Radio World July 2020 eBook: Trends in Codecs & STLs

What role are codecs playing in the new world of broadcasting?

July 2020 From the Publishers of Radio World

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Trends in Codecs & STLs (in the New Normal)

July 2020

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Welcome to Radio World’s latest ebook.

Like so many of the technology products familiar to the radio plant manager, codecs (and the remote, distribution and STL applications they serve) have seen a great deal of change in the last decade. Or, perhaps more accurately, several waves of change.

IP has established itself as omnipresent and Brett Moss reliable. The long-established copper infraRadio World Gear structure has been decommissioned. Soft& Technology Editor ware-based codecs have moved from twitchy app projects to core products, some reaching into all-digital audio ecosystems.

In putting this ebook together, we sought input from equipment manufacturers and users in the field, asking them to describe recent trends, discuss how the technologies are being used, and predict where it’s going next.

The experts we interviewed bring perspectives from organizations including 2wcom, AEQ, Barix, Comrex, Cumulus, Educational Media Foundation, Entercom, GatesAir, In:Quality, Multitech Consulting, SCA, the Telos Alliance and Tieline.

The transportation of high-quality digital audio has never been easier.

4 A s Remote Audio

26 C odecs Open Up “Hard-

9 C odecs Make More

28 C odecs Offer Redundancy,

12 D elivery of Service Is

29 I ncreased Quality, Less

13 A EQ Codecs Deliver

30 D igital Networks Must

Evolves, Fidelity Reigns

By Tom Hartnett

to-Reach” Sites With Bryan Waters

Robust Connections With Tim Neese

Backup and Failover With Eric J. Fitch

Key for Planning With Dan Jackson

Bandwidth With Shane Tove

From Minsk

By Sergio Sanchez

Facilitate Operations

By Anke Schneider

17 R emote Broadcasting 32 U se Case: Four Programs in Your Browser

By Paul Kaminski

to 14 Locations

By Alfonso Lopez

20 M ore Choices Than

33 T he Codec Has Become a

Ever for Radio STL

By Kirk Harnack

Chief executive Zillah Byng-Thorne Non-executive chairman Richard Huntingford Chief financial officer Rachel Addison Tel +44 (0)1225 442 244

Swiss Army Tool With Charlie Gawley

TRENDS IN CODECS & STLS Radio World | July 2020

3

As Remote Audio Evolves, Fidelity Reigns The next step involves finding ways to leverage existing systems into the studio

By Tom Hartnett consultants to know that telephone audio is grungy, thin and fatiguing to listen to at length. If the competitive challenge is to avoid listeners hitting the “next station” button, then maintaining listenable audio throughout your programming should be the primary goal. At the same time, it’s incredibly important that stations engage in their community (and monetize their brand) with remote broadcasting. Technology has helped combine these goals.

I’ll spare the reader a detailed history of this science, but a list of recent technology is helpful. Dedicated loops (when telephone tariffs reigned supreme), RPU radios, frequency extenders (maturing to multiphone line models), ISDN and POTS codecs each saw their era of popularity and utility wax; and each waned for its own reasons.

Something new was always available that was more cost-effective or easier to procure.

Tom Hartnett is technical director of Comrex.

Although it has been around for decades, FM broadcasting remains the most popular audio medium.

A lot of the reason FM thrives, despite the attempts to create a “better” digital alternative, is technical. FM was defined with technical standards that deliver a low-noise signal that allows for easy reception in most environments. But more than that, FM was defined as having deviation standards that allow for an audio bandwidth that covers the majority of the hearing spectrum.

Sure, modern audio media bests FM in frequency response and signal-to-noise, but the fidelity of FM remains “good enough” for the vast majority of listeners.

Much more than features like stereo imaging and dynamic range optimization, it’s the fidelity of FM that keeps listeners engaged. The ability to hear the funky bassline along with the high-hat cymbal or to derive the emotional nuances of a speaker’s voice is what makes FM radio shine.

If the competitive challenge is to keep listeners from hitting the “next” button, maintaining listenable audio throughout your programming should be the primary goal.

ADVANCING FIDELITY

But any broadcast airchain is only as strong as its weakest link. With digital recording and production, it’s relatively easy to make a great-sounding in-studio product. But generating live, remote audio has always come with its own set of challenges and costs.

Too often, broadcasters have been willing to compromise on the fidelity of remote feeds for the sake of cost and convenience, airing live audio from telephones. Telephone systems, by design, convey only the fraction of audio spectrum required for intelligibility. They filter out lows to avoid noise pickup, and they filter out highs for reasons having to do with the dated economics of 20th century digital telephony.

Comrex has built a company, and I’ve built a career, finding alternatives to live telephone audio for radio broadcasters.

It doesn’t take any scientific studies or high-priced

But the main objective — fidelity — was always either equaled or improved.

We all use IP pretty much exclusively for live out-ofstudio audio these days, due to ubiquity and cost. And luckily, IP makes carrying higher-fidelity audio feeds easy.

Audio coding science has come a long way and implementations are now cheaper and lower-power. Wireless IP has made the remote broadcaster’s dream a reality.

It’s now possible to carry a handheld, battery powered device into the field and generate programs that rival the sound of in-studio sources.

TRENDS IN CODECS & STLs Radio World | July 2020

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Audio Frequency Bandwidth Comparison OPUS encoder: 20 Hz - 20 kHz FM Audio Bandwidth: 30 Hz - 15 kHz Modern HD voice encoders (G7.22, AMR-WB, EVS): 50 Hz - 7 kHz Telephone: 300 Hz - 3kHz

0

5 kHZ

10 kHZ

15 kHZ

20 kHZ

Fig. 1

Mobile phone audio quality is a longtime frustration for broadcast. For programming with listener call-ins, there’s been a routine need to disconnect callers who are unintelligible. This makes programs suffer and wastes valuable airtime. But even here, we see that the industry has realized there’s not always a need to stick with legacy low-fidelity audio (see Fig. 1).

As mobile phones and networks mature, it’s becoming increasingly common to experience high-fidelity “HD Voice” calls between mobile callers. Modern audio encoders like G.722, AMR-WB and EVS are integrated into late-model phones, and the voice-over-LTE networks that support this traffic quickly are replacing the legacy networks. Several carriers are able to cross-connect high-fidelity calls between them, expanding the number of users who experience HD Voice on calls.

On VoIP and mobile networks the existing challenge is the same: There’s no easy way to “bridge the gap” and bring this high-fidelity audio into a broadcast studio reliably. So even when calls originate from these advanced networks, the caller audio is converted into the thin fatiguing sound we all know, in order to be compatible with legacy “bridging” systems (see Fig. 2).

So the next step in the evolution of high-fidelity remote audio for broadcast clearly involves finding a way to leverage existing systems into the studio. While that work is underway, there’s already one existing tool that can be used today to improve telephone audio.

BRIDGING THE GAP

So game over, right? What could possibly come next?

Problems remain to be solved. We still air telephone audio from listeners. Setting up a remote broadcast can still be a challenge for the nontechnical. And specialized audio encoding gear has significant cost.

Meanwhile, nonbroadcast industries have discovered that offering “toll quality” audio for communication isn’t good enough. Like broadcast, a competitive edge can be had by offering an experience with higher audio fidelity. The recent boom in video chat apps proves this point.

While audio challenges exist in that world with regard to echo cancellation and delay, fidelity has never been an issue. Developers saw early on that high-quality audio needs to be part of any system from the ground up. FaceTime, Skype, Zoom, Teams, Messenger and Duo all use high-fidelity audio encoders.

Voice-over-IP systems, now common in office environments, aren’t constrained by the legacy telephone system within their borders. They can by default deliver high-fidelity audio encoders when talking exclusively over their LANs. Even on a relatively poor audio system like a telephone handset, the difference between an in-office call and out-of-office call can be startling.

This is because calls outside the LAN must convert the fidelity of the audio to the “lowest common denominator,” which is the legacy phone system.

TRENDS IN CODECS & STLs Radio World | July 2020

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Fig. 2

WEBRTC

When I introduced the concept of WebRTC to broadcasters several years ago, it was a hard sell at first, as it was difficult to describe in a concise sentence. But don’t be afraid of the scary technicalsounding name.

WebRTC is essentially a video chat app that’s built into virtually every web browser, whether desktop or mobile. It’s an open standard and allows anyone to create a video chat service without requiring any software installation on the participant’s system. That’s because the critical pieces are already in the browser, waiting to be “woken up.”

Like other video conferencing apps, WebRTC uses a high-fidelity audio encoder by default. This encoder is called Opus, and it’s becoming the de facto standard for live web conferencing.

Because WebRTC doesn’t require the video part of a call, every web browser, both desktop and mobile, can now be considered a high-fidelity audio encoder using Opus.

Using WebRTC can be as simple as subscribing to an

By using WebRTC you’re leveraging the power of developments that were never intended to be used for broadcast. audio-only service provider like ipDTL, Cleanfeed or SourceConnect Now. This will require a pro-grade audioready computer at each end of the link. The Comrex Opal provides a pro-grade hardware solution that handles all the complexity within its server box.

Either way, by using WebRTC you’re leveraging the power of developments that were never intended to be used for broadcast. This is the way things have been done for decades — from POTS codecs, ISDN to IP, broadcast always finds a way to leverage new developments for their unique requirements.

We’ll continue to do that as existing “HD Voice” networks converge and interoperate. Maybe someday soon the goal of banishing telephones from the radio will come to pass.

TRENDS IN CODECS & STLs Radio World | July 2020

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Codecs Make More Robust Connections Technology brings us more locations via more methods at lower cost than ever broadcasters, codecs are the “magic” device that has allowed them to continue normal programming in a completely remote fashion.

Tim Neese is president of MultiTech Consulting Inc., a multi-faceted broadcast technology consulting and contracting firm.

RW: How many ways are there of making connections?

Neese: The number of connection transport methods has decreased in recent years.

Not long ago, it was possible to purchase codecs that could connect via POTS, cellular, ISDN, T1 and Ethernet.

In some cases, all of these connections were available via a single codec.

As telcos have begun to phase out ISDN, T1 and traditional POTS circuits Tim Neese. “Data connectivity via mobile in favor of newer transport technolphones and devices is now as common as patch panels in facilities once were.” ogies, codec manufacturers have focused on these technologies as well.

While the traditional connection choices have decreased, newer technology connections have become available in more locations, via more methods and at RW: How are today’s technologies solving problems in lower cost than ever before. creative ways?

For instance, data connectivity via mobile phones and Neese: One of the most common issues is studio and devices is now as common as patch panels in facilities transmitter sites that are unable to be linked via tradionce were. Numerous codecs are able to leverage that tional (RF) point-to-point methods. This, combined with connectivity via either a physical or wireless connection the ongoing sunsetting of the telco ISDN and T1 infrawith the device or as a software application that runs on structure, has propelled connection of these sites via the the device itself. public internet to commonplace.

Today’s codecs and encoding algorithms have proven RW: What would you like manufacturers of these to be more than up to the task of making those connectechnologies to add or offer in future? tions viable and reliable.

Neese: I would like to see more manufacturers include advanced security tools and options like firewalls, inteRW: What role are codecs playing in this new world of grated VPNs and secure web configuration services at-home broadcasting? within their codecs. That, I believe, would allow for more Neese: In the new world of broadcasting where, for secure codec deployment via direct connection to public some, every live segment is what was once considered networks and provide even greater deployment flexibilia “remote” broadcast, the codec has become as essenty for broadcasters. tial as a microphone or pair of headphones. For many Radio World: What’s the most important trend in the design and performance of codecs for remotes or STL? Tim Neese: Most codec manufacturers are incorporating and continuously improving stream redundancy and error correction techniques that allow for significantly more robust connections. These techniques allow codec users to take advantage of readily available transport methods and the public internet to make reliable connections for both remote program contribution and studio to transmitter links.

TRENDS IN CODECS & STLs Radio World | July 2020

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Delivery of Service Is Key for Planning Dan Jackson of SCA also sees trends moving from “boxes” to a software and op-ex model Gone are the days of running OB trucks and lengthy lead times on ISDN installs. Content teams want to be able to move at the drop of a hat.

Dan Jackson is head of audio operations for SCA, Australia’s biggest entertainment company. Its multimedia assets include more than 100 radio stations.

RW: How are today’s technologies Radio World: In this ebook we are solving problems in creative ways? exploring trends in the design and Jackson: Moving to IP-based techperformance of codecs. nologies has decreased reliability on Dan Jackson: At SCA we are a big links, so SmartStream and dual link believer in standards to ensure codecs have allowed codecs to be interoperability between products. used in place of expensive microWe like to talk in terms of SMPTE wave links, where historically the IP 2110 and AES67 for audio distribuservice is no good. tion, and AES67/SNMP for control.

As for codecs we tend to change RW: What role are codecs playing for our choice based on the application; we still use aptX for voice and AAC, Dan Jackson. “Gone are the days of running OB you in at-home broadcasting?

Opus and Tieline MusicPlus for audio trucks and lengthy ISDN installs. Content teams Jackson: During COVID we had want to be able to go as soon as possible.” around 20 shows at 50 locations distribution. using mostly Tieline VIAs. Their Cloud One of the most important trends Codec Controller was an excellent add-on as it allowed is delivery of service. As we move away from dedicated our engineers to manage the codecs remotely.

POTS and ISDN lines and onto 4G/5G services, we are at Historically we have been fairly flexible with our talent, the mercy of the tower operators. Having the ability to so broadcasting from home was not new for us. Holiday double-deliver the data through the same or multiple carhomes, pregnancies and simply being in another state riers ensures that content is delivered 100% of the time. have required us to have the capability to broadcast We’ve also noticed a shift away from having extremely remotely. low latency links, gone are the days where talent are monitoring off air, so things like Forward Error Correction RW: What recent features are offered that other engineers are welcomed to help ensure the quality of service. may appreciate?

Another important trend I see is the shift away from Jackson: Time-zone delay is a great feature. In a country traditional “boxes” and capital expenditure. We are seethat has five time zones delaying the content from the ing a large demand for a software and op-ex model, head end can be costly so codecs with time zone capamuch like the rest of the IT world. bility delay ensures maximum efficiency. Another great One of the biggest issues we faced during COVID was feature is the addition of control data as opposed to more the inability to scale our codec fleet. Companies that are traditional “pulses,” our playout system can now talk taking advantage of WebRTC like ipDTL are certainly pavmachine to machine so the decisions are a lot smarter. We ing the way for a software/op-ex future. even delay the data so it’s time aligned with our audio.

When we talk about remote codecs for remote broadcasts, the more portable, durable and simple, the better.

CONTINUED ON PAGE 35 TRENDS IN CODECS & STLs Radio World | July 2020

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Banks of AEQ Phoenix Stratos codecs installed at Minsk Olympic Stadium.

AEQ Codecs Deliver From Minsk A refurbishment project expands a stadium’s media capabilities

By Sergio Sanchez

The author is R&D engineer at AEQ and was the technical coordinator for the project described.

to take advantage of the growing media world.

Dynamo National Olympic Stadium is the largest multipurpose stadium in Minsk, Belarus. It was reconfigured a few years ago so it could hold not only football matches but also athletics and other kinds of events, making into a venue with an international scope. It was reopened in December 2017, in time for setup of the 2019 European Games.

This international profile is reflected by the attendance of sports journalists from all over the world for live broadcasting. With that expanded mission, Dynamo National Olympic Stadium managers selected AEQ commentary system with 150 positions, each one equipped

At times it seems that “media” is everywhere.

Smartphones provide immediate content from seemingly every corner humans visit. Drones and small GoPro-style cameras add another omnipresent dimension. The world is awash in multimedia material that can be turned into usable/sellable content.

Designers of new buildings already pack them with media-friendly digital infrastructure. Hoping to take advantage of an opportunity, many owners of aging, media-constrained venues are refurbishing their facilities

TRENDS IN CODECS & STLs Radio World | July 2020

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•

•

Sergio Sanchez with the AEQ Olympia commentary unit.

with an AEQ Olympia commentary unit. This allows more than 150 international TV and radio stations to broadcast events from Minsk simultaneously. Each of those can customize its language, style and media personalities.

The whole stadium offers Dante-AES67 AoIP multichannel technology provided by AEQ equipment. A powerful AEQ BC2000D audio matrix was installed, allowing for centralization and distribution of all the audio signals produced in the building.

In order for the commentary positions to communicate with the rightsholder stations or RHBs for each event, 150 AEQ Phoenix Stratos audio codecs were installed.

Each one is able to establish two bidirectional circuits between the station and the commentary units. One of them is normally used to carry the program and program feedback, while the other commonly operates as a coordination channel.

Each codec has front-panel controls; however, it would be chaos if each codec operator could take control of the system. Fortunately there is a control software available so AEQ’s ControlPhoenix software application is being used to establish and supervise the up to 300 simultaneous connections that can be established with the 150 audio codecs.

AEQ Stratos is a suitable audio codec for this purpose because: • It is a dual codec, so the same device can establish the program and coordination circuits.

• It is developed according to EBU N/ACIP standards, so it can communicate with most any third-party codec that may be in a broadcast facility inventory.

• Besides operating with IP networks, it can also establish connections using synchronous ISDN and even V.35 lines, allowing for an alternative path to establish a link.

• Stratos audio codec includes several state-of-theart audio algorithms, such as Opus, the new stan-

dard for broadcast-quality audio communications, as well as legacy codecs for ISDN communications and guaranteed third-party compatibility, such as G.722 or MPEG2.

It offers both analog and AES/EBU digital input/ output connectivity with external synchronization capability, plus ancillary data transport, seamlessly emulating RS-232 protocol over the IP audio stream and GPI/O connections for external signaling.

ControlPhoenix software provides full remote control of codec installations ranging from one to several hundred Phoenix family devices. Control is possible locally or from anywhere in the world, if so required, using an internet connection. This software application is provided for free with all AEQ Phoenix codecs while a license is only necessary to simultaneously manage more than two devices. It includes call list management, full configuration, real time VU meters, audio alarms, event logging, etc.

Some of the most important international sports events held in the remodeled Minsk stadium include the Second European Games and The Match Europe vs. USA track and field competition.

The Second European Game featured 4,000 athletes from 50 European countries competing in 15 sports.

For those games the real-time broadcasting production used a total of 12 OB vans and more than 200 cameras. Seven hundred broadcasting professionals ensured that the signals were properly produced and reached the 190 rights-holding broadcasters around the globe. More than 600 live and nearly 800 recorded coverage hours were produced.

The Match Europe vs USA, conducted in September 2019, gathered 300 world-class athletes who competed in 37 different track and field events.

Eurovision Sport was the event’s broadcasting partner.

Eurovision used the Phoenix Stratos audio codecs in order to send the audio from commentators speaking all the different languages throughout infrastructure.

Dynamo National Olympic Stadium in Minsk, Belarus.

TRENDS IN CODECS & STLs Radio World | July 2020

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Remote Broadcasting in Your Browser The ipDTL service takes on a higher profile during pandemic

By Paul Kaminski A subscriber of the service can send a link to another location, where that link is opened in a browser and a bidirectional studio quality audio link would then be established. ipDTL has additional capabilities: With the proper configuration, the subscribing user can also connect to legacy ISDN codecs (where still available) and even connect to a voice grade telephone. With SIP protocols and using a sip.audio account from In:Quality, a subscriber can also connect to hardware codecs (e.g. Comrex, Tieline, Telos, JK Audio) that are configured for Opus connections.

It is also compatible with G.722 and G.711 over IP. There is a video angle as well, VP8 and H.264.

And it lives on Windows, Mac, Linux and ChromeOS computers.

Paul Kaminski, CBT, is a longtime Radio World contributor.

Working from home or remote studios is in “normal” times a challenge for broadcast and voiceover talent.

When one factors in a pandemic lockdown and the ensuing scramble to move studio-quality audio back and forth, a service like ipDTL can make that task less of a challenge.

The ipDTL service has been enjoying more attention during the pandemic but has been around for several years. It comes from In:Quality, which “operates a worldwide network for the real-time transmission of professional audio.” The company says its users include the BBC, New York Public Radio, NPR and Global Radio.

The service is based on the open source Opus codec.

Founding Director Kevin Leach, a former radio host and BBC sound engineer, says ipDTL runs smoothly on any modern computer: “If you can browse the internet smoothly on your computer, then you can run a stable ipDTL connection.”

SUBSCRIPTION LEVELS

There are three levels of ipDTL annual subscriptions: Bronze users can send one connection link, Silver users two and Gold users may send up to four simultaneous connections. A version allowing six connections is in the late stages of testing.

The Gold subscription also includes basic video functionality. Prices start at $15 per month. Subscribers get a sip.audio address (XXXXX@sip.audio), which allows SIP-enabled devices to talk to the subscriber.

How does this work in the real world? I tested the service recently during a virtual NAB Show demonstration with Leach.

There are some caveats from my experience.

ipDTL screen image.

TRENDS IN CODECS & STLs Radio World | July 2020

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An online network map lets the user find a studio, search for voice talent or see which radios stations are “ipDTL ready.”

service. With a wired gigabit connection over Cat-6, I experienced no connection problems.

Leach says multiple SIP connections with ipDTL at the studio end provide maximum flexibility. He said devices such as Wheatstone SwitchBlade and Comrex Access MultiRack can receive multiple duplex real-time SIP streams from remote sites and guests. The studio so equipped can send a connection link to talent and guests so they can connect easily to the SIP hardware.

There’s also an option to connect from one of the company’s new range of SIP Opus Codecs.

Subscribers may opt to be listed in a database of ipDTL users around the world for an extra charge. That’s a useful resource for audio reporters and producers, and it could also help those producers and reporters to connect with newsmakers and subject matter experts.

“For too long now, there has been uncertainty and trepidation about the migration from ISDN to IP codecs,” Leach said. “With SIP now, it feels like we’re finally past the point of no return, but there’s still some work to do. A newspaper journalist should be able to ask a radio producer what SIP address they should call for an interview, and get a confident reply. Looking at the messages in our support inbox, we’re not quite there yet.”

Paul Kaminski hosts produces msrpk.com’s “Radio-RoadTest” program. Twitter: msrpk_com; Facebook: PKaminski2468

Subscribers may opt to be listed in a database of ipDTL users around the world for an extra charge.

All codec developers (software and hardware) caution users that sending a true mix-minus is vital for proper operation of the codec. For example: On a mixer like the Allen+Heath ZED 10, there are three ways to send a mix-minus (aux send, FX send and record bus).

The best one that seems to work with USB connections is the record bus, where inputs other than the USB connections are selected (mics, etc.). The USB out can be fed from the record bus. If the main mix were fed to the USB out, that would create a feedback loop. A standalone codec could be fed mix-minus from the aux or FX send.

During our demonstration, I fed the record bus with the USB output. When Leach talked about the importance of a mix-minus, I created a feedback loop (which happens when the codec or other receive channel is fed back to the other end of the connection).

Unless a laptop is within visual distance of a wireless router, and it is the only device on the network at that particular time, in:Quality strongly recommends a wired connection to the network router when using the ipDTL

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More Choices Than Ever for Radio STL Unprecedented technology growth expands options for quality and price considerations

By Kirk Harnack By the end of the 20th century, both wired and wireless STL systems began having some serious challenges.

Telcos became less interested in supplying equalized program circuits — so much so that most stopped accepting orders for new circuits. And, in major markets — often with clustered tower sites — the 950 MHz RF band had no room for additional users.

Today, it’s nearly impossible to even order new T1/E1 service, and observant engineers can easily see the deterioration of their telco’s outside wire plant.

In some markets, ISDN usage became “unlimited,” so a few broadcasters employed either dial-up or “nailed-up” ISDN service with ISDN codecs at each end. These days, ISDN service is withering and new connections are generally not available.

Despite the disappearance of old telco services and the congestion in some areas of the 950 MHz band, engineers actually have more STL choices than ever before.

The key to new STL options is internet protocol — IP. And the best news about IP is that it can be transported in more ways than you might think.

Let’s edit and append our list to strike through nonviable options and add new options:

The author is senior solutions consultant for the Telos Alliance in Cleveland.

Radio World reports there are more than 22,000 licensed radio stations in the United States. Of these, more than 15,000 are tallied by the traditional accounting of fullpower AM and FM stations. Low-power FMs, translators and FM boosters add about 7,000 more.

The vast majority of these stations employ some kind of audio link from the studio or other origination point to the transmitter site. That’s a lot of studio-transmitter links.

CHANGING TECHNOLOGIES

What kind of STL was used by the first radio station where you worked? Forty years ago our main STL choices were these: • Equalized program lines from your local phone company; • Discrete (L/R) 950 MHz (band) TX/RX radios — one for mono, two for stereo; • Multiplex (MPX) 950 MHz (band) TX/RX radios.

More choices arose during the 1980s — digital choices — such as the QEI CAT-LINK, which transported FM MPX over a T1 or E1 telco link, and Dolby’s 950 MHz STL transmitter/receiver pair. More audio data-reduction codecs allowed further choices in the 1990s, both in wired and wireless STL systems, including stereo and even multichannel digital STL systems.

Most of these used either T1/E1 telco circuits or worked in the 950 MHz band in the United States, and other UHF bands for other countries. The 21st century brought upgraded digital RF STLs as well as a variety of budget-priced analog composite systems.

1. Equalized program lines from your local phone company; 2. Discrete (L/R) 950 MHz (band) TX/RX radios — one for mono, two for stereo; 3. Multiplex (MPX) 950 MHz (band) TX/RX radios; 4. ISDN codec (came and went over a 30 year period); 5. T1/E1 link (unavailable for new installation and dying quickly);

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traditional and IP-based STL systems.

6. Digital, discrete (L/R) multichannel 950 MHz (band) TX/RX radios; 7. Satellite delivery (L/R) using audio codecs; 8. Discrete (L/R) audio using IP audio codecs with appropriate IP connectivity; a. Public internet; i. Fiber; ii. Cable; iii. DSL; iv. 4G LTE carrier; v. Wireless ISP; vi. Satellite; b. Managed, wired IP link; c. Private, wireless IP link; 9. Linear, discrete (L/R) audio using AoIP (Livewire+, AES67) with appropriate IP connectivity; a. Managed, wired IP link; b. Private, wireless IP link; 10. FM Multiplex (MPX) using µMPX bitrate-reduced multiplex technology (for FM); a. Public Internet; i. Fiber; ii. Cable; iii. DSL; iv. 4G LTE carrier; v. Wireless ISP; vi. Satellite; b. Managed, wired IP link; c. Private, wireless IP link; 11. FM Multiplex (MPX) using linear multiplex technology (for FM); a. Managed, wired IP link; b. Private, wireless IP link.

PURPOSE-ENGINEERED STL SYSTEMS

Purpose-built RF STL radio systems are generally considered to be reliable. They tend to work well 24/7/365 throughout their service life expectancy of perhaps 10 to 20 years. They tend to be thoughtfully engineered and well-built.

They should be, as some of these STL systems cost well over $15,000 for just the radios. Add-ons may include additional audio channels, ancillary data, serial transport, etc. Add to this the cost of antennas and coax cable for effective service at 950 MHz and a full-featured, purpose-built radio STL system can top out at $30,000 or more, plus tower crew installation. A full backup system will likely double that capex expense. The good news is that there is likely little or no continuing operation expense (op-ex), unless tower space and/or building rooftop rental is required for a given installation.

Lower-end RF STL radio systems can cost far less than the example above. A basic FM MPX STL radio pair can be purchased for under $4,000, though the antenna/coax/ placement expenses would be similar to those above.

Audio quality of RF STL systems can range from “acceptable” to “pristine.” Purely analog RF STLs balance between wideband system noise and demodulated distortion. Any distortion of any type in the system will end up being demodulated at the STL receiver and retransmitted by the main FM signal.

Traditional digital RF STL systems deliver crystal-clear audio with a very low noise floor. However, many of them are limited to about 15 kHz of top-end audio transport.

The brick-wall filtering required is, indeed, audible on some of these systems when A/B compared to 20 kHz STL systems in which only the FM audio processor is doing pilot-protection filtering.

Finally, unless the digital RF STL offers linear audio transport, there will be some kind of psychoacoustic coding algorithm involved. And most likely the aggressive, multiband FM audio processor will follow this coded audio in the chain. The net effect is exaggeration of any coding artifacts in the station’s audio.

Some key benefits of 950 MHz RF STL systems are: • Robust, reliable operation; • Independence from third-party service providers; • Op-ex costs ranging from “reasonable” to zero; • Excellent expected equipment lifespan; • Digital systems that offer very clean audio with low noise.

That’s quite a list of options we have now. And, yes, several sub-items regarding IP transport are duplicated.

However, that’s to make a point: As our society’s reliance on IP connectivity grows, so also grow our options to acquire and use it on a reliable and professional level.

KEY CRITERIA

We engineers generally have three key criteria for our studio-transmitter links: Reliability. We’re seeking at least five 9s of reliability.

That translates to about 26 seconds per month or 5 minutes and 15 seconds of downtime over an entire year; Audio quality. Every moment of audio that we broadcast to the public will traverse our STL. There’s no room for any compromise here; it really must be perfect; Cost. We should consider both capital expense and operational expense (cap-ex and op-ex).

Let’s consider each of these three criteria, comparing

Some downsides to 950 MHz RF STL systems include: • RF congestion that implies interference potential;

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• • • • •

•

Lack of RF coordination, which continues to result in occasional problems; Susceptibility to malicious interference; Higher capex costs, potentially much higher; One-way communication — no inherent return path; Psychoacoustic coding algorithms producing artifacts that are exaggerated by aggressive audio processing; Audio sampling (bit depth) likely only 16 bits.

categories — linear and “coded.” Another disambiguation of IP-based STLs could be “wired” and “wireless.” However, we can also use wired and wireless IP transport methods at the same time, or with one backing up the other.

Linear IP STL systems will transport digital audio perfectly with the same 1s and 0s that go into such an STL being delivered to the far end. Twenty-four-bit, 48 kHz-sampled linear audio typically requires about 2.5 Mbps to transport. While this audio delivery is basically perfect, it typically depends on having zero packet loss.

This zero-error scenario is often provided through redundant IP paths between the endpoints.

Coded systems will use some kind of coding algorithm to reduce the bitrate required for transport. The audio codecs in such STL systems are generally configurable from a wide range of bit rates, and perhaps even a selection of coding algorithms. Bitrates that are appropriate for broadcast audio range from about 128 kbps up to 756 kbps, depending on the chosen codec. Modern coding algorithms offer some “error concealment.” For example, the AAC family of algorithms claims inaudible error concealment with up to 20% of random packet loss.

Linear IP STL systems are particularly good for broadcasters in that they are absolutely transparent to the audio. Every sample of audio data from the studio arrives exactly the same at the transmitter site. The IP transport path must provide excellent packet delivery performance in a linear IP audio system. Indeed, a common approach to implementation is to simply extend the audio over IP network at the studio out to the transmitter site. With this approach we assume the AoIP audio channels we need at the transmitter already exist on the studio’s AoIP network. By providing a qualified IP path to the transmitter site, we can “subscribe” to the desired audio channel(s) from the studio.

One example of such a system is at Delta Radio in Greenville, Miss. The studio and rack room were fully Livewire AoIP already. A robust 5.8 GHz IP-radio link was established to the transmitter site, 13 miles away.

The only audio equipment required at the transmitter

Purpose-built wired STL systems typically depend on an incumbent telco provider for T1/E1 service or some other tariffed connection. These have had some assurance of reliability from the telco and, at least in the past, were given priority for clearing any faults. Indeed, before telcos became more interested in providing IP connections, such dedicated circuits as T1s and the like were considered very reliable, with “backhoe fade” being the most prominent risk. These days, anecdotal evidence suggests that T1 and similar services are no longer a priority for incumbent carriers.

Some key benefits of purpose-built, wired STL systems are: • Robust, reliable operation; • Immunity from malicious interference; • A variety of user-end connection options — audio, data, POTS extension, etc.; • Excellent expected equipment lifespan; • Two-way communication with optional equipment.

Some downsides of purpose-built, wired STL systems include: • Dependence upon telco service providers; • Poorly-maintained telco cable plant that portends more faults; • Higher capex costs, potentially much higher with optional add-ons; • Psychoacoustic coding algorithms producing artifacts that are exaggerated by aggressive audio processing; • Audio sampling (bit depth) likely only 16 bits.

Similar to digital RF STL systems, the audio is often brick-wall filtered at 15 kHz, with the attendant audible coloration of such filters. Also, psychoacoustic coding algorithms deteriorate the audio slightly, with exaggerated effect after the FM processing that follows.

IP-BASED SYSTEMS

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site, apart from the IP radio, is an Axia xNode. Four linear stereo channels are received there via the Axia xNode.

For remote monitoring, there are four linear stereo return channels over the same IP connection. All this audio is 20 kHz in bandwidth, using 48 kHz digital sampling and 24 bits per sample. The aggregate data rate across the IP radio link is about 10 Mbps in each direction. A thorough explanation of this system, including IP radio performance considerations, is available in this article.

Other examples of IP-radio STL systems are becoming more and more commonplace, from stations in rural Australia and small Pacific islands to the largest radio markets in the U.S. IP radios may be licensed or unlicensed, depending on the RF band selected for operation. While not an absolute guarantee against interference, licensed IP radio paths tend to be secure and free from such interference.

An RF path and interference study is required as part of the licensing process. Unlicensed IP radio pairs are likely to work well in rural or uncongested areas, but careful consideration and backup planning are suggested for unlicensed operation in built-up urban locales.

Coded IP STL systems offer several advantages in terms of bandwidth requirements and recovery from packet loss. While a linear IP STL, as discussed above, needs a near-perfect IP transport path to work properly, a coded-based IP STL is less stringent. Codecs generally offer more packet buffering, error concealment and reordering of packets arriving out-of-order than a linear system.

While audio codecs themselves will cost more than, say, a simple “node” AoIP endpoint, they offer more flexibility in their IP connection’s requirements. For example, using audio codecs at each end, one may install an IP radio link pretty much “out of the box” and have a working STL.

One may also use public internet for either the main or backup link between codecs.

Another IP-based STL method has appeared in the past few years. It’s an FM MPX transport codec over IP. This is similar in connection to the audio chain as an MPX RF STL; the FM audio processor is placed at the studio, and the full MPX signal is carried to the transmitter site where it’s wired to the FM exciter directly. These MPX over IP systems also appear in both linear and coded varieties.

Linear MPX STLs require anywhere from 3 to 7 Mbps to faithfully sample and transport an FM processor’s MPX output. These systems may not offer any error correction but usually do offer connections for dual-path redundancy.

An interesting application of MPX-over-IP technology is to use one FM audio processor to feed identical audio to several FM transmitter sites. There’s no need for separate FM audio processors at each site.

More recently, an interesting and useful method has arrived for coding the FM MPX signal to transport it at a much lower bitrate.

The trade name is µMPX (or Micro-MPX). This pro-

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•

prietary method does not use psychoacoustic coding.

Rather, it’s a novel application of mathematical data reduction that is well suited for the FM MPX signal. It is “well-suited” because artifacts of the µMPX algorithm tend to fall in time and frequency where they don’t affect the perceived audio. And, unlike psychoacoustic codecs, precise peak control is maintained through a µMPX STL system. The minimum bitrate for µMPX systems is 320 kbps, with a maximum of 576 kbps. The algorithm also offers dual IP path redundancy as well as forward error correction to rebuild occasional lost packets.

The IP transport(s) for µMPX can be Public Internet, IP radios, Wireless ISP, etc. or any combination of those.

Some key benefits of IP STL systems are: • Highest audio quality using 20 kHz (48k/24-bit) linear audio; • Lower-performance IP connections can use coded audio; • It’s likely the lowest cost option, even with redundant IP path cost; • Robust, reliable operation with redundant IP paths; • Additional uses thanks to IP connectivity, e.g. monitoring, backup, disaster recovery, etc.; • Two-way communication inherent in IP connections and AoIP; • One-to-many distribution for single-frequency networks; • µMPX distribution that offers excellent MPX performance at modest bitrates.

Some downsides of IP STL systems include: • IT networking skills are needed for most robust configurations; • Where applicable, psychoacoustic coding algorithms result in artifacts which are exaggerated by aggressive audio processing (does not apply to linear configurations);

Ultimate reliability is based on having at least two IP paths for redundancy.

IP-based STL systems easily offer the most flexibility in terms of connection options. If a private, high-quality IP link, using enterprise-grade IP radios, for example, can be obtained, then two-way linear audio is the best option for perfect audio transport. That same high-quality IP connection can also be used for remote transmitter control, remote backup, disaster recovery options, security video surveillance, remote telephony and so much more.

If only public internet or lesser-grade IP radio paths are available, then coded audio at the highest available bitrate makes sense.

In either case, having more than one IP path is truly important for redundancy. We know that packet loss and occasional outages from internet service providers can be a problem.

The transactional business world, as well as the worlds of social media and entertainment, are now dependent on IP connectivity. For broadcast engineers this widespread dependency implies competitive low cost and wide availability of IP transport. Moreover, this active marketplace of IP equipment and services is delivering flexible system solutions; redundant systems and connections; and generally better audio performance that we’ve heard before.

WRAPPING IT UP

IP connections support security video surveillance.

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Radio broadcasters truly have more options than ever for their studio-totransmitter link technology. Traditional 950 MHz RF links work well enough in most situations. Analog systems do exhibit some noise, while digital 950 MHz links may be constrained by filters and less bit-depth, or by audio coding artifacts. IP-based solutions afford engineers more options, including the possibility of the most pristine linear audio link they’ve ever experienced.

Codecs Open Up “Hard-to-Reach” Sites Whether the pipe is big or small, you have options to get your audio delivered Bryan Waters is chief engineer at Cumulus Media Atlanta.

After 25 years in broadcast radio, including 13 years in engineering, he says, “I learn something new every day.”

With most units giving built-in failover and/or multiple network interfaces, one piece of gear at a site can get you everything you need.

Radio World: How are design and performance of codecs for radio changing?

Bryan Waters: One of the more important trends I am seeing lately with codecs would really be on the STL side.

You see more MPX codecs hitting the market, giving more flexibility to broadcasters who want their gear at the studios where it is accessible, but still want the sound of the composite output at the transmitter.

With most units giving built-in failover and/or multiple network interfaces, one piece of gear at a site can get you everything you need.

With Atlanta’s ever-growing metro, a traditional 950 MHz link is not always possible. We deal with it by running GatesAir IPLink MPXp units as our primary STL system, using a mix of composite and AES running over fiber between five sites. With the web GUI, we are able to monitor everything from studio to transmitter site.

RW: What about broadcasting from home?

Waters: Codecs have kept the industry alive during our days of COVID-19 quarantine. From the standard Comrex and Tieline “remote gear” to online sites like CleanFeed.net, we’ve had to find ways to bring cohosts together virtually, enabling them to still interact with the audience, without it sounding like a train wreck. Though occasionally …

RW: How are these technologies being deployed to address problems in new ways?

Waters: Codecs have opened up the world for those “hard-to-reach sites,” the situations where you’d never get a STL shot but you’ve got internet.

RW: How powerful can codecs get?

Waters: I think the only limitation is our imagination.

Where there is a need, there will always be a person building something bigger, better and stronger.

RW: What functions and features are being offered on new models that engineers who haven’t bought a codec or STL in a while should know about?

Waters: I don’t know if there is a function or feature that would stand out to me, as much as the stability of IP codec units now.

Eight to 10 years ago, an IP codec seemed like a good alternative for remote broadcasts; but the stability, along with a basic lack of network infrastructure, made it a less-than-optimal choice for STL. Today, whether the pipe is big or small, you have available options to get your audio delivered and sounding good.

Bryan Waters

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Codes Offer Redunancy, Backup and Failover Entercom’s Eric Fitch on how his operation has benefited from trends in codecs and STL plex T-1 STL systems. Each site has three ISPs to provide triple redundant paths for the audio streams.

We have a fiber-based MPLS system as the primary connection to each location. That is backed up by a wireless internet connection at the studio and cable modems at transmitter sites. We have a third connection to each ISP via our business network on a second fiber network from the studio.

The IPL-200 is able to have three separate audio streams that can failover if one or two of the streams drops, keeping the station on the air, while notifying us via e-mail or SNMP that there is a fault. The system has an optional redundant power supply, which is great if and when the UPS fails.

The ability to access all 28 nodes of the IPL-200 on our network from work or home makes configuring and troubleshooting a breeze. We can see when an ISP has a failure at any of our sites because there are multiple ways to log into the codec.

Eric J. Fitch is director of technical operations for Entercom New England. He has been a broadcast engineer for 35 years, working in Syracuse, Albany and Boston. Today he is responsible for 14 stations in five markets.

Radio World: What’s the most important trend in codecs?

Eric Fitch: Redundancy, backup and failover.

With more facilities being managed by fewer people, there is a huge need for systems to be able to recover from outages without human intervention.

Remote access via a web GUI, as Eric J. Fitch well as having logging, SNMP and e-mail capabilities are critical for managing equipment at remote locations.

RW: How are these technologies helping you solve practical problems?

Fitch: We are able to connect our studios in Boston, Providence, Springfield and Worcester to their associated transmitter sites and the Westwood One Satellite NOC in Denver, which uplinks the WEEI Sports Radio Network and the Red Sox Radio Network, via GatesAir IPL-200 Audio over IP systems.

Our GatesAir IPL-200 systems have replaced our Intra-

RW: What role are codecs playing in the new normal of at-home broadcasting?

Fitch: We are lucky that high-speed internet is available in most people’s homes. Just 10 years ago we were struggling with DSL and 56 kb dialup modems.

Now with cable modems and fiber service, we sometimes have better internet connectivity at our homes than we have at our studios and transmitter sites. The connections are so good that our listeners don’t notice that the hosts are doing the shows from home.

With cable modems and fiber service, we sometimes have better internet connectivity at our homes than we have at our studios and transmitter sites.

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Increased Quality, Less Bandwidth Shane Toven on how codecs are becoming more powerful and useful rural areas were limited.

Once IP connectivity started becoming more ubiquitous and there were an increasing number of IP codec options on the market, Radio World: What’s the most I took advantage of that to execute important trend you see in the design some very complex remotes, one of and performance of codecs for which involved live events at two remote or STL use for radio broadcast different venues, and full talkback facilities? facilities between the studio and Shane Toven: I see a trend toward the two venues. combining multiple codec chanThe latest application for codecs nels in a single unit. This helps with at my current facility has been conconsolidation of facilities where verting multiple channels of audio multiple content streams and on the AoIP network at the studio locations are involved. I also see to encoded audio for carrying codecs becoming more powerful as Shane Toven. “The biggest advantage of a across lower bandwidth links. This newer and more efficient encoding software-based solution is ease of use and conversion is done entirely inside options are available. reduction in hardware costs.” the codec itself without any actual The most exciting development transition to AES or analog audio. Livewire I/O on one that I see is increasing quality with less bandwidth usage. interface, codec I/O on the other interface. It really makes As broadcasters shift toward consolidating more facilities for a very nicely integrated solution. and interconnecting remote talent, this will be an important consideration for balancing quality versus bandwidth cost.

RW: What role are codecs playing in this new world of at-home broadcasting?

RW: How are today’s technologies solving problems in Toven: Codecs have been critical in this role, though not creative ways, or being deployed in your own facilities? in the traditional hardware sense. Some broadcasters Toven: Codecs have been an invaluable tool for me, have chosen to deploy hardware codecs for this purpose, going all the way back to the original POTS codecs. but many others are using services such as CleanFeed Unfortunately, ISDN was not an option in rural Minnesota or ipDTL. Both have advantages and drawbacks, but the where I started my career. I purchased a Comrex Vector biggest advantage of a software-based solution is ease of at my first station when that technology became availuse and reduction in hardware costs. I could also envision able. It made a significant improvement in the quality of a scenario where the codecs themselves become an interemote broadcasts when the options for connectivity in grated software component of a virtual infrastructure.

Your smartphone becomes your codec and the talent can work from anywhere with very little hardware.

Shane Toven, CSRE CBNT, is senior broadcast engineer at Educational Media Foundation.

The most exciting development that I see is increasing quality with less bandwidth usage.

RW: How have AoIP technology developments been reflected in the look and function of codecs?

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Digital Networks Must Facilitate Operations But the modern digital network can do much more

by Anke Schneider The author is sales, marketing and PR manager for 2wcom Systems GmbH. These days most studios run their AoIP networks to produce audio content. In theory, keeping the studio’s contribution separate from distribution offers flexibility at all sites; and separating the audio portion from transmitting sources such as satellite, DAB+ or IP provides various benefits.

Practically speaking, it’s now necessary to change perspective: IP offers broadcasters significantly more flexibility for the transmission System setup to facilitate operation: Following the audio, left to right, from of content, so more and more broadcasters a studio, through 2wcom AoIP encoders into the 2wcom multimedia over and recording studios are deciding to expand IP network cloud (MoIN). Transcoded by MoIN, it then moves into available IP-based networks. distribution channels: internet, DAB multiplex or satellite - and the broadcast But broadband and fiber optic are growing transmitters. at very different rates internationally. So in addition to the use of IP-based structures, flexible alterMoreover, compatibility of the different frame sizes of native distribution paths must be available.

AAC Profiles and Opus is important. Put each manufacThis leads to the question of how best to link contributurer through its paces to ensure that its products suption and distribution. The answer is to keep it in segments, port all possible variants of an audio algorithm. because an increasing number of stations are multimedia, Third, the codec must support all common protocols streaming audio and video along their facility’s networks. as well as standards for internet interoperability such as Livewire+, Ravenna, AES67, EBU Tech 3326 or SMPTE FIRST SEGMENT: STUDIO SITES ST2110 full-stack.

To connect the studio’s networks with the headend and Fourth, flexible stream management must be possible finally all transmitter sites, we should consider several in means of channel scalability and MPEG multiplexing aspects for our audio setup. The AoIP codec chosen for facilities. This includes perfect network synchronization this should meet at least five key requirements. by supporting PTPv2, NTP (Network Time Protocol) and First, it must be stable even when operating in WANs. 1PPS. Especially for audio description of live events, synThis can be achieved by providing features for transmischronization down to the microsecond is essential. sion robustness like redundant internal or external power Fifth, management and control of each audio over supplies; software redundancy, e.g. forward error correcIP codec of all studios in a network should be available tion or SRT; and dual streaming or parallel streaming with remotely via PC web interface, supporting SNMP, Ember+, different audio bitrates. Also, please consider a backup JSON or NMOS. The main system control should be acceswith an alternative source to ensure that your content is sible hands-on via local hardware control in case a WAN transmitted, say, via satellite in case IP lines fail. becomes inaccessible.

Second, the codec must provide all audio formats norAs a result of the above, all studios in a static or mobile mally used in a studio, like Enhanced aptX, most ACC pronetwork can fall back on a unified codec solution while files or MPEG formats, Opus, Ogg Vorbis, PCM and Dolby. keeping their independence. From a budgetary point of TRENDS IN CODECS & STLs Radio World | July 2020

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view, a system as described above provides the chance for all studios of a network to rely on existing AoIP setups.

SECOND SEGMENT: HEADEND

The demand for the perfect link at the headend implies all of the requirements mentioned above. Moreover, the solution should just collect the forwarded studio programs to make them available by simply transcoding the streams respectively to the different distribution sources — audio over IP, DAB+ or satellite. This could be achieved by a multimedia-over-IP network server software, flexible, integratable into existing structures in hardware, VMs or as a containerized cloud service.

A system setup that fulfills the requirements described above ensures a “non-locked-in” arrangement. The headend/multiplexing system and studio systems are not hosted on the same network but kept separate from each other; thus, it is possible to replace one or the other if needed without bringing the companion operation down.

An aside about virtualization. We are at the beginning of the use of virtualized products. Be aware that virtualization counts on maintenance. This goes along with the wishes I have often heard from our customers that a broadcast network should be expandable as easily as possible, add new services with a mouse click and mirror the configuration of one device to another.

Scalability can be notably improved by using virtualization strategies. The possibilities that have been introduced by Docker or VMware to copy instances, take snapshots or run them across multiple hardware devices are a great improvement to scale and maintain networks.

That also has a major impact on needed rack space.

Thanks to virtualization, applications can share the same hardware or even run as a swarm across multiple hardware units with different hardware configurations.

As a result, the number of devices needed is reduced to a minimum, because server hardware has in most cases a lot more processing power than the specialized hardware of codec manufacturers. Thanks to AES67 and other AoIP standards, the requirements for real hardware

On-demand transcoders support production and post-production.

They allow flexible handling and further distribution of alternative audio streams, like audio description or audio commentaries for social media and the broadcast of a multimedia contribution.

interfaces are slowly disappearing, and that is opening the door for virtualized solutions dependent on an all-IP infrastructure. With high bandwidth and robust IP lines, audio processing in the cloud becomes possible. In consequence, manufacturers have to pick up the pace and offer their solutions as virtualized software.

THIRD SEGMENT — IMAGINATION

With a little imagination such networks have been utilized for a variety of transportation duties. Here’s a beginning list.

Icecast/HLS to DVB Transport Stream Transcoding: This is used by a number of customers who want to make webstreams available on a DVB transport stream that can be sent in cable networks or via satellite.

Streaming Encoder: Software can also be used to feed a streaming encoder, for example, the Wowza streaming cloud; or the solution transcodes the audio signals to adaptive bitrate protocols like HLS that can be distributed to the end customer by using a CDN. AES67 to WAN Bridge: With a great number of supported audio over IP protocols, a “multimedia over IP” network server can transcode signals from studio networks that use AES67, Dante, WheatNet, Ravenna or Livewire+ to a format that is suitable for wide-area networks. For example, the studio signals can be transcoded to Opus for a low-bitrate transmission with SMPTE 2022 conform error protection or using Secure Reliable Transport (SRT).

That enables a studio-to-studio bridge that can overcome even stressful network conditions.

On-Demand Transcoders: The multimedia over IP network server software offers scalable activation of codecs in means of number and time. This allows flexible handling of alternative audio streams such as an audio description of a video, to guarantee accessibility for blind and visually handicapped persons. Or, when a multimedia contribution is produced, operators are enabled to process simultaneous audio commentaries for the video, station website, social media and the radio broadcast.

Audio is directly fed into the 2wcom MoIN digital multimedia network, where it can be routed to a streaming encoder or directly to a content delivery network and then into web streaming.

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Use Case: Four Programs to 14 Locations Integrator puts Barix and Wheatstone gear to work

by Alfonso O. Lopez ANTOFAGASTA 106.3

COPIADO 97.7

PNCGUIPULLI 92.3

OVALLE 96.7

Signal 1

WHEAT-IP AES/EBU

Signal 2

Signal 3

WHEAT-IP AES/EBU

WHEAT-IP AES/EBU

WHEATSTONE SWITCHBLADE

CONNECTION VIA PUBLIC INTERNET NORTH ZONE

VALDIVIA 105.7

WHEAT-IP AES/EBU

DEDICATED 5MB CONNECTION, TRANSMISSION VIA SIP

CONNECTION VIA PUBLIC INTERNET

OPEN PUBLIC INTERNET

SOUTH ZONE

CONNECTION VIA PUBLIC INTERNET

DOWNTOWN AREA

OSORNO 89.1

Signal 4

CONNECTION VIA PUBLIC INTERNET

SOUTH ZONE

Summary: Distribution of four audio sources streaming WheatNet-IP audio to 14 locations using Wheatstone SwitchBlade, decoding signal via SIP protocol / OPUS codec with Barix M400 Opus Decoders.

Hardware required: 4 Channel WheatNet-IP license for audio servers 1 Wheatstone SwitchBlade 14 Barix M400 Minimum internet requirements: Head-end 5 MB Each remote site 512 K

SAN FELIPE 92.7

SAN ANTONIO 89.1

CONSTITUCION 92.9

Chillan 88.7

CASTRO 97.9

PUERTO AYSEN 94.1

COYHAIQUE 89.5

PUNTA ARENAS 90.7

The application uses 14 Barix M400 units and Wheatstone WheatNet-IP and SwitchBlade to serve 14 transmission sites via the internet.

extent needed in these circumstances.

A project that caught our attention to share for this ebook was in Chile. This customer needed to distribute four different programs to 14 locations. We suggested using a new Wheatstone SwitchBlade distribution 16-channel system together with the new Barix Opus Decoder. The combination of these units allowed the customer to have control of each stream individually from the main headquarters.

The diagram above tells the whole story. This configuration is not only cost-effective but also allows a level of independent control on each stream and studio.

The author is president of 305 Broadcast.

For 26 years I have been working as an integrator specializing in Latin America. My career started at a small company called Digital Link, then I became the owner of 305 Broadcast in Miami, where we distribute more than 150 brands in our industry, among them Barix.

In today’s chaotic situation with COVID-19, codecs have become an essential tool to every broadcaster both for remote and distribution unicast/multicast. We as integrators have found that many infrastructures were already prepared to receive and send audio via IP, but not to the

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The Codec Has Become a Swiss Army Tool Gawley says users want remote control, simple connections, cloud access and more mix them in live on-air or FTP files back to the studio.

As an example, we have a large national broadcaster in the United Kingdom that has set up a live mixing studio for both radio and TV programs from an engineers’ lounge room due to COVID-19 lockdown restrictions using two ViAs each in triple mono mode. They are connected to four presenters on ViAs in their own homes — all mixed in the lounge room with program audio sent to a Merlin PLUS multichannel codec in Master Control. A producer is connected over Tieline’s Report-IT app to the Merlin PLUS, where a comms channel is fed

Charlie Gawley is VP Sales APAC/EMEA of Tieline.

RW: Charlie, what’s the biggest trend in this segment of our industry?

Charlie Gawley: Remote control and simple connections are paramount these days. The pandemic has accelerated this demand, but thankfully Tieline was already wellplaced to put remote control of all equipment at the engineer’s fingertips.

From a network control perspective, cloud management of all devices is expected. As an example, Tieline’s Cloud Codec Controller Charlie Gawley lets engineers fully configure and remote control all their codecs remotely from the studio or home. Our Report-IT app can be connected, monitored and allow remote input level adjustment as well.

This has been extremely important during the pandemic, as a broadcast engineer can adjust remote audio levels and other settings as required from their own home.

Simple connections are also facilitated by a traversal server like Tieline’s TieLink, which allows creation of call groups, displays codec “presence” and facilitates NAT traversal.

to the talent off-air.

RW: How about this year’s “big story,” the sudden explosion in remote and at-home broadcasting?

Gawley: Codecs have played a crucial role in facilitating home broadcasting and keeping stations on-air after the pandemic forced networks to send people home.

For the seasoned Tieline user broadcasting remotely for over two decades, broadcasting from home is just another venue. However, for studio-based talent it would be foreign to them.

There have been two dominant use cases. One involves broadcasters at home using full-featured codecs like the Tieline ViA with record and playback capability and the ability to integrate live callers in a home studio. These codecs also delivered redundant streaming over multiple IP interfaces like cellular and wired interfaces and data aggregation technologies.

The second use case involved rapid deployment to multiple people in an affordable and simple way. Our ReportIT Enterprise app for iOS and Android allowed users to

RW: How do you see codec technologies being deployed now in clients’ facilities?

Gawley: Today there are demands to do more with less — essentially looking for that Swiss Army tool in your broadcast kit.

The ViA portable IP mixer/codec has enabled broadcasters to essentially set up a remote operational studio where they can take live calls over SIP, Skype, WhatsApp and mix directly live on-air. Users have been able to do their pre-recorded interviews or commercials and either

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download a software codec and tap “connect” to go live very simply. All the configuration was done remotely by the engineer at the studio or from their home.

RW: How many ways are there of making connections?

Gawley: Sales of ISDN and POTS-capable codecs have definitely tapered off, and everyone has either moved or is moving towards IP audio transport.

From an IP perspective, many codecs support unicast peer-to-peer connections or can multi-unicast to dozens of endpoints. Multicasting is also supported to unlimited endpoints over multicast-capable networks. Codec IP audio streams can be delivered over any IP network and integrate seamlessly with all AES67-compatible broadcast studios.

A national broadcaster in the U.K. set up a live mixing studio from an engineers’ home using two ViAs each in triple mono mode, connected to four presenters in their own homes.

foreshadowed that ISDN one day would cease to exist and wanted to have a similar level of interoperability over IP. They set up a working party, which Tieline was a member of, that gave rise to the EBU 3326 Interoperability standard over IP with the SIP protocol at its core, and Tieline was the first non-European manufacturer to implement the standard alongside AEQ, AETA, Orban and Mayah.

Fast-forward a number of years, the studio has caught up to where AoIP in the studio is rapidly becoming the norm. Tieline was ahead of the curve implementing the WheatNet-IP protocol in its codecs, from there others added Livewire, Ravenna and Dante. Given all these different proprietary AoIP standards, both the AES and EBU got the industry together and now we have AES67.

RW: And how powerful do you think codecs can get?

Gawley: Just as processors get more powerful, so do codecs. Today’s leading codecs increasingly include more features and options than ever before. For example, audio processing like EQ, compression and limiting is performed in some. Multiple connections can be configured with multiple redundant streams and data aggregation.

RW: What best practice tips should buyers be aware of in 2020?

Gawley: From an STL and audio distribution perspective, the codecs of today increasingly integrate high-density streaming features, which deliver scalable, space-saving options. It’s what our customers are demanding.

For example, our new Gateway multichannel DSP-powered codec delivers 16 codecs in a compact 1RU design with flexible analog, AES3 and AES67 I/O. From a remote broadcast perspective, the leading codecs can connect multiple streams for program and separate communications or can stream to multiple endpoints simultaneously.

When bandwidth becomes limited the leading codecs offer network data aggregation in addition to stream diversity.

Record and playback, FTP upload/download, audio processing (EQ, limiting and compression), redundant streaming and data aggregation, are just some of the features buyers should look out for.

RW: And what will codecs look like in the future, if we use them at all?

Gawley: Codecs will be required for as long as IP networks are imperfect, which is the foreseeable future.

It’s true that some networks can carry full bandwidth PCM audio over fiber, but these networks generally make up the backbone of larger networks and their primary studio-to-studio infrastructure. Transmitter sites often don’t have fiber runs due to their location or the expense of installation. Remotes are performed from anywhere and often rely on cellular and other wired services that still require a codec to reliably transport audio.

Lossy networks like the internet are imperfect and still require “smart” IP technology to reliably transport audio due to jitter and packet loss; this is where one should look for bit-stream diversity.

Despite significant advances and increases in available bandwidth, cellular networks can at times suffer from capacity constraints. This is where data bonding/aggregation comes into play and one should look for this to be included in a codec, rather than as a clunky peripheral piece of hardware that introduces an additional point of failure.

RW: How have AoIP technology developments been reflected in the look and function of codecs?

Gawley: Audio over IP has been Tieline’s bread and butter for over 16 year, enabling broadcasters to send audio over the public internet, and is nothing new to us.

Tieline had implemented strategies to mitigate packet loss with Forward Error Correction and Auto Jitter Buffer techniques while other codec manufacturers were stipulating use of the five 9 MPLS networks. In 2007 the EBU

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RW: What will the codec of the future look like, if we use one at all? How powerful can codecs get?

Jackson: As for the future: Fully flexible drop-and-drag DSP boxes, with inputs and outputs on AOIP. You pay for what you consume so you are able to scale up and down depending on your requirements at the time.

If you think about virtual environments and how much resource these environments have, there doesn’t seem to be a limit to power.

Consoles that are capable of doing multiple mix-minus feeds have been invaluable. We have been able to keep our staff healthy and sounding great on the air.

RW: What functions and features are being offered that engineers who haven’t bought a codec in awhile should know about?

Fitch: The ability to use multiple internet connections to back up the codec’s connection. The codec can use error correction and buffering to make sure that lost packets are recovered, which is a great asset. A cable modem can be backed up with a wireless hotspot to ensure a stable connection. The ability to remotely control the codec in the field from the studio and have the codec email if there is a problem.

RW: What best practice tips should codec buyers be aware of in 2020?

Jackson: Plan for the future. If you don’t have AoIP just yet, it probably won’t be long.

Think about the environment. Is it being installed into a data room or carted to every football game? How strong is the 4G/5G signal in your area. Do you require dual bonded SIMs?

RW: How many ways are there of making connections?

Fitch: We use whatever connection that is available to us: FIOS, cable modems, private MPLS networks, public and private WiFi and Plum cases that bond two cellular carriers.

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RW: What will the codec of the future look like, if we use one at all?

Toven: As connectivity continues to improve, we may in fact not require codecs anymore. I can envision a time where we are able to pass multiple channels of uncompressed AoIP between facilities directly. This would further simplify installations by eliminating one more step in the chain and improve audio quality by reducing the number of cascading codecs, a problem that has plagued engineers since the early days of bit-reduced encoding. I think what will become more important rather than codecs in this scenario is precision timing sources synchronized to GPS.

RW: How have AoIP technology developments been reflected in codecs?

Fitch: Since we have 15 of our show hosts broadcasting from home due to COVID-19, ease of use and control is the most important function that I have seen. We have to make sure that the codecs are as simple as possible to set up and operate.

Three of our morning shows each have three co-hosts connecting to the studio before 6 a.m. That is nine simultaneous remotes using Comrex Access codecs. Having one-button pre-programmed connections is a necessity to make sure everyone can connect themselves.

We use Comrex Fleet Commander and Comrex Switchboard to monitor and connect all of the codecs that our hosts are using from home. We can see the quality of the connection and make changes on one app, so we don’t have to login to multiple codecs to check connectivity issues.

Newer consoles are able to provide multiple mix-minus feeds so multiple hosts and phone callers can all be on the air simultaneously. We used to struggle with one remote and one caller, now we do three remotes and two callers without batting an eye. No echo or “I can’t hear you” complaints any longer.

The board can be set up by anyone, since the mix-minus in done by the consoles automatically.

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Toven: AoIP has made implementing multi-channel codecs much simpler. Instead of a rack full of AES or even analog audio wiring, the codec has no traditional audio I/O at all. One such product that we currently use is the Telos iPort. This streamlines the installation and implementation of codecs in our AoIP based facility considerably. The codec has very few physical controls and metering on it. Instead you have a 1RU box that can handle eight or more channels of encoding and decoding with all monitoring and control performed via the network.

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