Palm Beach Aquatic Centre has completed a major redevelopment, reopening as a modern community destination designed to serve swimmers, fitness users, families, and local community groups across the southern Gold Coast. To support the expanded facility, The Prestige Group designed and integrated a venue-wide JBL Professional audio system that provides flexible, multi-zone coverage throughout indoor and outdoor spaces while accommodating the operational demands of a high-traffic aquatic environment.

[The Experience Economy, Revisited]

When the Palm Beach Aquatic Centre reopened in October 2024, it marked far more than a cosmetic upgrade. Apart from the existing 164-foot pool, the revamped facility now includes a learn-to-swim pool, hydrotherapy pool, splash pad, gymnasium, group fitness studios, café, crèche and community gathering spaces. With multiple activities occurring simultaneously throughout the day, the City of Gold Coast required an audio system capable of supporting clear communication, background music and zone-specific announcements without disrupting adjacent areas.

The Prestige Group worked alongside MadisonAV to develop a solution that extended well beyond the basic paging system originally specified during the planning phase. Instead, the team implemented a true multi-zone design capable of providing localized control across more than 40 independent audio zones, allowing staff to manage announcements and music according to the operational needs of each area.

To meet the acoustic and environmental requirements of the facility, The Prestige Group selected a range of JBL Professional loudspeakers designed for both indoor and outdoor applications. Twelve JBL CBT 70J column loudspeakers were deployed in large open areas where controlled coverage and speech intelligibility were critical. JBL AWC82 all-weather loudspeakers were installed throughout pool halls and other moisture-exposed environments, while JBL Control Series ceiling and surface-mount loudspeakers were integrated throughout fitness areas, studios and ancillary spaces to provide consistent audio coverage across the site.

The system was designed to support the unique operational requirements of an aquatic facility, where humidity, chlorine exposure and changing occupancy levels can present challenges for conventional audio systems. By combining weather-resistant loudspeakers with a distributed, zone-based architecture, the installation provides reliable performance while allowing the venue to adapt audio content to different activities occurring simultaneously throughout the complex.

“The CBT70s are very focusable, with even dispersion,” said Nathan Wright, project manager from The Prestige Group. “They integrate cleanly with the architecture, deliver strong full frequency performance, and are well suited to the environmental conditions.”

Since reopening, Palm Beach Aquatic Centre has exceeded attendance expectations with strong community engagement across its aquatic, fitness and recreational programs. The audio system has become an important part of the facility’s daily operation, supporting communication, programming and visitor experiences throughout the venue. With its flexible multi-zone JBL Professional installation, the Palm Beach Aquatic Centre is equipped to accommodate continued growth while providing reliable audio coverage across one of the Gold Coast’s most active community facilities.

There’s nothing better than a successful deployment of new AV when everything is running smoothly. New auto-framing audio and video conferencing hardware and software connect in-person and remote participants, and a control system automatically initiates a meeting and starts the presentation on display. Participants collaborate, sharing content and video that is seamlessly switched to different displays. It’s AV/IT nirvana.

Until it’s not. We asked AV/IT industry thought leaders about how their company’s products and solutions help ensure interoperability and connectivity not only on the day of deployment, but also on Day 2 and beyond. Below are some excerpts.

For the full interviews, check out the sidebar on the right.

Omer Brookstein, CEO and co-founder at Xyte, said, “The real goal isn’t simply to connect devices. It’s to make AV easier to manage, easier to support, and easier to improve over time. Open cloud connectivity helps organizations monitor mixed fleets more consistently, move data between systems, automate routine tasks, and adopt new tools without rebuilding everything around a single vendor’s stack.”

“We focus heavily on open standards and network-based solutions, said Emma Eagle, manager of Marketing at Hall Research. “When AV technology works the same way other IT systems do, it becomes easier to deploy, manage, and scale across an entire organization—from small meeting rooms to large conference spaces.

“Established customer deployments may require upgrades to maintain long-term interoperability, as the protocols themselves are often evolving,“ Amy Zhou, director of Sales at Magewell, added. “For example, the popular NDI connectivity technology has changed over the years; it’s now on version 6.3 overall, and media formats within it have gone from NDI High Bandwidth to NDI HX, NDI HX2, and NDI HX3. This makes the upgradability of our products crucial for continued interoperability and compatibility.”

Joel Mulpeter, senior director of Product Marketing at Crestron, said, “Our products always need to work with the network—not against it. When we get deployed in a facility, the network exists, and we’re part of that ecosystem, so we ensure that we’re good citizens on that network.”

“By combining manufacturer-agnostic interoperability, robust management, and scalable design, Audinate’s solutions help ensure connectivity that works without friction, every day, at any scale,” said Jim Kidwell, principal product marketing manager at Audinate.“

“At ViewSonic, we approach product development with the full ecosystem in mind,” said Shane Roma, technical product manager of dvLED and Commercial Displays at ViewSonic. “A unified API structure reduces programming complexity, accelerates deployment, and minimizes support calls across diverse installations.”

Effi Goldstein, President of the HDBaseT Alliance, added, “Our commitment to an open, vendor-agnostic standard empowers businesses with exceptional flexibility. Organizations can mix and match products across multiple brands without risking compatibility issues, supporting business continuity, easier maintenance, and long-term investment protection.

“Sennheiser’s commitment to interoperability—across Dante, open APIs, alliance partnerships, and secure cloud management—helps organizations build AV ecosystems that are connected, scalable, and ready for whatever comes next,” said Jakub Kolacz, manager of Product Management Software at Sennheiser.”

Adam Sowers, senior manager of Business Development at Sony Electronics, added, “As a founding member of Open AV Cloud, we help to drive best practices around open ecosystems and interoperability for IP-based integrations, control, and monitoring that work across environments.”

“By prioritizing interoperability at every level, Shure empowers IT teams to deliver collaboration environments that are consistent, intuitive, and scalable wherever work happens,” added Justin Ramirez, senior director of Software Sales at Shure.

Crest Audio, a brand of Peavey Electronics Corporation, has introduced two new PoE++ powered loudspeakers: the CiP 26 and the CiP 5. Designed for today’s fully networked AV environments, both models combine professional sound quality with streamlined, network-cable installation—reflecting the growing shift toward smarter, more scalable commercial audio systems.

[Looking Back, Moving Forward]

As organizations increasingly adopt networked-based infrastructures, the need for loudspeakers that integrate seamlessly into AVoIP ecosystems continues to grow. The CiP Series answers that demand by delivering both power, control, and audio over a single Ethernet cable using PoE++ (IEEE 802.3bt Type 4 90W), removing the requirement for local mains power or separate amplifier racks. The result is cleaner installations, faster deployment and greater flexibility for integrators and end users alike.

“We are seeing a clear shift toward network-first audio design. Integrators want solutions that reduce infrastructure complexity while maintaining professional performance standards," said Christoph Sesseck, sales manager EMEA for Peavey Commercial Audio. "With these latest additions to our CiP speaker line, we are giving our partners scalable loudspeaker options that combine the latest networking protocols integration, PoE++ simplicity and the sonic character that Crest Audio is known for. It’s about making high-quality commercial audio easier to deploy and easier to manage.”

The CiP 26 is a compact yet high-output two-way loudspeaker designed for applications where full-range performance and musical impact matter. Featuring dual 6.5-inch woofers paired with a 1-inch dynamic tweeter, its bi-amplified architecture delivers clarity and headroom, reaching peak sound pressure levels of 103 dB at one meter.

With a frequency response spanning 65 Hz to 20 kHz and a wide 120-degree horizontal by 75-degree vertical coverage pattern, the CiP 26 is equally at home delivering dynamic foreground music in hospitality venues as it is supporting speech and presentation audio in larger collaboration spaces. A supplied wall-mount bracket simplifies installation, with connectivity coming via a single RJ-45 connector.

Complementing the CiP 26, the CiP 5 offers a compact solution tailored for a more controlled soundstage. Equipped with a premium 5.25-inch woofer and 0.75-inch tweeter in a bi-amplified configuration, it delivers up to 101 dB peak SPL with a more focused 90x60-degree sound dispersion pattern. This makes it ideally suited to boardrooms, classrooms, meeting rooms, boutique retail spaces and distributed background music systems where intelligibility and tonal balance are essential.

Like the CiP 26, the CiP 5 supports Dante and AES67 (sNET) audio networking and PoE++ power over a single Ethernet cable, with optional 24VDC external power available where required. Its compact enclosure and included U-bracket allow for quick, secure installation across a wide range of commercial settings.

Together, every loudspeaker in the CiP range shares the same design philosophy: professional performance delivered through simplified infrastructure. Whether supporting distributed background music or driving higher-energy foreground applications, the CiP Series gives integrators a cohesive, scalable solution that reflects how modern commercial audio systems are designed today—networked, efficient and ready to grow.

DECT NR+ is emerging as a new kind of 5G class wireless standard that brings high reliability and low latency into professional AV—but without relying on traditional cellular infrastructure. For those specifying wireless audio, intercom, and unified communication (UC) systems, it offers a complementary architecture to operator-driven or private cellular 5G and marks the start of an ongoing journey toward non-cellular 5G tailored to professional environments.

Crucially, DECT NR+ does not happen in a vacuum. For more than two decades, classic DECT and DECT Evolution have been quietly doing a huge amount of work in AV and UC: from cordless telephony and conference phones to business headsets and wireless presentation systems. In the professional audio world, wireless microphone systems like Sennheiser SpeechLine Digital Wireless and AVX have shown how DECT-based links can deliver robust, easy-to-deploy speech solutions in corporate, education, and video production workflows. That installed base has shaped expectations around reliability, range, and ease of use—and it provides the real-world experience on which DECT NR+ now builds.

From Classic DECT to DECT NR+

At its core, DECT NR+ is the revolutionary evolution of classic DECT, a new wireless technology standardized by the European Telecommunications Standards Institute (ETSI) as DECT2020 NR and formally recognized by the International Telecommunication Union (ITU) as part of the IMT2020 5G family. Often described as the world’s first noncellular 5G technology, it takes core 5G concepts—ultrareliable low-latency communication, flexible OFDM waveform design, and support for massive device densities—and adapts them to private, user-owned networks. DECT NR+ is designed to operate in frequency bands below 6 GHz, so its propagation characteristics are close to other 5G NR deployments while staying in the lower sweet spot for indoor coverage and penetration. Today, its operation is anchored in an often dedicated, protected 1.9 GHz DECT band that is, in many countries, allocated specifically to DECT and its successors. That gives AV designers a predictable, interference-managed canvas for high-performance wireless audio. While current products focus on this almost globally available 1.9 GHz allocation, the same physical layer is already being evaluated for future site-licensed operation in emerging private network blocks such as CBRS type 3.5 GHz ranges, and the 3.7–3.8 GHz and 3.8–4.2 GHz bands. This extends the concept of user-owned noncellular 5G into higher midbands. Alongside WMAS in the UHF band, DECT NR+ gives us a non-cellular 5G option in 1.9 GHz and, in the future, additional bands—extending the same design philosophy of controlled, low-latency wireless into new parts of the spectrum.

MERCI and Collaborative Innovation

Research initiatives such as the Franco-German MERCI project have helped translate the theoretical capabilities of DECT NR+—like URLLC (Ultra-Reliable Low-Latency Communication) class performance and high device density—into concrete architectures and prototypes for demanding live and broadcast environments. MERCI’s final demonstrations, from industrial guided vehicles to the world’s first MPEG-H object-based live audio transmission over DECT NR+, show that the technology is ready for industrial and demanding audio applications and workflows.

The ecosystem around NR+ is being built as an open, collaborative effort rather than as a closed proprietary stack. The OpenerInitiative.org community has become a focal point for open implementations, tools, and documentation around DECT2020 NR / NR+, lowering the barrier for new entrants and experimental projects. In parallel, the DECT Forum’s Academic Industry Roundtable (AIR) offers a platform where manufacturers, chip vendors, universities, and research institutes can align roadmaps, run joint trials, and feed requirements back into standards. Taken together with the long history of classic DECT and DECT Evolution in telephony, conferencing, and wireless microphone systems like SpeechLine and AVX, these initiatives ensure that NR+ continues that lineage into a 5G-class, non-cellular future for AV and UC.

RF Behavior and Network Architecture

From an RF perspective, the 1.9 GHz anchor band avoids the heavy congestion of 2.4 GHz and 5 GHz WiFi while sidestepping the fragmented ISM bands below 1 GHz. For AV engineers, that means starting from a cleaner spectrum environment where interference from consumer devices is reduced, and coexistence planning is more predictable. The DECT NR+ physical layer uses modern OFDM with adaptive modulation, turbo coding, and optional hybrid ARQ to maintain robust links even in challenging venues like arenas, theaters, and large broadcast compounds—all within that sub 6 GHz window.

Equally important is network topology. DECT NR+ supports point-to-point, star, and self-healing mesh configurations in license-exempt spectrum, with scheduling and routing decisions taken locally at the edge. Unlike cellular 5G, it is not tied to a centralized operator-managed core, so timing and resource allocation can be optimized directly for the venue’s requirements. This makes DECT NR+ behave more like a hardened industrial network than a consumer mobile system, suited to scenarios where latency and reliability must be guaranteed block to block rather than averaged across a wide area.

Cellular 5G, by comparison, is built around operator-run base stations, licensed spectrum, and a centralized core. That architecture excels at wide area mobility, outdoor coverage, and high throughput broadband, and it is increasingly relevant as a backhaul and contribution link technology in broadcast and live production. For many AV and UC projects, however, there is also a need for a fully self-contained, venue-owned RF fabric for the “last mile” between devices and users. DECT NR+ fills that space by providing a modern PHY and MAC that can be rolled out and controlled entirely on-premises, complementing rather than replacing cellular 5G.

What this Means for AV and UC Specifiers

The traffic profile reflects this division of roles. DECT NR+ is optimized for low to medium data rate, small packet traffic such as audio streams, control signals, and sensor telemetry across a single site. That aligns well with professional audio, intercom, and venue-wide UC ecosystems, where deterministic timing and high reliability matter more than peak throughput. Cellular 5G, on the other hand, is optimized first for broadband data, with URLLC and massive IoT capabilities available where needed. In many real-world designs, the result is a layered approach: 5G for wide area connectivity and heavy backhaul, NR+ for dense, low-latency device-level connectivity inside the venue.

[Coming Soon to Pro AV: DECT NR+]

To make NR+ practically usable across vendors and product categories, ETSI TC DECT is developing a family of application-specific profiles for DECT2020 NR, including a Generic Audio Applications profile. Building on earlier DECT audio profiles such as the Advanced Audio Profile, this work defines common behaviors and capabilities for a broad class of audio devices and use cases on top of the NR+ radio layer. The aim is that NR+ based microphones, headsets, loudspeakers, and other endpoints can follow shared expectations for latency, codec handling, and control signaling, giving specifiers more confidence that NR+ labelled solutions will meet a consistent baseline.

For AV and UC specifiers, the technology choice is already becoming a practical line item. A wireless microphone or intercom system that runs on DECT NR+ is not just “another DECT solution”; it is built on an ETSI standardized PHY/MAC engine operating in an often dedicated, protected 1.9 GHz band, with deterministic latency and high device density designed in from the start. That changes RF planning, interference management, and scalability. Instead of manually coordinating dozens of channels under tight constraints, professionals can specify a fabric that can automatically adapt while maintaining predictable performance. In UC, the same fabric can underpin headsets, speakerphones, and room control systems on a dedicated RF layer that does not compete with IT-managed WiFi.

Looking ahead, the ability of DECT NR+ to move beyond 1.9 GHz into site-licensed midbands such as CBRS-type 3.5 GHz allocations and the 3.7–3.8 GHz and 3.8–4.2 GHz ranges now opening for local networks adds another dimension for system designers. The same noncellular 5G technology that underpins license-exempt AV and UC deployments will, in the future, also be able to inhabit coordinated, locally licensed spectrum where extra isolation or capacity is needed—for example, in large campuses, broadcast hubs or critical infrastructure facilities.

In that sense, DECT NR+ is not simply “DECT with a new name,” but a new class of radio access technology that brings 5G-style performance into a spectrum and architecture tuned for professional AV and UC. As this journey continues—driven by collaboration between manufacturers, open technical communities, academic partners, and standardization bodies—DECT NR+ is set to become another proven building block on the AV designer’s shelf: a noncellular 5G foundation that aims to make “wireless as good as wired” a practical design choice in everyday audio and UC projects.

SCN: How long have you been with Peavey, and what are your responsibilities?

Courtland Gray: I’ve been working at Peavey for 24 years. I started in our finance division. I became more involved with various parts of the business as time went on, and have served as COO, CFO, and since our legendary founder, Hartley Peavey, has referred to me as CEO in a few emails to people, I claim that now as well. The responsibilities are broad and wide, covering both our Commercial Audio division and the musical instrument side of the business.

SCN: Last year, Peavey celebrated its 60th anniversary. How has the company been celebrating?

CG: From a product viewpoint, we have brought back several classic Peavey guitar amps from over the years and put them into a pedal preamp format, which have been very popular. And after many years of development, we are thrilled to release Scion, the next generation of MediaMatrix. We are immensely proud of our long-term legacy.

In September, we had an anniversary party in Bali for our APAC distributors and customers, which was a great event. We have also brought our 60th anniversary celebrations to trade shows, with both our ISE and InfoComm booths heavily leaning on our U.S. heritage with an old time Western bar theme showing our achievements over the past six decades. Like most established companies, we have had our share of ups and downs, but we continue to learn and grow as we look to the future with exciting new developments on the horizon.

SCN: I know it’s available in some of your mixers, so I have to ask: Are you for or against Auto-Tune?

CG: I’m for it. There is a big difference between the aggressive T-Pain effect of Auto Tune and its use for a tired singer trying to stay in key toward the end of a gig. Our mixers help tweak it just enough to make a pleasant experience for everyone.

SCN: What prompted Peavey to move into the Pro AV market back in the 1990s?

CG: We were approached by a startup company to distribute the technology that is MediaMatrix. At the time, there were some heated debates about doing this, as some saw Peavey as a hardware company and not a software company. But Hartley saw the opportunities this would have for the brand, and we flew people in and out of Meridian to teach them about the future of audio DSP. We were the first to market with this technology, and it proved to be a good decision that continues to give us opportunities for growth and expansion.

SCN: It’s all audio, of course, but what sort of technological overlap is there between your Pro AV products and your instrument/amp products?

CG: Peavey Commercial Audio is, essentially, made up of two halves—Crest Audio develops and manufactures products for the live performance space such as sound reinforcement, mixers, and amplifiers, while our MediaMatrix line focuses on networked audio, matrices, and sound distribution. Ultimately, it’s all about high-quality, reliable sound, so we use our extensive experience in speaker design to produce, for example, our superb Versarray speaker line.

But I will say the divisions have not overlapped as much as I would like. We need tighter integration of technology, but our engineering groups have been mostly apart and have not collaborated ideally. I see opportunity to take some of the most advanced audio technology on the market from MediaMatrix into our other sound reinforcement and MI products. The cost of driving the code has come down dramatically over the years, which could make the very different price points of each division less of an issue today.

SCN: What makes Peavey’s MediaMatrix unique in the industry?

CG: That’s an excellent question which I’m always happy to answer. Quite simply, Peavey was the very first to introduce networked audio protocols to the industry. Even Dante can trace its roots back to MediaMatrix.

SCN: Is CobraNet still an important selling point for your MediaMatrix DSP solutions, or has Dante taken over completely?

CG: We still have some customers running CobraNet and we can still, for a little longer, support them. However, anything new is moving to Dante or soon to sNET, our protocol that is backward compatible with current nWare installs as well as RAVENNA and AES67. The first CobraNet installation was with MediaMatrix.

SCN: What new Pro AV initiatives are we likely to see from your company?

CG: This is an exciting year for us as we launch the latest and greatest addition to our MediaMatrix range. It’s called Scion and is, quite simply, the most advanced and powerful media and control processor platform available. Scion is the third generation MediaMatrix DSP from Peavey and calls upon 30 years of our pro audio engineering excellence in both hardware and software design.

We have MediaMatrix systems operating in venues today that continue to run without issue for more than 20 years. This latest generation is purpose built from the ground up to meet the demands for today’s (and tomorrow’s) IP-networked installations. With proven 24/7/365 reliability in the most demanding and prestigious venues globally, we believe we are delivering the most powerful DSP based media and control system available today.

SCN: From an audio standpoint, what's next for the Pro AV market?

CG: There is little doubt that the future of all AV installations will be on the network. Across the U.S. and Europe, AVoIP solutions are now the norm, with the Middle East and Far East now embracing the technology. India continues to be a booming market for AV as a whole for both legacy and AVoIP technology.

Longer term, I wouldn’t be surprised to see zero cabling in certain projects in the years to come. Bluetooth and 6G will have almost limitless data capability, removing the need for cables altogether. I would expect many organizations to cling to their legacy infrastructure as the new technology is proven for security and reliability, but there will certainly be a lot less cabling as a whole. Similarly to the connection conundrums we have had around DVI, HDMI, and USB-C, it will be great if there was one wireless uniform connection protocol looming to make things easier for everyone, from manufacturers to end users.

Give your audiophiles the gift of a natural listening experience from Audio-Technica. The ATH-R50x professional open-back reference headphones are designed for all manner of critical-listening tasks. The headphones’ drivers are re-engineered to deliver bass that’s powerful for an open-back model, plus an accurate upper midrange to ensure the clarity of vocals and lead instruments without sacrificing overall balance.

Crafted with mixing, mastering, or creating in mind, the ATH-R50x provides precise, reliable reference across stereo, surround, and immersive formats. Whether they are for the professional or home studio, these headphones are the perfect stocking stuffer for any pro audio enthusiast.

SCN Stocking Stuffers is a seasonal series that highlights affordable Pro AV tools for industry professionals on your “nice” list.

If you've attended one of my webinars during the past six years or more, you'll know that I am a staunch advocate of high-quality audio. That sentiment goes beyond the workplace into my home. My top-of-the-line 65-inch OLED is paired with the best surround sound system I could afford.

Hear Me Out

Until three years ago, I had my interviews for articles transcribed by a woman who knew the industry. Not only did I need her to listen intently to sometimes two different people, but she also needed to discern different dialects and pick up on industry jargon. If my interview were 30 minutes long, it would take her 90 minutes to transcribe, and I'd generally receive it a day later. Two years ago, I began using an AI-generated transcription service that turns around my interviews in a few minutes with around 90 percent accuracy. Given I need to review everything, that percentage is acceptable. I'm only paying a few dollars per interview, and the speed is practically real-time. Unless 100 percent accuracy is needed, human transcriptionists are looking for a new career path.

Within the past year, companies offering highly sophisticated AI transcription services have exploded.

The floodgates have opened. In January, Future editors received a plethora of press releases that hit our inboxes announcing new products and services incorporating AI that enhance the benefits for end-users, integrators, and employees. It's an exciting time.

At Your Service

Like many of you, our company has its fair share of meetings. Every Monday at noon, ET, we have an all-hands B2B Tech Group meeting with, at last count, 53 invited guests. Future, the parent company of AV Technology, has an office in New York City, but the majority of our staff is remote, spread across the U.S. and the U.K., and in more time zones than I can keep track of. For the past few years, Google Meet has become our common meeting space. Having worked remotely since 2014, I have to admit, it's nice to see the faces of the people I work with (just not every Monday, please!).

After extensive research, internal testing, and security measures, Future deployed Gemini, part of Google's family of multimodal AI models and an AI assistant for companywide use. Like other AI assistants, Gemini's features are numerous and growing. I was skeptical because I had already tried its transcription feature and found it was not nearly as accurate as the paid service I was using.

In particular, during the past six months, I have noticed that almost every meeting I participate in, whether it's a Teams, Zoom, or Google meeting scheduled by someone outside my organization, soon after the conclusion, I often receive a meeting recap, and sometimes action items.

Much to my surprise, the summaries and action items have been mostly accurate and pretty helpful. The truth is, I can be more engaged and collaborate more effectively when I am not taking notes. That is, if the AI agent is picking up everything that all the participants are saying.

True Story

Several months ago, Future's B2B senior leadership team flew in from various locations and, along with New York City area employees, conducted a series of planning meetings. Each meeting also comprised three to six remote participants. Our NYC conference rooms are outfitted with a display, webcam/soundbar, and a microphone on the table. The meeting I participated in had approximately 10 people seated at a long conference table, with a microphone in the middle. Plain and simple, I couldn't hear everyone, and when more than one person spoke at once, it was a disaster.

It's not surprising that over the course of the 90 minutes, many sidebar conversations took place that weren't germane to the primary goal. Ironically, those were the conversations I could hear more clearly.

Gemini was not being used to capture this meeting, which was a good thing because the small group sitting together at the end of the table was discussing where to go to dinner that evening, and not everyone was invited. The problem was that I could hear that "quiet" sidebar conversation more clearly than some of the important voices during the meeting.

[Agentic AI, Cindy’s Dahlias, and a Cautionary Tale]

Cindy Predicts Agentic AI as the Top Trend for 2026

I have moderated countless webinars and written even more articles about the importance of high-quality audio in classroom and meeting spaces. As if on its own, being able to hear everyone clearly isn't enough to justify investing in high-quality audio; I firmly believe agentic AI will be the new driver.

Microsoft Copilot, Zoom AI Companion, and Google Gemini are very sophisticated AI-powered tools that have many agentic features. Some of these features have quickly become active participants, and one could argue that they are the most productive participants in a meeting.

High-quality audio in meeting spaces is no longer just about delivering "meeting equity" for the hybrid workforce and remote students. Even if there's no remote participant, using the agentic AI features in Microsoft Copilot, Zoom AI Companion, and Google Gemini can transcribe and summarize meetings and suggest action items. Think about all the time each person spends (or doesn't bother) on figuring out the salient points from a meeting and what action they need to take. More to the point, did they really listen and remember what action is required? For individuals who weren't able to attend the meeting, a summary with action items can be sent.

I believe the use of agentic AI in meetings in 2026 is going to grow exponentially. And I can attest to how much time it has saved me and how it has reminded me of the key points and actions needed.

The reality, and the real point of what has turned into a rant, is that if you don't invest in high-quality audio for your meeting spaces and classrooms, you are making a statement that you don't want to enable people to be more productive. And we all know that CFOs view productivity as the real measure of success.

In 2026, agentic AI will be the most significant driver of finally achieving buy-in for high-quality audio integrated into meeting spaces and classrooms. That's my prediction, and I am sticking to it.

There’s a new wireless transmission standard coming to the audio market: DECT NR+. Short for Digital Enhanced Cordless Telecommunications New Radio+, DECT NR+ is poised to deliver more reliable wireless performance in challenging audio environments.

When DECT NR+ starts to be implemented in pro audio gear, AV technicians and producers will see significant performance improvements in wireless microphones, in-ear monitoring systems, intercom solutions, and professional headsets, among others. In other words, DECT NR+ is set to redefine wireless audio.

The DECT NR+ project has been spearheaded by Sennheiser and the MERCI Project. Dr. Andreas Wilzeck is Sennheiser’s head of spectrum policy and standards, as well as head of the MERCI Consortium that is driving DECT NR+.

“I have been deeply involved in the field of DECT standardization for over a decade, beginning my journey in 2014,” he told SCN. “The MERCI [Media and Event Production via Resilient Communication on IoT Infrastructure] Project is a Franco-German, government co-funded project focused on developing cutting-edge solutions for private 5G networks, particularly those powered by or building upon DECT-2020 NR [marketed by the DECT Forum as DECT NR+] technology.”

Telephone Roots

If the acronym DECT seems familiar, that’s because it is—at least to people who used home cordless telephones a few decades ago. "DECT was initially developed and introduced for cordless telephones,” said Wilzeck. “The first ETSI [European Telecommunications Standards Institute] standard was published in 1992 to enable digital cordless telephony, delivering much higher voice quality and security compared to older analog systems.”

DECT was always envisioned as a flexible digital wireless telecommunication standard that could be used for much more than cordless telephones. In fact, while cordless phones became the most recognizable application, Wilzeck said the original DECT standard was drafted to also support enterprise systems, wireless microphones, baby monitors, industrial sensors, and even public access networks. “In practice, the overwhelming early market success was in home and office phones," he noted, "but DECT’s design always allowed for, and increasingly delivers, many more applications in digital communications.”

Operating in the license-free 1880-1900 MHz range in most countries, the original DECT wireless standard still offers several advantages to audio professionals. They include interference-free operation, good range and building penetration, stable and acceptable latency (for many applications), and secure communications.

Because of its technical strengths and suitability for professional use, DECT has gained a "strong foothold in the digital audio transmission market," Wilzeck said. “Its success is attributed to a unique combination of ease of use, audio quality, low and fixed latency, reliability, and security, making it the ideal choice for demanding audio scenarios.”

Case in point: Cordless headsets in call centers benefit from DECT’s ability to support high device densities, ensuring clear and interference-free conversations in crowded environments. “For audio-for-video applications, solutions like Sennheiser AVX deliver reliable, high-quality audio essential for professional video productions,” Wilzeck added. “Meanwhile, Sennheiser SpeechLine provides a user-friendly and dependable option for corporate and educational settings, simplifying wireless microphone deployment and operation.

"Additionally, DECT technology is widely deployed in intercom systems for live events and broadcast productions, where reliable, low-latency audio transmission and straightforward setup are critical for seamless communication. These advantages have made DECT the preferred wireless solution for mission-critical, professional audio and communication needs across industries.”

What's Different?

DECT is clearly good for wireless audio communications. But DECT NR+ is better.

The reason: DECT NR+ has been specifically designed to meet the ITU’s IMT-2020 (5G) requirements for Ultra-Reliable Low Latency Communication (URLLC) and massive Machine Type Communication (mMTC).

“DECT NR+ is a complete reimagining of DECT, offering a highly flexible wireless technology foundation that can be easily adopted across diverse industry ecosystems,” Wilzeck explained. “It operates below 6 GHz, including the globally available 1.9 GHz bands, and supports reliable, scalable deployments.”

When compared to the original DECT standard, DECT NR+ provides better multipath operation, improved radio sensitivity, enhanced resistance to interference, more efficient bandwidth utilization, low latency, high reliability, scalability, and automatic interference handling. DECT NR+ also supports various network configurations, including tar topology, mesh networking, and point-to-point cable replacement.

“At its core, DECT NR+ delivers a state-of-the-art PHY and MAC layer engine built on OFDM modulation, combined with Time Division Duplex [TDD] and both Time and Frequency Division Multiple Access [TDMA/FDMA],” said Wilzeck. “These technical advances provide enhanced interference resilience, robust multipath operation, and facilitate a broad range of deployment topologies—from simple cable replacements and star networks to advanced, self-organizing mesh networks. With support for both unscheduled and scheduled services, automatic frequency management, and high security, DECT NR+ is engineered to address a wide spectrum of industrial and professional communication needs, well beyond the capabilities of classic DECT.”

Next-Level Benefits

As previously explained, the common benefits of DECT technologies include user-friendly operation, true mobility, automatic frequency and interference management, and support for high connection densities. Thanks to its advanced foundation, DECT NR+ elevates these benefits to an entirely new level.

“By introducing modern radio technologies such as OFDM, adaptive modulation, turbo coding, and self-organizing network architectures, DECT NR+ dramatically improves latency, reliability, and data throughput compared to previous generations,” Wilzeck explained. “Ultimately, DECT NR+ combines the trusted ease, mobility, and flexibility of classic DECT with the performance, scalability, and resilience demanded by next-generation digital wireless communication. What makes it truly remarkable is that DECT NR+ is an independent, openly standardized, and user-deployed 5G technology.”

Currently, DECT NR+ is making its way out of the lab and into the design of future audio products, so there's no point asking Google for DECT NR+ audio products that you can buy today. (We know; we checked.) So, how will it be deployed? Wilzeck has some ideas.

“DECT NR+ is expected to be adopted by a wide range of industries and user groups that require reliable, scalable, and user-controlled wireless connectivity, especially where low latency, high density, and autonomy are critical,” he said. “With its advanced features and flexible architecture, DECT NR+ will not only reimagine classic DECT solutions, but also expand into applications far beyond the reach of previous generations. From industrial automation and smart buildings to healthcare, live event production, and smart cities, DECT NR+ delivers the flexibility and future-readiness needed for the next wave of digital transformation."

According to Wilzeck, DECT NR+ emerged in market-ready solutions between late 2024 and 2025, initially targeting professional and industrial technology sectors. As for audio applications, especially in professional wireless and event technology, Wilzeck said these are just beginning to accelerate. Successful trial demonstrations have resulted in fresh industry interest. "This phased market rollout highlights both the versatility of DECT NR+ and the varying readiness levels across industries," he concluded, "shaping a staggered adoption curve that responds to the unique needs and innovation cycles of each sector."

At the minimum, a DI box converts unbalanced high-impedance signals. At the maximum, a powerfully versatile DI network audio interface goes beyond the box and maintains signal integrity while improving the utility/efficiency of the pro audio chain. Crest Audio’s compact Stage 2x2D is a feature-rich DI box that’s PoE capable with an AD/DA interface for virtually any performance challenge. Designed with future-proof network audio streaming connectivity (Dante/AES67) and supporting dynamic microphone, line level, and Hi-Z instrument pre-amps, the Stage 2x2D delivers audio signals without transient audio loss or acoustic artifacts. On large network audio installations such as hotels, casinos, performance arts and convention centers, the Crest Audio 2x2D provides solid performance, but it’s equally at home touring or in a recording studio. Powered via network switch or available PoE injector, it can be mounted in a 1 RU single or dual rack mount or standalone with its protective rubber casing and feet.

SCN Stocking Stuffers is a seasonal series that highlights affordable Pro AV tools for industry professionals on your “nice” list.

If you want to fill someone’s stocking with a multi-tool, you might want to skip the pliers/knife/file combo and consider the Sennheiser Profile Wireless. I gave this system a serious test drive earlier this year. It’s basically a rectangle of audio goodness (see image left). The charging bar houses two wireless transmitters and a transmitter that connects directly to your phone via USB-C or Lightning (for older iPhones). Both transmitters have integrated mics, but you can also use the locking 3.5mm connector to add a lavalier for a more traditional on-screen look. Pop on the included windscreen on the charging bar and those stored mics become a handheld mic (see image right). Attach the charging bar to a small tripod (it has a 1/4-inch thread) and now you have a tabletop mic. It might not be a great multi-tool for home improvement projects, but the Sennheiser Profile Wireless delivers plenty of flexibility for video projects.

SCN Stocking Stuffers is a seasonal series that highlights affordable Pro AV tools for industry professionals on your “nice” list.

d&b audiotechnik is set to introduce the CCLi Compact Cardioid Line Array, designed specifically for installed applications. Building on the success of the CCL system, which was launched in January 2025, CCLi offers the same technical advantages and acoustical properties, while adding distinctive features tailored for fixed installations.

[No Clean Separation for AV and IT]

The CCLi system provides exceptional performance, compact dimensions (209x593x355mm) and a sleek design, making it ideal for system integrators, consultants and venue owners who are looking for an innovative, powerful and subtle installation solution. It offers broadband directivity control (down to 60 Hz) with low-frequency cardioid behavior to focus acoustic energy onto the intended audience areas. This reduces unwanted low-frequency reflections from the side and rear walls, minimizing reverberation in acoustically challenging spaces.

CCLi delivers extended high-frequency headroom and minimized distortion to ensure greater perceived intelligibility. In terms of efficiency and scalability, the CCLi excels thanks to its single-channel amplifier operation. CCLi also offers features that cater to the needs of the installation market: an unobtrusive design with a new front grille integrated with less visible rigging elements, side panels that replace the CCL handles with flushed side grills, and an optional Phoenix Euroblock connector panel with sealed cable gland.

CCLi is a versatile solution for installed applications, where the accuracy and extent of directivity control, power density, elegant design and cost-effectiveness are key, including multi-purpose venues, theatres, houses of worship, clubs and more.

“The new CCLi system combines a dedicated install-focused design with advanced d&b technology to provide superior acoustical performance for a wide range of Installation projects, ensuring that every word and note is delivered with great precision and power,“ said Santiago Alcalá, product manager loudspeakers, d&b audiotechnik. “Another major benefit of CCLi is its budget-friendliness, which is further enhanced when the systems are combined with the new Bi12 ground subwoofers and the recently introduced 25D amplifiers. This makes CCLi more accessible for many smaller venues and spaces.”

The CCLi system is available in 80 and 120-degree horizontal dispersion models (CCLi8 and CCLi12). As with other d&b loudspeaker systems, the system package also includes a range of matching subwoofers, such as a dedicated flying subwoofer (CCLi-SUB), the installation version of the CCL-SUB. The new Bi12 multipurpose passive cardioid ground subwoofer is introduced specifically for installations. A mobile ground subwoofer version (B12) is also available for the CCL system.

STAGETEC has introduced NEX:T axio, a solution designed to bring intelligence, flexibility, and scalability to the modern production workflows. NEX:T axio integrates premium audio conversion, onboard processing, logic control, embedded STAGENET system, and network management—all within a compact, modular form factor. Available in 8, 16, and 32-channel configurations, it empowers production teams to deploy high-performance audio wherever it’s needed most—in the studio, in the field, or on stage—without compromising quality or control.

[The Power of Predictable Audio Performance]

NEX:T axio enables a smooth, step-by-step integration to AoIP I/O Networks. Existing systems, whether STAGETEC or third-party, integrate effortlessly, preserving workflows and investments. With intuitive web-based control and a familiar STAGETEC GUI, users gain centralized management, full system visibility, and maximum future readiness.

The NEX:T axio was designed for organizations modernizing their audio infrastructure, especially where there is a mix of legacy and IP audio, and where flexibility, scalability, and hybrid workflows matter. It ticks many of the “modern audio network” boxes: multiple formats, IP compatibility, centralized control, and scalable architecture where you want to invest in something that can grow rather than rebuild.

Built in embedded STAGENET browser, the NEX:T axio delivers instant, unified control through the same intuitive, browser-based interface used across all STAGETEC products. No applications, multiple GUIs, or complex setup—just seamless device discovery, routing, monitoring, and GPIO management in one system. Users can save on costs and scale precisely with the right size (8/16/32) while gaining processing and logic in one device and potentially reducing the amounts of additional devices required.  And NEX: T axio is fully compatible with NEXUS and AVATUS, while opening the path to future hybrid IP workflows. 

From broadcast studios to live stages and remote setups, NEX:T axio delivers enhanced sound quality and seamless adaptability across any workflow. The mixture of legacy plus IP formats in one box is a strong feature for organizations that are not ready to rip and replace. Supported formats include analog, AES3, MADI, and IP-audio protocols such as Dante, AES67, and ST 2110-30.

The four-stage operation of TrueMatch converters removes the necessity for analog preamplification, effectively eliminating analog overload and component noise for a pristine audio experience, achieving a previously unobtainable, unmatched dynamic range of 164dB. With integrated four-way microphone audio split and integrated processing and logic providing onboard DSP and GPIO capabilities, NEX:T axio goes beyond a standard I/O box, enabling local signal processing, routing, and automation directly within the unit for smarter, more efficient workflows.

Crest Audio has expanded the capabilities of its latest CPL+ loudspeaker series with full support for EASE (Enhanced Acoustic Simulator for Engineers) and CLF2 (Common Loudspeaker Format) data integration. The move gives AV consultants, system designers, and integrators direct access to accurate, manufacturer-verified loudspeaker data for use in predictive acoustic modelling, coverage mapping, and design validation—essential steps in today’s data-driven installation workflow.

[Blueprint for Success: No Clean Separation for AV and IT]

“EASE and CLF2 integration ensures that CPL+ is not just powerful in performance, but also predictable in design,” said Christoph Sesseck, sales manager for Peavey Commercial Audio (EMEA). “We know that consultants and system designers depend on credible data. Now, they can drop CPL+ straight into their modelling software and trust that what they see in simulation is what they’ll get in the room.”

The CPL+ range combines tour-grade components, cabinetry, and refined voicing to deliver high output, clarity, and reliability in demanding environments. Designed for both fixed installation and mobile reinforcement, the series includes a family of full-range and subwoofer models, offering coverage and configuration options for a wide variety of applications.

Now EASE and CLF2-enabled, each CPL+ loudspeaker now ships with comprehensive GLL and CF2 data files, allowing users to perform detailed design simulations within EASE 5, EASE 4, and CLF Viewer, as well as most CLF compatible simulation software, providing 3D balloon and polar plots for precise directivity analysis; frequency response and phase plots validated to industry standards; comparative overlays to match CPL+ models against other manufacturers; and accurate SPL coverage and STI mapping when deployed in EASE-modelled venues.

Bose Professional has unveiled its DesignMax Luna DML88P pendant loudspeaker system. Luna was designed to deliver high-fidelity audio for open-ceiling environments such as retail stores, restaurants, breweries, fitness studios, and similar design-conscious spaces.

[An Unsung Microphone Hero]

Luna was crafted with Bose Professional’s proprietary Ring Array configuration, which combines eight precisely positioned 2-inch mid-high drivers and an integrated 8-inch woofer to create a smooth, articulate listening experience that fills the coverage area without hot spots or dead zones. This innovative design achieves exceptional clarity and balance down to 40 Hz, which eliminates the need for external subwoofers even in open-ceiling spaces. With a maximum SPL of 111 dB and consistent 140-degree coverage, Luna delivers powerful, immersive sound that requires fewer speakers to cover large areas—preserving both aesthetics and budget.

“DesignMax Luna represents a bold leap forward in pendant loudspeaker technology,” said John Maier, CEO of Bose Professional. “We’ve reimagined what’s possible in architectural audio, delivering a product that meets the highest standards of sonic performance and visual elegance. Luna is a testament to our commitment to technical leadership and addressing our customers’ evolving needs.”

Luna was specifically engineered to address installation issues commonly found in challenging open-ceiling spaces and adapt to constraints while remaining perfectly balanced. Luna supports both single and two-point suspension, with a streamlined profile and concealed wiring that provides a clean look from every angle.

“Luna is more than another pendant loudspeaker—it sets a new standard for performance in a unique, self-contained solution,” added Shawn Watts, VP of product at Bose Professional. “Luna delivers space-filling, crystal-clear sound and deep bass from a single pendant form factor. Designed to replace the traditional pendant-and-subwoofer setup, Luna is easy to spec, easy to hang, and made to blend in visually—letting the sound take center stage. We’re proud to bring this groundbreaking product to our global customers.”

ADI introduced Episode Pro Audio’s Episode Business Music Solution (EBMS), designed to help customers continue growing and succeeding in the commercial market. EBMS is a streamlined, all-in-one offering for small-to-medium business installations, featuring a suite of interoperable professional audio products that give integrators a complete, scalable, and easy-to-deploy sound system purpose-built for commercial environments.

Until now, integrators have often had to piecemeal commercial audio systems using a mix of third-party components, leading to systems with potential for compatibility issues, and limited service visibility. EBMS solves this challenge by delivering a comprehensive, tightly integrated lineup of professional-grade DSPs, amplifiers, mixers, and accessory wall plates — all engineered to work together seamlessly and with the ADI ecosystem, including Control4, OvrC, and RYFF streaming.

“There’s long been a gap in the market for a unified, high-performance commercial audio solution that’s easy for integrators to install, manage, and expand,” said Michael Buratto, senior director product management, at ADI. “With EBMS we’re empowering our customers to capitalize on a huge opportunity by making it easier to design, deploy, and support professional-grade audio systems while leveraging familiar ADI tools and platforms.”

From fast-casual and high-end restaurants to gyms, salons, and more, EBMS components are designed for demanding real-world applications. Whether a project calls for ambient background music, zone-based audio distribution, or paging and announcements, Episode Pro Audio, the new sub-brand introduced with this launch, delivers flexibility and reliability at scale.

Integrators can select from a range of components in the EBMS line, including zone mixer / power amplifier combination units, dedicated power amplifiers, front-end processors, and POE wall plate controllers.

EBMS is a key piece of ADI’s broader “Better Together” ecosystem strategy, enabling integrators to pair Episode Pro Audio with a growing portfolio of complementary products from Araknis, Luma, Binary, and Control4, and more. This unified approach enhances performance and streamlines system design, giving professionals the tools to deliver complete, high-value solutions to commercial end-users with confidence.

Additionally, with OvrC-enabled support, integrators are able to remotely monitor and troubleshoot any issues without having to roll a truck, saving them and the end-user valuable time and resources.

“EBMS is more than just a new product line,” concluded Buratto. “By offering a complete, high-performance solution that’s easy to specify and fully backed by our ADI platforms, we’re addressing an obstacle for our customers by making it easier than ever to pursue new opportunities, win more bids, and deliver exceptional results.”

QSC, a leader in professional audio, video, and control solutions, has announced a strategic decision to divest its DCS (Digital Cinema Speaker) product line to Moving iMage Technologies (MiT). This decision underscores QSC's long-standing commitment to expanding its Q-SYS networked device portfolio, a foundational component of its Full Stack AV Platform.

About the Transaction

As of October 31, 2025, MiT has assumed full responsibility for the manufacturing, distribution, and servicing of all DCS products. The deal includes the entire Digital Cinema Series (DCS) loudspeaker product line, including all products sold under it. This specifically covers the Screen Channel ("SC"), Subwoofer ("SB"), Surround ("SR"), and Screen Channel ("RSC") product lines.

Moving iMage Technologies (MiT), a publicly traded company and a prominent provider of technology, products, and services to the Motion Picture Exhibition industry, is headquartered in Fountain Valley, California. MiT was founded in 2003 and has a history of collaboration with QSC, having acquired QSC's USL Accessibility product line in 2022. “It feels nice to be passing this legacy onto a company with whom we have a strong history, and we trust will continue to do great things with this line,” Heyn said.

“As a longstanding focus to grow Q-SYS, the DCS loudspeaker product line has become a non-core part of our long-term growth strategy,” Heyn added. “However, the DCS product line’s strong legacy and customer base made it a natural fit for a focused cinema provider like MiT.” This transaction allows MiT to take the lead in supporting and growing the DCS product line, ensuring continuity of service for existing customers, while QSC simplifies operations and focuses on its core businesses.

A Laser Focus on Full-Stack AV

While divesting the DCS loudspeaker line, QSC assures its customers that the Q-SYS Full Stack AV Platform is primed for the cinema market. “Cineplexes operators are always looking to create seamless, connected experiences across their multiple space types— lobbies & concessions, hospitality zones, rental meeting spaces, arcades & immersive attractions, and more.

Heyn explained the Q-SYS customer value and key differentiator through the lens of Place, Space, and Applications. “Traditional AV providers tend to operate only at the Application level—offering point solutions that were not designed to connect those Applications across an entire Space, much less bridge the data-driven insights necessary to deliver a truly holistic experience for a given Place like a cineplex.”

The Q-SYS Full Stack AV Platform, however, was designed to unify Spaces and scale across diverse Places—whether high-performance workplaces, higher education environments, or entertainment destinations.

QSC will support this platform-focused vision, in part, through its robust portfolio of loudspeakers, including NL Series networked loudspeakers, high-performance installation loudspeakers like PL Series, commercial loudspeakers like AcousticDesign Series, and active loudspeaker solutions, including QSC K Class, L Class, and C Class.

In the world of microphones, it’s easy for boundary mics to be overlooked, both literally and figuratively. They're not flashy. They don’t swing from booms or stand tall on stage. Instead, they sit quietly, flush-mounted or discretely placed, doing their job with incredible efficiency.

But to those who understand their magic, boundary mics are indispensable. Whether you're capturing crystal-clear dialogue in a boardroom, micing an altar in a church, or unobtrusively capturing voices on stage in a school play, a good boundary mic can be the secret to exceptional audio.

Also known as a pressure zone microphone (PZM), a boundary microphone deploys a mic capsule mounted just a few millimeters above a rigid surface and facing toward it (not up or out). In practice, a down-facing condenser capsule is mounted above the rigid bottom plate of a mic housing and the housing is positioned flush onto a larger surface, usually a table, wall, or floor.

The concept is rooted in physics: A boundary mic capsule diaphragm responds to variance in pressure (hence "pressure zone”) created between the capsule and the rigid surface in the “boundary” between air movement and no possible air movement. Direct sound waves and early reflections off the rigid surface arrive sufficiently in-phase that the coherent signals are additive, resulting in a 6 dB boost in sensitivity while avoiding the comb filtering and phase interference that can muddy the sound when capturing direct and reflected sounds that arrive out of phase due to transmission path length. Directional boundary microphones are created by rotating the capsule within the boundary space to point outward horizontally.

Benefits and Limitations

Unlike traditional stand-mounted condenser or shotgun mics, boundary mics take advantage of their proximity to a rigid surface to provide clear, natural, and intelligible signals, particularly in environments where conventional mics might struggle. Their compact form factor and discreet appearance make them ideal for installations where visibility is a concern, or where foot traffic would knock over a traditional mic stand in seconds.

There are four key strengths of boundary mics. First, they offer natural sound with extended reach. The placement near a surface minimizes destructive phase interference and captures sources evenly. They also provide wide coverage. Omnidirectional or hemispherical patterns cover a broad area, ideal for group dialogue.

A low profile is another advantage. They blend into surfaces and stay out of sightlines, which is ideal for video capture and permanent installs. Plus, installation is easy. They can be permanently installed in tables, podiums, or ceilings for long-term setups.

These days, there are different boundary mics for different applications. For example, the Audio-Technica ES947c/FM3 is designed to be embedded directly into conference tables. Newer models like the ES964 feature multiple capsules for modern immersive and multi-zone applications. You can even integrate boundary mics into a Dante-based AV network with models like the ATND971a.

Of course, they’re not perfect for every situation. One common issue is mechanical noise. A pen click, finger taps, or coffee cup thuds on the same conference table can transmit directly to the mic capsule. You can mitigate this with good placement and surface isolation, but it’s something to be aware of in environments with fidgety users.

Placement Tips

When it comes to getting the most out of a boundary mic, placement is everything. Surface quality matters. Use hard, flat surfaces (for example, wood, glass, or stone) to reflect sound evenly. Avoid tablecloths or uneven surfaces.

You also want to avoid center placement. Don’t stick a mic dead center on a large table. Instead, space multiple mics equidistant from participants.

Mind the mechanical noise, too. Place felt pads or isolation rings under the mic to reduce transferred vibrations—and keep your mic clear of busy hands.

Finally, use the right pickup pattern. Select cardioid for focused speech pickup or omni for capturing sound evenly throughout the space.

[Unheavenly Acoustics]

Boundary mics may not be glamorous, but they are one of the most practical and efficient tools in the modern audio toolkit. Their ability to capture natural, intelligible sound across a variety of challenging environments makes them a must-have for any serious audio pro.

Whether you’re trying to maintain sightlines in a theater, clean up a conference call, or mic a choir without a forest of stands, boundary mics give you high performance with minimal footprint. And with advancements like Dante connectivity and programmable control, they’re more versatile than ever.

Fulcrum Acoustic has introduced its Driveflex amplifier line and version 1.5 of Fulcrum One software, extending the Fulcrum platform to include power amplification and advanced array functionality. The release gives users a unified workflow for designing, simulating, controlling, and monitoring amplifiers, loudspeakers, and immersive systems within a single environment.

[Dante and the Single Pane of Glass]

Driveflex amplifiers are compact 1RU models that deliver the power and processing required for Fulcrum’s DSP. They are automatically recognized and configured within Fulcrum One software, which loads optimized TQ presets and limiters for the entire loudspeaker catalog. This integration ensures consistent voicing and system protection while reducing setup time and on-site adjustments.

Fulcrum One version 1.5 adds Driveflex amplifier management to the platform. Used together, Fulcrum One 1.5 and the Driveflex amplifier series enable advanced array tools including array optimization, array voicing, and array limiter protection. These functions provide greater control and predictability in system design, helping users achieve uniform performance across a wide range of applications. These advanced capabilities are currently supported in the AHS, FL, and CC loudspeaker series, with additional models to follow.

“Fulcrum One now brings loudspeakers, amplification, DSP, and immersive tools together in one complete platform,” says Stephen Siegel, President of Fulcrum Acoustic. “With the introduction of Driveflex amplifiers and Fulcrum One 1.5, we have created a unified system that simplifies every stage of design and deployment. This integration reflects our focus on practical engineering. We are developing technologies that solve real-world acoustic challenges and giving audio professionals tools to deliver consistent results.”

I've been seeing a lot of AI-based transcription and captioning solutions on Pro AV trade show floors recently, so when viaim offered me the chance to review its new RecDot wireless earbuds with built-in AI recording and transcription, I couldn't say no.

Of course, RecDot is not designed for live programming or events. It's a personal recording and transcription service that can be used to document client calls or team videoconferences. RecDot can even double as your own personal dictation device.

The system consists of two earbuds and a charging case. Accessories include an illustrated instruction book, short USB-C charging cable, and four sizes of earbud tips to make sure you get the right fit.

You'll need to install the viaim app on your cell phone and then pair your phone to your RecDot via Bluetooth. Once setup is complete, start a call or a videoconference on your phone, then press the sensor (a small striped area that's easy to miss until you know what you're aiming for) on either earbud to start recording. In case you need to record in a small group setting, you can also record via the charging case by pressing the red button that's front and center.

Press the sensor or red dot again and the recording stops. Once your recording is complete, you can download the audio from your earbuds to your phone, listen to the conversation, and even transcribe it in multiple languages.

By the way, the system alerts everyone else on the call that you've started recording, which is a nice touch and probably a smart legal move when you use it in certain states.

What can I say? It works! The transcription stumbles over some proper names, as you'd expect, but overall it was very accurate and pretty quick. Bonus: Tell the app how many speakers there are on the call and it'll create a transcription that differentiates between them.

The basic transcription plan is free and allows for 600 minutes per month. That's 10 hours of calls. Need more time? You can upgrade to either the Pro or Ultra plan, which require a monthly or annual fee.

There's a lot you can do with viaim app. Beyond personalizing the earbud settings, you can edit your transcripts. And let's not forget the AI, which can summarize meetings and create to-do lists. You can literally choose your AI large language model (LLM) and chatbot model. (OpenAI's GPT-4.1 is the default for both.) Plus, you can export audio and/or text files.

I only have two real issues with RecDot, but they might be substantial for you. First is build quality. These earbuds certainly look nice—they are available in black or silver—but they are very plasticky. Even the charging case feels less sturdy compared to others I own from JBL, Poly, and even budget-friendly TOZO.

The other issue is the sound. From a phone call standpoint, I was told I sounded clear but not crisp using the built-in mics. Also, the actual recordings lacked fidelity from my side (think AM radio), though other participants sounded better. I can live with that; after all, I'm recording notes for personal use, not a podcast for mass distribution.

But even with the available sound settings, it took some work to get the headphones to produce high-quality audio. At first, when I played The Police's "Message in a Bottle," it sounded like the band was in, well, a bottle. However, I was able to improve the output significantly when I shut off my iPhone's EQ, but that creates a new problem when I want to use other headphones. Again, I'm not buying these to play music, but I sure wouldn't mind some microphone and playback refinements.

RecDot is priced at $224 on the viaim website. Keep your audio expectations in check—these will probably not be your favorite earbuds for music. The star of the show here is AI. If you attend way too many remote meetings, RecDot can help you stay organized with recordings, transcripts, summaries, and more. You can even translate meetings for international activities. There's plenty of customization available, too. If your phone feels more like your office than your actual office, RecDot could be the way to help you keep track of what's been said and what's next on your to-do list.

Audio has often been the overlooked partner to video. With the myriad of microphone and speaker solutions available at various price points, technologies, sizes, and distribution capabilities, this is changing. Microphones designed to capture voices from participants seated at movable desks, digitally steerable column loudspeakers to overcome acoustic obstacles, soundbars for small meeting spaces, or in-ceiling systems have changed the collaboration and meeting room landscape. Today, there’s no shortage of solutions tailored to specific use cases.

We reached out to AV/IT industry thought leaders to share their insights and best practices on achieving high-quality audio.

Randall Lee, Vice President of Marketing and Product Management at Bogen Communications

In today’s educational and corporate environments, audio solutions have evolved into powerful, intelligent systems that drive communication, collaboration, and safety. Designing audio for today’s meeting rooms, classrooms, and flexible workspaces should include everything from speaker placement and mic pickup to acoustic treatment, mobility, and system integration.

The best room audio setups start with a thorough understanding of how the space will be used. Microphones may need to capture voices from participants seated at movable desks or tables, while some spaces require sound reinforcement for presenters, particularly in large or acoustically challenging environments. Speaker placement may be fixed or adaptable based on room configuration. Additionally, paging, intercom, and panic button systems are often necessary and may need to connect with other rooms or administrators. These considerations are important when evaluating audio solutions for today’s audiovisual environments.

Beyond traditional analog systems, IP-based audio platforms offer the scalability and flexibility that meeting rooms, classrooms, and flexible workspaces require. Systems like Bogen Communications’ Nyquist deliver clear, consistent audio across a wide range of environments, from classrooms and conference rooms to lobbies and campuses. With support for virtually unlimited zones and endpoints, IP speakers and microphones can be deployed where they’re needed most while remaining centrally manageable through a unified interface.

IP-based solutions are especially powerful because of their ability to serve multiple roles, including reinforcing sound during instruction or presentations, enabling paging and intercom communication, supporting ADA accessibility features, and serving as the backbone for emergency notifications. With built-in support for cloud-enabled services such as email and SMS alerts—and integration with phone systems, access control, and video surveillance—platforms like Nyquist are a key part of a comprehensive safety and communication infrastructure.

Tyler Troutman, Strategic Market Development at Shure

AV and IT teams managing today’s modern meeting spaces are tasked with delivering consistent, high-quality audio in any circumstance. Whether participants are dialing in remotely or joining a meeting in person, high-quality audio has evolved from a nice-to-have to an essential component of productive and engaging meeting experiences. To create these meeting experiences at scale, IT teams need a combination of the right equipment, flexible design, and streamlined workflows.

Taking a room-first approach to setting up spaces is a good starting point. Evaluate the acoustic environment and room size before selecting the technology for the space. By evaluating the room first and paying attention to a room’s acoustic profile, IT professionals can tailor the microphone and speaker placement to the space while minimizing the impact of reflective surfaces, rather than relying on a one-size-fits-all solution. This helps to ensure speech intelligibility and minimize distractions from background noise.

Scalability is another critical component for consideration when setting up today’s meeting rooms. Building systems with interchangeable components that can scale across multiple rooms offers more flexibility and makes ongoing maintenance more manageable. This leads to faster deployment for IT teams and more consistent performance for end-users.

Supporting modern meetings means planning for flexibility. Spaces need to be adaptable to accommodate both in-room and remote participants. In physical spaces, equipment needs to be set up to capture audio clearly for those dialing in remotely. Prioritizing speech clarity helps to ensure everyone is present in a meeting, regardless of where they are located.

By focusing on flexibility, scalability, and providing equipment that meets the needs of the end-user, IT teams can create spaces that deliver high-quality audio experiences without increasing the complexity of their installations.

Jack Horry, Product Manager, Quorum at Ross Video

Good audio has always mattered, but in today’s meeting and learning environments, it has to do more than just sound clear.

At Ross, we’re thinking about how sound systems can work smarter to serve both in-room participants and those joining remotely. Modern audio technologies like ceiling microphone arrays, fixed mic positions, and AI-driven speaker identification are transforming spaces. This enables automation, reduces operator burden, and creates seamless experiences. That’s how we’ve approached Quorum. Because people speak from known positions, we can assist with switching and audio handling without needing deep operator knowledge.

AI and automation often trigger Orwellian anxieties, but we must consider the tools they provide. A moderator in a large discussion can instantly see who said what and when. Real-time captioning and speech-to-text have largely replaced stenographers—not because the role was unnecessary, but because the original goals of speed, accuracy, and accessibility are now better met through shared, real-time data. It’s not about replacing people; it’s about empowering everyone in the conversation with immediate access to what’s being said.

The more metadata your system captures—from speaker ID to location to timestamp—the more valuable your content becomes. It can be searched, archived, translated, or captioned in real time, improving inclusion and comprehension for everyone. Historically, this meant removing variables—fixing seating, and tuning rooms. That still helps, but modern systems can adapt in real time, offering a smarter path forward.

Ross is designing with this future in mind. You’re not just investing in sound quality; you’re future-proofing your environment. We believe the best systems don’t just hear the room; they understand it.

David Missall, Insights Manager, Consultants and Technical Application Engineer Manager at Sennheiser

Even with crystal-clear video, the past five years have underscored how critical distinct, reliable audio is for effective communication. Whether in a classroom, meeting space, or hybrid environment, poor audio—dropouts, echo, or unclear speech—can quickly derail engagement and productivity. But great audio doesn’t happen by accident; it requires upfront planning and coordination between AV professionals, IT departments, and architectural teams.

Effective audio planning starts with understanding how a room will be used. A two-person huddle space has vastly different needs compared to a 50-seat lecture hall, and this context informs decisions around integrating soundbars, ceiling array mics, or voice lift systems. Acoustic treatment is just as essential, as hard surfaces create reverberation, reducing intelligibility. Carpets, curtains, and acoustic panels help maintain clarity, especially when paired with a layout designed for optimal listening zones.

Additionally, voice lift technology is increasingly being adopted in meeting spaces to improve local speech intelligibility. By reinforcing voices just enough to make conversation feel natural and effortless, voice lift can reduce fatigue and support inclusive communication in large meeting spaces.

Equally important is device interoperability. Today’s AV ecosystems feature equipment from different manufacturers, and they all must work together seamlessly. Open APIs and brand-agnostic solutions support long-term flexibility are making it easier to evolve with user needs.

Platform certification is another critical piece. Sennheiser’s solutions are certified for Microsoft Teams and Zoom, ensuring dependable performance and a consistent user experience.

For those looking to streamline the planning efforts, Sennheiser’s free, browser-based Room Planner tool allows AV teams to map out room layouts and identify the right microphone and speaker configurations for each space.

Michael Bridwell, Vice President of Commercial Audio Sales at AtlasIED

Speech intelligibility is the foundation of effective communication in meeting rooms, classrooms, and corporate or higher education spaces. If participants can’t clearly hear and understand what’s being said, the purpose of the gathering is lost—whether in person, virtual, or hybrid.

Modern architectural trends present unique acoustic challenges, particularly with highly reflective surfaces such as glass, marble, and stone. In these environments, simply increasing volume isn’t enough; it’s about precision. That’s where beam steering technology comes in.

Digitally steerable column loudspeakers, such as AtlasIED’s Aimline ALX-D Series, are specifically designed to overcome acoustic obstacles by directing sound only where it’s needed—on the audience—while avoiding ceilings, floors, and other reflective surfaces. Each transducer in the ALX-D Series has its own DSP and amplifier channel, enabling precise, software-based sound targeting. This results in significantly improved direct-to-reverberant ratio and enhanced speech clarity.

The beauty of beam steering is that it eliminates the need for mechanical tilting or repositioning of loudspeakers. This means integrators can maintain clean architectural lines by wall-mounting or recessing loudspeakers, while still achieving pinpoint coverage. For reverberant or acoustically complex rooms, beam steering often isn’t just a best practice—it’s the only viable solution for delivering intelligible audio.

By embracing digitally steerable loudspeaker technology, AV integrators and facility managers can transform meeting and learning spaces into environments where communication is clear, engagement is higher, and outcomes are stronger.

Joel Mulpeter, Senior Director, Product Marketing at Crestron

The industry is moving in a very clear direction when it comes to audio-over-IP.

Commercial audio is a discipline in itself. Customers expect audio signals to be available on the network, where they can be routed and processed between audio-over-IP endpoints and controlled and configured remotely. We took that trend as an opportunity to not only develop video products with audio support, but also create an entire portfolio of comprehensive, flexible, and scalable network audio products that make deploying and managing large-scale audio systems as simple as possible.

The move to network audio from the old cabling of the past is a huge benefit, especially for commercial deployments. Network-based audio systems allow for the best scalability and the most flexibility in routing, processing, and distribution. Audio-over-IP solutions deliver sound at near-zero latency. They’re often interoperable with audio industry standards, easy to install, can support enterprise-grade network security standards, and can offer a wide array of streaming service capabilities. They’re often very energy efficient, to boot.

Whether it’s the sound system for a single hotel or company lobby, or a centrally controlled campus audio system, an AoIP ecosystem can make powerful audio systems a reality. There’s a long and growing list of potential use cases: Retail, hospitality, corporate campuses, higher education, training centers, conference rooms, bars, restaurants, and fitness centers can all benefit from audio-over-IP commercial audio systems. As for Crestron, our DM NAX technology audio systems are 100 percent scalable and can therefore cover a wide range of applications while allowing for future growth. And there’s another plus: These systems can be remotely managed and monitored, enabling support teams to troubleshoot and solve problems without putting a truck on the road.

Joe da Silva, Vice President of Marketing at Extron

Places where people gather for classes, meetings, and other events now require audio systems that have a considerable amount of flexibility. Extron solutions are a key part of spaces that are designed or retrofitted for flexibility in terms of audio connectivity and divisibility.

From a signal flow point of view, flexible designs often start with beam-forming microphone arrays such as those available from Extron partners. These microphone arrays easily handle a person speaking in any part of a room, and even actively track them as they move around. As powerful as that is by itself, it’s in the DSP audio processor where the real magic happens.

An Extron DSP audio processor is able to route any input signal to any output, with the capability of utilizing networked audio connectivity. That routing is managed by a matrix mixer that includes essential signal processing such as compression for evening out audio levels, filters for adjusting the tonal characteristics of the audio, and acoustic echo cancellation (AEC) for eliminating the echo that can otherwise come from the far end of a VoIP call.

When large spaces get divided into smaller rooms, or participants’ locations vary from day to day, Extron DSP audio processors support preset recall that quickly and accurately reconfigures the audio routing for each configuration. Preset recall and real-time changes, such as volume, are driven by intuitive Extron touch panel or button control systems.

A variety of Extron speakers get divided into zones, driven by Extron network audio power amplifiers, to cover different areas within the overall space. The audio assigned to each zone is separately managed by the DSP audio processor for optimal sound in any chosen room configuration.

A successfully designed space can handle practically any scenario that comes along, and Extron products are designed with the flexibility needed built right in.

Nancy Knowlton, Chief Executive Officer at Nureva

In today’s corporate and educational environments, design aesthetics often take center stage. Sleek modern furniture, bold color palettes, and open layouts dominate the conversation. Amid visual planning and transformation, one critical element is frequently neglected: acoustics.

Modern meeting rooms, classrooms, and hybrid collaboration spaces are increasingly designed with hard surfaces, exposed ceilings, concrete flooring, minimalist décor, and one or more glass walls. While visually appealing, these materials reflect sound, creating echo, reverberation, and poor speech intelligibility. The result? In-room participants struggle to hear clearly, and remote attendees are left with distorted audio, undermining the very purpose of these spaces: namely, to foster effective communication.

The rise of hybrid working and learning models has only amplified the need to prioritize acoustic design. A space that looks cutting-edge but doesn’t deliver a great listening experience fails to support productivity or engagement. Poor acoustics can lead to fatigue, miscommunication, and disengagement, especially in environments where clarity is crucial.

Fortunately, solutions exist. Carpeted floors, dropped ceilings, acoustic panels, ceiling baffles, sound-absorbing furniture, and strategic room layout can dramatically improve the sound quality without compromising on aesthetics. Advances in acoustic materials now allow for integration into design elements—things like decorative wall treatments that double as sound absorbers and sculptural ceiling clouds that enhance both form and function.

Designing for optimal acoustics should be as fundamental as choosing lighting or furniture. It’s time for architects, designers, and facility managers to give top priority to sound as the key user experience they’re creating. And it’s not just new spaces being designed from scratch that need to be considered; rectifying existing acoustically deficient spaces should be made a priority. After all, a space that sounds as good as it looks is not just beautiful—it’s functional.

Extron introduced its first PoE speaker, the NF 4C LP. This 4-inch ceiling speaker system features scalable power, DSP, and Dante connectivity. Capable of driving up to three external passive speakers from its integrated four channel amplifier, the NF 4C LP is a single‑cable solution for individual zones within larger distributed ceiling speaker systems. Using Extron’s DSP Configurator Pro software, the integrated DSP can be set up independently for each amplifier channel with essential processing per zone. Dante Domain Manager and AES67 support ensure wide compatibility with enterprise configurations and other network audio devices.

[Son of In-Ceiling Speakers]

“The NF 4C LP expands our line of PoE powered, Dante-enabled solutions,” said Casey Hall, chief marketing officer for Extron. “Hot on the heels of our instantly popular NetPA 204 POE, the NF 4C LP PoE powered ceiling speaker system offers an additional design option to our customers.”

The 4-inch (102 mm) full-range driver features a wide and smooth frequency range of 110 Hz to 20 kHz and is engineered with a practical 135-degree conical dispersion. In voice lift applications, the NF 4C LP provides excellent coverage and voice clarity. Available in packages of two, three, or four speakers, the NF 4C LP meets the requirements of complex, decentralized systems with a self-contained PoE powered speaker solution.

DiGiCo has introduced its smallest and newest addition to the family, the Quantum1¹², a single-screen, 12-fader desk that is DiGiCo’s first true flypack mixer. Designed to fit within a rugged Pelican Air 1637 case, the console, when packed, weighs only 50 pounds, qualifying it as a standard checked bag option on most flights. But despite its ultra-compact form factor, the Quantum1¹² is still a giant on features and functionality.

[Premium Sound for Every Seat]

Based on the latest seventh-generation FPGAs, DiGiCo’s newest product offers 80 input channels with 24 aux/subgroup busses, LR/LCR master busses, and a 12x8 matrix, all with full channel processing. Twelve control groups plus two solos are also available.

Sporting a 17-inch, 1000-nit, high-brightness multitouch screen with both a meterbridge and soft quick select buttons for speedy and intuitive operation, the Quantum1¹² is additionally equipped with 18 individual TFT channel displays to provide clear user feedback and interaction. Beneath the main display are a dozen newly-sourced 100mm touch-sensitive faders, selected for their smooth movement and precision control.

The DiGiCo Quantum1¹² naturally incorporates Quantum Range features like Mustard Processing channel strips (20), Spice Rack plugin style native FPGA processing options (6), and Nodal Processing (24). Twelve graphic EQs are further available, as are eight FX Rack slots.

Other standout highlights include Stadius 32-bit ADC and DAC conversion, dual DMI slots, and a host of versatile I/O options, including Dante (64 channels in and 64 channels out at both 48kHz and 96kHz), MADI (redundant or single), optional optics, AES, local connections (16 analog inputs and eight outputs), and more.

“Our new Quantum1¹² is a brilliant illustration of the idiom, ‘Good things come in small packages,’” said DiGiCo managing director Austin Freshwater. “We originally tasked our R&D team to create a rack-mountable Quantum console to replace the SD11, which is now 14 years old, and they absolutely outdid themselves. Not only is the Quantum1¹² rackable, but it’s also small and light enough when cased to be under the magic size and weight numbers when traveling by air.”

“The initial brief really challenged our team, driving us to research new materials, optimize design techniques, and inversely shrink topologies while increasing performance,” confirmed DiGiCo head of research and development, Michael Aitchison. “Our previously smallest console, the SD11, weighs 53 pounds without packaging, and 161 pounds when flight-cased, so it wasn’t a proper flypack product. The new Quantum1¹² is nearly 70% lighter, but still incredibly solid, and far more powerful. We’re extremely proud with how it has turned out and suspect that our customers will be quite pleased with it, too.”

It's here! Get your copy of the October SCN Audio Issue today.

Every October, SCN dives into the world of sound, bringing you insights from expert audio manufacturers for your next installation. This month, Chuck Gloman restores the spirit of radio, Graeme Harrison tells us how music can enhance the retail experience, and Matt Czyzewski dives into the power of predictable audio performance.

Also this month:

• Executive Q&A's with GraceNote and Vanco.
• AI and live events
• A deep dive into audio technology from Dante and Audio-Technica
• Insights from SCN The Nine Andie Mace

Find out more inside. Your copy of the October SCN Audio Issue is one click away.

In the early 1980s, Kim Wilde warned us that a new wave was coming. Of course, she was singing about the nightlife and emerging music trends that were tempting the kids in America. In 2025, there really is a new Wave. It's a speakerphone from Insta360 with a unique style and AI features that will no doubt tempt the Pro AV space.

Unlike traditional speakerphones, which tend to resemble hockey pucks, the Wave stands upright. Think cylindrical tower speaker with a circumference of a mini soda can. A bright, thin LED indicator light offers color-coded status updates, including green for power and red for mute.

Above the USB-C port on the back of the base is the power button. Press that button and the Wave doesn't just turn on—it announces itself to world. The speaker rises up from its base with its own intro music to expose a compact, circular touchscreen interface. Don't worry, you can shut off the chime in the settings, but the speakerphone will rise or fall with each power toggle.

It's about 9.75 inches when fully elongated (you have three height choices). The top of the unit was also designed as a perch for the company's Link 2 4K webcam, which has a magnetic base, but that means you need at least another 2.5 inches of clearance, so be careful with that desk hutch.

The Wave comes with a sturdy USB-C cable, USB-A to USB-C adapter, wireless dongle, quick start guide and other documentation, and a padded carry bag that offers some protection during transport. An integrated suction cup keeps the Wave in place, but a table mount is also available separately.

Audio Options

I tend to keep my speakerphone plugged into my computer or dock, but the Wave offers wireless connectivity via Bluetooth, Wi-Fi, or the included dongle, which can be stored conveniently (and magnetically) on top of the unit. Audio is saved as MP3 files by default. You have the option to record lossless WAV files, but that will result in larger files. Either way, the Wave has 32 GB of built-in storage, so you won't be worried about storage anytime soon.

There are eight microphones strategically placed on the unit, four around the body of the cylinder and four on top. You also have the option of two audio modes, one that captures all sound (you choose the pickup pattern) and another with noise reduction that's designed for meetings. I opted for noise reduction—and then had a videoconference in the middle of a thunderstorm. The South Florida rain was hammering the skylights and the thunder clapped loudly, but the person on the other side of the Google Meet never heard a thing.

The touchscreen interface makes it easy to mute your mic, adjust the playback volume, or start and stop recordings. You can also see the time, displayed in military time presumably for less confusing file names, and the date. A small on-screen arrow provides you with access to files, audio modes, and other settings.

As far as speakerphones go, it's solid. The audio is crisp and clear. I was told I sounded good, even when I was 8 feet away from the unit, and there were no problems with vocal pops. It's also a good speaker for playing music, though it could use some help with the low end. Perhaps an EQ is in Insta360's firmware upgrade pipeline.

When the unit arrives, you'll need to remove a couple of protective stickers and then connect it to your computer. Getting connected takes a little effort. There's a File Transfer option under the USB Mode in the settings menu. That'll help you access the software for download on your PC or laptop.

Once it's installed and you've created an account, it's time to access InSight, Insta360's cloud-based AI agent. It's where your audio files are automatically uploaded. From here, you can listen to your recordings, summarize them, get transcriptions in multiple languages, and customize your experience.

Like similar transcription services, there is a free basic plan that provides a few hours of transcription per month. Want more? The Pro plan offers 1,200 minutes of transcription per month and increased functionality for about $100 per year, or you can buy blocks of transcription time in increments as small as two hours.

I recorded that previously mentioned conversation, plus I recorded a short excerpt from an audiobook presentation of The Hitchhiker's Guide to the Galaxy. The transcriptions weren't perfect—AI will be stumbling over names and less common words for the foreseeable future—but they were solid and did provide mostly accurate speaker identifications.

Summarizing the Summaries

Now, let's talk about those AI summaries. I had access to the Pro plan for this review. There are more than a dozen summary templates that simulate various types of meetings, from interviews to brainstorming sessions, and present the content based on those parameters as easily digestible paragraphs and bullet points. You can even create a custom template for more specialized results.

Generating a summary will take a few seconds, and that time increases as your files/meetings get larger, but InSight produces a transcript at the same time. You don't get one without the other. You can also update the summary using a different template, but that will take more time.

The summary functionality is a bit of an acquired taste. Like other AI-based content generators, it tends to lean toward verbosity. Yes, results will vary based on the template you choose, but you can train it to deliver the results you want.

For example, I had a 24-second clip that included three sentences of me pondering out loud if this experiment was going to work, followed by two sentences from the Wave's instruction book. Using the "Keynote Presentation" template, the generated summary was more than 400 words and included a general summary, key themes summary, three bulleted lists, and a to-do list with one item. For reference, the transcript of my recorded file was 67 words.

To be fair, the AI was clearly trying to make lemonade out of very small lemons. Also, I simply told it to create the summary without making any changes to the prompt. In another test, I read the first two paragraphs of the Declaration of Independence. Using the same template, the summary for this two-minute monologue was significantly more useful. (FYI, the transcription was excellent, too.)

The lesson here is that you'll need to find the template that works best for you and tailor it to your purposes. Speaking of tailored, you can enhance your transcriptions with industry glossaries, improving accuracy with terminology from health, legal, technology, and other fields. You can even create a custom glossary.

[Editorial: More AI Action and Frustration]

The Insta360 Wave retails for about $300 and is available in graphite black or arctic white. On the surface, you're getting a sharp looking speakerphone that delivers quality audio. Beneath that unique exterior, you're also harnessing AI technologies that deliver customized meeting summaries and transcriptions. In other words, the Wave provides the tools to help you communicate clearly and then actually do something constructive with your conversations.

AV Technology's content director, Cindy Davis, had a candid conversation with Daniel Sennheiser and Dr. Andreas Sennheiser, Co-CEOs of the Sennheiser Group. They discuss the stories that shaped the company's 80 years, new milestone technologies such as Spectera, its latest wireless broadband ecosystem, what to expect in the near future, and thoughts on artificial intelligence.

Cindy Davis: Let's start with your perspective on Sennheiser's 80th anniversary.

Andreas Sennheiser: It's more about the stories over the past 80 years that really shaped who we are and the impact that Sennheiser made in the market. At the very beginning in 1945, an engineering-minded group of people were obsessed with bringing good and better audio to the world. That group of people injected the spirit of creative dissatisfaction with everything that existed in that field, which then inevitably led to a lot of innovations over the decades. In 1957, the first wireless microphone—the first of its kind globally—was a revelation for everybody who was attached to a cord on stage until then. Another example from the 60s and 70s was the shotgun microphone, which brought creative freedom to the film world because it allowed for recording from a distance without the need for close micing. Every decade has its own special footprint that we have left. And the most recent, I would say, is Spectera, our newest digital ecosystem, bi-directional wireless system, which is really the first of its kind. It's like the move from a push-button phone to a smartphone, which becomes anything you want it to be depending on software-defined features. And this, I think, from the very beginning of a microphone to Spectera, really, what's alive is the innovation spirit, the reliability of a product, and everything just to create remarkable audio experiences.

Cindy: How do the newer technologies, such as TeamConnect and Spectera, impact corporate and higher ed end-users and integrators?

Daniel Sennheiser: Everything starts and ends with a customer request or with a customer user experience, and that's what we are very much focused on. We have our TeamConnect family, microphone, and beamforming technology. We offer the most feature-rich collaboration bars on the market, but we also provide wireless platforms like SpeechLine Digital. Eventually, Spectera will also allow, especially higher education, but also corporations, to embrace the freedom and the new opportunities of that new platform. So, in the end, what Spectera offers us is a platform. Whether it's used for a conferencing system for a large corporate venue or for any monitoring system on a live concert, it doesn't really matter. It's about what the user does with the platform that we're offering. Therefore, we really believe in creating unique applications, unique experiences for the customers with the customers. That has been a stronghold of Sennheiser: co-creation with customers, whether it was with television in the 1950s or with universities today to create hybrid learning experiences, making collaboration and learning easier. It's always about understanding what the application is and how technology can help bring that promise to the customer.

Cindy: It seems like software has been driving innovation in our industry. How is this playing out at Sennheiser?

Andreas: I think you absolutely spot on with that and that's why that to Daniel's example, focusing on the user experience, which is an evolving request, that is very, very important that we really master this on the software side because a generic platform like Spectera will allow different applications, different user experiences, and the software enablement will also make sure that our products are future-proof because you can iterate much faster. We can bring in a lot more requests from customers when we can iterate on the software side than if we had to do a hardware spin. Everything that we're really doing going forward is thinking ecosystem, thinking in ever-growing and enlarging platform solutions, which then can be targeted for this application or for another application, really, depending on the front-end the customer wants to use.

Cindy: A lot of companies are talking about ecosystems, platforms, and iterations of products versus new hardware products. The iterative process can catapult an existing product into a new category, but on the surface, it might not look all that special. During our industry tradeshows, attendees are often looking for the shiny new object.

Andreas: A perfect example is our TeamConnect Ceiling solution. The ceiling microphone in its first iteration was just a very smart dynamic beamforming microphone. In its next firmware iteration, it had new features included, like camera steering, and used the positional information gathered from the speaker to give it to the camera so it could be steered. In the third iteration, we had voice lift integrated, so it became almost like a PA system in the room. In the newest iteration of the firmware, it now behaves like native devices that can be integrated into Q-SYS, Crestron, and Extron control systems completely natively. And it still looks the same from the outside, but in terms of functionality, it has developed fourfold since its inception.

Daniel: At the same time, the complexity is even harder because we as humans are still inherently analog. So, we cannot, at least for the foreseeable future, directly connect to our brains. So we need an analog microphone, we need an analog speaker, and at one point, that needs to be translated. And for that, we need very good hardware. And this is where Sennheiser has historically been strong: at the intersection of the best hardware and the best software. Hardware is something that a lot of companies, especially new software companies, take for granted, and that's where they fail, because if you have stable hardware and the software has to run on that hardware, we need to foresee all the iterations that are coming in the next two to four years and build that into the hardware. And this is where the magic of a long-term thinking company like Sennheiser, still privately owned, still a family company, we can build all that knowledge into the hardware to then enable the next iterations on the software side. It's this combination that makes us so reliable. So, if somebody is in front of 80,000 people or in front of 600 students, feeling completely confident that the system is running remotely, maintained, and being checked just before they go on stage, and talk to their audience.

Cindy: We're posting news about acquisitions just about every day. What keeps Sennheiser a family-run, independent company?

Andreas: It gives us the freedom to really focus on what matters for our customers. This is the be-all and end-all to staying independent. We have a lot of topics that we have followed up on for decades that sounded very promising, and in the far future. Immersive audio is a good example. We've been researching 3D audio algorithms for decades, but there was really no application until AR and VR started to take off, and suddenly, we could bring a lot of our ideas there. Then we brought it to the automotive space for three audio experiences in the cars or doing software plugins like the RIME [Reference Immersive Monitoring Environment] plugin from Neumann that allows you immersive mixing on the go. A lot of these forward-looking things that probably stock-listed companies or companies that are funded externally would not necessarily do to this extent.

Our relationship with the customers is also a commitment to try out things with them that may or may not work—to be in an exploration phase with them. That has proven to be very strong, particularly in times where others have failed. A perfect example is COVID, because we invest so much in our own supply chain, and we have very strong relationships with the major chip manufacturers. We hardly had any product that was not available during the whole COVID period, where most of the others have just said, "You know what? Come back 18 months from now." And I think in such times, it's not just about being a good friend who has coffee with a customer, but also being a partner, offering products and support. I think these are all ultimately a testament to the actions and the thinking that only independent family-run companies would probably do.

Cindy: What’s on Sennheiser’s roadmap?

Daniel: We believe that there are still a lot of great things that we can do, especially in the collaboration and higher education market. We are a company that is agnostic. Ecosystems are really important, but we don't want the customer to be locked into an ecosystem. We play natively with all the big ecosystems because we want the customer to be able to make their own choice. We want the integrator to play their important part, and ultimately, the end user, whether it's a university, a corporate hospital, or whatever, to have the best experience. And for that, we now also bring our Control Cockpit to the cloud, which is basically the control software that is able to monitor all your endpoints. Suppose you have multiple campuses or multiple buildings across the campus. In that case, being in the cloud is actually very helpful because you can manage all those things at the same time from your home office or from your office without having to walk around and check that every professor put the handheld back into the charging cradle or every microphone is working correctly. Looking out a little further, we believe that, especially in the field of 3D audio, there are a lot of things still to explore, and that promotes inclusion for people who are on the far end.

Imagine a large corporate meeting or a university lecture today. You often have people calling in from the far end, and they are very frequently second-level participants. With 3D audio, you're able to have less fatigue, you have better inclusion, and at the same time, you also include people on the ground. For example, our Sennheiser MobileConnect is the system for the hard of hearing. People can bring their own devices; they don't have to sit in a certain area. They can use just their smartphone and the already installed university network to feel included and participate in the lecture from anywhere, including advanced features such as voting. Understanding those verticals and especially those high-impact verticals, which we call education spaces, hospitals, large corporate boardrooms, or large training facilities, is where we understand the user really well. Our technology, whether it's our TeamConnect Ceiling, Spectera, SpeechLine, or Sennheiser MobileConnect, all play together because every solution is unique, and every solution needs to be tailored to the customer.

Cindy: The buzz has been about artificial intelligence. How is Sennheiser using AI now, and what is the plan for the future?

Andreas: AI is really cool. For us, AI has mainly two vectors. One, it's an enabler for speed and efficiency because it allows us to develop fast prototypes much quicker, validating assumptions and conducting research. The right use of AI can really propel everybody forward in acquiring knowledge and getting solutions ready. What we believe is even more important is how AI can make our solutions better, our product smarter, and more context sensitive, like a TeamConnect Ceiling sensing, whether it's a person speaking, a coffee machine grinding, or the ventilation sound. So, I think AI has the potential to make the products smart, which allows them, without human interaction, to react as if they're almost intelligent devices that preempts what needs to be done before a user may even have identified the need for a change. So, it's both enabling our products to be almost like smart agents that support whatever the customer wants to do, and at the same time, allowing us to be much more efficient and fast-forward in bringing a solution to the market.

Cindy: Final thoughts?

Daniel: Maybe just one point, which I find interesting, especially now, for our 80th anniversary, because we've decided that we want to talk about innovation and the iterative aspect of it. That's why we also decided not only to share the success stories, but also to share things where we successfully identified a path that does not work. And I think that's really interesting to understand as well. We tend to have a culture where you always talk about what's working, and you most often learn more from the things that are not working or not working yet, because the idea was great, but the execution wasn't. Look forward to some stories, such as the Surrounder, which was an idea out of the late 90s, where we created a thing around the neck where you could listen to surround sound. It sounded great, it looked really stupid, so it wasn't a commercial success.

But today, if we think about 3D audio, that's exactly the kind of frontrunner to that. Or we brought out the first true wireless headphones in 2008 when smartphones didn't have Bluetooth on board. The experience was bad because you had to plug in a transmitting device to then connect to your earphones. However, a couple of years later, when Apple introduced Bluetooth on its iPhone, it suddenly created the market. So we'll talk about these things where we can anticipate a future, and maybe it's the same a little bit with AI right now, where a lot of people think AI is the solution to everything, and it's probably not. AI is a solution to a lot of very smart problems, but sometimes a simple algorithm is much better because it gives you a much more predictable result. We want to elevate the discussion in our 80th anniversary with maybe also a little twinkle in the eye and show that we're different.

Introducing Extron's new MGP 641 xi 5K multi-window processor. It is designed for presenting up to four 5K sources on a single screen, including 5K and ultra-wide displays. Supported video signals can be up to 5120x2880 with 4:4:4 color sampling, and Extron's exclusive Vector 4K scaling engine ensures downscaled and upscaled content is presented with unequaled image quality.

[Commitment to Competition]

In addition to four HDMI inputs that support data rates up to 18 Gbps, the HDCP 2.3-compliant processors include an HDMI background input for presenting live, non-scaled content behind the source windows at resolutions up to 5K. The MGP 641 xi 5K SDI model supports 12G-SDI sources, along with HDMI. An HDMI output accommodates a local display, while the DTP3 output that supports 4K/60 and 5K resolutions drives a remote display located up to 330 feet (100 meters) away with identical content.

"System designers and integrators asked for a way to support 5K and ultra-wide displays. Our newest MGP 641 xi 5K delivers that and more," said Casey Hall, chief marketing officer for Extron. "It easily scales and presents up to four windows of 5K HDMI video on a single display, and the LinkLicense upgrade enables annotation over live content, making it one of the best scaling solutions for a diverse array of environments."

The scaling engine, rack-mountable form factor, and wide variety of presentation and transition effects of the MGP 641 xi 5K all combine to create the optimal signal processing solution for corporate boardrooms, training rooms, and live environments such as public auditoriums. They are equally effective in mission-critical venues that require display of live information, from a situation room or control room to cascading multiple MGP 641 xi 5K units in a command center.

These processors also provide customizable window layouts, video and logo keying, window transition effects, and output rotation. Two-channel embedded audio can be extracted or passed to the video output.

With an optional LinkLicense, users can easily annotate over live content to emphasize key points. Annotations can include free-hand drawing, shapes, lines, and pointer arrows, each in selectable colors. Providing enhanced content presentation with on-screen annotation and more, the MGP 641xi 5K processors are ideal for high-end environments and live events.

CODA Audio has launched its free SPACE HUB Producer software, which empowers users with ‘immersion without complication.' Designed to build immersive sound fields using their own computer and audio hardware, the free-to-use software also forms an update for its SPACE HUB immersive sound processor, with added functionality and new features.

[An Immersive, Ghost-Busting Adventure]

“Unlike other immersive platforms our SPACE HUB Producer software offers unlimited usage, allowing producers and sound designers the chance to develop and experiment within this exciting space, and be introduced to the huge potential of our fully supported CODA Audio SPACE HUB engine which delivers the ultimate in immersive sound,” said David Webster of CODA Audio.

SPACE HUB is CODA’s 19-inch /3U immersive processor which allows users to build audio experiences in three dimensions. With more areas of the performing arts adding an immersive dimension to productions, the decision to release a free-of-charge, fully-functional, downloadable version of the SPACE HUB software is ideal for sound designers to plan and preview their immersive audio environments in their pre-production facilities. It is also designed with educational and training purposes in mind and enables anyone to find out for themselves what SPACE HUB has to offer.

Identical to the software used in the SPACE HUB engine, when installed on the user’s computer SPACE HUB Producer offers 128 input channels. These can be sourced live from the user’s interfaces of choice, or from internal computer playback. It’s an object-based platform, where each input or group of inputs becomes an audio ‘object’. Objects can be placed anywhere in the three-dimensional audio space, and can be fixed or moving. Snapshots, routines, and movement can all be live or triggered by Midi or OSC, and it can be used in combination with performer tracking systems. Within the software there is also a comprehensive reverb engine and downmix facility.

For SPACE HUB Producer, output is restricted to 16 channels and the software does not have direct help and support from CODA Audio. Where ‘immersion without limitation’ is required with higher output channel counts, lowest latency, and premium interfacing, CODA’s SPACE HUB processor gives the full, professional solution with up to 128 x 128 channels.

Audinate is expanding support for SMPTE ST 2110-30 and AES67, further strengthening Dante’s integration with broadcast networks. Building on a foundation of interoperability and high-performance audio transport, the latest updates equip broadcasters with new tools to meet the demands of modern production workflows.

[AES70 in Control]

Dante is already widely used as the backbone for audio over IP in broadcast environments, powering everything from comms to remote production and studio infrastructure. As video workflows continue their shift from SDI to IP, most notably with the adoption of ST 2110, there is a growing need for convergence with established Dante audio networks. These updates help ensure compatibility and interoperability between Dante and ST 2110-based video systems and networks without sacrificing performance and ease of use.

“These enhancements strengthen Dante’s role in bridging audio and video workflows across modern broadcast networks,” said Mark Gerrard, group product manager at Audinate. “Interoperability has always been central to our approach, and as broadcast networks and production needs continue to evolve, we remain focused on extending Dante’s capabilities to meet real-world demands.”

The latest updates give broadcast teams more flexibility in how they manage ST 2110-30 and AES67 audio flows, whether working in a mobile unit or across a multi-facility operation. Core configuration tasks, such as defining flow parameters and adjusting PTPv2 clock settings, can now be completed directly in Dante Controller without the need for Dante Domain Manager. This simplifies setup for smaller or fast-paced environments where cloud or server-based tools may not be practical. For larger or more complex systems, broadcasters can continue using Dante Domain Manager to automate configuration, apply global policies, and coordinate settings across subnets, while also taking advantage of expanded device-level control.

Added support for 96 kHz audio enables broadcasters to meet the demands of high-fidelity production workflows, including music programming, immersive audio, and live event capture. To enable this, Dante now supports ST 2110-30 AX, BX, and CX conformance levels, ensuring interoperability with systems and equipment operating at higher sample rates.

New device-level configuration capabilities further extend Dante’s utility in broadcast environments. Features such as the ability to assign flows within a broader multicast address range, define custom RTP payload IDs, and specify PTPv2 clock DSCP values are among several enhancements designed to improve compatibility with third-party devices, support a wider variety of network designs, and provide long-term flexibility as systems scale and evolve.

Extron is now shipping two new additions to its expanding family of 8K products. Introducing the
DA2 HD 8K AE and DA2 HD 8K AI, a pair of two-output 8K HDMI distribution amplifiers.

[We Love Sports! The SCN September Issue Is Now Available]

The DA2 HD 8K AE extracts two-channel LPCM audio from the HDMI input signal and converts it to analog audio. Its analog output is configurable for stereo or dual-mono signals. The DA2 HD 8K AI amplifier accepts separate two-channel analog audio and converts it to 16-bit LPCM audio at 48 kHz for embedding onto the HDMI output signal.

These HDCP 2.3-compliant products are loaded with features and capabilities that streamline integration and operation. They feature Extron EDID Minder Plus. This technology supports additional memory blocks for displays that convey advanced features via both EDID and DisplayID for proper management of the extended EDID information required to reliably display and distribute 5K and 21:9 aspect ratio content. The DA2 HD 8K AE and DA2 HD 8K AI ensure high quality and reliable system operation in a wide variety of professional environments, from presentation venues to esports arenas.

“Many professional audio devices do not fully support 8K HDMI audio de-embedding and embedding functions,” said Casey Hall, chief marketing officer for Extron. “These new products feature Extron EDID Minder Plus, which delivers advanced EDID management and editing capability to ensure complete compatibility.”

The DA2 HD 8K AE audio de-embedder and DA2 HD 8K AI audio embedder are designed and engineered for commercial AV applications with 8K video resolution requirements. Both models provide two outputs that support HDMI 2.1 data rates up to 40.1 Gbps, HDR, Deep Color up to 12-bit, 3D, and HD lossless audio formats.  Other features include input cable equalization, output muting, audio configuration, Key Minder that continuously verifies HDCP compliance, and more. The embedder also offers audio gain and attenuation adjustment and +5 VDC, 250 mA power on each output for external peripheral devices. With these many audio and video capabilities, the DA2 HD 8K AE audio de-embedder and DA HD 8K AI embedder meet the industry needs for audio processing and distribution of high data rate HDMI signals to standard and ultra-wide aspect ratio displays such as 21:9 and 5K.

Extron recently introduced its new NetPA 204 POE, a Dante-enabled, PoE powered, four channel audio power amplifier that delivers up to 20 watts per channel. It is designed to be mounted above ceiling tiles, out of sight and near the speakers. Receiving audio and power over a single cable, the NetPA 204 POE automatically recognizes connected PoE, PoE+, or PoE++ to deliver the corresponding amount of output power.

[Commitment to Competition]

With its integrated DSP, the NetPA 204 POE includes essential audio signal processing that offers the ability to meet the requirements of complex, decentralized systems. Dante Domain Manager and AES67 support ensures wide compatibility with enterprise configurations and other network audio devices.

“The NetPA 204 POE simplifies design and integration by requiring only a single cable for power and network audio,” said Casey Hall, chief marketing officer for Extron. “The integrated DSP makes the NetPA 204 POE an ideal solution for multi-zone and user-reconfigurable spaces that utilize decentralized control and audio over IP.”

The NetPA 204 POE is housed in a plenum rated enclosure that is fanless for silent operation. Like all Extron products, the NetPA 204 POE has been engineered for many years of reliable service.

A recent rollout at Australia’s GPT Space&Co features Sennheiser’s TeamConnect Ceiling (TCC) microphone technology as the backbone of a scalable AV setup that supports today’s hybrid workforce. In partnership with AV integrator Programmed, the GPT Space&Co meeting rooms were outfitted with Sennheiser’s TeamConnect Ceiling 2 (TCC 2) and the newer TeamConnect Ceiling M (TCC M) microphones across key spaces, enabling clear, reliable, and automated audio in any room layout.

Unlike traditional co-working models, GPT Space&Co focuses exclusively on premium meeting and event rooms, with technology designed to be as invisible and intuitive as possible. In 2024, GPT Space&Co opened four new A-Grade sites, including a flagship at 181 William Street in Melbourne, which was designed to provide seamless, on-demand meeting spaces for businesses that prioritize quality, flexibility, and integrated technology.

The Ceiling-Mounted Advantage

Each meeting space is designed to accommodate a wide range of use cases, from executive briefings to team workshops and hybrid presentations. These rooms are walk-in ready with Microsoft Teams Rooms, BYOD support, and an AV experience that requires no technical training.

“For the larger rooms, flexibility was a key focus,” Camden Webb, flexible workspace technology manager at The GPT Group, said. “Traditionally, when AV equipment is placed on or under tables, it limits how the furniture and, consequently, the space can be used. Whether the room is set up in a theatre style, boardroom style, or workshop style, users still receive the same high-quality experience.”

Webb added that ceiling-mounted audio has transformed the user experience: “The Sennheiser TeamConnect Ceiling microphones ensure that no matter where someone is located, they can be heard clearly. Remote participants can follow discussions and feel engaged, even if they are not physically present.”

Rooms like Petition and Verdict use TCC 2 microphones, allowing layouts to change daily without any impact on audio performance. The microphones auto-adjust to the speaker’s position thanks to patented beamforming technology, removing the need for tabletop devices.

Anthony Aloi, AV solutions manager at Programmed, noted that while fixed microphones can work well, they restrict modularity. “Ceiling-mounted microphones give us the flexibility dynamic spaces require,” he said. Webb agreed, adding that the system provides granular control over pickup zones to minimize background noise.

This integration is made even more valuable by the system’s compatibility with platforms such as Microsoft Co-Pilot, supporting AI-enhanced features like transcription, meeting summaries, and intelligent note-taking.

[Workplace 2025: Sennheiser]

Form Meets Function

The Chamber room at GPT Space&Co’s 181 William Street location is a 50-person event space with wall-to-wall glass and a unique geometric layout. Programmed specified three TeamConnect Ceiling M microphones for the space, offering powerful performance in a smaller, circular form factor suited to the room’s shape and acoustics.

“Combining three TCC Ms gave us great coverage of the space while allowing our team to program exclusion zones to minimize street noise,” Aloi said. “They are more cost-effective than the TCC 2s, and they look amazing.”

To preserve the room’s design, Programmed worked with Sennheiser and the builder to color-match the microphones to the ceiling finish. “In a premium space, integration is about more than function. The technology should disappear into the architecture,” Aloi said. The Judge boardroom uses the same setup for design and performance consistency.

With a broad range of users—ranging from lawyers to creatives—the solution also had to be intuitive and reliable. “These rooms are intuitive and support all devices,” Aloi said. “The technology stays out of the way so people can focus on the meeting.”

Webb noted that while a site manager is always present, their role is customer-facing, not technical: “Every guest needs to walk in, start their meeting, and go. Sennheiser’s microphones make that possible; they’re invisible to the user but essential to the experience.”

A Crystal-Clear Mission

As hybrid work evolves, GPT Space&Co’s latest rollout sets a new benchmark for flexible, elegant, and functional meeting spaces. With Sennheiser, they’ve built intelligent AV environments that scale with the needs of modern teams.

For Aloi, the mission is simple: Technology should empower people to connect—not complicate the experience. With Sennheiser, GPT is delivering a collaboration experience that’s consistent, confident, and ready for the future.

AV Technology Manager’s Guide: The Audio Guide

High-quality audio is no longer a “nice to have,” it is a “must have.”

21 Industry Thought Leaders share their insights and best practices on achieving high-quality audio, device interoperability, and the best possible user experience.

25 Audio Products that are designed for scalability and flexibility, microphones and speakers that ensure intelligibility, robust and flexible routing, voice lift systems for everyone in the room, delivering near-zero latency, and so much more!

6 Use Cases Dante-Enabled Audio for a Shopping Destination | Concert-Grade Sound | Police Station Gets Newfound Clarity, Enhanced Safety | Powerful Audio for an Evening of Advocacy | University Union Amplifies Event Accessibility | Scalable and Sustainable Higher Ed Solutions

AV/IT Teams A sophisticated, future-ready meeting space that adapts to any team, setup, or work style.

BONUS! AV Technology Best of Show at InfoComm: Check out the winners and what the judges had to say.

DOWNLOAD THE GUIDE HERE

PK Sound has introduced the Ts15 intelligent subwoofer. As the compact low-frequency offering in the Trinity range, this high-output, low-profile enclosure offers low-frequency performance with an operating range of 33-130 Hz and peak SPLs of 136.5 dB.

[Laser-Focused Sound Coverage]

“Ts15 was designed to embody our core philosophy of empowering users to handle the widest array of potential applications with the fewest individual products,” commented Colton Schmidt, engineering manager at PK Sound. “This module delivers unmatched power and impact in its class along with incredibly tight transient response – all in a compact form factor to maximize versatility.”

Ts15 features a front-loaded, long-excursion 15-inch transducer in a bass reflex design for clean and punchy output. A large vent enables high SPL performance with minimal port distortion. PK Sound’s proprietary VE platform single field-replaceable module located on the rear of the cabinet contains the 2,400 W Class D amplifier, control electronics and power supply.

Onboard DSP and AVB end points simplify set-up and calibration while optimizing acoustic performance, all routed through Neutrik’s DR Series of IP65-rated connectors for maximum stability.

Network-based auto-array via PK .dynamics software allows each Ts15 to recognize its position within the system for improved accuracy and simplifies advanced deployments including cardioid configurations and arc delays.

Integrated rigging hardware allows for modules to be securely ground-stacked or flown as single units in a wide array of configurations via an optional Bracket Set. An M20 pole mount thread is also located atop the unit and an optional Caster Kit is easily installed for portability.

Its low-profile design allows Ts15 to fit in a wide variety of spaces and efficiently pack for global transport, making it an ideal solution for both mobile and installed applications.

Snap One introduced a new line of Episode Hybrid Amplifiers, designed to meet the demands of large-scale outdoor speaker installations. These high power and versatile amplifiers are more powerful than its predecessors with 1200W and 2400W models, giving customers more options than previous models for easy and flexible configurations.

[Brave New Vision]

“As a combined business, the size and scope of our outdoor audio systems is expanding, and our customers increasingly need a powerful amplifier to deliver the exceptional outdoor and indoor listening experiences their clients expect. Today’s end-users are no longer willing to sacrifice quality when it comes to outdoor entertainment—they want their outdoor audio systems to match the sound quality of their indoor setups,” said Michael Buratto, director of product management for Episode Multi-Room Audio at Snap One. “The new and improved Hybrid Amps are designed to meet this dual need, along with providing features that enhance versatility and simplify the installation and configuration process.”

The Episode Hybrid Amps support both 70V and 8-ohm systems, giving customers the flexibility to mix and match speakers and subwoofers, including Episode’s outdoor Surroundscape speakers and Radiance speakers. Customers can also customize DSP settings for each project using either an intuitive web-based interface or the amp’s high resolution front panel display. Inputs and outputs are fully adjustable to accommodate a variety of source types and levels, from balanced and RCA analog inputs, to digital coax inputs. Episode rigorously tested the Hybrid Amps to ensure lasting performance across a wide range of commercial and residential applications.

“We’re committed to continue investing in our product portfolio and to ensure we’re working closely with our customers in order to offer them the latest and greatest solutions they need to help grow their businesses,” added Buratto. “Together, ADI and Snap One are focused on innovating throughout the indoor and outdoor audio space, and these new Episode Hybrid Amps are a testament to our ability to do so.”

The redesigned Episode Hybrid Amps pair seamlessly with Episode’s Surroundscape speakers and Radiance speakers and enhances the ADI and Snap One Outdoor Solutions Suite, which includes TVs, lighting, networking, and more, allowing customers to create fully customizable outdoor entertainment experiences for their clients.

Yorkville Sound has expanded its Paraline series with the next generation of enclosures. The four new powered enclosures are designed for ground stack and flown configurations and offer essential features and high output in compact cabinets.

[KRK Studio Monitors Support Immersive MAD Arts Exhibit]

The full-range PSA26 and PSA28 combine Tom Danley’s patented Paraline Lens and Synergy Horn technologies to minimize comb filtering while maximizing efficiency and directivity. They also feature lightweight neodymium drivers, tone shaping controls, onboard limiting, and integrated rigging hardware.

The PSA15S and PSA18S subwoofers are designed to pair perfectly with the top boxes, and offer 3000 Watts of peak power with your choice of 15 or 18-inch woofers.

All four new models are IP54 rated so they are ideal for outdoor use and can provide powerful sound reinforcement that is easy to transport and set up in any venue. These new PSA speakers are also proudly built at Yorkville’s factory in Pickering, Ontario.

“We listened carefully to feedback from our users to design new speakers with even greater scalability, functionality, and output,” explained Jeff Cowling, VP of sales and marketing at Yorkville Sound. “With the new Paraline, we’ve succeeded in creating the most versatile line array series of its kind, and one that can adapt from small stages up to flown touring systems.”

Eastern Acoustic Works (EAW) recently expanded its installation amplifier line with the all-new UXA4807D. UXA installation amplifiers offer onboard processing that includes FIR and limiter settings called Greyboxes. Greyboxes come preloaded and can be assigned to each output from a pull-down menu using intuitive software. Ample user-adjustable EQ, limiter/compressor and delay settings are available to fine tune the installation.

[Go Inside the Historic Carolina Theatre]

“Through the creation of the UXA Installation Series and specifically the UXA4807D, we are making it easier and faster to specify and integrate EAW systems,” said TJ Smith, president of EAW. “We understand that today’s consultants and integrators do not have time to review mountains of specifications and download settings. EAW UXA installation amplifiers save time during specification, integration and commissioning to provide trustworthy and long-lasting results.”

Multiple input signal formats come standard; Dante, four analog, and SPDIF signals are available in the UXA4807D’s comprehensive input matrix mixer.

To eliminate tampering, there are no front panel controls. All amplifier control is accomplished using a web browser when connected to the amplifier’s integrated Wi-Fi access point or hardwired TCP/IP connection. For larger systems, multiple UXA amplifiers can be monitored and controlled with EAW’s Resolution 2 software.

The Power Factor Corrected (PFC) universal power supply is efficient and tolerant of mains disturbances. UXA amplifiers operate from mains voltages of 100-240 volts and 50 or 60 Hertz. ErP and EnergySTAR compliance mean operating costs are reduced. UXA amplifiers recover from a loss of mains power quickly, passing audio from the analog inputs within 10 seconds from application of power.

DOWNLOAD NOW! NOW AVAILABLE! AV Technology Manager’s Guide: AV & The Cloud

Cloud-enabled devices and systems have fundamentally changed the way AV/IT teams within enterprises and institutions operate, ultimately enhancing the user experience.

Learn how an industry initiative, launched by a coalition of AV manufacturers and technology providers to accelerate the AV industry’s transition to the cloud, is promoting openness, interoperability, and customer-first innovation.

20 Industry Thought Leaders share trends and insight into the future of AV in the cloud. Combining cloud and embedded intelligence creates limitless possibilities.

23 Pro AV Cloud-Based and Intelligent Products that increase efficiencies, manage, monitor, control, integrate with APIs, provide data and analytics, improve security and sustainability, ensure real-time responsiveness and accuracy, enable remote device deployment, and so much more!

DOWNLOAD THE GUIDE HERE

Extron has unveiled the two newest models in the CA Series of Column Array Speakers. Introducing the CA 83T 30-inch Eight Driver Column Array Speaker and the CA 43T 17-inch Four Driver Column Array Speaker.

[4 on the Floor: Lessons from InfoComm 2025]

These new models offer customers new size options for a broader range of sound reinforcement applications. Designed for outstanding speech intelligibility, especially in reverberant environments, these new speakers feature a sleek modern aesthetic and flexible mounting options, making them ideal for any installation.

“Customers asked for smaller column array speakers to join our CA 163, so we are now proud to deliver two new CA Series speakers, the CA 83T and CA 43T,” said Casey Hall, chief marketing officer for Extron. “These are perfect for adding column array speakers to small or medium size acoustically challenging spaces or to supplement sound reinforcement in larger spaces used alongside our existing CA 163 long throw column array speaker.”

The CA 83T and CA 43T can be used as either standalone or as supplemental speakers alongside the CA 163. Their matched 3-inch (76.2 mm) drivers, ensure consistent voicing throughout the entire series, making them ideal as fill speakers in multi‑speaker applications. Two mounting systems are available to accommodate different room layouts and sizes. PT models include a pan and tilt mount for maximum speaker aiming flexibility, while LP models include a low‑profile mount for minimal protrusion from the wall. The CA Series is available in black or white paintable enclosures.

Games Done Quick (GDQ) operates semiannual “speedrunning” marathons for gaming competitors that raise millions for causes like the Prevent Cancer Foundation and Doctors Without Borders. GDQ has honed its events to a well-oiled logistic and technical machine, combining live performances in hotel ballrooms with real-time Twitch streams viewed by millions. With each of these marathons running 24 hours a day—for an entire week straight, Dante AV has helped simplify and elevate their workflows.

[Historic Style, Modern Tech]

However, managing this intricate setup has not always been seamless. “Our primary show is unique in that we have to serve both an online audience, streamed on Twitch.tv, and an in-person audience at a hotel ballroom," Jason Deng, director of technology at GDQ explained. "We also provide CCTV so that anyone staying at the hotel can watch the event on their in-room TVs.”

This dual-delivery setup, coupled with GDQ’s reliance on volunteers to manage critical AV workflows, created challenges that their prior solutions weren’t designed to address. That’s when GDQ turned to Dante AV.

Challenges Solved

Before adopting Dante AV, GDQ relied on a mix of video specifications and connectivity technologies that demanded significant resources and effort to manage. For video transmission, their go-to solution came with notable limitations. “With NDI, we needed to set up a PC in the ballroom and another PC to capture the signal at the hotel’s CCTV location,” Deng said. This configuration required multiple machines and technicians, and it was prone to signal degradation and artifacting if there were any network disruptions.

GDQ also experimented with connectivity technologies such as HDBaseT, but as Deng noted, “We had problems if the signal had to hop through network switches.” The complexity of laying out SDI cabling further compounded the challenges in physical setups, particularly in tight or difficult-to-access areas.

These limitations, combined with the need for user-friendly, reliable systems that volunteers could easily manage, highlighted the need for a more robust, streamlined solution.

A Familiar Face

GDQ’s previous experience with Dante for its audio networking made the transition to Dante AV remarkably smooth. “Most of our audio flow already ran through the Dante platform, so we were quite used to that system,” Deng said. When the opportunity arose to incorporate Dante AV, GDQ deployed a solution that drastically simplified their video workflow: a single AJA Dante AV 4K-T transmitter in the ballroom, and a Dante AV 4K-R receiver at the hotel’s consumption area.

This combination proved to be a match for the fast-paced, visually demanding content of GDQ’s gaming marathons. Based on the Dante AV Ultra codec, these devices deliver 4K60 4:4:4 HDR video with ultra-low latency and visually lossless quality over standard 1GbE networks. They support SDI and HDMI inputs and outputs, which helped GDQ streamline their SDI cabling. Dante AV Ultra audio and video streams are fully time-aligned and individually routable, ensuring flawless synchronization across displays.

“With AJA’s Dante AV-enabled gear, we deployed a single hardware solution—a source in the ballroom and a receiver at the hotel’s consumption area. It was essentially plug-and-play,” Deng said. “We tested it first to make sure it worked, but we didn’t need to set up multiple machines or deal with four or five different pieces of hardware.” The result was a reliable, low-maintenance setup that eliminated the switch-hopping issues and frequent troubleshooting required by their previous workflows. “With Dante AV, when you’re using an unmanaged switch like ours, there are no such issues.”

Volunteer-Friendly and Built for Success

GDQ’s reliance on volunteers to operate its AV systems made ease of use a critical factor in technology selection. “Since this is a charity fundraiser, we rely heavily on volunteers to manage audio and video,” Deng said. “It’s important that we use software and equipment that are volunteer-friendly.”

For that, the team used Dante Controller, a centralized management platform that provides a clear visual interface for routing AV signals. “Dante equipment has been great because Dante Controller provides a one-stop shop where you can see exactly where the signals are going,” Deng said. “It can also be managed remotely by those who know the system, and we can write training documentation for volunteers.” This user-friendly approach not only made the system accessible to volunteers but also reduced GDQ’s reliance on external contractors, helping to keep costs down.

Championship Results: Reliability, Scalability, and Quality

The impact of Dante AV on GDQ’s events has been transformative. One of the most notable outcomes is the reduction in technical overhead. “Before Dante AV, we needed two technicians and three machines to get a signal from the ballroom to the hotel CCTV,” Deng said. “Now, with Dante AV, we only need one technician and a plug-and-play AJA Dante receiver and transceiver.”

In addition to streamlining operations, Dante AV provided 24/7 reliability for the seven-day event. “The image quality was excellent, and it worked 24/7 for the entire event without any signal degradation,” Deng said.

Dante AV’s interoperability and scalability also stood out. The platform integrates seamlessly into the larger Dante ecosystem, enabling GDQ to manage both audio and video signals through the same interface. “Adding Dante AV is a smooth transition whether you’re already in the Dante ecosystem or new to the platform. And with Dante Controller, you can see the signals which makes laying out networking much easier than SDI,” Deng said.

As GDQ continues to refine its technical workflows, Dante AV is poised to play an even larger role. “I’ve been wanting to deploy more Dante AV to the rest of our stack,” Deng said. While bandwidth considerations remain a factor for future expansions, GDQ’s successful deployment of Dante AV for its flagship events demonstrates the platform’s ability to meet demanding professional AV requirements. “Dante AV works for us on many levels,” said Deng. “It’s reliable, easy to use, and eliminates many issues associated with other AV-over-IP technologies.”

Alfatron Electronics recently introduced the Alfatron ALFC-657D full-range, 6-inch Dante ceiling speakers. The new ALFC-657D ceiling speakers deliver enhanced sound quality and are designed for use in distributed sound reinforcement applications. The range is sold in pairs as one main speaker (ALFC-657-DM ) with a built-in 2x30W PoE (Power over Ethernet) amplifier outfitted with a 2-channel Dante Input, and one secondary speaker (ALFC-657-DS) equipped with a Phoenix input.

[The Next Big Thing—What’s on the Horizon for Pro AV?]

The Alfatron ALFC-657D full-range, Dante ceiling speakers use a flush-mounted design that enables them to blend aesthetically into a wide range of environments. The main speaker provides an amplified speaker output to drive the secondary speaker. Each pair of ALFC-657-D speakers are powered via the network POE switch. The system utilizes a 2-way, full-range loudspeaker design with a coaxial HF transducer. The 6.5-inch woofer features a moisture/water-proof composite diaphragm. A 1-inch silk dome tweeter with an adjustable angle produces rich mids, delicate highs, and an overall warm sound.

The Alfatron ALFC-657D ceiling speakers feature a wide, flat frequency response ranging from 65Hz to 20KHz and deliver very pleasing and articulate audio quality. The system has a high maximum SPL output of 111dB, enabling it to provide a commanding sound presence.

"As the latest addition to our company’s loudspeaker product line, the ALFC-657D ceiling speakers are engineered using premium-quality components and advanced acoustic design techniques to ensure superior sound quality, clarity, and coverage," Kevin Gelman, national sales director, CFO at Alfatron Electronics, commented. "This includes an aesthetically designed borderless magnetic grill for a visually pleasing appearance coupled with weather resistant materials that offer superior moisture protection for installation in humid environments.

"For distributed sound applications such as ceiling speakers in restaurants, outdoor patio areas, meeting rooms, and other spaces, the new ALFC-657D Dante ceiling speakers meet the highest standards of performance and durability. I’m confident customers will find much to like.”

Having the right wireless microphone for quick interviews—from 30-second spots for social to three-minute shorts for YouTube—is no longer an option, but a necessity. With the Saramonic Ultra two-channel wireless microphone system, I found just what I needed.

[More from Cavadi: Premium Podcasting with the MV7+]

The way people digest content has seen a dramatic change over the past few years. While reading an interview is still a popular choice, people like to see it in action and put faces to the names they’ve read for years. Whether at a baseball stadium checking out a new videoboard or conducting a Q&A at an InfoComm booth, wireless mics bring that experience to life—which means crystal-clear sound, good pickup, and no drops are imperative.

As you may know from previous reviews, I spend my free time podcasting and Substack-ing about sports. I recently took the Saramonic Ultra wireless mic system to a baseball game for a test run. The results were extremely positive, and I am now ready for success on the trade show floor.

Ultra Unboxing

The Saramonic Ultra system comes in a tight, compact box with nearly everything you need included. In the box, you’ll find another boxed set of accessories as well as the hard-shelled charging dock that also serves as the holder for the kit. The charging station/travel case holds two wireless, clip-on transmitters, the receiver, and two adapters for iPhones, cameras, and USB-C smartphones.

A couple of notes here. The charging ports for the mics and receiver have a blue sheath protecting them. Be sure you remove them, or you may wind up at your event with a dead battery. Second, as it is with smartphones and almost any other USB-C device, you get a charging cable, but the wall charger block is not included, so if you plan on plugging into a wall, make sure you have one. The good news is once you get a good charge, you are good to go for up to 24 hours.

There is a slew of accessories, including a pair of windsscreens and an antenna that boasts an additional 924 feet of range, that fit in a soft bag with a pull-string close. Yes, you have to carry an additional bag with you, but it’s a non-factor for me, as I typically have my backpack with me to carry my tripod, computer, and everything else needed for my mini studio on the go.

Fine Tuning

Setup is pretty straightforward. Open the box, connect the receiver to your phone, clip on the mics, and off you go. There is a Saramonic app that you can use for recording, but I used the Blackmagic Camera app for my test. I simply flipped the switch to Saramonic within the app, and I was off and running. It was that easy.

The receiver has a clip, but you can use the adapters to plug it right into your phone or camera. An OLED screen and dial allow you to easily scroll through options like output, mic gain, recording settings, timers, mono/stereo settings, and more. The transmitters have a clip to attach to your jacket or button down, but included magnets help keep them in place in case you are wearing a thinner T-shirt.

A huge feature for the transmitters for me is that they are IPX5 water resistant. As someone who fancies football and baseball games in the elements, that is a home run (excuse the pun). They are also designed with omnidirectional microphones that feature auto-gain and advanced noise cancellation, plus, the system allows for 32-bit float audio and recording.

Take Me out to the Ball Game

To put the Saramonic Ultra to the test, I conducted a quick interview at a baseball stadium with another college baseball journalist before a game. It was not a quiet environment. There were teams warming up in the background, some wind, and the sounds of fans slowly trickling into the stadium as well as music over the PA system.

There was some of that music in the background, but it hardly affected the conversation in the final edit. I also talk with my hands, so when I decided to hold the microphone (instead of clip it to my clothing) and wave my hands, it didn’t lose much pickup.

Take a look (and listen) for yourself.

The mics are an ideal fit for journalists, content creators, podcasters, live streamers, and anyone else looking for a quality sound and wireless convenience for interviews. You cannot buy the pieces individually, so if you are looking for just one wireless mic, you may want to look elsewhere. However, with a list price of $269, the Saramonic Ultra is a hit for novices and pros alike.

Bose Professional recently unveiled the Forum series multipurpose loudspeakers. The Bose Professional Forum series offers exceptional fidelity and surprisingly high output in a compact, install-friendly form factor—bringing a level of performance that’s rarely found in loudspeakers of comparable size and price.

[EdgeMax Takes Loudspeakers to New Heights]

“Forum full-range coaxial point-source loudspeakers are incredibly clear and loud, delivering outsized performance in an ultra-compact enclosure that solves the most common issues designers and installers face,” said Shawn Watts, director of product management, Bose Professional. “Forum raises the bar by combining precision engineering with the installation flexibility required to configure mounting options on the fly. The proprietary Beamwidth Matching Waveguide ensures signature Bose Professional sound quality, delivering consistent frequency response, clarity, and projection, while the surprisingly small cabinets can be rotated to change its coverage pattern to solve real-world issues in an elegant, precise, and powerful way.”

Available in 8-inch and 12-inch coaxial models, Forum FC108 and Forum FC112 are designed for fast and easy installation on projects where a small footprint, high output, and consistent sound are a priority. High-excursion drivers deliver up to 132 dB max SPL, while the Beamwidth Matching Waveguide aligns low and high frequencies for tonal consistency across the coverage area. Forum series loudspeakers employ a fully tuned passive crossover with no additional processing required at installation, ensuring excellent performance with any full-range signal right out of the box.

The Forum FC108 is an 8-inch full-range coaxial point-source loudspeaker designed for small-to-midsized areas with a frequency range of 53 Hz–20 kHz, 122 dB max SPL, and 110×60-degree coverage. The Forum FC112 is a 12-inch full-range coaxial point-source loudspeaker with a frequency range of 48 Hz–20 kHz, 132 dB max SPL, and also provides 110×60-degree coverage. Forum FC112 fills mid-to-large areas with even sound while delivering pinpoint accuracy in high-impact zones. Both speakers can be rotated at installation to change the coverage pattern to 60× 110-degree.

Both Forum series loudspeakers are IP43-rated for long-lasting durability in indoor and protected outdoor environments. Installation points on the top, bottom, and sides of the enclosure connect to a frame that’s integrated with the driver—making the entire cabinet rigid and secure when suspended. The cabinet’s orientation can be switched to fit a space's unique demands while maintaining aesthetic consistency with adjustable logos. Mounting options include U-brackets, suspension rails, and optional eyebolt suspension.

On the opening day of InfoComm 2025, Nureva is previewing its new HDX series. AV Technology’s content director, Cindy Davis, sat down with Nureva’s executive vice president of Products and Strategy, Rob Abbott, for some inside intelligence.

The full product press release is at the end of the interview.

Cindy Davis: Tell us about the HDX series and where it fits in your current lineup of products.

Rob Abbott: The HDX series is a line extension, and so we're adding it to our HDL line to broaden our portfolio. We're not taking anything away from our customers. We're providing more choices. And really, what we're doing is building on what the HDL is all about, which is great microphone coverage for conferencing use cases. We're increasingly extending our use cases into in-room use cases. And so, although the HDX does cover larger sizes, it's more about expanding into broader audio use cases, including full-range distributed speakers for amazing in-room speaker output to consume content. We are extending our full-room microphone coverage to larger spaces, improving our sound location data. We're building on adaptive voice amp by introducing adaptive voice lift. This is a brand-new area and a new set of use cases, and a really interesting area in the market as a whole, and I think a really exciting area, with the potential for users. Lastly, we're introducing Hearing Assist, which is a new area for us. And we're leveraging Auracast, which is a Bluetooth technology. We'll talk a bit more about that. But really, this is about extending our line, adding for our customers, adding choice, focusing not just on conferencing but a broader set of audio use cases, and taking our product portfolio into larger and larger spaces.

CD: While the integration channel and AV end users are critical to success, Nureva has always spoken directly to the IT audience. Why?

RA: Integrators and resellers are an important part of what we do and how we go to market. But from day one, the source of truth is the end user, and it's the decision maker. It's their problems we need to solve. It is fundamental to what we do. They give us feedback on the problems in their rooms, and it's our job to solve them creatively. So they're the source of truth. However, we also saw that our customers are increasingly taking on more responsibility for the AV in their rooms. And, of course, they want to deploy it simply. They want to manage it simply. More and more of our customers do their own evaluations and testing. From the beginning of the conversation, you're dealing directly with an IT manager, and they ask questions, wanting to know the ins and outs.

They want to try things in their own spaces; they want to set it up, and they want to experience it firsthand before making a decision. On the other side, increasingly, organizations are taking on more of the support, management, and maintenance of their rooms. They want to be in control of their own destiny. It was in our DNA to start, but it was those factors and how they wanted to buy and manage their spaces. It reinforced our commitment to continue down the path of providing tools and the right feature set for IT managers directly.

Our product inherently lends itself to a really easy evaluation. We support our customers in doing evaluations directly. We will send them evaluation kits. And because of the way our product works, you don't have to put it in the ceiling; you can place it on a tripod in the room, and it auto-calibrates. You can experience it in your space as an end user, and it de-risks the whole thing for them, and they just know. That's another proof point of the way we've implemented big-room audio with an integrated bar with Microphone Mist. It's really easy to evaluate in your space.

CD: For those not familiar with Nureva’s Microphone Mist technology, please share how it differs from technologies in other microphone bars.

RA: The core analogy is that we take a series of physical microphones in a room and use them to create a set of virtual microphones that create a coverage map of virtual microphones throughout a space. And this is corner to corner. The whole space is filled with virtual microphones, and then using microphone missed technology, we're able to monitor and find the best sound source of speech and then apply the right audio characteristics anywhere in the room so that, unlike something that is beam forming, that has lobes, has logic that essentially has switching logic and where is the user in what lobe, let's identify it and then switch to it. What happens in that process is you get switching artifacts. You don't get uniform coverage throughout a space. You also get more dependency on tuning and calibrating the space. Microphone Mist allows us to fill a room with virtual microphones, so anyone in the space can be picked up anywhere in the room.

Because of how it works, it can continuously auto-calibrate, so you don't need to tune it. One of the things we like to say about our products is that they're really one-touch installations, and that's important for the IT managers we're talking to. You can set up the room once, and regardless of the room's layout, changes in the use case, or the number of people in the room, the product will continuously auto-calibrate and provide a great experience. You don't need to call an audio expert to tune. That's a really critical thing for us. What we've seen as we talk to people is that most large spaces now have to do many jobs. The tables and chairs are on wheels, and the space transitions from a lecture to a collaborative session every hour. This room is changing. A key theme we're considering as we bring our products to market is that these large spaces are highly dynamic. We're trying to deliver a one-touch audio system that provides full-room audio coverage and great, consistent audio pickup.

CD: Please explain the use of the terms “adaptive” and “amplification.”

RA: The reason we chose the word adaptive for voice amplification and lift is on a few levels. First, the system is adaptive to the speaker and their location, as opposed to whether they are in a specific zone or location. Again, it comes down to ease of setup and the flexibility needed in these real-world, large spaces. It's also adaptive within a lecture setting. If a lecturing instructor starts with, I want to speak; I want to be amplified through voice amplification at, say, a 20-decibel level, but at some point during this lecture, I'm going to pause and ask for questions and then I want the student to be heard at a more subtle level. We can adapt to that flow between an instructor-led to more of a Q and A, and then we can be adaptive to whether you are focused on the interim experience, or distributed student or participants. So, we can adapt to all of those flows in a lecture or a meeting in real-time without user intervention.

Lastly, we can adapt to what's happening in the room. Do you need to use voice lift? Do you need to use voice amplification? We can be adaptive to that because a room, the day in the life of a room, has many uses, and from hour to hour, it changes.

CD: Given we've been talking about the IT manager, what can they expect from the monitoring and maintenance of Nureva's products?

That's part of our ease-of-use story because it's not just ease of installation; it's ease of ownership. One of the things that we always subscribe to is to try to make it not require specialized installers to do the installation. Again, we will still go through the channel where needed. The HDX is an integrated audio bar. The cable runs are really easy. We utilize single-cable power over Ethernet.

CD: Parting thoughts?

RA: We're really excited about bringing this to market. It's a big program for us. It's early days. What we're doing at InfoComm is a preview announcement. We're telling our customers about what's coming, and because they've asked us, they need to know about things as they create standards and start to think about, "What do I need a year from now?" We're targeting our shipments to start in Q1 of 2026. We'll get some units throughout the fall, but it's important that we set the right expectations.

Nureva Previews Highly Versatile HDX Series at InfoComm

Nureva Inc., an innovator in advanced audio solutions, announces the Nureva HDX pro audio series. Each HDX configuration is a complete and flexible solution designed to meet the full range of in-room audio and remote conferencing requirements for higher education classrooms, corporate training rooms, flex spaces and large meeting rooms up to 60′ x 40′ (18.3 x 12.2 m). Building off the success of Nureva’s HDL pro series systems for audio conferencing, the HDX series is a direct response to the mounting costs and complexities faced by IT managers who need to support a growing set of in-room audio requirements beyond full-room microphone pickup. This includes high-quality in-room media playback, voice amplification, voice lift, assistive listening, lesson capture, meeting recording and support for AI agents. Traditional pro AV solutions that deliver similar functionality require trained technicians equipped with the necessary components and software for installation. In contrast, the HDX series delivers all this functionality to IT professionals in a tightly integrated, cost-effective solution that doesn’t require special expertise to install, set up or maintain. HDX systems automatically adapt to changes in room layouts and their associated audio requirements, reducing the resource load on busy IT teams and their budgets.

HDX systems come with the choice of two or four wall-mounted audio bars and one or two optional microphone pods for Adaptive Voice Lift and enhanced microphone coverage. The system hub has single-cable AV-over-IP connectivity. These components are designed to be simple to install and set up, and therefore highly scalable, and they can be configured in varying combinations depending on the room size and in-room audio requirements. HDX systems deliver full-room speaker coverage that combines distributed audio with full-range loudspeakers for exceptional in-room sound. Adaptive Voice Lift enhances voice clarity across large spaces, dynamically adjusting to the talker’s position without requiring dedicated microphones or a complex setup. Integral to every HDX configuration is Nureva’s patented Microphone Mist technology and its unique unified coverage map. The physical microphones of all the components automatically configure to create an expansive, unified microphone array to provide full-room coverage while delivering maximum gain and consistently clear audio pickup throughout large spaces. HDX systems are also compatible with AI agents such as Microsoft Copilot and Zoom’s AI Companion. Additionally, their performance is further enhanced by using trained AI models for specialized audio tasks. Voice detection AI accurately distinguishes human voices from other sounds, enhancing audio location accuracy and improving camera tracking. Noise suppression AI filters transient background noises like chair squeaks, rustling papers and slamming doors. This is an ideal complement to UC&C platforms that lack integrated noise suppression capabilities.

“Nureva continues to push the envelope by simplifying complex audio challenges in large spaces,” said Ira M. Weinstein, founder and principal analyst at Recon Research. “With the new HDX series, Nureva makes in-demand features like distributed audio and voice lift not just possible, but easy to deploy and manage.”

“With higher education institutions rapidly adopting HyFlex learning models and office environments transforming into flexible collaboration hubs, demand has surged for full-featured audio solutions that support a broad range of in-room and remote conferencing needs in larger spaces,” said Nancy Knowlton, CEO of Nureva. “Our HDX series meets this demand with an unmatched combination of simplicity and performance that helps IT professionals fully serve the needs of their users while making the best use of their resources.”

Key features of the Nureva HDX series:

High-quality in-room speakers deliver full-range audio for media playback,music, presentations, videos and recordings.

Adaptive Voice Amplification provides in-room amplification of presenter’s voice when using a wireless mic, while simultaneously maintaining full-room microphone pickup of in-room voices so everyone can be heard by remote participants.

Adaptive Voice Lift automatically boosts in-room voices across large classrooms and meeting spaces, without the use of wireless microphones. When one or two HDX microphone pods are included, the audio bars automatically configure to deliver maximum gain, ensuring clear communication everywhere in the space.

Advanced conferencing audio guarantees full-room microphone coverage, ensuring clarity for remote participants.

Audio recording capabilities seamlessly integrate in-room microphones, wireless microphones and external audio sources, making high-quality lecture and meeting capture effortless.

Assistive listening is simplified with built-in Auracast broadcast technology, ensuring easy accessibility for those who are hard of hearing.

Simple installation and scalability

Two or four audio bars easily mount on walls and an optional one or two microphone pods easily mount on walls or ceiling T-bar rails, with single-cable connections to the system hub.

A user-friendly configurator tool streamlines the component selection, setup and deployment process, eliminating the need for external technical support.

The Nureva acoustic check tool, available with the iOS version of the Nureva App, measures background noise and reverberation, providing a clear score of room acoustics and guidance on optimal placement of the audio bars.

Sustainably designed using energy-efficient sleep modes, passive cooling and environmentally responsible materials such as 30% post-consumer recycled plastics and 95% recycled fabrics. The HDX series also features eco-conscious packaging made from 99% recyclable materials, replacing hard-to-recycle foam inserts with biodegradable molded pulp. Packaging dimensions are optimized to reduce waste, improve palletization and help lower transportation emissions.

HDX systems will be available to order in October 2025 and will ship in early February 2026.

Me: It's trade show season. I don't have time for any product reviews.

Shure: Hey, did you see our new mic that lets you plug in a second mic or a musical instrument without a mixer?

Me: Well played, Shure. Well played.

At its core, the new Shure MV7i "smart" microphone is a podcaster's dream. It's based on the same form factor as the SM7B, an XLR mic that's based on the SM7 that debuted in the 1970s. Though it only weighs 1.25 pounds, this is not a small mic; it's more than 6 inches long and 3.5 inches wide.

[Welcome WMAS]

The back panel of the mic has a USB-C port for simple connection to your laptop or phone. There's also a 3.5mm headphone jack, so you can monitor your audio directly from the source. But the star of the show—and the reason I simply had to test this mic—is a combo XLR/1/4-inch input. It allows you to connect another mic or a musical instrument, such as an electric guitar, and record both sources at once without a mixer.

DSP and Other Details

What's in the box? The MV7i includes a yoke that allows you to angle your mic after it's attached to a stand. There's also a black windscreen that needs to be removed if you flip your microphone orientation for use with a boom mic stand, as well as a 10-foot USB-C cable.

A few technical specs to share: The MV7i features a cardioid pickup pattern, with a 50-16,000 Hz frequency response and adjustable gain up to +36 dB. It offers up to 24-bit, 48 kHz sampling, as well as 16-bit, 44.1 kHz sampling for iOS users. There's also phantom power for mics that you plug into the MV7i.

The diecast aluminum mic body sports an LED touchpanel that provides a little built-in branding opportunity, too. You can customize the colors on the touchpanel through Shure's MOTIV Mix Desktop app or MOTIV apps on your mobile device, so if you insist on putting your mic in the shot (don't get me started), the LED display can be tailored to match your company's logo color. You can also adjust the LED brightness and turn off the "pulsing" mode (for live metering) if you find it distracting. The touchpanel also serves as a mute button.

Several Shure DSP features are configured via the various MOTIV apps. For example, the Auto Level Mode with SmartGate provides real-time mic gain recalibration for consistent output and prevents overlapping dialogues during two-channel recording. There's also the adjustable reverb (good for music, not so much for interviews), Digital Popper Stopper, and a real-time denoiser, which does an amazing job removing ambient noise you didn't realize was there until you shut it off.

I really like the audio output choices. Not planning on a lot of post? Mixdown is the down-and-dirty option, combining both sources into one mono signal. Multi-track keeps both signals as separate mono tracks, so you can separate them later for editing. And the stereo setting records one stereo audio track, adding the mic input to both channels along with the stereo audio from the 1/4-inch jack. That's a good choice for a music demo or maybe a voiceover with some background music.

There is a catch: The MV7i only supports mics or instruments with an XLR or 1/4-inch output; there's no USB-C input. I suppose that could be on the product roadmap for the future, but for now, you're limited to analog.

Plug In and Play

On its own, the MV7i is an outstanding USB microphone. Use it and you'll be the best sounding participant in your next videoconference, or you can output the audio directly to your DAW. You need one of the MOTIV apps to change the out-of-box settings and set your output mode, but you don't need it to actually use the mic.

The multi-source audio magic happens in the MV7i itself. To test it, I plugged in an Audio-Technica AT2040, a hypercardioid dynamic mic that's also purpose-built for podcasting. I know it was supposed to work, but I was still amazed that I had two high-quality mics from two different manufacturers working in concert with only one USB cable feeding my laptop. It also worked flawlessly with my iPhone.

Both mics sounded great, but the Shure's extensive digital effects do not automatically carry over to the second mic. At one point, the MV7i sounded like it was deep in a concert hall, while the AT2040 sounded like it was, well, in my office. To fix this, go into a MOTIV app and adjust the XLR or 1/4-inch input separately. I decided to match both mics, but you can choose what sounds best for your project.

Once you configure both inputs the way you want them, the MV7i keeps those settings, even if you unplug the mic. So, when I connected the MV7i to a second computer and specified it as an audio track in my Mixcraft DAW, that concert hall sound came through on both mics.

For my next test, I dug out my old Audio-Technica PRO 3H, a handheld mic I think I've owned since parachute pants were a thing. The XLR mic plugged in without a problem. It still sounds just fine, by the way, but the on/off switch was useless—it was always on when plugged into the MV7i. Still, the MV7i played nice with my XLR mics, old and new.

Then, just to be thorough, I pulled out my electric guitar. Don't get excited—I know maybe five chords, but at least I could test the mic. It was easy enough to insert my cable into the 1/4-inch jack, which is hidden inside the XLR port.

I slipped on my beyerdynamic DT 770 headphones and plugged them into the MV7i's headphone jack, turned on the app, strummed a G, and there it was. No mixer, no external converters, no amp, just a boy and his guitar. Add a little reverb on the vocal and suddenly I'm getting delusions of bright lights and merchandising deals.

The MV7i is priced at $349, and Shure also offers a podcaster bundle that includes the no-frills MV7X (with XLR output) for $499. So, do I recommend it? Well, if you just need a USB podcast microphone, Shure's MV7+ offers the same look and performance as the MV7i for about 20% less. However, for podcasters on the go, or even corporate podcasters with limited space or technical know-how, it's hard to think of a simpler wired setup with this level of quality. Seriously, two cables—one of which comes in the box—and you're creating broadcast-quality audio interviews.

[Wireless Microphone Headaches]

Plus, it's no secret that a lot of Pro AV folks have roots in (or still perform) music. For them, the MV7i is an easy way to record vocals with an instrument—and the built-in features can improve the quality of the finished product without fixing it in post. I get the feeling the MV7i will be pulling double duty in a lot of facilities.

Initially showcased at ISE 2025, Bose Professional has officially released the Veritas series of smart mixer amplifiers. Veritas amplifiers combine sound clarity and streamlined operation for background and/or foreground music, paging, and multimedia applications in small- to medium-sized commercial spaces.

[Capturing the Audio Content 'Explosion']

“Veritas series smart mixer amplifiers were designed to maximize the combination of sound and simplicity,” said Shawn Watts, director of product management, Bose Professional. “Veritas amplifiers combine high-fidelity amplification, simplified installation and modern OLED screens for intuitive control right out of the box. This makes Veritas a perfect match for small- to medium-sized installations in high-energy restaurants, retail centers, fitness clubs, bars, office spaces and public areas, where intelligible sound and simplicity of operation are critical.”

All Veritas amplifiers feature Bluetooth 5.0 connectivity, which offers greater range, lower latency, and more reliable streaming, and include an OLED screen incorporating an embedded QR code for self-guided setup. Scanning this code launches an easy-to-use webpage that walks users through the entire installation, setup, and Bluetooth connection process—designed to make it easy to get up and running, right out of the box.

The Veritas series is comprised of four models at launch: The half-rack-size Veritas 1100BH and Veritas 250BL, and the larger, more powerful Veritas 2160BH and Veritas 2160BL.

The half-rack Veritas 1100BH features a front-mounted XLR-and-¼-inch input for convenient mic or audio source connection. Designed for 70V/100V high-impedance systems, it’s perfect for smaller rooms, offices, and boutique shops. Similar in features to the 1100BH, the Veritas 250BL is the perfect choice for DIYers looking for a half-rack-size, professional-grade amplifier for low-impedance 4/8-ohm audio systems.

For medium-sized 70V/100V distributed audio systems requiring more flexibility, the Veritas 2160BH efficiently powers multiple zones with uncompromising clarity and performance and is ideal for hotels, retail, and multi-room installations. Similar in features to the 2160BH and for applications requiring more power in a low-impedance 4/8-ohm system, the Veritas 2160BL delivers robust sound for larger rooms, restaurants, and performance spaces.

All models include a rear ControlCenter port, allowing easy connection to an optional Bose Professional analog wall controller for convenient volume adjustment. Mounting kits and other accessories are available to allow for discreet installation on a table, desk, counter, shelf or behind a television or monitor.

According to DemandSage, a data analysis firm, there are more than 4.4 million podcasts as of 2025, up from around 4.2 million in May 2024. Meanwhile, the number of podcast listeners could grow more than 6% to about 584 million this year and could rise to more than 651 million by 2027. Worldwide, the podcast market was worth more than $30 billion in 2024 and could reach $131 billion by 2030.

Based on those numbers, it's no wonder podcasts are driving microphone sales, since content creators need mics to be heard. Joe Nassar, brand manager of Saramonic USA, said this "explosion" of podcasting, live streaming, and similar content is "creating many new customers and consumers of these types of products."

Market Drivers

Podcasting is probably a consideration for many of your corporate, educational, and worship clients. Some may already be producing content, while others could be planning for future podcast projects. So, what trends are influencing purchasing decisions? Without a doubt, simplicity and cost lead the pack. Podcasters want microphones that are straightforward to set up and use, as well as affordable.

“Since COVID and the rapid growth of the podcasting medium, people have looked for easy and accessible audio solutions that don’t require an audio engineering degree to tell their story," explained Robb Blumenreder, Sennheiser’s customer and market insights manager for pro audio. “As everyone from your realtor to music teacher are starting podcasts, it has become important to enable all levels of content creators with the right tools. Solutions like the Sennheiser Profile USB microphone, a USB-C based solution, empower creators with an all-in-one, plug-and-play option that is easy to use, affordable, and offers professional-level sound quality.”

“Simplicity and versatility are key,” agreed Sean Sullivan, Shure’s associate director of product management. "The market is looking for less gear, fewer cables, and simple operation.”

This is why his company recently introduced the Shure MV7i, a USB-connected, yoke-suspended microphone. It features a combo XLR/1/4-inch input jack that lets you plug a second microphone (or an electric musical instrument) directly into the microphone. “Shure’s MV7i is a direct response to customer’s needs with high-quality audio with a two-channel interface and mic in one unit,” Sullivan said.

[Wireless Microphone Headaches]

Another trend driving podcaster microphone purchases is the desire for podcasters to visually brand themselves as serious broadcasters. This is why so many podcasters are seated behind hefty desk/arm-mounted microphones—as if they were hosting World War II-era news bulletins from London during the Blitz. Blumenreder characterized this trend as the microphone as status symbol.

"Before, you never wanted to see the microphone in a video recording,” he said. “These days, seeing the microphone has become synonymous with having a serious show, quality equipment, and investing in your craft as a podcaster.”

Because of this branding statement, microphone-equipped headsets are rarely used by podcasters, whether on camera or not. “I don’t think these are very popular because they are not seen as professional, even though these types of headphone/microphone combos are very versatile and give you a consistent sound,” said Nassar. “Instead, many users want the look of a large diaphragm dynamic microphone in front of them to legitimize what they’re doing.”

A related trend influencing podcaster purchases is conservatism. “Brands like Shure are creating innovative products to stand out and offer more features, like with the MV7i, but podcasting customers seem to gravitate to more standardized and accepted products,” Nassar told SCN. “This is why you see the same mics like the Shure SM7B, Rode PodMic, and to a lesser extent Electro-Voice RE20s in so many productions. Customers and new podcasters seem to think that having the same products as successful podcasters gives them more cachet and make them seem more legitimate. But unfortunately, this kind of 'me too' mentality will only stifle innovation and new product developments and customer affordability.”

Finally, podcasters tend to buy cardioid microphones with their directional pickup patterns, rather than omnidirectional mics that capture everything in the room. “The vast majority of mics we see for podcasting are cardioid pickup patterns,” Blumenreder observed. “These are ideal because they do a great job of rejecting the background noise and reflections, only picking up on the person speaking directly into it. Without fail, for podcasts, the cardioid is the most popular pickup pattern."

USB Domination

When it comes to podcaster microphone connectivity, USB is king. But XLR has a part to play in this brave new world of audio production. “The popularity of USB is driven primarily by the absence of other products needed to make this kind of audio production work,” Nassar said.

At the same time, relying on USB connections alone “does limit some users in the long run,” he noted. “This is why you’re seeing many [podcasting] products now that are both USB and XLR.”

The reason USB is so prevalent has everything to do with the current nature of podcasting. “Generally speaking, many podcasters start off by doing it on the side semi-professionally to augment their business or as a hobby,” said Blumenreder. “With this approach, and no formal audio training or budget for studio time, most podcasters tend to begin with a simple USB microphone that starts at a low price point. However, as we look towards established, professional podcasts, we can expect to find higher-end equipment, typically XLR-based, with much more infrastructure supporting the end user’s needs.”

[Executive Q&A: Big Changes at DPA Microphones]

Although some podcasters like to see themselves as being at the cutting edge of audio production, the microphones they choose often reflect the safe, proven form factors of yesteryear. Plus, traditional styled radio/production microphones are preferred for the legitimacy they impart to podcasters, even though some innovative products have been expressly designed to solve podcaster problems.

Time will tell if the Shure MV7i and similarly creative mics will catch on as the podcasting industry matures. But one thing appears certain: As more podcasts are produced, mic sales are bound to grow as well.

Sennheiser unveiled its one-channel version of its Profile Wireless 2.4 GHz all-in-one microphone solution. Designed to be lightweight and ultra-portable, the one-channel version covers typical usage scenarios of solo creators, while allowing for later expansion of the system by including the same two-channel receiver as the Profile Wireless 2-channel system.

The streamlined single-mic system comes in a compact bag and includes a second USB charging cable. Also, a 32-bit float firmware update for all Profile Wireless systems is available for download on the Sennheiser website. Just like the two-channel version, the solo system is quick and easy to set up—no app required. It can be used on cameras, with smartphones and computers, and can be combined with an external lavalier microphone or even used to create a wireless boom. The mic has an operating time of up to seven hours, and 16 GB of memory for up to 30 hours of internal recording.

The Profile Wireless 1-channel mic system comes in a rugged pouch that includes the clip-on microphone with mini windshield and magnetic mount, a two-channel receiver, two USB cables, USB-C and Lightning adapters, a camera cable and a shoe mount adapter. As all Profile Wirelesss components can also be bought separately, nothing stands in the way of expanding the system to two channels at a later point in time.

For 32-bit float recording, Profile Wireless combines the audio signals of the two A/D converters inside the Profile Wireless mics to create a single 32-bit float audio stream. Creators who prefer 24-bit recording due to its lower memory requirement still have the option to deactivate 32-bit float recording and turn on the Safety Channel Mode. This will output a 6dB version of the audio in addition to the original sound level and thus help protect against clipping.

To update their Profile Wireless system to firmware version V4.1.0, creators are can visit the Sennheiser website.

Learning equity is vital. Every student needs to have the best opportunity to learn the material for any course of study. This is especially true in higher education classrooms where students are more discerning about the value they receive from the institution. The classroom, especially the hybrid classroom, where remote and in-room students come together in one space, has become a cornerstone of modern-day AV technology. When envisioning a hybrid classroom, we typically think of the visual elements—projectors, smart boards, mobile devices, and cameras that make it possible. However, audio is the most foundational element in creating an exceptional, equitable, and inclusive learning environment. Simply put, when speech is intelligible, students can learn. For everyone in a room to hear clearly and naturally and participate in discussions with ease, large classrooms or lecture halls may require a voice lift system.

Far and Near-end Equity

In the last five years, the Unified Communications (UC) industry has exploded in education. UC technology made it possible for those on the far end (remote students) to attend and participate in classes, events, and discussions. In the hybrid learning space, the far-end experience has been prioritized to ensure students are fully included—they can hear clearly, see presentations, and participate seamlessly. However, for those students attending in-person classes, those classrooms may need specialized audio technology to make sure everyone feels connected and is receiving an equitable experience. Those in the classroom need to feel like they're part of the conversation as much as possible and can hear what's being said in the back of the classroom. Voice lift provides speech intelligibility for whoever is speaking, so everyone in the room can hear clearly.

What is Voice Lift?

Voice lift is an area of audio science that is often misinterpreted. It's not a public address (PA) or sound reinforcement (SR) system. With a conventional PA/SR system, all signals received in the room are boosted to the point where even the most distant presenter can be understood. However, two things happen: In order to achieve a sufficient gain before feedback (GBF), presenter microphones must be worn very close to the body, and good placement can be challenging. In addition, communication from students around the classroom is encumbered by either needing to pass around a microphone or having them move to one. What's more, the front-of-stage loudspeakers in those systems must be quite loud to reach the back of the room. This results in uneven volume levels, where front-row students experience overly loud audio while distant students may struggle to hear.

Voice lift is a more evenly balanced audio system. It was introduced to evenly distribute the presenter's natural speech volume, even to listeners at great distances. It maintains consistent speech levels throughout the room, regardless of distance. Voices sound as clear, close, and natural as possible without a hint of artificial tone or feedback. To achieve that intelligibility, it amplifies audio signals individually while increasing the speech level by 3 to 6 dB above the noise floor. Ceiling microphones and distributed loudspeakers are divided into different zones with their levels controlled by an external DSP system.

When is Voice Lift Needed?

A rule of thumb is that voice lift is used in a room when you have listeners that are 24 feet from the presenter, such as a lecture hall or auditorium, but acoustics can play a role too. If it's a well-balanced, acoustically treated room, then a voice lift system may not be needed. If it's a highly reflective room, then intelligibility will be compromised, making voice lift essential. Anything less than 24 feet, it's harder to control the zones of a voice lift system, but in some cases, such as instructors who are naturally soft-spoken or low-pitched, it may be beneficial. The goal is to achieve intelligibility for all participants in a room.

Voice Lift for Active Learning

Within a voice lift system, even distribution of sound with no feedback is achieved by splitting the loudspeakers and microphones into different zones within the room. The quantity and locations of the zones will be determined by the size and acoustics of the room, the number of listeners, the distance between the microphones and the loudspeakers, and the polar pattern of the chosen loudspeaker. Determining how loud the system needs to be for even coverage and speech intelligibility in the room will require calculating the potential acoustic gain (PAG) and the needed acoustic gain (NAG). This formula is very important in designing a voice lift, or even if one is needed.

The goal is to achieve a value of >=0 dB for the difference of PAG − NAG. This indicates a potentially stable voice lift system. If the calculated value is negative, the system is potentially unstable and could suffer from feedback and insufficient intelligibility.

The following formulas are needed to determine the PAG/NAG value:

PAG = 20*log((D0 * D1)/(D2 * DS)) − 10*log(NOM) − FSM

NAG = 20*log(D0 / EAD)

PAG − NAG = x dB

If x >= 0, this value indicates a potentially stable system.

If x < 0, this value indicates a potentially unstable system, which tends to suffer from feedback.

D0 = Distance between the talker and the farthest listener

D1= Distance between the ceiling microphone and the loudspeaker

D2=Distance between the farthest listener and the loudspeaker

EAD=Distance between the talker and nearest listener

DS=Distance between the talker and the ceiling microphone

NOM=Number of open microphones

FSM=Ceiling microphone feedback stability margin

Here's a PAG/NAG calculation to illustrate:

PAG=20*log((33*10)/(6*7))-10*log(2)-6=17.9

NAG=20*log(33ft/12ft)= 8.78

PAG-NAG= 9.12 dB

A Voice-Lift Scenario

Let's say we have a large classroom where there are two ceiling microphones paired with four loudspeakers. The room is divided into four coverage zones. The first coverage zone includes the teacher's area. When they are speaking, the loudspeaker in their zone does not need to be on. Likewise, the loudspeakers closest to the instructor will be at much lower volume because they can hear the instructor unassisted to some degree, so they need less amplification or voice lift. But as you move away from the speaker and the distance increases, more amplification is needed to hear in the sides or back of the room. Essentially, the back of the room will sound the same to those students as those sitting in the front row. With the ceiling microphones and audio zones, when someone starts to speak in the back, the system will do the opposite, muting the loudspeaker within that zone but applying appropriate amplification so other students, as well as the instructor, can hear that student naturally and equally. The loudspeaker in that zone is muted to avoid a feedback loop. Likewise, the other ceiling microphone for the instructor is muted.

When multiple open microphones are operated at the same time, the systems must be balanced against each other. As interactions between presenters and the audience can often be lively, it is important to manage the interaction between microphones, loudspeakers, and the persons speaking. This is why a DSP is needed. It allows for mix-minus, which refers to the muting or lowering of loudspeaker volume in zones that have active microphones. This will help to avoid feedback in each zone, regardless of who's speaking in the room. The speakers will automatically adjust, depending on which microphone is active.

Challenges with Voice Lift

Feedback can be very problematic in a voice lift system. This is why, in addition to a zoned distributed loudspeaker system, the microphone itself will be an important part of creating an optimal voice lift. It should be a beamforming microphone with strong third-party DSP and matrix mixer manufacturing partnerships. A beamforming ceiling microphone will be able to cover a large area but in very narrow beams. This increases its discernability in what audio it picks up and helps to reject the speakers around it. It's like a virtual shotgun microphone. Microphones that integrate well with the DSP and matrix mixer will allow for the creation of multiple zones of mics and loudspeakers that can be fine-tuned so that mics can be muted as needed and the gain of the speakers can change, depending on which microphone is live.

Microphones Specialized for Voice Lift

In addition to beamforming technology, microphones should include features specifically designed to suppress feedback. Microphones that feature voice lift technology significantly amplify speech within the classroom and activate many processes to automatically suppress feedback. When specifying, designers should look for a solution that employs frequency shifting procedures, which automatically apply algorithms to the microphone output signal to mitigate the risk of feedback from the loudspeakers in the room. This ensures that various frequencies are shifted in real time while maintaining the voice quality. Shifting the frequencies considerably reduces the risk of feedback, as sound is transmitted not as a single wave bundle but in multiple wave packets. It also allows you to achieve higher gain. This frequency-shifting technique is used to achieve the best possible speech results.

Of course, frequency shift won't be needed on the far end, so within a Dante stream, one will be a frequency-shifted stream while the other will be a clean mic signal without frequency shift that's sent to the far end. This gives integrators the ability to balance the system nicely and even get more volume than they might need.

Good Acoustics Can't Be Overlooked

Of course, attaining the best listening and speaking environment comes down to the acoustics of the room. The designer and/or architect should prioritize good room acoustics. A highly reverberant room, which many classrooms can be due to hard, reflective surfaces, should be mitigated before considering a voice lift system. Because it would be impossible to get rid of all reflections, integrators should deploy microphones that make use of any remaining reflective properties of a room. A narrow beam of 30° ensures that excellent speech intelligibility is achieved even in rooms with many reflective surfaces. This focused beam is able to pick up each individual's speech cleanly, even when the presenter is standing with their back to the microphone (e.g., speaking and using a whiteboard at the same time).

Acoustic education for integrators, consultants, designers, and architects is also key. Understanding the impact good acoustics have in a room is a conversation as old as time and stands apart from any trends.

The Future of Learning

Students in higher education today are much more discerning in the educational programs and experiences they're paying for. Schools have to be ready to demonstrate the value of in-person learning. All students entering the classroom, as well as the instructors, want to have a good listening experience. As accessibility and equity become a central focus for all grade levels, audio is an important factor for everyone, especially for those who are hard of hearing.

Live theaters such as Broadway have been implementing voice lift for years. They don't rely solely on stage loudspeakers but also loudspeakers that fill in at balcony seats and under the balcony so the audio for those seats sounds just as even and intelligible as other seats that are closer. People pay good money to go to a play. Why shouldn't the higher education classroom be the same?

As schools get deeper into the journey of designing classrooms built around equitable learning for remote and in-person attendees, it's also important to think about how those same ceiling beamforming microphones integrate with cameras and controllers. For example, if a student in the classroom asks a question, the beam (location) from the microphone works with the camera to activate the correct zone. That way, the person at home can not only hear who's speaking but see them as well. Sennheiser is one of the first manufacturers to provide positioning data and presets for its TeamConnect ceiling microphones for Q-SYS, Crestron, and other camera controllers.

More schools and other organizations are starting to look at voice lift systems and how it can improve the in-room experience. For instructors, they don't have to strain to hear questions, and discussions can happen more organically and naturally, which benefits students who may be anxious about speaking. People want to learn, and instructors want to get the message across easily.

L-Acoustics recently introduced the LA1.16i, an ultra-compact 16-channel amplified controller specifically engineered for the installation market. This latest innovation delivers enhanced channel density in a space-saving 1U format, making premium L-Acoustics sound accessible for a wider range of venues and applications.

[The Nine 2025: Setting the Tone]

Purpose-built for near-field and short-throw applications where higher sound pressure levels aren’t the priority, the LA1.16i is ideal for hospitality venues, retail environments, cultural spaces, and residential installations. Its high channel count enables the deployment of large quantities of speakers – for example, up to 48 X4i compact enclosures – while its space-efficient design maximizes rack space utilization and optimizes installation costs.

“Following the success of our LA7.16i amplified controller, we identified a clear need for a complementary solution for small to medium installations,” said Genio Kronauer, executive director of R&D, technologies and platforms at L-Acoustics. “The LA1.16i applies the same innovative approach to these applications, significantly reducing the cost-per-channel of amplification while maintaining the sonic signature and reliability that L-Acoustics is known for.”

Like its predecessor, the LA1.16i incorporates L-SMART, L-Acoustics’ patented power management technology that intelligently matches real-time power delivery to the needs of the loudspeaker system. This innovation optimizes efficiency and ensures reliable performance while reducing energy consumption compared to traditional amplifier designs.

The LA1.16i is ideal for front-fill and delay systems in theaters and performing arts centers, distributed background music systems in restaurants, hotels, and retail spaces, L-ISA spatial audio and Ambiance active acoustics installations, and home cinema and yacht entertainment systems.

For network integration, the Milan-certified LA1.16i supports Milan-AVB and AES67 audio network protocols with seamless redundancy, alongside AES/EBU and analog inputs.

Installation-friendly features include terminal block connectors for all audio outputs, GPIOs for third-party control system integration, and a 24V DC backup power input for the DSP card. The LA1.16i is remotely configurable, monitored using L-Acoustics LA Network Manager software, and offers integration with popular media control systems through Q-SYS plugins, Crestron modules, and HTTP API.

LA1.16i represents a significant step forward in making L-Acoustics solutions more accessible for projects where high channel count and precise audio control are required. Its unique combination of features creates a compelling amplification solution that delivers uncompromised quality at an approachable price point.

Eastern Acoustic Works (EAW) launched its latest loudspeaker: NT206L, the first in the innovative NT Series of next-generation speaker products aimed at simplifying audio setup and operation for production events.

NT206L is a compact, self-powered line array loudspeaker perfect for production companies, system integrators, and venues seeking high performance, ease of deployment, and operation, along with cost efficiency in a lightweight design. Unlike conventional compact line arrays, the NT206L delivers superior sound quality, premium materials, and an easily adjustable horizontal horn dispersion in an enclosure made of both wood and metal. This new EAW Premier Class (P-Class) offering marks the first product conceived and designed entirely at EAW’s new facility in Franklin, MA.

Engineered for power and precision, NT206L features two 6.5-inch low-frequency drivers with 2-inch voice coils and a 1.75-inch high-frequency driver, brought to life by 1500 watts of onboard power. A newly patented EAW waveguide leverages EAW’s Focusing technology to provide exceptional high-frequency response. NT206L also debuts a new EAW Core Technology called Adjustable Horizontal Directivity (AHD). With AHD, users can quickly choose between four horizontal coverage profiles through simple hand-operated adjustments in the field. NT206L plays loud (139dB) and goes low (65-20kHz), all in a premium Baltic birch, steel and aluminum enclosure. Weighing under 32 pounds (14.5kg), this active loudspeaker is lighter than most passive, plastic enclosures in its class. NT206L is designed for rapid and straightforward setup, featuring familiar analog inputs and controls, without need for complex audio networking. It includes a comprehensive assortment of compatible accessories for transport, stacking, pole mounting and rigging.