When Bon Iver's eponymous sophomore album, Bon Iver, Bon Iver, dropped ten years ago, it was widely praised by music critics, debuting at the number two spot on the Billboard 200 U.S. chart and earning the group Grammy Awards for both Best Alternative Music Album and Best New Artist. To properly celebrate the anniversary of this milestone release, frontman Justin Vernon and his band brought the album to life in its entirety, plus additional material, with a pair of immersive live shows dubbed "Sincerely Grateful" at the new YouTube Theater at Hollywood Park using L-Acoustics L-ISA technology.

Not only did this exclusive two-day event mark the first concerts that the group had performed in more than a year and a half, but it also represented the first official worldwide deployment of the new L-Acoustics L-ISA Processor II. Using this next-generation 3D audio processing engine, FOH engineer Xandy Whitesel mixed two shows that garnered huge praise from both patrons and press alike for their ultra-high-resolution, panoramic sound.

[L-Acoustics L-ISA Immerses an Opera Audience in Open-Air Sound]

"My goal with this band is always to enhance the music and connect with the audience, not distract them with the sound," Whitesel shares. "When it comes to immersive audio, there's a fine line between glitz and gold, but the intimacy that L-ISA can bring to even a big venue like the YouTube Theater is amazing. I equate it to what it must have been like going from listening to mono recordings in the '50s to first hearing stereo in the '60s. Suddenly there's a whole new world out there, and it's much more sonically interesting and engaging."

[Cal Poly's Spanos Theatre Comes Back to Life with L-Acoustics Kiva II]

He adds that the L-Acoustics immersive audio platform allows him to "more easily make my mix sound huge and beautiful, and I can use spatial panning in ways that I'd previously feel guilty for," he laughs. "That can radically change how one mixes because normally we make so many choices and compromises based on building a clean and big-sounding 'mono-ish' left-right mix. With L-ISA, the mix from my FOH position translates consistently to almost the entire venue-not just to a narrow strip of audience standing in the middle of the room. I heard someone describe these YouTube Theater shows as a 'visceral, communal explosion of emotion,' which perfectly sums up the ideal Bon Iver concert experience that L-ISA helps create."

Whitesel is no stranger to L-ISA. Not only did he use the technology for the second leg of the band's i,i tour in early 2020, but he also first explored it with a still-legendary one-off concert at the Santa Barbara Bowl two years prior. The recent availability of L-ISA Processor II, he says, makes an already great thing even better. 

[Wayne State University Chooses L-Acoustics for New Basketball Arena]

"The new spatial front-fill is an uber-cool improvement," he enthuses. "Before now, front-fill was mono, and so this new ability finally brings a full spatially immersive experience to the very front rows. L-Acoustics has also developed a new reverb that's remarkable. The ones that I normally use are stereo, whereas the onboard L-ISA reverb spreads out the 'verb tail amongst the whole L-ISA configuration, and it sounds fantastic.

"The new Controller software adds several new features that greatly enhance usability as well. There is a feature for tying solo on the desk-in my case, a DiGiCo Quantum7 feeding out to all of L-ISA's 96 available objects-to focus on the correct object in the L-ISA Controller software. And they added compatibility with stereo channels and groups, which also helped me easily translate my left-right session and left-right brain to the L-ISA Scene/Extension setup. Even though it does take some training and practice to optimally use object-based audio, I don't think it's possible to have a bad mix with L-ISA. 

[The Factory Plans Ahead With New L-Acoustics System 'No One Can Say 'No' To']

Although it would be easy to assume that transitioning a 6,000-seat venue's fixed stereo loudspeaker system into a 7.1 L-ISA Immersive Hyperreal spatial audio design might require a significant investment in additional gear and time, these two shows quickly disproved that. YouTube Theater's stage PA primarily features 57 L-Acoustics K2 and 30 Kara II enclosures that make up the LCR main arrays and out-fill hangs. For Bon Iver's shows, local Certified Provider 3G Productions brought in only 11 more K2, plus two dozen extra Kara II for the panoramic system's Extension arrays. Furthermore, the switchover time to lower, reconfigure, and re-fly the entire loudspeaker system to match the L-ISA design required only five hours from start to finish.

For these shows, five Scene arrays spread out and flown over the width of the stage, all featured ten K2 over four Kara II down-fills, paired with far left and right Extension arrays, both composed of a dozen Kara II. Nine K2 over six Kara II per side served as out-fill hangs, while six Kara II per side functioned as over-balcony out-fills. Two arrays of eight KS28 subwoofers each were hung behind the center array in a cardioid configuration, with a further 12 KS28 positioned on the floor across the full arc of the stage. A dozen Kiva II spread out across the stage lip delivered spatial front-fill for the first several rows of seating. LA12X amplified controllers fully powered the entire system, which was driven by an L-ISA Processor II and three P1 digital audio processors, all on redundant Milan-AVB networking. 

[Solotech Goes With L-Acoustics K3 for No. American, European Projects]

L-Acoustics Head of Product and Technology Marketing Scott Sugden described the end result of what more than 180 loudspeakers combined with object-based mixing delivered: "Although most fans walking into these shows didn't know what to expect, what they heard was much more clear, detailed, and nuanced than they had ever experienced before. Even if they were seated all the way at the back of the venue, it felt like Justin was ten feet in front of them. The effects of his voice engulfed the entire theater and enveloped each patron with sound, giving the shows the intimacy of listening to the band on headphones. But it also very much had the dynamic raw power of a live performance and the ability to connect the fans with the artists on stage.

[L-Acoustics, Mixhalo Partner Up to Advance 5G Live Entertainment Experiences]

"L-Acoustics has spent the last 30 years developing technologies to ensure that everyone in the audience has an amazing experience, and L-ISA is the absolute culmination of those efforts. It will be hard for those fans who were at one of these shows to go back to a 'regular' concert again after experiencing this."

Every year Systems Contractor News announces its Installs of the Year Award in November. These awards are given out to firms that go above and beyond to create ultimate user experiences. This year was challenging for integrators to work around the Covid-19 restrictions while crafting personalized experiences for their clients. CommLink Integration Corporation won the Install of the Year Award for its installation at Concepts, a luxury clothing retail store (created by Amazon). Located in Boston, Massachusetts, the new retail store combines authentic skateboarding roots with the brand's ability to convey a narrative and translate its identity through an immersive audiovisual retail experience designed and executed by the crew at CommLink Integration. Brands and products employed to help deliver the conceptual narrative at Concepts include Sonos, Sharp NEC Display Solutions, BrightSign, AVPro Edge video matrix, Symetrix audio DSP and UniFi network hardware.

[Symetrix Radius NX, Barco Clickshare Conference Now Compatible]

Concepts also uses an original custom control system, called Latitude, conceived and designed by CommLink's engineers Hunter Grayson and Tim Adams. It is based on CSS, HTML5 and JavaScript.  Latitude runs on an in-house spec'd 1U micro server, which serves a custom GUI via a local-only web page, which is interfaced with iPads in the store. They selected a Symetrix Radius NX 12 X 8 DSP and xOut 12 for the system integration.

Evan Landry, CTO or CommLink Integration Corporation, said many of the reasons why they chose Symetrix as the DSP for the system were the sound quality, performance and the ease of installation. Landry said, "Two of our favorite things about Symetrix are, one, the Dante Brooklyn chip is baked-in and you get the full 64 x 64 channels; and two, the hardware reliability is top notch. By the time of the installation, we had become heavily entrenched in Symetrix DSPs, with all our engineers being Composer certified. One of the biggest points for us, aside from rock-solid reliability, was being able to remotely control any parameter simply by assigning a remote-control number."

The café has the same set-up and can be its own zone for watching the game or join with the rest of the system for regular BGM. This zone joining and dividing is accomplished by two matrices, which are switched simultaneously by the preset controller in the  Latitude control system: audio by Symetrix and a video (HDBaseT) from

[Symetrix Introduces xIO Bluetooth Dante-Enabled Endpoint]

What made The Concepts store installation unique and different is that it is a multi-use space. It is a clothing and skate boutique, as well as a café, an elite sports bar and a night club. There is a DJ booth, which has its own set of XLR inputs thanks to a QSC Atterotech unDx4i. So, in essence, the audio system performs two job functions. It is required to provide even coverage for background music during the day and provide bumpin' power for the night-club scene. This presented its own set of challenges during the design), as the needs for these modes are very different. 

The smaller scenes are vignettes of the audio system: the skate room has its own 86-inch display, as well as a Sonos ARC sound bar. This does double duty serving as the audio source for featured videos, as well as playing background music (BGM) during normal business hours.

Aurora is introducing the new DTX series of audio products.  At the core of the product line is the DTX Dante audio DSPs, with a variety of analog and digital inputs and outputs suited for systems of any size.

Dante is part of the entire DTX DSP series with models ranging from 4 x 4 to 64 x 64 channels digital audio.  Additionally, the 4 x 4, 8 x 8 and 16 x 16 models include analog input and output channels for integration with traditional inputs and speakers. 

At the heart of the system is the Dash software.  Dash serves as the all-in-one control and configuration tool for DXT DSP hardware. Systems can be designed to perfectly match specifications and conform as requirements change or evolve. All classes of audio processing modules for signal routing and shaping are provided.

Dash provides a new way of thinking about DSP software. This is a software designed by audio professionals for audio professionals. It uses an interface layout that will be very intuitive to even the most novice integrator yet extremely powerful in its capabilities.

It is often said that simpler is always better – and for venues and spaces with minimal audio input requirements, that may be the case. Many are turning to solutions with internal audio processing capabilities, which decrease the footprint of their network while reducing the need for separate components for signal conditioning, audio processing, and amplification.

“With design advancements, almost all amplifiers these days have some level of audio processing built in,” said Sanjay Kulkarni, senior director of amplifier product development at U.S.-based audio manufacturer QSC. “This audio processing includes signal conditioning, loudspeaker voicing, loudspeaker monitoring – for shorts, opens, impedance changes, etc. – as well as loudspeaker protection which happens within the amplifier, as they are directly attached to the speaker.”

With the pandemic changing the way spaces are designed, there has been increased usage of products with integrated amplification, including powered ceiling speakers and surface-mounted speakers. These PoE-powered speakers are excellent examples of the benefit of integrated amplification and receive audio via a network audio transport.

“In this example, the amplifier can be ‘tuned’ to the specific drivers within the speaker to achieve both higher fidelity and higher SPL than is possible via traditional speakers without built-in amplification,” said Tom Lureman, director of engineering and product management with commercial audio solutions provider AtlasIED. “In some cases, the drivers in the speaker can be bi-amped, further improving sound quality while eliminating the need for a passive internal crossover.”

According to some manufacturers, integrated amplification is one of the biggest processing and amplification trends within the residential and commercial audio markets, as seen through the popularity of products like Bluetooth speakers and portable PA speakers. When discussing their needs around audio processing and amplification, clients emphasized products that are flexible, reliable, and can easily be installed and managed in a variety of settings.

Lureman said customers are increasingly asking for “turnkey audio systems rather than separate components that must be piecemealed together. Declining are requests for processing from manufacturer X, amplification from manufacturer Y, and loudspeakers from manufacturer Z. Consequently, many audio manufacturers have shifted their focus on the development of complete systems that simplify installation and optimize performance rather than on individual product specifications.”

Other trends include flexible distribution of power, remote monitoring, and enhanced amplification. According to Kulkarni, flexible distribution of power per channel is becoming the norm to “help increase design efficiency and maximize an amplifier’s available power through all the connected speakers.”

With labor shortages and the pandemic forcing many staff members to work remotely, many clients are looking to secure easy- to-use remote monitoring capabilities that leverage the amplifier to monitor the overall health of the audio sub-system. From those running front-of-house at live venues to those building out the sound landscape within corporate meeting spaces, powered loudspeakers with internal audio processing and amplification built into the loudspeaker cabinet are frequently requested.

Manufacturers that have taken note of these requests are focusing on growing the audio processing capabilities within their products. The use of integrated amplifiers alleviates the possible “unknowns" when using separate equipment and integrators, and end-users are clearly seeing the benefit. “By integrating amplification into products, settings such as EQ/filters and limiters can be dialed in and optimized for that exact configuration, ultimately providing a better sounding and more reliable product that’s easier to set up and maintain for both integrators and end-users,” said Lureman.

Current Technology Options

Bogen Communications heard the functionality requests from clients and designed a suite of new solutions to address the market needs for integrated amplifiers with advanced audio processing capabilities. Its Nyquist product line and Platinum Series amplifiers were designed to handle the stringent demands of modern audio systems.

“These product lines implement new engineering and design technology, greatly reducing the power demands of the amplifiers, offering a greener solution with less power consumption, and a myriad of great features commercial sound contractors have come to expect from Bogen amplifiers,” said John Minnick, Bogen vice president of engineering.

The Platinum Series includes features rarely found in other commercial amplifiers, including five-band parametric equalizer with independent gain, frequency, and Q controls. Bogen also recently launched its Nyquist firmware release, increasing the capabilities for several Nyquist devices, including SIP-compliant endpoints for VoIP phone system paging applications and standalone network managed audio equipment.

QSC has continued to enhance its Flexible Amplifier Summing Technology (FAST) and the core amplifier feedback topology found within its PL380 and CX-Q Series amplifiers through recently awarded patents. This next generation of FAST enhances the overall usability of the technology while removing the need for special wiring connections dependent of particular use cases. These new capabilities will also allow for easy, 1:1 amplifier redundancy.

During the pandemic, AtlasIED introduced Atmosphere, a digital audio system that acts as a comprehensive ecosystem made up of interoperable processors, accessories, and amplifiers. The system offers a powerful, flexible DSP and uncomplicated user interface for enhanced customer control and engagement.

Atmosphere also includes patented ambient noise compensation (ANC) technology. In the past, ANC technology was difficult to integrate and provided limited end-user controls. Atmosphere responds and learns from user inputs, simplifying the overall setup, and uses a patented adaptive filter technology that uses voice in its measurement of the overall noise level without voice-based audio being sent to the Atmosphere processor.

What’s Next?

With manufacturers constantly looking to improve their integrated amplification capabilities, the list of possible future enhancements can sometimes seem never-ending. “In the future, we see the need for higher efficiency amplifiers that minimize loss and maximize the given amplifier power across the design, easy-to-use interfaces, built-in intelligence using machine learning, and real-time data analytics. Essentially cost-per-watt and cost-per-channel,” said Kulkarni. Also, continued enhancements to PoE are allowing for higher levels of product integration within appropriated audio ecosystems.

“As the world becomes increasingly IT-focused and compliant, many established and emerging trends from IT run parallel to the needs of AV,” Lureman noted. These crossover trends, which will only increase within integrated amplification, include remote cloud management, frequent software updates (both free and licensed), and higher power endpoints using PoE++/UPOE.

The What: Barix has unveiled the Exstreamer M400, the first model in a new generation of the company’s Exstreamer IP audio decoder family. The new hardware is designed as a modern, extensible, and easy-to-deploy decoder platform supporting the latest technology standards.

The What Else: Powered by Barix’s Linux-based IPAM 400 audio module, the Exstreamer M400 has significantly higher processing power than previous Exstreamer models and a more programmable, software-driven architecture for easier expansion, rapid prototyping, and customization. An increased buffer size and advanced buffer management technology bolster the Exstreamer M400’s resilience to network performance disruptions. Expanded support for IT security standards includes HTTPS: support for audio streams and the web-based management interface, plus secure, encrypted SIP (voice-over-IP) communications using TLS and SRTP.

[Barix Ships Analog-to-IP Speaker Upgrade Solution]

Installation of multiple Exstreamer M400 units is simplified by the devices’ freely-included System Configuration Tool. Accessible through a web browser, a dashboard enables installers to efficiently configure up to hundreds of Exstreamer M400 devices at once. It can be used to easily assign devices to grouped zones and specify up to five prioritized source streams for each unit—for example, emergency notifications can be set to override background music playback—as well as the relative volume of each stream. Settings can be copied from one Exstreamer M400 to multiple others simultaneously to significantly speed up multi-unit configuration.

The Exstreamer M400 features stereo, line-level, analog audio output with RCA-type connections and can decode streams in MP3, AAC-HE, FLAC, PCM, Opus, and Ogg Vorbis formats, as well as supporting SIP and secure SIP in peer-to-peer and server modes. Other new features relative to earlier Exstreamer models include IPV6 support; internal audio file storage with externally-triggered playback; and support for Singlewire Software’s InformaCast and Syn-Apps (now part of Intrado) emergency notification platforms with automatic endpoint registration.

“Our earlier Exstreamer products helped make Barix a leader in streaming decoders and earned a strong reputation for their longevity, exceptional audio quality, and feature-richness,” said Davide Nossa, product manager, Barix. “We are excited to continue the Exstreamer family’s hallmarks on a modern hardware foundation that offers customers ‘more of everything’—more power, more flexibility, more security features, more protocols supported, and more extensibility for the future. More than 20 years of IP audio experience and market expertise went into our development of the Exstreamer M400, creating a decoder platform that integrators and users can trust for their AV needs today and in the future.”

The Bottom Line: The Exstreamer M400 initially targets IP audio applications in the professional AV market including background music, paging, SIP/VoIP integration, and emergency notifications. Direct support for Barix’s flagship IC Paging and Simple Paging solutions, scheduled for release later in Q2, will allow Exstreamer M400s to be added as endpoints. The highly extensible device will also be expanded with robust functionality for radio broadcasters in future firmware updates, such as AES67 support for high-quality audio transport between facilities and interoperability with Dante-based devices.

The What: Extron has begun shipping the SSP 200 high-performance Surround Sound Processor designed for pro AV applications in corporate and commercial environments. 

The What Else: This new processor automatically decodes Dolby and DTS formats from HDMI sources to discrete audio outputs. Providing up to 10 built-in outputs, plus additional outputs via an EXP port, the SSP 200 provides the flexibility needed for multimedia system requirements, including support for the latest immersive formats of Dolby Atmos and DTS:X. An up-mix function synthesizes exceptional multichannel audio from stereo content.

Related: Audio's Next Frontier: 3D Soundscapes

Featuring an HDMI input with loop through and EXP expansion port connection to DMP Series audio DSP processors, the SSP 200 is designed for integration into pro AV installations. The SSP 200 is housed in a 1U, half-rack-width enclosure that is rack‑mountable and occupies a fraction of the space required by a consumer surround processor or AV receiver. It can be controlled and configured using RS‑232 serial control or via a network connection.

"The SSP 200 addresses the increasing demand to provide immersive audio systems in boardrooms and auditoriums,” said Casey Hall, vice president of sales and marketing at Extron. "It offers a host of integrator-friendly features not found on other surround sound processors, all provided in a compact half-rack package that makes the SSP 200 the AV professional's clear choice over bulky consumer receivers."

The Bottom Line: The Extron SSP 200 automatically decodes Dolby and DTS formats from HDMI sources to discrete audio outputs, and is designed for use in corporate and commercial environments.

Last summer, AtlasIED launched Atmosphere, a digital audio platform comprising an entirely new suite of products and technologies. Under development for two years according to the company, the platform features a combination of digital audio processors, amplifiers, graphical user interfaces, programming software, and controls, and is engineered to enable the utilization of touchless control, automation, and artificial intelligence. The components were designed to work together to minimize training time and installation, configuration, and programming for AV integrators, and to reduce the learning curve for end users. So just how well does it all work?

Setting it Up

I evaluated the Atmosphere platform utilizing the digital audio processor, multi-channel amplifier, and several of the accessories. In terms of design, it is refreshing to see a simplified product offering that meets 90 percent of the multi-zone use cases with only a few models and accessories. The flexibility and features offered at a reasonable price point make the Atmosphere lineup appealing to systems designers. The installation is also simplified for quick turnaround on a jobsite.

At the heart of the system is the digital audio processor. Available in four- or eight-zone configurations, each includes all the necessary features and functions of a modern DSP. The web interface makes setup and system audio equalization a breeze. Preloaded EQ presets for AtlasIED speakers are quick to set up and sound good. The control interface allows for a user-friendly experience. One of the best features of this particular DSP is the ambient noise compensation, which is impressive with its ability to adapt based on a changing environment. 

The amplifiers are available in both 400 and 800 watts and have an expandable Dante receiver card that will offer more than the four analog line inputs. The available power allows for versatile configurations of speakers in almost any type of environment.

There are a wide variety of accessories including inputs, control, and monitoring wall plates. These were all easy to set up and allowed for high-quality microphone inputs, accessory RCA input, or even Bluetooth. The Bluetooth had an impressive signal range, although as expected, it worked better on the latest generation iPhone versus an older iPad in our facility. This is based more on the generation of Bluetooth hardware in those specific devices. The ability of operators to quickly control volume and zones on the intuitive wall plate is both user friendly and simple to install.

I like that the DSP features an automated message player that is built in with scheduling ability; however, I did not test this feature. It does add more to this platform, setting it ahead of the competition.

Impressions

My favorite feature is the simple and personal control offered. The ability to control sources, messages, scenes, and more with any smartphone application (iPhone in my specific case) is quick and intuitive. The ability to add user-defined access takes it one step further.

I like the tagline used on the Atmosphere website: “Simple doesn’t have to be basic.” This is a true statement. I found the entire platform to be easy to use, install, and control, while offering a full-featured product line. An interoperable and wholistic system sets the entire Atmosphere platform and product offerings ahead of the competition. 

Overall, the platform sets a new standard for digital audio with multi-zone processors, amplifiers, and a variety of accessories. Touchless control, easy-to-program automation, and artificial intelligence offer integrators and users an effortless way to create the perfect Atmosphere for their business. It will be a great tool in any fully featured system design.

Jeremy Caldera, CTS-D, CTS-I is the senior vice president of audiovisual technology at Pearl Technology, is a member of the AVIXA University senior faculty, and a member of the 2021 SCN Hall of Fame.

The What: Focusrite has announced the availability of its new Red 8Line, a 58-input, 64-output, Thunderbolt 3 audio interface. 

The What Else: The Red 8Line features two Red Evolution mic pres, independent L/R monitor outs, eight balanced line inputs and outputs on DB25 connectors; 32 channels of Dante I/O; 16-channel ADAT input and output (at 48kHz), and stereo S/PDIF input and output. As with other Red interfaces, it has 24-bit/192kHz A-D and D-A conversion for no-compromise recording and audio playback. With 119dB dynamic range A-D and 121dB dynamic range D-A (A-Weighted), the converters have been selected and calibrated by Focusrite to balance sound quality, dynamic range, and conversion latency.

Related: Focusrite Launches ISA 828 MkII Preamp

Red 8Line features the latest implementation of Thunderbolt 3, with Focusrite's ultra-low-latency Thunderbolt drivers. This speed provides the flexibility to use DAW plugins in real time, so you can use your DAW to track through plug-ins and record virtual instruments. With two Thunderbolt 3 ports as standard, it's also possible to daisy-chain Thunderbolt devices, such as third-party plug-in accelerators, hard drives, external 4K displays, processing units, and Thunderbolt hubs.

On-board Dante connectivity enables Red 8Line to communicate with industry-standard audio-over-IP equipment from hundreds of manufacturers. Dante enables system expansion using Focusrite RedNet interfaces or devices from third parties such as Neve, Yamaha, and DiGiCo. Red 8Line has built-in monitor control for mono, stereo, and multi-channel monitoring environments up to 7.1.2. Compatibility with the new Focusrite RedNet R1 desktop remote controller allows even greater flexibility, and customized presets for immersive monitoring environments including Dolby Atmos.

The Bottom Line: The Focusrite Red range is well suited for professionals who work in Pro Tools | HD and other DAWs, on both Mac and PC. With both mini-DigiLink and Thunderbolt 3 connections, it enables instant switching between Pro Tools and other DAW applications without the need to reconfigure your audio settings. Users just change the host mode either in RedNet Control software, or using the Red 8Line front panel, to change DAWs in seconds.

It’s going to take some doing to get people back out to entertainment venues once the pandemic is behind us. It’ll require an enticing experience they can’t get at home—something like a captivating immersion in spatial sound.

To help venues deliver this, Meyer Sound released Spacemap Go, an iPadOS app that offers a revolutionary new way of experimenting with sound. Available through a free app download and firmware update on the 4,000-odd GALAXY systems already installed in the field, the system can be controlled by a single iPad or with multiple iPads—enabling the creation of on-the-fly and pre-programmed sound shows by operators with nearly any technical background.

Related: Audio's Next Frontier: 3D Soundscapes

To get a better idea of just how powerful and intuitive the tool is, I dropped by the company’s demo setup in a rehearsal room at Jazz at Lincoln Center in New York. I walked into a space decked out with speakers—24 lining the perimeter at ear level and four hanging overhead—mounted on trusses, providing an enveloping bubble of audio. In the center of the room was an iPad on a stand, connected to a mixer controlling two GALAXY 816 processors. 

The demo began with a pre-programmed sequence using sound effects designed by Tony Award-winning sound designer Leon Rothenberg for a theatrical performance of Dracula. The Spacemap Go software on the iPad displayed eight panels corresponding to the different tracks in the production, each showing an overhead map of the speaker layout and the location of its respective source. For tracks that moved about the room, the path between speakers was clearly drawn: a “moody wind” whistled around the space in a neat circle, and Dracula’s “heavy breathing” arced back and forth across the speakers on the front half of the room. Upon pressing play, the source dots glided about on their paths as the sound enveloped my ears. 

To show off the real-time operation of the system, the Meyer Sound team queued up a vignette of a battle scene. “War ambience” tracks were set on a loop at opposite corners of the room, and a helicopter was placed overhead. I was invited to experiment with the movement of the chopper: Using a stylus I dragged the source about the Spacemap, resulting in the instantaneous movement of the sound. I dragged it toward the front of the room, and the sound hovered at a spot between the lower ring of the speakers and the height channels. The process was incredibly satisfying and couldn’t have been more intuitive.

According to John Monitto, director of business development and sales manager at Meyer Sound, Spacemap Go allows sound designers to create dynamic immersive environments with GALAXY processors that previously would have required significantly greater investment with the company’s D-Mitri processors—plus a lot of technical programming. Positioned for use in theatre, concert halls, houses of worship, and specialized settings like theme parks and planetariums, Spacemap Go gives operators of nearly any ability the tools to shape sensational aural environments with very little training. Further, while the system is capable of working with as many as 32 inputs, it can be employed in setups with as few as just two , and between eight and 256 outputs—creating possibilities for inventive sound design for nearly any application.

Further Details

Per Meyer Sound, Spacemap Go offers compatibility with popular sound design and show control programs such as QLab. For example, a complex sound design can be assembled using QLab in the studio, perhaps initially combining all multichannel cues into a stereo rough mix for reference. When on site, the tracks can be quickly expanded into a multichannel spatial mix using Spacemap Go’s templates for common multichannel configurations. The templates can be customized for a particular setup, while panning trajectories can be created with the touch of a finger, and if desired, captured for playback during performance. Although spatial trajectories can be created, edited, and stored using the graphical interface, the detailed matrix remains immediately accessible for reference or direct data entry.

In addition to QLab, Spacemap Go can be automated by popular DAWs such as Ableton Live, Logic Pro X, Pro Tools, Digital Performer, and Reaper, taking advantage of the iPad’s multi-touch interface for effortless spatial automation. GALAXY now supports the RTTrPM protocol, allowing integration with real-time tracking systems such as BlackTrax.

Systems are configured by Spacemap Go using any combination of GALAXY 408 or GALAXY 816 using Milan AVB, AES3, or analog as inputs and outputs. Spacemap Systems can be configured for up to 32 inputs and as many outputs as are provided by the processors. Each GALAXY provides up to 232 matrix crosspoints. Multiple iPads may be used to control a system, providing immediate access to additional channels or allowing simultaneous access to different screen views.

Spacemap Go is available as a free download from the Apple App Store. Spacemap Go is compatible with any iPad capable of running iPadOS version 13 or later.

The What: QSC has released the multipurpose NV-32-H (Core Capable) native network video endpoint for the Q-SYS Ecosystem, which allows users to toggle between “Core Mode,” which transforms the device into a fully capable Q-SYS processor with local HDMI switching capabilities, and its existing “Peripheral Mode,” which is designed for network HDMI video distribution over standard AV networks.

The What Else: The new Core Mode on the NV-32-H offers a fully featured audio processing engine for the Q-SYS Ecosystem, including 32 x 32 network audio channels, eight AEC channels, one VoIP softphone instance, and optional support for Software-based Dante. As with the rest of the Q-SYS processing portfolio, it also features a robust software-based control engine allowing for simple control integration for native Q-SYS devices as well as third-party connected devices.

Related: The Technology Manager's Guide to Audio in the New Era of UC

Additionally, as the new NV-32-H (Core Capable) is specially intended to support in-room video collaboration spaces, it also offers an integrated 3x2 video switch for locally connected HDMI sources. This feature represents time savings and cost efficiency for small-to-medium sized meeting rooms and classrooms that require easy toggling control between in-room inputs (PCs, BYOD devices, document cameras) and outputs (multiple displays or projectors) as it removes the need for a standalone switching device and costly integration requirements.

Peripheral Mode toggles the device into a multi-stream network video endpoint for HDMI video encoding/decoding for video distribution for the Q-SYS Ecosystem. This mode allows video distribution needs within larger collaboration spaces to strike a balance between quality, latency, and network efficiency.

“Our definition of meeting rooms has dramatically changed over the past nine months with an increased need for smaller rooms with collaboration at the forefront,” said Josh Arnold, senior product manager, Installed Systems, QSC. “We used this opportunity to reimagine the NV-32-H and develop a multipurpose, all-encompassing endpoint that gives integrators flexibility to meet specific room needs, while reducing hardware, especially for rooms where real estate comes at an absolute premium, and simplify integration and optimize system costs.”

“We have asserted from the start that Q-SYS is merely software,” said Mike Brandes, product manager, Q-SYS Video, QSC. “The dual personality of the new NV-32-H is proof positive of this belief. Our trust in industry-standard Intel processing combined with our ability to innovate at the software level has allowed us to infuse a new level of flexibility and scalability of Q-SYS, particularly to meet the rapidly changing needs of the collaboration space.”

The Bottom Line: In either Core Mode or Peripheral Mode, the NV-32-H (Core Capable) offers simplified integration for web conferencing applications like Microsoft Teams Rooms, Zoom and Google Hangouts Meet. This devices act as a plug-and-play access point for USB delivery of Q-SYS audio and conference room cameras.

Dan Dugan, the inventor of the automatic microphone mixer, will be awarded an Emmy at the upcoming 72nd Engineering Emmy Awards. His award, for “Outstanding Achievement in Engineering Development,” honors his invention of gain sharing automixing technology and will be presented in a live-streamed ceremony on Thursday, October 29.

In a press release issued earlier today, the Television Academy said “Engineering Emmys are presented to an individual, company or organization for developments in engineering that are either so extensive an improvement on existing methods or so innovative in nature that they materially affect the production, recording, transmission, or reception of television.” Nine companies and five individuals are being recognized this year.

Related: AV Technology Names Best of Show Special Edition Winners

“The idea for automatic mixing technology came to me about 50 years ago when I was a sound designer for live theater,” Dugan said. “While watching an operator struggle to manage a large number of microphones, I thought ‘there has to be a better way.’ It took a few years of subsequent research and design work in my lab, but the gain sharing automixer was the result.” 

The gain sharing technology invented by Dan Dugan enables everyone to be heard at the right time, and it does so much more quickly than a human operator could. Overall system gain is held at the level of one microphone, regardless of who is talking or how many people are trying to talk at once. The result is fast, transparent crossfades without upcutting, choppy sound, or shifts in background noise. Connection is simple: microphones are connected, channel gains are set, and the automixer takes care of following the action. The operator is then freed up to focus on the quality of the mix rather than trying to ride herd on multiple faders at once.

Dugan hardware products, which patch into existing sound consoles, and the licensed Dugan algorithm in other manufacturers’ products (such as those made by Sound Devices, Yamaha, and Waves) improve audio quality in broadcast news panels, sports commentary, talk shows, corporate meetings, political debates, distance learning, live theater, and other applications. 

“It’s such an amazing honor for me to receive an Emmy for my invention,” Dugan said. “This recognition is, without a doubt, one of the highlights of my life. I’ll be thanking a lot of people in my acceptance speech for their unflagging support and inspiration.”

The 72nd Engineering Emmy Awards will be presented on Emmys.com Thursday, October 29, at 5:00 PM PDT.

It’s becoming more common for external EQ and mixers to be replaced by a versatile integrated amplifier with audio processing. We asked four industry experts from different companies to share the benefits of these simple yet robust amplifiers and describe some applications where they will make a difference.

Noel Larson, Vice President of Marketing and Business Development, Ashly Audio

The benefits of integrated products include needing less space to install, a faster installation overall, and a reduction of cabling. At Ashly, we look to also provide you with better control and easier-to-use software. For example, with our new AquaControl software and the mXa-1502 integrated mixer amp, you can connect remotely to the box and verify the system from your office. No need to roll a truck if the bartender erased a preset. Just fix it from your office.

The most significant benefit you will see in the investment of an integrated amp will be in remote view and diagnostic capabilities. The ability to sell service contracts with the installation delivers customer peace of mind, and a recurring revenue stream.

By implementing integrated solutions, you not only reduce the space, time, and cabling associated with an install, but you’re also lowering the barrier of entry for pro solutions.

[The Evolution of Ashly Audio]

At Ashly Audio, we can’t see any installation being less productive after installing integrated amplifiers. This is especially true as solutions of all sizes are now coming to market. The mXa-1502 brings this functionality to a whole new customer.

John Minnick, Vice President of Engineering, Bogen Communications

In most applications where the sound reinforcement requirements call for no more than four to six input channels, the audio can be easily mixed and properly equalized using an integrated amplifier.

Rack space and total solution cost are the top two benefits of an amplifier with built-in audio processing. In a smaller venue, rack space is always at a premium. Bogen’s portfolio of integrated public address mixer amplifiers covers the gamut of price points that fulfill a broad range of power output and audio processing requirements.

In every application, you will always achieve better sound reinforcement results if you have audio processing—at a minimum, you should have parametric equalization. Here at Bogen, we’re looking forward to launching our Platinum Series, which will be available in early Q4, which features a 5-band parametric equalizer in addition to a suite of Bogen’s traditional audio enhancements that fulfill the requirements of any legacy application.

Mark Meding, CEO of Magnum Power Security, Exclusive U.S. Distributor of Deva

The simplicity of operation is increasingly important to the end user community. With integrated amplification technology being so reliable, there will be fewer failure points in the audio chain when using external devices. Also, by using fewer components, you can use less space and minimize power consumption.

Integrators can consider amplifiers with built-in audio processing because the programming process is less cumbersome, and the user operation is typically easier. Further, having one unit with built-in processing means a slimmer form factor.

[Audio's Next Frontier: 3D Soundscapes]

The additional benefits I see are the integrated solution becoming a smarter device with processing capability becoming more robust and easier to use, and higher efficiency. For the end user, it means more efficient cabling and connections and lower power draw.

Integrated amplifiers will be the most useful in hospitality, including restaurants, bars, and hotels. Outdoor areas such as zoos, communal city spaces, and amusement parks where an integrated solution may be required are a great fit for Deva.

Paul Zielie, Product Development Lead, SAVI Controls

The primary benefits are the consolidation of multiple components into a single box, reducing the complexity and the cost of installation and maintenance.

The benefits depend on the processing available in the amplifier. In addition to the baseline DSP functions of EQ on the amplifier channels, more advanced functions increase the benefits.

Matrix mixing, where any input can be routed to one or more outputs, allows common program sources to be routed to multiple zones. Matrix mixing can also be used in paging, either through automated DSP features or external control.

Audio delays on sources can facilitate lip sync where video is being distributed as IPTV and the audio takes a non-IPTV path. An analog or audio-only signal will often lead an IPTV signal by up to hundreds of milliseconds.

Network audio inputs and outputs (Dante, AES67, etc.) allow the sharing of program and paging sources across multiple amplifiers, each installed near where they are needed.

[SCN Hall of Fame 2020: Paul Zielie]

The ability to subscribe to “regular” streaming media such as IPTV audio streams allows the integrator to support more potential sources without requiring an analog input for each source.

The ability to control and monitor the entire audio chain remotely will be the biggest long-term benefit of an integrated DSP/amplifier.

The advantage of integrated DSP grows with the number of amplifier channels. In retail and hospitality settings, the ability to route program material to multiple individual zones, each with their own amplifier(s), becomes much easier with an integrated DSP.

The What: QSC has announced the addition of two smaller-sized AV and control processors to the Q-SYS Ecosystem: Q-SYS Core 8 Flex and Q-SYS Core Nano. 

The What Else: Driven by the proliferation of networked endpoints and the expanding processing needs of a wider variety of spaces and topologies, the Core 8 Flex includes onboard analog audio I/O and GPIO plus network I/O while the Q-SYS Core Nano offers network-only audio I/O processing and control. Built on the same flexible and scalable software foundation as the rest of the Q-SYS processor portfolio, including the best-in-class Q-SYS Core 110f, these new Cores expand design options to meet a wider variety of installations.

Related: QSC Joins Microsoft Teams Device Certification Program

The Q-SYS Core 8 Flex includes a 64x64 networked I/O channel capacity with eight onboard FLEX audio channels and eight GPIOs to integrate analog audio and control devices into the Q-SYS Ecosystem. Q-SYS Core Nano offers the same 64x64 networked audio I/O without the onboard analog I/O to support installations with smaller spaces with centralized processing and fully networked endpoints. Both of these new Q-SYS Core processors occupy a smaller half-width, 1RU footprint and include pre-installed 8x8 Software-based Dante audio channels (license upgradeable up to 32x32 channels), driverless USB audio, and AV bridging capabilities. 

“These new processors answer the call of smaller spaces that require fewer I/O as well as the increased popularity of fully networked peripherals with a solution that stays true to the foundation of our platform, delivering the rock-solid processing power and extensibility you have come to expect from QSC,” said Trent Wagner, audio product manager, QSC. “The Core 8 Flex and Core Nano rightsize the onboard I/O and physical footprint without compromising on features, performance, or flexibility.”

“The introduction of these two new Core processors brings even more design flexibility to the Q-SYS Ecosystem, delivering on the promise of a standards-based software platform that can easily scale up or scale out,” said Jatan Shah, executive vice president, chief operating office and technology officer, QSC. “I’m proud of the innovation and engineering strength of our research and development teams, who continue to deliver solutions that meet the evolving needs of the markets we serve, while reducing complexity and cost for our partners and customers.” 

The Bottom Line: These smaller Core processors utilize the same software-based architecture, control engine, and design software suite (Q-SYS Designer Software) as the rest of the Q-SYS Core processing portfolio. They both offer scalable DSP processing, video routing and bridging for web conferencing, as well as third-party endpoint integration without the need for separate dedicated control processors. 

The What: Extron has begun shipping the HAE 100 4K Plus audio de-embedder. 

The What Else: The HAE 100 4K Plus extracts audio from a 4K HDMI source and provides outputs for stereo and dual mono analog audio signals, as well as digital audio multi-channel or two-channel formats. It is HDCP compliant to ensure display of content-protected media and support data rates up to 18Gbps, HDR, up to 12-bit Deep Color, 3D, and HD lossless audio. 

Related: Extron Shipping NetPA Ultra Dante Amplifier with DSP 

"HDMI audio embedders and de-embedders are often critical tools in AV system designs,” said Casey Hall, vice president of sales and marketing at Extron. "Offering 18 Gbps performance, the HAE 100 4K Plus offers the integrator friendly features and performance to meet today's AV system requirements."

The Bottom Line: The HAE 100 4K Plus is equipped with integrator-friendly features such as EDID Minder, HDMI input cable equalization, comprehensive LED status display, and a compact enclosure.

The What: Extron has begun shipping the XMP 240 Expansion Matrix Processor, featuring an extensive mix matrix with 24 channels of AEC (Acoustic Echo Cancellation) and 48x48 Dante connectivity in a half-rack space. The XMP 240 can also be used standalone for matrix processing in an all network audio system. 

The What Else: When connected to a DMP Plus Series processor via Dante or the EXP expansion port, a complete system is created that offers up to 36 channels of AEC, a USB audio interface, analog connectivity, and optional VoIP. A system expanded with an XMP 240 is capable of supporting multiple beamforming microphone arrays alongside numerous other Dante sources, with enough outputs to support multiple zones and destination devices. Designed for network audio systems based on Dante or AES67, the XMP 240 allows for high-channel-count audio input and output processing.

Related: The Technology Manager's Guide to Networked Audio 

"The ability to add 24 channels of AEC via a Dante network to a DMP based system gives our customers powerful flexibility and scalability for larger systems,” said Casey Hall, vice president of sales and marketing at Extron. "This much processing power is unprecedented in a half-rack device that can be placed under a table or virtually anywhere requirements dictate."

The Bottom Line: With connectivity for up to 48 sources and 48 destinations via Dante or AES67, the XMP 240 is well suited for applications that utilize many network audio devices. Dante Domain Manager support provides user authentication, role-based security, and expansion of Dante systems over any network infrastructure.

Propelled by Netflix’s deep dive into Mexico’s media content industry, Cinematic Media’s new post-production facility, part of GGM Estudios and located just north of Mexico City, is poised to contribute to the growth of Spanish-language prestige content. When completed by the end of 2019, the facility will house eight stages/cutting rooms, all certified for the Dolby Atmos broadcast immersive-audio format. Keeping the workflow manageable at a high pace will be components from Focusrite’s RedNet range of Dante-networked audio converters and interfaces.

When finished, the audio studios will have a complement of six RedNet HD32R 32-channel HD Dante network bridges (one per stage), 22 Red 4Pre 58-In / 64-Out Thunderbolt 2 and Pro Tools | HD compatible audio interfaces (three per Stage plus Edit Suites), six RedNet AM2 stereo audio monitoring units, and six RedNet X2P 2x2 Dante audio interfaces.

“The work is already here, so we had to get started as quickly as possible,” said Martin Hernandez, the sound supervisor and designer at Cinematic Media (and a two-time Academy Award-nominated supervising sound editor), adding that the new facility’s location in a commercial area about an hour north of Mexico City offered them the room needed to construct a large enough facility to address Mexico’s burgeoning media business. 

Six stages will be open by the end of the year, plus the editorial suites. Hernandez said that the RedNet interfaces provided Cinematic Media’s facility with the technology needed for a fast start to a reliable workflow. The equipment actually came before the space was done, set up in temporary studios even as the facility itself was being built. The efficiency afforded by RedNet technology meant that they could immediately begin work as soon as equipment was installed. A speedy workflow was especially important in light of the fact that mixes would have to be transported to outsourced mixing stages temporarily while the new ones are built.

“We are cutting all sound to picture in this temporary space facility, then taking those tracks to a mixing stage for the client to listen to, then back here for the pre-mastering and final mixes,” Hernandez said. “It’s rather hectic to have to travel so much, but it’s the only way to keep up with the workload while we wait for the soundstages to be built here. But the RedNet interfaces let us meet deadlines that we might not otherwise be able to. The RedNet units are reliable and easy to use, and they are truly plug-and-play. The Red 4Pre and RedNet HD32R interfaces connect directly to the three Pro Tools HDX systems we have in each mix studio, so we have a fully Dante-enabled network for audio between all of the rooms. The speed and ease of the workflow is amazing.” 

And so, Hernandez, added, is the sound from the Focusrite interfaces. “RedNet sounds fantastic, and combined with the efficiencies it brings us, it was the right choice for what will be Mexico’s leading post production facility. No other system could have provided the kinds of capability RedNet does at that price. It is an excellent investment.”

Broadcast audio engineer Jeff Cohen (the Super Bowl Pre/Post, Orange Bowl Halftime, NCAA Basketball) chooses Waves plugins and Waves SoundGrid for NFL broadcasts. Nine-time Emmy Award winner Jeff Cohen is owner and senior engineer at DigiMAXX Productions and has over 30 years of experience designing and mixing audio for live events and broadcast productions. He remarks, “I mix an NFL game every Sunday, starting with a preseason game and in alternating years, one divisional playoff game. This year we will go until week 17. Most of the Waves equipment in the truck is owned and rented to FOX by DigiMAXX Productions, and we use Waves in two different ways on our NFL show. One is for post-production and includes ProTools, which is used for editing packages built by our onsite video editor, and the other is for real-time, on-air mixing.” 

The first production system setup includes ProTools 12.4 running on a 15-inch MacBook Pro, Waves SoundGrid Mobile Server, Waves Mercury bundle, Waves DTS Neural Surround Collection, a Dante Virtual Soundcard, and a Focusrite RedNet 6 Dante to MADI converter. The second setup is a Lenovo ThinkCentre Windows 10 Pro PC, Waves SoundGrid Extreme Server, Waves Mercury bundle, Waves DTS Bundle, Waves SSL 4000 Collection, the Dugan Automixer and a Calrec I/O card in the Artemis Beam Console.

The setup also includes an additional Lenovo PC running Spot-On music software and an additional MacBook Pro running other music and sound effects from Soundboard. The Mac also runs Dante Virtual Soundcard and the Lenovo runs AES through a Digigram Cancun. Additionally, there is a Windows computer with the Calrec, which also runs RTS ADAM AZedit and IPEdit software for communications, as well as the Calrec Artemis operating system. The broadcast truck features Neumann KH-120 monitors with a Genelec 7050 Sub. There are also additional speakers for PFL, hot mics, and overhead speakers. “It was built to be ready for immersive monitoring,” Cohen said.

The truck is running multiple Dante networks, a Calrec Hydra 2 network, and a Waves SoundGrid network. One Dante network is with the truck and runs all of the communications using RTS Omneo and Studio Technologies communications devices as well as microphones connected directly to the Calrec though a Dante I/O card. The team uses Studio Technologies’ Dante Announce Consoles and OnAir Belt Packs for on-air talent along with Shure Axient RF for the field reporter, as well as six RF Klover Parabolic microphones with Neumann KM 183 and 184 microphones. Other field mics include Sennheiser 416s, 8070s, 816s, and MKH70s, along with Shure VP88s for the crowd. The second Dante Network includes the ProTools computer, Mac Music Computer, the RedNet 6 (connected to the Calrec MADI I/O) and a Calrec Brio effects submix console. The Brio is also connected through the Calrec Hydra2 Network.

For the announcers, Cohen and his team uses Waves SoundGrid on individual inserts as well as group inserts. For each announcer, they use the Waves Scheps Omni Channel, F6 Floating-Band Dynamic EQ, and then the SSL E-Channel. “I prefer to use the dynamic EQs for the announcers,” Cohen said. “In sports broadcasting, with announcers using the HMD 27 headsets, mic placement is very important to the sound quality. While at AES, I heard Andrew Scheps speak and do a demo with his Omni Channel plugin. He was describing using the de-esser in a way other than on the high end. He spoke of how it was purposefully full range and not limited to just the high end. I immediately put this into effect to reduce the excessive low mids, but only when above a threshold. If you use a standard EQ, you are removing the frequency no matter what. The use of dynamic EQs and the de-esser dramatically improves the audio quality in a very dynamic way. When something is out of bounds, it is adjusted.”

The team also uses the Waves Dugan Automixer on all of the announcers to minimize background noise coming in from the mics. The IFB (Interruptible Fold-Back) mixes are preprocessing and the inserts go to air. On the Announcer subgroup, they use a Waves SSL G- Master Buss Compressor to hold things in line. “The quality of this buss compressor is so good and a great addition to the signal chain,” Cohen said. “This is the final stage of maintaining announcer levels and delivering -24 LKFS (loudness, K-weighted, relative to full scale) levels in a live environment. We also use the Waves WNS Noise Suppressor to try and remove a bit of the stadium ambiance as well.” 

For the LFE, Cohen and his team uses Waves’ Submarine for Subwoofer enhancement. “We used to use a DBX 120a Subharmonic Synth, but it was not available to us any longer,” he said. “I had to look for an alternative. Before Submarine came out, we used Waves LoAir and experimented with Renaissance Bass and MaxxBass. That worked OK, but Submarine is delivering exactly what we were looking for. We use it to add subharmonics to graphics, music and other enhancements. I am very impressed with the effect this has on our final product and the quality of the LFE.”

The team also uses the Waves WNS Noise Processor on the effects submix to try to remove some of the stadium ambience. “We look to get as much of the field sounds as possible,” Cohen said. “Hits, whistles, and other sounds from the field bring the viewer into the stadium and closer to the action. We also fight with the PA. It is excessively loud in some stadiums now, and anything we can do to bring out the field sounds is highly desirable.” 

In the ProTools world, the team uses the DTS Neural Surround Collection up mixers to create a 5.1 mix from stereo elements. “This year is the first year we have gone to 5.1 for most tape elements and archival,” Cohen said. “So previous elements as well as the music and sound effects need to be upmixed to the 5.1 environment. If we stay in this environment, the quality of the surround product, delivered to the home, is much better and less of a distraction.

“Using Waves has allowed me to use outboard processing that I do not have with the console by itself. It provides the ability to use devices that, in the past, had to be hardware devices. For example, we used to have an analog hardwired version of the Dugan on our show; now it is all software. Having all of these tools at my fingertips allows me to create a mix and solve mix related issues without the addition of external hardware. We now have all of the tools to create a consistent, high quality mix, bringing additional intelligibility and defined field sounds while creating the 5.1 surround canvas all in the box.

The What: Biamp's TesiraFORTÉ is now Certified for Microsoft Teams, delivering Biamp's professional audio quality to Microsoft Teams Rooms in medium and large conferencing spaces. 

The What Else: Tesira DSPs use Biamp's advanced Acoustic Echo Cancellation (AEC) technology to eliminate common sources of echo during videoconferencing in medium-to-large collaboration spaces. Tesira also supports bidirectional mute synchronization to share mute or unmute commands with Microsoft Teams Rooms. This means the microphones accurately display mute status (red or green) and pass mute commands to the overall system with the push of a button.

"Achieving Microsoft Teams Rooms certification uniquely positions Biamp to serve the needs of our customers," said Joe Andrulis, executive vice president, corporate development, Biamp. "As a certified DSP, Tesira is easy to deploy, delivering professional-grade AV faster and more efficiently for enterprises using Microsoft Teams Rooms in medium and large meeting spaces."

"Having the Biamp TesiraFORTÉ DSP now be Certified for Microsoft Teams is an important milestone for Microsoft Teams Rooms," said Ilya Bukshteyn, partner director, microsoft teams devices. "Certified DSPs enable Microsoft Teams Rooms to deliver superior audio experiences, compatibility, and reliability in a broad range of rooms and complex environments, including customized environments such as executive boardrooms and multipurpose rooms."

The Bottom Line: The Biamp products certified for use with Microsoft Teams Rooms include the TesiraFORTÉ AVB VT4 conferencing solution and the Tesira EX-UBT expander when paired with Sennheiser TCC2 microphones. Additional solutions are currently in process to be certified for Microsoft Teams Rooms.

The What:  AudioScience, designer and manufacturer of DSP-based digital audio peripherals for the original equipment manufacturer (OEM), broadcast, installed sound, and entertainment industries, has upgraded the functionality of its Iyo Dante family of audio-over-IP interfaces with the release of a Crestron Module.

The What Else:  With the new module, any Crestron programmer can create custom applications that fully integrate the Iyo Dante in any Crestron ecosystem. Supported controls include audio metering and analog level/gain adjustments as well as microphone input phantom power.

"We are excited to introduce this new capability utilizing the Crestron Module for the Iyo Dante family as it will provide more installation options for the broad audio over IP marketplace,” said Nicole Santiago, AudioScience network product manager. “AudioScience recognized the market didn’t require more DSP boxes; the market needed more I/O at a cost-effective price point. AudioScience made it happen, while still providing a professional-level microphone and connector options for all of our integrators requirements.”

The Bottom Line:  Supporting Crestron's latest SIMPL# module format, the Crestron module allows complete control of an Iyo Dante over TCP/IP.

The What: TASCAM has introduced the Model 16 all-in-one mixing studio for live multi-track recording.

The What Else: The Model 16’s 14 inputs provide top panel access for microphones, line inputs, and instruments. Channels one through eight feature premium XLR mic inputs utilizing TASCAM’s Ultra-HDDA (High Definition Discrete Architecture) microphone preamplifiers with 48-volt Phantom power. This setup allows combinations of vocals, drums, miked instruments, and more to be connected along with direct output instruments such as keyboards, bass, or acoustic guitar, multi-effects, electronic drums, or any line sources.

Channels nine through 10 and 11-12 feature a single XLR mic input, each with stereo 1/4-inch line inputs. Channels 13-14 feature a stereo RCA and mini jack input. Or, alternatively an external Bluetooth device can be connected to the Model 16’s wireless channel for controlling streaming audio volume via the Channel 13-14 fader control.

The internal 16 track multi-track recorder records WAV files (up to 48kHz/24-bit) directly to SD, SDHC, and SDXC cards. Punch in/out is available for up to eight tracks simultaneously, a versatile feature for songwriters and solo artists alike.

The Model 16’s robust built-in USB Audio Interface can send/receive (16 in/14 out) audio data with up to 24-bit/48kHz recording quality—to and from a wide range of popular DAWs, including GarageBand and more. Via USB, external tracks can be summed to a single or stereo-paired Model 16 fader to control custom or stereo sub-mixes for drums, vocals, and more using one or two faders.

Using the onboard channel Mode switch, each channel (or stereo paired channels) can independently select the input source to fit the recording or mixing environment or setup. The input source for each channel can be selectable as live, internal multi-track recorder (MTR), or USB for increased flexibility in signal path processing and workflow.

Each channel strip features 60mm faders, a one-knob compressor for inputs one through eight, three-band EQ with a sweepable mid-range frequency and 100Hz Low-cut Filter (three-band only on channels nine through 14). Model 16 has a built-in effects processor with selectable and editable effects (reverb, delay, chorus, flange).

Three aux outputs are offered on each channel for flexibility in monitoring and effects routing—MON 1 [pre], MON 2 [pre/post], and FX [post]. Selectable mute, MAIN/SUB, and PFL switches are available on each input channel for versatile audio and signal routing options during recording or mixing.

Hi/lo shelf and wide/narrow-Q semi-parametric EQs (for main and monitor outputs) provide additional tweaking options for dialing in the perfect final mix. MAIN output via XLR balanced and SUB output via TRS balanced connectors, 12-segment L-R paired LED Output Level Meters and a Stereo Headphone Output round out the long list of versatile recording and mixing features and functions.  The unit can also be rack mounted with an optional rack mount kit.

The Bottom Line: Designed to combine the ease and feel of an analog mixer but with the recording quality of a modern digital recorder, the Model 16 is a full-featured compact mixing console that fits comfortably in a rehearsal studio, house of worship, home studio, or any production environment where recording quality is paramount, and space is limited.

The What: Symetrix has released Composer 7.2 for Windows, the latest version of its programming software for Edge, Radius NX, Prism, and Solus NX DSPs. Composer enables integrators to program a complete end-to-end DSP signal path using just one application.

The What Else: Version 7.2 adds several important features. The new Emailer Module enables sending email notification directly from the DSP hardware, triggered by any event in the DSP design. By using commonly available email-to-text gateways, SMS text messages may also be sent.

Amplifier monitoring and control has been added for the Powersoft Duecanali, Quattrocanali, Ottocanali, X4, and X8 amplifiers, and for Panasonic WP-DM9XX amps. This enables Powersoft amplifiers to communicate current, alarms, thermal, and impedance stats directly to the DSP for real-time display on portable and fixed devices.

The number of simultaneous TCP/IP connections Control Server can host has been dramatically increased from 6 to 1,000, allowing far more simultaneous users and control interfaces.

A new right-click Change Unit Type option has been added that transforms a site file DSP into any other Symetrix DSP hardware type, making it easy to convert a design from, for example, a Radius 12x8 EX to a Radius NX. In-place copy transfers all processing, control assignments, and SymVue control screens that are not connected to physical device I/O. This makes every design file a viable template.

Composer 7.2 adds a host of other features. The new UpDown fader type changes any fader images to create user-friendly touch controls for incrementing or decrementing a parameter. You can edit text in place for buttons, label, and text objects, which is faster and more intuitive than using the Properties pane. A hot key enables changing font size when designing control screens. A reorganized Toolkit in Site View lines up audio, network, and control tools logically by type. In matrix selectors, the display of the selected input now shows the input name instead of the channel number.

The Bottom Line: Composer for Windows handles end-to-end configuration of Radius, Prism, Edge, and Solus NX DSPs, including creation of customized GUIs, scheduling, presets, logic, security, and integration of Symetrix and third-party control solutions. A complete end-to-end DSP signal path can be designed using this single app.

The What: TASCAM has unveiled the new SERIES 8p Dyna 8-channel mic preamplifier with analog compressor, the company’s first standalone mic preamp and A-D expansion device. 

The What Else: TASCAM has long been known for its ultra-low-noise preamps, and eight TASCAM HDIA (High Definition Instrumentation Architecture) preamps are its best yet. They feature noise level (EIN) of -127dBu, THD of 0.0007 percent, dynamic range of 116dB, and crosstalk of 119dB. Each SERIES 8p Dyna channel also features a single-control analog compressor with true bypass for handling hot input signals and creating smooth audio input. True bypass is available per input.

With two instrument/microphone combo inputs on the front panel, users can quickly plug in mics, instruments, and more. Six balanced mic/line inputs grace the rear. Every microphone input has individually switched +48V phantom power for use with condenser mics.

Dual S/MUX optical outputs on the rear of the SERIES 8p Dyna enable use of the unit as a front-end expander for S/MUX or ADAT-equipped audio interfaces, such as TASCAM’s SERIES 102i and 208i, and US-20X20. Using the two S/MUX outputs simultaneously, users can send all eight SERIES 8p Dyna channels at up to 96kHz (four channels at 176.4 or 192kHz).

For additional versatility, the SERIES 8p Dyna’s eight line-level analog channel outputs are provided on both 1/4-inch balanced TRS connectors and a balanced 25-pin D-Sub connector, so users can route signals to interfaces, recording mixers, front-of-house consoles, and more.

Word clock in and out BNC connectors with an In/Out/Thru switch enable proper synchronization of digital signals. Front-panel analog overload indicators, compression indicators, and digital peak and level meter bars enable monitoring the unit’s performance. Connect its D-Sub output to a TASCAM ML-16D or ML-32D Analog Dante converter, and it becomes a front-end for an audio-over-IP network.

The Bottom Line: Featuring versatile connectivity and redesigned microphone preamp circuitry, the TASCAM SERIES 8p Dyna is designed for both studio and stage use. Housed in a rugged 1U rack-mount casing, its dual S/MUX optical ports make it well suited as an expansion unit for a TASCAM SERIES 102i or 208i, US-20X20, or other S/MUX-equipped audio interfaces including ADAT, or it can be used as a standalone front end.

The What: Extron has introduced the DMP 64 Plus series of compact Digital Matrix Processors, featuring six DSP input channels with AEC, in a half rack space. 

The What Else: The AT models can be powered by PoE+ and provide 32x16 Dante connectivity for Dante microphone arrays and other Dante devices, while the V models include up to eight VoIP lines for collaboration applications. Featuring an ACP bus for audio control panels, an expansion port for bidirectional digital audio connection with a DTP CrossPoint or another DMP, eight audio file players, and configurable macros, the DMP 64 Plus can be used in a wide variety of smaller systems.

"The smaller size of many collaboration spaces often means fewer inputs are needed while retaining key features like AEC and multiple control options,” said Casey Hall, vice president of sales and marketing. "The DMP 64 Plus series can be powered by PoE+ for installation simplicity and offers Dante and VoIP models to meet any connectivity need."

All six main DSP channels of the DMP 64 Plus offer FlexInput source selection to route any Dante input, expansion input, or one of the six mic/line inputs, to a main DSP channel. This enables the DMP 64 Plus to process audio from any source with the full range of DSP capabilities, including AEC.

The Bottom Line: With a flexible mix matrix, a USB audio interface, and 6x4 analog connectivity, the DMP 64 Plus provides the ability to easily connect and route signals to accommodate system requirements.

The What: QSC has introduced the firmware v2.0 update to its TouchMix-30 Pro compact digital mixer. The firmware v2.0 aims to further satisfy the needs of demanding professionals with a comprehensive feature enhancements.

The What Else: Improvements include new custom fader banks function supporting three fader banks assignment, each for eight channels, that are saved as part of a Mixer Scene; the capability to interface with third-party external control surfaces with motorized faders (QSC has verified operation with iCon Platform M+, Behringer XTouch Compact, and PreSonus FaderPort 8 control surfaces); 28 new instruments presets (cajon, accordion, harmonica, ukulele, cello, clarinet, various percussions and more). In addition, CP Series Compact Powered Loudspeakers Presets are available directly in the mixer as well as CP8 and CP12 loudspeakers’ input gain settings recommendations.

“Listening and responding to the needs of our customers has always been a priority for QSC,” said Jon Graves, QSC mixer product manager. “Already proven as an incredible tool to help users achieve great results, with this new firmware, the TouchMix-30 Pro is now far and away the absolute best-in-class small format digital mixer on the market today. We’re really proud of the effort and know our customers will appreciate the new functionalities.”

The Bottom Line: Well suited for production professionals, musicians, and bands as well as live performance venues, the TouchMix-30 Pro’s 32 mixing channels (24 mic/Line, 6 line, stereo USB) and 16 outputs offer signal management flexibility for a wide range of applications. Advanced features such as Class-A microphone preamplifiers, Anti-Feedback and Room Tuning Wizards, two independent 24- channels Automatic Microphone Mixers, two real-time analyzers (RTA), patch matrix, 32-channel direct- to-external-drive recording and DAW interface with Mac OS and Windows computers make the TouchMix-30 Pro equally competent as both a professional live sound or recording system.

In our series of #InfoCommInspo blogs, we're getting inspired by InfoComm 2019 exhibitors and learning what trends they expect to see at the show—plus a special sneak preview of what they'll show in their booths.

Q&A with Glenn Booth, director of marketing, Professional Audio, Yamaha Corporation of America

Q: What about InfoComm inspires you?

Glenn Booth: As InfoComm has grown in size and scope, it’s never lost that “small show” feel where you can actually have meaningful conversations, even in the middle of your booth, and that’s important to me as an exhibitor. InfoComm is a great opportunity to stay in touch with so many different aspects of our industry and really talk to our customers, dealers, and integrators. We can also meet with AV end users from a number of different markets and learn what they need to do their jobs better or differently, what challenges they face and talk to them about how we can help them. That’s exactly the type of direct feedback we need to keep our technology development on track and relevant.

Q: What is the most exciting trend you expect to see at the show?

GB: There’s so much focus throughout the commercial AV industry on some form of connected workflow, whether it’s unified communications, cloud collaboration or networking, or a combination of each. Interestingly, Yamaha is highlighting that same theme of connectivity in our exhibit, where many of our new products feature Dante audio network connectivity or our ProVisionaire software that allows control and monitoring over multiple devices in a system.

Q: How will your solutions inspire integrators?

GB: Integrators are dealing with more challenges than ever before: more elaborate AV system requirements, tight deadlines, and even tighter budgets. By making every piece of the audio chain work together and talk to each other—from consoles to amplifiers, to loudspeakers, to fixed install racks—we’re making the system integration and installation process easier and more efficient from initial design to final completion.

Q: How will your solutions inspire tech managers?

GB: Just like our integrator partners, our end user customers are dealing with challenges of their own. It may be a house of worship needing to spread their message to satellite churches internationally, or a university broadcasting a lecture to multiple campuses, or a stadium distributing content throughout a venue. The end user’s focus needs to be on their operations, and not on the AV technology that makes it all work. By designing our audio solutions to be as connected and intuitive as possible, we’re making it easier for integrators, and ultimately, the end users, to focus on expanding their creativity and getting the most of out of technology.

To learn more, visit Yamaha in Booth 6461 at InfoComm 2019. 

InfoComm is giving our readers a free Exhibit Hall pass to the show! Use VIP Code "FUTURE"  to redeem your free pass. Click here to register.

The What: Bose Professional is now shipping its four recently introduced ControlSpace engineered sound processors: ESP-880A, ESP-880AD, ESP-1240A, and ESP-1240AD [the “A” designation indicates Bose AmpLink connectivity, and the “AD” designation indicates both AmpLink and Dante connectivity]. 

The What Else: They feature analog audio I/O—mic and line-level—and a Bose AmpLink output to send digital audio to AmpLink-equipped Bose amplifiers. Engineered for precise performance, each includes advanced signal processing with 48kHz/24-bit audio conversion, low-latency, and ultra-low noise operation. In addition, the ESP-880AD and ESP-1240AD feature integrated Dante audio networking as well as a front-panel RJ-45 Ethernet connection for localized configuration and monitoring.

ControlSpace ESP processors can be configured using ControlSpace Designer software and its large set of signal processing modules. This includes automatic mic mixers, predictive feedback suppressors, room combiners, multiband graphic and parametric EQs, Bose loudspeaker libraries, signal generators, routers, mixers, AGCs, duckers, gates, compressors, source selectors, and delays. Each processor also supports integration with industry-standard control systems, such as Crestron and AMX.

The Bottom Line: ControlSpace ESP open-architecture DSPs offer enhanced flexibility for a wide variety of applications—from small, self-contained projects to large, networked systems. The processors are compatible with three new, user-friendly digital ControlCenter controllers: the CC-1D, CC-2D and CC-3D. They also work with the ControlSpace Remote app, which allows end users to easily adjust volume, mute, tone, source selection, and more from their mobile devices.

The What: ClearOne has announced the immediate market availability of its new CONVERGE Pro 2 48VTD that includes VoIP, Telco, and Dante interfaces in a single unit.

The What Else: The new CONVERGE Pro 2 48VTD, now shipping worldwide, functions the same as ClearOne’s existing V and T offerings providing a 10-watt amplifier, four configurable mic/line audio inputs, and eight configurable mic/line outputs. Additionally, all models are Avaya, Cisco, and ShoreTel compliant, and support built-in USB audio, and built-in Skype for Business along with interfaces for VoIP, Telco, and Dante.

The new VTD series works with all existing CONVERGE Pro 2 units and all existing P-Link peripheral products, including the ClearOne BMA CT, Beamforming Microphone Array 2, USB Expander, GPIO Expander and DIALOG 20 Wireless Microphones. All CONVERGE Pro 2 mixers feature advanced audio DSP algorithms, including acoustic echo cancellation, noise cancellation, feedback elimination, gain and level control, and microphone gating.

“Once again, ClearOne has reimagined its leading solutions to provide AV practitioners with optimal, cost-efficient solutions for every unique installation,” said Zee Hakimoglu, ClearOne chair and CEO. “By combining VoIP, Telco, and Dante capabilities in a single unit, integrators can reduce rack space requirements and deliver added functionality. The four-input CONVERGE Pro 2 48VTD can also lower costs when paired with our new P-Link-based BMA CT, allowing integrators to bring a low-cost Dante interface to the BMA CT.

The Bottom Line: The new 48VTD pairs with ClearOne’s BMA CT and allows integrators to simplify system designs, increasing installation flexibility and reducing costs. It can also reduce design uncertainty on projects where network connectivity may be unknown early in the project specification cycle.

The What: Bose Professional is expanding its ControlSpace EX range with two new processors optimized for conferencing rooms (EX-440C and EX-12AEC) and a high-powered, general-purpose digital signal processor (EX-1280).

The What Else: With an open-architecture, all-in-one design, the ControlSpace EX-440C conferencing signal processor facilitates high-quality microphone integration and audio processing for small-to-medium standalone meeting rooms. Various inputs and outputs allow for flexible configuration: four mic/line analog inputs, four analog outputs, onboard VoIP and PSTN, USB, Bose AmpLink output, eight-channel acoustic echo cancelling (AEC), and 16x16 Dante connectivity.

With an open-architecture design, 12 acoustic echo cancellers (AEC), 16x16 Dante connectivity, and flexible signal processing, the ControlSpace EX-12AEC conferencing signal processor provides a robust expansion of control options and capabilities tailored for conference rooms using the ControlSpace EX-1280C processor—well suited for larger conference rooms, boardrooms, and any space where Dante microphones are employed for conferencing.

With an open-architecture, single-rack-unit design, the ControlSpace EX-1280 is a robust digital signal processor equipped for general-purpose audio processing applications. Twelve mic/line analog inputs, eight analog outputs, Bose AmpLink out, and 64x64 Dante connectivity allow for flexible configuration and high-quality sound system control. The EX-1280 digital signal processor is designed for performance venues, places of worship, hospitality and resorts, and other spaces requiring intensive processing and scalability, offering increased functionality beyond the existing ControlSpace ESP engineered sound processor line.

The Bottom Line: ControlSpace Designer software simplifies the setup process for all three processors with intuitive drag-and-drop programming and conference-specific software tools that help reduce installation time and on-site errors.These new models are compatible with Bose Professional’s line of Dante endpoints and end-user controllers—including wall-mount and mobile device control using ControlSpace Remote.

The What: ClearOne has begun shipping COLLABORATE Live, a series of four video collaboration solutions designed to provide a complete video collaboration experience for any size room. 

The What Else: For large boardrooms, auditoriums, conference rooms, lecture halls, courtrooms, training centers, and telemedicine facilities, COLLABORATE Live 900 delivers a complete, professional-quality collaboration system solution featuring a combination of video components integrated with advanced audio DSP technology.

COLLABORATE 900 features a suite of built-in videoconferencing capabilities including a Skype for Business client, SIP/H.323 videoconferencing, wireless presentation, interactive whiteboard, and recording and streaming. It also includes a CONVERGE Pro 2 DSP mixer, advanced audio processor, and ClearOne’s Beamforming Microphone Array 2, which features adaptive steering technology to provide audio pickup coverage of an entire room. For high-quality video capture of all participants in the room, the COLLABORATE Live 900 comes with the UNITE 200 PTZ 1080p60 camera with 12x optical zoom. The system also includes a 90-day free trial of the ClearOne COLLABORATE Space cloud-based videoconferencing application.

COLLABORATE Live 600 is a video collaboration system that delivers clear, full-duplex audio for medium-sized conference room environments. It offers the same suite of built-in videoconferencing capabilities, the UNITE 200 PTZ camera, an ultra-friendly user interface, and a 90-day free COLLABORATE Space trial subscription. For audio, the system features two ClearOne CHAT 150 speakerphones that daisy chain with CHATAttach.

The COLLABORATE Live 300 system is engineered for users looking for a complete small-room and executive office video collaboration solution. Featuring the same user interface found in the larger room systems, COLLABORATE Live 300 includes a free 90-day COLLABORATE Space web conferencing subscription, a Skype for Business client, SIP/H.323 videoconferencing, wireless presentation, and interactive whiteboard capabilities—along with one CHAT 150 speakerphone and one UNITE 150 PTZ camera with 1080p30, 12x optical zoom.

Rounding out ClearOne’s new COLLABORATE Live product line is COLLABORATE Live 200, a new video collaboration system with ultra-wide angle video capture, critical for viewing all conference participants in huddle spaces and smaller room environments. Designed specifically to meet huddle space budgets, COLLABORATE Live 200 features the UNITE 50 EPTZ 1080p30 camera with 3x digital zoom and a 120-degree, wide-angle field of view. Other features are similar to those found in the COLLABORATE Live 300 system, including the free 90-day COLLABORATE Space subscription.

“Video is central to the 21st century collaboration experience,” said ClearOne CEO and chair Zee Hakimoglu. “Users should be able to have the video collaboration system that is right for them in their own unique environment. Also, COLLABORATE Live enables users to enjoy a user interface that is as intuitive and simple as using their familiar mobile or tablet devices.

“From the smallest huddle spaces to the largest auditoriums and training facilities, ClearOne now makes it possible, practical, and affordable to leverage all the audio and video benefits of collaboration in cloud-based solutions that are custom designed for every size of room. ClearOne’s mission is to enable the video generation with all of the tools needed to achieve new levels of collaborative productivity.”

The new COLLABORATE Live Series will be showcased at ISE 2019 at Stand D-130 in Hall 11 at the RAI Centre in Amsterdam, and is available for immediate delivery.

The Bottom Line: COLLABORATE LIVE is designed to empower users to collaborate in any size room environment, with high-quality audio and video, interactive whiteboard, presentation, recording, streaming, and cloud connectivity.

MSRP for the systems is as follows: COLLABORATE Live 900, $9,999; COLLABORATE Live 600, $6,999; COLLABORATE Live 300, $3,499; COLLABORATE Live 200, $2,799.

As both staff and a freelance producer/engineer at leading New York City studios like The Hit Factory and Quad Recording in the 1990s, Chris Curran has accrued credits on recordings for artists such as Foreigner, Ice Cube, Jeff Buckley, Philip Bailey, and many others. Curran discovered a passion for podcasting several years ago, and as a podcaster and educator, he has brought that deep knowledge and technical understanding to the podcasting world. 

After relocating to Colorado Springs in 2015, he decided to exchange ideas with the proliferating base of podcasters through his Podcast Engineering School, disseminating practical audio understanding through live interactive online training and mentoring from his studio, Fractal Recording. And one constant piece of the podcast puzzle for Curran throughout his podcast career has been Focusrite’s Scarlett USB audio interfaces.

“I discovered Focusrite microphone preamps when I was still an assistant engineer, working my way up,” Curran said. “When the Scarlett range became available, I knew that’s what I wanted to use, because it’s Focusrite, which automatically means it sounds great.” He’s used several of the Scarlett models, including the first-gen 8i6, and second-gen 18i20 and 2i2 (of which he has four) units he currently works with, along with a Focusrite Red 3 dual compressor/limiter. But he said that the Scarlett 2i2 is the unit best suited to the broadest range of podcasters. “It’s small, it’s affordable, and it has two microphone inputs, which is exactly what the vast majority of podcasters need,” he said. “And you know you’ll have great sound right from the start, because of the audio quality that Focusrite brings to it.”

Curran also praised the simplicity of design and operation of the entire Scarlett range, which enables podcasters to get good sound out of the box, allowing them to focus on their content. “When I started podcasting, I’d see other podcasters sticking a microphone on a table between four people and just going for it, so I knew there was a need for education around how to get the best sound for podcasting. With the Scarlett 2i2, they can spend more time on the podcast itself, knowing that Focusrite is helping them get good sound quality. It has the basic functionality they need, and with those Focusrite mic pres, the sound they want, at a price they can afford. It’s solid and reliable. There are other interfaces out there, but Focusrite did it right.”

It can be argued that quality audio is the most critical component in the effectiveness of an AV system. More so than inadequate video or complex operation, unintelligible audio makes it very difficult—or impossible—to convey a message. The inability to hear a presentation or participate conference call can be one of the most frustrating and embarrassing situations that a user will encounter.

Audio systems are designed through basic principles of AV, including selection of the proper type and quantity of microphones and speakers, understanding distance measurements for placement, and following installation best practices. And, in most cases, a programmable digital signage processor (also known as an audio DSP) is utilized to configure the audio paths, mixing, gain structure, and echo cancellation. Since the audio DSP is the component that requires the most setup in the audio chain, it can also be linked to the cause of most audio issues if not programmed properly.

Audio DSPs are software-driven devices that are considered “programmable”; however, the skillset required is more of configuration and is better performed by AV system designer or audio engineer than a traditional programmer. It is critical that the person responsible for programming the audio DSP has thorough knowledge of audio and is familiar (or better yet, certified) in the brand of audio DSP that is specified in a project. Like control system programming, the software, hardware components, and approach required to program an audio DSP vary from manufacturer to manufacturer.

Most audio DSP platforms can be categorized in two flavors: open architecture and closed architecture. Open architecture refers to the lack of predefined structure in the design in the audio system. For open-architecture systems, programming of the audio DSP starts with a blank canvas and provides ultimate flexibility in the design of the audio flow and processing, resulting in more options and support for more variations. However, it requires more skill, discipline, and knowledge to avoid pitfalls.

Closed-architecture systems are less flexible, but easier to design due to static components within the audio system that are always present and set up in advance. Although these systems may be easier to program and have fewer variables, the rigidity of their structure may not suit all needs or applications.

Similar to control systems, selecting the appropriate audio DSP solution needs to incorporate a number of factors, including the system application, the size and layout of the space, user requirements, and future considerations. For example, the audio DSP that would be a great fit for a meeting room, conference room, or classroom may not the be same as the audio DSP that would be ideal for a multipurpose room, event space with room combining, theater, or stadium.

Most audio DSPs are scalable; however, there can be limitations to the number of pathways between units, processing power, and availability of echo cancellation. Lastly, the ability to support Dante or other audio network protocols is a valuable feature for interoperability and expansion.

In addition to hardware specifications and installation parameters, an audio DSP program or configuration takes into consideration the following factors that need to be defined in software where they apply.

• Presentation audio routing
• Audio conference mixing and echo cancellation
• Videoconference mixing and echo cancellation
• Audio conference combined with a videoconference
• Speech reinforcement
• Mix minus voice lift
• Room-combining presets
• Compensation for room design and acoustics
• Integration with a control system
• User controls and requirements

In addition to programming the audio DSP to satisfy the system requirements, the audio levels need to be set up and calibrated to provide optimal performance. This important service of audio commissioning is often overlooked due to lack of time, knowledge, or scope requirements. Since audio depends very much on the specifics of physical space and environment, proper audio DSP programming accompanied by precise field commissioning is what is required to ensure success.

There is more than meets the eye when it comes to delivering an effective audio system. Considering how important audio is to the success of a presentation, meeting, or event, proper programming and commissioning of an audio DSP needs to be handled effectively and looked at as a priority.

As with control system programming, it is critical to identify a specialized and reliable resource to ensure your audio system will perform effectively and efficiently. The idea that any field tech or programmer can provide audio DSP programming is not a good solution.

Don’t put your meeting, presentation, or conference call at risk with an unproven field technician or programmer who is trying to figure out how to program the audio DSP on the fly. Instead, enlist the skills of a specialized audio DSP professional and have confidence that users will communicate with comfort and clarity.

Steve Greenblatt, CTS, is president and founder of Control Concepts, a provider of specialized software and services for the audiovisual industry.

Want to Learn to Program DSPs?

Many industry manufacturers sponsor training programs to learn more about DSP and the AV industry at large. For example, Biamp offers a wide variety of certification training in a number of formats to fit any schedule, no matter how hectic. The company hosts live webinars, on-demand self-paced courses, and more advanced hands-on, instructor-led sessions providing the information and the knowledge to build the solutions that customers are seeking. Biamp’s trainings cover a broad gamut, from topic specific subjects to become more fluent in networking, VoIP and DSP, to individual product training from Tesira to Vocia.
Visit www.biamp.com/training to learn more.

The What: Yamaha console users can now access Stage Tec's NEXUS system over IP in conjunction with the Dante interface. Yamaha consoles can control NEXUS microphone inputs and can use the NEXUS converters.

The What Else: With the TrueMatch microphone converter, NEXUS offers AD/DA converter technology at an internal resolution of 32-bits and a dynamic range of 158dB (A), eliminating the need for microphone preamplifiers.

NEXUS requires the Dante audio-over-IP interface XDIP and the control board XACI for this interaction. A Dante board provides up to 64 channels. Up to four Dante boards can be used per mixing console. This IP-based variant replaces the serial modification (via MADI and NEXUS XCI).

The Bottom Line: Yamaha has expanded Dante device support with the latest update of the CL/QL series of mixers. Yamaha mixing consoles can now be accessed over IP in Stage Tec NEXUS routing systems to control the parameters of the TrueMatch microphone inputs (phantom power, low cut, and gain). The license for NEXUS remote control by Yamaha mixing consoles must be purchased from Stage Tec.

The What: Meyer Sound has been granted a U.S. patent on proprietary digital technologies that enable the Bluehorn System to reproduce complex musical signals with absolutely flat frequency and phase response across the full audio bandwidth, an achievement in high-power studio monitors.

The What Else: The new patent, number 9,992,573 B1, is formally titled, “Phase inversion filter for correcting low frequency phase distortion in a loudspeaker system.” The award document outlines the digital signal processing techniques involved in canceling out the phase anomalies inherent in all loudspeaker systems due to the physical mass of the loudspeaker drivers and resonance of the loudspeaker enclosures. The patented technology restores the original phase relationships, even at the lowest audio octaves, by applying an inverted phase response.

The patented filter algorithms are hosted in a dedicated hardware processor placed in front of both the two-way, mid-high loudspeaker and the separate low-frequency loudspeaker. The system-specific correction algorithms hosted by the processor were developed by meticulously analyzing all physical and electrical non-linearities throughout the entire system, from 25 Hz to 20 kHz, in order to bring acoustic output — at any level — into phase and frequency alignment with the input signal. Essentially, the digital processing nullifies the non-linearities of drivers and enclosures, resulting in a phase-coherent response previously achieved only by open-air electrostatic headphones with their extremely low mass diaphragms.

The Bottom Line: Complete Bluehorn Systems are available in either stereo or LCR configurations, with each channel comprising a two-way mid-high loudspeaker, a separate low frequency element, and the dedicated digital processing.

Bluehorn System is designed for high-resolution monitoring applications in mid-sized rooms for music recording, cinema post-production, music mastering or other applications requiring the highest accuracy in sound reproduction. Externally, Bluehorn System’s two-way component shares the same cabinet footprint, driver dimensions and waveguide configuration as Meyer Sound’s Acheron Designer loudspeaker. On the inside, however, it utilizes an upgraded HF driver and new amplifiers with ultra-low noise and distortion.

The What: miniDSP has launched the all-new Streaming HD (SHD) series stereo processor featuring Dirac Research's Dirac Live room correction platform.

The What Else: The Dirac Live-enabled miniDSP SHD series brings to market a stereo processor designed from the ground up to achieve the highest performance with the most flexibility. With its floating point processor, 32-bit converters, quad-core ARM streamer, multichannel USB audio, Volumio network audio support, and more, the SHD series delivers natural, realistic, and transparent sound to streaming audio.

The SHD series joins an extensive portfolio of Dirac Live-enabled miniDSP products that combine miniDSP’s expertise in digital signal processing with Dirac’s expertise in audio optimization and room correction.

“Few in the industry are as passionate about audio as those at miniDSP, which is why our years of extensive collaboration have yielded such astounding results,” said Niklas Thorin, Dirac’s GM of high performance. “The Streaming HD series is the ideal product to introduce Dirac Live to the streaming market, as it truly captures the full potential of our room correction capabilities. With this SHD series introduction, the bar has now been raised for high definition audio streaming.”

Dirac Live is a patented room correction solution that not only corrects the frequency response, but also the impulse response, of the loudspeakers in a room. The solution provides true impulse response correction over a large listening area, improving the depth, the positioning, and distinction of individual voices and instruments.

In an effort to satisfy the needs of all miniDSP customers, the company will release two versions of the SHD series: the SHD full rack-mount unit, an analog and digital All-In-One-Does-It-All unit, and the SHD Studio, a more compact, all-digital solution available at a lower cost and suited for those who already own a DAC. SHD Studio comes with all features of the full version, but without the analog component.

Both versions of the SHD series come equipped with an embedded Volumio network audio music player that allows for the streamed audio to be controlled via smartphone, tablet, or desktop-based app.

“Regardless of hardware design and sophistication, certain external variables are always going to negatively impact audio quality—that is unless you implement a digital room correction solution like Dirac Live,” said Antoine Rouget, CEO at miniDSP. “Dirac Live reduces these variables and equips our products with the degree of performance consistency necessary to satisfy the discerning ears of the miniDSP community. We’re excited to work yet again with Dirac to introduce its Dirac Live room correction solution to the streaming processor market.”

The Bottom Line: Using multiple measurement and mixed phase correction, Dirac Live helps audio systems to create a natural, realistic, and transparent sound with tighter bass and reduced room modes, in a way previously not possible.

The What: Yamaha Professional Audio has released Version 5.0 for Yamaha CL and QL Digital Audio Consoles.

The What Else: In addition to support for more Dante devices, the V5.0 update adds to the consoles’ growing set of tools. The Portico 5045 Primary Source Enhancer by Rupert Neve Designs is now available in the Premium Rack. This device makes it easier to control background noise from microphone sources while enhancing clarity and increasing feedback margin, and is a valuable tool for houses of worship, stadiums, performance halls, and any other venue where feedback must be carefully controlled. Individual Fade Time functionality allows different fade times to be set for each fader, an important feature for users who need complex fader control, especially for musicals and other theatrical applications. Dante devices newly supported by V5.0 Future updates will provide support for other Dante devices.

“Yamaha CL and QL console updates continually contribute to a wide range of applications such as live music, corporate events, broadcast, and theater,” said Marc Lopez, director of marketing, Commercial Audio products. “And now, the Version 5.0 update not only adds new functionality, but through close collaboration with third parties, also provides connectivity with more Dante devices.”

The Bottom Line: This no-charge update is designed to provide enhanced convenience, new features, and expandability. V5.0 for the Yamaha and CL and QL Digital Audio Consoles is available via download at www.yamahaca.com.

Datapath has signed global distributor Stampede for Latin America.

Stampede President and CEO, Kevin Kelly, said the Datapath partnership dovetails with the company’s current product offering. “For more than 35 years, Datapath solutions have helped to revolutionize video distribution and management for thousands of customers across the world and now, with the support of Stampede, they will be able to reach broader and deeper into the Latin American market,” he said. “Datapath’s solutions are a perfect complement to our current portfolio of offerings and they will enable our regional resellers to custom configure system solutions across every major vertical market.”

Small and medium scale digital signage continues to thrive in Latin America, and Datapath’s processors and controllers bring reliability and power to display projects of any size.

Greg Babbs, Datapath sales manager for the region, said: “Our Fx4 controller is known and trusted worldwide and we’re sure it will be a popular choice for Stampede’s customer base. What’s more, with the availability of our new Hx4 and the very latest additions to the iolite range, we now have a solution for any multi-display or video wall requirement.

He added: “From Chile to Mexico, Latin America continues to embrace digital signage. Security projects and smaller control centers are a growth area but there is also strong demand in the rental market—ideal for our robust but portable control hardware.”

To coincide with the partnership, Datapath has recently provided day-long training sessions at Stampede’s LATAM offices, ensuring all staff are up to speed with Datapath’s latest technology. Further training sessions are planned towards the end of the year.

The What: Radial Engineering has announced that the Catapult Mini TX and RX are now shipping.

The What Else: The Radial Catapult Mini is the latest addition to the modular Catapult snake system, which provides the ability to drive four analog audio channels over a single shielded Cat 5 cable. The Catapult Mini TX (transmitter) and RX (receiver) both feature a compact steel chassis with a hardwired fan-out to four XLR connectors, providing a simple and rugged solution for delivering balanced analog audio between any two destinations without signal loss or degradation.

The Catapult system connects via Cat 5 Ethernet cable, which is widely available, lightweight, and economical. As many buildings are already pre-wired for Cat 5, the Catapult Minis can take advantage of this to easily transmit audio between distant rooms without the need to install or supply new cabling.

The Bottom Line: All of the Radial Catapult modules are compatible with each other, and each is sold individually, providing the option to mix and match as needed. The Catapult Mini TX and RX can be used together as a pair to create a compact four-channel audio snake, or they can be matched with other Catapult modules that include splitting capabilities or transformer isolation—allowing you to select whichever combination best suits your application.

The What: Glensound, the UK-based manufacturer of audio products for the broadcast, sound reinforcement, installation, and theatre markets and ASK Sales have introduced the AoIP22 Dante/AES67 Two Channel Bi-Directional Audio Interface.

The What Else: The new AoIP22 offers versatile connectivity that ensures easy integration into a variety of environments. The interface offers two balanced analog audio inputs on XLR connectors, two balanced analog audio outputs on XLR connectors, and a Neutrik Ethercon RJ45 network connector. When using the free Dante Controller by Audinate, which can control all Dante enabled devices on the network, the AoIP22 is automatically identified. Once discovered, users route audio circuits to/from the AoIP22 with just the click of the mouse.

Glensound’s AoIP22 interface uses the standard 24-bit Dante network resolution and recognizes Dante network sample rates of 44.1/ 48/ 88.2 and 96kS/s. Being AES67 compatible, the AoIP22 supports the AES67 network sample rate of 48 kHz.

A removable belt clip is supplied and fitted to one side of the AoIP22. While not necessarily designed to be worn, this clip makes the interface flexible enough so that the AoIP22 can be hung in many different places. The unit also includes rubber feet—enabling it to be positioned, for example, on a desk or the back of a workbench, without concern for the unit sliding around. The AoIP22 is also designed with cable tie holes in all four corners of the case. These are extremely strong and are well-suited for threading cable ties—providing yet a viable option for temporary installations.

The Bottom Line: Designed to quickly and easily interface existing analog equipment to a Dante/AES67 network audio system, the AoIP22 is powered by PoE (Power over Ethernet)—so only one cable needs to be connected to the network to carry both audio and power, providing flexibility, and saving installation time.

RTI, a control and automation manufacturer for residential and commercial environments, announced that its XP-8v Advanced Control Processor was honored with a SCN InfoComm Installation Product Award in the category of "Most Innovative AV Automation/Show Control Product" at InfoComm 2018 in Las Vegas.

The third generation of RTI's XP-8 advanced control processor, the XP-8v combines the advanced control options of its predecessors with a new Cortex A8 CPU for increased processing power and expanded two-way driver support. Expanding the processor's capabilities even further, its RAM memory and internal non-volatile Flash memory have been upgraded to 1GB and 512MB, respectively. The addition of an HDMI output enables intuitive control via on-screen display, providing users with the convenience of a customizable interface on their TV.

[SCN Announces 2018 InfoComm Award Winners]

"We loaded the XP-8v with all the horsepower and features our dealers need to deliver reliable control and integration in large-scale homes and commercial installations," said Brett Stokke, director of marketing for RTI. "It's an honor to have our efforts recognized at InfoComm by SCN magazine and its readers with this prestigious award."

The SCN InfoComm Installation Product Awards showcased the most innovative commercial AV products of 2017. Finalists were selected by an independent judging panel of experienced integrators, with winners chosen by SCN readers via online voting.

The What: Audionamix has introduced its latest plug-in, IDC: Instant Dialogue Cleaner, at this year's Sweetwater GearFest on June 22-23 in Fort Wayne, Indiana.

The What Else: The IDC plug-in is versatile and easy to use due to its integrated DNN artificial intelligence that automatically detects and separates speech, no matter the surrounding content. With the turn of a knob, users can clean up dialogue from interviews, dailies from movies and TV, during the post-dialogue editing process, re-recording mixing, news and sports recording, radio broadcast, and more.

The Bottom Line: The IDC plug-in is a real-time, cloudless solution, and will offer unique and effective dialogue cleaning technology through an intuitive interface, compatible with many workflows. The software will be available for pre-launch sale starting on July 2 and will officially be released on July 17. Audionamix will be in the R. Joseph Group Tent 7 at GearFest.

Extron Electronics has been selected by Avaya for membership as a Technology Partner in the Avaya DevConnect program; its DMP 128 Plus V Audio Processors are compliant with key unified communications solutions from Avaya.

The Avaya DevConnect program promotes the development, compliance-testing, and co-marketing of third-party solutions that are compatible with standards-based Avaya solutions. As a DevConnect Technology Partner, the Extron DMP 128 Plus V models have been compliance-tested by Avaya for compatibility with the Avaya Aura 7.1 and IP Office 10.1 Platforms, enabling companies to confidently integrate local A/V solutions with their VoIP communications.

“Membership in the Avaya developer community underscores our commitment to providing solid, real world solutions to our integrators and end users,” said Casey Hall, vice president of worldwide sales and marketing, Extron. “Our shared customers can be confident that Extron’s VoIP solutions have been engineered and tested for ease of integration, and years of reliable and cost-effective use.”

The Avaya DevConnect program currently includes thousands of software and hardware developer companies, integrators, service providers, and customers. Members have created a broad array of solutions tested for Avaya compliance, including natural language speech recognition applications, mobile and emergency notification services, specialized computer telephony integration and reporting capabilities, and applications tailored for specific vertical industries.

“Technology Partners like Extron Electronics are helping Avaya promote complete, integrated solutions,” said Eric Rossman, vice president, partnerships and alliances, Avaya. “Extron’s new VoIP solutions bring a wealth of A/V experience that enable clear, natural communication experiences for all conferencing participants even in challenging conditions. We are pleased to have Extron join our developer community.”

The What: Bose Professional broadens its ControlSpace sound processor line with six new DSP models and three attractive digital wall controllers.

The What Else: The new ControlSpace CSP-1248 and CSP-428 commercial sound processors offer right-sized I/O and simplified configuration for small-to-medium-scale projects. They are designed to serve as standalone DSPs in commercial applications such as retail stores and restaurants, or any public place where modest processing, high-quality sound and facilitated installation are desired. They include balanced analog I/O, RCA inputs, eight GPIs, one GPO, a mute contact, RS-232 serial, Ethernet network (for configuration or wall control) and Bose AmpLink out, for simplified digital audio connectivity to supported power amplifiers.

Bose has also rolled out an update to two popular ControlSpace ESP fixed-I/O engineered sound processors. New variants of the Bose ESP-1240 and ESP-880 models have been upgraded based on integrator feedback to include options for Dante™ audio networking and Bose AmpLink connectivity. The new variants – the ESP-1240A, 1240AD, 880A and 880AD – inherit a fresh visual appearance while maintaining key features from their predecessors.

To complement both the new ControlSpace CSP and ESP processors, Bose has also introduced three new user-friendly digital wall controllers: the CC-1D, CC-2D and CC-3D. Each model comes in white and is available in regional variants.

The Bottom Line: In addition to the new digital wall controllers, the new CSP and ESP processors are supported by ControlSpace Remote, where customized control panels can be built and deployed to end users for wireless control of their systems from mobile phones, tablets or laptops.

The What: Radial Engineering has begun shipping Presenter, a compact mixer that combines a microphone preamp and a USB program input for presentations, from school lectures to corporate audio.

The What Else: The USB input on the Presenter allows for direct connection to a laptop for high-quality audio playback, for presentations that include digital audio files, or for DJ applications. This connection also provides power for the Presenter, eliminating the need to carry around a separate power adaptor. For connection to mobile devices such as smartphones or tablets, a 3.5mm stereo input jack is included on the front of the unit, and can be adjusted using the program level control. The XLR microphone input is paired with low and high EQ controls to tailor the frequency response, along with a low-cut filter that rolls off excessive low end.

Both the microphone and the playback signals are mixed to stereo XLR balanced outputs, which can be connected directly to a PA system or a pair of powered speakers. A mono switch is provided if only one speaker is needed, and a ground lift ensures that the output signal is clean and free of buzz or hum from ground loops.

The Bottom Line: The Presenter's compact size allows it to fit on a desktop or podium, and it includes a port for a Kensington lock to secure it for permanent installations. As an added benefit, the Presenter can also be used as a USB recording interface, so speeches or lectures can be recorded for later use.