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Steve Zaretsky I T: 1 (212) 315-1111 ext. 15 I [email protected] Coast -

George Horton I T: 1 (213) 249-9229 ext. 15 I [email protected]

Broadcast Audio. This is S51.Solid State LogicSOUND I I VISION

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TABLE OF CONTENTSVOLUME 53 I NUMBER 9 I SEPTEMBER 2011

BroadcastEngineering

FEATURES40 SPECIAL REPORT:

DIGITAL AUDIO SYSTEMS44 MN does more with less

Engineers with MTV Networks had six monthsto revamp a studio overlooking Times Square.

52 New loudness metering standardWith the revisions to BS.1770, a new test suitewas needed to test meters for conformity.

60 Fixed-point vs. floating-pointcalculations in audio processingA case can be made that fixed-pointnotation can yield improved resolution.

64 Processing audio for mobile DTVMobile DTV devices present newchallenges for audio mixing engineers.

67 Adaptive, automated loudness controlMultiloop technology provides a way tointegrate loudness control with automation.

BEYOND THE HEADLINESDOWNLOAD

12 Next -generation captioning sytemsCaptioning provides a way to reuse andresell content without costly dubbing.

FCC UPDATE18 Analog LPTV ends in 2015

Channels 52-69 must cease LPTVoperation by the end of the year.

continued on page 6

JUST THE FACTS!According to a new report for Euroconsult, the fixedsatellite sector grew both in terms of transponderdemand, which was up 4.4 percent, and revenue,which reached $10.8 billion in 2010.

Learn more at www.broadcastengineering.com.

4 broadcastengineering.com I September 2011

In a studio overlockirq Times Square, MTV ergineersmade the conve-sion fr3m SD to HD in half as much spaceas :he previous ;tud o.

That's a bold statement, but a truE one As one of the world's largest brcadcast and professional solutionssuppliers, Grass Valley' has over 3,000 active 'broadcast' customers, and ters of thousands of professionalusers generating content using Gass Valley tools. For more than 50 yeas, 3rass Valley has been, andcontinues to be, at the forefront of on -air innovation, creating some of t -e most accomplished productsand services available. When you're watching news, sports, or entertainment programming, whether ona TV, the web, or a mobile phone, you're watching Grass Valley at work.

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TABLE OF CONTENTS irNTINUED)VOLUME 53 'NUMBER 9 1SEPTEMBER 2011

DIGITAL HANDBOOKo RANSITION TO DIGITAL

20 Digital video processingGain a greater insight into the system byexamining how the parts work together.

COMPUTERS & NETWORKS24 Navigating the uncharted cloud

Cloud computing brings opportunities andchallenges to professional media companies.

PRODUCTION ROOM28 Nonlinear editors

Here's a look at some of the technicaldistinctions of today's NLE systems.

SYSTEMS INTEGRATIONNEW MEDIA NL TWORKS

32 MultiviewersIncrease your understanding of this importantpiece of broadcasting equipment.

NEW PRODUCTS & REVIEWSTECHNOLOGY IN TRANSITION

72 IntercomsToday's intercom systems are high-tech.

NEW PRODUCTS74 Miranda's Kaleido-XQUAD

multiviewer and more ...

DEPARTMENTS8 EDITORIAL

79 CLASSIFIEDS81 ADVERTISERS INDEX82 EOM

SEE IT ONLINE!Learn from the experts with Broadcast Engineering'sSpecialized Training eBooks and video tutorials.

Learn more at: www.broadcastengineering.com/specialized -training/.

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6 broadcastengineering.com 1September 2011

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EDITORIALDEPARTMENT

Tethered users andtoilet tablets

Alate August e-mail subject line captured myattention. While the e-mail appeared to havelittle to directly impact OTA broadcasting, thetitle could not be ignored: "Owners take their

tablets to the restroom." As curiosity got the better of me,I clicked to read the entire press release.

It seems that Staples Advantage, a division of theStaples office supply company, conducted a survey to seehow tablet owners were using their portable devices. Thestudy asked 200 owners of the computers, across multiplecompanies and industries, about their use of the technol-ogy. The study revealed that 60 percent of tablet usersadmitted to taking their devices on vacation, and almost

80 percent said they use them in bed. A third of tabletowners said they take them to restaurants. Okay, I can be-lieve these numbers, but it was the next study data pointthat caused me to pause.

The study showed that more than a third of tablet users,35 percent, take their devices into the restroom. Clearly,one of the key advantages of these relatively new devicesis their portability. One often sees tablets being used inrestaurants and other non -office spaces. Combined withWi-Fi, they can be an effective communication tool. Andfor parents, tablet games help address the issue of fidgetykids when dining out or traveling.

If broadcasters have their way, viewers soon will beable to connect USB receivers to tablets and enjoy OTADTV while away from the home and even on -the -go.

After all, connection -free OTA is one of broadcasters' sev-eral advantages over cable.

Tablets also enable a wide range of both business andgaming applications to be carried out on portable andlarger screens. No external or folding keyboard is required;owners can type right on the screen. On the surface, a tab-let solution seems ubiquitous and perfectly suited for to-day's busy and fast -paced executive.

But to me, it just seems creepy that some Americans havebecome so tethered to technology, they cannot even use therestroom without being tethered to an electronic gadget.

A telephone survey asked adults if they always washedtheir hands when in public restrooms. While 96 percentclaimed they did, actual observations indicate otherwise.

A survey from the American Society for Microbiologyand the American Cleaning Institute reported that in re-stroom observations, 93 percent of women washed theirhands in public restrooms, but only 77 percent of men didso. But that's an improvement over the results from 2007,when only 66 percent of men did.

I've observed some of those non -washers in our com-pany restroom. Call me a phobic, but I even employ thepaper towels used to dry my hands to open the restroomdoor. If nothing else, such hygiene can help prevent thespread of colds and flu.

All this leads me to regard the use of any tablet com-puter, which may have been close to a recent flush, at adinner or restaurant table less than appetizing.

The next time someone wants to share something withyou on their tablet, let them hold it. Or, ask them where it'srecently been.

EDITORIAL DIRECTORSend comments to: [email protected]

BE


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DOWNLOADBEYONDTHE HEADLINES

Captioning's futureAs platforms continue to expand, dubbing's alternative

is an economic means to use and resell assets.

Generally, there are tworeasons for captioningcontent: for accessibil-ity - both for the hear-

ing impaired and for when audio isunavailable (e.g. within public spacessuch as at airports, gymnasiums, etc.)- and for language translation to en-able access to content by foreign lan-guage audiences.

Particularly in the digital television,multichannel world, an increasingamount of content is reused acrossregions. Captioning provides an eco-nomical means to allow broadcastersand content providers to reuse andresell their assets across multiple plat-forms. (See Figure 1.)

Dubbing may make sense in someregions. However, in terms of workflow,cost and multi -language access, it doesnot provide the optimum solution.

The alternative to dubbing -captioning - provides a number ofadvantages that make it a highly at-tractive solution for reaching the

BY FD HUMPHRFY

Accessil)files

Audiodescription

Video signing

Captioningand subtitling

Internet

AFlash: e.g. DFXPReal: e.g. RTSilverlight: applicationQuicklime: e.q. QTWindows media: e.g. SAMI

0 n0

3Open burn -in: e.g. 032, CG2, XIFFlash: e.g. DFXPQuicklime: e.g. QTWindows media: e.g. SAMI

e evision

US SD -CC CEA608: e.g. CAP, NCI CAP, FLCalUS HD -CC EIA708: e.g. CAP, NCI CAP, FLCWST teletext e.g. 890, STL, PAC, RAC,132, X32DVB text e.g. 890, STL, PAC, RAC, T32, X32BD/DVD text: e.g. CG2, XIF, 032, Xew

DVD/Bluerey

BD/DVD bitmap: e.g. SON, SPL, XIF, X32BD/DVD text: e.g. SCC XIF, L32

CiremaDigital cinema: e.g. DTS, DCI

Figure 1. This chart shows the wide rangesubtitling can be deployed.

greatest number of viewers. The re-duced cost makes it more viable fora greater range of content. It is easierand more practical to offer captions

FRAM E GRAB A look at the issues driving today's technology

Looking ahead, traditional pay TV still topsOTT products continue to evolve and expand, but projectedrevenues are still far behind.

$250,000,000,000

$200,000,000,000

$150,000,000,000

$100,000,000,000

$50,000,000,000

$0

Source: IMS Research, August 2011

Pay TVsubscription

revenue

OTTrevenue

OTT VODrevenue

www.imsresearch.com

of platforms to which captioning and

using efficient state-of-the-art work-flows, especially for content that islive -to -air or has completed produc-tion very close to time -of -air. And,captions provide the greatest acces-sibility not just to the main programcontent, but also to advertisements,which widens the demographic servedby the advertiser. Equally, as notedabove, captions allow for content tobe accessible in public spaces.

Multi -language captioningIn addition to the necessity to com-

ply with legislation in the target loca-tions, captions provide an inexpensiveway to open up content to the widerlocal and international audiencesthrough language translation.

Depending on the format of cap-tioning used, it is possible for multiplelanguages to be delivered to viewerswho then can choose their preferredlanguage. In formats such as CEA-708, WST Teletext and DVB, it is also


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possible to switch on and off captionsvia remote control.

Captions also provide a rich sourceof content -relevant metadata for thevideo asset. Enhancing the asset withthis additional data aids repurposing,as well as enables detailed search-ing and indexing of content. Withintegration into centralized mediaasset management systems, contentcan also be more easily monetizedthrough clip resale, redistributionand syndication.

Evolving production andbroadcast workflows

To aid format and resolution con-versions for diverse distribution for-mats, many broadcasters want to storevideo assets as a single common mez-zanine format. This represents thehighest quality version. Thereafter, allsubsequent broadcast and streamingversions will be derived from it.

To optimize repurposing, the stor-age of caption data should align withthis principle and be stored as a high-level generic form of caption data.With this approach, there are two keymethodologies to consider. One relieson the creation of a master caption,which has as much information aspossible related to it. This allows lesssophisticated derivatives to be readilyproduced. In effect, this becomes themezzanine format caption and relieson informed choices being made dur-ing the creation/preparation phasefor presentational aspects such asfont, color, positional and alignmentinformation, drop shadow, and char-acter edging.

Alternatively, there is the transcodeapproach, which relies on a lowestcommon denominator format file(such as an CEA-608 compliant cap-tion or Teletext caption) being cre-ated and effectively upconverted tothe target format. This will optimizeease of use and speed, but does nottake advantage of the sophisticatedoptions available within higher -endstandards (such as CEA-708, DVBETS-300-743 and DVD Bitmap).

Many broadcasters choose to create

Creation

Caption and subtitlecreation andrepurposing

Early

Ingest to DAMS ormedia server systems

>>>>>>>>>>>>

Late

Either ingest to DAMS/media server systems,or send for live playout

Live

Live input gatekeeper Live playout of files Live transcode of data Capture to file for

replay and repurposing

Time to playout

Figure 2. Once captions have been created, they must be bound to content.Options have increased as encoding technologies have grown to faster -than -real-time capabilities.

a hybrid of the two. They implementsome of the capabilities while lim-iting the overall time dedicated tocreating the caption by constrainingand automating some choices. Whenconsidering which approach to take,it is worth noting the variations thatdifferent standards offer in terms oflevels of control and sophistication.

Captioning forfile -based workflows

The caption creation technologiesmost commonly in use fully supportnonlinear video - thereby allowingcontent providers to digitally sendencrypted and/or watermarked videoclips. This saves costs associated withtape -based methods.

The best creation systems allow forfar greater caption operator productiv-ity through technology enhancementssuch as shot change detection, semi -

automated time -coding functions,advanced reading speed algorithmsand so on. Modern caption authoringsystems should also be Unicode-capa-ble, in order to create and repurposecaptions in almost any language. Forcomplete flexibility, they also shouldfully cater for HD, VOD, Web, mobile,digital cinema and the burgeoning ste-reoscopic 3-D space.

The employment of a wide networkof caption operators makes expediteddelivery ever more possible. Often,these operators are located aroundthe globe in order to maximize timezone benefits. As a result of this highlydistributed mode of production, a

substantial worldwide network of pro-fessional freelance caption operatorshas emerged, equipped with the latestcreation workstations and using high-speed broadband links to receive se-cured video clips and job instructions.

Finally, creation is sped -up throughcaption agencies sending proofs backto their clients electronically. This isachieved by generating an all -digitalapproval, which has captions overlaidto video so that the client can quicklyand easily assess placement, timing,font choice and other factors.

Binding captionsto content

After the creation phase has beencompleted, the caption or subtitle datamust then be bound to the content,enabling presentation to the viewerwhen they watch the programming.

This binding can be considered asoccurring in one of three periods oftime, as illustrated in Figure 2:

Early binding. The pre -prepared fileis linked to the program content wellahead of transmission.

Late binding. Similar to early bind-ing, but occurs closer to air time andonly becomes possible due to faster-

than -real-time encoding technologies.Live binding. For either truly live

content or for pre -prepared contentthat only becomes available very closeto airing, thereby eliminating the pos-sibility of pre -binding captions.

In modern workflows, files are ei-ther sent for time -of -air transmission(a live bind), or are transcoded into a


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file -based video asset (during early orlate binding).

Driving time -of -air transmissionwill be a system that tightly inte-grates into the automated workflowof a master -control facility - withthe caption playout system approv-ing files in advance of airing and thenairing the correct file at the right timeautomatically, either with or withoutexternal time -code.

The time -of -air system can alsobe used as a gate keeper for real-timecaptioning, where the system au-thenticates the caption operators andtheir work slot, prior to allowing passthrough to air.

Additionally, the time -of -air sys-tem can extend the control of theautomation system over the live cap-tioning by switching the data sourceto the in -line caption encoder basedon the automation schedule. Thisremoves the dependency on the livecaption operator to remember tomanually switch control at the righttimes. This prevents situations suchas open connections blocking playoutof pre -prepared content during com-mercial breaks.

Increasingly, a hybrid of ingest andtime -of -air methods is becomingthe workflow of choice, resulting ina system that intelligently arbitrates

between ingest to video servers when-ever possible and time -of -air playoutas appropriate. In this role, the time -of -air playout system is elevated to acentral caption management platform.

The time -of -air caption system canalso provide interfaces to other ancil-lary data signals and XDS informa-tion such as wide-screen signaling,vChip parental controls, BroadcastFlag information, DRM controls suchas CGMS-A data, Digital Program In-sertion (DPI) data, etc.

Transcoding captions forflexible asset management

As well as supporting different out-put distribution formats, modern cap-tioning solutions support reversioningof video assets. This occurs when anasset is manipulated in the time do-main or split into different programsegments. The process often takesplace within an NLE and can effec-tively destroy the captioning data as itbecomes disassociated with the videoand audio content. Modern transcod-ing solutions can circumvent this issueby using the edit decision list from theNLE (and other sources of data de-scribing the differences between theoriginal and the derived version of thevideo), bridging the caption data fromthe original to target version.

By using the mezzanine format forvideo and caption data described ear-lier, the caption data component canbe transcoded appropriately at thesame time as the video, ensuring thesame quality or better of captions asfor the broadcast version. The samealso applies for situations where thelowest common denominator mezza-nine route has been followed. It maynot provide better presentation, butit still provides the other benefits ofeasier repurposing and greater effi-ciency through predictable output.

Seamless integration innext generation workflows

Any platform for caption play-

out and management must providethe stability needed to ensure confi-dence in data making it to the viewer.Therefore, it is vital to seek out tech-nology providers who have the ex-pertise and experience in developingand delivering the specialist systemsrequired, as well as those who haveproven their worth in the field. Usingthis expertise, it becomes possible toimplement highly fault -tolerant so-lutions that integrate seamlessly intomodern workflows. BE

Ed Humphrey is president of theAmericas, Softel.

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FCC UPDATEBEYONDTHE HEADLINES

Analog LPTV end nearingSome channels will cease LPTV operation this year.

BY HARRY C. MARTIN

In July, the FCC released its long-awaited decision announcingthat Sept. 1, 2015, will be thedrop -dead date for analog Class

A TV, LPTV and TV translator opera-tions (together, "LPTV"). However,analog LPTV operations on Channels52-69 must cease by Dec. 31, 2011,regardless of whether the licensee hasbeen able to find an available lowerchannel. But, anyone with a compan-ion channel or flash -cut CP has un-til Sept. 1, 2015, to convert to digital.Here are some of the implications ofthe FCC's decision:

Repacking. The FCC anticipatesthat a four-year transition periodending on September 1, 2015, shouldprovide adequate time for LPTV sta-tions to accommodate themselves tochannel "repacking?' The commissionhas embraced the notion of repack-ing the TV spectrum to make morespace for broadband use by repurpos-ing up to 20 broadcast TV channelsand either relocating affected stationsto other channels or partial channels,or by having the stations relinquishtheir frequencies through "incentiveauctions" -a plan now being consid-ered in Congress. (See "FCC Update"in Broadcast Engineering's July issue.)The repacking process, if it moves

DatelineNoncommercial TV stations in

Iowa and Missouri must file theirbiennial ownership reports on orbefore October 1, 2011.

By October 1, TV and Class ATV stations in the following loca-tions must place their 2011 EEO

reports in their public files and postthem on their websites: Alaska,Florida, Hawaii, Iowa, Missouri,Oregon, Puerto Rico, Virgin Islands,Washington and the Pacific Islands.

forward as the FCC wants, is certain toaffect a sizable number of full -powerDTV allotments, which would in turnshrink the spectrum available for digi-tal LPTV.

Federal money for the transition.The FCC has encouraged NTIA to askCongress to extend the existing pro-gram for reimbursing LPTV digitaltransition costs. Some $30 million re-main in unspent funds in that program.The FCC does not discuss either: (a)the program's statutorily -mandatedeligibility criteria, which strongly favorthe most rural stations and complete-ly exclude urban stations; or (b) theprogram's dollar limits of $6000 and$20,000, neither of which fully coversconversion costs.

Procedural relief The FCC has au-tomatically extended all currentlyoutstanding digital CPs for flash cuts(i.e., on -channel conversions to digi-tal) or digital companion channelsfor existing analog stations. No mat-ter when those permits were issued orhow many extensions were previouslyrequested, all of these have now beenextended to Sept. 1, 2015. Permitteeswho can show extraordinary unfore-seeable circumstances will be eligiblefor extensions until March 1, 2016.Construction permits for new stationshave not been similarly extended; theyare good only for their original three-year terms.

Notices to the public. Migrating li-censees must broadcast announce-ments 30 days before terminatinganalog operation if they have programorigination capability. Stations lackingsuch capability must find another wayto notify the public - e.g., newspapernotices.

"Minor" change redefined. LPTVstations applying for displacement toa new channel are restricted to a 30michange in transmitter sites. Otherchanges may exceed that distance and

still be classified as "minor" as long asthere is not any overlap of licensed andproposed protected service contours.The FCC will now impose the 30milimit on all minor changes in additionto the contour overlap requirement.

Use of VHF channels. The FCC'sspectrum repacking proposal mayinvolve moving LPTV stations toavailable VHF channels. To enhancethe attractiveness of VHF, the FCChas increased the LPTV power limitto 3000W, on all VHF channels.

Class A gets option to choose chan-nels. Class A stations with both ana-log and digital operations will now befree to elect either of their channelsfor permanent digital operation. Theycan apply for a construction permit toflash -cut their analog channel or maymigrate their Class A protected sta-tus to their digital channel without aCP by filing a Form 302 -CA licenseapplication.

Fees for ancillary digital services.Both full -power and LPTV licensedstations that provide non -broadcastdigital data services in addition tovideo program streams are requiredto file Form 317 each December andpay 5 percent of their gross ancillaryservices revenues to the government.Digital LPTV stations operating underan STA without a license have been ex-empt, but no longer. If they provide nosuch ancillary services, they may say soand pay nothing. BE

Harry C. Martin is a member of Fletcher,Heald and Hildreth, PLC.

rilSend questions and comments to:[email protected]

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TRANSITION TO DIGITALDIGITAL HANDBOOK

Digital video processingGain insight into how the parts work together.

In this column earlier this year,we mentioned "light -to -light,"the concept of an overall trans-fer function describing a com-

plete camera -to -display video sys-tem. We then zeroed in on samplingand reconstruction, two elements ofthis cascade of signal processing ele-ments. This month, we'll look at thisconcept from a holistic perspective,to gain insight as to how the variouselements interrelate.

Characterization ofvideo systems

In systems analysis, the behaviorof a system with an input and out-put can be characterized by severalmeans. With audio systems, it is cus-tomary to describe the frequency re-sponse of a system, i.e., the gain ofthe system from input to output as afunction of the frequency of the sig-nal applied to the input. In mathe-matical terms, this gain is a complexfunction that can be evaluated interms of magnitude and phase, and

BY ALDO CUGNINI

Imagersystem

Lenssystemsystem

Decompression

Digitizer

Storage/transmissicn

Colordecoder

4 -

Colorencoder

Bit -streamprocessing

Proc

Proc

1

Compression

D/A

Displaysystem

Figure 1. Camera -to -display video processing elements.

in this form is generally called thetransfer function of the system.

With video systems, a similar con-cept applies, wherein the input andoutput "signals" are comprised oftwo-dimensional (or more) images.In fact, as video is concerned withmoving images, the signal can be con-sidered as having three dimensions:vertical, horizontal and temporal.

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Breaking down the image into lumi-nance and chrominance componentsadds a "dimension" as well. The com-plete description of a video system,from input image to output (display)image can thus be characterized byone or more transfer functions, andwe are predominantly concerned withthe amplitude aspect of those.

Transfer functions can be linear ornonlinear, meaning that the input-out-put amplitude relationship at any onefrequency can be plotted as a straightline or as a curve, respectively.

When dealing with linear systems,transfer functions are commutative,i.e., cascading different systems willresult in an overall transfer functionthat does not change when the orderof the different systems is changed.But video systems tend to be nonlin-ear, as the transfer functions dependon many different characteristics ofthe images and often will vary overtime as well.

The typical elements of a completevideo system are shown in Figure 1.A number of these processes can takeplace simultaneously, i.e., in real time,or can operate at different points intime, such as upon retrieval from astorage medium. Also, some of the


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elements shown can be omitted orinterchanged in order; one example isa totally analog system, which wouldnot contain a digitizer, or compres-sion, or the inverse functions.

Lens systemWe start with the optics of the

video camera; the concept of trans-fer function applies to optics as wellas to electronics. In the optical do-main, we are generally concernedwith the optical transfer function,which includes magnitude andphase. Because phase is usually ofless interest when a wide spectrumof light is considered, with videosystems we are more often interestedin the modulation transfer function(MTF), which concerns amplitude

Modern codecsalso include loopfilters that reduceblocking artifacts

but may cause lossof subjective detailor objective fidelity.

only. The MTF describes the perfor-mance of an optical system as a func-tion of spatial frequency, measuredin cycles per unit angle, and providesa way to characterize the maximumspatial resolution of the system. Oth-er measures of the behavior of opticsinclude monochromatic and chro-matic aberration, which are spatialdistortions of the ability of an opti-cal system to form well -focused andcolor -registered images.

Imager systemState-of-the-art cameras use CCDs

or CMOS sensors to capture images.These devices inherently perform thesampling function, i.e., they converta continuously valued image into adiscrete spatially sampled representa-tion. While most professional cam-eras use three imagers and a color

separation system of optical prisms,other video cameras use a colorseparation Bayer filter that is incor-porated in a single imager. The per-formance of an imager is defined bymany characteristics, including maxi-mum resolution (in two axes), mini-mum sensitivity, signal-to-noise ratioand linearity.

DigitizerImagers usually do not digitize the

signal, i.e., convert the amplitude val-ues into quantized digital values. Thatfunction resides in the digitizer, alsocalled an analog -to -digital converter.(There are some imagers, however,that integrate the digitizer on thesame device.) The Nyquist Theoremstates that information signals mustbe sampled at a rate at least twice ashigh as the highest frequency compo-nent; otherwise, aliasing will result.The combination of imager and digi-tizer will determine the resolution,aliasing performance and pixel rate ofthe video.

Color encoding usually includeschrominance subsampling to reducesignal bandwidth, with contributionand distribution signals typically us-ing 4:2:2 sampling, and transmissionsignals using 4:2:0 sampling; this sub -

sampling will introduce its own alias-ing. Color space modification can alsobe integrated into this process.

Post processing consists of a num-ber of image transformation func-tions that provide special effects andimage improvement. The simplest ofthese are luminance and color cor-rection, which adjust for poor lev-els from cameras and make videoconsistent across content, and peak-ing, which applies a high-pass filter(both dimensions in a digital sys-tem) that accentuates image detail_Peaking alone can emphasize noisein the image, so noise reductionshould accompany its use, by intel-ligent coring and/or filtering. Thesetechniques lower noise by adaptive-ly quantizing and changing selectspatial frequency bands of the im-age, depending on the distribution

of luminance and chrominance val-ues. Format conversion is anotherprocessing element that involves aresampling of images in both thespatial and temporal dimensions,with subsequent changes to imageresolution and aliasing.

Compression is used to lower thetransmission bit rate and/or storagerequirement of digital video. Whilethe process can be lossless, i.e., theinput/output transfer function ap-proaches a "straight wire connection,"high compression factors are com-mensurate with image distortion,generally causing a number of knownartifacts, including blockiness andquantization noise. Modern codecslike MPEG4-AVC/H.264 also includeloop filters that reduce blocking arti-facts but may cause loss of subjectivedetail or objective fidelity (accuratepixel reconstruction). The effect of allof the preceding system elements onthe performance of the compressionsystem should not be underestimat-ed. Poor processing and interfacingof signals in both analog and digitalform can cause an increase in artifactsand a decrease in the efficiency of thecompression system.

Compressed bit streams can bemodified by numerous methods ofbit -stream processing, such as re -

coding for multiplexing, splicing toswitch program content and logoinsertion. Storage and transmission,which in the ideal are lossless pro-cesses, can nonetheless delete infor-mation through errors in the media.While storage is usually designed ina way that makes errors nonexistent,transmission errors can result in irre-trievable data corruption and loss.

The decoding sideThe processes on the decoder side

can be distributed among variousdevices or completely integrated ina display. Decompression is usuallydeterministic, within the accuracyof the arithmetic operations. Thus,any two decoders implementing thesame fixed-point (or floating-point,when so encoded) mathematical



operations should produce the sameoutput. However, there may be somesavings in processor power dissipation,memory size, etc., by taking someapproximation shortcuts in the de-coding algorithm, which can have aneffect on output picture quality.

With mostconsumer displays

based on eitheran LCD, plasma orsmall projection

panel, the displaycan become

part of the finaldigital -to -analog

conversion process.

Color decoding is typically thestraightforward inverse of the encod-ing process. Processing on the decod-er side provides image improvementand allows user preferences, includ-ing sharpening (peaking) and noisereduction. Spatial down -resolutioncan occur either within this process-ing or as part of the decompressionprocess. The D/A (digital -to -analogconverter), while ideally provid-ing an inverse to the digitizer func-tion, always requires a low-pass filter(LPF) that can potentially degradehigh -frequency response, i.e., theability to reproduce image detail.

The display system attempts to con-struct a faithful reproduction of theoriginal image. With most consumerdisplays based on either a large LCDor plasma panel, or a small projectionpanel (the latter usually involving op-tics), the display can become part ofthe final digital -to -analog conversion

process. While signal reconstructionin the system sense usually requires alow-pass filter, and one will be usedwhen driving analog video outputs,some digital display panels elimi-nate the D/A converter and LPF en-tirely. An optical spatial filter is usu-ally absent in panel displays due to itscomplexity and the ability of the hu-man eye to integrate the pixels into acontinuous image. BE

Aldo Cugnini is a consultant in the digitaltelevision industry.

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COMPUTERS & NETWORKSDIGITAL HANDBOOK

Navigating the cloudCloud computing brings opportunity, and challenge.

If you are wondering how the"cloud" relates to professionalmedia applications, you are notalone because one of the hottest

topics in the IT world these days iscloud computing. So, this month wewill look at several concepts behindcloud computing and explore both op-portunities and risks for our industry.

What is a cloud?Clouds are usually used as a sym-

bol for a network - frequently, theInternet. Clouds were used prior tothis by telephone companies to sig-nify their nationwide switching net-work. The Internet cloud that keepsshowing up on just about every facil-ity diagram is just a lot of switches,routers and servers interconnectedwith fiber optic lines and UnshieldedTwisted Pair (UTP) cables.

BY BRAD GILMFR

router caches close to the consumer,rather than being immediately delet-ed, so that frequently accessed mate-rial could be served from the routerrather than from the original source.Then, the concept of managed net-works emerged, guaranteeing perfor-mance through the cloud. Since thattime, the cloud gained and continuesto gain capabilities as it morphs frombeing an interconnection of privatecomputers to a collection of serversthat are able to provide a wide vari-ety of services to the public.

Whatever as a serviceWhen you get into a conversation

about the cloud, the services topicis not far behind. A quick Internetsearch for "as a service" returned thefollowing: Storage as a Service (SaaS),Software as a Service (SaaS), Platform

The cloud has become a system of connected servers instead of computersThis growth expanded potential capabilities, but it has also compoundedorganizational complexity.

Early in the history of theInternet, the cloud idea was prettystraightforward. Packets transitedthe network from source to destina-tion based upon the destination IPaddress in the header. As problemswith the Internet were addressed andsolved, however, equipment in thecloud became more complex. For ex-ample, content started being saved in

as a Service (PaaS), Computing as aService (CaaS), Infrastructure as aService (IaaS), Security as a Service(SaaS?), Data as a Service (DaaS),Communications as a Service (CaaS),and my favorite, Monetization as aService (MaaS). The list seems end-less. That said, some of these servicesmay have applicability to the profes-sional media industry, particularly

Storage as a Service and Software asa Service.

Storage as a ServiceStorage as a Service (SaaS) has

been around for some time. Thisservice originated when some verylarge companies decided to try sell-ing excess storage capacity on theirInternet servers to the public. Earlyefforts at offering remote storage ser-vices faced several challenges. First,bandwidth bottlenecks, at both theserver and consumer ends, presenteda major problem. Over time, how-ever, most users have been able togain access to very high bandwidthInternet connections. Ironically, somesuccessful remote storage servicessuffered from their own success: Asthey became popular, the bandwidthneeded at their locations increasedtremendously as the number of us-ers with high -bandwidth connectionsincreased. In the end, only very largecompanies with servers distributed allover the world remained viable.

A second challenge was providing amethod of connectivity to the remotestorage. Early implementations re-lied on File Transfer Protocol (FTP),Network File System (NFS), CommonInternet File System (CIFS), SAMBAor some other remote file systemimplementation. Media profession-als are well acquainted with the issuessurrounding large file transfers andFTP. FTP's file size limit varies withoperating systems, but even now itis not unusual to find limits of 4GB.Also, unless you are particularly care-ful, FTP's throttling mechanism willcause exceedingly slow transfers orcomplete failures of transfers if thenetwork delay is too long.

Users experienced interoperabilityproblems with NFS, CIFS, SAMBA andother remote file system implementa-tions as well. Perhaps some of these


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problems were due to configurationerrors, but still, implementations ofthese shared file system architectureswere less than plug -and -play. Firewallsalso caused connectivity problemswith some remote storage implemen-tations. IT departments were unwill-ing to open up the communicationsnecessary to make remote file systemswork across corporate boundaries.

Later implementations relied onHyperText Transfer Protocol (HTTP)and Secure HyperText Transfer Proto-col (HTTPS) for file transfer. In par-ticular, HTTPS alleviated many of theconfiguration/interoperability issuesand also provided relatively securetransit across corporate firewalls. But,HTTPS still did not allow the sort offlexibility and overall architectural de-sign that many remote storage userswere seeking. A new, simple, comput-er -friendly method was needed to ac-cess remote storage on the Internet.

Storing video in the cloudTwo powerful ideas combined to

enable SaaS. The first is the conceptof cloud services. In order to under-stand this better, it may help to look ata specific example. Amazon has beena major provider of cloud services.Amazon's Simple Storage Service (S3)offers large amounts of storage at verycompetitive prices. The main differ-ence between remote storage and S3is the way Amazon presents the stor-age to the user.

From a business perspective,Amazon is offering storage in thecloud; you are not connecting to aspecific server. Instead, you send con-tent to Amazon, and then they put itsomewhere. Later, you can search forit or retrieve it. You interact with S3using Representational State Transfer(REST). REST allows users to inter-act with services using many com-mon Web programming languages,including Perl, Java and PHP. Howyou interact with the service is upto you; the results are the same. Butthe important point is that Amazonprovides a storage service to you as abusiness. How they do it, where they

store the content and how they main-tain it is invisible to you.

The second concept that has beencritical in introducing services in thecloud is the clever renaming of theInternet as "the cloud:' To manypeople, the concept of a service exist-ing in the cloud is a lot easier to un-derstand than the concept of remotestorage offered as a business over theInternet. Which would you ratherbuy? But, whenever you hear SaaS,realize that vendors are talking abouta service being offered on distributedhardware connected to the Internet.

SecurityI have attended several professional

media conferences where SaaS hasbeen introduced. Without question,the number one issue raised right offthe bat is security. Vendors can say allthey want about the security meth-ods they use, but we users will alwaysbe concerned about whether contentstored in the cloud is safe. In fact, Iquestion whether we will ever be total-ly comfortable with a business wherethe essential premise is that you do nothave to worry about how and wherethe content is stored because the ven-dor has you covered.

That said, major broadcasters useVirtual Private Networks (VPNs)for on -air video delivery every day.Perhaps over time, financial consid-erations will overcome our concernsfor security, or better yet, perhapsvendors will offer professional mediaservices such as SaaS on secure, man-aged networks.

Software as a ServiceAnother area that may ultimately

be interesting for professional mediacompanies is Software as a Service(also SaaS). Two professional mediaapplications that immediately cometo mind are video editing and videotransformation. Imagine a video edi-tor application available on the Web.Now, also imagine that you are will-ing to store content in the cloud. Onecan then quickly see a professionalvideo application where you can

work on your project from virtuallyany computer. You can give othersaccess to your project, allow them tosee rough -cuts in real time and inter-actively participate in an editing ses-sion - even though editors and cli-ents may be hundreds or thousandsof miles away. Editors can access thesame content for different projects (anews story and a segment of a long -format news show, for example).Editing applications can be writtento allow differing editing functional-ity based upon the capabilities of theend -user device (workstation vs. tab-let) and bandwidth available.

Video transformation as a serviceinvolves giving content to a serviceand asking the service to transform thevideo in some way. This has been donefor some time in the Hollywood com-munity, although the concept of offer-ing this as a service is relatively new.

Software as a Service challengesseveral long-standing practices. Forexample, the software is not ownedor maintained by the end user; thefunctionality of the editor dependson the end -user device, and content isstored in the cloud. However, we maybe willing to accept these changesgiven the flexibility and capabilitiespresented by SaaS.

ConclusionNew cloud services are coming,

and security and viability concerns ofthese services for professional mediaexist. But, realizing them as Internet -

based server options helps frame in-formed talks with vendors. BE

Brad Gilmer is president of Gilmer& Associates, executive director of theAdvanced Media Workflow Associationand executive director of the VideoServices Forum.

111Send questions and comments to:[email protected]

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PRODUCTION ROOMDIGITAL HANDBOOK

Nonlinear editorsHere's a look at some of the technicaldistinctions of today's NLE systems.

These are heady days forhigh -end nonlinear editingsystem providers. But, wemay have recently lost one

of the biggest players in the game.Despite the fact that Apple's Final

Cut Pro 7 software is being used by54.6 percent of the professional ed-iting community, according to a re-port by market research firm SCRIInternational, in June Apple releaseda completely rewritten version calledFinal Cut Pro X. Much to the surpriseof many, despite that this new incar-nation is completely 64 -bit, so manyprofessional features were removed

BY L T. MARTIN

have more than a few editors scratch-ing their heads.

So this makes it an apt time totake a look at some of the technicaldistinctions under the hood of the stillbroad spectrum of NLE systems fromthe major manufacturers who stillwant to offer solutions in this space.

Avid Media ComposerAvid's Media Composer is still

the most used NLE on prime -timeTV productions, being employed onup to 90 percent of evening broad-cast shows. One reason is its designphilosophy, called Avid Intelligent

Avid's Media Composer is now compatible with Pro Tools hardware for greaterworkflow flexibility.This includes audio capture and monitoring of Media Composerprojects through Pro Tools hardware and voice-over recording into the NLE toenable faster collaboration between sound and picture departments.

that it appears Apple may be switch-ing its focus away from the high -endvideo editing market segment. FinalCut Pro X cannot import projectsstarted in Final Cut Pro 7 or any otherNLE. It cannot ingest from or outputto tape and can't even be connectedto a broadcast monitor. These notica-ble (and important) missing features

Architecture, that Avid instigatedback in 2008 with the release of MediaComposer 3 and the DX hardwareline. Avid Intelligent Architectureis a way of constructing softwarethat constantly looks at the Mac orWindows computer being used, thegraphics card installed and whateverhardware is attached, and selectively

sending a given task to the compo-nent best suited to accomplish it.

For example, if an editor needsto do something that is CPU -inten-sive such as decoding a codec, thosefiles get sent directly to the CPU.But effects work will be directed tothe GPU, and DNxHD encoding(short for Avid's proprietary DigitalNonlinear Extensible High Defini-tion compressed file format) or thinraster format expansion will go toAvid's DX hardware. That enables theMedia Composer software to con-stantly evaluate the resource that canmost efficiently get the job done.

Avid has also been successful im-plementing phonetic indexing intoits systems starting with the 2007 in-troduction of its ScriptSync feature.ScriptSync analyzes the audio trackson clips in the Media Composersource bin and links them to the writ-ten script that has been imported intothe system. Clicking on a line of scriptgives access to all the takes containingthat specific dialogue.

This year, Avid introduced Phrase-

Find for unscripted productions suchas documentaries or news packages.PhraseFind is a search engine plug-inthat can find specific words or phras-es in a production without referenceto a script.

Autodesk SmokeOne of the most cost-effective and

powerful nonlinear finishingsystems isAutodesk's Smoke. Originally avail-able only as a turnkey system onLinux, Smoke became available forthe Macintosh in a software -only, 64 -

bit version in December 2009. Thelatest incarnation, Smoke for Mac2012, is an online NLE with extensivegraphics capabilities that can importEDLs in all major formats. Smoke

28 broadcastengineering.com 1 September 2011


gains its image quality by automati-cally converting every compressedincoming file to uncompressed RGBand performing all its graphics/effectswork in 4:4:4:4 color space.

Smoke is also the only NLE thatcan use the platform -independentAutodesk FBX file format to load 3-Dmodels and motion graphics in theiroriginal, unrendered native format.FBX, which originally stood for film -box, is similar to a PDF format for3-D graphic objects.

For stereoscopic, dual channel 3-Dwork, Smoke for Mac 2012 can alsoscan each video track to detect dif-ferences between the left- and right -eye clips.

Grass Valley EDIUSGrass Valley has been adding some

amazing capabilities to the latest 6.02version of its EDIUS software, whichexpands its usefulness in broadcastapplications. The company acquiredthe system from Canopus in 2005

Grass Valley has added a new live capture option to EDIUS.The option allows usersto ingest live video on one EDIUS client while they simultaneously begin to edit iton a different EDIUS client.

and is marketing EDIUS as a soft-ware-codec edit system for Windows,highlighting the efficiency of its con-stantly upgrading codecs. In fact, inNovember 2007, EDIUS was the firstto crack the challenge of decoding thecomputationally -demanding but lowbit rate AVC Intra HD format.

Thanks to its robust codec tech-nology, EDIUS can now edit 4K files(4096 x 2160) in 8- and 10 -bit. To editthese files, the system automaticallygenerates proxy files during capturethat are easier for the CPU to han-dle. This lets EDIUS perform all itstasks in real time without ever need-ing to render, even on platforms oflesser capabilities.

The system's Source Browser putsthe functionality needed for import-ing all files into a single viewer, sup-porting from P2 to Blu-ray applica-tions. EDIUS can handle a wide rangeof output options, and can even ex-port back to an AVCHD card or aniPad/iPod.

Boris FX Media 100After being spun out of Data

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Abstraction Layer, has never usedApple's QuickTime proprietary multi-media framework to play out its files.

Of course, the Media 100 systemdoes use QuickTime for handlinggraphics formats and system -level co-decs. But it does not use QuickTime'splayback services because the Me-dia 100 system was always intendedfor broadcast use and has becomeknown for its ability to quickly playout its timeline without waiting forQuickTime intervention.

Although Media 100 can play outa sequence of shots directly from itstimeline, it uses Adobe After Effectsas its compositing system, and can ac-cess Apple's Color software for colorcorrection. In fact, many editors useMedia 100 and Apple's Final Cut Pro 7software side -by -side. Media 100 canbe faster when assembling a simplesequence thanks to its ability to use anA/B roll style of timeline, which givesthe editor simple access to the pre- andpost -roll handles of each shot. Evenso, many editors may find that FinalCut Pro 7 provides greater effects andimage manipulation capabilities.

Sony Creative SoftwareVegas Pro

The Vegas Pro NLE from SonyCreative Software boasts several firsts.

It was the first editor to go 64 -bit onWindows, and was the first NLE toincorporate a digital audio work-station within the NLE itself. Withsupport for industry -standard VSTplug -ins, Vegas Pro provides the abil-ity to edit audio down to the sam-ple level and perform 5.1 surroundsound mixing.

The latest version, Vegas Pro 10,enables integrated stereoscopic 3-Dediting and the ability to deliver 3-Dprojects either as two track files orsingle files with side -by -side, top/bottom or line -alternate encod-ing. It also provides ingest, previewand output of closed captioningdata, supporting industry -standardclosed -captioned file types for eitherbroadcast or Web delivery.

Adobe SystemsPremiere Pro

Many editors may suggest thatAdobe Systems' Premiere Pro NLEbenefits most from the presumedretreat of Final Cut Pro from thebroadcast editing arena. As part of itsCreative Suite 5.5 collection of soft-ware applications, Premiere Probenefits from the development ofthe Mercury Playback Engine, whichruns natively in 64 -bit on bothWindows and Mac platforms.

Adobe has optimized Premiere Profor maximal use of a platform's GPUto enable it to handle file sizes up to5K and beyond. It works with boththe broadly accepted OpenCL pro-cessing language and also NVIDIA'sproprietary Compute Unified DeviceArchitecture (CUDA) parallel com-puting architecture. This design letsPremiere Pro offload many process-ing requirements from the CPU ontoa qualified GPU.

The result is that Premiere Pro hassufficient power to handle multiplelayers of Red Digital Cinema's 4K filescomplete with color correction. Ado-be has pursued an aggressive releasecycle of its Premiere Pro softwareupgrades. But unlike some other ven-dors, the company maintains a highpriority to ensure each new versionis completely backwards compatiblewith its predecessors. BE

L.T Martin is a freelance writer and post -production consultant.

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NEW MEDIA NETWORKSSYSTEMS INTEGRATION

MultiviewersIncrease your understanding of this important piece

of broadcasting equipment.

Multiviewershave becomeubiquitous inmost broad-

cast control rooms. The flex-ibility of displaying multipleimages in various resolutionsin an active and situational -

aware display has many ben-efits. Building on some ofthe key points learned in thelast article, "Understandingblocking capacitor effects,"(August 2011), let's continueour review of broadcast andproduction equipment de-sign considerations, this timewith respect to multiviewers.

As you may remember, myfirst task in the video broad-cast industry was to build anoptical -to -electrical converter and toevaluate small form -factor pluggable(SFP) integrated circuits from dif-ferent vendors. That assignment waspromptly interrupted due to a changein priority, and I moved to the multiv-iewer group to increase the manpower.At that time, I had little first-handinformation about multiviewers, andWikipedia didn't help much either. SoI had to learn the hard way. Fortunate-ly, as I look back on the experience, Ican say that we had a talented team,and the product we developed turnedout to be a great success.

A full discussion of the multiv-iewer by itself could take more spacethan this entire issue, so I will attemptto explain it from a 10,000ft view. Ifportions of this technology gener-ate questions or you want to discussthese issues, please let me know, andin my blog or through BroadcastEngineering's Disqus, I can explainthings in more detail.

The global functionality of the

BY RFNAUD LAVOIE

Some multiviewerseach input signal towithout errors.

also have the ability to analyzebe sure the signals are correct and

multiviewer can be summarized likethis: Take multiple video and audiosignals and combine them to createattractive and flexible layouts. In ad-dition, some multiviewers also havethe ability to analyze each input signalto be sure the signals are correct and

Video/inputs

Graphicaloutputs

without errors.Basically, multiviewers

are composed of severaltypes of common cards:input video card, input au-dio card, GPIO card, outputcard, interconnect card and,in some cases, router cards.(See Figure 1.)

The input video cardThe input video card is

responsible for the recep-tion of the multiple videofeeds. The number of inputsoften varies from four to 16inputs, depending of thesize of the multiviewer. Thefirst task of the input videocard is to receive the signalsand perform any needed

equalization or light -to -electricalconversion. One of the new trends isto use SFP cards to handle this task.Because of the variety of signal types- coaxial SDI, optical SDI, compos-ite, DVI, video component - and thepressure to release products faster and

Control andsynchronization

Figure 1. A multiviewer is typically built from only a few types of common cards,using as many as necessary to complete the design.


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-0

" 111

Equalizer Deserializer

Probeand

ancillarydata

extraction

Deinterlacerand

scaler

stage 4

Serializer

Stage 5Stage 1 Stage 2 Stage 3

Figure 2. Shown here is an example video input card with multiple processing stages.

faster, it makes sense for manufactur-ers to use a generic connector (SFPcage), which enables them to supportall video formats without the need tocreate a variety of different cards.

In the first stage, the shape of thesignal is restored and converted to SDI.After Stage 1, the serial data is con-verted to parallel data for processing,but why? The reason lies in the speedof the SDI signal. To properly convertthe data (with cost -justifiable technol-ogy), it is advantageous to first convertthe serial data to parallel data. This iswhat the deserializer does - take theserial data and parallelize it to ease thefurther processing of the video.

The video can now be upscaledor downscaled more easily. We willsoon have the technology to processthe data without deserializing it, butfor now, the data must be processedin parallel form. The next processingsteps are now feasible: deinterlacing,upscaling, downscaling, color conver-sion, audio analysis, audio extraction,etc. All of these functions are possibleon the parallel video data.

Depending on the multiviewer'sarchitecture, the functions imple-mented by the input video card varyfrom deserialization only to full scal-ing. To better understand the inputcard functionality, let's reconsiderthe example multiviewer input card.(See Figure 2.)

In this case, each of the eight videofeeds on each input card represents a3G -SDI, 1080p signal. Each signal isfirst equalized/converted at Stage 1and then deserialized at Stage 2. This

represents 8 x 2.97Gb/s, a total of23.76Gb/s of aggregate data.

Nb_input

Total bandwith = E Bandwithsnput

input = I

The third stage is to probe and/orextract auxiliary data, which may in-clude audio, teletext, closed caption-ing, etc. This is an optional step. Thevideo data itself is processed at thefourth stage. In the fourth stage, thesignal maybe deinterlaced,which is not required for1080p, and then scaled.This stage is crucial, andeach multiviewer manu-facturer has its own se-cret sauce to maintain thequality of the content.

Let's assume that wedid downscale all the in-put video in stage fourby a factor of eight to allinputs with the same sizeon one display. The aggre-gate data is now 8 x 2.97Gb/s / 8 =2.97Gb/s between this input card andthe output card.

Nb_input

Scale bandwith = E (Bandwith,nput x scale_factor)input =

To send this data to the outputcard for the final scaling, color cor-rection and picture -in -picture, alongwith closed captioning, VU meters,VITC, etc., the data channel needsto support from almost 6Gb/s (whenSD -SDI signals are downscaled) upto 23.76Gb/s for nonscaled eightinputs. To accomplish this in a cost-effective manner, the input card usu-ally reserializes the data (Stage 5)before sending it on the interconnectcard. An important factor to considerwhen designing a multiviewer inputcard is to perform this step with uni-form latency. Any delay induced byprocessing must be constant across

all inputs. This becomes evenmore important when process-ing 3-D imagery that originates astwo separate sources.

Interconnect cardIn many cases, the interconnect

card is a passive card. It can be seenas a large array of cables that connectevery input card and every GPI card

Input card4

Input card3

lap:it card2

Inp it card

Output card4

Output card3

Output card2

Output card

Figure 3. The point-to-point video patharchitecture has become the more commondesign for video links.

to every output card. This card alsocontains the communication linksto control every other componentsin the multiviewer: input cards, out-

put cards, GPIOcards and audiocards. One of

Figure 4.The daisy -chain video path architecture can be used for the communicationchannel.


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Figure 5. Shown here is an example communication link with the master path inred and the redundant path in orange.

the most important links on the in-terconnect card is the synchroniza-tion link. This includes Hsync, Vsync,Fsync, clock and proprietary links tosynchronize all the cards together.

allow multiple users to share the samecontrol path and statuses.

Finally, the synchronization link iscomposed of Hsync, Vsync and othersynchronization signals. This link can

Vsync from theGPI card

Vsync at 1in(output card 1)

Vsync at 19in(input card 4)

Ops 150ps 2.850ns

e

Figure 6. This graph shows how card length affects delay on the Vsync.

Different architectures have beenimplemented to connect inputs tooutputs. These implementations areoften thought of as daisy chain, busor point-to-point. However, with theincrease in signal speed, the point-to-point architecture has become themore common design for video links.(See Figure 3 on page 34.)

For the communication channel,the daisy -chain approach can be used.(See Figure 4 on page 34.) But likeabove, with the increased data speed, ithas become more difficult to do. Usu-ally, the internal communication linkis composed of two or more commu-nication paths to ensure system redun-dancy. In our example, (see Figure 5),output card 4 is the origin of the mas-ter path (colored red), and output card3 is the origin of the redundant path(colored orange). The communicationlink also contains external commu-nication to allow the user to take thefull control of the multiviewer. Nowa-days, the external communication linkis typically Ethernet (with SNMP) to

be a daisy chain, like that shown withthe blue line in Figure 4. However, thedelay between the first card in the sys-tem to the last one should be almostthe same. The delay on a printed cir-cuit board is typically 150ps to 180psper inch, which is not a significant

Let's use Figure 3 as the reference forour multiviewer discussion.

Each input card sends one videolink to each output card, and thereare four output cards. This meansthe interconnect card is composed of4 x 4 high-speed links that carry videosignals. But this is theoretical, and thisdepends of the bandwidth betweeninput cards and output cards. Inpractice, two or more links are usedfor video links. Let's now take a lookat the output card.

Output video cardIn our example, each output card

can receive video from four inputcards. In this example, that meanseight video inputs per input card. Un-der a worst -case scenario, each out-put card will receive the entire videostream from all input cards (a 32 -in-put stream). Let's assume the outputcard is capable of driving one moni-tor with a 4K resolution maximum.In this example, the output card willdrive one monitor capable of 4K reso-lution each (3840 x 2160 pixels).

Because we can have 4 x 1080p im-ages displayed on this monitor, theworst -case aggregate bandwidth willthen be 4 x 2.97Gb/s = 11.88Gb/s.We will ignore any saved bandwidthfrom removing the blanking period.(Only the active picture is usually

Different architectures have beenimplemented to connect inputs to outputs.The point-to-point architecture has becomethe more common design for video links.

factor, but buffers in the path are criti-cal. Let's assume a standard rack unitchassis where the printed circuit boardwill be 19in long. (See Figure 6.)

If this delay is not acceptable (archi-tecture dependant), a point-to-pointarchitecture can be used instead.

For the current crop of multiview-ers, each vendor does it differently,and the interconnect card can be cre-ated in a thousand different ways.

transmitted between the input andoutput cards to save bandwidth.)

The question the system designerneeds to answer is: Can the 4 x 1080pimagery come from one input card? Ifso, the bandwidth of each input cardis now 11.88Gb/s maximum. With to-day's new programmable logic FPGAcards, 11.88Gb/s can be achieved withone high-speed link, commonly calledserializer/deserializer or SerDes.


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Each input card can send one ortwo links to the output card for pro-cessing, for a total of each output cardreceiving up to eight links from the

input's cards. The processing powerof the output card therefore must be4 x 11.88Gb/s, or 47.52Gb/s of videodata. This is a large amount of data to

Deserializer

Stage 1

Final scalingand

color spaceconversion

Stage 2

Positioningand

graphicmusing

Stage 3

Graphicgeneration

andcontrol

GPU

Stage 4

Serializer forgraphicoutput

Stage 5

0E

Figure 7. This block diagram shows an example of an output video card.

process in real time. Often the videodata is encoded to ease the deserial-ization process using a well-knownencoding scheme like the 8b/10b.Using 8b/10b, the final data band-width becomes 47.52Gb/s x 1.25, or59.4Gb/s. This requires that the out-put card be equipped with a powerfulenough FPGA to be able to processthis amount of video data.

The first stage performs the de -

serializing process. (See Figure 7.)Once this is done, further data pro-cessing can be done in Stage 2, suchas final scaling and color conversion(from YCbCr to RGB). Following thesecond stage is the positioning andthe graphic muxing. This is wherethe images are placed at the correctlocation in the final layout. Graphicsare also muxed to allow the user toadd transparency, pictures -in -pic-tures and fancy layouts like rotatingimages. Multiviewers also must be

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capable of displaying closed caption-ing, VU meters, tallies, and other sig-nals and alerts.

The final stage is another serializa-tion, this time to format the data to bereceived by the monitor or to send toanother destination (SDI, streaming,etc.). The graphic objects are gener-ally created by an internal CPU thatcontrols the layout of the objects andthe system behavior, SNMP, etc.

The softwareThe software is a key component

of any multiviewer. This operationalfeature applies not only to the graphi-cal objects and final layout, but alsoexternal devices the multiviewer mayneed to control. Such software shouldremain easy to understand and touse. With remote access, this softwareshould run on different platformsand sometimes from different loca-tions throughout the world, thanks tothe Internet and the thumbnails!

Complementary cardsThe multiviewer's complementary

cards include any other supportingfeatures needed to make this devicework in the real world. While thesecards are important, they are also of-ten less complex. Let's review some

of the more common cards usedon multiviewers.

The GPIO card is used to add evenmore functionality to a multiviewer.These cards control the tallies and vi-sually report alarms. The GPIO cardis typically equipped with either 64 or128 general-purpose inputs and out-puts to control other devices in thestudio (such as microphone mute,communication request, ring alert,start a play -out system, red-light, stu-dio warning lights, etc). The inputsare tolerant to 24V, and the outputsare isolated by opto-couplers to per-mit easy interfacing.

The genlock card is used to syn-chronize the output of the multiview-er to the reference signal of the stu-dio. The genlock card can also createall the clocks used in the multiviewerto ensure that all boards receive thesame clock (,remove the drift betweenclocks) and maintain clock phase.

The need for audio monitoring isobvious. A multiviewer provides aconvenient platform to see visuallywhat is happening on both video andaudio sources. Also, multichanneloperation precludes operators frombeing able to hear the audio fromperhaps hundreds of channels. Vi-sual displays, along with automated

silence sensors, make the process farless cumbersome. Instead of requir-ing a separate device to analyze theaudio sources, the multiviewer can doit and alert the operator only whennecessary.

Multiviewers and IPAs IP technology intrudes into the

traditional serial digital video mas-ter control room, multiviewers willneed to adapt. Some multiviewersalready support the encoding format;others are able to receive thumbnailsfor monitoring.

Another growing trend is to embedthe multiviewer in routers and otherequipment. Some vendors alreadyembed the monitoring router func-tion inside the multiviewer. For someapplications, it may make sense tominimize the interconnect betweenthe router by locating the devicesclose together, even by making themultiviewer a component inside therouter. When this approach is taken,manufacturers will need to make thisa robust device because the router isthe most crucial device in any broad-cast or production environment. BE

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September 2011 broadcastengineering.com 39

40broadcastengineering.com

I Septem

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MTV DOES MORE WITH LESS 44

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special Report supplement I September 2011 I broacicastengineering.com 41

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The production control room featuresa Sony MVS-8000A HD switcher and a

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n the spring of 2010, Michael Bivona,vice president of engineering at MTVNetworks, was faced with a perplexingchallenge. He was tasked by the com-

pany's Music Group division with rebuildingthe physical and technical production facilitiesof Viacom's MTV studio overlooking TimesSquare in NewYork City from SD to HD. Whilethis might seem to be an engineer's dream proj-ect, Bivona had to do it with less space and alimited budget; and his staff still had to oversee

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Oh, and one othercaveat: He had tolaunch the new HDstudio on New Year'sEve 2011, a mere sixmonths after firstbreaking ground onthe new second -floorfacilities that now in-clude a new studio,production controlroom, audio mixingsuite, ingest/playback

and video control/transmission areas. Addi-tionally, numerous HD updates and enhance-ments were also required for the post -pro-duction facilities, delay rooms and updatedtransmission facilities that are co -located onalternate floors of the building. They alsoupdated two rooftop robotically controlledcamera systems.

Taking it in stride, Bivona said, "Our team isoften asked to do the impossible with limitedresources. The success of this project is a trib-ute to the great professionalism and expertiseof the entire project team."

Considering all of these factors, and look-ing to partner with an established systemsintegration firm with a proven track record,Bivona called upon the expertise of JosephPolicastro, a founder and senior director ofBroadcast Integration Services (BIS, UnionCity, NJ). The two had previously workedtogether in 1997 to build the original SD fa-cilities in the previously larger space. Theywere now challenged to develop an integrat-ed project program that could achieve morewith fewer resources.

Since Viacom had surrendered half of itsoriginal space to a new tenant in the build-ing, a new optimized space development planwas required (based on a design by architectNeil Tucker), and a totally new mechanical,electrical and plumbing (MEP) infrastruc-ture was required (based on a design fromAMA Consulting Engineers, NY). Most im-portantly, the broadcast technical infrastruc-ture - which is based on HD and 3G -com-pliant equipment and systems - had to becarefully selected and integrated in a "tightlycoupled" architecture in order to providemaximum flexibility, scalability and futureexpansion capability.

The design team also had to deliver the ba-sic workflow processes that production andoperations crews had become adept at whileintroducing enhancements to take advantageof new HD file -based advancements.

Policastro said the original design phaseposed many challenges since there was muchless space to work with, yet they had to ac-commodate all of the necessary systems and

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The audio control room houses a Solid State LogicC140 console mixing surface and several roll -

around racking carts to accommodate a variety ofoutboard audio and peripheral support equipment.

equipment requirements. The design planthat ultimately evolved is practical and re-tains its traditional production workflow, yetprovides a highly flexible platform to accom-modate new technologies as they becomeavailable. Several examples include the cre-ative use of MADI systems to conserve wir-ing requirements and maximize the audioI/O channel capacity.

Production studio with a viewThe new 4000sq ft production studio fea-

tures Sony HDC1500 (1080i) cameras, usedin various configurations. The studio camerasystem, which is wired to accommodate upto 14 cameras, is based on a predominantlySMPTE camera fiber backbone. A glass street-

side view overlooking New York's TimesSquare, it provides a stunning backdrop for

the studio, completewith an outdoor45ft-wide electronicbillboard in a largepicture frame acrossthe street that Bivonaand his team oftenutilize to great ef-fect as part of theshow's imagery.

To support theshow production re-quirements, the teamalso had to imple-ment more than 73wireless channels formicrophones, IFBand party line in oneof the most active RF

environments in New York City. Since the stu-dio also accommodates audience provisions,and considering the addition of dynamic livemusic -based production components, there arealso audio console mixing positions for bothfront of house (FOH) and monitor mix (MM).

The studio lighting system is based onhigh -efficiency LED fixtures, which solved theproblem of suitably lighting a smaller studiospace for HD production while simultane-ously reducing power and heat loads.

HD production control roomThe newly installed, state-of-the-art produc-

tion control room features a Sony MVS-8000AHD switcher, Chyron graphics and a monitordisplay wall composed of 20 flat -panel LCD

monitors of various sizes with multiviewer soft-ware for dynamically assigning sources, aspectratio sizing and tally. In order to optimize thespace for the production and operations staff,a three-tier console configuration features thenecessary ergonomics while providing suitablesight lines to the front monitor wall. Analogousto a typical mobile production truck environ-ment, these design efficiencies fully maximizethe space while providing what Bivona said wasa comfortable environment for the crew.

Adjacent to the control room is the audiomixing suite, which was acoustically designedto accommodate a conventional surround -sound monitoring environment. The au-dio control room houses a Solid State LogicC140 console mixing surface and several roll-

around racking carts to accommodate a vari-ety of outboard audio and peripheral supportequipment. To optimize space and minimizecabling, the console interface engine, audioI/O frames and jackfields are all co -locatedin the adjacent equipment room racks. AGenelec GLM surround -sound monitoringsystem provides a software -based time delayand phase alignment application to optimizethe room monitoring environment. Also, con-sidering the recent passing of the CALM Act,Linear Acoustic loudness monitoring is alsoimplemented in the audio system chain.

Audio communicationsImplementing 64 -channel MADI technol-

ogy within the audio console core as well asthroughout the entire plant facilitated the ef-ficient distribution of multiple channels ofaudio (via a fiber -based infrastructure). TheMADI transport stream allowed multiple au-dio mixer positions, routing switcher, inter-coms and ancillary interface devices to shareand process audio channels simultaneously.In some areas, discrete AES signal subsystemrouting and support provisions were deployedas a backup.

The overall communications system isbased upon an RTS ADAM Matrix with acombination of analog, MADI and IP-basedRVON interface cards. The matrix, keypanels,telephone interface units and other gear areall seamlessly interfaced to provide a plethoraof point-to-point, IFB, four -wire (e.g. cam-eras, etc.) and party -line communicationsrequirements. Although the majority of theIFB and party line are interfaced to the talentand operations via wireless provisions; there

46 broadcastengineering.com I September 2011 I Special liepurt aupw,t,

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Post production/playbackTo support production and post capabili-

ties, graphics and video files are shared overan extensive IP-based file sharing network.

Project team

MTV Networks:Michael Bivona, VP of engineeringJim Brizzolara, director of studio engineeringBill Anchelowitz, director of project managementThayne Knop, director of content creation technologiesTyler Marinello, studio engineeringAdrienne Bonfrisco, director of administration

Broadcast Integration Services:Joseph Policastro, principal senior directorKevin Henneman, senior design engineerAdam Semcken, senior design engineerAndy Morris, engineerRobert Gilmartin, engineerJudi Southard, logisticsChris Butler, integration supervisorWilliam Frederick, lead techJavier Casilimas, engineer

Eauioment list

ADC video and audio patchingAvid ISIS storage, Media Composer Nitris editingChyron HyperX graphics platformEvertz EQX multiformat/hybrid router, VI P -X multiviewer,

VistaLINK SNMP monitoring, test and measurement equipmentEVS XT[2] video serversGenelec speakers and loudspeaker manager softwareGepco fiber systemsImage Video tally systemJoseph Electronics fiber systemsLinear Acoustic 5.1 monitoring and loudness systemsMiddle Atlantic Products equipment racksQTV telepromptersRTS intercoms

Sony HDC1500 cameras, HDCAM SRW-5800 tape playback,LCD displays, MVS-8000A HD switcher

Solid State Logic C140 audio consoleSennheiser wireless microphonesTBC Broadcast consolesTektronix test and measurement gearTelecast Fiber Systems SHED systemTelemetrics/Canon rooftop camera roboticsVinten studio pedestalsWohler audio monitoring

The team has also implemented an EVS harddisk system for multicamera recording andplayout production support. Studio and post -

production teams share resources, linking thestudio EVS system to the post -productionAvid Media Composer editing workstationsand ISIS Interplay media storage.

The Avid workstations are located on anacjacent floor and networked to an ISIS stor-age system that is tightly integrated with EVSX7[2] server and IP Director interface.

This provides for tapeless studio recordings,simultaneous Avid -based high-res editing andAvid Interplay -based ingest and low-resolu-tian review of media from either the playbackor producer areas in the studio facility locatedon the concourse level. Additionally, this pro -vices seamless file -based capabilities from thenumerous producers' offices and worksta-tions situated on various floors throughoutthe building.

Although the facility has heavily seguedto a tapeless workflow environment, thereis r complement of Sony HDCam SR tapedecks to supplement additional productionreq lirements and legacy tape playback andrecord functions.

The equipment room,from both sides

The main equipment room posed severalunique challenges due to the space constraintsand required HVAC provisions. Again, takingcues from mobile production trucks and theteam's skillful use of space, the room featurestwo rows of 36in-deep racks, 8ft high, withequipment and components carefully in-stalled on the front and rear sides. To providesuitable equipment cooling in such a denseenvironment, the HVAC system is imple-men-ed in a hot-aisle/cold-aisle configuration(e.g. racks oriented back-to-back for hot -aislereturn). Maintaining the individual pieces ofgear can be cumbersome at times, but Bivonaand Policastro spent many hours planningwhich pieces of equipment were most likelyto need more focused attention and which didnot. Meticulously planned cable managementpractices were also implemented to best uti-lize die available cable routing paths.

Among the key equipment system com-ponents, the plant's routing infrastructureis based on a hybrid AV router from EvertzMicrosystems. The router connects the facil-ity's production equipment, audio mixers,

48 broadcastengineering.com 1September 2011 ,,ort supplem,

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and utility/external sources via fiber andcopper cable.

Space saversLooking to conserve real estate, the de-

sign team used onboard processing andfiber-optic and network connectivity forsubsystems like production routing, audiomixing and intercom communications.Multiplexing typically handled by multipletrays of outboard devices and cabling withpatching points were replaced with a com-mon audio standard (MADI), as well asnumerous embedders and de-embedderslocated in the router frame itself. This savedon rack space as well as reduced cooling andpower requirements.

While some engineers might cringe at hav-ing to troubleshoot such a system, Bivona saidthe use of Evertz Vistalink SNMP monitor-ing software streamlines the trouble shootingprocess while monitoring system health andalerting them when a problem does arise.

Video control/transmissionThe video control/transmission area con-

tains all of the respective camera CCUs,OCPs and two SVO camera shading/QCpositions. Although the system is currentlywired to accommodate a total of 14 camerachains, only seven camera systems were in-stalled initially to accommodate the currentproduction requirements. Bivona indicatedthat for any ad -hoc and additional futureproduction applications, it is simply a matterof purchasing or renting additional camerachain components.

There are also several broadcast servicepanels (BSPs) judiciously located in the studioas well as located in several alternate areas ofthe overall concourse studio facility, includ-ing support areas such as the green room andcommon corridors. This allows the additionalflexibility for stand-up camera positions as re-quired. All of the SMPTE camera fiber dropsterminate back at a SMPTE patch panel in theequipment room. Operations then has the

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flexibility to assign any camera to any respec-tive CCU.

In addition to the studio camera compli-ment, there are also two roof cameras withremote PTZ provisions from the SVO posi-tions. These cameras provide a stunning viewoverlooking Broadway and Times Square. Theinterface panels on the roof areas include aTelecast Fiber Systems SHED system that con-verts SMPTE baseband signals to light (E/O)for distribution over single -mode fiber. HDsignals are then converted back (0/E) at thereceive end for interface to the core systems.Leveraging existing fiber drops on numerousfloors within the building, the SHED allowsHD camera signals to be transported over twoordinary strands of single -mode fiber to nu-merous locations throughout the building.

Additional provisions within the videocontrol/transmission area also include utilitycolor correctors and discrete routing switch-er control panels to support MTV's releasetransmission path assignments.

Final comments andfuture applications

The original concept of ultimate flexibilitywhile minimizing day-to-day operating costshas resulted in the ability of MTV Networksoperators to quickly and easily recall presetshow configurations and setups as necessarywith minimal reconfiguring of the infrastruc-ture in place.

With ongoing close attention to detail andthe continual liaison of all of the project teammembers throughout all phases of the project,the new facility is one that MTV Networks canlive with for many years to come, accordingto Bivona. As for other potential applications,the core infrastructure even provides the flex-ibility of producing 3-D content by addingsome option cards if the network ultimatelydecides to experiment with a dual -stream 3-Dformat in the future. =IMichael Grotticelli regularly reports on theprofessional video and broadcast technologyindustries.

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New loudness

metering standard

Dr. Richard Cabot uses atest suite developed for thePrism Sound dScope Series

Ill audio test instrument.The test measures whetherloudness meters conform tothe new BS.1770 algorithm.

oudness issues have been aroundsince the beginning of broadcasting,or at least broadcast advertising, but

ila they were exacerbated by the DTVtransition and inconsistent use of the dial-

norm metadata. Listener dissatisfaction in-creased, ultimately culminating in passage ofthe CALM Act.

In the design stages of DTV, considerableeffort was put into development of loudnessmeasurement technologies to support the dial -norm mechanism. This work culminated in theoriginal version of the International Telecom-munications Union (ITU) standard BS.1770,which describes a fundamental loudness mea-surement algorithm. It was validated throughmultiple sets of listening tests on various piecesof program material. The ITU standard alsodescribes a true -peak meter for determiningthe peak amplitude expected when a digital au-dio signal is reproduced in the analog domainor transcoded into another digital format.

Several years ago, the ATSC developed rec-ommended practice A/85, which describes

I

how broadcasters should ensure a satisfactorylistening experience for DTV viewers. TheCALM Act mandates that the FCC enforcethe loudness related portions of ATSC A/85,and its successors, through appropriate rulemaking. The loudness measurement portionof A/85 is based on BS.1770.

The CALM Act created an obvious oppor-tunity for equipment manufacturers to pro-vide loudness measurement tools. Although afew loudness measurement products were onthe market by 2009, many more were intro-duced at NAB and IBC in 2010, and there arecurrently more than a dozen loudness mea-surement products of various forms on themarket. Each is vying for a piece of the large

as broadcasters equip their fa-cilities for loudness measurement.

In late 2010, the ITU committee that main-tains BS.1770 accepted (after much nego-tiation) changes submitted by the EuropeanBroadcasting Union (EBU). The result is asignificant improvement in the calculation ofloudness - one that makes the measurementmuch more sensitive to the loud portions of anaudio segment. The effect is to prevent adver-tisers from significantly increasing the loudnessof a portion of a commercial by manipulatingthe loudness elsewhere. Consider a hypotheticalexample where an announcer screams at wide-ly spaced intervals throughout a commercial inan effort to get the viewers' attention. Thoughthe average loudness might seem reasonable,the peak loudness will be quite annoying. Thenew method puts more emphasis on the louderportions and assesses this spot as louder thanthe original technique would have.

This new version of BS.1770 has just beenpublished. Consequently, meters in use willneed to conform to the revised specification.Most manufacturers of such products havebeen following the developments in the EBUand ITU and have upgraded their software toaccommodate the change. Unfortunately, how-ever, they haven't necessarily done it correctly.

The problemAs a user, how do you assess whether a

meter you are considering meets the revisedspecification? How do you know if its softwarehas been modified to conform to the new ver-sion standard? It's not as easy as testing a VUmeter or a PPM. The EBU specifies some ba-sic tests in its technical recommendations andprovides the necessary waveforms on its web -site. However, these tests are basic and do notthoroughly test all aspects of compliance.

We will describe a suite of tests developedspecifically to check every aspect of a meter'sdesign. These tests also give diagnostic infor-mation about any implementation issues thatexist. They are available at no charge as de-scribed at the end of this article.

Our test suite was developed by craftingsignals whose parameters change dynami-cally so as to stress individual portions of themeasurement in isolation. Each test can thenmaximize its sensitivity to the specific imple-mentation errors it was designed to detect.The signals were developed using mathemati-cal models of the algorithm, including modelswith various intentional imple-mentation errors. The signalswere optimized to give the larg-est difference between readingsobtained by the correct modeland those obtained by incorrectimplementations.

The new BS.1770 algorithmoperates on multiples of a basic100ms interval, so readings dif-fer slightly with variations be-tween the start of the measure-ment and the start of the signal.These reading differences followa cyclic pattern, with alignments50ms apart creating maximaldifference. Consequently, thetest signals were evaluated ata reference alignment and at an alignment50ms delayed. Signal characteristics wereadjusted to minimize this difference, thoughsometimes this was in direct conflict with thedesire to maximize the sensitivity to imple-mentation errors.

Understanding the ITU standardThe original ITU loudness measurement

algorithm is shown in Figure 1. The audiochannels (except the LFE) are independent-ly filtered with a low -frequency roll -off to

simulate the sensitivity of the human ear anda high -frequency shelf to simulate head dif-fraction effects. The combined response ofthese filters is referred to as "K weighting"and is illustrated in Figure 2. (See page 54.)Surround channels are given a 1.5dB boostto account for the relative gain provided bytheir position on each side of the listener.The individual channel powers are summedto obtain the surround program's total pow-er. This is averaged over the entire program,yielding a single number metric for the pro-gram loudness. If a "dynamic" indication ofloudness is desired, a three -second movingaverage is typically used. Readings are re-ported in LKFS (Loudness, K -weighted, rela-tive to Full Scale) which may be thought of asloudness dBFS.

The ATSC recommendation specifies thatloudness measurements should focus ondialog or an alternate anchor element. Theintent was that viewers would set the dialogloud enough to be intelligible in their envi-ronment, and that maintaining constant dia-log loudness would maintain intelligibility.

LF

CF-

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K -filter

K -filter

K -filter

K -filter

K -filter

Meansquare

0dB

+15dB

+1 5dB

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Start/pause/reset

Short-termloudness

., Programloudness

Figure 1, Shown here is the original ITU loudness measurement algorithm. Audiochannels (except the LFE) are independently filte-ed with a low -frequency roll -off.

This assumed "well behaved" content (manycommercials don't fit this description), andalso depended on proprietary loudness mea-surement technology. In an effort to addressthese and other issues, the EBU PLOUDcommittee revisited BS.1770. Their work re-sulted in the 2011 revision of BS.1770.

This revision maintains the same filteringand power measurement method used in theoriginal standard, but changes the way mea-surements are averaged and presented. Theintegrator stage of Figure 1 is replaced with

S i ecial Re i ortaRITAI AllfilItSYSTFMS

+5dB

0dB

5dB

-10dB

-15dB

-20dB20Hz 100Hz 1kHz

Integrated loudness20kHz

intervals overlap by 75 percent,so a new value is obtained every100ms. Results are gated with astart/stop control to allow selec-tion of the audio segment to bemeasured. An absolute gate of-70LKFS is applied, which auto-matically eliminates lead-in andplayout portions of isolated au-dio segments.

The algorithm focuses onthe foreground portion of theaudio by a two-step averag-ing procedure (the orange ele-ments in Figure 3). The 400msmeasurement values are aver-aged over the content beingmeasured. The resulting LKFSvalue is decreased by 10 andFigure 2. K weighting, illustrated here, is the combination of results between

low -frequency roll -off and high -frequency shelf filter responses. used to gate the 400ms mea-surement values. This "relative

the processing shown in Figure 3. The channel gate" focuses the assessment on foregroundpower is summed over 400ms intervals. These sounds, the elements that generally dominate

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viewers' judgments of program loudness. Thevalues that pass the relative gate are averagedto form the final reading called "Integrated"loudness (abbreviated "I").

The standards also specify a true -peak me-ter. This is a device that measures the peak val-ue a digital audio waveform will reach when itis reproduced in the analog domain, or whenit encounters many forms of digital process-ing. To understand the problem,recall that digital audio represents acontinuous analog signal by a seriesof samples, taken at regular intervalsdetermined by the sample rate. AsFigure 4 on page 56 illustrates, thereis no guarantee that samples willland on the audio waveform peak.

However, these samples do repre-sent the underlying audio waveform,and when it is reconstructed, thepeak will be restored. This peak canalso occur when the samples are sub-jected to many types of processing-anything that introduces phase shiftor time offset - such as sample rateconversion, filtering or delay. If thishappens in the digital domain, thenew samples may clip, even if the

400ms average100ms update -70LKFS

Start/pause/reset

L

Average

Integratedloudness

1.

4,

Figure 3. This chart reflects theEBU PLOUD committee's 2011revision of BS.1770.The revisionchanges the way measurementsare averaged and presented.

original samples did notreach digital full scale. Be-cause many peak metersmerely display the maxi-mum audio sample, theyincorrectly gauge the sys-tem headroom.

The EBU recommen-dation introduces other

If a peak happens in the digital domain,

the new samples may clip, even if the

original samples did not reach digital full

scale. Because many peak meters merely

display the maximum audio sample, they

incorrectly gauge the system headroom.

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S l ecial Re l ortDIGITAL AUDIO SYSTFMS

Listeners' perception of loudness is best described with a

longer averaging time.The EBU recommends a running three-

second average that it calls the "Short -Term" loudness.

Figure 4. This graph shows there is no guarantee thatsamples will land on the audio waveform peak.

measures that are still under considerationby the ITU. Intended to assist mixers andprogram personnel in creating and character-izing content, their acceptance by the ITU isunlikely to impact CALM Act requirements.However, given their potential usefulness inproduction, it is helpful to understand them.

The EBU specifies "Momentary" loudness(abbreviated "M") as the stream of 400msmeasurements that drive the gating mecha-nisms described earlier. When displayed on ameter, they look much like a VU display sincethe 400ms averaging time is close to the 300msof a classic VU meter. Mix engineers watch thedisplay for help estimating the program loud-ness of live productions. Listeners' percep-tion of loudness is best described with a lon-ger averaging time. The EBU recommends arunning three -second average that it calls the"Short -Term" loudness (abbreviated "S").

The EBU also defines a measurementcalled "Loudness Range" (abbreviated LRA).This is derived from the Short -Term loudness

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56 broadcastengineering.com I Special Report supplement

using a relative gating process similar to thatdescribed above but with the gate set at -20.The LRA is the span from the 10 percent to95 percent points on the distribution of Short -Term loudness values that pass the relativegate. The LRA is descriptive of the programmaterial dynamic range. Using the 95 percentpoint allows occasional extremely loud events,while the 10 percent point ignores modest si-lent intervals during the program. If the LRAexceeds about 15, it is likely that viewers willbe unable to find a single volume control set-ting appropriate for the entire program.

Loudness meter evaluationThe test suite described here is avail-

able for testing BS.1770 compliance of anyloudness meter. It may be downloaded asa dScope III script or as .wav files fromwww.prismsound.com/loudness1770, andalso as a series of wave files and docu-mentation from www.qualisaudio.com.More complete documentation of the tests,

their design and their expected re-sults are included in the downloadpackage. Any new tests will be addedas they are developed.

The current 16 -test menu forthe script -based implementationis shown in Figure 5. All tests,except Test 2, are stereo signals andshould be applied to the LF and RFchannels of a surround loudnessmeter. All tests, except the firstthree, comprise a 1 kHz sinewaveat varying amplitudes. When theexpected result is a range ratherthan a specific target, this is due tothe 100ms alignment uncertainty.

Test 1 checks the accuracy of the True -Peakmeter. The initial waveform is a one -eighthsample rate, -6dBFS sinewave in which sam-ples are chosen to correspond to the sinewavepeaks. After three seconds, the frequencychanges for one cycle to one-fourth samplerate, and the amplitude increases to -2dBFS.

resT EXPEC ,,E,ULT HELP

1. True Peak Meet

2 Channel Gaon and Summation

3 K weighlmg and Calotsalion

4. Absolute Gahng

5.18eiatove Galng St1

E Relative Gating 02

7 EBU 3341 g3

8 EBU 3341 g4

9 EBU 3341 115

11. Loudness Range 81

1 Loucgoess Range St2

12. Loudness Range g3

13. Louisess Range 84

14 NLR Loudness Range

b. WIR Loudness Range

S. Relative Gate Value

.2 OdBES Peaks

Steady .23 OLKFS

Loudness ..23LKES E

..E2 to .7.7LKES Eto .22.0LKES

L7

[11LRA. 5 OLU

LRA15 CCU

IL -10 to -10 21.1(FS'

IL Y EZIELKES

=.23.0LKFS

IL Y -2.3.0VES

IL -23.0LKES

LRA v 10 OLU

5 OLU

20 OLU

.15 OLU

moce.quaiosaudio corn

Figure 5.This imageshows the menu for script -based implementation.The full suite for testingBS.1770 compliance of anyloudness meter currentlyconsists of 16 tests.

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Figure 6a.This graph shows changes to the -6dBFS sinewave after the frequencychanges for one cycle to one-fourth sample rate and the amplitude increases to

-2dBFS.The samples are chosen to occur 45 degrees off the sinewave peaks.

The samples are chosen to occur 45 degreesoff the sinewave peaks as shown in Figure 6a.When this waveform is properly interpolated(as would occur when reproduced in the ana-

Test 2 is Dolby Digital -encoded and stimulates all channels simultaneously,

including the LFE. If the meter incorrectly sums in the time -domain, the

reading will cycle. If the meter includes the LEE, the reading will be too high.

log domain), the result is the waveform shownin Figure 6b. All meters will read -6dBFS ini-tially. After three seconds, the reading shouldincrease to -2dBFS. Non -interpolating meterswill increase to -5dBFS.

Test 2 is Dolby Digital -encoded and stimu-lates all channels simultaneously, includingthe LFE. Power summation is checked by us-ing sinewaves of slightly different frequencies.A compliant meter reads -23LKFS. If the me-ter incorrectly sums in the time -domain, thereading will cycle. If the meter includes theLFE, the reading will be too high.

Test 3 checks the filter response at six fre-quencies: 25, 100, 500, 1k, 2k and 10kHz us-ing sinewaves of varying amplitudes to give aconstant reading of -23LKFS.

Test 4 alternates between -69.5dBFS and-90dBFS to exercise the absolute gating func-tion. A compliant meter reads -69.5LKFS. Ameter that does not implement absolute gat-ing reads -71.5LKFS to -72.7LKFS.

Test 5 steps the amplitude between -23and -6dBFS at intervals between 0.5 and 1.4seconds. A meter that correctly implementsrelative gating reads -7.7LKFS to -8.2LKFS; anon -compliant meter reads between -13.2 and- 13.5.

Test 6 checks aspects of relative gat-ing missed in test 5 by alternating between-36dBFS and -20dBFS. A compliant me-ter reads between -22LKFS and -22.5LKFS,whereas non -compliant meters read between-24.3LKFS and -24.7LKFS.

Tests 7, 8 and 9 are corrected implementa-tions of EBU Tech 3341 Test Cases 3, 4 and 5that evaluate basic loudness meter operation.

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The versions created by the EBU have slight er-rors discovered in the test suite's development.

Tests 10, 11 and 12 are implementations ofEBU Tech 3342 Test Cases 1, 2 and 3, whichevaluate basic loudness range meter operation.Each test takes 40 seconds and spends half itstime at each of two amplitudes 10dB, 5dB and20dB apart, respectively. The loudness rangemeter should read these same values.

Test 13 implements EBU Tech 3342 TestCase 4, which evaluates loudness range metergating. It spends 20 seconds at each of the am-plitudes -50dBFS, -35dBFS, -20dBFS, -35dBFSand -50dBFS. A compliant meter will ignorethe -50 amplitudes and read 15LU. If gating isnot implemented, the meter reads 3OLU.

Tests 14 and 15 are sinewave-based alter-natives to EBU Tech 3342 Test Cases 5 and 6,which are narrow -loudness -range (NLR) andwide -loudness -range (WLR) program clips.The NLR test uses amplitudes of -50, -40,-25, -20dBFS, -15dBFS, -20dBFS, -25dBFS,-40dBFS and -50dBFS, while the WLR testreplaces the -25 amplitudes with -35dBFS. Inboth tests, the -40dBFS amplitudes are main-tained for three seconds and the -15dBFS am-plitudes for two seconds, while the others have23 one -second durations. The durations at-40 and -15 test the 10 percent and 95 percentstatistical processing defined in the loudnessrange algorithm.

Test 16 measures the meter's relative gatethreshold, displaying -8 or -10 as appropri-ate. This is useful to determine if your meteris designed around the old standard.

If a loudness meter gives the expected re-sults for each of the tests above, the likelihood

Figure 6b.This figure shows what happens when the waveform is properlyinterpolated, as would occur when it is reproduced in the analog domain.

is high that the implementation is compliantwith the latest version of BS.1770.

Richard Cabot is the CTO of Qualis Audio. He waspreviously chairman of the AES digital audio mea-surement committee for the development of theAES-17 standard. Ian Dennis is technical director ofPrism Sound and, as vice chair of the AES digitalaudio measurements committee, wrote the docu-ment which became the true -peak meter specifica-tion in BS.1770.

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Special Report supplement I broadcastengineering.com 59

Si ecial Re ortDIGITAL AUDIO SYSTFMS

Fixed-point vs. floating-point

numbers in audio processing

In digital audio, using a fixed-point number format can

yield benefits in resolutionand dynamic range. 1

he traditional view is that the float-ing-point number format is superiorto the fixed-point number formatwhen it comes to representing sound

digitally. In fact, while it may be counter -intu-itive, there is a case to be made that the use offloating-point numbers yields lesser resolutionthan the use of fixed-point notation.

Floating-point numbers definedFloating-point numbers are like scientific

notation on calculators: They have a mantissa,the number part, and an exponent, a multi-plier used to scale the number part. For ex-ample, 1.414 x 103 is a floating-point numberwith a mantissa of 1.414 and an exponent of3. The attraction of this form of notation isthat it can be used to express numbers overa much larger range than would be possibleif the same number of digits were used in afixed-point (integer) number.

To understand the impact number formathas on digital audio systems, we must con-sider two properties: resolution and dynamicrange. Resolution or numerical precision isdetermined by word length. As it increases,

resolution improves. Dynamic range is alsodetermined by word length, but in the float-ing-point format, it can be dramatically ex-tended by the choice of exponent. Compare,for example, a 24 -bit fixed-point number thathas a dynamic range of about 144dB, to a 24-

bit floating-point number where eight bits aredesignated as an exponent. The latter has adynamic range of more than 1500dB.

Choosing the rightnumber format

To choose the right number format for adigital audio system, the dynamic range andresolution must be large enough to afford faith-ful representation of all audio signals that maybe encountered. Sound pressure level (SPL)is a logarithmic measurement of sound levelswhere 0dB represents the threshold of humanhearing. (See Table 1 for real -world examples.)

In general, music, sports and drama donot demand the full range of sounds - fromjet engine to gently rustling leaves. Nonethe-less, the 24 -bit fixed-point format remainsunsuitable for digital audio systems becauseaudio processing can introduce errors thatare manifested as audible noise unless addi-tional resolution is provided. Note that it isresolution, not dynamic range, that is needed.To illustrate this, let's take a look at the mostimportant audio processes: gain, mixingand equalization.

Gain, mixing and equalizationApplying gain to a digital audio signal

means multiplying by a big number for louderand a small number for quieter. The problemcomes when the product is a number thatdoesn't fit neatly into the number of digitsyou have to represent it. There is usually anextra bit that you have to get rid of, a processcalled truncation. How you truncate affectssound quality. Simply rounding up or downintroduces an unpleasant quantization noise,so a better idea is to add a random number

1

September 2011

to the leftover bits and then round up ordown. This idea is known as dithering, andit makes low -amplitude signals sound muchbetter. The downside is that the number for-mat must carry additional resolution in theform of footroom bits to dither against. Here,floating point does not help because there isno need for extended dynamic range. In fact,any bits given over to carrying an exponentare extraneous and would be better deployedextending notation in the mantissa.

If multiple signals are to be mixed (added),then it is a good idea to provide some addition-al headroom bits to extend the dynamic rangeduring the calculation, but this does not have tobe a large number. If a floating-point numberwere used to generate intermediate headroom,as the mantissa is scaled up, footroom bitswould be lost, affecting any subsequent dithercalculation and introducing noise. Hence, inmixing calculations, the floating-point formatis in fact a liability.

Equalization is a more complex case. Tomake it more convenient for processing, theubiquitous biquad calculation used in digitalfilters may be arranged in different forms. Forexample, the popular direct form II is usedto reduce the number of computations insystems where DSP cycles are at a premium.The tradeoff is that the resulting intermediatecalculations have such a large dynamic rangethat floating-point format must be used, atleast in a system constrained to a fixed wordlength such as a DSP chip. In other words, theneed for using floating point is a consequenceof cost-cutting rather than a consequence ofthe pursuit of high quality.

Limitations of a rigidfloating-point format

The problem that arises from the use ofa rigid floating-point format (for example,that found in ADSP SHARC chips) is that theresolution is fixed by DSP architecture, not bythe requirements of the calculation. Most ofthe time, this is adequate, but there are certainfilter configurations where it is not. The el-evation of the noise floor that results from theresolution limit of the floating-point formatis significant. (See Figure 1.)

If high -quality sound is the goal, the bestapproach is to first decide on the desirablelevel of performance and then select thenumber format to achieve it. In the case ofEQ, a high level of resolution is needed in

Real -world sounds

Calm breathing, or gently rustling leaves

Normal conversation

Passenger car at 10 meters

Hea-ing damage (long-term exposure)

Vuvuzela

Level of sound that can cause physicalpair

Jet 3ngine at 30m

M1 rifle at lm

Stun grenades

Sound levels

10dB

40dB to 60dB

60dB to 80dB

85d B

120dB

130dB

150dB

168dB

170dB to 180dB

parts of the calculation to avoid generatingthe kind of noise evident in Figure 1. A flex-ible architecture allows word length to beincreased so that it most precisely matchesdesired performance.

Using the same filter in an audio consolethat relies on fixed -point -based digital signalprocessing shows how the high word length,

Tabie I. here are someexamples of the dynamicrange of sounds inthe real world.

000100.00

-5.00-105.71

-10.00-111.43

-15.00L#5 -117.14

-20.00-122.86

-25.00-128.57

-30.00-134.29

-35.00-140.00

1.00Hz 10.00Hz

7.,

100.00Hz 1000.00Hz

fixed-point approach has reduced the noisefloor of the filter to almost exactly that of thetest set. (See Figure 2 on page 62.)

Lack of resolution produces a secondary ef-fect that also impairs filter performance. Thisis because you cannot add very big numbersto very small ones. Let me demonstrate thiswith an extreme case. Imagine that you haveadopted a seven -digit floating-point formatwith a four -digit mantissa and a three -digitexponent. You can represent the number

Figure 1 This plot shows theTHD+N for a 30dB notch filterat 50Hz executed in 40 -bitfloating-point (direct form Ill.

September 2011

S ecial RetortDIGITAL AUDIO SYSTEMS

000100.00

-5.00-105.71

-10.00111 .43

-15.00-117.14

-20.00-122.86

-25 00-128.57

-30.00-134.29

-35.00-14000

1 00Hz 1000.00Hz

Figure 2.This plot shows theTHD+N for a 30dB notch filter at 50Hz executed infixed-point notation.The blue trace is theTHD+N of Calrec's Bluefin2 processing.

The red trace is theTHD+N of the the test set looped output to input.

1 million by writing it as 1000 x 103. Now,add the number 999 to this. You should get1,000,999, but since you have only four digitsavailable, the result you end up with is 1000 x103 - the same number you started with. Inother words, adding 999 has no effect on theresult, an illustration of the limitations of thefloating-point system's lack of resolution.

In conclusion, it is inevitable that the use offloating-point numbers will deposit arithme-tic errors when they are subject to the math-ematics of audio processing. Candidly, theseerrors are small and often irrelevant, especial-ly when compared to the many other threatsencountered by audio quality on the path be-tween microphone and living room. Still, ifnumerical precision is our goal, as it shouldbe, then we should strive for objective analysisof the virtues of competing notational systems- floating point vs. fixed point - rather thanfailing prey either to intuitive explanations orthe common wisdom.

Patrick Warrington is technical director at Calrec.

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Processing audio for mobile DTV

As mobile device usagegrows, so has the needfor high -quality sound.

ust

Broadcasters have to understand the limitations of

their one-fourth rate (or less) transmission channel

and mix the content knowing it will be heard on

tiny earplugs or little speakers. Reducing dynamic

range is a way to ensure comfortable listening.

when audio mixing engineerswere finally getting comfortable with5.1 surround sound and other typesof multichannel audio formats for

sophisticated home and public theater use,along comes the need to correctly processstereo audio signals for portable devices withtin speakers.

The goal for con-tent distributors is

to create a consis-tent and predictableaudio experiencewithout having topreprocess every-thing upstream ofthe device. Sending

out the proper tags and metadata, the en-coder will then instantly know it has to makesure the content is rendered (decoded) prop-erly for the mobile device.

In most cases, the dynamic range needsto be more tightly controlled due to physicaland electrical limitations of cell phones, tab-let PCs and other devices. Couple these limi-tations with the fact that these devices arenormally used in very noisy environments

(trains and buses), and program audio canbecome unreliable and frustrating for usersif not mixed right.

Dolby Labs has done some testing of por-table mobile DTV reception devices, as part ofits ongoing research into this area. The com-pany's Dolby Mobile product (a suite of postprocessing technologies for mobile devices)includes a series of custom compression al-gorithms that are used to improve the audioexperience on mobile devices. Some of thesealgorithms can take a surround -sound mixand "virtualize" the 5.1 signals for renderingover ear buds, headphones and even a largerdevice's compact internal speakers.

One of the things Dolby's engineerslearned during their tests is that TV -basedcontent (movies, sitcoms, etc.) are mixed

terms of peak levels and dynamic range, thanMP3 music files. Modern music is typicallyproduced with peak levels frequently at orjust below clipping and the loudness averag-ing approximately 11dB below that. There-fore, it doesn't have a lot of dynamic range,whereas movies and TV content offer a va-riety of levels relating to dialogue, sound ef-fects and ambient noise. In this case, the typi-cal average loudness for this type of contentis in the -22 to -29 below clipping - 12dBto 18dB lower. (Remember, a 10dB differ-ence in loudness is approximately equiva-lent to a subjective doubling, or halving, inperceived loudness.)

Dolby's research in this area sought to es-tablish a new reference level for what soundedgood on small speakers, but also looked to le-verage metadata computed at the encoder up-stream (either in Hollywood or at a broadcastfacility). The decoder at the device could thenidentify the incoming bit stream and activatedifferent modes to get the best results.

The company has taken the two main legacydecoder operating modes, Line and RF (usedfor DVD and TV content), and developed anew reference level for portable devices thatare based on the same principals. It's called

wptember 2011

Portable Mode and is present in the latestgeneration of Dolby decoders and processingtools for mobile devices.

Jeffrey Riedmiller, Director, Sound Plat-form Group, at Dolby Labs, worked with afew of his colleagues last summer to developa comprehensive white paper that was pre-sented at the Audio Engineering Society con-ference in London in May 2010. It describesa nondestructive method for controllingplayback loudness and dynamic range onportable devices and is based on a worldwidestandard for loudness measurement as de-fined by the ITU.

They worked with a database of 25,000songs, along with hundreds of TV and movietitles, and found that when looking at music,the mean or medium of that large data setended up being about 11dB below maximumlevel. So, Dolby decided to establish -1 1 dB asthe new audio reference point for its decod-ers embedded within portable devices. Usingthis level, portable device manufacturers couldleverage the metadata contained in the latestDolby formats to non-destructively process thewide dynamic range content, such as TV showsand movies, in a manner optimized for thesedevice types all while providing a better subjec-tive match (in terms of loudness and dynam-ics) to music files being played. The result givesthe consumer a much improved (and moreconsistent) experience in terms of loudness,intelligibility and peak level control.

With this metric as a baseline, most en-coders can be used to control the level anddynamics of content processed with Dolby'stechnology and metadata generated with oth-er companies' technology as well. This newmetric is also being deployed inside Dolby'sMobile platform and its suite of core technol-ogies that are used for Dolby Pulse and DolbyDigital Plus decoder products.

Riedmiller said this new Portable Modealso makes things easier for mobile video pro-ducers to create content that complies withnew government mandates to protect againstaudio loudness. It does this while also bring-ing the dialogue out in front to make it moreintelligible (and enjoyable) when viewing innoisy environments. (This is especially im-port for live sporting events, where the crowdcan drown out the announcer if the mix is notdone right.)

Meanwhile, the associated metadata is stillattached to the media so that if a person plugs

his or her iPhone into a docking station, itcan play out that music or TV show on a largehome theater speaker system and sound great,with the originally mixed full dynamic range.

However, there area number of chal-lenges with process-ing audio signals formobile devices thatengineers should beaware of, accord-ing to Tim Carroll,founder of LinearAcoustic. He said thataudio programs mayvary from mono to5.1 channels, but thecurrent mobile au-dio pipeline is stereo(and mono in manycases). This requirescompatible and au-tomatic downmix-ing that is stereo- andmono- compatible.

Linear Acoustic offers its AERO.mobileproduct, which handles up to two separatemobile audio streams. It combines down -mixing with processing and psychoacoustictools to produce a controlled and intelligibleresult for mobile devices. The mobile pro-cessing will soon be available in the compa-ny's AERO.file file -based product as well.

The Dynamic range processing has to betailored not only for controlling range, butalso for protecting dialogue, and this requiresmanaging a different set of pyschoacousticchallenges. Carroll said that multiband pro-cessing, coupled with these new psychoacous-tic tools, could work extremely well. Atten-tion must also be paid to the codec itself asdifferent versions and mixes of encoders anddecoders can produce surprises.

Peter Poers, Director Sales & Marketing,at Jiinger Audio-Studiotechnik, said broad-casters have to understand the limitations oftheir one-fourth rate (or less) transmissionchannel and mix the content knowing it willbe heard on tiny earplugs or little speakers.Reducing dynamic range is a way to ensurecomfortable listening.

Junger makes a Television Audio Proces-sor that combines audio conditioning andencoding in the same unit. It also performsdynamic rangemanagement,loudnessandpeak

Linear Acoustic's AERO.mobile can downmixand process separate audio streams to providecontrolled, intellig ble results for mobile devices

;eptember 2011

S I ecial Re I ortAIGITAI A111:1111 SYSTEMS

level control, consistent spectral balance (bySpectral Signature), and metadata control forthe transport stream encoder.

Broadcasters also have to recognize the lim-itations of their legacy encoders. Poers said ifa station arranges and mixes for a "consistentspectral energy balance," the encoding resultswould improve by estimating low transmis-sion bit rates in the system.

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Therefore, broadcasters launching newMobile DTV services should talk with equip-ment vendors to ensure that they support newdecoder operating modes, like Dolby Portable,and also the generation of metadata in theirsystems. The encoder also has to generate theproper metadata.

This should not be a problem as the origi-nal MPEG and DVB specs have had bit fields

within it that carry audio metadataacross decoders. That said, manymanufactures have not implement-ed it to date.

1111111111111111111111111111111111111

Junger's Television Audio Processorcombines audio processing and

encoding in the same unit.

Standards are always key to suc-cess in broadcasting, and right nowconsistent use of a way to process au-dio for mobile devices is still evolv-ing. Yet, progress is being made inseveral places (including within theAdvanced Television Systems Com-mittee) as new ATSC A/153 -compli-ant encoders and decoders becomeavailable to help ensure contentsounds good and levels meet withconsumer needs. A satisfying userexperience is perhaps the most im-portant element to a successful mo-bile TV service. Being able to hear itcorrectly is a big part of that. IN

Michael Grotticelli regularly reports onthe professional video and broadcasttechnology indstries.

66 broadcastengineering.com I September 2011 I Special Report supplement

Adaptive, automated

loudness control:..

Today, we live in a world of automa-tion. You see it in our industry, inour daily working lives and even athome. Everywhere you look, there

is technology that is designed to make thingshappen faster by replacing human effortwith machines.

The broadcast industry is no exception tothis rule. Automation is everywhere, and thetrend to speed things up with technology isapparent, especially now that the Internet hasgiven us new ways to disseminate audio andvideo. The traditional technologies that weused to rely on for production and distributionare changing to encompass this new world andthis, in turn, is further reducing the influenceof human control. This begs the question: Willthese changes alter our experience of movingpictures and audio? Should we preserve the hu-man control of audio signals in such a highlyautomated broadcasting world?

Impressions of abroadcast facility

Whenever I imagine a totally automatedtelevision broadcast facility, two things imme-diately spring to mind. The first is what I callthe "fleece jacket syndrome?' No matter howwarm it is outside, everyone inside is wearinga fleece jacket or a sweater because the air hasbeen cooled down to such an extent that it feelslike a permanent winter. No wonder broadcast-ers are so keen to grab one of those fleece jacketswhenever a manufacturer is giving them away!

The second image I have is of the controlroom -a huge place with dimmed lights andpictures flickering away on massive video walls.No matter where you are in the world, thesecontrol rooms all look the same - incomingchannels, transmission channels, leased lines,etc. In front of this, there is inevitably a largedesk with a number of control panels andintercoms, as well as computer screens show-ing program lists, scheduling information,the condition of the transport streams and

perhaps the satellite uplink configuration. Ev-erything in there needs to be controlled, butwhere are all the people? Usually there are veryfew of them. The image of any individual chan-nel on the multiscreen display gives the channelname and visual information about the videocontent. There may alsobe some bargraph-likevisualization that tellsyou something aboutthe running audio. Theengineers who are pres-ent are surveying theoperation of all the dif-ferent technical gear in-volved in the process. There might be a stationfor quality monitoring of the audio and video,but you don't find this in every control room.This is what a totally automated broadcastingtransmission center looks like. This image is, infact, the reality for broadcasting centers, turn

Malaysian broadcaster Astrohas installed more than 30channels of JUnger Audio'sLEVEL MAGIC automatedaudio processing at itsfacility in Cyberjaya.

When I image a totally automated television

broadcast faciity, everyone inside is wearing

a fleece jacket or a sweater no matter

how warm it IS outside because the air

has been cooled down to such an extent

that it feels like a permanent winter.

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around facilities, uplink teleports, and cableand IPTV headends everywhere in the world.

If you look around one of these facilities andtry to find an audio fader or a video switcher,you will be disappointed because there aren'tany. Nor is there anyone there to control aswitcher or fader. In modern facilities, staff aremanaging data that are being stored on massiveservers with the unbelievable hard drive storage

of many terabytes. Programmed schedulers arerunning the playout for hundreds of TV chan-nels. The monitoring that is being carried outis primarily to control the basic condition ofthe running video and audio. There isn't anyquality monitoring in place.

In such an environment, how does oneensure that certain quality standards for au-dio and

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video are being maintained for eachdedicated transmission or distribu-tion network? This is a delicate issuebecause, in an automated facility,there is a need to achieve optimumresults while keeping human effortto a minimum.

In terms of picture content, thisisn't too difficult because through-out the production chain, the videosignal is subject to many dedicatedcontrols in order to keep it withinlegal transmission requirements.Video format conversion is com-monplace, and various levels of per-formance are available. Video is datathat is delivered in a compressedformat for the digital domain, andall those compression devices areperforming a number of video fil-ters and algorithms to guarantee thebest possible signal quality within anacceptable bandwidth for the trans-port stream.

However, when it comes to au-dio, things are different. Of course,there is technology on the mar-ket that is designed to convert theaudio into the format required bythe transport stream and to meetthe technical specifications neededto guarantee the best sonic perfor-mance. But that is where the simi-larity stops because, with audio,there is no common overall techni-cal specification that is designed tocheck or legalize the content.

Audio engineers do have the ben-efit of some technical recommen-dations, but sometimes this doesn'tsolve the problem. For example, ifaudio is coded into the digital do-main, the highest possible value inlevel should be OdBFS. But that ba-sic recommendation doesn't reallyreflect the wide variety of different,practical ways in which one can


deal with digital audio. Take, for instance, CDaudio. At 16 -bit audio resolution, CD audiouses almost all of the available coding space(theoretically 96dB system dynamic). Thismeans that if a broadcaster is getting musicfrom a CD, it will be controlled and masteredto reach OdBFS as its maximum value. That'svery different from a typical broadcast sig-nal, which now uses -18dBFS or -20dBES as

its alignment level in order to keep it in linewith the recommendations issued by inter-national regulatory authorities such as theITU, the EBU and ATSC. What this meansfor broadcasters is that audio content comingfrom different sources can have very differentlevel conditions. Content from CD audio canbe more than twice as loud as audio comingfrom a standard TV broadcast.

Differences in program loudnessare painful

As an industry, one thing we are all realizingis that the level of audio sources used in broad-cast transmission can vary wildly. Of course,nothing is transmitted that hasn't already beenprocessed and quality -controlled, but we stillhave a number of issues to contend with relat-ing to audio levels and audio control.

Over the last 10 years, broadcasters havebegun to understand that technical -orientedlevel control doesn't necessarily solve theirproblems when it comes to delivering betterquality audio transmission. Audio loudness is

now a hot topic, and there have been many ar-ticles published that discuss this issue and givethe background to it. In simple terms, all audiosources that have been processed to controlloudness should deliver the same overall loud-ness impression. And proper loudness controlis definitely improving the quality of audio indigital broadcasting systems.

But what remains an issue is how loudnesscontrol is applied in today's world of auto-mated broadcasting. Broadcasters have nochoice but to use integrated or external audioprocessing to perform this control. I can al-ready hear the complaints that some people,most notably skilled audio engineers, willmake in response to that statement. I knowthey will be asking how automated onlineloudness control can be pleasant to the earsand be done in a way that isn't detrimental tothe audio.

I can understand their concerns, but in aworld where automation is king, there is noother choice because broadcasters are notlikely to install an audio booth with propermonitoring and fadercontrol where someonecan sit and perform thetask manually. We haveto recognize that somekind of automated au-dio control is requiredif broadcasters are tocomply with the new

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True Visions, a cable antisatellite television provider inThailand and Southeast Asia,has installed LEVEL MAGICautomated audio processorsto balance audio levelsacross its pay -TV channals.

One thing we are all realizing is that the level

of audio sources used in broadcast transmission

can vary wildly. Nothing is transmitted that

hasn't already been processed and quality-

controlled, but we still have issues to contend

with relating to audio levels and audio control.

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Special Repert supplement I broadcastengineering.com 69

S l ecial Re! ortDIGITAL AUDIO SYSTFMS

In

Look aheaddelay

t

Processing

parameter Kx

Multiplier 1

T

Multiplier 2

o>

WeightingK2*Ust

T

Multiplier n

Tt

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Figure 1.The digitalimplementation of the

multiloop design permitsa short delay time to be

introduced into the audiosignal path.This allows thegain changing elements to

"look ahead" and determinethe correction needed before

applying it to the delayedsignal just in time to control

even the fastest transients.

loudness standards and recommendationsand maintain the highest level of quality fortheir audiences.

Given that there is no other choice, all thatremains to be discussed is what characteristicsthis online automated loudness control sys-tem should have.

Adaptive audio controlUnlike an ordinary line amp, the gain of

a loudness control processor is not constant;it varies with time depending on the specificcontrol algorithm of the loudness processorand the changing loudness and amplitudeof the input signal. These variations in thegain, which represent the real control process,should take place without any bothersome

side effects to the audio signal itself, effectssuch as pumping, signal distortion, sound col-oration or noise modulation. In other words,they should be inaudible.

The setting of the attack time parameterof a loudness control element affects how theunit will react to rapid amplitude changesin the audio signal. A long attack time canlead to overshoots (and consequent distor-tion) because the system is not fast enoughto reduce the gain. A short attack time mini-mizes the chance of overshoots, but themore rapid gain changes in such cases haveaudible side effects such as clicks and othermodulation artifacts.

Single -band and multiband designs are be-ing used to develop hardware and softwarecontrol units and, depending on the architec-ture, both designs are capable of doing a per-fectly good job. One advantage offered by asingle -band design is that it doesn't touch thesound as there are no filters involved.

The question is: How can one guaranteethe best possible performance if no one hasthe opportunity to realign the circuits whilethey are being used? In the absence of a hu-man, what is needed is a system that is adap-tive and performs like a human. Such an algo-rithm needs to change its behavior based onthe resources available, and for this you needa multiloop principle.

I suggest the use of a multiloop approachbecause we believe this is the best solution.

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The various loops each work over the entirefrequency spectrum (wired band). They workin parallel, each with a different set of attackand release parameters. Each loop developsa control signal, which is then summed withthe controls from the other loops to produce asingle gain control signal applied to one finalgain control element.

The way of summing the individual controlsignals requires unique technology. The digitalimplementation of the multiloop design per-mits a short delay time to be introduced into theaudio signal path. This allows the gain chang-ing elements to "look ahead" and determinethe correction needed before applying it to thedelayed signal just in time to control even thefastest transients. This is particularly importantfor the limiter, which provides a precisely leveledoutput signal absolutely free of overshoots (clip-ping). (See Figure 1 on page 70.)

The proprietary algorithms used in the mul-tiloop design also allow the automatic adjust-ment of the attack and release times accordingto the evolution of the input signal over time.This is called adaptive loudness control. By an-alyzing the incoming audio, the system can setrelatively long attack times during steady-statesignal conditions but short attack times whenthere are impulsive transients.

ConclusionDespite the views of some critics, adaptive

multiloop technology does provide a way to

integrate audio control systems into fully au-tomated broadcasting systems. It is obviousthat a file -based control philosophy will nevercover 100 percent of all broadcast transmis-sion content - not leastbecause television stillincludes live content,and even when tape-less technologies areemployed, this contentis still live. In a typicalbroadcasting facility, wewill continue to have amix of file -based and stream -based contentcontrol in order to guarantee proper outputsignal condition anytime and independent onthe source.

Regarding the tasks of the engineers in anautomated multichannel control room envi-ronment, any kind of signal processing mustbe a set -and -forget solution because no onehas the time or the resources to use a humanfor this task, especially not for audio. In thisscenario, the only way to create "automated"and "quality" real-time online processing al-gorithms is to use adaptive technologies. Pro-vided they meet the requirements of all thecurrent different standards and recommenda-tions, these adaptive technologies will alwaysguarantee the best audio control performancein a fully automated system. IN

Regarding the tasks of the engineers in

an automated multichannel control room

environment, any kind of signal processing

must be a set -and -forget solution because

no one has the time or the resources to use a

human for this task, especially not for audio.

Peter Poers is managing director of Anger Audio.

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Special Report supplement 1 I broadcastengineering.corn

TECHNOLOGY IN TRANSITIONNEW PRODUCTS & REVIEWS

IntercomsToday's intercom systems are high-tech.

What could be morefundamental thancommunications inan industry where

individuals collaborate to createcompelling content? But don't makethe assumption that intercom sys-tems are not high-tech these days. Wehave come a long way since the eraof carbon microphones and WesternElectric headsets, and what is fun-damental is that function has notchanged, only the sophistication withwhich we attack the technology. Ofcourse, this is largely due to tools thatsimply were not possible in the earlydays of television production. It is,however, remarkable that we still plugheadsets into the backs of camerasand use gooseneck microphones onrack -mounted communications pan-els as the user interface for commu-nications just like we did more than ahalf -century ago.

Two drivers are largely responsiblefor the wholesale change in commu-nications underlying the user inter-face mentioned above.

First, in the latter part of the 20thcentury, around 1970, a method wasdeveloped of using a duplex two -wire(actually three -wire) system for par-ty -line communications that carriedpower along with the audio. Party -line communications was nothingnew, and in fact was the basis of tele-phone communications for about100 years before RTS and Clear-Comimproved on the approach.

Around the same time, flexible sys-tems were developed that used activeand passive audio hybrids to allow ef-fective communications on multipleparty lines at the same time withoutjoining all of them together. This de-velopment of the "matrix intercomsystem" may have had even more farreaching impact. Every control roomtoday is built with private matrix

B r JUriN Lut-

At the Eurovision 2011 song contest, organizers EBU and NDR used multiple Riedelsystems, including Artist digital matrix intercoms, Performer party -line intercoms,and various digital and analog radios.

communications channels for differ-ent parts of the crew to communicateselectively instead of using one largeparty line system. I am sure this savesmany productions from devolvinginto chaos. It also allows problemsolving to move on in private whenit might otherwise have to wait for aquiet period. Best of all, matrix sys-tems allow things to be centralizedthat otherwise would require multipleoverlapping and redundant systems.

IFBTake, for instance, IFB. IFB can be

built without a matrix system andwas done that way from day one inpermanent studios. But by connect-ing talent earpieces through a matrixsystem, the production staff needs tohave only one microphone for com-munications. Using software -defin-able systems for switching the audiosignals allows them to be routedflexibly for monitoring, with inter-rupt or noninterrupt outputs easilycreated. It is the routing, ne matrix,aspect of systems like this that makehighly flexible and powerful systemspossible. Many modern systems alsoadopt digital displays for control

panels, which allows signals to benamed uniquely for each production- for instance, labeling the IFB forthe anchor with his name instead oftape labels or fixed electronic labels.

Sports and live productionNowhere is this more important

than in major sports production.Indeed, much of the advancementof matrix systems, and in particularthe highly flexible control and pro-gramming systems that have evolved,are a direct result of systems de-signed to support the needs of majorsports and location entertainmentshow production.

I had the fortune to work in thatsegment of the industry and saw therise of an "A2" position whose respon-sibility was just communications. Onlarge productions it is a full-time,high-pressure position. Over time, ithas grown to be a complex part of aproduction as well. Major produc-tions, like the Academy Awards, spendmonths planning every part of the in-stallation of communications, whichincludes multiple production unitsfor domestic and international pro-gram distribution, as well as facilities


Ukrainian broadcaster TV CompanyUkraine uses two Riedel Artist 128

digital matrix intercom systems at itsKiev studio.

to support single -camera standups(with IFB) for affiliates around theworld. Every department in a largeproduction needs private, thoughoverlapping, communications, in-cluding lighting, audio, producers,director's "camera" circuits, engineer-ing channels and more.

Wide area systemsBut there are applications today

that are far more technically interest-ing and that make use of more com-plex digital transport mechanisms.Large, distributed broadcast organiza-tions have complex communicationsneeds. I was first introduced to widearea systems like this when I starteda 30 -year association with Eurovisionin the late '70s. Even then, Eurovision(the European Broadcasting Union's"operations" arm) maintained a four -wire communications system thathad drops in every control room inEurope that originated or receivedshared signals. At that time, it wasanalog and hard to keep levels bal-anced. Today, Eurovision, and anylarge operation, particularly thosedoing news production from manyinterconnected sites, have a cryingneed to connect local matrix inter-com systems into a vast WAN inter-com system that facilitates complexlive breaking news production.

Solving this thorny problem to-day is made considerably easier bythe ability to connect matrices to-gether digitally to form a huge virtualmatrix. Audio is often connected asVOIP trunks, and IP-based controlsystems allow configuration of com-plex productions without involvingpersonnel in each site. It is possibleto pass mnemonics for labeling pan-els, configurations of who can speakwith whom, even mix minus selec-tions for IFB to be flexibly built withonly modest preplanning. Web -basedprogramming permits graphicalrepresentations of the topology aswell as simple Web service calls forsome operations. Leveraging IT tech-nology for voice communications isnot unique to media operations, butthe immediacy of our industry putsparticular stress on systems for qual-ity of network service and guaranteesof service availability. The PSTN canalways be a backup, but relying onvoice circuits would severely limitthe flexibility we have come to expectfrom modern communications.

I find it interesting that in an eraof smart phones and wireless every-thing that we have not more fullyadapted communications technol-ogy to wireless approaches. That isnot to say that wireless intercom hasnot been available for a long time,for it has. But why aren't we usingBluetooth headsets instead of ty-ing ourselves with copper leashes?Perhaps we give camera people atouch screen to adjust volume andmute their mics, or flash the name ofsomeone calling a camera on a smartOLED tally indicator. Why not allowforwarding a call to an absent opera-tor to their smart phone over Wi-Fiwithin the facility? A little creativitymight make the sophistication of to-day's communications technologylook simplistic in the future.

John Luff is a television technologyconsultant.

BE

111Send questions and comments to:[email protected]

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September 2011 broadcastengineering.com 81

EOMDEPARTMENT

Workflow's new worldSome are braving it now; it's inevitable for the rest.

BY ANTHONY R. GARGANO

The world of high tech, it

seems, is always rife withbuzz words, and certainlybroadcast is no exception.

One of the buzz words we hear to-day constantly being bandied aboutis "workflow:' Companies talk aboutproviding workflow solutions, increas-ing workflow efficiencies, end -to -endworkflow, workflow ad nauseum.

So, what is this new technospeak,workflow?

Don't be intimidated by the fancyterm. Underneath it all it is quite sim-ply the process that you undertake,from beginning to end, to accomplishwhat you need to get done. Break thatprocess down into its individual steps,and now you have a new technospeakterm: workflow elements. One such

format conversion.Today's broadcast facilities find

themselves dealing with an increas-ing number of video formats. Theseformats run the gamut from networkdistributed satellite content to user -

generated cell -phone video (and inquality levels that can range fromHD to Skype), not to mention every-thing in between. In this multiformatenvironment, a common workflowelement is the transcoding of thatcontent to a different format. Formatconversion may be required on theinput side for incoming material inorder to get it on -air. The days of sta-tion output being solely confined to"On Air" are long past. Today's broad-casters find themselves supportingthe requirements of multiple outputplatforms and, as a result, may be re-quired to recode that same content tomake it available for Internet stream-ing, for use on the station's websiteor for transmission via the station'sMobile DTV service.

For this particular workflow ele-ment, there are a number of offerings

with various levels of capability. Re-cently, I had the opportunity to lookat several of what might be termed en-terprise solutions for transcoding andformat conversion. The full extent ofoptions may perhaps seem like over-kill for a small local broadcaster today,but given the ever proliferating appli-cations for video (more and more ofthose are in a file -based infrastructure,thankfully), it might be just what youneed tomorrow. And, at the broadcast

Should your routinenot yet involve

formal workflowproducts, be

prepared because itwill someday soon.

network level or for station groupsthat use centralized aggregation anddistribution points for content, it maynot be overkill at all.

Three of the systems I looked atwere from RadiantGrid, Rhozet andTelestream. Some key parameters ofinterest in format conversion work-flow are speed, quality and the num-ber of formats handled. The best wayto evaluate any product is by dem-onstration in your own facility, usingthe content you have to deal with on adaily basis and evaluating the results inthe practical environment defined byyour own needs and requirements.

Speed, for example, is a functionnot only of the workflow productprocessing algorithms, but also it isregulated by the computer hardwareinstalled in a facility. Output qualityis going to be largely driven by thequality level of codecs used in theprocessing software but can be in-fluenced by the amount of time that

is devoted to processing the content.Need a quick turnaround? Shortenthe cycle, and normally one wouldneed to be prepared to pay the cost inquality level. Perhaps not though, ifyou are using RadiantGrid's technol-ogy. Its grid processing technique canslice the content into bite -size chunksand essentially parallel process thetranscoding effort across as manycomputer platforms as can be madeavailable on the grid.

Then, there is the horsepower raceof number of formats handled, whichis strictly defined by the manufactur-er. Give or take a few, Telestream canhandle about 120 different formats.Rhozet claims upwards of 150.

RadiantGrid offers the option ofcontainers, codecs, audio formats and

then mix and match in any way theysee fit to satisfy the needs of the re-quirement. In essence then, the actualnumber could be hundreds of formatcombinations.

Should your daily routine not yetinvolve formal workflow productsand systems, be prepared becausethey will someday soon be an integralpart of your workday life. File -basedworkflow is already driving someautomation systems from ingest toquality control to transfer to the play-

out server.If you would like to delve into

a more in-depth treatment ofworkflow and workflow process-es be sure to check out BroadcastEngineering's webcast series at:broadcastengineering.com/webcast.There is a range of excellent tutorialsoffered as live webcasts or on an ondemand basis. BE

Anthony R. Gargano is a consultant andformer industry executive.

ISend questions and comments to:[email protected]


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