WASHINGTON—FCC Chairman Brendan Carr has called for new spectrum auctions for wireless mobile as well as revisiting rules regulating TV loudness.

In a blog posted on Wednesday, the new chairman outlined his plans for the first FCC meeting scheduled for Feb. 27 and cited the aftereffects of Hurricane Helene to advocate for expanded wireless spectrum availability, particularly for first responders.

“Last Friday, I took my first official trip as FCC Chairman, visiting parts of North Carolina that were hit hard by Hurricane Helene,” Carr said. “I met with 911 operators who handled a sixfold surge in calls during the peak of the storm, first responders who led rescue and recovery efforts, telecom workers who quickly restored communications in difficult conditions, and local broadcasters who provided vital information to their communities. These visits only underscored the importance of robust, resilient, and affordable connectivity for Americans across the country. That is why we are taking action right out of the gate to get more spectrum into the hands of consumers—spectrum that can power new connections and innovations.”

To address these deficiencies, Carr said the commission will be “laser-focused on accelerating efforts that can get more spectrum into the marketplace. “

Carr characterizes the auction of AWS-3 spectrum licences—a band of RF used primarily for mobile broadband and public safety—as a “win-win.” He said he hopes to “kick-start” the process this month to complete it by June 23, 2026.

“It brings new spectrum into play for commercial use,” he said. “And the proceeds from this auction will also cover the costs of the national security initiative known as ‘rip and replace’—an effort that is removing untrustworthy technology, like Huawei and ZTE gear, from networks.”

In a move sure to get the attention of broadcasters, Carr also called for opening up additional portions of the C-band spectrum for 5G. Used mostly by TV and radio broadcasters for satellite downlinks, broadcasters have opposed efforts to expand usage of C-band over concerns of interference, particularly for ENG.

Carr hasn't been shy in advocating for more spectrum auctions. At last year’s Advanced Television Systems Committee (ATSC) annual meeting, Carr suggested that auctioning off additional broadcast spectrum would be on the block.

Carr also said the commission plans to vote on final rules to give emergency managers and consumers greater flexibility when it comes to Wireless Emergency Alerts, in particular, deciding when to issue alerts strategically in order to avoid so-called “alert fatigue” (aka “Boy Who Cried Wolf” syndrome).

Carr also said he will revisit the CALM Act, which was passed by Congress 15 years ago to require broadcasters to work with MVPDs to reduce the volume on TV commercials.

“Loud TV commercials are a frustrating headache,” Carr wrote. “You’re sitting there, chilling out, and then BOOM some commercial breaks in at a high volume. I don’t like them, and I’m pretty sure you don’t either. In fact, the FCC has recently seen an uptick in consumer complaints about excessively loud commercials.

“Back in 2010, Congress passed a law to address this issue, but given the rise in complaints I think now is the time for the FCC to revisit the issue,” he added. “Accordingly, we will consider a notice of proposed rulemaking to see if there are additional actions the Commission could take today to make sure TV viewers aren’t inundated by exceedingly loud commercials.”

The AES Technical Committee for Broadcast and Online Delivery will hold an in-person meeting next week during the AES Show in New York.

The one-hour meeting will be in Room 2D02 of the Javits Convention Center starting at noon on Thursday, Oct. 26.

Chairman David Bialik said the group and meeting are open to all.

“The committee creates technical documents and standards for broadcast and streaming. Most recently we released TD-1008, which has been elevated to the AES77 standard ‘AES Recommended Practice Loudness Guidelines for Internet Audio Streaming and On-Demand Distribution.’”

The group also recently created a loudness educational web portal to help educate the public about the importance of loudness. It currently is exploring the issue of dialog intelligibility for television.

BEACONSFIELD, U.K.—The National Film and Television School, a leading film, TV and game sound design school in the U.K.,, is relying on NUGEN Audio’s Loudness Toolkit to help educate students and deliver mixes that comply with today’s various output formats, NUGEN announced today.

“We don’t want to distract our students from telling stories, and at the same time we want them to learn how to use the modern tools necessary for their craft and feel at ease with them,” says Jeremy Rodeschini, senior supervising engineer at NFTS.

The toolkit, which includes NUGEN’s VisLM, ISL and LM-Correct, enable NFTS students to gain experience with audio measurement, analysis and correction technology, he said.

“With film soundtracks, the quality of the orchestration can be maximized by having a team focused on the narrative, a team where every member contributes to the story and collaborates while preserving a production’s artistic intent,” said Rodeschini.

Students also use the company’s intuitive software to maintain consistent, compliant loudness for television and online projects, NUGEN said. With the company’s VisLM software, students ensure consistent dialog and hit the appropriate delivery specifications of the TV production.

To adjust their final TV and online mixes, students use NUGEN’s ISL and LM-Correct. ISL’s True Peak limiting ensures their work is True Peak compliant. With LM-Correct, quick check analysis and correction can be performed to fix last-minute mistakes.

“When making mixes for YouTube and other online platforms, we restrict the dynamic range while mixing, which sometimes requires an extra overall boost or attention to make it consistent with the rest of our online deliverables,” says Rodeschini. “LM-Correct can correct these mixes quickly and easily, which is a massive time saver.”

More information is available on the NUGEN website.

The face of television audio has developed some interesting new wrinkles since we last looked at loudness—namely, immersive audio, object-based audio and audio for personal devices. We’re just beginning to unwrap the specifics and demands of these new delivery formats as well as learning how to produce for them, but loudness is an important factor of each.

The best news on the loudness front is that measurements are still based on the ITU-R BS.1770 standards we’re already using. Existing loudness meters remain valid, though some updates will likely be required. First up, we’ll take a fresh look at the foundations of loudness before we examine the impact the new formats will have on it.

LOUDNESS METERS
Loudness meters are neither volume unit (VU) nor peak meters, but are K-weighted meters designed to measure loudness in a manner that more closely approximates how humans hear. They display Loudness K-weighted Relative to Full Scale (LKFS), where each unit of LKFS (aka Loudness Units (LU)) is equivalent to one decibel.

Fig. 1: Loudness meter

Measurement is done by first applying a pre-filter that simulates how a spherical human head acoustically influences audio; then a second-order high pass filter is applied. Next, a mean square calculation is performed on all channels before they are summed, with each surround channel getting a 0.5 dB bump in the calculation and the LFE being dropped from it.

Gating was added in BS.1770-2 to ensure that quiet passages and silence don’t unduly influence the reading, so these gates make up the final stage of measurement. Samples are taken in overlapping 400 ms blocks, with the first gate at –70 LKFS, and the second at –10 dB relative to the first gated measurement.

THE MEASUREMENTS
As we look at the measurements audio engineers will actually encounter, it’s important to remember that all loudness measurements are made over time. Therefore, each measurement will fluctuate throughout the program and displayed values will rarely, if ever, remain static.

The Momentary (M) reading is an ungated measurement of loudness over a sliding 400 ms window and is a good representation of loudness at a given moment. Short-term (S) is also ungated and displays loudness over a three-second continuously sliding window. Integrated (I) is the gated overall loudness of content from beginning to end and is the measurement used to ensure CALM Act compliance.

NEWER MEASUREMENTS
A few less familiar measurements include Loudness Range (LRA), Peak to Loudness Ratio (PLR) and Maximum True Peak Level (TPL).

Loudness Range is the overall variation between quiet and loud sounds within a given piece of content, but it’s really a little more complicated than that. According to EBU Technical Document 3342, Loudness Range is “Based on the statistical distribution of measured loudness.” So the measurement looks at the entirety of the content and uses a combination of absolute and signal-dependent gates to ensure that individual loud or quiet moments don’t skew readings.

Peak to Loudness Ratio measures the dynamic range of a mix and is primarily used for music production, with some interesting potential as a broadcast tool for mixers who want a measurement of the dynamic range of their mix. PLR is a real-time measurement, usually applied to the entire piece of content, though it could also be used to determine the dynamic range of individual sections of audio. Higher PLR numbers indicate a mix with a wider dynamic range and values closer to zero indicate less dynamic range.

Maximum True Peak Level is simply the highest audio level measured at a given point within the content. It should not be allowed to exceed either the –2 dB TP specification in ATSC A/85 or the True Peak value of the content delivery specifications.

NEW FORMAT LOUDNESS
When it comes to measuring loudness for immersive and object-based audio formats, testing seems to indicate that the loudness of immersive mixes closely track the loudness of rendered 5.1 or 7.1 full mixes.

Fig. 2: MasterCheck Pro meter showing PLR measurement

In e-Brief 352, from the AES Berlin 2017 in May, the authors presented data showing the loudness variation between an immersive movie mix rendered into a Dolby E-AC-3 5.1 mix. When decoded into multiple Speech Gated Loudness configurations the variation was a maximum of 0.3 LU, and maximum variation of 0.7 LU when decoded and replayed into multiple Relative-gated Loudness configurations.

Additional evidence that current loudness tools will work for NGA audio formats can be found in the Atoms Production Suite manual, which states that, “You should use the 7.1 full mix re-render for loudness measurement during post production to ensure that content meets delivery specifications.”

Suggestions have surfaced for specific multichannel loudness tools for NGA formats, but if measuring surround mixes is equivalent to measuring NGA mixes, then these new loudness tools may not be necessary.

PERSONAL DEVICE LOUDNESS
The outlook is a little murkier for personal devices such as mobile phones and tablets because there’s so much variation between them. With differing acoustic and electronic outputs, the potential listening options seem almost limitless. Content services seem to recognize the need for loudness management because most have instituted normalization, though not at the same loudness level.

Fortunately, work is already in progress to address the technical needs of these devices. The AES has one study group working on loudness standards for streaming and the AGOTTVS subcommittee is working on loudness standards for OTT and video streaming. Both AES groups have published preliminary guidelines and have broad support from the broadcast industry and content services. However, the sheer variety of available devices makes the labor of both AES groups extremely involved and technically challenging.

A lot has changed in the 14 years since Dolby introduced the LM-100, the first broadcast loudness meter, yet ITU-R BS.1770 and its refinements are holding up well in the face of new audio delivery formats, proving that it was a wise choice for modern loudness management tools. We’ll likely see more refined measurement algorithms in the future, as well as additional specialized measurements, but for now it appears that the loudness tools we’re using will continue to serve us for the foreseeable future.

Jay Yeary is a broadcast engineer and consultant who specializes in audio. He is an AES Fellow and a member of SBE, SMPTE, and TAB. He can be contacted throughTV Technologymagazine or attransientaudiolabs.com.

Immersive audio took off in a huge way in 2014, especially in cinema with the release of films in Dolby Atmos and Auro 11.1 formats. The Traumpalast in Backnang, Germany was one of the first theaters to install a Dolby Atmos system, which includes a total of 57 speakers.

Here we are at the beginning of 2015, and as with all new years it is fashionable to analyze how the previous year went and take a look at how this new one might unfold. In this column dated Jan. 1, 2014, I made some predictions about what I thought the big stories of the year would be in the world of television audio. So let’s assess my prognostication skills and see whether they’ll hold up this year.

# 1: REFINEMENT OF THE CALM ACT—INCORRECT
My first prediction of last year was that the CALM Act would be further refined to make it easier to understand and implement, and this simply did not happen. There was almost no movement around the CALM Act this year other than the FCC finally adopting the 2013 revision, and the ATSC winning a well-deserved Primetime Emmy for A/85.

The rest of the year turned out rather quiet on the loudness front (pun definitely intended), with consumer complaints tapering off from an initial surge. Whether this is due to consumers actually being satisfied with the results of CALM implementation— or because they’ve given up reporting because they see no public vilifying of broadcasters over loud commercials—is something we don’t yet know; but most broadcasters seem to be doing a good job of keeping loudness under control.

It will be interesting to see what happens as initial two-year waivers expire; whether the FCC will allow extensions to those waivers due to the adoption of A/85:2013; and whether there will be any significant enforcement of the CALM Act. I think we’ll see some minor news regarding CALM this year, but it is unlikely to be anything significant.

# 2: AUDIO OBJECTS, NOT IMMERSIVE AUDIO—INCORRECT
It may not have been possible to be more wrong about this prediction than I was, and it’s partially the result of attending events where discussions take place regarding technology that remains years away.

Since audio objects for broadcast are tied directly to the rollout of ATSC 3.0, and since we won’t see a candidate standard for it until 2016, audio objects for broadcast are a technology of our future rather than our present. The work on this is so current that it was just a month ago (Dec. 5, to be exact), that the ATSC issued a Call for Proposals for ATSC 3.0 Audio Systems, with initial complete system submissions due by Jan. 12, 2015.

It appears that whatever the audio system ends up being in ATSC 3.0, it will be a complete solution from a single company as opposed to a system with technology sourced from many companies. Needless to say, we won’t see audio objects delivered via broadcast in the home for quite some time.

Immersive audio, however, has taken off in a huge way, especially in cinema with the release of films in Dolby Atmos and Auro 11.1 formats. With the inevitable Blu-ray and streaming releases of immersive films to consumers, we will undoubtedly see additional speakers, immersive sound bars and sound frames from a variety of manufacturers cluttering living rooms across the country in the very near future.

# 3: LOUDNESS MANAGEMENT FOR MOBILE, ONLINE DELIVERY—HALF CORRECT
Mobile and streaming is a bit of a conundrum since the devices used to access the content over the air and over the Internet can be the same, though different technologies in the device get used for each.

During a trip to South Korea last year I learned that essentially every mobile phone there comes with unlimited data; that watching television on phones, while riding the subway or driving a vehicle, is something everyone does; and that an awful lot of the phones have an antenna to receive over-the-air DMB television broadcasts.

Compare that to how people use their mobile phones to watch television in the United States, where the majority of viewing seems to be Internet streaming with very little obvious viewing of over-theair broadcasts (based on my observations from the daily commute).

With the big cellular providers making handsets with receivers available and accessory manufacturers providing plug-ins and add-ons for tablets and other devices, there’s no reason people can’t watch broadcast television while they’re on the go, but I still wonder how many people are actually doing so in the United States.

Of course, one big problem with television on the go, whether broadcast or streaming, is that external environmental noise means dynamic range of the audio must be kept to a minimum and loudness management is critical.

On Jan. 30, 2013, the ATSC released their recommended practice A/154:2013, which specifies –14 LKFS as the target loudness level for audio content. This means that a pretty serious loudness practice already exists for mobile television delivery. However, we still have nothing solid for streaming Internet delivery loudness that I’m aware of.

NPR conducted a study on this in 2013 and I’ve been taking some content measurements— both of which we’ll cover in another column—but the results leave me wondering whether this will be the next battleground of consumer audio complaints as viewing of streaming content increases.

THE MISSING PREDICTION
I’m still kicking myself for omitting audioover- IP and AES-67 adoption as an important technology for broadcast audio in 2014, possibly the most important one, in fact. I simply focused too closely on the other predictions and ran out of room and left it out even though I already had plans to begin implementing the technology myself.

Obviously AoIP took off last year with lots of AES-67 compatibility announcements, culminating with a very successful AES-sponsored plugfest in Munich.

After a bit of actual experimentation with AoIP products I found that they behaved pretty much as I expected with high-quality audio, but there were some discovery issues when connecting devices from different manufacturers together, even when using the same AoIP technology. Still, I have high hopes for AoIP and think adoption will continue to escalate this year.

As you can see from this analysis of my 2014 predictions, my crystal ball is on the fritz, so either take my predictions with a grain of salt or just wait awhile, because I have the feeling that all of them will eventually come to pass—my timing is just a bit off.

Jay Yeary is a broadcast television engineer specializing in audio. He can be contacted via TV Technology.

How do file-based and real-time loudness processors fit into an overall facility audio system? Each has its place. If a facility is primarily file-based, then file-based processors can be integrated into the overall front-end workflow to conform content to house loudness and true peak targets. Real-time processors, on the other hand, are most appropriately used at the end of a signal chain just before transmission to catch and correct any errant loudness errors.

FILE-BASED PROCESSING
For ingest, file-based loudness processing is generally set up as part of an overall video and audio workflow that can include transcoding, quality control and more audio-specific processes such as channel verification and assignment, Dolby E decoding and encoding and metadata verification, besides loudness and true peak processing.

Workflow is generally set up via software interfaces instead of knobs and buttons on a front panel. The processors are integrated into the file-based environment via IP infrastructure.

Linear Acoustic AERO.2000 audio/loudness manager Software-only loudness meters are available as plug-ins for digital audio nonlinear editing systems for the craft editor to use during the edit session. However, there is a school of thought that advocates using a file-based loudness processor on a piece after it has been edited.

“Glancing every 17 seconds [at a loudness meter] is antithetical to getting work done and stultifies the creative juices,” said Oliver Masciarotte, director of customer experience at Minnetonka Audio Software Inc., maker of AudioTools. “Let audio mixers mix and stop worrying about loudness. Mix and make it sound good. Then give it to something to take care of loudness and not screw up the mix. Do this after the mix is locked.”

While post-edit loudness processing could be done in the edit suite, it does tie up that resource, which is why taking it out of the NLE environment will probably make more sense for most facilities. The edited piece can be copied to a “finished jobs” folder on the network, which would put it in a queue for automated “back-office” processing.

Simon Pegg, senior software pipeline architect for RadiantGrid, said he’s heard concerns from audio mixers about the introduction of loudness metering into the edit suites, fearing that it would take away from their creative process. Pegg’s response was that loudness processing actually gives audio mixers the freedom to do their job properly.

Jünger Audio’s T*AP TV audio processor

“We’re not taking away your skills,” Pegg said, referring to audio mixers. “Rather we’re liberating you to use your skills. Let the loudness process be automated. You use your ears to create, and we do the technical stuff, so you play well with everyone else.”

REAL-TIME PROCESSING
Compared to file-based processing, real-time loudness processing is best used at the end of the audio chain before final transmission. In this application, a real-time processor acts as a final gatekeeper to make sure no potential loudness violation sneaks by.

“Typically you want a real-time processor to do nothing for most of the time,” Pegg said. “But if something goes disastrously wrong, clearly going to a violation, then you want it to take effect.”

If every piece of content is checked and made to conform to the correct loudness level up front, then real-time processing at the end of the signal chain can be light and audio quality will be minimally affected.

“Scale everything first [with a file-based processor], then the real-time processor can be set to do less,” said Tim Carroll, Telos Alliance CTO and Linear Acoustic founder. “Otherwise the real-time processor has to be set for the worst offending audio and then apply it to all programs.”

Linear Acoustic has the AERO series of processors for television, as well as software versions for scaling and optional processing and upmixing for file-based content. Also, licensed versions of its processing can be found in products from Wohler/RadiantGrid, Axon, Cobalt Digital, Miranda, Ross, Snell and others, Carroll said.

One of the limitations of a real-time processor is adapting the processing to different types of audio content. As Pegg said, “loudness correction has to be applied sensibly, or it can squash the signal entirely. You can get a good loudness number, but the sound [quality] will be poor.”

While real-time loudness processors typically have less flexibility than file-based processors, they don’t necessarily have to be stuck with a worst-case setting. These types of processors come with factory-defined presets that cover many scenarios, and usually allow custom presets to be saved. Much of this can be done with front-panel controls. Just remember that, unless in bypass, a real-time processor is always doing something.

It’s probably best not to set and forget with a real-time processor, or take the setting that comes out of the box, but to consider the types of programs that may have to be dealt with. A live musical event will probably need a wider dynamic range than a prerecorded situation comedy, for example.

Automation can be a solution. A playlist can be set to trigger different presets for different types of content. Peter Pörs, managing director, Jünger Audio GmbH, suggested sending a trigger to a real-time processor when content is switched to interstitials and then back again. “This clears the measurement memory from the previous program and starts the measurement all over again,” he said.

Jünger Audio calls its loudness solution Level Magic II, and can be found in such products as T*AP Edition eight-channel TV audio processor and D*AP4 LM Edition four-channel digital audio processor.

A real-time processor can also be put in bypass if a piece of content is known to have the correct loudness, but take care. Carroll said that even if a program is scaled correctly with a file-based processor, it could still have a wider dynamic range than the viewer’s system can comfortably handle. Since this could then engender viewers’ complaints, some real-time processing may actually be needed.

Another caution regarding automation is that if it somehow gets out of sync or was programmed incorrectly from the start, the wrong preset could end up being triggered— processing when not needed and vice versa.

If a real-time processor must be set with only one configuration, then try picking a setting somewhere in the middle between hard limiting and really wide dynamic range.

“If it’s too far in one direction, then Hollywood will complain; but if you go the other way, the consumer will complain,” Carroll said.

While a typical form factor for a real-time processor is a 1RU chassis, there are products available in a modular format, especially for facilities that generate multiple channels. Examples are Cobalt Digital’s Compass and Fusion range of modular products that have an option for the inclusion of Linear Acoustic AEROMAX 5.1 and/ or 2.0 channel loudness processing for at least 17 Cobalt openGear cards. Jünger Audio’s C8000 modular line includes the C8086+ eight-channel Level Magic II processor card for 5.1 or 2.0 mixed mode or four 2.0 channels.

Any processing will leave its effect on the audio signal, real time generally more so than file-based. So be aware if any of these undesirable audio effects are heard—distortion, pumping, reduced dynamic range, collapsed soundstage, image shifts from front to back and left to right, spectral skewing and reduced intelligibility.

With a collapsed sound field, “you lose the sense of depth and panning,” Masciarotte said. “The center channel may be lost or get buried. The sound may shift from front to back for no apparent reason.”

Remember, intelligibility can be compromised any time that gain or power spectrum is changed.

Used wisely, loudness processors can keep a facility compliant with the latest loudness rules and regulations and still produce good quality audio.

Mary C. Gruszka is a systems design engineer, project manager, consultant and writer based in the New York metro area. She can be reached via TV Technology.

The CALM Act is nothing new, especially to those who have been working on loudness management for years. The general assumption is that everyone “gets it” now, yet I continue to encounter others who don’t fully understand the rule and find themselves floundering with aspects of implementation, typically with measurement. In this column, we’ll look at how to measure content by delving into the official documents of the FCC Report and Order 11-182, FCC Order and Further Notice of Proposed Rulemaking 13-141, and ATSC RP A/85:2013.

METERS
Measurement of all content must be done with a loudness meter that measures LKFS and that preferably uses the ITU-R BS.1770-3 algorithm. The FCC R&O states that, “Compliance with the [recommended practice] requires industry to use the International Telecommunication Union Radiocommunication Sector (ITU-R) Recommendation BS.1770 measurement algorithm.” Gating has been added to the newer BS.1770-3 measurement for audio content measuring below −70 LKFS, and it includes a sliding relative gate for content that falls 10 dB below the absolute gated measurement.

This gating is designed to insure that quiet sections of audio don’t lower the overall loudness reading, which should help measured content match more closely. FCC document 13-141 explains it this way; “BS.1770-3 employs ‘gating’ that will exclude very quiet or silent passages of a commercial when calculating the average loudness of that commercial.” 1770-3 was adopted by the ATSC in the 2013 refresh of A/85, and while the FCC currently allows the use of 1770-1 or 1770-3, the newer version will likely become the official version in November 2014.

THE ANCHOR ELEMENT
The anchor element is whatever piece of audio the listener will be most interested in hearing when viewing the program and is critical to measuring content according to A/85. The anchor element will usually be dialog, but it could be music or another audio element if the program is not heavily speech dependent. A/85 defines it as, “The perceptual loudness reference point or element around which other elements are balanced in producing the final mix of the content, or that a reasonable viewer would focus on when setting the volume control.” In long-form content, the anchor element is to be isolated from other audio elements, measured, and its loudness measurement is considered the loudness value of the content.

MEASURING SHORT-FORM CONTENT
Short-form content is defined in A/85 as, “Advertising, commercial, promotional or public service related material or essence (also termed “interstitial” content). The typical duration is less than approximately two to three minutes.” To determine the loudness value of short-form content, measure the average loudness of the full mix of the content for its entire length. All short-form content is to be measured using this method.

The document does go on to note that this type of measurement could be a problem for wide dynamic range short form content because, “The louder elements of this type of material will increase the loudness measured with a long-term integrated method, and consequently reduce the perceived Anchor Element loudness after normalization. This can cause an unacceptable match to long-form material measured with an anchor-based method.”

No alternative measurement methods are suggested but if loudness-matching problems are experienced with wide dynamic range short-form content, remeasuring using the anchor element measurement method may solve them.

MEASURING LONG-FORM CONTENT
Long-form content is everything that isn’t short form and measuring it is not nearly as straightforward. When it comes to long-form content it’s all about the anchor element. A/85 suggests that during production and post-production the anchor element is to be isolated and measured. If it can be isolated, then a “Representative sample of the Anchor Element,” can be measured instead of the entire program according to A/85.

However, if the anchor element cannot be isolated then it is necessary to measure the average loudness of the entire length of the entire program. For long-form content that is already finished, “A section of the content that is representative of the Anchor Element (typically dialog) should be isolated and measured and reported as the Dialog Level of the long-form content.” A/85 goes on to state that if the anchor element cannot be determined and isolated enough to measure “The loudness of the element of the content that a reasonable viewer would focus on when setting the volume control should be measured and reported as the Dialog Level of the long form content.” If even that isn’t possible then the full mix of the entire program will have to measured.

Measurement during live production is covered briefly in A/85 but the process doesn’t seem clearly defined, though some excellent suggestions are made about mixing using a set control room volume level and focusing on listening with the ears rather than watching meters. Based on information in the document and from experience, a process similar to this should work well: Mix engineers should use a LKFS meter and measure the average loudness of each segment (or full program if continuous) while mixing to the predetermined target loudness for the program. At the same time, the mix engineer may choose to monitor a short-term sliding scale on the loudness meter to keep track of the loudness of the last 3–10 seconds as a guide.

MEASURING DOWNMIX
Measuring the downmix loudness of content has become critical because its loudness value may not match the loudness value of the surround original. Whether the loudness of the downmix and surround version of content match is dependent on where elements are placed in the soundfield. For example, placing the same source across all front channels can boost the downmix loudness by several dB, which means it may no longer match the loudness of adjacent content.

This is an extremely condensed version of how to measure content according to CALM but I hope it makes measurement a little bit clearer for anyone having trouble with it. A/85 and the FCC documents can be a bit daunting, but they contain valuable information that is worth the time you’ll spend reading through them.

Jay Yeary is a television audio engineer who spends his days working for a large media corporation. He can be reached through TV Technology or via Twitter at @TVTechJay.

File-based loudness processing, as one might expect, operates on audio files, either standalone, or extracted from an audio/video file such as MXF. The strength of a file-based processor is that it can gain foreknowledge about an entire audio file (or segment)—start to finish—before doing any processing. Because of this, file-based processing could potentially be less intrusive than its real-time counterparts.

Typically, file-based processing happens in multiple passes. First, the audio signal is analyzed, then—based on preset rules—the processor determines what type of functions or operations is needed and then applies them. Depending on the adjustments needed, further analysis and processing steps may occur in an iterative fashion, to arrive at the final result. While this may sound time-consuming, typical processing times are faster than real time. What kinds of measurements and processing are typical of file-based loudness processors?

DEFINED LOUDNESS TARGET
The first is fairly obvious—loudness. Adjusting the audio signal to reach a defined loudness target is fairly straightforward for file-based loudness processors. Unlike a real-time processor, which effectively rides gain throughout, a file-based processor inserts either attenuation or gain. if it’s needed, to reach the target loudness. This is done after performing a loudness measurement on the file, per ITU recommendation ITUR BS.1770-3, “Algorithms to measure audio programme loudness and true-peak audio level.”

Fig. 1: Example of a file-based workflow that includes loudness measurements and adjustments, and true peak limiting for the AudioTools family of products (Courtesy of Minnetonka Audio Software, Inc.)

This single gain shift (scaling factor) doesn’t change the dynamic range. “It’s like adjusting the gain control on a fader and leaving it alone for all of the content,” said Bob Nicholas, director of international business development for Cobalt Digital in Urbana, Ill.

In addition to loudness, ITU-R BS.1770-3 describes how to measure another parameter, true peak level. Both file-based and real-time processors can provide this measurement. According to ITU-R BS.1770-3, “true-peak level is the maximum (positive or negative) value of the signal waveform in the continuous time domain; this value may be higher than the largest sample value in the 48 kHz time-sampled domain.”

If only the sampled peak value were used, problems such as inconsistent peak readings, unexpected overloads, and under-reading and beating of metered tones could occur. Again from ITU-R BS.1770-3: “The problem occurs because the actual peak values of a sampled signal usually occur between the samples rather than precisely at a sampling instant, and as such are not correctly registered by the peak-sample meter… [The] use of a true-peak indicating algorithm will allow accurate indication of the headroom between the peak level of a digital audio signal and the clipping level.”

DYNAMIC RANGE CONTROL
File-based processors can be used for other related functions as well. One example is dynamic range control (as with the AERO. file option for RadiantGrid or the Wohler loudness appliance solution). Another is measuring and controlling maximum short-term loudness, maximum momentary loudness, loudness range, and dialog level (as with Minnetonka’s AudioTools family, which includes AudioTools Server, AudioTools FOCUS and AudioTools Loudness Control for Harmonic ProMedia Carbon, for another example.)

Simply scaling an audio file, even though it meets a loudness target, may not be enough to stop viewer complaints, if dynamic range remains wider than their systems can handle, as Tim Carroll, Telos Alliance chief technology officer and Linear Acoustic founder, pointed out. That’s the reason for adding dynamic range control to loudness processors. For a theatrical release movie, for example, dynamic range may need to be narrowed to make it a better fit for TV, while a sitcom may not need much, if any, dynamic range adjustment.

File-based processors are capable of making a variety of measurements to determine how to adjust the final output to reach a predefined target. This can often be an iterative process, as one adjustment may affect another parameter. Fortunately iterative processes are well-suited to software-based products. If the target isn’t reached after a certain number of attempts, the processor can send out a notification for human intervention.

File-based loudness processing is typically just one part of an overall file-based ingest or quality control workflow that can include audio up or down-mixing, Dolby E decoding and encoding, metadata adjustment, channel assignment detection and conforming, watermarking, pitch and time control, sample rate conversion, channel management (muting, copying, reconfiguring, replacing), and third-party functions and integration, according to Oliver Masciarotte, director of customer experience at Minnetonka Audio Software.

Fig. 2: Example of a more complex file-based workflow that includes multiple languages, program correlation and channel order verification, downmixing, loudness measurement, processing and peak limiting for the AudioTools family of products (Courtesy of Minnetonka Audio Software, Inc.)

As an example, the AudioTools family can integrate with Telestream Vantage and Harmonic ProMedia Carbon. File-based processing is also amenable to complex automation, according to Masciarotte.

The audio functions can occur alongside any video file-based functions such as transcoding, aspect ratio, standards conversion or quality control.

BRANCHING AND ITERATIVE WORKFLOW
Branching and iteration are keys to efficient and flexible workflows in file-based processors. Let’s look at a couple of workflow block diagrams for AudioTools to see how this works.

Referring to Fig. 1, start at the left side of the block diagram, where the audio essence is extracted from an MXF file. The audio is then checked for the presence of Dolby E. If Dolby E is detected, then the channels are first decoded to PCM before being sent to the measurement and loudness normalization stages. According to Masciarotte, loudness measurement and control, for the most part, needs to happen with PCM (pulse-code modulated) digital audio signals.

While not shown in Fig. 1, the workflow could be set up with another branch to check if the loudness meets the required target. If “yes,” the signal would continue on to the next stage, but if not, the audio channels would be subjected to further loudness level adjustments and measurements.

After the loudness adjustment sections, the workflow checks to see if the signal needs to be converted to Dolby E. Depending on the answer, different true peak limiting is applied. If encoding is required, it happens after the true peak limiter, and the signal gets re-wrapped into the MXF file.

Fig. 2 shows a more complex workflow that includes multiple languages, program correlation, channel order verification, and downmixing, in addition to loudness measurement and processing and true peak limiting.

With software-based systems, workflows such as these are defined according to each customer’s specific requirements. According to Masciarotte, these types of file-based systems can be modular to some extent, scaleable, and easily interoperable across WANs, MANs, LANs and SANs. The file to be processed can be local or remote depending on the system.

Some IT knowledge is typically needed to set these up and to ensure a good user experience.

Mary C. Gruszka is a systems design engineer, project manager, consultant and writer based in the New York metro area. She can be reached via TV Technology.

As a practical matter— to ensure compliance with the CALM Act or to produce deliverables to a specific loudness target—stations and content-producing facilities are relying more on loudness processors to automatically make adjustments to the audio content.

There are two broad categories of loudness processors: real-time and file-based. Within each type, loudness processing can be a stand-alone function or incorporated as part of a total processing package.

Real-time loudness processors operate in nearly real time (with some buffering and processing delay) to meter loudness of an incoming signal (mono or multichannel) and then typically continuously make audio level adjustments depending on the rules or presets it is given.

Fig. 1: General signal flow of a multiband real-time loudness processor, Linear Acoustic Aeromax

A file-based loudness processor, on the other hand, analyzes loudness from audio that was recorded as a digital file, typically as part of a video file. The analysis occurs over the entire length of an audio piece and then a scaling factor (gain or attenuation), if needed, is applied to the entire content, based on preset rules, so that the audio output is delivered at that new (target) loudness level.

REAL-TIME PROCESSORS
While they can be used for treating loudness on archival material on tape (instead of in a digital format), real-time processors are typically installed in the last stage of the audio signal chain, before an encoder, to catch any loudness problems that would make the program non-compliant.

A real-time processor fixes loudness “by adjusting dynamic range,” said Tim Carroll, Telos Alliance CTO and Linear Acoustic founder.

The processor continuously measures program loudness of the input audio signal according to the ITU-R BS.1770 standard (currently version 3, from August 2012). Then automatic gain control, or in other words, compression, is applied to adjust the level to meet the target. Think of this as riding gain with an audio fader.

Fig. 2: Comparison of multiband (left) and wideband multiloop (right) block diagrams for real-time loudness processing

“The [processor] is constantly adjusting the level,” said Peter Pörs, managing director, Jünger Audio GmbH. “The fader is never in a fixed state. It’s moving so slowly that you don’t perceive it.”

For content at relatively controlled levels, adjusting the AGC too fast will be audible, yet if a loud transient occurs, the processor must be able to react quickly to pull it down. This is why processors typically have an output limiting stage.

WIDEBAND AND MULTIBAND PROCESSING
Processors differ in how the AGC is applied. Some apply AGC across the entire audio spectrum (wideband). Others use the multiband approach where they break up the full audio spectrum into sections or bands and apply AGC individually to each band. Different attack and release times can be set for each band.

According to Carroll, if the processor runs in straight wideband this, in general, can make processing adjustments more audible. As an example, a loud thump could bring down the level of a whole program, even though it’s only the lower frequencies that caused the level to spike. With multiband processing in this scenario, only the lower frequencies would be reduced, (Fig. 1).

Four or five bands generally are adequate for multiband loudness processors, according to Carroll, and this is what is typical.

Another idea behind multiband processing is the way us humans perceive sound. We don’t hear linearly across the audible frequency range. We are more sensitive to mid-range sounds compared with those of higher and lower frequencies, but the difference changes as the audio level changes.

For example, for normal hearing at low audio levels, a sound at 100 Hz must be raised about 15 dB higher than one at 1,000 Hz for the two tones to be perceived as equally loud. As audio levels increase, the lower frequencies don’t need to be raised quite that much compared to the mid-range to be perceived as equally loud.

Two researchers from Bell Labs, Harvey C. Fletcher and Wilden A. Munson, studied this phenomenon and in the early 1930s published their results with graphs of equal loudness curves across the audio spectrum and at different audio levels. These have come to be known, not surprisingly, as the “Fletcher-Munson curves.”

Fig. 3: Block diagram of Jünger Audio Level Magic loudness management

That’s why if you compress the entire signal you change the relative levels of the high and low frequencies to the mid-range, and that, according to Bob Nicholas, director of international business development for Cobalt Digital, changes the character of the sound. This can have a negative effect on intelligibility.

“Multiband is not trying to keep things spectrally flat, but to keep things spectrally balanced,” Carroll said.

Nicholas said that multiband AGC is more applicable to a sound source that’s a mix of different signals and wideband is more for single source signals, like that used on a channel strip of an audio console.

Taking a different tack, Pörs said that the multiband approach can produce anomalies when the different frequency bands are summed together. “The possible difficulty is that [with] the overlapping zones of the filters, a precise summation of the signals is nearly impossible [and that] leads to coloration,” he said. “That’s why we came to wideband.” (See Fig. 2.)

Wideband with a twist, that is. “We have a different processing design approach,” Pörs said. “We call it multiloop design.” This design incorporates a series of gain controls, with each “fader” controlled separately. (See Fig. 3.)

“The various loops each work over the entire frequency spectrum,” Pörs said. “They work in parallel, each with a different set of attack and release parameters. Each loop develops a control signal which is then summed with the controls from the other loops to produce a single gain control signal applied to one gain control element.”

The algorithms in the processor provide automatic adjustment of the attack and release time based on how the input signal changes over time. “This is called ‘adaptive dynamic range control,’” Pörs said. “By monitoring the waveform of the incoming audio, the system can set relatively long attack times during steady-state signal conditions, but very short attack times when there are impulsive transients.”

In addition, the Jünger multiloop design allows for a very short time delay to be put in the audio signal path. “This lets the gain-changing elements ‘look ahead’ and determine the correction needed and to apply it to the delayed signal just in time to control even the fastest transients,” Pörs said.

No matter what design, it must be set and used correctly. More on this later.

Mary C. Gruszka is a systems design engineer and consultant based in New York. She can be reached via TV Technology.