meet demand. Storage and Internet speed nearly double every year, and new solutions from the world of mobile are improving power consumption within the cloud. With 10 Gb/s intranet backbones in service now, throughput will continue to improve drastically.

The greatest efficiencies come from leveraging client computing to undertake as much of the processing workload as possible. Though it might sound odd coming from a provider of cloud solutions, such a scenario provides immediate scalability, reduces latency to negligible levels, cuts costs and allows access with even the most modest of hardware set-ups.

Better Picture Quality

The cloud is well suited to the post-production convention of working with proxies, a method that, at least for the foreseeable future, will be more efficient than working with high-resolution content. Proxies are continuing to improve in both quality and resolution, with, for example, higher resolutions provided for fine-cut editing than for logging. Conforming from original HD sources ensures minimal generation loss on the final output.

Many are already making 4K home videos in the cloud aided largely by the fact that connection speeds have doubled every year while storage costs have halved. As these factors continue to improve, the cloud will look more and more attractive as an alternative in a wide range of post-production workflows.

Technology has made post-production in the cloud easy. Now the focus is on improving and perfecting reliability, responsiveness, interface design, price, and integration to other applications.


Data security is an issue with any IT-based system. The risks of losing data can be mitigated by keeping multiple copies of data in multiple sites — something that is much easier for a cloud provider than for an individual broadcaster.   As long as users have and use a back-up strategy, the security of their video assets can be assured.

Workflows and Adoption

The prospect of change takes people out of their comfort zones, so it happens much more slowly than changes in the underlying technologies. This is especially true of broadcast and creative workflows that come with their own unique tricks, work-arounds, and emotional attachments. With people now using the cloud in their everyday lives, they are more open to adopting it professionally. An increasingly technology-literate young workforce — already fluent with cloud technologies in their personal lives — is accelerating cloud acceptance and adoption.

Beyond Post-Production

The future of cloud-based broadcast infrastructures goes well beyond editing. Maintaining different equipment for the many different components of a post-production workflow becomes less important as the cloud offerings improve. Already, the cloud can provide fully featured solutions for ingest, graphics creation, post-production, publishing, hosting, search, distribution and advertising.

Producers of genres such as near-live sports for Internet and mobile distribution can already do all their post-production in the cloud. Dedicated cloud post-production platforms can take in multiple live HD video feeds (at one point during testing for the Summer Games, there were 200!), incorporate graphics from various cloud creation systems, perform all the editing and finishing, conform at full HD resolution, and hand over to other cloud solutions for mass distribution. Clients can mix and match. In fact, Microsoft Azure is designed for just such a scenario. An ever wider variety of options is opening up to broadcasters, working flawlessly together to create customized, end-to-end cloud solutions.


The relentless advances in technology are transforming post-production. Traditional desktop IT solutions are being replaced with more cost-efficient and productive cloud-based solutions. At the same time, technological change is transforming the distribution and consumption end of the business. Suppliers of traditional post tools and their users would do well to embrace these changes.

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Tech Talk

Myth and Reality of Auto-Correction in File-based Workflows

File-based workflows are ubiquitous in the broadcast world today. The file-based flow has brought enormous efficiencies and made adoption of emerging technologies like Adaptive Bit-Rate (ABR), 4K, UHD, and beyond possible. Multiple delivery formats are now possible because of file-based workflows and its integration with traditional IT infrastructure.

However, the adoption of file-based flows comes with its own set of challenges. The first one being – does my file have the right media, in the right format and without artifacts? Fortunately, the leading auto QC tools have kept pace with the growing technology advances to provide us with this peace of mind. However, there are still many unsolved video artifact issues that the auto QC tools grapple with. Video dropout, for instance, is still a subject of research and significant advances are expected over the next few years to more accurately detect these issues.

Once we know the issues, a natural question is – if the auto QC tool can detect the problem, can it also fix it? The answer to this is not so straightforward. In the analog and tape world, the content was as good as it was created. Correction was limited to simpler processes like signal level clipping, or color phase correction, which could all be done at the delivery stage.  Of course if the content deteriorated due to tape issues like tears, twists etc., one went through expensive film restoration techniques, if one could afford them – these were manually assisted processes done under the fine eye of the editor. Can auto-correction take care of these?

It so turns out that correction is not that simple in the file-based world. The content is often stored and delivered in a compressed format. It is also wrapped in containers to keep the audio, video, sub-titles and a host of metadata information for the tools to work properly in the workflow. In the file-based workflow, correction of the content requires not only changes to the baseband content, but also  re-encoding and re-wrapping of the corrected content back to the compressed format.

We see the following challenges to the auto correction process:

  • Firstly, there are several baseband issues that are not even detected automatically (in other words, they are outside the scope of auto QC), forget about auto correcting them. Remember, analog world people used manually assisted processes under editorial supervision.
  • Secondly, after the corrections are applied, through manual or automated process, if the transcode including the re-wrap processes are not managed properly, auto correction will introduce new set of issues – the corrected content may even be worse than what you started with, resulting in an unproductive looping.

How then can you depend upon an auto QC tool to do auto-correction? Well, there are some issues that are amenable to auto-correction, albeit a few.  Most of the issues can be categorized in three types: metadata inconsistency, video essence issues, and audio essence issues. Whenever re-encoding and re-wrapping of content is required to be done after correction, auto-correction via auto QC tool may not work very well. On the other hand, metadata inconsistencies and audio essence issues are more amenable to auto-correction. However, correction of video essence issues typically do not converge when performed on encoded files.  In fact some of the video issues take place due to encoding / transcoding process as well. The right workflow, tools and techniques are needed to be deployed to make the auto-correction flow work well for you.

It is a misconception that auto-QC tools can also auto-correct all kinds of content issues. That is a sweeping generalization, and QC tools should not fuel that false notion. 

Figure 1 below, provides a typical high level file-based workflow. After ingest, content is edited to create a mezzanine file which is of high quality but with minimal compression. Different facilities can select their own mezzanine format ranging from Motion JPEG to ProRES to AVC Intra. Mezzanine content is then transcoded to multiple compressed formats for different delivery formats. 

Fig. 1 –  Typical File-based Workflow

Since content undergoes some form of complex transformation at each stage, this can potentially introduce stage specific issues within the content.  Each stage can introduce different types of issues. Similarly, different levels of correction are possible at different stages. Auto-correction works well if it is done on uncompressed digital content, as it can be modified and corrected, before being compressed and wrapped, much like the correction in the analog world. However, if the content is already encoded and wrapped (e.g., transcoded content), then auto-correction gets far more complicated – the re-encode process after the auto-correction introduces other issues, making the correction process divergent, less effective and even infeasible.

The ingest process often introduces artifacts like dropouts, signal level errors in video and transient noise, wow and flutter in audio. Auto-correction can work well at this stage if the digitized uncompressed and un-wrapped content is available. Issues like video signal levels, RGB color gamut, audio loudness etc., can be corrected in the uncompressed digital content, which is then encoded and wrapped into the mezzanine format like AVC intra and J2K. However, there are a host of baseband video and audio issues which should not be auto-corrected as one runs the risk of modifying the content.

The ingest process often introduces artifacts like dropouts, signal level errors in video and transient noise, wow and flutter in audio. Issues like video signal levels, RGB color gamut, audio loudness etc., can be corrected in the uncompressed digital content, which is then encoded and wrapped into the mezzanine format like AVC intra and J2K. Although, auto correction will work properly for most of the error scenarios, however, the correction may modify the content to an unacceptable level. For example, while correcting VSL or RGB color gamut errors, the characteristics such as hue, saturation or contrast might be changed affecting the perceptual experience of the viewer. Similar example holds true for correction of transient noise in case of audio. In these cases, manual inspection is also required after correction of the content.

The similar argument holds for the editing stage. However, there are several other types of issues that can crop up at the editing stage which cannot be auto-corrected. Trying to merge two different media during the editing process can lead to field order issues. We often see customers complaining about VSL, RGB errors which get introduced at the editing stage while adding special effects and graphics/text in the content.

After transcoding, auto-correction can be done to a very limited extent. Transcoding for delivery purposes is a complex transformation where content is converted from one format to other. Many issues such as audio / video corruption, blockiness, blurriness, pixelation, audio / video dropouts, motion jerks, audio clipping etc. have been found to occur during the conversion process, not to forget non-compliance with audio/video formats or delivery specifications. Transcoders can also get affected by buffering issues during transcoding process, leading to overflow/underflow like situations. This can lead to introduction of freeze or silence frames within the content. Even if we were to auto-correct the issues, re-encoding the corrected content has the potential of introducing similar issues in different forms and different parts of the content. It is the best that auto-QC and transcoding tools collaborate to correct these issues.

With this background, let’s now have a look at different kinds of issues that an auto QC solution can detect in an encoded content and what needs to be done to correct those issues.

Any QC solution will typically detect four kinds of issues:

  • Conformance Errors: These errors are primarily non-compliance to different audio video standards. For example, an MPEG-2 video stream must be compliant with the MPEG-2 video standard. Any non-conformance needs to get reported. This category also includes checking compliance of the content against different regional/delivery specifications like DPP, IMF, AS-02 etc. Correction of these kinds of issues generally requires the files to be re-encoded and re-wrapped. Baseband correction is not required for these kinds of errors.
  • Metadata Errors: Each workflow and each stage in the workflow has its own requirement in terms of metadata. For example, an HD delivery requires resolution to be 1920 x1080. Content meant for broadcasting in the USA needs to have a frame rate of 29.97 fps. Each delivery or stage can have further restrictions on parameters like scanning type, GOP structure, profile and level of encoded media, the number of audio / video tracks etc. Any deviation from the acceptable values will lead to content being rejected. So a QC solution is expected to check for such metadata properties at different stages of the workflow. Moreover, certain information like resolution, time-codes and field order are present / encoded at both the wrapper and at the audio/video level. If there is any inconsistency between the layers, a QC solution should be able to report the same. If the issue is with the wrapper layer (MXF, QuickTime, Transport) then only re-wrapping needs to be done to correct the content. But in cases, where metadata information at video/audio level is incorrect, one will need to perform basic re-encoding along with re-wrapping. Example for such a case would be if the US media environment requires content with 29.97 fps but underlying media has frame rate of 24 fps. Simple fix for this issue would be to introduce cadence pattern of 3:2 at video layer. Such correction will need basic modifications to video layer and further re-wrapping of compressed media.
  • Baseband Errors: These errors are different audio / video artifacts which lead to deterioration in perceivable quality of content.  These errors are introduced because of stage specific transformations as discussed earlier. This includes errors like freeze frames, blockiness, dropouts in video and issues like silence, different kinds of noises in audio. Correction of such artifacts first needs to be done at the baseband level followed by the re-encode and re-wrap processes.
  • Regulatory Compliance Errors: Different regions of the world have their own regulations in terms of content quality. It includes loudness control regulation all over the world. We have CALM Act in the USA, EBU R128 standard is widely followed in Europe. Likewise, the UK broadcasting market requires content to be checked for any possible flash patterns to avoid photo sensitive epilepsy situations. It is possible to correct these kinds of errors via baseband correction followed by transcoding process. 

Many kinds of errors discussed above will require compressed content to be processed (re-encoded or re-wrapped or both) to remove the errors. This processing is not as straightforward as it looks.  It is critical to decide where in the workflow, and with what tools these errors should be corrected.  Let’s take the case of an auto QC tool also claiming to provide “good” correction capabilities.

The tool comes with its own encoder. The workflow would look like this:

Fig. 2 – QC Tool-based Transcode & Correction Workflow

So a typical flow for auto QC and correction flow will work somewhat as below:

  • Mezzanine file is converted to delivery format using facility specific transcoder
  • The transcoded content is then checked using a QC tool and an output report is generated; The report will contain detected errors, if any
  • If the content has no errors, it goes to the play-out stage otherwise it moves to correction workflow which is an extension of the QC tool here
  • The QC tool then performs correction on the basis of the reported errors – it uses baseband correction algorithms along with its transcoder for correction of the content

The corrected content is then ready and can be moved to the play-out stage for final delivery. Up to this point everything looks good and quite rosy. But users of this type of workflows may be in for a shock when corrected content fails to meet the delivery requirements and gets rejected. The situation is quite common because the corrected content may not be of desired quality and may have additional new issues which were not there in the first place. Let’s now look at the challenges involved in the above correction process.  


Transcoding is a complex transformation process involving conversion of content from one form to another. A transcoder output is controlled based on a host of input settings to handle varying flavors of container and media formats, and to meet various kinds of delivery specifications, in order to get the media with a required level of quality and so on. The input settings control various internal processes of the transcoder which includes motion estimation techniques, bit budgeting, rate distortion model, selection of QP values and matrices, the block interpolation/estimation processes, reference frame selection and more. The final output of the content is dependent on the quality of the said processes being used inside the transcoder as well as the input parameters selected. Inappropriate selection and usage of input settings to transcoder may result in output content not meeting the intended requirements. A wrongly selected bitrate parameter can degrade the quality of the output content with new artifacts (out of RGB color gamut errors, video signal level errors, blockiness, softness etc.).  Another such scenario can come up while selecting display field order for the output content. An SD DV content (bottom-field first by default) when transcoded to MPEG-2 video (top-field first by default) will lead to motion-judder issues in the output because it was required to change the default field order input value to the required one. Thus in order to create good quality and optimally compressed content, several parameters need to be fine-tuned and managed as per the facility’s requirement. Setting these parameters/options even for the best transcoder requires expertise. One cannot expect another ‘generic’ transcoder to be able to perform at the same level.  It is hence to be expected that any attempt to re-encode the content with another encoder could lead to negative effects. The second encoder, while trying to encode the corrected content at the same bitrate may follow a different bit allocation strategy leading to compression issues like blockiness etc. It is also highly possible that a new encode process can completely miss certain information that is vital for the content.  To name a few, user data present at video level may get lost in the process of transcoding. Another example would be watermarks, where the generator leaves a special mark in the video/audio to establish publisher information. It is impossible to replicate or reinsert these watermarks unless the same set of tools is used during correction. It is also quite possible that some of the settings are not even consistent among the two transcoders. For example, the other transcoder might be using different motion estimation techniques or rate distortion algorithm inside it or it may also happen that the original set of tools inside a profile or level is not supported. That will cause the correction process to generate media data with unacceptable profile / level and content quality, which will be rejected later at the play-out stage. At a minimum, one should use the same encode process and tools as used during content creation.

However, issues don’t end here. Even if we plan to use the same transcoder, it can potentially introduce new errors while correcting existing ones.

  • The re-encoding process leads to loss of some audio / video information which in turn impacts the quality of content. Degradation in quality, though minimal for most of the cases, will depend on the encoding parameters and the content itself. If re-encoding is done to reduce bitrate of the content, it will lead to compression artifacts like blockiness, pixelation etc.
  • Conformance errors may also get introduced because of faults in encoder under certain conditions.
  • In a few cases, it is also possible that metadata errors may be introduced, if the wrapper information is not set correctly; one such example could be the field order – assume a case where the field order has changed after re-encoding but the same is not reflected at the wrapper level. Such inconsistencies can arise and thus there is a need for a better management of such issues.


Another big challenge in a correction flow is to re-assemble / re-wrap the corrected and compressed media with exactly the same properties as the original file. Transcoders come with their own built-in Muxers or can be integrated with third party Muxers to wrap compressed media into a container. The media workflows in the broadcast industry use their own set of unique tools to transform and assemble media information. A different re-wrap tool or the same tool with differently encoded essence will produce different results.  This implies that the corrected output file may be different in properties in comparison to the input source file. An example is MXF version, where the original file may have been assembled using a lower version. But if a new Muxer used during the correction process uses a higher MXF version, it may cause interoperability issues in the workflow. Also, the MXF specification allows addition of new proprietary ULs that can be generated and interpreted by specific Muxers / applications. For other tools, it acts as ‘Dark Metadata’ that will be ignored while processing. Hence, the second Muxer for such cases will ignore the dark metadata and the proprietary information would be missed in the corrected content. Hence, it’s an imperative to avoid the usage of two different Muxers in your correction workflow.

Baseband Correction

In baseband correction, there are issues like video signal level, RGB gamut, field order, digital dropout, loudness related errors which can be intelligently corrected. For such issues, the content is first decoded. Algorithms are then applied on the baseband / uncompressed signal to intelligently correct them for the reported issues. Once the baseband correction is done, the content is re-encoded and re-wrapped. Can we fully rely on baseband correction? Perhaps not. It is possible that a certain correction may introduce fresh errors during re-encoding process. For example, VSL / RGB correction may end up altering the block boundary pixels which in turn leads to blockiness like issues in the corrected content. There are additional set of errors which cannot be auto corrected like: freeze frames, silence and certain noises. If the capture device, for some unknown reasons, fails to capture a few frames, it can potentially lead to a freeze like situation. It’s not possible to re-create those dropped frames during correction cycle until we have access to the dropped frames. It is also possible that certain special effects that are added to video may cause QC solution to detect those effects as blurred or pixilated area in the video frame. In this scenario, it is not desired to correct the content. Hence, there is a need for manual intervention to understand these anomalies and then take appropriate corrective measures. Some of these aberrations maybe intentionally be introduced as special effects, and therefore, needs no correction.

Proposed Workflow

Auto correction has its own set of challenges as mentioned in the previous section. Because of these challenges, it is not practical to expect an auto-correction tool to be a panacea for all issues. In fact, there is a class of issues that can be auto-corrected. Coupled with the right set of tools and workflow, one can make auto-correction work under these limited circumstances, such as:

Legalization of audio and video content and some cases of regulatory compliances. These include audio loudness, true peak, loudness range, audio levels, audio noises like background noise, crackle. On the video side, it includes video signal levels, RGB color gamut, cases of video dropouts and also flashiness patterns. The proposal here is to limit the role of QC solution to baseband correction. The correction flow can then rely on facility specific transcoder for its encoding needs. For example, a facility may depend on Dolby tools for encoding of AC-3 /Dolby-E content. In such a scenario, the role of auto QC tool is to perform baseband correction for audio and then submit the encoding job to Dolby tools. This would ensure consistency between the original content and the corrected content in terms of metadata and quality.

Another practical use case here would be integration of auto QC tools with the workflow automation / transcoding solutions like Telestream Vantage. Once the content is transcoded, QC tools can then perform content analysis followed by intelligent correction depending on the detected and correctable errors. The workflows can be further configured to feed the corrected but uncompressed output from QC tools to in house transcoders like Vantage for re-wrapping/re-encoding. Submission of transcoding jobs to the transcoder after correction can be initiated in multiple ways. In some cases it can be as simple as dropping a file in a Watch Folder while for other cases, a QC solution may need to invoke transcoder’s web services to start the required job. For larger workflows, it would also make sense for MAM /workflow automation solutions to create some kind of correction / self-healing workflows so that transcoding action can be invoked once correction process is done. These discussed approaches would require the QC solution to be integrated with some of the widely used transcoders / workflow solutions so that a large number of customers get the benefit. Such a flow would typically look like this.

Fig. 3 – Proposed Transcode & Correction Workflow

The steps followed in the proposed flow are listed below:

  • Media file is analyzed using the QC tool
  • The content moves to the play-out folder if it passes. In case of a failure, the content is de-muxed, decoded and then corrected for anomalies at the baseband layer, if required
  • Content is then submitted for transcoding using facility specific tools
  • The correction process may also need to specify new parameters / settings to be used during the transcoding stage
  • Modes for submission of transcoding job can vary as discussed earlier

The use of the same transcoder will eliminate lots of potential issues and make the above flow more practical and amenable to correction.

Most of the challenges in correction process arise because of re-encoding / re-wrapping. Correction which does not change the size of the compressed content can be handled without a possible re-wrap.  That is true for uncompressed content based on baseband correction. Audio content in a lot of cases is stored in an uncompressed manner using formats like PCM, AIFF, BWF or AES3, owing to the fact that audio requires much less data size as compared to video. Since uncompressed content occupies fixed block sizes at certain offsets, it is not necessary to re-wrap the whole media. A smart correction tool can simply perform what we call as in-place correction.  The goal here is to un-wrap the audio, record the length of each uncompressed audio block with the corresponding file offsets. Once baseband correction has happened, corrected content can then be written back to the main file block by block using the recorded information. This way wrapper information or media data from other tracks remain untouched.

The above strategy is really useful for correcting audio errors like program loudness, loudness range, true peak etc. and it works efficiently in an iterative correction process. Errors like loudness, loudness range cannot be corrected in a single run. They may require multiple correction runs to reach desired levels. In-place correction ensures that no temporary file or buffer needs to be maintained for storing intermediate media. Corrected output values can be re-written to the final file for each iteration. This strategy works out not only to be efficient but also fast.  The concept of in-place correction can also be extended to uncompressed video formats like YUV, RGB. But since uncompressed video formats are not widely used, it may not be very beneficial to the end customer.

Another class of issues that can be corrected is metadata inconsistency errors. In cases where the encoded content is correct, but container metadata has been wrongly encoded, the problem of correcting the content requires only metadata changes for specific fields. These corrections can be applied without the need for transcoding or rewrap of the content and are very amenable to auto-correction. This again falls into the category of in-place correction. For -example, if there is a discrepancy between the resolution information present at the MXF layer compared to the actual video resolution, the resolution information at the MXF layer needs to be corrected by directly accessing the headers and there is no need to transcode or re-wrap the file. This scenario would include correction of metadata fields like frame rate, chroma format, aspect ratios, sampling frequency, encoded duration etc.

To conclude, auto QC is now an essential component in file based workflows and is widely used these days. This has triggered the need for a QC solution which can auto-correct errors in order to save time and resources. It is based on the thought that if a tool can detect error, it can also potentially fix it. But auto-correction in the file-based world is a more complex process and should not be trivialized. A QC tool having in-built support for auto correction including transcoding has issues of its own. Transcoding and re-wrapping processes if not managed properly, can introduce fresh issues into corrected content leading to further degradation of content quality. Hence, it is not possible to fully rely on such auto correction flows. A more practical approach would be to reuse facility specific tools for encoding needs during the correction process. In such scenarios, the role of a QC tool is limited to baseband and metadata correction or setting the transcoder correctly. A smarter in-place correction strategy can also be adopted in case of uncompressed content. Having said this, there is still a set of issues which requires manual intervention and thus cannot be auto corrected. Hence, the scope of QC tools for auto correction is limited but feasible for a set of issues provided we use the right tools, workflows and techniques.

The advanced auto QC tools can be used to automatically detect the video and audio artifacts, focusing on legitimate auto-correction in a controlled and restricted manner.  

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Tech Talk

The Critical Characteristics of a 4K Quality Test Solution

The buzz about 4K is at full volume now – how to record it; how to edit and package it; how to display it. And there is another consideration: how to test it for quality. Even though there is no widespread distribution of 4K content yet, that day is rapidly approaching, and broadcasters and equipment manufacturers alike have to be ready with equipment and workflows that will yield the most reliable, pristine signals and the best possible 4K viewing experience and also in every format on every screen. Not only do these factors affect viewer satisfaction and loyalty and, in some cases, subscriptions; they also affect ad revenue, service level agreements, and compliance. So, testing and verifying quality is no small thing for a content distributor.

The industry trend is toward more programming and more devices with varying screen sizes. At the same time, new and more efficient encoding standards will allow for additional channels over the same or smaller data rates. These trends make for a seemingly infinite number of formats to manage, making it all the more critical to put the right equipment and workflows in place now.

Broadcasters and service providers who are planning to launch 4K services are in the process of determining their ideal distribution scenarios right now, using real-world analysis and QC tools to research the technology requirements of delivering 4K content to a wide range of services: real-time TV channels delivered through cable IPTV or satellite, file-based delivery of VOD content, and streaming content over the Internet. Based on that research, they will define their workflows and assemble the right combination of gear to meet the rapidly approaching demand for 4K content.

And when those theoretical exercises become reality, and the content distributors must put their carefully planned 4K workflows into practice, they’ll use those same tools to conduct testing and QC after the content goes live. A typical content distributor could receive a program feed from a number of sources, such as a satellite, an existing network, or an uncompressed file from a postproduction process. From there the operation provides both high-bit-rate encoded video for broadcast and at the same time puts out compressed “adaptive streams” as a number of profiles, in different bit rates to support any number of OTT platforms. As time goes by, more and more of those uncompressed source streams will originate from 4K content — content that must be decoded on the fly, recorded, ingested for QC testing, and encoded using the compression codecs required by the target devices. During the testing phase, the system must provide an automated means of testing key attributes of video quality, audio quality, lip sync, and loudness.

What sort of solution can handle testing for 4K video delivery both now, in the research phase, and after the content goes live? What goes into the ideal testing tools?

Must-Have Items for Testing 4K Video Quality

Most important is the ability to ingest, record in 4K resolution, and test content from any compressed or uncompressed video source using today’s SDI-type interfaces, whether the signal comes from an IP network, a file, or an uncompressed feed from a video infrastructure. This also includes files from the editing suite, such as multiple frames of uncompressed sequences of any length, or lightly compressed mezzanine-level files.

The testing tools should also be able to decode any encoded file in the current MPEG or JPEG standards, as well as in emerging encoding standards such as HEVC. In addition, it is critical that the system be able to play back the content in every resolution required for delivery to every screen, from UHDTV and HDTV to resolutions for mobile devices.

Another important characteristic is a flexible input/output architecture that lets users test individual devices and then place them into a test network structure — either at a live broadcasting plant, a content delivery network, or in the lab of the product’s manufacturer or developer.

The ideal test solution should also support not only 4K, but the full range of resolution levels, from streaming formats for handheld devices and PCs; to today’s most active broadcast-level HD formats; to high-frame-rate HD.

The Importance of Audio Measurements

A 4K testing tool should include audio, because audio requires several tests in addition to video. In order to account for all audio variables, the testing tools must perform three key audio measurements. The first, perceptual quality testing, simulates a human perception test and creates a measurement that is as close as possible to an actual subjective study done in a standardized environment.

The second critical audio measurement is for performance, which identifies performance issues with audio devices or audio in the network chain. These tests find audio faults such as silences and glitches, and a sufficient solution should log the failures for instantaneous review. Audio performance testing also should cover lip sync measurements with millisecond accuracy.

Audio loudness testing, which has become extremely important with the adoption of global loudness standards, is the third audio measurement that the ideal test solution should include. Effective loudness testing involves applying loudness standards to every distributed program’s audio channel groupings and reporting a measurement.

When searching for a way to analyze 4K video quality, it’s important to pay attention to those characteristics. The right test solution can mean the difference between success and failure in the world of 4K content delivery.

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Auto QC in the Digitization Workflow


Over the years, content creators and broadcasters have accumulated large libraries of assets in analog formats. With the switch to the digital workflow, there is a critical need to digitize all these assets because of multiple reasons:

  • Risk of losing the asset forever if not digitized
  • Preservation of asset for posterity, since digital format offers immunity from degradation over time
  • Getting the asset ready to be used and monetized in today’s digital workflows
  • Space and operational costs optimization

As a result, the digitization of analog tapes and archiving into digital libraries is critical to complete the transition to the file-based digital world. Post digitization, the content becomes the master – the analog tapes are thrown away. The asset, however, is only as good as the digitization process. What if the process was faulty, or there were errors introduced during the digitization process itself – the tape head was not aligned, the tape was read twisted, there was audio and video drift, or some audio track went completely missing, color bleeding happened, or maybe there was too much hue or red in the ingested material, or the tape head was not clean and it inserted a vertical line on all the frames – the possibilities of things going wrong are immense. And all of these are known to happen. If the process is faulty in anyway, the loss is immense – priceless assets will be lost forever. So what does the archiving process rely on – eyeball QC of the ingested content.  But with thousands of hours of content to be digitized, manual QC is neither a practical nor a good option. The manual process is unreliable and is fraught with errors. It lacks consistency and with human fatigue setting in, it tends to be unreliable. Further, there are several errors that are not human detectable, but only machine detectable in the file-based digital workflows. Additionally, metadata checking can be erroneous – the asset with wrong metadata may also be practically lost in the archives, never to be retrieved when needed.

To overcome all the shortcomings, auto QC is now an essential and central part of the digitization workflow. The process is fast and efficient, consistent and reliable. When coupled with a manual review process on random and/or erroneous digitized content, the results are vastly improved in terms of the digitized content quality. Good auto QC tools with deep video quality checks for analog dropouts are increasingly deployed in the tape archiving process. Auto QC quality checks need to be enhanced to handle many digitization specific issues.  Sensitive and detailed video dropout checks are critical for good archiving and one cannot take shortcuts with simple file-checking tools – industry grade QC tools with in-depth video dropout checks specially developed for analog tape ingest need to be deployed. A word of caution – the field of video dropouts is a subject of R&D with several checks still being evolved. This paper explores some of the checks in depth and how auto QC is being deployed in the digitization workflows.  

The Need for Digitization and the Process

Most archives, broadcasters, universities, governments and television stations have thousands of hours of content accumulated and stored in analog tapes over the years.  A typical broadcaster may have nearly 100,000 to 200,000 tapes of one hour duration collected over a ten-year period. While a few stations might have Super 8 or U-Matic tapes, majority of the tapes are a mix of Betacam (SP/SX Digital/IMX), XDCAM or HDCAM.

Preservation of the tapes requires not only space, maintaining correct tags, sorting of the tapes in correct sequence (like all sequence of sports together), the mere number of tapes creates a huge problem for the management to effectively use the tapes when needed. Also, the quality of the tapes deteriorates with time due to the inherent nature of magnetic tapes. In many cases the recoverability of the programs from the old tapes can no longer be guaranteed. Maintaining the tapes is a costly affair while the quality of the content is still not guaranteed. The digital workflow offers a solution to this.

Once tapes are digitized, facilities can achieve multiple benefits from the same. These include:

  • Preservation of assets without the fear of quality loss or degradation
  • Optimization of space and operational costs – retaining large archives of tapes in temperature and humidity controlled spaces is expensive while storing digital content in files is a lot less expensive
  • Faster access and retrieval from the archives with enhanced metadata search capabilities
  • Online content for new audiences and monetization possibilities

Facilities are fast migrating to complete digital and file-based workflows and getting rid of the old tape archives.

Figure 1 below provides a typical digitization process deployed during migration.

Stage 1.  Tape Preview

At this stage, the different type of tapes (IMX, Betacam, HDCAM, XDCAM etc.) are sorted and tagged. Tapes are also physically checked for tape quality, presence of any foreign body in the tape, physical damage, tape twisting etc. If some of the important tapes are found to be damaged considerably, the tapes are usually sent to external specialist for restoration.

Stage 2. Tape Preparation and Cleaning

The tapes identified and sorted for digitization are moved to the ingest area at least 24hours before the actual ingestion of the tapes to avoid sudden expansion/contraction of the tapes. The ready tapes are then loaded in a tape cleaning machines to remove the dust and residuals like oxide deposits. With a huge number of tapes to be digitized, barcode labels are generally put on the tapes for better tracking and mapping of the metadata to the assets.  Barcodes are also used by the downstream tools to automatically select the transcode profile during the digitization process.

Stage 3. Digitization of Tapes

At this stage, the tapes are played back and the ingested digitized content is encoded to house formats like Jpeg2000, AVC Intra etc. For large scale digitization, automated robots are deployed which can feed the tapes to VTR automatically from the stacked tapes using barcodes.  Apart from creating the digitized files, a database containing the metadata is also updated for the digitized assets.  A low-res proxy file is also generated along with the hi-res files.

Stage 4. Quality Checking

The quality of the ingested content must be checked to ensure proper ingestion. Post digitization, the digital content becomes the master and the tape becomes redundant. It is therefore essential to ensure that the right quality has been achieved in the digital master, before the tape is thrown away. If the content volume is low, one may rely on eyeball or manual QC to check the digitized content quality. However, with hundreds of hours of digitized content, manual QC is not a practical option.  The manual process is also fraught with errors:

  • Manual QC lacks consistency
  • Some errors are not perceptible, but manifest themselves only during playback on some equipment
  • Human fatigues sets in, leading to unreliable QC process
  • Metadata checking can be erroneous – the asset with wrong metadata may also be practically lost in the archives, never to be retrieved when needed

Large scale digitization hence relies on auto QC tools to assist in the quality checking process.  However, as we will see in the next section, there are a host of issues that can crop up in the storage of tapes and the playback process, which impact the video quality of the digitized assets. Similar issues can arise in the audio as well.  

All these can lead to various different kinds of artefacts in the ingested digitized content. These artefacts are classified as “analog dropouts” in the video and the associated audio. A good auto QC tool should be able to reliably and accurately detect such artefacts. While one can identify these artefacts with visual inspection, identifying all such issues automatically through auto QC tools is still a subject of research and a lot of R&D is being done on the same (we will go into more details on this in the next section).  Some of the advance auto QC tools provide a higher degree of reliability, accuracy and coverage of these analog artefacts, and are much better suited for deployment in the digitization workflow.

The auto QC process can be complemented with a manual review process to finally accept or reject the digitized content.

Stage 5. The Archiving Process

Once the digitized content is accepted, it is then archived using the selected archiving software. Metadata is updated, along with the proxy file. The process is complete and the corresponding tape can be discarded.

The Auto QC Process

As discussed above, auto QC is now an essential and central part of the archiving workflow. The auto QC process is fast, efficient, consistent and reliable. When coupled with a manual review process on random and/or erroneous digitized content, we achieve higher levels of productivity with vastly improved results.  

Figure 2, below, shows a typical auto QC workflow in a digitization project.

There are mostly four types of checks that are done as part of a quality checking process on digitized files:

  • Checking the compliance of the generated content
  • Checking for timecodes and metadata
  • Checking the baseband quality in audio and video
  • Checking for encoding/transcoding errors, if content is compressed

Compliance and metadata checking is a straightforward process needed to ensure that digitized content will work with all downstream tools. It is similar in nature to the checking done in current file-based workflows.  The real complexity comes with ensuring that baseband quality of the digitized content is above the defined and agreed to threshold level. This becomes even more challenging when the same has to be done reliably with auto QC tools. Video issues can manifest themselves in different ways, and each one of them requires deep R&D to detect them reliably and accurately. With one large broadcaster, we saw over 50 different types of video quality issues in the digitized tapes. In the next section, we describe some of these in more details. 

Quality Issues and Detection

The information embedded within tapes is in the form of voltage signals. The formation of each pixel, frame or picture is attributed by stored signal values on magnetic tapes. Alteration in natural variations in these signal values will lead to incorrect color values for captured pixels and cause errors in formation of fields, frames or pictures. These alterations are caused due to mishandling, ageing and improper maintenance of tapes. These can also be due to errors within the digitization process being used. The resulting video artefacts in this way are collectively termed here as analog dropouts. Some examples are blotches, scratches, miss-tracking, head clog, skew error, horizontal/vertical sync pulse loss, etc. The following sections will discuss some of the commonly observed analog video dropouts in further detail.

Horizontal / Vertical Sync Pulse Loss

A video frame consists of multiple horizontal scan lines spread across the vertical resolution. A specific voltage level exists at the end of each scan line indicating its end and start of the next scan line. Any variation in the voltage level (due to noise) will shift content lines, perceptually viewed as horizontal lines. This is shown in the snapshot below (Figure 3 a).

Vertical sync pulse is another such voltage level controlling the start/end of a new video frame. Any deviation in the voltage level will disturb the start of the formation of the next frame. Vertical sync pulse loss merges the two adjoining frames at the frame boundary (Figure 3b).

Skew Error

A magnetic tape can have dimensional changes due to continuous expansion or shrinkage of the tape surface over time. Due to this, the recorded tracks are affected by changes in length and angle resulting in misalignment with respect to the playback head. During playback/recording, this loss in alignment will shift a band of scan lines at the top/bottom of the picture. This horizontal shifted portion of the video frame at the top or the bottom part is termed as Skew Error (Figure 4).

Line Repetition Error

An analog to digital conversion device gets the video data in the form of scan lines. The buffers that store each scan line data are updated regularly after each sample and hold duration. The line repetition error is caused due to issues in controlling the signals – the current scan line is not captured and is replaced by the previously fetched scan line. This error in the control signal continues for a while and the same is manifested as a repeated set of horizontal content lines. The artefact is shown in Figure 5.


Blotches occur due to presence of dirt/sparkle on the surface of a magnetic tape. Dirt/blotches disrupt the reception of signals during video data capture. The area for which the data is not received, appears as white or black spots. Snapshot of the video frame with blotches is shown in Figure 6.


Scratches appear in the video frame due to removal of oxide on the tape surface. The loss of oxide is due to wear and tear after prolonged or continuous usage of a tape. Generally, these scratches are in the form of thick horizontal line with some break at the boundary. The artefact is shown in Figure 7 below.

Chroma Phase Error

Composite video signals consist of chrominance components combined with luminance component using the phase modulation method. Any deviation in the phase will affect all the constituent components. With the phase error, the hue and saturation for the pixel colors may change and this will result into deviation of the colors from its natural values e.g., skin color, natural color of leaves or flowers sky etc. One of such examples is shown in Figure 8.

Dot Crawl / Rainbow Artefact

While capturing from a tape using composite signals, sometimes luma can be misinterpreted as chroma or vice versa. If chroma is treated as luma, the resulting artefact is termed as Dot Crawl. On the other hand, if luma is treated as chroma, the resulting artefact is termed as Rainbow Artefact.


The ghosting artefact is perception of weak shadows around the edges of the primary visible objects within a scene. It happens due to transfer of magnetic signals across the adjacent tapes. A snapshot frame for this error is shown in Figure 9.

Apart from the above listed set of errors, other errors may also get introduced while capturing color values corresponding to each of the pixel location in a frame. In some cases, values are not retrieved at all; localized patches are created abruptly within the content. If the captured values are different from its natural value, video signal level and out-of-gamut errors are introduced in the captured video sequence. Apart from these errors, different kinds of noise or noise patterns can be perceived due to noise introduced while capturing analog signals.

Fortunately, there are processes and tools to correct not all but some of the errors introduced after analog to digital conversion. These tools or processes consider specialized steps to correct the tape device or the conversion process itself. There are post-processing tools as well to remove any noise in the digitized content, to correct the hue / saturation / balance / contrast of colors etc. But before applying any such correction step, it is required to know if there is an error and what type of error it is. The knowledge about the type of errors will help in selection of the correction steps to be followed.

Similar to videos, audio samples too are stored as voltage signals on magnetic tapes. Any aberration while capturing the audio signal during the digitization process can lead to audio distortion of different types as discussed below. 

Audio Click/Audio Crackle/Transient Noise

Click/Crackle/Transient Noise/ Glitches are introduced due to scratches and dust on the surface of a tape. These are localized degradation that only affect certain groups of samples and thus cause a discontinuity in the waveform.

Scratches lead to disrupted audio samples during of the digitization process. These are perceived as ticking/popping/scratchy kind of noise lasting for a very small duration.

Audio Dropout

Audio Dropout is defined as distortion in audio signals in which silent frames of small duration (from 4ms to 300ms) are introduced in midst of normal audio data. It is characterized by abrupt fall in the signal level at the event of audio drop and abrupt rise at the end of audio drop frame.

Audio Dropouts are mostly introduced during digitization due to damage appearing on the tape. If a certain part of the tape is damaged, it won’t be possible for the head to read the corresponding audio data resulting in audio loss for that specific duration.

In addition to the above defects, the digitization process can also cause Audio Clipping. Because of dust and dirt contamination, it is possible that voltages become so high that it causes few of the audio samples to go above the legal range of 0 dB.   

For detection of audio defects, checks like loudness checks, audio dropout, audio clipping, checking for different type of audio noises are very common during the quality checking process.

If closed caption and burnt-in subtitles are present in the content, advanced quality checking tool will not only check for their consistency and dropout, but it will also make sure they are present in safe area of the screen.

Unlike the errors in compressed digital data, the errors in analog medium are difficult to model. The analog data errors are random and do not follow a known pattern. This is also due to variations in the conversion processes or varieties of electro-mechanical components used inside tapes. Because of this uncertainty, detection of these artefacts is quite tedious. Highly specialized image and video processing concepts and algorithms are required for accurate and reliable detection of errors in the digitized data. 

The Challenges Faced by Auto QC Tools

Selection of a correct auto QC tool for digitization is not only critical, but it has direct impact on the quality of the digitized content. A good auto QC tool can make the digitization process more efficient by detecting issues accurately and reliably. Algorithms to detect analog errors are more complex than that of digital errors. The detection algorithms need to consider and model various kind of non-linear processes followed during analog to digital conversion.  Error detection algorithms have been developed for detecting some but research is continuing for the difficult ones where it is complex to model the actual error context. The auto QC tool you deploy in your workflow will provide benefits which are only as good as the depth and accuracy with which it detects such analog dropouts. Some QC tools just do a lip service in the name of detecting such issues, and it is advised that a proper tool be selected after due testing of the results. Fortunately, there are some industry grade QC tools, with in-depth video dropout detections available now. These tools have exhaustive checks dedicated to the analog tape ingest process and have been successfully deployed at large archiving projects.

The digitization of tapes and archiving to digital formats is a necessity to complete the transition to the file-based digital workflow. During this process, it is critical to use the right set of tools to ensure the quality of the content being archived. Artefacts can manifest in multiple ways in these tapes and need to be detected. Detection of these artefacts called analog dropouts is complex and several deep algorithms have been developed for the same. While a lot more research needs to be done to cover a larger set of analog dropouts, using the right auto QC tool during the archiving process helps detect these complex analog errors more accurately and reliably, and enables you to preserve and deliver high quality of the generated content. 

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6 Misapprehensions of MAM

Implementing a media management system is considered to be a risky business. Like national infrastructure programmes, the perceived complexity, upheaval and perception of failure can be so daunting that the risk may seem to outweigh the reward. But, like so many business technologies, MAM have been transformed in recent years with the advent of open standards and interfaces.

It’s time to debunk some common MAM myths.

Perception: MAM implementation projects cost millions

Truth: MAM software is available from as little as £10k

It’s not surprising that MAM projects are perceived as expensive – before its collapse the notorious BBC Digital Media Initiative had racked up a £96m bill and even comparatively ‘small’ projects at a large facility or national broadcaster in recent years have averaged a few million. For most operators those figures are unimaginable. But a pragmatic approach to scope and implementation can deliver real value at low cost. The principal enabler here is the cost of the MAM software.

Rather than starting at £1m as they did ten years ago the cost for modern product has dropped to around £150k for a base system – with some tools available for as little as £10k. Cloud or hybrid service models make it possible to pay-as-you-go for a fraction of the cost and without long term commitment.

For on-prem’ deployed systems the cost reduction is largely as a result of use of open, service based technology as well as vendors designing the foundation tools to be less specific to a given function or workflow. Modern MAMs are architected on the basis of a core which is functionally augmented with bolt-ons or plug-ins.

Perception : MAM’s rarely deliver value on investment

Truth : A targeted approach delivers immediate benefits

The return on an investment of millions will be long – often too long to make a strong business case, but now that MAM system implementations can be achieved for a fraction of the cost, it’s easier to prove real value in day-to-day operations and to the bottom-line.

With relatively inexpensive entry points, operators can focus on a specific difficulty in the operation and prove or refine an approach before tackling other areas in the workflow. Defining and building workflows is no longer time consuming so demonstrating operational benefit may be achieved in hours instead of months.

An important caveat here is scalability though – do ensure that the chosen MAM vendor can scale as you broaden deployment – a modern MAM should readily scale from a few, to a few hundred users without even taking the system off-line.

Perception : Implementation of a MAM system takes months – if not years

Truth : You can be up and running in minutes

MAM history is littered with projects that took years to implement and were obsolete by the time they were delivered. This was generally caused by an over-focus on up-front analysis driven by the need to understand the operating model ahead of configuration which itself was driven by the complex and time-consuming nature of configuration.

A modern MAM can be up and running in minutes – or seconds in the case of cloud-based system. From the ‘vanilla’ base an operator can quickly create users, set access permissions and build simple production workflows for transcode and delivery or review and approve. A modern MAM allows operators to build and test workflows as they go – an agile approach which delivers rapid results. Because it is now quick and easy to configure and implement a base system, and then customise the system to your needs, there is less pressure to perform a perfect up-front analysis, and more tolerance for changing requirements.

Some products such as the Blue Lucy Asset Manager (BLAM,) have built-in tools that deliver immediate benefits.For example; BLAM will index existing storage pools and create browse clips without moving the original media and while maintaining project hierarchy. This tool means that projects can automatically be brought under management of the MAM without any user intervention or interruption to operations.

Perception: Deploying a new MAM means replacing the systems we have.

Truth : An effective MAM will integrate with legacy and 3rd party systems.

Many MAM projects are initiated on the premise of replacing one or more existing systems. The narrative runs “… we can replace these three systems with this one new system. Operations will be simpler, delivery will be faster, and support will be easier……” This, all too common, one big new universal panacea system mantra rarely (never) actually delivers. Subsuming functionality, particularly in active systems is fraught with risk – risk which always translates to cost. The simplest and most cost effective approach is rarely replacement.

A modern MAM vendor should be able to provide connectors to commonly used 3rd party systems and be willing to build these for legacy components. These connectors are used to get essential data from 3rd party systems, process it and if necessary, put data back. The MAM should be capable of acting as the integration layer that connects systems and, importantly, provides a single view of the operation.

This is far less expensive and risky than incorporating functionality in a new system. It is also likely that specialist systems provide a better, or more specific, capability than one system trying to do everything. Do be wary of vendors who are reluctant to integrate with third parties or boast that their systems can absorb operational capability.

Perception: MAMs only manage media.

Truth : Modern MAMs provide broader capabilities including workflow orchestration and enterprise reporting.

Very few modern MAMs are just asset managers, but perversely the term MAM has stuck because it’s one that everyone understands. Rather than just being limited to a static repository where media objects are stored and made accessible, many MAMs include key operational functions such as automatic and task-driven workflows. The automation of tasks such as transcoding in a workflow obviously removes manual process but of equal benefit is a task driven workflow which drives operation procedure – i.e. the people.

More sophisticated systems (like BLAM) can provide data and intelligence on a broadcast / media operation, enabling workflows to be adjusted for maximum efficiency. Efficiency for a content producer translates to saving money in the production or distribution process or reaching wider audiences on an ever increasing number of consumer platforms.

Perception: MAM projects require big investments in infrastructure and storage

Truth: Modern MAM’s use the cloud for scalability

Although the cost of MAM software is largely flat (or at least should be,) the cost of storage is dependent on the volume of media being produced and stored. When planning on-prem’ deployments, system designers – understandably – plan for the worst case scenario, which means high storage costs and big CAPEX spend.

Using a combination of on-premise and cloud storage allows media operators to avoid spending money up-front for capacity that they may or may not need in the future. MAM vendors like Blue Lucy are able to seamlessly blend the on-prem’ and cloud-based services, providing access through a single user experience. This approach offers huge flexibility for ‘burst’ capacity and de-risks a strategy of cloud migration.

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How to diagnose faults in a hybrid IP/SDI video network

The most commonly cited advantage of deploying IP video networks in production and other operational applications is the ability to use commercial off-the-shelf (COTS) IT-based infrastructure, which takes advantage of the economies of scale of the IT industry when compared with the relatively small broadcast industry. Additional advantages of reduced cabling cost and weight along with the much greater routing flexibility that offers more flexible production options. These advantages have captured the industry by storm and broadcasters are already working on early deployments of IP video networks. Not far behind deployment is the need to efficiently diagnose and resolve faults.

IP introduces new technical and skills challenges. These include jitter, latency and the risk of dropped packets and network asymmetry that results in different path delays upstream and downstream.

Deploying IP for video production applications is effectively the collision of the two worlds of video and network engineering. Video engineers are comfortable with the use of SDI, coaxial cable, patch panels, black burst and tri-level sync for timing and above all, monitoring signal quality. The challenge for the video engineer is to understand IT technologies and impact of an IT infrastructure on the video.

On the other hand, network engineers are familiar and comfortable with, IP flows, protocols, network traffic, router configuration, Precision Time Protocol (PTP) and Network Time Protocol (NTP) for timing. The biggest difference however is that in most data center applications, lost data can be re-sent – this is not the case with high bitrate video. The challenge for the network engineer is in understanding video technology and its impact on IT infrastructure.

Overcoming jitter

In any digital system, jitter is any deviation from the regular periodicity of the signal. In IP networks jitter is the variation of the packet arrival interval at a receiver. If the network routers and switches are all configured and operating correctly, the most common cause of jitter is network congestion at router/switcher interfaces.  

The application within a network element will likely require the data to be received in a non-bursty form and as a result, receiving devices adopt a de-jitter buffer, with the application receiving the packets from the output of this buffer rather than directly. As illustrated in Figure 1, packets flow out of the buffer at a regular rate, smoothing out the variations in the timing of the packets flowing into the buffer.

Figure 1. Packet jitter is deviation from the periodicity of the packet arrival interval.

The rate of packets flowing out of the de-jitter buffer is known as the “drain rate.” The rate at which the buffer receives data is known as the “fill rate.” If the buffer size is too small then if the drain rate exceeds the fill rate, then the buffer will eventually underflow, resulting in stalled packet flow. If the sink rate exceeds the drain rate, then eventually the buffer will overflow, resulting in packet loss. However, if the buffer size is too large, then the network element will introduce excessive latency. 

Jitter can be measured by plotting the time-stamps of the packet inter-arrival times versus time as shown in Figure 2.

Figure 2. Packet inter-arrival intervals plotted versus time.

This is useful to identify variances in jitter over time, but it is also useful to be able to plot the distribution of inter-arrival intervals vs. frequency of occurrence as a histogram. If the jitter value is so large that it causes packets to be received out of the range of the de-jitter buffer, then the out-of-range packets are dropped.  Being able to identify outliers such as the example in Figure 3 is an aid in identifying if the network jitter performance is either likely to or already the cause of packet loss.

Figure 3. Packet inter-arrival intervals plotted versus frequency of occurrence.

A series of packets with long inter-arrival intervals, will inevitably result in a corresponding burst of packets with short inter-arrival intervals. It is this burst of traffic, that can result in buffer overflow conditions and lost packets. This occurs if the sink rate exceeds the drain rate for a period of time that exceeds the length of the remaining buffer size, when represented in microseconds.

Establishing de-jitter buffer size

To establish the necessary de-jitter buffer size, an alternative form of jitter measurement known as delay factor (DF) is used. Delay factor is a temporal measurement indicating the temporal buffer size necessary to de-jitter the traffic.

In IP video networks, the media payload is transported over RTP (real-time protocol). One form of DF measurement takes advantage of the fact that the RTP header carries time-stamp information that reflects the sampling instant of the RTP data packet. This is known as time-stamped delay factor or TS-DF (as defined by EBU Tech 3337) and represents temporal buffer size in microseconds as shown in Figure 4.

Figure 4. TS-DF represents temporal buffer size in microseconds.

The TS-DF measurement is based on the relative transit time, which is the difference between a packet’s RTP timestamp and the receiver’s clock at the time of arrival, measured in microseconds. The measurement period is 1 second, with the first packet at the start of the measurement period being considered to have no jitter and is used as a reference packet.

For each subsequent packet, the relative transit time between this packet and the reference packet is calculated and at the end of the measurement period, the maximum and minimum values are extracted and the Time-Stamped Delay Factor is calculated as:

TS-DF = D(Max) – D(Min)

Finding root causes

To establish root causes, it is necessary to understand whether visible impairments are being caused by IP errors or if some other fault is causing the impairment. Figure 5 shows how a network monitoring tool can be used to track time-correlated video and IP errors. This is made possible by correlating the time stamps of the video errors and the RTP packet errors.

Figure 5. Time-correlated video and IP errors.

A video CRC error does not in itself confirm that the video is impaired making it desirable to use traditional monitoring methods such as picture and waveform displays as well as audio bars.

Figure 6. Traditional video monitoring and audio bars are useful in confirming errors.

Tracking PTP errors

Device clocks in IP video networks have no inherent concept of system time, so precision time protocol (PTP) is used to synchronise these clocks. The most recent version is IEEE 1588-2008, also known as PTP version 2 with the SMPTE ST 2059 PTP profile being specifically intended for broadcast applications. 

In effect, PTP provides genlock functionality equivalent to that delivered by black burst or tri-level sync in SDI networks. The overall PTP network time server is referred to as a PTP grandmaster, with devices that derive their time from PTP being referred to as PTP Slaves. PTP grandmasters are usually synchronized to GPS, GLONASS or both.

For the foreseeable future, many video networks will use a combination of SDI and IP. To allow frame accurate switching between SDI and IP derived content, it is essential that the timing of the Black Burst/Tri-Level Sync is not offset relative to the PTP clock.

This is achieved by measuring the timing offset between the PTP clock and BB/Tri-Level Sync and then making any necessary correction by skewing the SDI syncs with reference to the PTP clock.

Figure 7. Measuring the time relationship between BB/Tri-Level and PTP.

A final consideration

In live production applications, network experts may not be present on the production site and networking equipment also may not necessarily be in a location that is easily accessible. It is desirable that network and video engineers can control any diagnostic equipment remotely.

Figure 8. Remote control ability enables on-location access to network expertise

An all IP infrastructure is the vision for most broadcasters around the world and is already starting to happen in many facilities. The reality is however that the transition won’t happen overnight leading to the need to manage hybrid SDI and IP infrastructures, and thus a need for IP and video engineers to work closely together to ensure seamless operation and quickly track down faults.

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