HEBDEN BRIDGE, U.K.—Calrec Audio will introduce Type RT, a new 6-, 12- and 18-motorized fader console range that condenses everything a broadcast console needs into a compact, self-contained unit, during IBC2026, Sept. 11-14, at the RAI Amsterdam Convention Center.

The launch is part of the wider IP ecosystem Calrec will show. The new console is built around Calrec’s established Argo consoles, which lets broadcasters build productions their own way, sizing their investment to the project, the company said.

Type RT relies on proven Calrec technology and is well-suited such applications as stadiums and sports venues, radio stations, local TV/podcast studios, corporate events and houses of worship. Type RT combines the best of Calrec’s Argo and Type R platforms in a ready-to-use, self-contained console.

Everything a user needs is built in, including DSP, control processing, comprehensive I/O, monitoring and integrated USB audio connectivity. Created as an accessible entry point into the Calrec IP ecosystem, Type RT is well-suited for new users who want the confidence of proven broadcast audio performance in a straightforward, all-in-one unit, Calrec said.

Built for fast deployment, Type RT is a compact, fanless tabletop console that ensures silent operation, delivering professional live broadcast performance in environments where space, time and budget matter, according to Calrec.

Dedicated information and metering displays beneath each fader give clear, instant visual feedback. The user-configurable control and monitoring panel adapts to how each operator likes to work. Up to 138 assignable buttons, 24 rotary encoders with dedicated displays, three ancillary displays and a four-inch TFT screen with user-configurable meters put critical controls and status information where needed.

Calrec's Medium Capacity Card delivers native SMPTE ST 2110 AoIP connectivity with full support for existing Calrec AoIP boxes. Dante is available via an optional card, and MADI via a software license.

Type RT combines optional True Control 2.0 for remote production and ImPulseV virtualized DSP in an ultra-compact form factor. Calrec Assist, the browser-based control interface, is included as standard.

Large and small soft panels from the existing Type R line are fully supported, with one soft panel per six-fader section as standard.

Every model shares the same capability, I/O and licensing options, so users can choose the surface size that suits their needs and budget rather than compromising on performance.

The company will also unveil Apex M6 at IBC. Apex M6 is a compact, cost-effective 1U modular I/O platform for audio interfacing, routing, bridging and network edge connectivity.

Designed to expand and extend connectivity across broadcast audio infrastructures, Apex M6 supports ST 2110-30, AES67, Dante, MADI, SDI, Waves SoundGrid AES3 digital formats and analog workflows.

The platform can be deployed independently of Calrec consoles, making it a good solution for broadcasters transitioning to IP, expanding existing networks or integrating mixed-format environments, the company said.

Apex M6 works with Calrec's full range of consoles while opening opportunities in standalone conversion, virtualized workflows and multivendor environments. Its 512-512 channel internal router allows any input to be sent directly to any output without patching signals back through the network, while NMOS-native operation ensures interoperability across multi-vendor ST 2110 ecosystems.

The company will also preview a series of Argo and ImPulseV software enhancements at the convention that reinforce its commitment to the EBU Dynamic Media Facility project and give broadcasters more freedom in shaping infrastructure around their priorities.

See Calrec at IBC 2026 stand 8.C47

More information is available on the company’s website.

DARMSTADT, Germany—All sound engineers probably agree that delivering high-quality audio mixes for sporting events requires a versatile mixing console, a good speaker setup, and carefully selected microphones with an excellent tonal balance.

For my work at tier-one events, I need a console that provides quick access to all critical features, is easy and fast to configure, and is flexible. For this, I use as many direct access buttons and custom functions as possible. Designed to address these needs, the Lawo mc²56 console makes it easy to check the availability and status of hundreds of audio sources.

Dolby Atmos Sound
This is especially important for Dolby Atmos productions where I need to create and monitor several 5.1.4 presentations, as well as 5.1 and stereo renditions provided by the mc²56’s downmix function—both with and without commentary. Near-instant tweaks are of the essence when the world is watching.

Automation tools such as audio-follows-video, Lawo’s KICK software for tracking-based, close-ball mixing that delivers crisp ball and other on-pitch noises—and increasingly AI, which I use to clean audio signals such as commentary—have become essential. Without them, the high audio-quality standards viewers have come to expect would probably be impossible to achieve.

I prefer to mix remotely, in an optimized acoustic environment, which usually involves audio sources coming from several locations. The Remote Audio Control Room (RACR) in Germany, where I mix frequently, provides such an environment. While this might work with proprietary MADI over Ethernet connections, only an IP-based network allows me to mix signals originating from different parts of the world.

The total absence of bleed from the venue allows me to make informed decisions about the sound. Today’s audio and video workflows more than justify using a networked ST 2110 backbone for the production of top-notch TV events, be they World Cups, four-year athletics jamborees or music performances.

Lawo mc² consoles have become the de facto standard in all productions I work for. Almost all OB trucks in Germany and the Benelux are equipped with Lawo desks, and the host broadcaster for global soccer events also provides them. So, for me, there is definitely a muscle-memory dimension to Lawo consoles: I know them inside out and find their layout highly intuitive. Working with an mc² console almost feels like my fingers can reach inside the desk and directly touch what I need to refine. This gives me speed, reliability and confidence.

I also use a lot of plug-ins, such as Waves, high-quality Seventh Heaven reverbs, AI tools such as Clear or Clarity and many others, depending on the production, as well as new metering tools that we developed at the university.

Quest for Big Audio
I am very lucky to work on so many immersive audio mixes: the whole crowd is cheering, music is playing, different camera angles are shown in rapid succession—and everything is immersing you. This conveys the impression that viewers at home are actually at the event.

We spent the last 20 years looking for “bigger” audio as video moved from SD to HD, then to UHD and HDR. The tricky part was figuring out how to get rid of unwanted noises in the audio for optimal channel separation. We now have the tools to provide an enveloping sound experience, and we can even specify the scale of an immersive mix to match the average screen size viewers have in their homes—or make it as huge as we like.

More information is available on the Lawo website.

EMERYVILLE, Calif.—Systems integrator Advanced Systems Group has hired Mark Vinten as a senior account executive on its audio and postproduction team.

An industry veteran with experience in professional recording, broadcast technology, manufacturing and systems integration, Vinten will be tasked with fortifying relationships with existing ASG clients and developing new opportunities throughout Southern California.

Prior to ASG, Vinten worked for 10 years at Evertz, progressing from service technician to director of West Coast sales and service. Earlier in his career, he worked for eight years as an engineer for rapper and music producer Lil Jon, contributing to projects with R&B singer Usher that earned four Grammy nominations and two Grammy awards, ASG said.

“Mark understands our business because he’s lived it from multiple perspectives,” said Hamid “Gadget” Hopkins, director of audio and post at ASG, to whom Vinten will report. “He doesn’t lead with products. He begins with conversations about workflows and solving problems.”

Vinten said he joined ASG because of his longstanding relationship with its president, Dave Van Hoy, and a desire to offer clients a broader range of solutions.

“I've worked with small and large customers in the media space,” Vinten said. “I think of facilities as more than a new box here and there or compartmentalized systems. I look at the facility workflows as a whole and identify efficiencies and new technology that can maximize value.”

He also cited the company’s in-house expertise as a reason for signing on.

“ASG has subject matter experts covering traditional on-prem solutions, hybrid and cloud-based solutions, and the ability to provide creative and technical staffing to support customers as needed,” he said.

Vinten can be reached at mvinten@asgllc.com.

CLEVELAND—Telos Alliance will highlight its latest product line for TV, streaming, and radio at the 2026 IBC Show, Sept. 11-14 at the RAI Amsterdam.

Visitors to Stand 8.D37 will find two European product debuts as well as major updates and additions across intercom, live production audio, and file-based audio workflow tools.

New this year for Jünger Audio flexAI, is flexAI CastCompanion. Powered by JAIC, the Jünger Audio Intelligent Companion, flexAI autonomously manages live voice sources that may arrive at very different levels, with varying quality and background noise. JAIC analyzes the incoming audio and helps optimize parameters, while Jünger’s deterministic algorithms handle the actual processing, delivering predictable latency and broadcast reliability. CastCompanion combines automixing, AI-assisted de-noising, voice enhancement, and loudness management to create consistent, intelligible mixes without an operator constantly riding levels.

Telos Alliance will show new flexAI support for 12G-SDI and NDI and will demonstrate how flexAI now integrates with the TVU MediaMesh cloud-native live production platform, allowing producers to extend Jünger’s powerful processing ecosystem across hardware, server, IP, partner, and cloud workflows.

The company will also highlight new Infinity MK2 hardware panels for the Telos Infinity IP intercom range. Designed for use in the studio and in the field, the larger buttons feature bright TFT displays, programmable colors, flexible Talk/Listen controls, three shift pages, a Reply to Last Caller function, and four-way “compass” controls.

For sports commentators, Infinity VCP Virtual Commentator Panel now integrates directly with Elgato Stream Deck+, adding compact physical controls to its browser-based commentary environment without requiring a traditional dedicated commentary box. VCP also uses adaptive bitrate control to help maintain contribution audio intelligibility and network reliability.

Also making its debut in Europe is Axia Pulsar, a compact AoIP console, that features an 8-fader Master Module that uses dual operating layers to provide access to 16 inputs; an optional Expansion Module gives physical access to all 16 faders. Axia Pulsar features bright TFT displays, touch-sensitive encoders, spill-proof 100mm motorized faders, programmable controls, Show Profiles, and integrated EBU R128 loudness metering. Powered by the 2RU StudioCore engine, the compact Pulsar integrates directly into all Livewire and AES67 environments. Optional Pulsar Soft browser-based control also makes remote operation possible from a computer, tablet, or smartphone, according to Telos Alliance.

Minnetoka AudioTools Server v7 will debut at IBC, with new A.I./machine-learning modules that include Language Detection, Dialog Intelligibility, and a powerful new tool, AudioTools Source Separation by AudioShake. This new subscription module can extract dialogue, music, and effects from finished program audio to give broadcasters, studios, streaming providers, and content owners new ways to work with existing material when original multitrack sessions are unavailable or incomplete.

Dialogue can be isolated for dubbing or intelligibility enhancement, music can be replaced for creative or licensing reasons, and clean M&E tracks can be created for alternate-language versions and new distribution opportunities. Because source separation runs inside AudioTools Server, it can also be combined with existing loudness, QC, transcoding, language, and delivery processes to create repeatable, automated workflows at scale.

For radio, the Omnia XII will mark its European debut at the show. Its new processing architecture combines wideband AGC, five bands of intelligent multiband AGC and compressors, six bands of multiband limiting, smart dynamic EQ, refined bass management, and a new “Clemenza” clipper, designed to deliver competitive loudness without the distortion and listener fatigue that can come with aggressive processing.

Along with products featured at IBC, Telos Alliance will showcase its work with industry partners to advance interoperability and more consistent audio workflows.

As part of the IBC Accelerator “Crystal Clear” project, Telos Alliance is collaborating with Nugen Audio, RTW, Steinberg, Fraunhofer IDMT, Channel 4, BBC, ITV, and others on standardized dialogue intelligibility measurement across the industry.

Telos Alliance will take part in Grass Valley's AMPP demo, with Minnetonka Audio ARC loudness processing and UPMAX up/down/cross-mixing interconnected via MXL to enable higher-quality, lower-latency audio processing within AMPP workflows.

Telos Alliance will also participate in Qvest’s Dynamic Media Facility demonstration, a portable, multi-vendor production environment handling live feeds, real-time audio and video processing, Dolby Atmos upmixing, signal monitoring, and multiviewing. Telos Alliance processing will operate alongside tools from other vendors, with MXL providing the common connection between applications. The demo shows how MXL can simplify integration and reduce latency across on-prem and cloud workflows.

Telos Alliance will be in Stand 8.D37.

I have written about supercomputers in the pages of TV Tech for more than six years. I even had an ABC director tell me my musings on computer dominance in broadcasting were pure folly and that I was “full of it.” However, my earliest contemplations did not take into account learning models which, to me, now further legitimize the possibilities of AI in broadcasting.

Let’s use my friend at ABC as an example. Bob has less than a dozen cuts in his directing repertoire, with only a handful of permutations between them—very predictable. Now, if you take it to the next level, Bob-Bot can remember these camera sequences and their outcomes, plus how effective their entertainment value was. You could even teach Bob-Bot the patterns of different directors and producers and mix up a pretty good show.

FIFA World Cup soccer and Host Broadcast Services have led the way in advancing sound quality and also in developing computer-assisted sound.

Not only was the concept of mixing matches in the broadcast center’s master control developed at the World Cup, but Lawo also created a mixing algorithm that compares sound intensities from different microphone positions, selects the best ones and creates the most desirable sports sound. I wrote in the pages of TV Tech that I thought the 2022 World Cup was the best-sounding World Cup ever and better yet, the best-sounding sporting event ever!

From Assistant to Controller
The difference between the early AI systems and current (and future) systems is that AI brings the ability for the computer to learn and remember. For example, a machine model can combine audio sources and create a mix from the existing data. A learning model will be able to elevate the entertainment value with more and better permutations each learning cycle.

Computer-assisted is quickly moving to computer-controlled. It is easy to see how the entire production chain can be computer-controlled. Cameras do not need operators anymore, with autofocus and simple commands like “follow the ball” or “follow car No. 6.” I was resistant to “audio-follow-video” for many years because I thought the switch did not sound very good and subjective control was the job of the mixer. Basically, my prejudices were ego-related and not engineering-related.

The Lawo “mixing assist” compares sound levels from different microphones and adjusts mix levels for the most desirable combination of sounds. So, what about computer-generated sound fields and sporting sound effects, like what Ben Shirley and Rob Oldfield of Salsa Sound have been developing? The Salsa program not only can mix, but can change the timbre, texture and composition of sounds. I imagine a ball kick with a little thunder or dynamite explosion added to it. That is entertaining!

Entertainment vs. Reality
When gaming came along, I realized my role as a sound mixer was to entertain the audience and not necessarily document the sound image. Some sounds were boring or not easily captured, which is why, in the early ’80s, I started to use a sound sampler—a primitive analog computer to play back and enhance the sound field—everything from pit sounds and tire squeals to canoe rows and crowds. Engaging sound, camera work and graphics drive broadcast retention, but also cost money, unless perhaps there is a heavy dose of generative computer influence and output.

Every sport can develop an audience via the internet, and there is clearly an abundance of content that is captured and streamed. Not only are supercomputers good at automating tasks and improving efficiency, but they are capable of creating realistic content. How about nude beach volleyball? How will content production keep up? Through AI.

Supercomputers with learning and adapting capabilities are known as “generative,” where computers create new content using similar or existing data. Think about the commentators. You rarely see them, and they tell the viewer what they just saw and then read something about the person, place or thing that was just shown. It is called play-by-play and color, and I think these folks are on the way out.

I have read postings on LinkedIn for audio practitioners to train AI algorithms. That makes sense, since the power of artificial intelligence lies with its ability to ultimately learn and think for itself. It also seems that the best models have a deep reservoir of knowledge to build on.

A Better Model Me
But where is this knowledge coming from? Well, one significant avenue has been through theft of intellectual property.

America has long had a beef with China over intellectual property theft, but it is happening right here in the United States—with impunity. I don’t know who is to blame—the publisher or the AI company.

I heard on the news recently that Anthropic, a leading AI company that I thought had scruples, has settled a class-action suit for copyright theft, which included two of my broadcast publications with Focal Press. If you are going to train a computer to do my job, then learning the fundamentals of signal processing as well as the aesthetics of signal balancing will make a better Model Me.

All sound disciplines are changing and the evolution probably will not stop. No doubt, analytical tools for quality control are soon to arrive, if they are not already here. Things like distortion, phase and loudness are easily quantified and corrected. While working on this article, one of my sources said they have developed a profanity eliminator that works basically in real time and even in different languages.

I used to think the safest jobs in television were held by those who set up and maintained the equipment. I don’t see that as the case anymore when I see stories of Chinese robots playing soccer. Television broadcasts are not hard to set up, and any deficiencies can be compensated or covered up by the computer.

Are you scared yet?

TAMPA, Fla.—As the A1 audio mixer for Tampa Bay Rays, Tampa Bay Lightning and Orlando Magic home broadcasts, I try to make viewers feel like they’re part of the experience. Whether watching from their living room or on a mobile device, the fans should feel like they’re sitting behind home plate, courtside or inside the arena.

The best broadcasts don’t draw attention to microphones, mixers or technology. They create a listening experience that feels authentic. I don’t want viewers to think that I’m trying to make them hear something. They should hear it naturally.

Sports audio is complex, with crowd noise, public address systems, venue acoustics and the pace of the game all competing for attention. My job is to present the sounds that matter most naturally, so viewers experience the game, not the technology behind it.

Natural Sound
That philosophy is what first drew me to the Shure DCA901 microphone array. I was introduced to the tech through colleagues working on major national broadcasts, and my first opportunity to spend significant time with it came during NBA All-Star Game productions. What stood out immediately was how the DCA901 allowed me to focus less on chasing individual sounds and more on creating a complete listening experience.

In basketball, I originally relied on a large number of microphones spread around the court. Capturing the sounds required constant attention as the action moved from one end of the floor to the other. The DCA901’s steerable lobes and automixing capabilities simplified that process while still giving me the detail and coverage I needed. Instead of constantly chasing the action, I could focus on getting the sound where I wanted it.

What impressed me even more was what I heard when using the DCA901 for baseball. For years, I’ve worked with high-end parabolic microphone systems that are considered industry standards in sports production and they do an excellent job. But when I first listened to the DCA901 behind home plate, I noticed something different.

With many traditional systems, you hear the impact of a baseball hitting a bat. With the DCA901, I felt like I could hear the wood itself. A clean hit sounded different from a foul tip. A player finding the sweet spot sounded different than one getting jammed inside. When a bat splintered, the detail of the wood breaking apart came through with remarkable clarity. As an audio engineer, those details matter, because they’re the sounds that make fans feel connected to the game.

Flexible Coverage
Today, I regularly deploy DCA901 systems on MLB and NBA broadcasts. For basketball, I place units on the basket stanchions. For baseball, I position them along the first- and third-base lines to capture key game sounds throughout the field of play. The steering flexibility gives me options that would traditionally require a much larger microphone deployment.

Pairing the DCA901 with Shure Action Isolator software has also been incredibly valuable. In baseball, I route my effects mics through the Action Isolator before bringing them back into the console. My crowd mics remain separate, so crowd energy stays natural, while Action Isolator cleans up the effects mix and makes it easier to feature key game sounds.

For me, the two technologies complement each other. The DCA901 gives me flexible coverage across the field and court, while Action Isolator helps reduce ambient noise that naturally exists in live sports environments. One feature I’ve especially appreciated is the ability to independently adjust processing on individual channels, allowing me to tailor the mix to the sport, venue and production requirements.

What continues to impress me most is how naturally the system captures important moments. Whether it’s the crack of the bat, the sound of the ball hitting the mitt, a pickoff attempt at first base or action developing across a basketball court, the DCA901 helps reveal details that might otherwise get lost in the mix.

More information is available on the Shure website.

BRISBANE, Australia—I’ve spent nearly two decades working in sound design and voice-over, and for almost half that time my focus has been “Bluey.” I lead sound design on the show and voice Uncle Stripe, working between my home studio and my studio at Folklore Sound in West End Brisbane on a series that started in Queensland and has grown into one of the most-watched children’s programs in the world.

My brother Joe created “Bluey” with a clear brief: No exaggerated cartoon effects, just an organic soundscape built from the sounds kids actually recognize from their own backyards. That direction has shaped nearly everything about how I work, and for almost the entire run of the show, the tool I reach for most is the Sennheiser MKH 416 shotgun microphone.

One Career Constant
My path into voice-over happened almost by accident. I was working at a postproduction house as a sound designer, spending my days recording other people, when a director walked in one day and asked me to step up to the microphone and read an advertisement myself. That one moment changed the direction of my career, and the microphone that has stayed with me through nearly all of it has been the 416.

I first used the 416 at postproduction house Cutting Edge, where it was the standard microphone for commercial voice-over recording. In 2016, I bought my own for my home studio, where it remains in daily use. I now own three. In Australia, the 416 has long been the benchmark for commercial voice-over work, and once I began using it on “Bluey,” its directional control and durability made it an obvious choice.

Capturing the real sounds of suburbia means stepping out of the studio and into unpredictable environments. I frequently do what I would call “guerrilla” field recording, using the 416 to capture audio directly from the locations reflected on screen. For slides down playground equipment, I want the real sound, which has occasionally meant climbing into large enclosed slides with the 416 in hand, banging around to capture real physical movement.

For years, I paired the 416 with a Zoom H5 recorder while gathering effects for “Bluey.” Today I use a Zoom F3 and a Sennheiser MKH 8018 in MS stereo, a microphone I’ve been thoroughly enjoying for location work.

Over eight years of field recording, my 416s have endured Queensland’s challenging conditions without missing a beat. The tight directional pattern and strong off-axis rejection let me isolate precise foley and ambient sounds outdoors, while the RF condenser design shrugs off the humidity and temperature swings that come with recording in Queensland.

The 416 has also defined my career as a voice-over artist. In Australia, it’s the microphone performers reach for, because you can work right up into its barrel and presence region to get the classic, relatable read advertisers want.

When we began production on “Bluey,” I brought in a Neumann TLM 103 to record several of our child actors. I have since added a Neumann U 87 for studio Foley and an MKH 8018 MS stereo microphone for location work.

When working from Brisbane with production and post teams on the other side of the world, having a kit built around trusted names carries real weight. When international studios connect with me over Source Connect, seeing an MKH 416 and TLM 103 on my equipment list immediately signals broadcast-quality standards before recording even begins.

Sounds of Australia
I love that I get to show the rest of the world how beautiful Australia sounds. Our native birds and suburban noise might feel mysterious to international viewers, but for us, they are simply the sounds of home. Having a microphone kit I can rely on, whether I am out in the field or mixing in the studio, means every detail translates cleanly to millions of homes around the globe

More information is available at on Sennheiser’s website.

SAVANNAH, Ga.— What started in 2015 as a college summer baseball team selling just a handful of tickets in its opening months has evolved into the phenomenon known as the “Savannah Bananas,” selling out every single game since we adopted the moniker in 2016.

This year marked the Savannah Bananas’ biggest broadcast season yet. In June, we brought “Banana Ball” to ABC for the first time, taking on the Party Animals at Autzen Stadium in Eugene, Ore., and our new Banana Ball Championship League is bringing more fans into more packed stadiums than ever.

Our exclusive distribution deal with Disney+ and ESPN—the most ambitious in Banana Ball history—allows for every game to be streamed on the ESPN App and Disney+, with select games airing across ESPN networks and ABC. True to our “Fans First” ethos, every broadcast remains free to watch on YouTube regardless of where else it airs.

REMI Banana Ball
With our audience growing at pace, BTV (Banana Ball TV, our production arm), has needed to rapidly build up its technical capabilities. We’ve seen a lot of growth for the 2026 World Tour season, so we’ve rebuilt our facilities over the past two years to increase our efficiency and output. Prior to this latest expansion, we were operating with consoles not designed for broadcast production, which caused a number of challenges.

Working closely with Sean McCluskey, our head of engineering, we designed and implemented a REMI workflow that gives our production staff consistency without compromise as they move from room to room and site to site. Spread across two main control rooms at our remote facility in Savannah, and with flypacks positioned out in the field, we’ve placed Calrec’s audio network at the core of our REMI operation, building in the capacity our rapidly expanding operation demands.

We adopted an agile, distributed system, installing three 36-fader Argo M consoles, three Type R systems, and True Control 2.0. This gives us a combination of analog, embedded SDI, MADI and Dante. All of our audio gets a first-pass mix and level set before it’s embedded in our video transmission back to Savannah. We make sure lip sync is correct on-site, so when everything reaches our control rooms, we can focus on mixing the broadcast as if we were at our home stadium.

The Calrec Assist GUI gives us full remote console control, keeping our operation agile whatever the demands. Using a browser means we can access any console from inside the stadium to make changes, and I can also work remotely from home to support our shows and fix issues quickly. It’s a genuine game-changer.

Range of Venues
That agility matters because no two venues where we play are the same. We love massive stadiums, but we also love the smallest ones equally. We recently played at Kyle Field in College Station, Texas, one of the largest venues in the state with a seating capacity of more than 100,000; two days prior, we played at a high school with free admission and a maximum capacity of 1,000. We are here to deliver a consistent entertainment experience.

That range of venues also demands an adaptable approach, which is why we work with local audio operators rather than carrying a full crew with us. Many of these engineers come from an entertainment and film background rather than broadcast, but the Calrec consoles have made it extremely easy for them to make that transition and excel.

Banana Ball is as much about performance as it is sport. It’s designed for casual fans who love great entertainment as much as for die-hard sports fans who want a legitimate season to follow, and creating that fun, positive environment is what resonates with people.

With our Championship League in its inaugural season and our television and streaming output expanding, Calrec’s Argo consoles have dramatically improved our production, putting us in the best position to broadcast the next chapter of one of sport’s most extraordinary success stories.

More information is available on Calrec Audio’s website.

DALLAS—Being selected to host nine 2026 FIFA World Cup matches, including a semifinal, was a great honor for AT&T Stadium and a tribute to the work of our entire team. From an audio perspective, the project brought our in-stadium production together with an international broadcast operation requiring several new signal combinations and delivery paths.

Our main audio control room is built around a 48-fader Solid State Logic System T S500m surface. It is the central point for audio distributed throughout the venue and for the feeds we exchange with broadcasters.

Game-Day Planning
We began meeting more than a year before the tournament and as plans developed, we learned that a central hub would connect the stadium’s audio, video and communications systems to the International Broadcast Center in downtown Dallas and the international broadcast truck at the stadium.

The show content was not radically different from a Cowboys game, where we regularly manage dozens of inputs routed to around 30 auxiliaries and patchable outputs, together with several stage submixes.

For the World Cup, the principal difference was how that content needed to be packaged and distributed. We had to send our show audio to both the IBC and the international broadcast truck, while continuing to supply feeds to the stadium PA and back-of-house areas. Our production partners at Media West provided audio support for the entertainment stage and sent their mixes back to me.

Broadcast requirements called for several distinct mixes and delivery paths. Music, video playback and entertainment content were supplied in stereo, while commentator and host microphones required a separate mono mix-minus. Later in the planning process, we received an additional request for crowd microphones and introduced a separate submix position in an adjacent control room, where the operator managed a combination of sources that changed during each match.

The main change to our System T game-day showfile was adapting the existing routing for the new broadcast workflows. We expanded the routing and monitoring setup to support the additional feed requirements, while keeping our control-room operation intact.

System T allowed us to make these additions without affecting our established stadium workflow. Its routing and I/O options gave me the flexibility to add sources, build new submixes and position the additional playback channels where I needed them, all on top of our typical configuration.

Touch and Done
During each match, the S500m remained at the center of the production, with a remote System T TCA and fader tile allowing an operator in the bowl to adjust EQ and levels as crowd conditions changed. Our bowl mix was embedded into the video feed returning to the IBC. The music mix and the mix-minus for commentator and host microphones were sent separately, while our crowd microphones were routed through another video stream with embedded audio.

I received the international broadcast audio feed and routed it to our back-of-house submixer. My mix and a return submix from the IBC, which needed adjustments during the match, were then sent to another control room. From there, a further submix was created and returned to both the IBC and the international broadcast truck.

The stadium’s Dante infrastructure continued to carry signals between our Shure wireless microphone receivers and System T, while connections to the World Cup broadcast operation were made through the IBC hub in our broadcast rack room.

Redundant System T engines provided confidence for both our team and FIFA’s technical team. At source level, I used the console’s A/B inputs for the primary and backup music playback, making either source immediately available while preserving my normal layout.

Delivering nine matches required preparation, coordination and an ability to adapt our established operation as broadcast requirements evolved. System T gave us that flexibility. When a technical manager requested another source in a submix during one match, it was simply a case of “touch, touch, touch…done.”

More information is available on Solid State Logic’s website.

HEBDEN BRIDGE, U.K.—As the broadcast industry moves toward more flexible production platforms, Calrec will present various improvements to its audio ecosystem at IBC 2026 (Stand 8.C47) that include upgrades and solutions developed as part of the company’s ongoing commitment to the EBU's Dynamic Media Facility (DMF) project.

More specifically, its IBC demos will include previews of enhancements to its ImPulseV DSP software, developed to support DMF. Calrec is also committed to adding support for the Media eXchange Layer (MXL) to ImPulseV.

“The ImPulseV DSP software enhancements previewed at the show align with the EBU’s Dynamic Media Facility (DMF) project,” explained Henry Goodman, director of product management at Calrec. “The idea of a shared server stack that can host all the production software a facility needs, with users spinning up whichever tools a given job requires, is a powerful driver for broadcasters focused on reducing costs and maximizing their investment to deliver a broader range of content.

“The DMF project and the associated Media eXchange Layer (MXL) development aim to solve many of the real-time challenges of moving to software-defined workflows,” he added. “It vastly reduces compute by enabling containerized software functions from multiple vendors to use shared memory to transfer video, audio, and metadata on the same server. It gives broadcasters the ability to take advantage of best-of-breed technology within the same flexible software stack, while retaining the tried-and-trusted feature set they've come to expect from Calrec, without committing to a one-off investment in processing hardware up front.”

In addition, Calrec will explain how its Argo ecosystem combining physical and virtualised user interfaces as part of its software-defined ImPulseV engine, giving broadcasters freedom of choice in how they build, scale and evolve their infrastructure.

Handling up to 256 processing paths and supporting all immersive audio formats, ImPulseV can be deployed on-prem on a standard COTS server, on public cloud infrastructure, or across a hybrid. It can be accessed from any Calrec Argo surface, or via the browser-based Calrec Assist interface, giving broadcasters the freedom to scale depending on the size of the project.

New, smaller DSP pack sizes of 96 and 140 processing paths join the existing 256-path configuration, tailored for smaller systems that still demand uncompromising, broadcast-grade audio performance. Calrec has also introduced short-term, flexible licensing options, including 1-year, 12-week and 4-week subscriptions. This lowers the barrier to entry for ImPulseV, giving broadcasters production capacity exactly when they need it, without long-term commitments or high upfront costs, making it ideal for temporary, variable or one-off productions.

Introducing ImPulseV v1.5

As part of Calrec’s ongoing work with Grass Valley’s AMPP ecosystem, ImPulseV will support the AMPP control SDK, giving applications running on AMPP direct control over console parameters, including fader control. This is supported by ImPulseV 1.5 update that will be demonstrated at the show.

Calrec's new AMPP Bridge control application installed as a workload on an AMPP node, will provide bidirectional control between an ImPulse core and the AMPP environment. Calrec-Bridge carries control data with AMPP handling audio through Dante, NDI and ST-2110 interfaces.

ImPulseV will also be available on the NEP Platform, a service offering orchestrated, rapid deployment of leading production applications like ImPulseV on server-based hardware, unifying multi-vendor applications into a single interface for end-to-end production workflows.

Calrec noted that support for COTS hardware reduces reliance on niche equipment, with Calrec currently benchmarking ImPulseV's performance on standard server hardware. ImPulseV will also include NDI connectivity as an alternative to Dante for routing audio into and out, supporting up to 256 mono-equivalent channels across a maximum of 16 streams, with up to 64 channels per stream.

ImPulseV will also add support for automated deployment using Cloud-Init and Terraform, tools that let broadcasters set up and configure new systems automatically rather than manually. Cloud-Init modules can configure surface layout, system and core names, manage licenses, and DSP engine addressing. New 'Online' licenses handled through an online licensing server, allow licenses to be added via key rather than file, and work with automated deployment. Terraform support is provided via example scripts from Calrec, which can be edited for individual deployments.

“The ImPulseV DSP software enhancements previewed at the show align with the EBU’s Dynamic Media Facility (DMF) project,” explained Henry Goodman, director of product management at Calrec. “The idea of a shared server stack that can host all the production software a facility needs, with users spinning up whichever tools a given job requires, is a powerful driver for broadcasters focused on reducing costs and maximizing their investment to deliver a broader range of content.

“The DMF project and the associated Media eXchange Layer (MXL) development aim to solve many of the real-time challenges of moving to software-defined workflows,” he added. “It vastly reduces compute by enabling containerized software functions from multiple vendors to use shared memory to transfer video, audio, and metadata on the same server. It gives broadcasters the ability to take advantage of best-of-breed technology within the same flexible software stack, while retaining the tried-and-trusted feature set they've come to expect from Calrec, without committing to a one-off investment in processing hardware up front.”

Calrec will also give visitors to IBC 2026 a first look at its Argo v1.5 software release, part of its ongoing commitment to continually developing and enhancing the Argo ecosystem.

Key enhancements to Argo v1.5

The new update will allow Offline Assist users to access the Argo Assist UI to configure shows, memories and presets without an ImPulse core or Argo surface, including importing a 'Desk Package' to configure against a specific physical surface. Also included are new True Peak Limiters, keeping audio in line with broadcast loudness standards, with 64 channels available in Assist, assigned per leg.

I/O patching has been streamlined: Desk Outputs and Desk Inputs have been renamed, and fader inputs now sit in their own discrete category, delivering faster switching when patching.

A new Bulk Advanced Configuration option lets users save advanced device settings across multiple devices in one action, specifying which options to save, and which devices to save them to. Calrec Connect will also display richer stream information for third-party transmitters, including channel count, packet time, sample rate, codec, media clock offset and payload type.

"Ultimately, these updates are about giving broadcasters greater freedom by delivering an infrastructure that fits today's production requirements and adapts to tomorrow's," says Goodman.

That freedom extends across the whole Calrec IP-native ecosystem, with the award-winning Argo console range at its center. It's a platform that supports a broad range of production workflows, giving broadcasters the flexibility to deploy resources where they're needed most while driving greater operational efficiency and cost savings.

Nugen Audio is launching two new flexible subscription offerings designed to give audio professionals short term access to the brand’s most essential plug-ins at the 2026 IBC Show, Sept. 11-15 at the RAI Amsterdam.

Built for project based workflows, “Post” and “Producer” plans provide a cost-effective way for users to access the exact tools they need, only for as long as they need them, Nugen said, which will provide guided demonstrations at the show, giving attendees a closer look at how each subscription supports real-world workflows.

The new Nugen Audio subscription model allows mixers, editors and producers to scale their toolset in real time. Whether tackling immersive upmixing for a streaming series or mastering a single track for release, subscribers can activate the appropriate plan for a specific project, pause it when the work is complete and restart it whenever new demands arise, according to the company.

The Post Plan delivers Nugen’s full suite of audio post-production tools, including VisLM, Halo Upmix, Halo Downmix, Halo Vision, Paragon, LM-Correct with DynApt, ISL with DSP extension, DialogCheck and more. It’s designed for dialogue editing, loudness compliance and multiplatform delivery for surround and immersive workflows.

The Producer Plan offers Nugen’s essential music production plug-ins, including MasterCheck, Paragon ST, ISLst, SEQ-ST, Stereoizer, Monofilter, Visualizer and others and is ideal for mixing, mastering and pre-production.

A key advantage of these new subscription options, Nugen says, is the ability for audio professionals to align plug-in costs directly with client billing. Instead of absorbing the expense of a full bundle purchase, users can activate a subscription for the duration of a specific project and invoice it as part of the production budget. This model supports freelancers, independent studios and project-based teams who need access to high-end tools without committing to long-term licenses, according to Freddy Vinehill-Cliffe, Product Specialist for Nugen Audio.

“Subscriptions can make it easier to align software costs with client work, helping you access the tools you need when you need them, without a large upfront investment,” he said. “Now, your plug-in subscription can be part of your per-project costs, paused during projects that don’t require the tools and then restarted when one of the solutions needs to come to the rescue.

Both plans are now available at https://nugenaudio.com/subscriptions/

Nugen Audio will be in Pod P27 at the show.

HEBBEN BRIDGE, U.K.—Calrec has integrated its ImPulseV virtualized DSP software with NEP Platform, bringing its scalable audio processing capabilities to NEP’s growing software ecosystem.

NEP Platform is an orchestration layer that brings software applications together in a unified software-defined environment, enabling broadcasters, rightsholders, leagues and live content producers to deploy, orchestrate and scale resources up or down rapidly as production requirements evolve.

"Pairing the power of NEP’s software orchestration platform with Calrec’s ImPulseV allows NEP to offer customers virtualized audio production environments and services that best fit the needs of each production. The ability to add temporary virtualized processing resources means broadcasters can scale to support their biggest events without significant up-front hardware investment,” said Calrec director of product management Henry Goodman.

"It also simplifies technical complexity. Wherever the processing is located and however a workflow is designed, it should feel seamless to the operator. Working closely with NEP, Calrec’s ongoing ImPulseV development simplifies deployment and gives customers the freedom to build more flexible, scalable production workflows.”

Seamless blending of on-premises infrastructure with remote, virtual or hybrid operations across multiple processing environments has become an increasingly established way for media companies to generate more content. Calrec's ImPulseV DSP software gives companies the freedom to choose how they work with processing resources that flex to meet any production demand.

“NEP Platform brings those trusted applications together, making it easier to deploy, connect and manage them through a unified operational framework. Adding Calrec’s ImPulseV to the NEP Platform ecosystem further expands those choices, giving customers access to scalable, virtualized audio processing while streamlining how modern production workflows are deployed and managed. It's another example of how we're growing an open ecosystem that gives customers the flexibility to build the workflows that best meet their operational needs.”

More information is available on the company’s website.

LOS ANGELES—Cineverse has launched VAUDIO, a new ad tech offering that takes existing podcast, radio or streaming audio commercials and adds the visuals required for television.

This allows advertisers to launch campaigns across connected TV (CTV) inventory without requiring a traditional video shoot.

Advertising partners at launch include A24, Aura Entertainment, NEON, Signature Entertainment, and Well Go USA.

"VAUDIO brings immediate opportunities to brands looking for an easy way to get more out of their campaigns using CTV, and for our supply-side partners, including media owners, platforms and OEMs, who can now unlock new ad formats and creative that support a better viewer experience, increase ad inventory fill, and drive new revenue," said Cineverse president and chief strategy officer Erick Opeka. "The economics matter here too. At steady state, we believe VAUDIO can contribute up to $12 million in annual revenue at a 15 to 20 percent contribution margin, and we are targeting that run rate by the end of our fiscal year. Because VAUDIO is built on IndiCue's existing ad infrastructure and is developed and marketed by the same team, it required minimal incremental investment, and that contribution flows through to the bottom line."

Spearheaded by the executives that joined Cineverse with the acquisition of IndiCue, and built by the company's industry-leading Matchpoint team, Cineverse said that VAUDIO was created to help advertisers reach new audiences in a high-impact, lean-back environment without requiring traditional video production.

VAUDIO transforms existing podcast ads, streaming audio creative, or host-read campaigns into visually-compelling CTV placements that drive awareness and recall.

Three example VAUDIO ads can be viewed at www.indicue.com/vaudio.

"CTV inventory is too often underused or wasted, leading to poor viewer experiences, money left on the table for platforms, and missed opportunities for brands looking to engage with consumers that are increasingly tuning in to ad-supported streaming programming," added Cineverse executive vice president of revenue Nicholas Frazee. "With VAUDIO, advertisers can easily reach addressable audiences by leveraging existing assets, extending the ROI of podcast campaigns, and doing so with content that engages audiences rather than putting them to sleep with repetitive jingles and countdowns. Cineverse continues to integrate IndiCue's monetization and engagement offerings into their broader ad solutions ecosystem, and brands and media companies alike stand to quickly benefit."

NEW YORK—The Audio Engineering Society has elected David v.R. Bowles to serve as the group’s new president-elect in 2027.

He’ll join the AES board of directors as president-elect on Jan. 1, then succeed Cesar Lamschtein as the group’s president in 2028. He’ll then serve a year on the board as past president in 2029, followed by a subsequent two-year term as an agency governor.

The society’s newly elected regional vice presidents and governors will serve for two-year terms beginning on Jan. 1, 2027. The new AES vice presidents are: Malik Williams, VP, Eastern Region, U.S./Canada; Jess Berg, VP, Western Region, U.S./Canada; and Merlijn van Veen, VP, Northern Region, Europe.

Newly elected as at-large governors are Helen Caddes, Ivo Marković and Angela Piva. Valerie Tyler was reelected as Secretary.

“As a volunteer-driven professional membership society, the AES board elections are a critical part of our operations,” outgoing AES President Brecht De Man said. “I want to congratulate the elected leaders for 2027 and look forward to working with each of them. I am also very grateful to each and every potential candidate, whether nominated to be on the ballot or not, and hope the Society is always supported by this level of volunteer energy and commitment.”

The members of the 2027 AES board are:

• Cesar Lamschtein, president
• Brecht De Man, past president
• David v.R. Bowles, president-elect
• Valerie Tyler, secretary
• Jayant Datta, treasurer
• Agnieszka Roginska, director

One additional director will be elected by the AES board of governors within the next 60 days.

In addition to the board members, the following AES members will serve on its 2027 board of governors:

• Masataka Nakahara, VP, Asia-Pacific Region
• Merlijn van Veen, VP, Northern Region, Europe
• Piotr Majdak, VP, Central Region, Europe
• Huseyin Hacihabiboglu, VP, Southern Europe/Middle East/Africa Region
• Ezequiel Morfi, VP, Latin American Region
• Malik Williams, VP, Eastern Region, USA/Canada
• Kelvin Grimble, VP, Central Region, USA/Canada
• Jess Berg, VP, Western Region, USA/Canada
• Jamie Baker, governor
• Fariuska Lira Barully, governor
• Leslie Gaston-Bird: governor
• Helen Caddes, governor
• Gary Gottlieb, governor
• Ivo Marković, governor
• Angela Piva, governor
• Bernarda Ubidia, governor

ATLANTA—At the 2026 IBC Show, Sept. 11-14 in Amsterdam, swXtch.io will launch groundSwXtch for Audio over IP (AoIP), a software application that lets broadcasters and media organizations move audio in all major AoIP formats, including Ravenna, AES67 and ST 2110-30, between facilities and into the cloud over standard IP networks, including the public internet.

The application achieves this without codec hardware and without reconfiguring switches or routers to support multicast. By carrying audio in its native, uncompressed form end to end, groundSwXtch preserves bit-identical quality while removing the encode/decode latency, hardware cost and deployment complexity that come with traditional AoIP transport, according to the company.

Unlike current methods, which mostly rely on dedicated codec pairs to move audio between facilities by compressing, transporting over a standard connection and decoding on arrival, groundSwXtch removes that step by carrying the existing multicast AoIP streams on a facility's network across the connecting network (WAN, 5G or public internet) through a centrally managed software overlay.

A lightweight application on standard hardware/servers or virtual machines forms the overlay and moves native AoIP traffic across existing IP infrastructure, including over switches and routers that were never configured for multicast routing. Broadcasters connect studios, production centers, remote facilities and data centers into one unified fabric without touching physical network hardware, with built-in packet protection for hybrid links including SMPTE ST 2022-7 hitless failover, and a single-pane view of every flow, according to swXtch.io.

Because groundSwXtch shares its core architecture with swXtch.io's cloudSwXtch platform, the two can be paired to extend the same multicast fabric into the cloud—giving broadcasters one hybrid network spanning on-premises and cloud environments rather than separate ground and cloud workflows.

Timing accuracy across that fabric is maintained through integrated clock synchronization including support for PTP, preserving the accuracy required for live production, intercom systems and distributed audio mixing. Because groundSwXtch transports AoIP streams natively in their original format, with no compression or transcoding at any point, audio arrives exactly as it left the source.

swXtch.io has validated groundSwXtch with broadcast and professional audio partners across live sports and remote event production, intercom connectivity between facilities, distributed audio mixing, remote production for entertainment venues, and radio workflows linking studios, newsrooms and regional bureaus into a single software-defined audio network.

"For decades, the industry has focused on making audio codecs faster and more efficient," said Brent Yates, CEO of swXtch.io. "We took a different approach by asking whether broadcasters need codecs at all. If audio already exists as native AoIP and we provide an IP network that can transport it reliably, removing that encode-and-decode layer simplifies infrastructure, lowers costs and preserves the integrity of the original signal. groundSwXtch allows broadcasters to move audio exactly as it was created while making distributed production far easier to deploy and scale."

swXtch.io will demonstrate groundSwXtch for Audio over IP during IBC2026 in meeting room 5.MR13 (Hall 5) in the RAI Amsterdam.

OXFORD, U.K.—Solid State Logic (SSL) will feature the evolution of its comprehensive System T platform that embraces a range of software-defined workflows during IBC 2026, Sept. 11-14, at the RAI Amsterdam Convention Center.

The company will demonstrate how its System T ecosystem is delivering scale with a range of physical and virtualized user interfaces as part of its software-defined Virtual Tempest Engine (VTE) for deployment across both on-prem and cloud workflows.

Supporting up to 256 processing paths and all immersive audio formats, System T’s VTE can be located on-prem on a standard COTS server, on public cloud infrastructures or use a hybrid of both. Delivering scalability, it can be accessed from any System T surface, whether that is a virtualized controller on a PC, a compact desktop tile or a top-of-the-range S500 console.

​The ability to deploy broadcast audio processing wherever it is needed and provide remote control of the processing from anywhere positions the System T VTE for the industry’s transition to more agile, software-defined workflows, said SSL broadcast product manager Berny Carpenter.

“At IBC, SSL is previewing enhancements to VTE’s software-based processing engine as part of our commitment to align with the EBU’s Dynamic Media Facility (DMF) project,” he said. “The idea of accessing a shared server stack that can host all the production software a facility needs, with users spinning up whichever tools a given job requires, is a powerful driver for broadcasters focused on reducing costs and maximizing their investment to deliver a broader range of content.

“The DMF project and the associated Media eXchange Layer (MXL) development aims to solve many of the real-time challenges of moving to software-defined workflows. It vastly reduces compute by enabling containerized software functions from multiple vendors to utilize shared memory to transfer video, audio, and metadata on the same server.”

Highlighting new and improved automated deployment functionality to streamline software-defined production both in the cloud or on-prem, SSL is inviting visitors to discover how its Virtual Tempest Engine complements MXL.

​Part of the same DMF-aligned push, but designed for more versatile use cases, SSL’s compact µVTE also will make its IBC debut. Launched at NAB and delivering the same expansive broadcast features as the wider System T ecosystem, it delivers SSL’s lowest entry point to virtualized production. It provides 96 virtualized processing paths and has a range of licensing options to support temporary, variable or event-based productions.

​SSL’s Net I/O ST 2110 Bridge, a high-capacity standalone ST 2110 and Dante converter that provides up to 512 channels of conversion in a compact 1RU chassis, will also make its IBC debut. With NMOS support and the ability to integrate with any Dante or ST 2110 connected audio console or network, the Net I/O ST 2110 Bridge builds on System T’s native IP architecture to allow users to deploy whatever they need, wherever they need it.

​In addition, SSL will also showcase the wider System T range at IBC, including the lightweight S500m console with redundant TE2 Tempest Engines and native ST 2110 and Dante connectivity. Designed for outside broadcast and heavy-duty fly-pack systems, the S500m is a compact version of SSL’s flagship S500 console.

Visitors to the SSL stand will also be able to get hands-on with a 32-fader S400 control surface as well as the ultra-portable TCA Tour. Designed for use on the road, the TCA Tour is a self-contained 16 fader control frame with integrated onboard I/O and a built in network switch.

See SSL at IBC 2026 stand 8.B45.

More information is available on the company’s website.

Banana Ball TV (BTV), the production arm for the Savannah Bananas baseball team, has installed three Calrec 36-fader Argo M consoles, three Type R systems, and True Control 2.0, to expand remote production capabilities.

The Savannah Banana exhibition baseball team tours packed baseball stadiums worldwide, using altered baseball rules that favor trick plays, all within a two-hour limit to increase audience engagement. Think Harlem Globetrotters, but for baseball.

This year, the team introduced the “Banana Ball Championship League,” which adds even more opponents and an even busier broadcast schedule covering games across 75 stadiums and 45 states.

Spread across two main control rooms at BTV's remote facility in Savannah, and with flypacks positioned out in the field, Calrec’s flexible audio network sits at the core of a significant expansion of BTV's REMI capabilities. Designed to extend capacity and accommodate future growth, it makes BTV's workflows much more fluid, according to Savannah Bananas’ Head of Technology Matt Webster.

“We’ve seen lots of expansion for the 2026 World Tour season, and part of re-building our facilities from 2025 to 2026 was to expand capabilities and increase overall efficiency,” Webster said. “The Calrec consoles are all being used to support REMI broadcasts; we have Argo consoles in our main control rooms and we deploy the Type R frame in the field in our mobile units and flypacks.

“This gives us a combination of analog, embedded SDI, MADI and Dante, and all the audio is given a first pass mix and level set before it is embedded in our video transmission system back to Savannah. We ensure lip sync is correct on-site, so when it comes back into the control rooms, we just focus on mixing the broadcast as if we are doing it at our home stadium.

“We also have remote control of the consoles via the Calrec Assist GUI for changing things on the fly. The most challenging part of any audio console is how you have to interact with it if you are not standing directly in front of it, but nothing beats having access to a web page; it means we can access any console from inside the stadium to make changes, and I can also work remotely to support all our shows from home and fix issues that arise quickly. It is a game-changer for sure.”

Access to Calrec’s True Control 2.0 protocol, which provides even greater levels of remote control and unparalleled flexibility to scale up when required, gives scope for BTV to further develop its workflows. Webster says he plans to introduce it more as the season progresses.

It demands an audio workflow as dynamic as the Savannah Bananas’ antics on the field, Webster added.

“We want the technology to outpace production,” he said. “It’s all about scale and simplicity, and Calrec allowed us to achieve this in our second generation of REMI workflows. With broadcast consoles like the Argo and Type R, our workflow has dramatically improved. The Calrec equipment allows us to truly make sure we have the best audio – sweetened and observed.

“It brings us a level of consistency that we have been looking for; it just works, and it is simple enough for any audio operator to just jump in and start driving. We have many local audio mixers in the Savannah area that have an entertainment and film focus rather than a broadcast one, but the Calrec consoles have been extremely easy for them not only to make the transition, but to excel.

“It means all our production staff are able to move from room-to-room and from site-to-site and not have any compromises or workflow limitations.”

NEW BERN, N.C.—Wheatstone will demonstrate its VMX virtual mixing platform at IBC 2026, Sept. 11-14, in Amsterdam.

The VMX platform is server-based and web-browser accessible for mixing in any environment on a variety of devices, from recessed touch screens in the studio to tablets at the ballpark.

VMX can be installed on an existing server or a dedicated appliance and offers a selection of tiles including basic utility mixers, a simple virtual mixing interface and complete virtual broadcast consoles for the television newsroom or the radio on-air studio, Wheatstone said.

Virtual consoles are accessible through a web browser, allowing users to quickly add newsrooms, control rooms, voicetrack booths and remote mixing instances on demand—complete with backup mirroring synchronized in real time for instant failover.

VMX is AES67-compliant with an easy-to-install setup wizard, the company said. It is scalable from large enterprise systems to the VMX Nano, priced and scaled for remote or emergency broadcast applications, according to Wheatstone.

VMX is part of Wheatstone’s Layers software-based broadcast suite for virtualizing studio functions as an extension of the WheatNet IP audio network.

Wheatstone will demonstrate the VMX total virtual mixing experience at IBC 2026 stand 8.A51, along with its WheatNet IP audio network of virtual and fixed consoles, talent stations and intelligent I/O access units.

More information is available on the company’s website.

STOW, Ohio—Audio-Technica has introduced the 20 Series Control Application, a Stream Deck plug-in designed to provide streamlined control of compatible Audio-Technica USB microphones directly from Elgato Stream Deck devices.

Developed for streamers, content creators, podcasters, gamers and audio production professionals, the application delivers fast access to essential microphone functions through customizable Stream Deck keys, dials and foot pedals.

Compatible with the Audio-Technica AT2020USB-X, Audio-Technica AT2020USB-XP and Audio-Technica AT2040USB microphones, the 20 Series Control Application allows users to mute and unmute microphones instantly, adjust microphone input levels, and, for AT2020USB-XP users, control Automatic Gain Control (AGC) and Noise Reduction (NR) functions directly from the Stream Deck interface.

The plug-in integrates with the Elgato Stream Deck ecosystem through simple drag-and-drop configuration, enabling users to assign controls to specific keys, dials or foot pedals and customize layouts to match individual workflows for streaming, recording, gaming, podcasting or content production.

Among the plug-in’s key functions is instant mute/unmute control, allowing users to silence or reactivate their microphone with a single press. The feature is especially useful for live streamers and gamers; it can also be assigned to compatible Stream Deck foot pedals for convenient hands-free operation.

The Virtual Fader function enables precise microphone input adjustment directly from the Stream Deck interface. Users can configure the control to set fixed microphone levels or increase and decrease levels incrementally using keys or rotary dials.

For users of the AT2020USB-XP microphone, the application also provides direct access to the microphone’s onboard AGC and Noise Reduction functions. AGC helps maintain a consistent output level regardless of changes in speaking or singing volume, while the Noise Reduction control allows users to select from multiple levels of ambient noise suppression or disable the feature entirely depending on their recording environment.

The Audio-Technica 20 Series Control Application is available now through the Elgato Marketplace.

For more information, read a recent blog from the company.

In Australia, the patch bay is called a tail board. In the United Kingdom, the engineer responsible for matching cameras is often called a “racker” rather than a shader.

At the FIFA World Cup, those differences disappear remarkably fast.

The world’s largest sporting event brings together hundreds of broadcast professionals from different countries, cultures and technical traditions. Within days, engineers from Australia, England, the United States and beyond must operate as a single crew delivering one of the most recognizable sounds in sports.

They arrive speaking slightly different technical languages. The same piece of equipment or role can have multiple names. Yet by kickoff, those differences largely vanish. The audience hears one broadcast, not the dozens of production cultures working behind it.

What viewers remember is the roar after a goal, the tension before a penalty kick, the swell of a national anthem and the eruption of a stadium after a dramatic winner. They remember how the World Cup feels. Audio is a major part of that feeling.

Australian broadcast veteran Tim Stapleton, audio guarantee for Host Broadcast Services (HBS) in Kansas City, is one of the people responsible for creating that experience. A veteran of UFC, cricket, combat sports in the Middle East and the recent Milan-Cortina Winter Olympics, this is his first World Cup. The scale stood out immediately.

"The World Cup is unique because of the sheer number of moving parts," he said.

Big Event, Big Undertaking
Commentary facilities are handled by a completely separate company, with 10 dedicated booths inside the stadium. Outside, the production compound fills a parking lot with mobile rack rooms and support units from multiple vendors.

The biggest difference from a normal match is the atmosphere. “The crowd makes it sound like a magical moment, not a regular weekly game,” Stapleton said. “You can tell it is something special.”

HBS deploys a standardized audio plan across all venues while adapting to each stadium's characteristics. In Kansas City, the bowl design allows crowd noise to rise naturally, giving the mix team exceptionally clean audience capture.

Capturing that atmosphere requires far more than the standard soccer microphone package. The foundation remains familiar: pitch microphones positioned around the field to capture ball strikes, player movement and referee whistles, along with microphones mounted on cameras to follow the action.

Beyond that foundation, the World Cup expands significantly. Additional crowd microphones and immersive audio arrays are deployed throughout the venue to capture the atmosphere that defines the tournament. In Kansas City, ambient microphone arrays are positioned on the roof, camera deck, and pitch level, allowing the team to capture everything within the roar of the stadium.

The goal isn't simply to make the crowd louder. It's to make viewers feel present.

GEHA Field at Arrowhead Stadium (or, FIFA is calling it during the tournament, Kansas City Stadium) presents a unique advantage in that regard. Stapleton noted that the stadium's bowl design allows crowd noise to rise naturally through the venue, creating excellent opportunities to capture clean audience reaction.

A Mix That Tells a Story
For Stapleton, however, the technical design is only part of the equation. His approach to mixing is rooted in storytelling. Rather than simply balancing levels, he actively follows the action around the field, searching for sounds that draw viewers deeper into the match. The crowd remains a constant emotional foundation while individual moments are allowed to breathe naturally. "You're telling a story," he said.

Goals are allowed to build organically rather than exaggerated artificially. As the action moves around the field, microphones mounted on handheld cameras, Steadicams and other mobile systems help bring viewers closer to the moments they are seeing on screen. There is even a microphone mounted on the referee, though it is only used during specific review situations.

The Lionel Messi hat trick in Kansas City was a perfect example. “The reaction inside the stadium was immediate and overwhelming,” Stapleton said. Moments like that are exactly why so much effort is placed on crowd capture. The goal isn't simply to hear the reaction; it's to transport viewers into the stadium and allow them to experience the emotion of the moment alongside the fans in attendance.

The technology behind those moments may be sophisticated, but for many of the visiting engineers, the most memorable part of the World Cup has been adapting to life in the United States.

For Felix Harris, an HBS A2 from London with extensive Premier League and Club World Cup experience, the technical approach feels familiar, just on a larger scale. American stadiums, compounds, and travel distances are noticeably bigger than what he's accustomed to in the U.K.

"It's really tomato-tomah-to as far as nomenclature goes," Harris said. "But it's definitely a bit more." A lot more.

Andrew Lilley, another U.K.-based HBS A2, pointed to one practical difference: U.K. vehicle-length restrictions limit how much equipment can fit in a mobile unit. American trucks simply have more room, allowing for larger technical footprints and expanded microphone deployments. “A good tech sheet keeps everyone on the same page,” Lilley said.

Teamwork Comes Quickly
While terminology and workflows may vary, crews quickly find common ground. According to Stapleton, the real key to building a successful crew has nothing to do with technology.

“The first few days are essential,” he said. Crews learn each other’s strengths. They figure out how people communicate. They discover who works best in which roles. Shared meals, a drink after work and long production days help transform a group of strangers into a functioning team. “You spend so much time together that you become a family.”

Harris described the welcome from U.S. crews as warm and accommodating. Lilley joked about adjusting to the food, the lack of public transportation and a level of humidity that felt different from anything he experiences at home. “You do not do green vegetables in the States,” he joked. “And everything is on a bun.”

Stapleton has been impressed by Kansas City itself. “The best part is the locals,” he said. "Everyone has been friendly and willing to help.”

By kickoff, the terminology has been sorted out. The accents have become familiar. The workflows have aligned. Engineers who arrived from different continents are now focused on a single objective, helping billions of viewers feel connected to a match happening thousands of miles away.

Shared meals, long production days and countless hours spent solving problems together have transformed a group of strangers into a crew. The microphones, trucks and technology make that possible. The people make it memorable.

Together, they create the sound of the World Cup.

SKOKIE, Ill.— Studio Technologies has unveiled Model 385, a compact, lightweight Dante-enabled beltpack designed specifically for broadcast and live-event audio professionals who use parabolic microphones, boom poles and other specialized microphones to capture critical on-air audio.

The portable Model 385 combines a high-performance microphone preamplifier with a single-talk/two-listen intercom interface, delivering a unique combination of on-air microphone and intercom audio resources in a convenient beltpack form factor, the company said.

Supporting Dante audio-over-Ethernet networking and powered via Power-over-Ethernet (PoE), the Model 385 integrates seamlessly into modern fixed and mobile broadcast environments, it said.

“The Model 385 was developed to address the evolving needs of broadcast sports and live entertainment production teams that require reliable, high-quality microphone and intercom functionality in a highly portable format,” Studio Technologies President Gordon Kapes said. “By combining exceptional audio performance, Dante networking and flexible monitoring capabilities into a single rugged device, the Model 385 gives field audio personnel an efficient and intuitive solution for demanding production environments.”

Using a standard 3-pin XLR connector, the Model 385 supports both dynamic and condenser microphones, with selectable P48 phantom power available for condenser microphone applications. A low-noise, low-distortion microphone preamplifier circuit offers adjustable gain ranging from 20 decibels to 65 dB in 1 dB increments. The analog microphone signal is converted to 24-bit digital audio and output through a Dante transmitter channel.

A 5-pin XLR connector allows connection of a broadcast headset, with adjustable gain and low-voltage electret power support for the headset’s microphone. The output of the headset’s microphone is routed through a second Dante transmitter channel, activated via a pushbutton talk switch, to provide intercom communication with other Dante-enabled beltpacks and intercom devices.

“The flexibility of Dante networking was a major design priority for the Model 385,” said Kapes. “Users can easily integrate the unit with Studio Technologies products as well as a wide range of Dante-enabled equipment from other manufacturers.”

The unit is Dante Domain Manager-compatible and compliant with AES67. Crews can quickly get up and running with just a PoE Ethernet connection, microphone and headset while maintaining dependable communications and premium audio performance, Kapes said.

Model 385 allows users to create highly customizable headphone mixes by combining up to four audio sources routed to the headset’s dual-channel headphone output. Three push-in/push-out rotary controls enable adjustment of listen levels for the main microphone preamplifier output and the two incoming Dante receiver channels.

These channels are commonly used to support intercom and program audio feeds. An integrated sidetone function provides confidence monitoring of the headset’s microphone signal.

More information is available on the company’s website.

HEBDEN BRIDGE, England—Calrec Audio has updated its ImPulseV Virtualised Audio Mixing Engine to expand the market for virtualized audio workflows.

The ImPulseV portfolio now includes two new, smaller DSP package options—96- and 140-path configurations—alongside the existing 256-processing path option. Tailored for smaller systems that still demand uncompromising, broadcast-grade audio performance, all three packages share the same comprehensive feature set, including full processing on every path and support for 5.1 and immersive audio formats up to 7.1.4. DSP resources can also be dynamically allocated across channel and bus paths in real time, ensuring maximum efficiency for any production setup.

Calrec has also introduced shorter, more flexible licensing terms—one year, 12 weeks and four weeks—lowering the barrier to entry and allowing broadcasters to scale their production capabilities with no need for long-term commitments or high upfront costs, according to the company. This makes ImPulseV “an ideal solution for boosting resources for temporary, variable, or one-off productions,” Calrec said.

“Traditional broadcasters are reinventing their operations so that they can produce more content for less, faster and across more diverse platforms,” said Henry Goodman, director of product management at Calrec. “Remote and distributed production, combined with virtualization, reflect the broader trajectory of live production: away from fixed infrastructure and towards flexible, hybrid workflows that blend on-premise, cloud and distributed resources.

“This transition is enabling broadcasters to increase agility and cost efficiency through new business models," Goodman added. “Hybrid workflows that incorporate a blend of existing and new infrastructure allow broadcasters to build resource efficiency into every production—and that is quite an attractive proposition.”

AudioShake said it will launch its Copyright Compliance System, an end-to-end workflow for detecting, identifying, removing and documenting copyrighted music in mixed media files.

The Copyright Compliance System is already in use across broadcast, sports, postproduction and rights administration, with early customers including ESPN, NFL Films, Jaywalker Music, CrunchLabs, Music Reports and amp, the company said.

The technology combines several manual processes into one automated system, helping studios, broadcasters, sports teams and other distributors use enormous volumes of library content that can’t be redistributed because copyrighted music is baked into the original sound mix, according to AudioShake.

Music copyright compliance is a meaningful bottleneck in monetizing and distributing such content on FAST channels and streaming platforms, AudioShake said.

The AudioShake modular system allows customers to use the full pipeline or adopt individual components (detection, identification, removal) inside their existing workflow. Extending upon AudioShake’s Music Removal model, the Copyright Compliance System combines:

• Music detection, flagging whether music is present in a piece of content, including noisy, fully-mixed audio where music is buried under dialogue, crowd noise and effects. This helps downstream cue sheet teams know where music is present and where to fill out the relevant information.
• Music identification, isolating the music in any media file and identifing the artist, song title, album, label, release date and ISRC for each segment. Teams can quickly QC the identified music, combine repeated segments and correct start and end times to accelerate cue sheet creation and approval.
• Music removal, removing music from a clip while preserving dialogue, ambient sound and effects. In scenarios such as social distribution, archival relicensing, or platforms where rights would otherwise create friction, AudioShake opens the downstream path for rightsholders to license replacement tracks, swap in their own catalog or recoup royalties for music that previously went unreported.

One early user is ESPN, which manages a vast library of live, archival, and digital content, AudioShake said. Constraints such as copyrighted arena and walkout music, expired licenses on archival programming and dialogue embedded into crowd noise have historically limited what content ESPN could distribute and how quickly it could do so, AudioShake said.

ESPN uses AudioShake to remove or isolate copyrighted music from sports highlights, preserving original commentary and crowd noise, and to accelerate the turnaround of rights-cleared assets across television, streaming, digital and social platforms. Similarly, NFL Films and the NHL’s Calgary Flames use AudioShake to clear sports clips and game footage for broadcast and social distribution, where music rights would otherwise create friction.

AudioShake said its solution addresses music-licensing problems that have restricted distribution of digital properties like ESPN’s “The Ocho.” Previously unusable content can be distributed across new platforms and to new audiences, extending its lifespan and the reach of ESPN’s archives, AudioShake said.

ESPN used this capability in a 2026 Super Bowl ad. AudioShake isolated the voice of Phil Simms, the Super Bowl MVP quarterback of the 1986 New York Giants, saying “I’m gonna go to Disney World” in the famous 1986 postgame commercial for Walt Disney World. Because of ambient in-stadium music, licensing the full clip would have been cost-prohibitive for just a few seconds of a 60-second commercial. AudioShake removed the ambient music and extracted Simms’ vocal, making the clip usable.

“As ESPN continues to expand the ways we reach and serve sports fans in an age of content abundance, it’s important to lean into groundbreaking technology solutions that can help us bring the right content to the fans that want it as quickly and efficiently as possible,” said Kevin Lopes, vice president, business development and innovation, ESPN. “Working with AudioShake to leverage their innovative audio separation lets us unlock more content for fans by accelerating and modernizing workflows and ensuring we can deliver more high-quality sports content to fans wherever they are.”

More information is available here and here.

EMERYVILLE, Calif.—Brian Gross has joined the audio team at systems integrator Advanced Systems Group as an account manager.

He’ll be a part of the Burbank, Calif.-based group led by Director of Audio and Post Hamid “Gadget” Hopkins and support clients across applications ranging from music recording to studio design to immersive audio, ASG said.

Gross’ career is rooted in recording studios, music production and audio post, ASG said, and he has designed, built and run commercial recording facilities.

“I love music technology, but when I’m working with clients, it’s not just about selling equipment,” Gross said. “I’ve been in their shoes. I know the problems they’re trying to solve, and I know how to get them there. I’m the first person to say when they don’t need something. It’s about nurturing relationships and providing real value to people.”

ASG said Gross’ experience should be particularly valuable as recording and postproduction studios transition to fully digital environments.

“Brian knows audio technology, has a deep curiosity about it, has run large studios and truly wants his clients to flourish,” ASG President Dave Van Hoy said. “He understands the nuances of music production and post workflows, which helps design systems that stay out of the way of creativity. That perspective, combined with his consultative approach with clients, is particularly valuable as immersive audio and next-generation studio design continue to evolve.”

Among Gross’ top emerging technology priorities will be immersive audio and modular studios, ASG said. The company recently inked an exclusive reseller agreement with Nu-Studio, which offers a modular, portable control room.

“I’m really excited about the Nu-Studio platform as well as immersive audio and helping push both forward,” Gross said. “At the end of the day, I want to give people the best possible environments to create.”

I like to watch the beginning of big sporting events and was looking forward to hearing and seeing the band The War and Treaty sing the Star-Spangled Banner on “Monday Night Football” back in November.

The sound started out bad and, surprisingly, never got better during the entire performance. Clue one: Basic troubleshooting. Two singers were on wireless microphones, accompanied by an acoustic guitar—direct—three faders. Each channel was distorted and the balance between the sound elements never changed.

Before submitting this article and blaming the “MNF” crew, I decided to check out YouTube, where The Sports Video Channel was credited, not “Monday Night Football.” I am glad I did, because I was shocked—this certainly was not the same mix I heard live on my ABC affiliate.

What happened and who is really listening?

I can only ask, what was the mixer listening to? What was master control listening to? It reminds me of a favorite saying from the late, great television producer Fred Rheinstein: “Are we doing the same show?” What are you listening to?

Staying in Phase
In the early days of stereo sound over analog copper wire, it was not uncommon for the left and right channels to get out of phase. Often, this would occur in transmission, clearly beyond the event mixer’s control. It became common to send a “split track” of the announcers on the left channel and the other sounds on the right channel, with the two-channel (stereo) mix taking place back at master control. That worked, but I never thought it sounded very good, especially when you used an Orban stereo synthesizer to create stereo—“phase-y” stereo at best.

Surround sound was a nightmare with the widespread use of Dolby Pro Logic, an analog synthesis of surround sound delivered over two analog audio channels. If you had the decoder, then you could decode the surround, but if you didn’t, you had what was dubbed “super stereo”—once again, phase-y-sounding stereo. Dolby Surround was problematic until digital transmission and the first set of ATSC standards.

As a very green sports location sound mixer, I quickly learned to listen to and monitor the output of the OB van and as many other places as possible. There could be several processing or signal-splitting stages before the sound leaves the truck, and a problem could easily happen at a spot where you may not be listening.

After the audio program leaves the audio production space, further processing may happen somewhere in the audio signal flow. Isn’t someone listening to the sound? I remember the story about a master control technician who told the field mixer, “the meters looked fine there.” Who is listening to the sound?

But the question still lingers on how can the audio mixer produce sound for the masses when the consumer listens on earbuds, TV speakers, and sound bars? Not to mention the sound may be listened to in stereo, surround and even immersive by a few. How about language? Language intelligibility has plagued the broadcast sound mixer since advancing from mono to stereo. In mono, there is no gimmick like a “phantom center” that could disappear when the left and right channels are out of phase.

Immersive Challenges
Surround sound was difficult for the sound mixer because there were four channels of event sound and music and only one channel for dialogue. Compounding the mixing challenge is speaker alignment and placement, and often the surround speakers are too close to the mixer. Significantly, if the center speaker is too close it can give the sonic impression that the voices are too loud, resulting in the mixer turning the voices down and making them hard to hear.

With immersive sound the problems are further complicated by the fact that you have just added four overhead speakers. Now you have between eight and 10 effect and music speakers and still just one voice channel.

Part of the problem is how we define channels and how we mix them. I do not see any reason to not put voice in the left and right channels or in the front immersive channel, in addition to the center channel. You might argue that true reproduction in the home may be off, but then consider how sound bars project the sound. Who are you mixing for?

Mixing audio beyond stereo is arduous because of speakers and speaker placement, but significantly because of the acoustic mix space. World Cup soccer is hosted in multiple locations with different equipment and mix spaces making a cohesive consistent sound challenging.

Beginning in 2010, Felix Kruckles and Christian Gobbel of HBS (Host Broadcast Services) devised a signal flow and schedule where all the World Cup matches were produced in stereo in the venue and stems were sent to the International Broadcast Center. Then, a surround or immersive sound overlay was blended and mixed in a proper mixing room. All matches were mixed in the same audio mix studio—that is consistency! Felix and Christian were listening.

Who is listening? The misuse of compression is at an all-time high, and I would bet that it did not sound like that in the audio room. When there were only a couple of channels of compression, compression was tricky. Now virtually every audio channel and signal path has compression available, and maybe that is the problem—the compression is cumulative over the signal flow. I hear overcompression every time I turn on the TV. I bet it has to do with meeting the loudness numbers required by law!

Who is really listening? Maybe quality control is the best use of AI. You could program a gazillion qualitative and quantitative factors into a “QC bot” and it could “steer” the mix with some DSP into perfection so we can all listen in high fidelity.

The real question is, what did I really hear? I know I am not crazy! Yet.

Adobe has expanded its work with Speechmatics to deliver cloud-grade speech recognition on-device for its Premiere editing software that the company says can run accurate on-device transcription locally and is powerful enough for professional work.

Speechmatics has been Adobe's partner since 2021, when Adobe became the first non-linear editing platform to include speech-to-text (STT) in Premiere. The new feature deepens with a new on-device STT model in Premiere that delivers near-cloud accuracy while keeping all audio local to the device.

When Adobe launched STT for Premiere, large enterprises couldn't always use cloud-based services due to privacy concerns. Speechmatics was one of the few providers with on-device models—a key reason for the partnership.

Five years later, those privacy requirements haven't changed. With the rise of LLMs and data sovereignty concerns, the need for secure deployments has, in fact, increased. What has changed is the performance gap: Speechmatics' new on-device model brings local transcription on par with cloud accuracy with optimisations to run efficiently.

This means, Adobe explained, that studios, agencies, and production companies handling content before it goes public can now work seamlessly from anywhere: on a film set, between client meetings, on a flight—at full accuracy, with no dependency on a connection and no interruption to the work.

The new Speechmatics on-device model has been trained on millions of hours of speech to deliver high accuracy for accented speech, non-native speakers, and noisy environments like field reporting or film sets. As a result, Adobe said that the new on-device model in Premiere:

• Is within 5% relative to cloud accuracy, evaluated across nearly 10 million words of diverse real-world data
• Processes 1 hour of audio in about 55 seconds
• Leads the way against the closest competitor, with a 12-16% improvement against Whisper-powered creative solutions
• Runs on Windows & Mac, making use of the latest AI acceleration techniques to ensure efficient processing across a range of hardware, including broad hardware support for the latest Mac M5, NVIDIA RTX, AMD GPUs and older hardware such as Intel Macs.

Speechmatics on-device joins Speechmatics cloud and Speechmatics on-prem as a purpose-built option for ISVs and OEMs where data residency, offline capability, or predictable costs make local execution the right architectural call. It integrates as a C/C++ library on macOS and Windows.

More than ever, technology suppliers in 2026 are meeting media entities where they live and work, with a hybrid, flexible and often invisible ecosystem.

Audio infrastructure manufacturers and providers agree that while the “sound” still must be perfect, the exact location of the gear no longer matters.

“We are not moving towards a future where the physical location of the audio suite doesn’t matter anymore,” said Calrec Director of Product Management Henry Goodman. “We’re already there.”

WHO, HOW AND WHERE

While the “where” has shifted, the “who” remains the foundation of the broadcast. Humans are still largely mixing and shaping program output. “But how and where they do all this has changed beyond recognition,” Goodman said.

The show’s exhibit floor offers ample evidence for television, radio and streaming contexts.

Wheatstone is showcasing its VMX virtual mixing platform, which can serve as a single mixing backend for multiple consoles or surfaces, at its booth.

Marketing Director Dee McVicker said the cost benefits from virtualization are a focal point. “Not to mention removing the massive physical limitations of the studio as we know it,” she said.

For manufacturers, the challenge is ensuring that as consoles move from isolated desks to complex networks — whether on-site, on edge or in the cloud — the engineer’s experience remains intuitive.

Solid State Logic intends to decouple the “big console” experience from the giant desk. Berny Carpenter, SSL’s broadcast product manager, said that through the company’s Virtual Tempest Engine, the software can be deployed across public clouds or private data centers.

Does this flexibility mean that mixing can be performed truly anywhere? Wheatstone’s McVicker offered a nuanced take.

“We’re not taking away from the tactile feel of a fixed console, yet we’re not limiting you to that or where you might be mixing at the moment,” she said.

Goodman too said Calrec’s approach is to provide the “big console” experience in a portable package.

FIXED HARDWARE TO MOBILE APPS

In the West Hall, Lawo is preaching a philosophy of agility through its HOME Apps platform. The concept is a departure from the “one box for each task” style of broadcasting.

For a media outlet using a generic commercial off-the-shelf server, Lawo believes the platform is ideal for managing a succession of workflows, allowing a broadcaster to decide on Monday that it’s an audio mixer and on Tuesday that it’s a multiviewer by switching containerized microservices.

“Hardware infrastructure no longer dictates workflows: Workflows now define the agile processing stack,” said Wolfgang Huber, Lawo’s public relations manager.

According to Huber, that kind of agility is important in mobile production trucks, where every inch of rack space and watt of power counts.

Wheatstone’s VML, an HTML5-based virtual console, can be accessed through a web browser, offering mixing on a laptop, tablet, smartphone or desktop. “You have a lot of flexibility to set up remote broadcasts, temporary studios in a disaster situation or even add on a permanent production studio,” McVicker said.

BY YOUR SIDE

But how does an engineer harness all of the power available now, particularly from AI and automation?

At Telos Alliance, AI is framed as a “mixing assistant riding shotgun,” said Clark Novak, its marketing content manager.

Telos’ flexAI platform utilizes the Jünger Audio Intelligent Companion, an AI-powered automixer.

Novak explained that flexAI pushes background noise down while keeping dialogue consistent, allowing the engineer to focus on the overall mix while the platform handles “small corrective moves.”

SSL offers a similar philosophy with its System T Dialogue Automix. Carpenter said that the system focuses on the active speaker across a large number of sources to reduce crosstalk.

“It allows the operator to focus on the output, rather than reacting to who is currently speaking,” Carpenter said.

Lawo, for its part, sticks to deterministic algorithms in its KICK sports mixing technology, for example, to avoid “AI hallucinations” in high-stakes environments.

But with the platform’s close-ball mixing, which is automated based on camera tracking data, Lawo’s Huber said the technology allows the audio engineer to focus on creative storytelling rather than manual gain-riding.

But a recurring theme is that disappearance of large surfaces. For instance, Telos Alliance’s Axia Altus and Studio Essentials lines allow mixing to be controlled entirely via a web browser.

“Virtual mixing isn’t about replacing the studio,” Novak said, “it’s about expanding it to other locations.”

Wheatstone’s McVicker called it a hardware-agnostic philosophy.

As NAB Show exhibitors will demonstrate, the movement to flexibility is not an all-or-nothing proposition. Most facilities are adopting a hybrid approach — keeping hardware where tactile control is needed, but moving the “heavy lifting” to the cloud.

As Calrec’s Goodman concluded, the tools must provide the operator with the ability to adapt to demanding environments.

Whether the processing happens in a basement rack or a distant data center, the mission remains the same: clear and compelling audio.

© 2026 NAB

LAS VEGAS—In a notable effort to improve the quality of audio in OTA broadcasts, Dolby Laboratories and TV Azteca have announced a project to bring Dolby Atmos to free-to-air television in Mexico over ATSC 1.0 transmissions.

The two companies said the distribution of immersive audio within the current ATSC 1.0 transmission standard marks a significant step forward in the evolution of live television and opens the door for other broadcasters to make similar improvements to audio quality of OTA broadcasts.

Those involved in the project said immersive audio has a particularly powerful appeal in live sports events, where sound is an essential part of the emotion. With Dolby Atmos, sound moves above and all around, placing viewers at the center of the action and making every game more thrilling and memorable, Dolby said.

“Our work with TV Azteca represents a decisive step in bringing immersive experiences to mass audiences in Mexico,” said Pankaj Kedia, vice president of Americas, commercial partnerships at Dolby Laboratories. “Imagine watching a game from your living room and hearing where every cheer in the stands comes from, the sound of the ball, and the voice of the commentator moving around you. That is what Dolby Atmos makes possible today in Mexico.”

Pedro Manuel Carmona Ortiz, CTO of TV Azteca added that “this project with Dolby Laboratories marks a technological milestone, as well as a turning point for the industry, which would not have been possible without the support and collaboration of our strategic partners in this journey.”

In addition to Dolby, those include Imagine Communications and Ateme, a global provider in video compression and distribution solutions.

“We are collaborating with Dolby to demonstrate that innovation in audio can transform free-to-air television,” added Pedro Carmona Ortiz. “Our goal is to bring people closer to the real emotion of every story, with a quality that was previously reserved for premium environments.”

The announcement was made in Las Vegas, during the 2026 NAB Show, where Dolby, Imagine and Ateme are exhibiting.

LAS VEGAS—Calrec (Booth:C6907) and Grass Valley (Booth: C2408) are announcing a long-term broadcast audio technology partnership that they will be showing to attendees at the 2026 NAB Show.

The companies are combining their industry expertise to deliver a software-defined production environment that will see Calrec's ImPulseV virtual DSP software integrated with Grass Valley's AMPP cloud-native live production platform. The deployment will provide broadcasters with a unified environment for managing live broadcasts.

Coupling Grass Valley’s solutions in video production with Calrec’s broadcast audio expertise, ImPulseV, strengthens audio workflows within Grass Valley’s AMPP market leading Ecosystem.

The companies noted that the partnership provides all the benefits of close integration and orchestration with proven applications from renowned broadcast partners on a single platform. Calrec’s pathway into AMPP OS will start with AMPP Control integration and evolve towards a fully native ImPulseV application running on AMPP.

The collaboration also strengthens AMPP's tier 1 audio capabilities alongside its tier 1 video production strengths, creating a single platform for high-end live productions.

The integration comes at a time when the industry is moving towards software defined production, open standards and collaboration. As hybrid workflows continue to develop, the partnership gives broadcasters the confidence to define and scale their own infrastructure requirements, and to fully utilize their resources.

“We are delighted to be partnering with Grass Valley, an organization that shares our belief that innovation, people, and technology define the future of our industry. In today’s fast-moving production landscape, strong solutions and trusted collaborations are more critical than ever,” explained Sid Stanley, managing director at Calrec. “Our virtual ImPulseV technology, combined with the Grass Valley’s AMPP Ecosystem, creates a powerful proposition, uniting best-in-class solutions to give broadcasters the freedom to build, scale, and adapt without compromise. This marks the next evolution of Calrec’s strategy, empowering our customers with true choice, flexibility, and control in a software-defined world.”

Adam Marshall, chief product officer at Grass Valley, added that “with the integration of Calrec’s ImPulseV into AMPP, we are extending the power of software-defined production with tier-one audio workflows. Media organizations can now deploy premium audio capabilities in a fully software-defined environment while maintaining the quality, responsiveness, and familiar operator control expected for high-value live production. Whether on-prem, in the cloud, or across hybrid environments, it all operates within a single, unified platform”

Stanley stressed that “as our industry transforms, the need for agile, scalable, future-ready solutions has never been greater. Together, Calrec and Grass Valley are setting a new standard, combining cloud, virtualization, and multi-vendor integration into one seamless ecosystem for next generation production.

“The introduction of ImPulseV into this ecosystem is a key milestone, enabling customers to deploy everything from micro-control rooms to global production environments with speed and precision - shaping infrastructure around their needs, not the other way around,” he added. “This is not just an important announcement for today, but a meaningful step towards the future. Both Grass Valley and Calrec are deeply committed to continuous innovation and investment - because in a world moving this fast, standing still is not an option.”

Both companies noted that audio has remained one of the most challenging workflows to virtualize without compromising quality, latency or the operator experience. The cohesive Calrec ImPulseV and Grass Valley AMPP integration is designed to meet these challenges, by delivering:

• Common orchestration
• Shared control
• Scalable infrastructure
• Cloud and on-prem flexibility
• Elastic resources aligned to live production demands

Calrec's ImPulseV creates flexible workflows across COTS hardware, public and private cloud, existing processing hardware and distributed resources, contributing to an efficient, flexible hybrid production environment. ImPulseV is part of Calrec's Ecosystem, powered by True Control 2.0, which unifies the company's high-end control surfaces with remote production capabilities. Broadcasters can use True Control 2.0 to manage up to five Calrec consoles or processing cores simultaneously.

By integrating ImPulseV with AMPP, broadcasters' flexibility to manage their assets is further expanded. One control surface can adapt – without loss of quality – to local, cloud and remote productions.

Both ImpulseV and AMPP provide OpEx based deployment models, enabling broadcasters to expand event coverage. The integration delivers audio capabilities that scale, without compromising the operator experience.

Calrec is exhibiting at the NAB Show between April 19 and 22 on Booth C6907, and Grass Valley is on Booth C2408. The integrated ImPulseV and AMPP workflow will be demonstrated on the Grass Valley Booth C2408.

CLEVELAND—Broadcast audio supplier Telos Alliance will be showcasing four new products as well as upgrades to existing solutions at the 2026 NAB Show from April 18 - 22 in the Central Hall of the Las Vegas Convention Center, Booth #C1819.

During the show, the company will feature its portfolio of traditional hardware products alongside virtualized and software solutions.

Several brand-new products will be introduced for the very first time at this year’s show, including:

• Omnia® XII, the brand-new flagship FM/HD/DAB audio processor from Frank Foti and the Omnia team that combines all-new algorithms and Frank’s effortlessly loud and clean “Clemenza” clipper with forward-thinking I/O and features users require to support today’s workflows (https://omniaxii.com)
• Axia Pulsar, a compact new AoIP console surface with 8 motorized faders - expandable to 16 - a touchscreen monitor section, and the same top-tier audio performance as the company’s larger Quasar XR and SR consoles (https://telosalliance.com/pulsar)
• Telos Infinity IP Intercom MK2 Master, Expansion, and Desktop panels, featuring 16 bigger, brighter, easier-to-read TFT button displays (https://telosalliance.com/infinity)
• Telos Infinity VIP SE, the latest member of the growing StudioEssentials family of compact hardware products, which merges the flexibility of Infinity VIP software with the convenience of easy-to-deploy compact hardware (https://telosalliance.com/VIP)

Improvements to several of the company’s existing offerings will also be demonstrated. The Jünger Audio flexAI platform features the Jünger Audio Intelligent Assistant (JAIC) to aid in managing complex multichannel mixes in real time, and the flexAI platform is now fully deployable in the cloud (https://telosalliance.com/flexAI)

Meanwhile, the Minntonka Audio AudioTools Server automated file-based processing platform addresses one of the top issues faced by television viewers: Dialogue intelligibility. ATS can analyze audio content and, if necessary, make corrections to ensure viewers no longer struggle to hear and understand dialogue within the overall mix (https://telosalliance.com/ATS)

Partner ip-studio will showcase its STUDIO ZERO ecosystem in the Telos Alliance booth, a fully-realized, 100% virtual cloud-hosted solution first introduced at the 2023 NAB Show. STUDIO ZERO merges a suite of Telos Alliance products with ip-studio’s advanced automation and workflow management software, making practical, reliable cloud-based studios a reality (https://ip-studio.com).

More information is available at https://www.telosalliance.com/.

NORTHRIDGE, CALIF.—TSL today announced the acquisition of JLCooper Electronics, a pioneer in professional control, automation, synchronization, and interfacing products.

The announcement was made after the death of founder Jim Cooper was announced on March 16. TSL says the acquisition “ensures continuity of service and support for its global customer base spanning live events, music, radio, pro-AV, and education markets.”

Cooper got his start at UCLA AND was also Chief Engineer at Oberheim Electronics before launching JL Cooper in 1979. He was also a former President of the MIDI Manufacturers Association.

Notable JL Cooper customers include Cirque du Soleil, NBA, Grand Ole Opry and global artists including Depeche Mode, Pearl Jam and Hans Zimmer, as well as a wide range of OEM clients that align with TSL’s strategic growth plans.

TSL says the move further expands its capabilities into adjacent markets while broadening its technology offering to its core broadcast customers.

“JLCooper’s range of interfaces will be great additions to TSL’s Hummingbird control ecosystem, giving creators, operators, and engineers expanded options and greater flexibility in how they connect and control systems within the latest software-defined, cloud, and hybrid workflows,” the company said in a statement. “The addition of MIDI control and communication solutions further enhances TSL's capabilities and broadens TSL’s industry reach into music and audio production, opening up the enormous breadth of devices compatible with the MIDI industry-standard, which Jim Cooper himself helped to create and establish.”

Following the acquisition, TSL will move JLCooper’s production operations from El Segundo, CA to its base in Northridge, CA after which TSL will commence production and shipping of JLCooper solutions.

“With the sad news of Jim Cooper’s passing, it was important to find a way to secure the future for JLCooper’s revered technology, while ensuring continuity for its customers and we’re glad we’ve been able to move so quickly to achieve this,” said Matthew Quade, CEO, TSL. “Bringing its solutions into the TSL Hummingbird ecosystem, which already combines and has expanded the much-loved Tallyman and DNF Controls capabilities, will only enhance the flexibility and choice available to our customers, while broadening the markets that Hummingbird can support.”

“We were all terribly shocked and saddened by Jim’s passing, and have been working round the clock to find a way forward for his business that would ensure his legacy continues and maintains support for the users that have grown to love JLCooper solutions for the past 46 years,” said Chuck Thompson, General Manager, JLCooper Electronics. “I’ve been delighted by how TSL has been able to provide a route forward in a short timescale, and the professional understanding and support the TSL team has shown. I can’t imagine a better home for the brand and products and look forward to working with TSL on providing customers with enhanced options for their future workflows.”

BURLINGTON, Mass.—Avid has released data showing that the filmmakers, editors and sound teams worked with Avid Media Composer and Pro Tools to create the vast majority of this year’s Academy Award®-winning pictures.

In total, 87% of winning productions were crafted using Avid solutions, reinforcing Avid’s long history as the creative platform trusted for the world’s greatest storytelling, the company reported.

“Well-deserved congratulations to all the filmmakers, editors and sound teams recognized at this year’s Academy Awards,” said Wellford Dillard, CEO of Avid. “These films didn’t just entertain audiences – they moved, thrilled and inspired millions of people. We’re proud to see the world’s most important stories shaped by artists using Avid tools as part of their creative process. That’s what drives us every day – building technology that helps the world’s greatest creators deliver their best work.”

Filmmakers working with Avid creative solutions were recognized across major categories including Best Picture, Best Film Editing, Best Sound and Best Animated Feature Film. Winning productions include:

• Best Picture: `One Battle After Another'. Producers: Adam Somner, Sara Murphy and Paul Thomas Anderson.
• Best Film Editing:`One Battle After Another'. Editor: Andy Jurgensen.
• Best Sound: `F1: The Movie'. Sound team: Gareth John, Al Nelson, Gwendolyn Yates Whittle, Gary A. Rizzo and Juan Peralta.
• Best Animated Feature Film: `KPop Demon Hunters'. Producers: Maggie Kang, Chris Appelhans and Michelle L.M. Wong.
• Best International Feature Film: `Sentimental Value' (Norway).
• Best Visual Effects: `Avatar: Fire and Ash'. Visual effects team: Joe Letteri, Richard Baneham, Eric Saindon and Daniel Barrett.
• Best Documentary Feature Film: `Mr. Nobody Against Putin' Producers: David Borenstein, Pavel Talankin, Helle Faber and Alžběta Karásková.

Alongside a wide array of this year’s winning productions, Avid also reported that its editing and audio solutions were used across the majority of nominated works in several major categories, with 8 of the 10 Best Picture nominees were crafted using Avid, 4 out of 5 editors in Best Film Editing cut their films with Media Composer, and and all Best Sound nominees used Pro Tools.

For example, Andy Jurgensen, winner of the Best Film Editing Oscar for `One Battle After Another', cut the film using Avid Media Composer®.

“This film had so many tones – comedy, drama, action – it’s so hard to compartmentalize,” said Jurgensen. “One thing I really got into on this movie was Avid’s automatic transcribing tool… When you need to find just a word or a phrase you can search and find it in all the different takes. It saved me so much time.”

The sound team behind `F1: The Movie', which won the Oscar® for Best Sound, used Avid Pro Tools workflows to craft an immersive racing experience that captured both the authenticity and excitement of Formula 1.

“What makes F1 exciting is that it is the peak of motorsport racing. It is the top of the top,” said Juan Peralta, re-recording mixer at Skywalker Sound. “One of the edicts we got from the producers of this movie was to try to make it as authentic as possible… We also were making a fun movie – we want people to enjoy it and just be on for the ride.”

Learn more about the creative process behind `F1: The Movie' here: https://connect.avid.com/f1-the-movie-lp.html

Find out more about Andy Jurgensen’s editing experience on `One Battle After Another' here: https://theroughcutpod.com/one-battle-after-another/

HEBDEN BRIDGE, U.K.—Calrec will unveil solutions for broadcasters that unlock the full potential of hybrid workflows that provide greater freedom, flexibility and choice in how they configure its products during the 2026 NAB Show, April 18-22, at the Las Vegas Convention Center.

Building on its hardware foundation, Calrec’s compact IP-native, 48-fader Argo M audio console with integrated digital signal processor (DSP) will make its U.S. debut. With up to 356 channels of internal DSP, Argo M delivers the power, familiarity and reliability of Calrec’s Argo platform in a streamlined 24-, 36- or 48-fader footprint.

The Argo M maximizes desk space with full-sized fader strips and intuitive controls for the most demanding production. Plug-and-play SMPTE ST 2110 and cloud connectivity deliver immediate high-performance control.

The ability to connect Argo M to an ST 2110-networked environment and to an existing Calrec ImPulse core alongside other Argo surfaces means broadcasters can quickly create multi-console IP environments. Seamless connectivity to ImPulse, ImPulse1 and ImPulseV means additional external DSP paths can be added as needed.

Calrec will also showcase key software updates and a feature preview at the show, including:

• Argo v1.4.24, providing the functionality of 1.4 and operator enhancements, which is available immediately.
• Type R v3.2, delivering improved Mix Minus outputs, Assist Upstand meter tiles, Off Air record contribution dialogue and Syslog support with a Q2 release.
• A preview of Argo Offline Assist, which allows offline configuration and will be available in Q4.

The company will also preview ImPulseV running on virtualized server hardware. Calrec's ImPulseV virtualized audio mixing engine creates workflows across multiple platforms whether COTS hardware, public cloud, private cloud or existing processing hardware and distributed resources, contributing to an efficient, flexible hybrid production environment.

Calrec’s ImPulseV employs the same familiar user interface and can be controlled directly from any Argo or True Control 2.0 enabled console or via the company's Assist web UI.

The company will also demonstrate its IP-native 24-fader Argo M console with ImPulseV cloud DSP software, which is designed to be deployable across multiple hosting environments, giving broadcasters ultimate choice as to where and how they deploy.

Calrec will also demonstrate True Control 2.0, the glue that connects Calrec’s IP ecosystem together. Its IP Ecosystem powered by True Control 2.0 allows users to blend on-premise infrastructure with remote operations and virtual cloud resources. By decoupling audio and video from hardware, teams can scale dynamically, orchestrate thousands of audio channels across distributed DSP environments and virtualize entire workflows.

It will also feature Argo S with IP connectivity, version 1.4 Argo software, Argo Q and its Type R modular, expandable IP mixing system,

See Calrec at 2026 NAB Show booth C6907.

More information is on the company’s website.

Broadcast audio has undergone a dedicated development process over the years, progressing from mono to stereo, analog to digital and baseband to Audio over IP (AoIP). Despite this, it continued to live in the shadow of broadcast video and was almost subsumed when embedded audio was introduced in the 1970s. Today, with dedicated standards and protocols such as AES67, Dante and Ravenna, the position of audio in TV has grown in stature while also coming under the aegis of the more video-oriented SMPTE ST 2110.

This, however, does not appear to be a hindrance to the ongoing evolution of TV/streaming audio, now firmly established on an AoIP foundation and looking towards the continuing implementation of Next Generation Audio (NGA), with AI promising more practical benefits for processing and control.

Building on the Backbone
The advent of ST 2110 enabled the discrete transport of audio and video, giving operators the freedom and flexibility to send either stream to different destinations. This is part of the “second wave of IP,” which Christian Struck, senior product manager for audio expert infrastructure at Lawo, predicted will be at the 2026 NAB Show, April 18-22 in Las Vegas.

“It’s fair to say that AoIP is the de facto standard now, but I think what we will see at the NAB Show will build on that as the backbone architecture [for systems],” he said. “This means using the abstraction layer or the generic infrastructure that is capable of dealing with audio and video in an equal manner…to abstract further away from bespoke hardware and move the offering towards software values.” He also highlighted the EBU’s recommendations for the Dynamic Media Facility [DMF] as “a clear trend” towards generic backbone architecture that will “allow operators to define how to utilize these facilities for whatever purpose and on what scale.”

Costa Nikols, executive team strategy adviser for media and entertainment (M&E) at Telos Alliance, picked up on the concept of the DMF, adding that ST 2110, NMOS (Networked Media Open Specifications) and virtualized audio engines are now the “baseline” for new builds and major refurbishments.

“MXL [Media eXchange Layer] is [also] becoming predominant in the broadcast space, allowing for different virtual machines and processes as well as memory share, with ST 2110 for signal flow and a bunch of transport standards like SRT and NDI and, in the audio world, Dante,” he said. “As well as those, there are software-as-a-service [SaaS] platforms from companies including Grass Valley, Vizrt and Matrox, [all of which can enable] virtualization like the DMF, configuring different rooms via software depending on the function you need to have that specific day.”

Going Full 2110
The Dante AoIP protocol is now firmly established in broadcast and its developer, Audinate, has expanded further into the market with video functionality. “Operationalizing IP and software workflows at scale” for broadcast audio is a growing trend, Principal Product Manager Will Waters said.

“New facilities are being brought about with IP at the core, with ST 2110 as the backbone,” he said. “But it’s not just about the [audio] quality, which is there on the digital side. It’s more about how you operationalize that, scale the network and ensure you don’t get into challenges because we really need a deterministic behavior in the audio, where networks are inherently packetized, so they’re meant to be agnostic to what’s flying over them. The trend I see is operationalizing that alignment with PTP [Precision Time Protocol] across the facility to get better control of multicast flows.”

Jason Waufle, global strategic business development manager at Shure, observed that Dante and AES67 are now being backhauled into ST 2110 networks as part of OB truck integrations.

“The industry consensus is a full ST 2110 or IP infrastructure is the way to go,” he said. “You’ve got significantly more flexibility, control and interoperability of audio and video and transport. Shure has been making Dante and AES67 native products for nearly a decade, and advancing our onboard DSP [to] make our array microphones essentially an edge device for a network, capturing more audio data than a single analog mic could.”

Phil Owens, senior sales engineer at Wheatstone, said there are also advancements in routing and distributing AoIP in live broadcast.

“The preferred method of operation now is to be able to grab any source or signal and send it and use it anywhere you want,” he said. “That kind of distribution and sharing is right in the IP wheelhouse, whether it’s long-distance transport with AoIP or just within the same building.” Right now, Owens added, Wheatstone is focused on server-based systems that can be in the cloud, but he sees on-prem setups as more practical. “Server-based apps can provide scalability and redundancy that bespoke hardware is lacking.” he said.

Immersive and Personalization
While AoIP is now either being implemented or is part of broadcasters’ plans across the sector, NGA, immersive systems such as Dolby Atmos and personalization of audio for alternative languages and commentaries are more confined to streaming services.

“Amazon and Netflix are overtaking the conventional linear TV broadcasters [when it comes to] Dolby Atmos and binaural,” Struck said. “It has always been a significant challenge to make the infrastructure and complete signal chain immersive-capable. It’s much more common to the large streaming services than it is for broadcasters, where only exceptional and selected things are transmitted in immersive.”

Nikols is more positive, saying Telos is seeing immersive and personalized audio “showing up in strategic plans” for new projects and installations. “They’re no longer just nice to have,” he said. There is now a “software-first virtualization mentality” in broadcast, Nikols continued, coupled with AI-assisted workflows. “It can assist in the creative process, [through] loudness control, compliance monitoring, speech enhancement and noise reduction.”

Another assistive application is in audio mixing, with Shure and others using AI to develop systems that can handle the more mundane aspects of a production. “We came up with the capability of developing DSP that, from an AI standpoint, allows the engineers not to have to worry about the audience mix because it does it automatically,” Scott Sullivan, vice president of strategy and innovation, explained. “But the things that are more in the hands of the professionals, that’s going to be significantly down the road. Right now, I think AI is a complementary tool.”

No doubt AI in all its forms will be prominent at the 2026 NAB Show, but it could be the less-showy aspects of audio, such as the second wave of IP, that really point the way ahead for this ever-evolving sector.

Driven by advances in IP, the cloud and AI, audio technology is more than just about sound—it's about providing viewers with a more immersive and accessible experience. In our latest Guide to Audio, we examine the latest trends in audio technology and discuss how broadcasters and media companies can bring these innovations to life across live, linear and streaming workflows.

Download the Guide here.

BRAKEL, Germany—Schulze-Brakel has unveiled SchulzClip, its dedicated branding solution for wireless clip-on microphones.

Designed for modern wireless microphone systems, SchulzClip allows broadcasters, production companies and brands to place their logo directly on camera without compromising sound quality or usability. The product clips securely onto shirts and collars and provides a clean, professional on-camera appearance while reducing wind and pop noise.

Manufactured using SchulzFoam technology, SchulzClip ensures clear, natural audio and features a non-reflective flocked surface suitable for professional broadcast environments. High-resolution, full-color logo printing enables strong, customized brand visibility.

“SchulzClip combines professional audio performance with visible brand presence in a way that has not been available before,” says Jens-Uwe Rumsfeld, CEO of Schulze-Brakel.

SchulzClip is compatible with wireless microphone systems from Shure, Sennheiser, DJI, RØDE, Saramonic and Deity. The product is ISO 9001 certified and made in Germany.

More information is available on the company’s website.

STAMFORD, CONN—NBC Sports has selected Audio-Technica to once again provide professional microphones, broadcast headsets, headphones and related accessories for its production of the 2026 Milan Cortina Winter Olympics and Paralympics.

NBC Sports will utilize Audio-Technica equipment to deliver much of its audio capture requirements across all sporting venues for its production of the 2026 Winter Olympics and Paralympics, helping NBC Sports route the excitement in Italy back to viewers watching coverage of the Games in the United States.

“NBC Sports is proud to mark 25 years of partnership with Audio-Technica, whose advanced audio technology solutions have played a vital role in supporting NBC Olympics’ production efforts around the world. Throughout this longstanding partnership, Audio-Technica has continued to deliver premium products and pioneering innovations in microphone technology, serving as the foundation of an audio signal chain dedicated to superior sound quality and reliability. Audio-Technica consistently delivers outstanding performance, Games after Games, whether in the host city as well as at our Stamford broadcast facility,” said Karl Malone, senior director, audio engineering at NBC Sports.

“We are delighted to be able to once again provide an extensive package of microphones, headsets and headphones that meets the demanding of technical requirements of NBC Sports”, states Rodrigo Thomaz, project manager – broadcast partnerships at Audio-Technica with technical responsibility for the collaboration; “and that they continue to have the confidence in our brand to deliver both on audio quality and logistical excellence.”

Click here for more on NBCUniversal’s Milan Cortina 2026 programming.

TV Tech’s overall coverage of NBC Sports' overall tech plans for the 2026 Winter Games can be found here, here and here.

Talking to Karl Malone, senior director of audio engineering at NBC Sports and Olympics, about the sound he intends to capture and deliver during the Winter Games in Milan-Cortina reveals a passion and excitement for the event he hopes viewers will sense at home.

A critical part of making that happen will be the extensive use of immersive microphones to take advantage not only of the five speakers and subwoofer of a 5.1 surround mix, but also of four height channels dedicated to picking up mics elevated to capture a fuller soundscape and immerse the audience in the moment.

“It’s usually quite cold and potentially not too windy up there [in the starting hut of downhill skiing], but it’s isolated, and I think immersive mics are really going to give us that feeling of isolation,” said Malone, offering an example of how channel-based immersive audio will benefit the coverage of one event.

“How we mix these normally, you’ll never hear any crowd in the mix because it’s all about the athletes and their time and the focus on them, the zoom in on the eyes and listening to the audio,” he said. “You see the breath, and you hear the breath. Then, as they go down the hill and they’re vying for a world record, you start to bring in the crowd and immersive.”

Dolby Atmos and Audinate Dante
Delivering immersive audio leveraging 10 channels of Dolby Atmos audio will be a major priority for the network during the Games. “We have large 8.0—basically four height and four sort of lower-level—arrays that we will use for the general ambience at pretty much all venues just to get that larger immersive crowd sound whether indoors or outdoors,” Malone said.

“Then we have some smaller immersive mics as well, which are Dante-based,” he continued. “That’s a huge advancement in being able to get immersive audio into tighter spaces and also being able to do it on a single Ethernet cable, which is wonderful.”

Using Dante enables the A1s mixing coverage at the NBC Sports hub in Stamford, Conn., to control mics strategically positioned around venues in Milan-Cortina. Accessing the mics through a remote desktop, the A1s can not only operate them, but also steer pickup in real time, as necessary, from 4,000 miles away.

“One of our hopes is to get these types of mics into the starting houses of the downhill races where the venue is smaller,” he said, specifically identifying how he hopes sound will convey “the pressure and the tension of being very high and very alone on the top of a mountain.”

“You can feel it, and you have to be able to hear that, too,” Malone said.

Meanwhile Back in Stamford
NBC will use 14 control rooms in Stamford, four of which can be considered primary control rooms, he said. One will be used for primetime, a second for daytime coverage, a third for USA Network and a fourth for NBC’s “Gold Zone,” its whip-around streaming coverage on Peacock, presenting the most exciting action from various venues. Each primary control room will handle 5.1.4 immersive mixes.

The remaining control rooms in Stamford are what the network refers to as “venue control rooms,” used to mix audio for ice hockey, cross-country skiing, speed skating, Alpine sliding, aerials and moguls, as well as at the Olympics Snow Park, where freestyle skiing and snowboarding will take place.

“Those control rooms act like trucks, feeding into a master audio control room where they’ll take the eight channels of immersive height information and mix that down into four,” Malone said. “So, it’s a finished product in 5.1.4.”

For this year’s Games, the network is giving its editors in Stamford the ability to edit 16 channels, not simply 5.1 surround. “Ultimately, you want those 16 channels to go back into a control room so that the A1 can play it back as live and have access to those eight channels of height microphones to be able to mix down into four channels,” Malone said.

“We championed the ability to maintain that whole 16-channel workflow through the building—from the venue all the way through edits and then back to a live control room,” he said. “That’s pretty important.”

There will be two exceptions to this remote integration model (REMI) workflow, however: the opening ceremony and figure skating.

U.S. viewers and the network regard figure skating as a “top-tier sport,” Malone said, and as such, NBC wants to have more production personnel on site.

“They [NBC Olympics reporters and talent] have more access to athletes,” he said. “It’s a bigger production with more cameras and more NBC cameras. There are more NBC microphones. It’s just bigger.”

The same is true of the opening ceremony, so both will be produced from an on-site mobile unit.

Distribution to Various Platforms
Audio from Stamford will be handed off as discrete 5.1.4 PCM to be encoded to Dolby Atmos and delivered to Comcast, Peacock, multichannel video programming distributor (MVPD) partners and the network itself. At that point, the network will remove the height channels for distribution to affiliates. “All of the NBC O&Os will get AC-4 [the codec used by Dolby Atmos and integral to ATSC 3.0],” Malone said.

“What we did for Paris [2024 Summer Olympics] is they [the owned-and-operated stations] took our network 1080i and up-res-ed it to 1080p HDR with Dolby Vision and Atmos for NextGen [TV],” Malone said, adding that some affiliates transmitting NextGen TV will do the same this time.

“The AC-4 pipe has already been laid out to NBC O&Os,” he said. “It’s really testing the pipe at this stage, and we’ve done very successful AC-4 tests with both immersive AD [audio description]—being able to take audio description and present that in a 5.1.4 presentation.”

To ensure its 5.1.4 immersive audio presentation is available to Peacock subscribers, the streaming service has conducted extensive testing of home devices, such as Roku boxes, Amazon Fire TV Sticks and others, to ensure glitch-free decoding and presentation, Malone said.

“That’s why it took a long time in my mind for Peacock to go immersive,” he said. “It’s because there are the creatives, including me, saying: ‘We’re ready to do this creatively. We can give you 5.1.4.’ But Peacock had to test every single device. They aren’t going to launch without making sure everything is right.”

Planning for the 2026 Winter Games started right after the Summer Olympics, with the goal of one-upping NBC’s award-winning coverage of the Paris Games.

“You don’t sit on your laurels,” Malone said. “You think: ‘What can we do better? How can we engage the audience more? How can we tell the story our directors and producers want to tell?’”

For the Milan-Cortina Games, when it comes to audio, the answer is clearly immersive.

Sports television has always chased immersion. Higher frame rates, longer lenses and increasingly complex graphics all served the same goal: putting the viewer closer to the action.

But for all the visual advances, sound remains a powerful shortcut to realism.

At the U.S. Figure Skating National Championships in St. Louis on Jan. 11, a quiet piece of audio innovation brought audiences closer to the ice than ever before by placing microphones directly inside it.

Cutting Into the Ice
Broadcast audio technicians arrived in St. Louis early to help the ice operations team install prototype Audio-Technica contact microphones directly into the skating surface. These contact microphones converted vibrations in the ice into an electrical signal that could be amplified for broadcast.

From there, a narrow trench ran from the mic location back to the dasher boards, allowing the cable to exit cleanly at the rink perimeter. Once the microphones were seated, the cabling was dressed tightly against the wall so it could later be tied into the broadcast infrastructure.

After installation, the ice team packed slush over the microphones and trenches, resurfaced the area, and painted the ice to conceal any visible lines. When finished, there was no visible evidence that 10 microphones were sitting just beneath the surface, positioned evenly around the rink.

Once the physical work was complete, the broadcast audio team tied the microphones into the broadcast truck, where the real test began.

A Different Kind of Microphone
The prototype microphones were contact mics, capturing only vibrations from the ice. Audio-Technica supplied the mics, with Gary Dixon, director of broadcast business development, closely involved in testing and exploring their applications.

“The goal is always the same,” Dixon explained. “How do we make sounds more interesting to people?”

Because the element was embedded directly into the ice, the microphones were inherently isolated from ambient noise. The mics use standard phantom power and were connected to conventional microphone inputs, with an internal amplifier bringing the piezo-element (an electrical charge or voltage generated by mechanical stress) signal up to microphone level.

Crowd reaction, PA spill and arena noise simply did not factor into the signal path. What remained was pure mechanical sound: blades carving ice, shifts in pressure, landings, takeoffs and the rhythm of the skating itself.

The microphones used in St. Louis were available through Audio-Technica’s rental program, with plans for a future commercial release. Dixon sees broad potential well beyond figure skating, such as basketball rims, soccer goalposts, swimming environments and even engines. Anywhere vibration tells a story, contact microphones offer a new way to listen.

Mixing the Ice
For Randy Pekich, senior audio mixer with NBC, the first listen was eye-opening.

“When I heard them, my first reaction was, ‘No way that’s a shotgun mic,’” Pekich said. “I was completely fascinated.”

With more than three decades in television audio and a background spanning music recording, studio television and large-scale remote productions, Pekich had embedded microphones into just about every imaginable place. But contact microphones in ice were new territory.

His initial concern was physics. With roughly an inch of ice covering each microphone, Pekich expected the sound to be thin, fragile or highly localized. Instead, he found that each mic captured a surprisingly wide area, roughly a 6- to 8-foot diameter, with enough detail to hear skaters working at center ice.

“They’re outstanding,” he said. “One of the best inventions to happen to the sport.”

Historically, figure skating audio relied on a combination of shotguns, PCC-style boundary microphones and occasionally lavaliers embedded in the dasher boards. As overall venue levels increased, those microphones inevitably behaved more like crowd mics, capturing PA and audience noise instead of isolating on-ice action. The in-ice contact microphones changed that equation entirely.

Simplifying the Mix
During the event, Pekich continuously monitored ambient sound levels in the arena, including crowd noise, music playback and general environmental spill, to understand what he was up against. With the contact microphones, much of that fight disappeared.

“I worried whether 10 microphones would be enough,” he admitted. “Then I heard everything.”

Because the microphones only responded to vibration in the ice, fader management became significantly easier. Instead of chasing skaters around the rink or riding levels to avoid crowd surges, Pekich could focus on storytelling. Subtle compression and minimal EQ were all that was required.

“Considering the massive frequency restrictor of the surrounding ice, they performed flawlessly,” he said. “They’re very true to sound.”

Importantly, the sound resonated with those who knew it best. Former skaters serving as on-air analysts commented that the audio matched exactly what they remembered hearing while skating. In some cases, they could even identify blade placement, helping them explain technique and judging decisions to viewers.

The Ice Team Perspective
Embedding equipment into the field of play is always a sensitive issue, particularly in a sport where surface quality directly affects athlete safety. John Monteleone, director of education with the U.S. Ice Rink Association, who oversaw the ice operations, acknowledged the initial hesitation.

“Most ice operations staff shy away from placing any foreign matter in the ice,” Monteleone said. “But after refining the process since 2019, we’re confident in the installation.”

The key, he explained, was precision and consistency. Keeping the cable runs straight, thoroughly removing ice shavings and carefully packing slush over the microphones ensured a uniform surface. Throughout the event, there were no impacts to ice quality, no interference from resurfacing equipment and no contact from skaters’ toe picks.

Removal, however, remained the most time-consuming part of the process. Locating and extracting the microphones after the event could take time and required chipping and hot water. Even so, Monteleone said he would be comfortable deploying the system again.

“My preference is always to keep the field of play as clean as possible,” he said. “But with the experience we’ve gained, I have no reservations about continuing to use in-ice microphones for these events.”

A New Layer of Immersion
Figure skating is a deceptively difficult sport to mix. The action is fluid, the pacing varies and the most important moments are often quiet, such as a subtle shift before a jump or the scrape of an edge setting up a landing. These embedded contact microphones added an entirely new layer of that story.

The on-ice sound translated exceptionally well, earning praise from the production team, leadership and viewers alike.

For audio, innovation doesn’t always mean more microphones or more channels. Sometimes it means listening differently. By installing directly to the ice, these contact microphones offered a rare combination of isolation, authenticity and emotional impact, bringing audiences closer to the sport without ever being seen.

Ever feel like simply keeping up with the mixing demands of live production prevents you from doing your best work?

With so many microphones across the field, court, or set and so many channels to monitor and mix, it can be overwhelming. Adding to the challenge are ever-present budget realities that demand A1s do more with fewer people and tighter timelines.

But that equation is changing. The new Shure DCA901 digital broadcast mic array combines 78 individual microphones to create eight distinct software-steerable channels, an audio DSP that powers a built-in automixer, and control software to ease setup and reduce the burden of managing multiple microphones used to create a conventional mix.

With the DCA901 array, A1s can build their mixes as they truly intend them to be rather than out of concern over the precise positioning of mics, microphone failure or unwanted noise from, for instance, fans in the stands who may drown out the sound of a shoe squeak on court or a player calling for the ball on a football (soccer) pitch.

Ultimate Control

Consider a conventional live audio production for a televised basketball game. An A1 working with assistants would attach shotgun microphones to the stanchions supporting the backboard and point each toward the court, providing a wide pickup pattern. A pair of lavalier mics—one at each net—would also be attached to pick up the ball hitting the rim or the swish of a nothing-but-net free throw—and another shotgun mic at midcourt.

Add into the mix an assortment of on-camera mics, other mics positioned around the arena for crowd noise, and a couple of cameras with shotgun mics attached to capture the sound of those in-your-face under-the-basket shots.

Now consider replacing most, if not all, of that with a DCA901 mounted to each stanchion. With the microphones’ easy-to-use control software, individually steerable coverage lobes are positioned to cover each half of the court. Using the array’s Auto Position feature found in its web-based GUI, an A1 selects a channel and instructs an assistant to make a sound at a desired location. Using that sound, the system automatically aims one lobe at the sound to pick up audio from that location.

The DCA901 offers several key benefits in this scenario. First, there are far fewer possible points of failure—fewer mics and backup mics, less cabling, and fewer wireless channels that might fail. Next, setup time drops dramatically—from hours to as little as 30 minutes. Gone is the need for back-and-forth between an A1 and an assistant, tweaking each mic's orientation. Instead, DCA901 lobe steering is done in software away from the array. Fewer mics, backups, and support gear, along with faster setup, translate into lower production costs.

More Intentional Mixes

Perhaps the greatest benefit of using the DCA901 is that it unlocks the full potential of A1s to mix live productions the way they want, not as they must. The array’s powerful, yet easy-to-use functions and software give A1s the freedom to mix with intention.

For instance, replacing so many conventional microphones with a pair of DCA901s in the basketball scenario means A1s can rely upon the array’s automixer to create a submix of the action on court as it moves from coverage lobe to lobe rather than mixing multiple audio channels coming from stanchion shotgun mics, lavaliers, camera mics, and the rest.

Rather, taking the submix from the DCA901 automixer and a few mics around the arena for crowd noise, an A1 has enough time to focus on the other sound elements that go into the main mix, such as sound effects associated with animated graphics, which might otherwise have been missed.

Of course, these same benefits can enhance audio production by opening up new possibilities for A1s to mix intentionally and achieve their creative vision for other sports, studio shows, and even reality TV.

Having that extra set of virtual hands makes all the difference not only for A1s but also for producers and directors, who expect their shows to look and sound great.

Click here to learn more about the DCA901 and schedule a demo in your environment.

Broadcasters, production companies, and show producers alike know that engaging audiences more deeply in their productions not only makes their work memorable but keeps the public coming back for more.

That’s why there’s been so much attention paid over the past few years to enhancing resolution—taking it to 4K and even 8K UHD—and transitioning to high dynamic range (HDR) pictures offering a much wider color gamut than standard dynamic range. Each helps to put lifelike images—those that pull viewers into content—on screen.

But what about audio production? Equal effort is being given to enhancing viewers’ audio experience as well. A number of technologies have emerged to enhance the audio experience, such as speech enhancement and descriptive audio. But it’s immersive sound that most directly engages audiences in the action by creating a captivating audio experience that’s the perfect match for 4K/8K UHD images and HDR.

Until recently, capturing immersive audio, especially for live productions like sports, has required elaborate, time-consuming mic setups and exceptional mixing diligence to create an immersive sound experience. Not only was a good deal of work needed to precisely position and tweak the mics' aim to pick up the channels required for the immersive mix, but cabling and mounting complexity grew with each new microphone.

A new digital broadcast microphone array from Shure, however, is changing that equation. The Shure DCA901 is equipped with 78 individual microphone elements that form eight lobes, which can be steered remotely in real time to create a sound field that changes as the action moves across the screen. During setup, having steerable lobes means adjusting the pickup from a central computer, not physically at the mic. During production, they mean being able to capture audio as the action unfolds.

For instance, for basketball games, two DCA901 arrays—one mounted to each backboard stanchion—provide A1s with total court coverage and the ability to steer audio pickup to capture dribbles and shoe squeaks as play moves around the court.

Beyond sports, the DCA901 is leaving its mark in live studio production, newscast production, reality TV production, and even after a shoot is finished in post-production.

Bill Oakley, Shure Associate Director of Global Product Development, recently sat down with TVTech contributing editor Phil Kurz to discuss immersive audio and the role the company’s DCA901 is playing in simplifying production of live immersive audio.

Click here to watch the interview.

NEW YORK—Audience complaints have increased as TV soundtracks evolved from the intelligible speech of the analog era to the cinematic-style wide dynamic range digital multichannel productions of today. Addressing this omnipresent issue, the AES Technical Committee for Broadcast and Online Delivery has published the comprehensive Technical Document (TD1009), “Improving Dialogue Intelligibility in Media,” which diagnoses the intelligibility problem and presents solutions.

“Dialogue intelligibility affects today’s audiences,” said Jim Starzynski, NBCUniversal Director and Principal Audio Engineer and Chair of the working group. “Increased reliance on captions has become apparent and indicates a problem. This study reveals why audiences can struggle to understand their favorite shows and offers in-depth solutions.”

Sound for picture delivered through broadcast and streaming is the primary focus of TD1009. Solutions also apply to audio-only media. Important takeaways from the study include detailing the “Dialogue Intelligibility Ecosystem”—dialogue’s journey from acquisition to the ear of the consumer and the application of how humans understand and interpret dialogue. Solutions presented include fostering a culture of dialogue intelligibility in the post-production process; addressing the limitations of TV sets, set top boxes and audio/video receivers; and identifying what can go wrong in distribution.

Production insiders and subject matter experts have collaborated for over two years in the development of TD1009 to guide creative professionals, content distributors, device manufacturers and consumer enthusiasts, AES said. The guide was created under the leadership of the AES Technical Committee for Broadcast and Online Delivery (TC-BOD), an international group that addresses audio quality chaired by David Bialik, Kimio Hamasaki, Matthieu Parmentier and Jim Starzynski.

The publication of TD1009 follows its initial public presentation at the AES Show 2025 – Long Beach Convention, during the session “Improving Dialogue Intelligibility in Media: A Report from the AES Broadcast and Online Delivery Technical Committee.” “With AES TD1009 now publicly released,” notes Rafael Kassier, who co-chairs the AES Technical Council with Lesley Fogle, “we encourage broadcasters, streaming platforms, content creators and audio engineers worldwide to explore and apply its recommendations to enhance speech clarity and overall listener experience.”

This paper is distributed openly by the Audio Engineering Society and may be viewed/downloaded at https://aes.org/community/technical-council/aestd1009/. The TC-BOD can be contacted by email at broadcast@AES.org.

On an NFL game day, the mix isn’t just sound, it’s the heartbeat of the broadcast. The goal is simple: Make viewers feel like they’re standing on the field without overwhelming the action. Neither Daniel Haggard nor Jeremy Katz originally imagined themselves shaping the audio for primetime NFL games, but once they found their way into broadcast audio, both discovered the rare mix of pressure, precision and excitement that keeps them coming back.

Haggard started in the early 2000s in Syracuse, back when the workflow was still mostly analog and troubleshooting was half the job. His big break came in 2015 when veteran mixer Wendel Stevens invited him to join NBC’s “Sunday Night Football,” a leap that meant six months on the road with a brand-new crew.

“It was terrifying and thrilling,” he said. “Only 6 million people watching.”

Seven years as an A2 gave him hands-on experience with RF, comms and field audio. That training made stepping into submixing a natural next move.

Katz’s route looks different. He cut his teeth working with bands, eventually landed at regional sports networks before joining Prime Video’s “Thursday Night Football” four seasons ago. His philosophy is straightforward: “My job is to capture everything happening on the field and hand the A1 something solid and clean. I try to be as aggressive as I can, and the A1 can decide what’s safe for air.”

Katz said the key is chasing clarity, not noise for the sake of noise.

Showing and Telling
Both mixers talk about audio in practical, picture-first terms. If something is happening on screen, the viewer should hear it—simple as that. From the quarterback’s cadence to a clean catch or the thump of a tackle, the goal is to match the energy and detail the cameras are showing. Anything that’s unnecessary, distracting or muddy doesn’t belong in the mix.

One tool they fully agree on is the mic mounted on the SkyCam. Hovering just above the action, it’s positioned closer to the players than almost any other microphone. Katz calls it “the best mic on the field,” and Haggard uses it as a bed of natural sound.

From there, player mics provided by the league give him extra detail on the line. He brings them up after the huddle breaks, rides them through the snap, then pulls them down quickly to avoid smearing the rest of the soundscape.

Parabs are another staple. Both mixers prefer committing to a single parab per play rather than blending multiple at once.

“Quality over quantity,” Katz said. “You get a clearer picture when you’re not stacking too much.” Haggard agreed that picking the right mic and trusting it is part of the craft. Sometimes you guess wrong, but when you guess right, it feels perfect.”

Nowhere is that more true than with the most coveted sound in NFL audio: the “doink.”

Across crews, mixers keep a friendly, season-long competition over who can capture the best goalpost hit. Katz joked that the entire “TNF” truck starts cheering the moment a kick drifts toward the upright.

Haggard remembered one playoff game where everything lined up: the contact mics on the post, the 416s under the crossbar, the tension of the moment and Mike Tirico’s call. “If that mic was down, it would’ve ruined the whole moment,” he said. “Instead, it was perfect. I rode the adrenaline all the way back to the hotel.”

An Art and a Science
Mixing the NFL isn’t easy—you’re dealing with massive crowd surges, stadium PA music, sideline chaos, sweaty linemen, unpredictable ref mics and dozens of camera perspectives that shift direction every few seconds. You’re anticipating where a play is going before it happens. You’re making choices that millions of viewers will react to instantly.

Both mixers emphasize that it’s a creative craft, not just a technical job. You learn by watching others, borrowing ideas, shaping your own style, and adjusting to each game’s personality. “There’s no single right way to mix,” Haggard said. “You pick up techniques, try new things, and keep refining. That’s what makes it fun.”

For viewers at home, the result is seamless: Every snap, tackle and catch lands with clarity and intensity.

Behind the screens, it’s hundreds of subtle decisions: mic choice, fader ride and EQ tweaks that make the difference between a flat, lifeless broadcast and one that truly feels alive.

NEW YORK—The Audio Engineering Society will hold the 2026 AES Show, Oct. 30–Nov. 1, 2026, at the Gaylord Opryland Resort & Convention Center in Nashville, Tenn.

AES Show Nashville will offer attendees and exhibitors new and exciting programming, which will extend beyond the normal scope of previous AES Conventions. From pre-show events and experiences to extensive studio tours and involvement with the local audio community, the new location will open up Nashville’s rich audio legacy to the international community of AES Show attendees.

Nashville has earned its nickname “Music City” by way of a uniquely concentrated ecosystem of world-class studios, live music venues, leading audio education institutions, pro audio manufacturers, engineers, artists and audio professionals. Music creation and production are woven into daily life of Nashville.

The city is home to many world-class recording studios, production and rehearsal facilities, a local network of top engineers and producers as well as several major record labels, music award shows and specials, film and TV scoring studios, live sound venues, broadcast facilities and top-tier professionals pushing the art and science of audio.

AES Show Nashville will also take advantage of the unique offerings of the Gaylord Opryland Resort and Convention Center, located steps from the world-famous Opry House facilities, home to the “Grand Ole Opry” live radio broadcast, the longest-running live radio broadcast in U.S. history. The radio show started in 1925 on WSM Radio. The AES will offer exhibitors and attendees specially priced hotel and registration packages.

More information is available on the AES Show website.

Video-streaming services delivering content to the state of California will soon be required to change the way they handle advertising to alleviate one of viewers’ biggest annoyances: Loud commercials. Thanks to Senate Bill 576, signed into law in October 2025 and taking effect on July 1, 2026, commercials can no longer be noticeably louder than the surrounding program.

Broadcast and cable television providers have been adhering to the Commercial Advertisement Loudness Mitigation Act—better known as the CALM Act —since it was signed into law in 2010 to address this same issue, but it has not applied to streaming services.

The new legislation closes that gap, requiring streaming platforms to follow the same practices outlined in CALM, which itself is based on the recommended practices outlined in ATSC A/85, ensuring that ads are delivered at the same volume as the rest of the program content.

Are You Being Served?
To understand how this impacts streaming services, it helps to look at how ads are actually delivered on these platforms. The two main approaches are client-side ad insertion (CSID) and server-side ad insertion (SSID).

Client-side means the ads are added into the stream by the app or device itself. CSID is flexible and allows for targeted ads, but controlling levels is challenging, as every device handles playback differently.

Server-side, on the other hand, stitches ads into the stream before it ever reaches the viewer. SSID behaves more predictably and is often preferred as it helps counter ad blockers, but it introduces its own challenges since different vendors use different encoding pipelines, which can result in inconsistent loudness.

Streaming providers employ these models in different ways. National broadcasters like NBC, CBS, ABC, and Fox already comply with the CALM Act for their traditional broadcasts, so extending those practices to their streaming apps is a natural next step.

Major streaming services such as Netflix, Hulu, Disney+, and Amazon Prime Video are also affected. Netflix recently introduced an ad-supported tier, Hulu and Disney+ lean heavily on server-side insertion, and Amazon Prime’s hybrid model mixes subscription and ad-supported content.

Local broadcasters face a unique challenge, as they often depend on third-party server-side ad insertion vendors for their streams, meaning they will need to ensure those partners are meeting California’s new requirements.

The biggest impact may be felt by FAST (Free Ad-Supported Television Services) channels like Pluto TV, Tubi, and Freevee. These platforms rely almost entirely on server-side ad insertion, pulling ads from a variety of suppliers, making them especially vulnerable to loudness inconsistencies.

Video-on-demand providers face their own challenges, as many of them use client-side ad insertion that requires normalization at the device level. This approach makes consistent loudness more difficult and requires VOD platforms to ensure loudness metadata is included and preserved so that the playback device can adjust levels accordingly.

Overcoming Technical Challenges
The technical hurdles to comply with this new regulation are significant. Different server-side ad insertion providers use different encoding methods, which means ads can arrive with varying loudness levels. Ads themselves often come from multiple sources, each mastered to different standards.

On top of that, playback devices—from smart TVs to mobile phones—interpret audio differently, which makes it even harder to guarantee a consistent experience. Content created before the legislation was enacted adds another wrinkle, since it may not meet modern loudness standards and would need to be reprocessed.

So how do streaming platforms tackle this? The first step is becoming familiar with ATSC A/85 recommended practices, which guide CALM compliance and, in turn, adherence to the new streaming regulations. Streaming providers will need to integrate file-based and, in some cases, real-time processing and loudness control into their server-side commercial insertion workflow, just as they currently do for their primary programming.

For client-side insertion, ensuring that loudness metadata is included and maintained is a promising option. Measuring loudness at ingest (to determine the need for corrective measures) and downstream (as a final QC step before distribution) using the latest iteration of ITU-R BS.1770 is critically important.

California’s new law is a big step toward making streaming feel more polished and viewer-friendly. For national broadcasters, it’s an extension of practices they already follow. For FAST channels and VOD providers, it’s a wake-up call to tighten up their workflows.

The challenges are real, but the solutions are within reach. By adhering to established loudness measurement, enforcing metadata-driven normalization, and holding vendors accountable, streaming platforms can not only comply with the law but also deliver a smoother, more enjoyable experience for viewers.

LAS VEGAS and NASHVILLE, Tenn.—Shure technology played a critical role during the broadcast productions of the 59th Annual CMA Awards at the Bridgestone Arena in Nashville and the 2025 Latin Grammy Awards at the MGM Grand Garden Arena in Las Vegas.

At the CMA Awards, Shure microphones and wireless systems delivered clear, reliable audio. The country music event’s production team relied heavily on Shure Axient Digital Wireless and PSM 1000 in-ear monitoring systems to deliver pristine sound to the crowd, artists and those tuning into the ABC-produced event around the world.

ATK provided wireless support with 34 channels of Axient Digital and 24 channels of PSM 1000 powering 116 belt packs per side, the company said.

“I still continue to be amazed by the rock-solid performance, operating range and tolerance to interference [of Axient Digital],” James Stoffo, RF coordinator for the CMA Awards, said. “One diversity set of antennas upstage center covered an entire arena from the stage to the back of house. When the transmitters were anywhere in the arena, I had completely pegged RF indication at the receivers. Exactly what you want to see.”

The show featured Shure KSM11 microphones for host duties and standout performances, including Luke Combs, who also paired the KSM11 for his collaboration with BigXthaPlug.

“I have been using the KSM11 for over three years for both speech and performance applications,” said Paul Wittman, production sound mixer for the CMA Awards. “They have a smooth, natural response and are consistent across many voices.”

Other artists relying on Shure Axient Digital and mic capsules included: Old Dominion and Megan Moroney, using a Shure SM58; Chris Stapleton, using a wired SM58; Riley Green, using a Beta 58A; Zach Top, using KSM9; Little Big Town, using the KSM11 and SM58; Ella Langley using the SM58; and Patty Loveless, using the SM58.

Latin Grammy Awards
This year’s Latin Grammy Awards saw Shure play key roles from the opening number, featuring Carlos Santana, to performances by Aitana, Alejandro Sanz and Karol G. Artists and the production team relied upon Shure’s Axient Digital Wireless Microphone System and the Axient Digital PSM In-Ear Monitor System

In addition to the main show, the Latin Recording Academy Person of the Year Awards and Latin Grammy Premiere were both supported entirely by Shure wireless, including Axient Digital Wireless and Axient Digital PSM In-Ear Monitoring Systems.

“The Latin Grammy Awards were a big success,” Univision Technical Manager Michael Karsch said. ”I attribute this to the professional audio crew and the support of Shure. Due to frequency limitations in Las Vegas, our Shure representative loaned us the new Axient Digital PSM system, which enabled us to provide an interference-free monitoring system for the talent. The talent loved the sound as well as the smaller belt pack. This made life easier and delivered a stress-free production.”

Production teams deployed eight channels of Axient Digital PSM for Stage A, while the rest of the show featured the trusted Shure PSM 1000. Wireless podium audio was delivered flawlessly using Shure AD3 transmitters, and announcers Maluma and Roselyn relied on Axient Digital systems paired with Shure KSM11 Cardioid Condenser Vocal Microphone capsules, Shure said.

“We had the opportunity to implement multiple channels of Digital PSM In Ear monitor systems,” said James Stoffo, RF coordinator for the Latin Grammy Awards. “The analog PSM RF spectrum was full of back-to-back DTV so I requested that we add as many Digital PSM channels as we could get our hands on. Everything went off splendidly and the monitor technicians were happy. The wide RF tuning range and audio quality, even in low RF signal conditions, makes the RF tech’s job so much easier. My intention is to specify the Digital PSMs on all future live productions.”

More information is available on the Shure website.

SAN FRANCISCO & THE COLONY, Texas—Dolby Laboratories is making what it is calling a “new chapter” in its retail efforts as part of an agreement with NFM (Nebraska Furniture Mart) to launch the first-ever Dolby Home Experience, at NFM's The Colony location on Saturday, Dec. 6.

Dolby said that the Dolby Home Experience is an immersive retail destination where shoppers can see, hear, and feel how Dolby Vision and Dolby Atmos transform their favorite movies, music, sports, and games.

"The Dolby Home Experience at NFM represents a new era of experience-driven retail," said Ketrina Dunagan, head of Americas marketing at Dolby. "This launch reflects how Dolby is meeting today's consumer through experiences that show, not tell, the difference our technologies make. Together with NFM, we're bringing the power of Dolby Vision and Dolby Atmos to life in a way that's simple, accessible, and built for how people shop today."

Built as a dynamic, interactive environment, guests can explore the ultra-vivid picture of Dolby Vision and the immersive sound of Dolby Atmos in a setting that feels like home guided by Dolby-trained experts and surrounded by the latest TVs, soundbars, and devices from leading brands. From inspiration to hands-on demos, the space helps consumers understand the value of a better entertainment experience, no technical expertise required, the two companies said.

"We are giving our Dallas-Fort Worth customers a new way to explore home entertainment," said Robert Kincheloe, store director at NFM. "The Dolby Home Experience transforms our store into an interactive destination where shoppers can try out the latest products, get hands-on guidance from experts, and see how premium entertainment technology fits into their everyday lives."

Dolby and NFM will host a two-day grand launch event from Saturday, Dec. 6 to Sunday, Dec. 7, with interactive demos, exclusive offers, giveaways, and hands-on opportunities to explore Dolby Vision and Dolby Atmos.

More information is available at the Dolby Home Experience webpage.

AJA Video Systems has joined the RAVENNA community, a network of partners devoted to advancing the RAVENNA standard for distributing real-time audio over internet protocol (AoIP).

AJA’s DANTE-12GAM and OG-DANTE-12GAM audio embedder/disembedder products support RAVENNA. The products make it easy to bridge between SDI and AoIP infrastructures. Both include dual 12G-SDI input and outputs, providing up to 32 channels of audio embedding and 32 channels of disembedding simultaneously for a total I/O capacity of 64 channels.

“As the transition to IP accelerates, interoperability is essential for our customers, many of whom are working with a combination of SDI and IP tools,” AJA President Nick Rashby said. “Joining the RAVENNA community underscores our ongoing commitment to meet customers where they are in their IP journey, broadening AJA product support for the larger AoIP ecosystem. RAVENNA is a powerful AoIP option, and now our customers can take advantage of it.”

Welcoming the company to the RAVENNA community, Lawo RAVENNA Evangelist Andreas Hildebrand said RAVENNA support in select AJA products will “unlock seamless interoperability with a range of AoIP protocols—and that benefits all professionals looking to make the most out of AoIP.”

More information is available on the AJA Video Systems and RAVENNA websites.

NEW BERN, N.C.—Wheatstone has named company veteran Darrin Paley vice president of business accounts, effective immediately.

In his new role, Paley will head global business sales for its Wheatstone, Audioarts Engineering and VoxPro product lines, the company said.

“Darrin brings to this new role his deep understanding of the issues facing our industry today and his experience as a trusted partner to our customers and dealers for more than 25 years," Wheatstone president and founder Gary Snow said.

Paley joined Wheatstone in 2001, initially as a senior sales engineer. Prior to that, he spent six years with family businesses Oakwood Broadcast and Ron Paley Broadcast (now GS Technical Broadcast Services), a specialist in broadcast-facility design, project management and installation, particularly for Wheatstone products, the company said.

Paley will be based from Wheatstone’s New Bern factory and can be reached via email at darrinpaley@wheatstone.com or at 252-638-7000, extension 116.

LOS ANGELES—Major League Soccer will introduce a broadcast audio enhancement tonight during the LAFC vs. Austin FC playoff match.

Partnering with EDGE Sound Research, MLS will debut “Embodied Sound,” a breakthrough technology that uses real-time object tracking to isolate the soccer ball as a unique audio source, the company said.

The game will mark the first-ever use of isolated ball audio in a live soccer broadcast, enhancing the control and clarity audio engineers will be able to achieve. By routing the ball’s sound into a dedicated channel, the broadcast mix will deliver more precise and immersive effects and capture the energy of each kick, post hit and player reaction without overwhelming ambient noise, it said.

The system provides mixers with a clean, controllable ball channel that will enable them to emphasize key moments in real time without affecting other ambient sounds, it said.

EDGE Sound Research is a member of the MLS Innovation Lab. The company, a U.S. based leader in multimodal audio, is pioneering new forms of multimodal sound to bridge the physical and emotional dimensions of sound.

The MLS Innovation Lab identifies and supports cutting-edge startups driving advancements in fan engagement, on field performance, and media technology.

More information is available on the company’s website.