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

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

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

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

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

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

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

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

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

Brutal güet, a Swiss broadcast services provider, has rolled out a state-of-the-art outside broadcast (OB) vehicle built on a Lawo AoIP (audio-over-internet protocol) infrastructure to power IP-based workflows.

The new vehicle is outfitted with a Lawo mc²56 MkIII production console with 48 faders, running on a redundant A__UHD Core licensed for 256 DSP channels. The core itself features a software-defined architecture providing up to 1,024 DSP channels within a compact 1RU housing. With native support for SMPTE ST2110, AES67, and RAVENNA, the system enables fully IP-based signal processing, Lawo said.

“The combination of the mc²56 and HOME [Lawo’s cloud-native control and management platform] has fundamentally streamlined our production workflows,” brutal güet Chief Operating Officer Jonas Zubriggen said. “We can configure complex setups in no time, automatically integrate devices into the network and react flexibly to new requirements—without manual IP configuration or lengthy prep work.”

The setup offers brutal güet maximum operational reliability and scalability, qualities that are essential for productions involving complex audio configurations and simultaneous multilingual broadcasts, it said.

“With Lawo, we can reliably manage parallel commentary feeds in German, French and Italian—all from one compact yet powerful vehicle,” Zurbriggen said.

The Lawo console supports demanding live productions and immersive audio workflows and offers snapshot automation, loudness metering and direct integration of Waves plug-ins, the company said.

Lawo’s A__stage80 and A__stage48 I/O units provide signal distribution. The A__stage80 offers 32 mic/line inputs, 32 line outputs, and eight AES3 stereo I/Os. It also supports SMPTE ST2022-7 seamless protection switching for redundant networking. Both are equipped with Class-A microphone preamps, MADI interfaces and dual IP ports.

The Lawo HOME platform manages brutal güet’s AoIP infrastructure. It automatically detects compatible devices, integrates them securely into the network and enables centralized configuration via an intuitive interface. HOME supports NMOS IS-04/05, IEEE 802.1x and RADIUS. With the new HOME 4.0 release and the HOME Pass licensing model, infrastructures can be scaled precisely, it said.

Broadcast Solutions designed the OB vehicle based on brutal güet’s operational requirements using its Streamline concept. The interior layout follows a clear workflow structure: the front section houses the video control area featuring a Sony MLS-1 video switcher for UHD/HDR productions, along with slo-mo and multifunction workstations in an expandable bay.

The center section accommodates the Lawo mc²56 MkIII console, installed perpendicular to the driving direction to maintain direct line-of-sight between the audio engineer and the director. A side window further enhances communication. The rear section includes camera shading and technical control workstations, along with server racks and the HI human interface control software system providing centralized process automation.

“We’re new to the OB truck business, but from the outset we wanted to make a clear statement about quality,” Zurbriggen said. “With Lawo, we’ve found a solution that not only meets our current needs but is also fully scalable for the future.”

More information is available on the Lawo and brutal güet websites.

NEW BERN, N.C.—Wheatstone Corp. has named veteran audio engineer Kelly Parker as director of product development.

Parker returns to the audio equipment supplier from broadcaster Townsquare Media, where he was vice president, broadcast operations, overseeing the technical operations of 349 radio stations in 74 markets. His new role calls for him to direct development for the company’s audio-over-IP, streaming, virtualization, mic, on-air processing and software server lines.

“Kelly has that rare perspective that comes with first-hand experience both as a developer and user of our technology, which will be important as we enter what is expected to be a transformational period for broadcasting,” Wheatstone CEO Gary Snow said.

Parker had been a systems engineer at Wheatstone for 14 years during the peak development of AoIP, from 2004 until he left to join Townsquare Media in 2019.

"I’ve been on the other side of the industry running 300-plus radio stations for the past few years,” Parker said in a statement. “Things are definitely changing and Wheatstone is uniquely positioned to adapt technology to those changes."

He cited the nature of Wheatstone’s centralized operations as an advantage, as it allows the equipment maker to quickly design and test new product developments for broadcasters.

Parker played a lead role in the development of Wheatstone’s WheatNet IP audio network in early 2005, the company said, introducing the first AoIP system that combined routing, control and utility mixers into one intelligent input/output unit. In subsequent years, he led the development of Wheatstone’s flagship LXE console, the first unit to use soft controls as a reconfigurable AoIP console surface.

He will be based in Wheatstone’s New Bern, North Carolina, factory.

The most significant advances in audio for today's media have emerged from broadcasters' transition to IP and the cloud. More than any other development, audio over IP is expanding the ways audio can be enhanced and customized to expand viewer options. 

 In our latest Guide to Audio, we take a look at how IP is changing the way audio is processed, managed and delivered in today’s broadcast environment and how it is revolutionizing the viewing experience.

Download the free ebook here. 

TSL will showcase new control and audio monitoring solutions combine conventional and IP audio infrastructures while reducing complexity and cost at its booth at the 2023 NAB Show, April 15-19 in Las Vegas.

Debuting at the show is X-Connect, TSL’s new IP routing control system, which the company says is effectively a virtual router built on industry-standard hardware and protocols. Perfect for incremental upgrades, X-Connect uses vendor-agnostic NMOS, making it compatible across a wide range of existing devices—including TSL’s own Virtual Panels, for a totally-customizable installation.

Distributed and remote production are now commonplace alongside conventional production workflows. With audio sources arriving over an increasingly diverse range of sources, flexible and accurate confidence monitoring has never been more essential. TSL’s new MPA1-MIX-NET has a clear and easily understood control panel and a wide range of input formats, including ST2110-30, RAVENNA and AES67. A 1G AoIP connection provides up to 64 channels, with an additional 64 channels of MADI as an option.

“Our priority at TSL is to provide an affordable, scalable and sustainable approach to future workflows,” said Ian Godfrey, President of TSL. “We make IP and traditional technologies work together, employing modern techniques to maximise productivity and reduce complexity for operators.”

TSL will be in Booth C2416.

The transition to audio over IP over the past decade has revolutionized the broadcast industry, bringing added flexibility and improving quality and options for the viewer. That progress, however, has not come without its concerns as media companies try to keep up with consumers who are viewing content on myriad devices.

In our latest free TV Tech Guide to Audio, we hope to lessen that anxiety by taking a look at how audio professionals are tapping into that flexibility that AoIP brings to our industry and the approaches taken to maximize the consumer audio experience. 

NEW BERN, NC—At this year’s NAB Show, April 23-27 in Las Vegas, Wheatstone is introducing several new AoIP-based audio consoles plus an upgrade to its Blade 4 access unit.

Tekton 32, Wheatstone’s latest AES67 and WheatNet-IP audio networked TV console will show for the first time at show. Tekton 32 interfaces with all major production automation systems through a unique control interface for tight integration of automation and mixing in one native IP audio environment. Tekton 32 has standard 64 channels (layered) and motorized faders tracked to automation so whoever is producing or directing, or doing both, can keep an eye on what the automation is doing and make adjustments as needed.

The AoIP console is also very easy to navigate for the occasional news report or sporting event that requires hands-on mixing. Tactile faders on the one hand and a touchscreen interface on the other make it easier for the busy producer or director to adjust EQ, fix levels and mix in feeds. All routing and control for the Tekton 32 is handled by the WheatNet IP audio network, an AES67 compliant and SMPTE 2110 supported IP audio network. If it’s on the network, it’s routable, programmable, and accessible – often automatically. For example, when a field reporter’s mic turns on, the correct mix-minus can be automatically sent back to the field reporter’s headset for IFB. 

The compact size of Tekton 32 is due to AoIP carrying much of the load that once sat on the console. Plus, by connecting routing, mixing and studio control through Ethernet cabling, AoIP opens up accessibility and gets rid of outdated wiring and layers of audio infrastructure. Tekton 32 is also WAN and REMI ready. Tekton 32’s WheatNet-IP audio network can scale from one location to several geographic locations across a WAN for use in REMI or other remote broadcast applications.

Also new is the GSX console surface, Wheatstone’s latest fully reprogrammable AoIP console and the newest addition to the WheatNet-IP audio network family. Wheatstone’s flagship LXE console surface, introduced in 2016, was the first completely customizable control surface for IP audio systems. 

The GSX AoIP surface features:

• Scriptable knobs and controls for customizing by show, talent or studio. Customize once and set it or quickly re-configure controls and displays for any purpose or application using Wheatstone’s ConsoleBuilder™ software.
• Up to 32 input channels with as few as four physical faders, or any combination that fits the studio size and function needed.
• Metering on-screen instead of on the console for a low-profile workspace that lets talent focus attention where it’s needed.
• Tactile surface on the one hand and touchscreen interface on the other for the best of both worlds. Faders and cue events on the surface. Pinch and drag EQ settings from the touchscreen.
• Modular design. Configure as a drop-in or tabletop console surface, or split into fader wedges in separate rooms connected via WheatNet-IP audio networking for fast and easy talent and show collaboration.
• Remote operation. Optional standalone virtual mixer or remote mixer that mirrors the physical surface offers an independent, yet shared user experience.
• Optional Automix, Layers, ScreenBuilder™, Smart Switch panels and IP accessories. Add any or all at any time to adapt the GSX to evolving needs.
• Empowered by WheatNet-IP, a complete intelligent network of connected elements, audio tools, and third-party products and applications.

Wheatstone has also added new audio tools on its AoIP network. This latest AoIP unit includes audio routing, control, codecs, processing, mixing, operating system and NMOS/AES67 interoperability in 1 RU. Blade 4 can be integrated into any new or existing WheatNet-IP network and includes:

• Selectable Opus, MP3 and AAC codecs for integrating workflows from remote venues or home studios into the studio operation as needed. All codecs are routable in native AoIP; no additional studio hardware required.  
• Built-in OS for running customized scripts and specialized software, metering apps and virtual interfaces.
• AES67 compliance, from .125 ms to 5 ms packets, for a wide range of interoperability with other AES67 devices and networks.
• Support for SMPTE ST 2110, including the NMOS discovery standard for AES67 and next generation television networks.
• Dual Ethernet ports available on every Blade 4 for failsafe redundancy.
• Integrated audio codecs, processing, mixing and operating system in one native AoIP environment for resource sharing.
• Two separate audio clip players, enhanced to play compressed or uncompressed audio files from the built in USB ports to virtually eliminate memory storage issues.
• Full AoIP I/O and intelligence in one unit for routing audio, mixing sounds, processing feeds, and controlling mics, consoles, and other studio appliances. Blade 4 has all the standard Blade features, including two 8x2 stereo utility mixers for online mixing of sounds or segueing remotely between feeds; routable stereo processor with parametric equalizer, compressor and limiter; and 12 universal logic ports plus 128 software LIO ports, programmable as inputs or outputs, and routable through the network.
• Codecs, software apps, mixing and audio processing, plus AoIP routing, control and interoperability in 1 RU for reducing rackroom real estate and associated cooling, cabling and other expenses.  

Wheatstone will be in booth N2631. For more information on the NAB Show, visit nabshow.com/2022.

LAS VEGAS—Sound is part of the many different aspects of broadcasting that will be featured at the 2022 NAB Show. Two elements in particular—immersive/personalized audio and intercoms—will attract a lot of attention in Las Vegas.

A trend that is still in its relatively early days is immersive and object-based audio (OBA). Both of these, which come under the general heading of “next generation audio” (NGA), have been discussed and promoted as “the next big thing” to hit broadcast television for at least the last decade. 

Dolby Atmos
This year will mark a decade since the first film in Dolby Atmos—Disney’s “Brave”—was released in cinemas, according to Mathias Bendull, vice president, Living Room, for Dolby Laboratories. “It’s incredible to see the progress we’ve made and the growing excitement from consumers and the industry demanding immersive sound experiences brought to life by spatial audio,” he said.

Today the world’s top streaming services—Apple TV+, Disney+, Netflix, and many others—deliver their most-watched movies and streaming shows in Dolby Atmos (and Dolby Vision) and Dolby Atmos devices are widely available across various price points and form factors from home theater systems to smartphones and TVs, Bendull said. 

With Samsung’s recent announcement that it would support Dolby Atmos in its upcoming Micro LED and Neo QLED TV, this means all major global TV manufacturers are now delivering TVs with Dolby Atmos, according to Bendull, who added that a majority of these TVs in North America also support ATSC 3.0 tuners, which will provide consumers access to NextGen TV delivered via Dolby AC-4.

“Consumer interest in Dolby Audio technologies, such as Dolby AC-4 and Voice+, has also shaped how broadcasters promote the benefits of NextGen  TV,” Bendull said. “This includes the recent NextGen TV holiday campaign, which aired commercials in participating ATSC 3.0 markets promoting the features of Dolby Audio, such as improved dialog audibility enabled by Voice + and consistent loudness.” 

Dolby is also teaming up with Pearl TV to amplify these messages at the NAB Show and through consumer education and retail point of sales initiatives,” he added.

Personalization & Immersive
On the horizon, the benefits of personalized audio features like home and away commentators, multiple language support, and audio description for 5.1 and Dolby Atmos are being explored by key industry players jointly with Dolby.

There is the view, from institutions such as Fraunhofer IIS—the main developer of MPEG-H 3D Audio—that the personalization aspect of NGA is the more compelling and necessary feature, rather than immersive sound. Christian Struck, senior product manager for audio production for Lawo, comments that the current trend appears to be creating presentations—in other words, “audio stems”—that can be used for different applications whose volume levels are set by viewers at home.

“A stem-based approach means that more channels and busses are needed than for a 5.1 setup,” Struck explains. “This means the ability to set channel levels for a variety of presentations becomes paramount and a flexible array of monitoring options is required.” 

On the subject of immersive audio, Struck describes it as “the perfect complement to 4K picture quality” and says it is being increasingly demanded by rights holders. “The main features for audio consoles in an immersive workflow, such as high channel counts and the bussing, monitoring and mixing infrastructure to support them, will become available in 2022.”

In terms of greater capacity on mixing consoles, at this year’s NAB Show, Lawo will be showing a 48-fader version of the mc²36 at the booth in the Central Hall. Other introductions include a software version for the mc² range, the A_UHD Ultra-High Density IP audio engine and new software features for the V_matrix routing, processing and multiviewing platform.

While sport and drama have fully embraced immersive audio—for future-proofing post-production if not for transmission at this time—there are areas that do not appear immediately well-suited to the technology.

“Our ‘bread and butter’ is TV news and it will be some time before that sector gets into the immersive environment,” comments Phil Owens, senior sales engineer at Wheatstone. Of personalization he adds, “There are a lot of levels to that. You can pick and choose different channels for things like second languages. But that’s not really immersive audio.”

At its booth in the North Hall, Wheatstone will launch Tekton 32, its latest audio over IP (AoIP) console for TV. The networked console is based on the company’s WheatNet-IP protocol that is aimed at smaller, local TV stations, which are either installing automated production systems or downsizing staff. It has 64 channels, which are layered, and motorized faders that can be controlled by the producer or director through the automation system.

Also on the subject of IP, a highlight at Calrec's booth will be a demo in which three independent consoles will be operated from a fully redundant pair of ImPulse cores; a 48 dual fader Apollo console, a 40 fader Artemis console and a headless console running Calrec Assist on a PC. The ImPulse DSP features native SMPTE 2110 connectivity and is compatible with the Calrec Assist web interface as well as Calrec’s Apollo and Artemis consoles to provide a simple upgrade path for existing Calrec customers moving to an IP domain.

It also provides 3D immersive path widths and panning for next-generation audio with height and 3D pan controls, flexible panning and downmixing built-in.

AoIP for Comms
IP is now firmly established in most areas of broadcast audio, from mixing to networking to distribution and contribution links. Another area that is now exploiting the full potential of AoIP is communications. 

The major intercom manufacturers—Clear-Com, Riedel and RTS—each has its own take on the technology, while other, smaller developers, or those entering this market more recently, have put their own spin on comms through IP. These include Green-GO, Glensound, Sonifex and AEQ, all of which will be exhibiting at the show.

Clear-Com, which will be exhibiting in the Central Hall, began investing in IP technology during the mid-2000s. Product Manager Kari Eythorsson explains that this was primarily to add remote connectivity for large comms systems in an efficient way. This, of course, proved beneficial during the height of the pandemic when production teams were either operating remotely within a broadcast center or working from home.

“In TV and film production, there was a need to build bubbles for people to work within,” Eythorsson says. “It was possible to reduce the number of people in a bubble [by connecting] with simple wireless systems.” He adds that the AES67 interoperability standard has allowed for expansion of IP systems, while virtual intercoms were facilitated through linking key panels, computers or smartphones over Wi-Fi, LTE or 5G connections.

Riedel also saw IP and remote communications become more in demand because of COVID-19, with the creation of hybrid environments.

 “Due to remote and virtual productions, as well as hybrid workflows, the IP connected intercom is becoming a new normal or a new requirement in projects and applications,” comments U.K. Sales Manager Nacho Lee. Despite the growing ubiquity of IP intercoms, Lee adds that more established technologies, such as matrices, still have a place in comms setups.

Audio may be a smaller percentage of the companies at exhibitions like the NAB Show but it is certainly making an impact with technologies that are significantly changing both the way people work in broadcasting and what TV services can offer their viewers. 

If you’re new to AoIP topologies, you have probably heard it’s best to set up a central core stack of switches in the TOC with edge switches at each studio or group of studios. There are two good reasons to do this: 

1. Should a studio lose connectivity with the central stack for any reason—fire in the TOC, flood, power outage—individual studios can continue to operate independently via their local switches.

Cisco has a topology called Stackwise, where the back planes of multiple switches in a TOC, for example, can be joined at very high bandwidth (somewhere on the order of 160 gigabits-per-second links) in a daisy chain configuration, as shown above. If any one of the switches should drop out of the stack, the other stack members can still communicate with each other. 

2. All local I/O is handled by the edge switches, which provides a more efficient networking and traffic control by cutting down the number of “home runs” from the studio to the central core stack.

Shown above are smaller, 12-port edge switches that handle the local I/O for the studio and have a trunk connection back to the central stack. For additional redundancy, you can take a baseband connection out of one of the local I/O nodes (or BLADEs, in the case of Wheatstone) in the studio and run it into the rack room. This gives you a baseband audio connection directly from the studio to the TOC in case you need to quickly patch programming into the RF chain. 

As a sales engineer for Wheatstone, Phil Owens has a working knowledge of AoIP topologies and best practices for designing an effective IP audio networked studio.

HEBDEN BRIDGE, U.K.—Mobile TV Group is launching its third FLEX mobile unit, which like its predecessors is fully equipped with Calrec technology for its audio setup.

The 47 FLEX unit is stocked with Calrec’s Artemis digital audio console, with audio-over-IP capabilities via Calrec’s H2-IP Gateway system. The truck also features MTVG’s Cloud Control capability.

The FLEX mobile units and dual-trailer outfits, with a 53-foot expanding trailer for Unit A and a 53-foot long Unit B. The infrastructure is based around the Grass Valley Kayenne K-Frame X IP switcher, an Evertz 384-port EXE 2.0 IP router and an Evertz Magnum control system.

MTVG has also announced that it is building the 48 FLEX OB truck, which will also feature Calrec Artemis.

The 47 FLEX OB truck is predominantly used by Marquee Sports Network, the network home of the Chicago Cubs.

HEBDEN BRIDGE, U.K.—Calrec’s Sound Institute is adding to its curriculum, announcing that it is now offering an Audio over IP training course.

Calrec’s AoIP training course will be delivered by the company’s IP network specialist, Dave Sampson, covering topics like networking, how to multicast through to PTP and the importance of intelligent network design. There will be seven lessons in total, four of which are currently available online; the remaining lessons will be released one per week.

“Our goal with this certification is to give broadcasters a comprehensive understanding of not only the value of AoIP but the details of how it works, implementation and day-to-day use,” said Sampson.

To participate in the AoIP training course, users can register or sign-in with an existing Calrec account, watch each video and answer a set of questions, then download and share the official certificate.

Calrec launched its Sound Institute back in August.

For more information, visit calrec.com/calrec-sound-institute. 

LOUISVILLE, Ky.—The newsroom at WDRB TV 41 and WBKI TV 58 in Louisville, Ky., has been getting quite a workout lately.

In a normal week, we produce more than 58 hours of live and local news. At the height of the civil rights protests in June, we were doing back-to-back coverage for five-six hours at a time with reporters embedded with protesters in the hot zones while conducting live interviews on Zoom, Skype and FaceTime.

Already, our new Strata 32 audio board from Wheatstone was easily juggling dozens of feeds at once and bouncing from one newscaster or reporter to the next—some reporting remotely—due to the pandemic.

TIMING IT JUST RIGHT

The Wheatstone Strata 32 is our first AoIP console, purchased months prior to the events of 2020 after a Wheatstone D8 TDM routed console gave us years of service.

We didn’t make a conscious decision to go IP; we just simply decided to go with the next generation of audio mixing available. It was really a stroke of luck that we made the transition to a fully IP audio-networked board when we did.

What we liked about the board right away was that it packed a lot of capability into a compact frame. It has dedicated faders for eight subgroups and two masters along with 32 physical faders that can be layered for 64 channels, all in a 40-inch frame with a central touchscreen interface. The Strata 32 puts all the controls on the surface and uses the Gibraltar IP Mix Engine for networking, which handles digital signal processing.

Because it’s an IP audio-networked board, there are no limitations with fixed connection points on the console chassis itself. Strata 32 has access to all sources in the network and any channel can connect to any audio source or destination, using any preferred audio format at any time. Strata 32 is an AES67-compatible board, which helps make it future-proof.

Once installed, we quickly discovered the Strata 32 makes simple tasks like being able to create mix-minuses for interviews a lot easier. This board has routable mix-minus with talkback interrupt, as well as routable tracking, routable mic presets and—most important to us at the time of purchase—routable EQ, filtering, compressing/limiting and expander gate.

I’ve actually been able to get my staff to embrace dynamics processing as a result. It’s always been a challenge cutting from the sports guys, who really like to project to the point of screaming, and then going to the weather announcer, who has more of a “normal” tone of voice.

Unlike in the past when we had to train staff on voice processing techniques live on-air, with Strata 32’s IP audio networking, I can train staff on the new console offline without every mistake going directly to air. Next, I plan to establish floating presets specific to each sportscaster, newscaster and reporter that will give operators a starting point for processing mics on the fly.

OFF TO THE RACES

We added the Strata 32 console to our news operation last September just as we began planning for the Kentucky Derby Festival’s “Thunder Over Louisville” fireworks and air show and the Kentucky Derby race, content which our 200-person staff produces every year at nearby Churchill Downs.

COVID-19 has bumped the race from the usual April and May timeframe to Sept. 5, and we plan to set up in the usual location near the Kentucky Derby Museum. For this year’s Derby, we’re told that protesters will be on site, so it could be interesting.

Throughout it all, the Strata 32 will be in the newsroom as usual, taking it all in.

David Callan is the media director of production for Louisville’s WDRB TV 41 and WBKI TV 58 TV stations. He can be contacted at dcallan@wdrb.com.

For more information, visit www.wheatstone.com.

As the broadcast industry transitions to IP networks, certain challenges remain in their implementation and operation. Cognizant of potential pitfalls, manufacturers are stepping up with new products that help customers move into a networked future.

The promise of full interoperability is tantalizingly just out of reach, reports Martin Dyster, project director, Telos Infinity. He said high-profile content providers from across the globe are asking product designers and suppliers in the wider media-over-IP marketplace to engage more directly and to work with each other as a team to move the industry forward. “I feel that we are within touching distance of turning the promise of IP into reality, but without further cooperation and teamwork, it remains more challenging than it should be.”

“Where a few of the video standards fall a little short with AES67, and even ST 2110,” said Shawn Anderson, product manager, RTS wired systems, “they’re really focused on the audio transport rather than the control. So one of the things that we helped spearhead was AES70, which is the Open Control Standard, to try and give some people at least a pathway to have unified control across the different systems. But that’s never going to completely replace the individual manufacturers, because they have far more capabilities with their native protocols,” he said.

Phil Owens, senior sales engineer, Wheatstone, said, “With the ubiquitous move to IP, it would be nice to say that all IP streams are the same, but they’re not. Wheatstone is designing our next generation Blade I/O devices with this in mind, allowing for compensation for various streams, flows and essences in the most flexible way possible.”

IMPLEMENTATION

Discovery, registration and connection management, essentially missing from current AoIP standards, means that implementation demands extra time and effort from the system integrator. As Tom Knowles, product manager, broadcast systems, SSL UK, observed, “Today’s standards-based approaches can be complex and time-consuming.”

But technology-based solutions like Audinate’s Dante, supported by SSL’s broadcast equipment, provide a full-stack solution, Knowles said. “Defined technology such as Dante makes AoIP routing a simple reality today while including the current transport-based standards and evolving to include the emerging discovery and connection management standards, future-proofing investment.”

For the foreseeable future, customers beginning to transition to AoIP workflows will likely take a hybrid approach, combining silos of equipment using proprietary protocols interoperating via standards-based networks and gateways.

“The biggest challenge I see is for stations that need to support some of their legacy SDI architecture while transitioning to IP,” said Owens. “Fortunately, the model makes it easy to do a stepped implementation, installing IP ‘islands’ that will ultimately be joined in the final system.”

AoIP has been and will undoubtedly remain a great enabler and is certain to drive the evolution of future workflows.

“The rise of IP and remote production that we’re seeing will naturally lead to fully virtualized workflows becoming more prevalent,” said Dave Letson, vice president of sales, Calrec Audio. “DSP resources will be centralized and allocated to productions as and when they are required, so a production hub can cover any event anywhere by tapping into centralized resources via IP networks.”

Looking further into the future, “Undoubtedly a major driver will be the lower cost and availability of ultra-high-speed data networks, the precursor being the rollout of 5G. Media data transport bandwidth will be ubiquitous,” said Dyster. “The real fun here is figuring out what our user interfaces will look like and how people might use our products differently.”

In this webinar, experts with Audinate and Solid State Logic will share perspective on how Dante helps provide a more powerful, stable environment for the broadcast world. 

 This highly informative Webinar will discuss and demonstrate the following:

In addition to many common questions we hear from broadcasters, representatives from Audinate and Solid State Logic will take questions, time permitting, at the end of the webinar which can be submitted using the attendee console.

If you find you’re unable to attend, or have a question that you would like us to be aware of before January 14, you can send it by clicking here. We will be sure to either answer your question live during the webinar (time permitting)

 A follow-up email with a link to watch the presentations On-Demand will be sent to everyone who registers within 48 hours after January 14.

Speakers

Phil Kurz is a contributing editor to TVTechnology. For more than 30 years, Phil has covered the television and non-broadcast video markets. During that time, he has served as the editor-in-chief of three different industry magazines and helped launch several successful e-newsletters on television technology-related topics. Phil also has written well over 1,000 articles, columns and editorials on technology topics. He is a graduate of the University of Missouri-Columbia with a bachelor and a master’s degree in journalism

Patrick’s career has spanned live production, recording, broadcast, installed sound and automation industries, especially in the adoption of new technological innovations. He excels at explaining not only the theoretical aspects of new tools, but also in bringing practical advice and instilling confidence for new systems to run as planned. Over the last five years, he has been active with Dante licensees and their adoption of AES67 and ST 2110. In his current role at Audinate, Patrick leads training efforts around the Dante audio-video network solution.

Responsible for the strategic planning and realisation of Solid State Logic’s Broadcast product portfolio, Tom has been the driving force behind the company’s shift to networked solutions. He specialises in the design and integration of networked / AoIP systems and works with a comprehensive engineering team to specify, develop, manage and deliver SSL’s Network I/O and System T ranges.

In recent years, during the move to IP in broadcast production, broadcasters have predominately focused on video transport due to it requiring so much bandwidth. However, audio also presents challenges. Compared to video, not only does audio involve a substantially greater number of flows, but it also uses a diverse number of standards in production.

Professional audio over Ethernet has been around since before the turn of the century, with broadcast radio being an early adopter of standardized network technology. Over the years, several competing proprietary approaches and standards for audio over IP have emerged, including DANTE, REVENA MADI (AES10) and AES67. However, the compatibility between the various approaches and even between implementations of specific formats has been a long-standing issue in audio transport and processing. With an increasing number of broadcasters moving to IP in the facilities, this issue of ensuring audio compatibility has become critical.

As broadcasters look to overcome the complexities presented by audio in the move to IP, they must consider the following key issues:

THE STREAMING PLANE

The streaming plane refers to the basic transport of the audio over the network. In that context, AES67 has become key. First issued in 2013, the AES67 standard has been adopted and integrated by most manufacturers, including providers of products based on proprietary approaches. Crucially, AES67 is also the basis of the recent SMPTE ST 2110-30 standard, which means that compatibility on the streaming plane between most of the popular solutions is largely assured.

That said, within the SMPTE ST 2110-30 standard, three levels of conformance are defined—only some of which are currently supported by vendors. The mandatory Level A provides support for 48 kHz streams with one to eight audio channels, at packet times of 1 ms. Level B adds support for packet times of 125 µs. Level C increases the max number of audio channels allowed per stream to 64. The latter means that MADI, which continues to enjoy a lot of popularity, may be carried as-is over the audio network.

What broadcasters often fail to note as they consider their move to IP is that many audio-over-IP systems are currently only able to handle the basic level A. They may also have limitations when it comes to the total number of audio network streams supported, and what combinations of channel count and stream count can be used. So, while the manufacturers can genuinely claim support for SMPTE ST 2110-30, the limited scope of their compliance should be taken into careful consideration when selecting audio equipment as they could place a restriction on the flexibility of the overall workflow.

TIMING

As part of implementing AES67 compatibility, Precision Time Protocol (PTP) version 2, or IEEE 1588-2008, can now be used for timing of the network by the different manufacturers. This also fits with the SMPTE ST 2110-10 standard that mandates use of PTP v2. SMPTE has also published the ST-2059 standard, which generalizes the media clock concept of AES67 to any kind of periodic media clock, including video and timecode.

CONTROL PLANE

The common production environment has many more audio sources than video and an even greater number of destinations. A major sports production could have thousands of audio channels travelling across the network, for example. So, while audio may not necessarily place high demands on bandwidth in an IP network compared to video, it certainly creates a challenge in terms of control and orchestration.

Audio engineers expect to be able to “plug and play” equipment and connect sources and destinations without concerns about protocols and standards. On the other hand, in a broadcast facility, inter-studio routing must be centrally controlled both for the integrity of signals, but also for security and access control.

The apparent strength of some of the proprietary approaches is that they include a comprehensive control plane, whereas standards like AES67 or indeed SMPTE ST-2110 do not define how the streams should be controlled.

PROPRIETARY APPROACHES

While the proprietary control planes are effective on their own, they are not compatible with each other. More crucially, they are designed for a local studio environment (LAN) and therefore aren’t suited to a seamless distributed production environment, such as for big campus or inter-campus use, or for remote production (over WAN).

Furthermore, these control planes rely on audio being made seamlessly available to any equipment in the network by default, meaning no explicit routing of streams is required. This could be a security concern, especially in a distributed, multidepartment or multi-organization environment.

In addition, the fundamental assumption behind this approach that no controlled bandwidth management is needed may be flawed when the size and complexity of the network increases.

MADI TIELINES

One approach to overcoming the issues with control plane interoperability, and address security and stability concerns, is to bridge different IP audio “islands” by using MADI baseband tielines. However, this adds complexity to the management of audio routing in the campus and reduces flexibility and agility. Essentially, this approach largely defeats the purpose and promise of using a converged media network in the first place.

NMOS

The Networked Media Open Specifications (NMOS), developed by the Advanced Media Workflow Association (AMWA), offers a way to address endpoint control for audio in a way that may deliver the true promise of distributed IP production. The standard is now gaining traction in the industry—although its uptake among audio equipment manufacturers is lagging behind that of video equipment vendors.

SDN

In the meantime, the most promising approach to control audio flows in an IP network is to use software defined networking (SDN). This not only provides an easy way to connect diverse sources and destinations, but it also adds a layer of predictability, performance guarantees and security by managing bandwidth and only allowing authorized destinations access to specific audio network flows.

PROTECTION

As production transitions from the LAN environment and into the WAN, and IP audio networking is converged with video networking, audio signal protection is becoming an issue. The SMPTE ST 2022-7 dual path protection standard has now been extended beyond video—to cover any RTP media stream—and provides a great way to ensure audio signal reliability. There may still be compatibility and network addressing issues where different parties need to exchange audio signals, e.g. between different organizations, or simply between an OB van and the live audio system. Broadcasters can address these concerns through IP Media Edge devices and/or SDN controlling which flows can cross the boundary and how—a better approach than using MADI tielines to bridge the gap.

WHAT’S NEXT FOR AUDIO?

The move to audio over IP will make it easier for broadcasters to keep up with the latest developments. For example, immersive audio authoring typically requires up to 127 dedicated audio object channels, in addition to a base surround signal containing up to 22+2 audio channels. Traditionally MADI has been used to interface this high channel count in the production stage, but IP networked audio has higher capacity and can fit all these channels in one cable. Audio over IP is also more flexible with regards to routing and does not require any expensive and dedicated MADI routers when more complex topologies than point-to-point links are required. The use of audio over IP means there is less need for dedicated or custom hardware, allowing for virtualized and flexible workflows. With increased capabilities to adopt new trends such as immersive audio, broadcasters will be able to offer an enhanced viewing experience to audiences.

Olivier Suard is the vice president of Marketing for Nevion.

NEW YORK—The Audio Engineering Society will receive a Technical & Engineering Emmy Award for its work with audio over IP, sharing the award with six partners who were involved in developing the AES67 standard: ALC NetworX, Audinate, Kevin Gross, QSC, Telos Alliance and Wheatstone.

The award is for “Development of synchronized multichannel uncompressed audio transport over IP networks,” and will be given in a ceremony at the NAB Show at the Wynn Encore on Sunday, April 19. The Technology & Engineering Emmy Awards actually are given for developments or standardization in engineering technologies that affect television.

AES67 is a protocol that established a standardized language for audio transport. AES noted that its AES67 standard for high-performance streaming audio-over-IP interoperability was introduced in 2013. It stated: “AES67 compliance allows audio content interoperability between the proprietary IP-based audio networking protocols developed by the Emmy co-winners: RAVENNA, Dante, Q-Sys, Livewire+ and WheatNet-IP.”

AES Fellow Kevin Gross led the AES67 Standards effort and is the chair of the AES Technical Committee on Network Audio Systems.

In the AES announcement of the award, Gross was quoted: “The improvement from audio networking born in the mid-1990s to new IP-based solutions emerged as a simultaneous invention from the honored companies. While collectively this represented a technical improvement, interoperability was not addressed until the AES initiated the X192 project on audio interoperability.”

He thanked the late Steve Church, Rich Zwiebel, Philip Lawo and Andreas Hildebrand as leaders of companies who “understood the potential for a standard to take audio networking to the next level,” and thanked then AES Standards Manager Mark Yonge for mentoring the process.

AES Executive Director Colleen Harper said AES67 “fundamentally changed the broadcast audio landscape and paved the way for recent similar developments for video.”

LOS ANGELES—As AoIP transports have coalesced around the AES67 interoperability standard over recent years, IP communications equipment manufacturers have introduced products that extend far beyond simple intercom functionality. No longer confined to restricted geographic areas with limited partyline functionality, the latest generation of IP comms gear has become a gateway to the LAN or even the WAN, offering worldwide connectivity and the ability to transport all kinds of data, not just audio, and certainly not just voice communications.

MATRIX-FREE

Last year, Telos Alliance launched its Infinity IP Intercom system, which converges voice communication and contribution audio on a single IT backbone, employing the latest standards-based VoIP and Livewire+ AES67 AoIP transport. Infinity introduced a matrix-free paradigm to broadcast communications that enables the system to be scaled up without worrying about how many ports are available. Infinity additionally integrates into existing analog, AES, SDI, and MADI systems via an AoIP interface that could include the manufacturer’s xNode baseband-to-IP interface line.

More recently, Telos introduced Infinity Link, an OPUS codec-based software product that converts WAN-side VoIP to LAN-side Livewire+ AES67. The royalty-free OPUS codec offers manually and auto-adjustable bitrates that are optimized for voice communications over the Internet and for storage and streaming applications.

The way that Infinity Link encompasses VoIP as part of the entirely matrix-free AoIP solution is unique, according to Martin Dyster, Telos Infinity project director.

“With a simple license key, each Telos Infinity device, whether a beltpack, master panel, or expansion panel, can host a group of Infinity Link codecs,” Dyster said. “This enables any IP audio path within its own network to connect across WAN to a Link-equipped codec at any remote location, bridging the local system to the outside world via its own embedded processing engine.”

TUNING OUT AMBIENT NOISE

Riedel Communications has introduced a unique wireless comms service aimed specifically at sports officials. Initially developed with German Football League (DFL) to address their requirements but applicable to many sports, Riedel’s Bolero S wireless hardware connects via AES67 to a local Riedel Artist frame, which interfaces with the Riedel’s Remote Operations Center in Wuppertal, Germany.

At the ROC, engineers at Riedel’s newly managed services group monitor audio quality, RF performance, and battery life while also adjusting talk and listen levels on the officials’ voice-activated headsets in real time as ambient noise levels change.

“This was the result of great synergies between Riedel, DFL and the refereeing department of Deutscher Fußball-Bund,” said Riedel’s Corporate Business Development Manager, Jacqueline Voss. “A stadium with 80,000 cheering fans, like Signal Iduna Park in Dortmund, is a very demanding environment for clear communications.”

The company is reportedly now in discussions with several North American leagues about adopting the service.

The Artist-1024 node packs 1024 non-blocking ports into a 2RU box that can be fitted with up to eight additional software-definable Universal Interface Cards (UIC). These new interface card combine networking, mixing, and management, and can be configured to act as an AES67 or MADI subscriber card, or as an Artist fiber/router/processor card through Riedel’s Director configuration software. The integral mixer on each subscriber card can be scaled from 8 to 128 ports per card and can access all 1,024 ports of the Artist backbone.

MAKING THE MATRIX MORE ACCESSIBLE

RTS recently launched its ODIN OMNEO digital intercom matrix, touting the new product as the only multiformat intercom matrix that works with all industry-leading standards, protocols, and connection types, including AES67, AES70, ST 2110, VoIP, G.711, G.729, G.722, Dante, and 4W analog. It also connects directly to partyline systems, the company said.

The feature set and compact form factor of the 1RU device were designed to make a matrix solution more accessible and easier to use than previously, according to the company, widening the potential IP-based communications user base. ODIN also provides a path for system expansion for smaller users who want to upgrade existing RTS systems to the latest technologies, the company states.

ODIN supports RTS IP-based ROAMEO wireless systems and KP-Series keypanels and users can flexibly define ports to use any of the available connector types. Providing a path to scale up the system, a single ODIN can grow from the basic 16 ports to a maximum of 128 ports and as many as eight ODIN units can be connected via an optical Inter-Frame Link to create a single matrix with up to 1024 ports. Up to 16 analog keypanels may be connected.

Joe Watson, staff engineer for ATK Versacom recently used ODIN to provide comms for “American Idol” auditions. “Everything I needed fit in one small 13-space rack—a huge improvement compared to my multiple-rack setups previously,” he said. “This makes ODIN a great fit for flypack operations.”

EXPANDING IP CAPABILITIES

Several years ago, Clear-Com introduced its LQ Series IP interfaces, compact devices that enabled users to extend their intercom footprint by connecting virtually any type and brand of intercom or audio device via AoIP, SIP, GPIO, 2-wire or 4-wire. The interfaces are a cost-effective method for connecting visiting systems or ad hoc users.

Clear-Com’s professional intercom communication portfolio now encompasses a broad range of IP-based products, including matrix, wireless and mobile IP solutions for all kinds of production workflows, including remote production.

“The breadth and depth of our ever-evolving product line proves how far Clear-Com has come with IP, building on our unrivalled expertise and industry experience to create solutions for any application,” said Clear-Com President Bob Boster.

At the NAB Show, the company further expanded its product line with the addition of the new V-Series Iris intercom Panel, which provides Eclipse HX users with a low latency AES67 AoIP connection. The V-Series Iris Panel can monitor and send communications over three concurrent full-duplex uncompressed audio IP streams, enabling systems to deliver high quality, uncompressed AES67 audio from user to user with significantly reduced latency. Users can deploy up to 64 Iris panels for each high-density E-IPA audio-and-intercom-over-IP card fitted. The card is natively AES67 compliant.

Clear-Com’s IP products also include its FreeSpeak II wireless intercom system. The FreeSpeak II IPT Transceiver features AES67 compatibility, low latency signal distribution, high radio performance, and high audio bandwidth, enabling FreeSpeak II wireless beltpacks to be deployed across a user’s LAN. The IP Transceiver connects to an AES67-compatible IP router and to Clear-Com’s Eclipse HX system via the E-IPA card.

Last year, Clear-Com unveiled the latest version of Agent-IC, a mobile app Apple or Android mobile devices that connecting securely to core intercom users across multiple IP networks over 3G, 4G and Wi-Fi, a potential boon to outside broadcasts or productions where crews are spread across large areas.

MADRID—TRECE is a Spanish TV station that has recently renewed its image. We have simultaneously moved to new facilities located in Madrid, which now comprise two studios and a central TV control. In our previous setup, we used a Kroma digital intercom system with 14 user panels, connected to outside of the facility via an AEQ Systel IP 12 voice coordination system. The all-analog system was connected around a Yamaha 02R console.

In order to integrate the existing equipment into the new facilities, we acquired an AEQ CONEXIA matrix to manage the studios’ audio and intercom signals.

INTERCOM AND BROADCAST AUDIO WITHIN THE SAME HARDWARE

CONEXIA is a redundant and modular audio matrix that simultaneously manages audio for broadcast and four-wire intercom circuits. We have defined a structure comprising five 64-channel AoIP Dante modules, 12 four-dual-channel AES/EBU cards and two additional 8-channel modules for Kroma digital intercom panels. The included switching and processing capability sums more than 300 inputs and outputs.

The 14 series-4000 intercom user panels are reused and connected by digital link, while the new 16 series-8000 panels are connected to AoIP circuits. CONEXIA’s new acoustical design and audio processing—which includes dynamics, equalization and feedback control—provides improved voice clarity. Twenty-four matrix input and output circuits have been routed to two Systel IP VoIP systems and two STRATOS audio codecs for external intercoms.

Three audio circuits are not used for intercom: General purpose digital audio inputs and outputs in the CONEXIA chassis itself; Analog and digital circuits at the four NETBOX 32, which are Dante AoIP terminal devices featuring 16 analog inputs and outputs plus eight dual AES/EBU ones each, located in several locations at the studios; and Dante AoIP links with each studio’s mixing console.

For external communications, two Systel IP 12 systems are connected to external 4-wire inputs and outputs of the CONEXIA matrix. The operator establishes the communication to the remote peers using a Systelset+ terminal.

We have found that Systelset+ greatly simplifies operation, as it includes a small physical support and a wide touchscreen. This allows us to execute most of the commands and operations required during TV production coordination, where most of the calls are generated by ourselves. Simultaneously we can receive and route them using matrix circuits to either the sound mixing console or, more frequently, to a panel or panels group for internal studio communications.

We use this setup mainly to coordinate ENG devices through mobile phones. The journalists receive the program and command return via their earset, and the camera operator communicates using a 4-wire circuit through micro-headphones installed in the phone.

The audio and Intercom system has been enhanced for two studios and a central control. A high-availability audio router with large processing capability has been incorporated. Our existing AoIP terminals have been used to distribute inputs and outputs. The whole installation has been deployed using IP, unifying broadcast and intercom audio. The intercom system has been expanded using new panels while keeping the old ones, and the external intercom system has been enlarged by incorporating very intuitive, efficient and small-sized control terminals.

Daniel Cemillán is the technical manager for TRECE. He can be contacted atdaniel@trecetv.es.

For more information, visitwww.aeq.eu/products/conexia-1.

LOS ANGELES—The universe of AES67- compliant products has expanded over recent years and with the interoperability standard now published in the SMPTE 2110 standards suite has become the “Rosetta Stone” of audio-over-IP. But while digital mixing console system products were among the first to adopt AES67, to what extent are end-users networking compliant products? What barriers to adoption remain, and what can be done to promote more AES67 network implementations?

ALL IP

“In the TV market, AES67 interoperability is part of almost every client conversation,” said Martin Dyster, vice president of business development-TV, TV Solutions Group, The Telos Alliance. Proof that AES67 is ready for primetime can be seen at University of Notre Dame Martin Media Center, a benchmark for what is possible in an all-IP, multivendor environment, where his company supplied the console and routing pieces, Dyster said.

Across the campus-wide media production environment, says Dyster, “All video and audio are transported as IP from the cameras and mics to PCR, and in the case of the Telos Alliance Axia Fusion consoles, there’s not a physical router in sight. Everything is routed at switch level, so it truly is a system achieving what was intended with AoIP all along.”

The installation includes Evertz core video and control elements. “The interchange of around 450 AES67 streams between it and the audio production system is seamless,” Dyster says. “Other vendors in this installation include Riedel, who supplied an AES67-equipped Artist system. Other examples where we’ve connected at AES67 level include products like the Genelec 8340 IP monitors—but any AES67 compliant product should work fine.”

AOIP IS THE GOAL

Don Bird, vice president of Business Development and Marketing for Lawo North America, agrees that AES67 is a basic requirement in almost every customer project. “In some cases, as a natural subset of their request to deploy a complete ST2110-compliant infrastructure, but even in those projects involving an upgrade of existing facilities where AoIP is the goal and the rest of the production pipeline is going to be migrated to IP over time,” he says.

Lawo has been providing AES67 capabilities across distributed and remote production architectures as well as in local studio environments for some years now, according to Bird. “This is nothing new at this point, as customers clearly recognize the savings in cost, flexibility and efficiency of operation.”

Phil Owens, senior sales engineer, for Wheatstone in New Bern, N.C., has observed less enthusiasm for AES67 thus far. “Customers are primarily concerned with their central infrastructure,” he says. “AES67 tends to address add-ons, in most cases. This will change in the TV market as SMPTE 2110 and its audio component, SMPTE 2110-30, become more prevalent.”

Owens has primarily seen client interest in interfaces with a live sound component in a system, and interfaces with some AES67- or Dante-capable intercom systems. “And we have seen smaller local studio implementations, such as interfacing with Dante-equipped mic preamps via AES67,” he reports.

“We are currently working on two university projects that have an auditorium equipped with a Dante-equipped live sound board tied into their broadcast facility via AES67,” adds Owens. “Another university has expressed interest in interfacing a Wheatstone system with an RTS intercom using their Omneo protocol, which is in fact Dante; that will be an AES67 interface.”

Calrec Audio has supplied AoIP networks to several broadcasters for facility-wide installations, reports Dave Letson, the company’s vice president of sales. “More often, though, we are seeing pockets of deployment in POC or smaller applications, so the broadcaster or facility can cut their teeth and gain an understanding of SMPTE 2110 networks.”

The real benefits and savings will come from greenfield builds that can implement the networking facilities from the get-go, Letson believes. Thus, some manufacturers have developed products, such as Calrec’s H2-IP Gateway devices, to bridge proprietary IP infrastructures and create hybrid networks.

That said, Calrec consoles are being networked with products such as Sonifex’s AES67-based monitor pre-hear products and Grass Valley routers, says Letson. “Calrec’s AoIP I/O solutions allow any piece of third-party equipment that is AES67 compatible to talk to Calrec’s I/O boxes and enter the [company’s proprietary] Hydra2 domain.”

BARRIERS TO ADOPTION

Manufacturers almost universally note that AES67’s lack of discovery, control and connection management, which were never part of the standard’s brief, are a barrier to adoption. Lawo’s Bird observes, “We offer a range of products that enable us to provide all-AES67 network infrastructures for our customers, so we don’t really encounter any barriers in that respect.”

“AES67 is an audio-only interface that has to be set up manually by entering a number of stream parameters at both the send and receive ends,” says Owens, “or ‘semi-manually’ by using an SAP utility to advertise those stream parameters to the various endpoints. Control—logic events—has yet to be addressed, although yet another standard, AES70, is a possible solution.”

Letson points to a recent survey by Grass Valley that indicates that IP adoption is slower than previously forecast. Two-thirds of respondents stated a preference for a hybrid approach, with cost a major concern when considering a move to IP. “Cost is obviously a barrier unless you are already planning to replace equipment,” he says, including the costs to deploy completely new infrastructure and train staff, and costs incurred through disruption.

“A fear of the unknown and a feeling that AoIP is brand new and unproven” also militate against wider or faster adoption, Dyster suggests.

The Networked Media Open Specification [NMOS] is currently being finalized. The IS-04 part of that spec addresses discovery, and the IS-05 part addresses connection management, according to Owens. “NMOS will make AES67 much easier to set up and use,” he said.

Calrec, like many other manufacturers, is developing a stream manager, Calrec Connect, which will allow complex network information to be displayed and streams to be created in a user-friendly manner, according to Letson. “This helps to reduce the burden of training needed,” he said.

Bird noted the importance of ongoing training in fostering adoption. “Through involvement in associations like AIMS and various technical recommendation groups and standards bodies we put great emphasis on education and awareness,” he said. “We encourage any customers interested in deploying an AES67 infrastructure to have open dialog with their consultants, integrators, and equipment providers to verify that all components specified are fully supporting the standard in an interoperable way.”

At this year’s NAB Show, I was engaged in a discussion about audio over IP with Phil Wagner, recently named president of Apogee, when he mentioned that no one was really discussing how to merge the technology into existing television environments. My own columns have been pointed more toward implementing it into new builds.

THINGS TO KEEP IN MIND

There are really just a few key things to keep in mind when tossing AoIP equipment into the technology mix of a current broadcast plant, but with SMPTE ST-2110-based facilities coming online and AoIP beginning to manifest itself in a very physical sense, this seems like a good time to look at them.

Just as we’ve done since television audio became digital, everything in the plant needs to be referenced to the same master clock. This has traditionally meant providing either black, tri-level, word clock or AES reference (DARS), but network connected devices—surprise—look for their clock on the network, which means we now need to provide them with an IEEE 1588 Precision Time Protocol (PTP) clock, which has been referenced to the house master. In the past this might have been achieved with a master clock from the IT world, but broadcast device manufacturers now provide master sync generators that include PTP alongside standard reference signals. Reference remains critical because the audio going through those network cables is still digital, so nasty tics, pops or complete lack of audio is the result when the reference clock is missing or incorrect.

Choosing a specific AoIP technology can seem like a daunting task since incompatibilities remain even with the publication of the AES67 standard. Remember that AES67 is not an end-to-end solution but is a defined set of internet protocols that, if followed, will allow audio to pass from one AoIP device to another. The good news is that virtually all AoIP equipment manufacturers ship their newer products with an AES67 compatibility mode of some sort, which ensures that audio will pass when connecting a box with one AoIP technology to a box with a different AoIP technology.

The bad news is that the audio is all that is guaranteed. Sticking with a given AoIP technology, Dante, LiveWire, Ravenna or WheatNet for instance, provide a more end-to-end solution so that all AoIP equipment on the network is aware of each other and information can be shared between them. It also means having full control of those devices, allowing routing and management. Mixing AoIP technologies doesn’t preclude the possibility of doing any of these things, but the lack of information sharing between technologies can make it very, very difficult.

OPERATING IN AES67 MODE

Careful planning is the best course of action before heading down the AoIP road. Since AES67 is the PCM audio transport of ST-2110, any AoIP devices in the broadcast chain will most likely need to operate in AES67 mode, so it is necessary to determine what, if any, features are lost when switching devices to this mode. Since discovery, control and management are not part of AES67, it is also critical to determine how this will be done prior to adding AoIP devices to the plant.

As previously mentioned, sticking with one technology or manufacturer will solve this problem, but this may not provide all the necessary pieces. There are now some third-party software products that promise to make differing AoIP solutions work together despite their differences, but a demonstration of the software controlling actual hardware seems in order before committing to this.

A solid solution seems on the very near horizon however, and it is already part of the ST-2110 package of standards. Obviously, ST-2110 will need discovery and registration, connection management and control, so the Advanced Media Workflow Association (AMWA) Network Media Open Specifications (NMOS) IS-04 and IS-05 have been chosen to solve those needs for ST-2110 devices. This means that AoIP devices in television facilities will use AES67 for audio transport, IS-04 to locate each other, and IS-05 to connect. Actual implementation of these specifications will be through additional software, but this is the current solution planned for SMPTE ST-2110.

A temporary workaround to the lack of discovery, connection and control across all AoIP technologies is to build an AoIP system from a single technology, then isolate this AoIP island from the rest of the plant with the exception of converting signals going in and out of these rooms. This could work for an audio suite (or group of suites) or even live mix rooms where IO to the room is self-contained and does not need to interface with other IO other than for ingest and playout. Outfitting professional broadcast audio consoles (and some semi-pro models) with AoIP alongside SDI, MADI, AES, analog IO as part of the system’s configuration is becoming standard practice. Achieving the same thing in video edit suites and the greater facility will prove a bit more difficult, and will involve more interstitial pieces. But again, advance planning is key, and part of that planning should be looking toward the facility’s future and not just its current state.

Probably the easiest part of adding AoIP devices to a television facility is connecting existing audio and networked audio devices together. There is more to it than just plugging in Ethernet cables for the AoIP devices themselves, since they need to be set up properly on the network. The type of network and its particular setup will determine how complex this actually is. Yet when it comes to interfacing AoIP devices with traditional audio devices there are plenty of solutions available of all types.

Attention needs to be paid to the underlying technology and whether it has an AES67 mode, but not every piece will be part of the broadcast chain.

Jay Yeary is a television engineer who specializes in audio. He can be contacted throughTV Technologymagazine.

BOTHELL, WASH.—The Alliance for IP Media Solutions (AIMS) will join forces with the Audio Engineering Society to host the first audio-over-IP technology pavilion at AES New York 2018, at the Jacob K. Javits Center in New York City, Oct. 17–19.

The AoIP Pavilion will consist of six demo pods, an AIMS demonstration area and a Pavilion Theater, all of which will highlight the latest in AoIP technology and compatibility, including AES67 and associated technologies as they relate to SMPTE ST 2110.

“AIMS and AES created this pavilion because we recognize the growing importance of media networking in the pro audio and broadcast industries,” said Michael Cronk, AIMS board chair. “Whether you deal with networked audio for broadcast and streaming, live sound or studio recording in any number of applications, everyone from audio networking novices to experienced audio and networking engineers will walk away from the AoIP Pavilion with valuable insight into media networking from many of the leading experts in this field.”

AIMS members will participate in the demonstration area, which will feature IP audio devices that include compatibility between AES67 and SMPTE ST 2110. The demo will consist of three racks of interoperable audio equipment exchanging IP streams.

The Pavilion Theater will offer educational presentations from representatives of industry groups, manufacturers and AIMS members. The presentations will cover topics such as the basics of AES67; the relationships between AES67 and SMPTE ST 2110; IP audio-stream monitoring; the anatomy of a full-stack implementation; and successful real-world deployment. The Pavilion Theater’s 30-minute presentations will run continuously throughout the three days of the AES convention and are free to all attendees of both AES New York 2018 and the 2018 NAB Show New York.

The AoIP Pavilion will also include six individual demonstration pods, four of which will be occupied by AIMS members ALC NetworX, Riedel Communications, Studio Technologies, and The Telos Alliance. Several of the pods will be connected to the main AIMS demonstration area, providing a live demonstration of audio networking over IP.

Companies participating in AIMS demonstration area are:

• ALC NetworX

• Archwave

• Artel Video Systems

• DirectOut

• Focusrite Audio Engineering

• Genelec

• Lawo

• Luminex Network Intelligence

• Meinberg

• Merging Technologies

• Riedel Communications

• Solid State Logic

• Studio Technologies

• The Telos Alliance

• Ward-Beck Systems

• Wheatstone Corp.

• Yamaha

To learn more about what’s happening in the AoIP Pavilion Theater visit the AES New York 2018 website.

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DEVENS, MASS.—Comrex will showcase ACCESS NX, the company’s latest addition to the ACCESS codec line at the upcoming NAB Show New York. ACCESS NX has been designed from the ground up to accommodate the technological advances in IP audio transmission. ACCESS NX features an all-new hardware platform that is optimized for running CrossLock.

The company will also feature the Opal IP audio gateway, which provides near-studio quality audio with consumer grade equipment. It is suitable for coordinating call-ins with guests who have no technical experience. Guests can connect by clicking a link. Opal makes it possible to do long interview segments in HD voice quality, without requiring guests to install apps or fiddle with settings.

Comrex will also demo the LiveShot codec, which delivers low-latency video and audio over a range of IP networks, and is capable of providing two-way media simultaneously. At only 3 pounds, LiveShot is versatile for a range of applications, even over challenging connections.

Comrex will be on the show floor in booth N820. Visit the NAB Show New York website for more information.

Audio over IP (AoIP) continues to gain traction with TV broadcast and production facilities for many good reasons: it reduces overall costs, improves workflows, minimizes equipment needs and greatly simplifies and reduces infrastructure. Additionally, AoIP solutions leverage the Internet of Things (IoT) by seamlessly accommodating most commercial off-the-shelf (COTS) products for fast and easy integration, enabling a gradual introduction of islands of AoIP on an as-needed or budgeted basis. All of these greatly increase the overall cost efficiencies of AoIP solutions with significant cost-savings that can be realized when broadcast and business operations are integrated onto a single AoIP platform.

Axia Fusion AoIP mixing console

Another significant advantage of AoIP is that it supports enterprise-level solutions. Since most broadcast and production facilities have already been converted from analog to digital, and employ internal networks, it makes sense for their next step to be the transition of audio to IP. This applies to the broad base of critical operations across TV broadcast and production facilities, including: distributed and decentralized routing; mixer hardware and software; site-to-site connectivity; multiple-line call management; signal distribution and format conversion; real-time audio processing; audio measurement and monitoring; and IP intercom and communication.

Before implementing an AoIP solution however, it is most important to understand the technology and a little of its history to better evaluate the equipment and interoperability protocols available from different manufacturers.

AUDIO TRANSMISSION EVOLUTION

The classic analog studio takes much of its inheritance from the telecommunications industry when only two-way point-to-point audio was possible. The move to digital and the adoption of early standards such as AES3 allowed for audio transmissions to be synchronized within the studio environment and capabilities were later enhanced with a matrix, which enabled more channels and more control. This configuration was replaced by audio over Ethernet for network audio inside the studio and was then followed up by audio on an IP network, allowing audio to move in and out of the studio environment. The next iteration with a centralized server room allowed a distributed architecture for supporting geographically separated locations.

One driving force behind the transition from an analog structure to an IP audio network is reduced costs. Broadcast and production studios can achieve overall reductions with a common transport mechanism for audio, control, messaging and other data traffic such as files and email and VoIP phones. Even more savings are realized through faster installations because baseband to AoIP interfaces (such as Telos Alliance xNodes) can be conveniently placed near the equipment they serve; CAT 5e or 6 cabling can be used to the Ethernet switch and configuration is done using a PC and browser.

STANDARDS-BASED

These benefits provided a concrete value proposition for vendors to develop and market a variety of AoIP protocols. The current AES67 protocol, issued in 2013, ensures the interoperability of the various AoIP protocols available in the marketplace and was rapidly adopted by all major broadcast audio equipment vendors. Today, the most current and yet most mature AoIP protocol is Livewire+ AES67, which is ideally suited for all levels of TV broadcast and production workflow. This latest version picks up where AES67 leaves off by increasing system capacity and reducing costs, among other advantages.

With Livewire+ AES67, all connections including audio, logic, metadata and backfeeds are made using a single IP connection. The full interconnectivity via IP includes linear, low-latency audio, GPIO (General Purpose Input/Output), discovery of audio streams and more sophisticated control and monitoring protocols for instant routing at the destinations. Livewire+ AES67 leverages standard, off-the-shelf Ethernet switches to route audio and one switch can handle thousands of channels. The packets and streams are fully compatible with other IP traffic, so a single cable can carry real-time uncompressed digital audio, device control messages, program-associated data and even routine network traffic including email, file transfers and web browsing.

Livewire+ AES67 is based on a set of communication standards including STP (Scheduled Transfer Protocol) and UDP (User Datagram Protocol), QoS (Quality of Service) and IGMP (Internet Group Management Protocol), among others. STP and UDP are used to provide fast, efficient transmission over the network, while QoS is used to ensure the overall performance of the service (making sure high priority data gets sent first) and IGMP is used to establish multicast groups. In short, by using the right set of well-established networking standards, Livewire+ AES67 assures broadcast users highly reliable and low-latency delivery of audio streams.

HOW IT WORKS

AoIP is intended to run exclusively on a controlled local network infrastructure. It replaces much of the dedicated broadcast infrastructure such as routers, distribution and patching systems that have traditionally been used for managing multiple audio signals as they pass through a production or broadcast facility.

Audio sources are connected to an audio I/O device and the baseband to AoIP interfaces convert the audio to uncompressed, 24-bit/48 kHz digital audio and then packetize the audio into RTP streams. Each audio source is given a channel number and each interface is assigned an IP address for identification and routing purposes. The interfaces are networked to each other over a standard Ethernet IP network.

FUTURE PROOF

AoIP environments can easily adapt to any audio channelization format as well as accommodating control-data channels alongside the audio payload. The packet switching and serial design of Livewire+ AES67 allows greater flexibility and responsiveness in accommodating future changes in I/O configurations if needed. And IP audio networks provide broadcasters the flexibility to grow and change at will. Network capacity can be extended simply by adding more switches.

Finally, when choosing an AoIP protocol, it’s also important to carefully look at the feature set and user interfaces of the gear. Have some of your on-air peo­ple examine the control surfaces and talk through what’s on their wish list to see what capabilities they need right away and what they may want down the road. Drill down on specific pain points or throw out new scenari­os—like handling phone calls or having a producer in a different room. Having a highly configurable system, and one that picks up where AES67 leaves off, will allow you and your team to make clever solutions.

John Shur is president of the Telos Alliance TV Solutions Group.

LOS ANGELES—With the widespread adoption of AoIP networking transports by broadcast audio mixing console manufacturers, their products no longer necessarily conform to the stereotype of a desk-sized piece of equipment covered in knobs, switches, and faders. Instead, mixing, routing, and processing resources may now be distributed across a network, locally within a venue or over far greater geographical distances, heralding an evolution of broadcast audio workflows and a potential seachange in remote production methods.

At-home production, pioneered at events such as the Olympic Games and by Europe’s LiveIP project, a collaboration between the EBU, Belgium’s VRT, and a host of technology companies, now appears poised for adoption in the United States as well. The next logical step in remote production, the at-home workflow—whereby audio from a remote event is routed over an IP network to the home studio for either mixing and broadcast or recording and subsequent post production—is certainly on the radar for console makers.

Wheatstone’s Series Four IP television audio consoleEFFICIENCY AND COST SAVINGS

“This workflow is gaining in popularity as a way to lower production costs, often reducing what would be a truck full of equipment to mostly just the microphones and cameras needed on-site along with a small crew,” explained Lon Neumann, TV audio sales engineer for Wheatstone in New Bern, N.C. “Through IP audio networking, an IP console located at the home studio in, say, Los Angeles, can mix live sound from, say, the Wrigley ballpark in Chicago. IP-connected microphones and preamps deployed in the field can be controlled by the IP-64 or Dimension Three Touch [console system], instead of adding crew and equipment to do the same.”

Wheatstone’s WheatNet-IP is an AES67-compatible network comprising I/O and specialty Blades supporting audio mixing, routing, and control. For example, HD-SDI Blades can source and de-embed SDI audio directly from the camera, de-embedding it, and sending it home as discrete audio.

At the remote location, Blade functionality—DSP supporting audio ingest, processing, and routing—can reduce the need for other devices, saving weight and cost. At the home studio, or anywhere in the network, Neumann noted, “Wheatstone’s IP-64, Dimension Three Touch, LXE or Series Four mixers can turn mics on, control levels, and trigger IFBs remotely. And because most IP audio networks are now AES67-compatible, WheatNet-IP included, it’s a relatively small matter to transport audio from, say, a live sound group that might have one network platform, such as a WheatNet-IP system, and a broadcast truck that has another, such as Dante.”

Calrec’s RP1DIRECT CONTROL

Calrec rolled out its own remote production engine at the 2016 NAB Show. The RP1 unit incorporates local DSP, enabling the generation of monitor mixes and IFBs. Calrec’s offline editing interface allows remote engineers to set up latency-free IFB mixes for commentary or talent monitoring at the venue.

The RP1 embeds transmission audio into existing video transport mechanisms, allowing it to connect via Calrec’s Hydra2 analog, AES, MADI or SDI interfaces. AES67, Ravenna, Dante and SMPTE 2022 AoIP transports are supported in the 2RU unit.

“It gives an operator in a remote studio direct control over channel functions such as mic gains, aux send/monitor mix levels, and fader levels,” said Dave Letson, vice president of sales for the U.K.-based company. “On the console located at the main studio, all these remote I/O resources appear like any other local I/O box, so workflows are exactly the same as any other broadcast.”

Letson added, “Remote broadcasting using RP1 means fewer resources are needed on-site, and controlling audio from a remote console saves money on setup time, crew, logistics and equipment. It’s simple to set up, very easy to use, and enables broadcasters to cover a greater number of specialized events, making it possible to maintain a widening range of content.”

Lawo’s Remote Production kitDIAGNOSTIC TOOLS

Lawo similarly offers a standalone audio processor, the mc² MicroCore, which provides 192 DSP channels and a 512x512routing matrix. The unit may be controlled using Lawo’s touchscreen-optimized mxGUI software as well as by various external studio control systems, such as VSM, Evertz, Quartz, BFE, Pharos, Ross RAP, GV Ignite and Vizrt Mosart. Audio I/O is via AES67 and Ravenna, or over MADI, using Lawo’s optional A__madi 4 interface.

“The device is a perfect option for small-sized automated applications where external studio control systems are triggering the operation, such as in automated newsroom productions,” said Christian Struck, senior product manager audio production for Lawo in Germany. “The processing core provides extensive diagnostic tools for remote maintenance, including log files and http access, just like any mc² console.”

Additionally, Lawo’s 1RU Compact Engine supports AES67 and Ravenna plus a very broad array of analog and baseband-digital I/O. “It’s the perfect product when you’re ready to begin the transition to IP, but not quite ready to toss out everything and begin anew,” Struck added. “This device is ideal for collecting signals from a remote location for long distance remote productions with the possibility of on-site low-latency IFB mixes.”

The Ravenna/AES67-capable Compact Engine’s functions—EQ and dynamics, mixing and routing, delay, and other DSP capabilities—can be remote-controlled from the home mixing console. Additionally, VisTool software running on a tablet may be used to create custom control panels or touch-based mixing consoles for local control of the Compact Engine.

ALL-IN-ONE

Studer’s compact and budget-friendly Micro Series, an all-in-one digital audio mixing system, was initially designed for radio production but is also suited to remote production, according to Harman spokesman David Glaubke. The 3RU chassis houses DSP and I/O (identical to the Studer Vista 1 console), including eight Dante channels.

“The VoiceMix algorithm alleviates high-pressure broadcast situations by automatically mixing and processing up to 12 inputs,” said Glaubke. “This feature is incredibly beneficial in live, unscripted productions.”

A GUI provides full control. “All that is needed to operate the console is a network connection to the core, via the core’s integrated LAN or Wi-Fi hotspot, and an HTML5-capable web browser,” said Glaubke. “Simultaneous control from multiple devices is possible, allowing both on-site and remote control from the studio.”

Conversely, a single laptop or tablet can control multiple cores. But a computer is not even necessary; a screen, keyboard, and mouse may be directly connected to the unit. For those preferring tactile control, Studer offers an optional 6-fader Micro Series surface.

Click on the Image to Enlarge
KEEP IN MIND
Before proceeding, there are a number of items to keep in mind. First is that AES67 could be considered the lowest common denominator for successfully sending audio across a network using IP. Many pieces required to provide a complete AoIP solution— most notably discovery, management and control—are not included in AES67 and were never intended to be. Click on the Image to Enlarge

One of my missions at this year’s NAB Show was to take a deeper assessment of the state of interoperability of audio-over-IP devices in preparation for an upcoming project. I was specifically interested to see whether a complete system could be built using disparate components from different manufacturers, utilizing whatever AoIP technology had been chosen for the product.

The project is for a broadcast facility audio control room with no significant connectivity to the video router, so the key pieces are the digital audio console, intercom system, digital snakes and peripheral equipment. This system needs be configurable in minimal time with minimal hassle. All components must be guaranteed to work together and need to be as interoperable as possible, which means complying closely with AES67.

As a longtime proponent of AoIP it was my opportunity to take a closer look at what could be assembled into a functional AoIP system.

Fig. 1: OSI layers

It is also important to remember that the internet protocol (IP) can be carried over a variety of transmission paths, including Ethernet, ISDN, ATM, FireWire and DSL, making it quite versatile.

The scope of the network itself should be kept in mind, whether it will be a small internal network or one that expands outside the boundaries of a given subnet, building, campus or continent. AoIP equipment working at OSI Layer 2 will suffice for small internal networks, but any AoIP network that will roam farther afield will need to operate at Layer 3.

Finally, and most important for us, the audio settings of the technologies must match.

My mission didn’t get off to a particularly auspicious start, even before the NAB Show was underway. At the DTV Audio Group meeting the day prior to the show, Thomas Edwards, vice president of engineering at Fox, presented findings from his experiments with AoIP interoperability, concluding that interoperability wasn’t working, primarily due to transport differences.

Fortunately, he didn’t leave things there as he went on to elaborate that this situation was just temporary as video-over-IP standards were beginning to gel around SMPTE 2022 and with AES67 as the likely audio transport.

Later on in the day, Tobias Kronnenwett of Lawo, talked about how they were having continual success with AES67-based AoIP technologies on a large scale with Moto GP races, proving that the technology itself is proven in real-world use.

As I started my own examination of each AoIP technology I made the disheartening discovery that simply claiming compatibility with AES67 does not necessarily indicate that the technology is interoperable.

Table 1: AoIP compatibility matrix

In the case of Dante, upgrades made to their devices means that Dante and AES67 streams can coexist on the same network, which is about the same as having roommates— you share the same space, but that doesn’t mean you get along. In Dante’s case, they legitimately want to integrate AES67 streams, but may be hesitant to modify their robust and very successful core technology just to play nice. The players who do claim full compliance with AES67 include Ravenna and Telos, but each have their own methods for discovery, control and management to fill in the gaps.

TECHNOLOGY CHOICES
When we get down to the AoIP equipment itself we find most manufacturers making definitive technology choices, with Dante appearing to be most popular. Among large intercom systems, Delec and RTS/Omneo have chosen Dante, while Riedel has announced support for Ravenna with their Tango intercom to complement their longstanding support of AVB.

Taking a different tack, intercom vendor Clear-Com has announced their own technology called Instant Voice Core (I.V.Core) which claims AES67 compliance and which, perhaps providing the “Rosetta Stone” of the intercom world, purports to have also been tested for compatibility with AVB, Dante and Ravenna.

The world of audio console players doesn’t provide any real surprises with all the name players getting into the game, though in decidedly different ways. Digico consoles and racks can be outfitted with modules that provide Dante AoIP connectivity.

Calrec has very diplomatically decided the best way to accommodate customers is by letting them have the AoIP technology of their choosing, whether it be AES67/Ravenna, AVB, Dante or SMPTE 2022-6.

Ravenna’s parent company Lawo offers AoIP options across a large portion of their products including consoles, routers, racks and interfaces.

Dante solutions have been arriving from SSL for years and, at the 2016 NAB Show, they were showing their System T high-end, large-format broadcast console.

Wheatstone offers AoIP in five of their audio consoles and other products through Wheatnet, their AES67-compatible solution.

Finally, audio industry stalwart Yamaha offers Dante console cards and remote rack units.

Unsurprisingly, no single manufacturer makes all the equipment necessary to build an entire broadcast audio system. Very few have anything close to an entire AoIP ecosystem within their own product lines, something I find more than a little disappointing since it leaves us using an awful lot of glue to stick onto non-AoIP equipment.

The bright spot is that AoIP works and works well if all components use the same base technology. It can also be successful to a lesser degree with equipment from different manufacturers, but complete compatibility can only be guaranteed when all equipment is using the same AoIP technology.

An all-Dante system will obviously work, as will an all-Ravenna system, and any technology operating in AES67 mode will pass audio to other AES67 devices, but the method that manufacturers utilize for discovery, control and device management may mean that the components won’t know about, recognize or pass control to each other.

The challenge for my project at this stage is to choose a base technology that allows everything to work seamlessly while remaining open enough to accept other technologies as AoIP matures.

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

NEW YORK—At CCW next week, Linear Acoustic plans to showcase its latest products for audio processing and Audio over IP. Among the products the company plans to show are the AERO.soft enterprise class TV audio processor and AERO.10 DTV audio processor.

The AERO.soft features the same processing engines as Linear Acoustic’s AERO.100, 1000, and 2000, with an optional Dolby coding while also supporting the AoIP transition. It features customizable I/O, audio and loudness control and logging. The AERO.soft engine can host eight AEROMAX 5.1 or AEROMAX 2.0 processing instances.

AERO.10

Meanwhile, the AERO.10 is an audio processor supporting up to 10 channels of PCM audio via AES, SDI or stereo analog I/O. It also comes with the same AERO.100, 1000 and 2000 processing engine. Tools include AEROMAX loudness and dynamics control, UPMAX II automatic upmixing and downmixing algorithms, as well as ITU and EBU compliant loudness metering and logging. Downloadable 7.5 day rolling and event-based logs include multiple ITU-R BS.1770-3 loudness measurements, as well as true peak values.

The Linear booth, located at stand 1402, will also feature several AoIP products from Telos Alliance partners. This includes PathfinderPRO, the xNode product family, and xSelector. The IP-Audio Driver PC software, which allows multiple channels of stereo PC audio to be sent to and from Axia networks, and the iProbe Audio Network management software will also be on hand.

CCW will take place from Nov. 11-12 in New York. For more information, visit www.ccwexpo.com.

CLEVELAND—The Telos Alliance has announced that the Linear Acoustic SDI xNode is now shipping. The latest in the series of xNode Audio over IP interfaces is able to fully distribute /O and AoIP for HD/SD-SDI signals.

A SDI xNode system can de-embed multiple channels of audio from its dual SDI inputs, taking audio from standard-definition and HD SDI audio channels and translating them to real-time Livewire+/AES67 audio streams. Each SDI xNode manages dual, independent HD/SD-SDI paths, and can de-embed, shuffle and re-embed up to 16 channels of audio on each SDI path, with full routing channels between SDI and AoIP I/O on each SDI interface.

As part of the Telos Alliance xNode series, the SDI xNode also includes analog, AES/EBU, microphone, GPIO and mixed-signal versions from Axia Audio. It also comes with dual-redundant Ethernet ports and a redundant power plan.

AMSTERDAM-Audio over IP connectivity and signal transport in radio is commonplace, but TV broadcasters have been slow to catch up – until now, says Martin Dyster, VP, business development/TV, Linear Acoustic, part of Telos Alliance

Anybody involved in broadcast technology will have noticed the subject of media-over-IP emerge as the ‘next big thing’. IBC2015 looks set to be dominated by IP and while the video side of the business wrestles with formats and protocols, standards based audio-over-IP is already here. Mature, proven and in the case of radio, AoIP has been around since the early 00’s.

TVB Europe has the rest of the story. 

BUENOS AIRES, ARGENTINA—The Dakar Rally, held over several weeks in January, relied on matrix communications systems from Clear-Com to handle communications among racing staff and the media. The rally–which garnered a billion viewers in 190 countries–began at Buenos Aires and ended in Rosario, Santa Fe.

Dakar Rally has used Clear-Com systems for the past 25 years and this year, organizers decided to upgrade their existing systems to Clear-Com’s Eclipse HX 8.0, which enables standard IP-connected V-Series user control panels to operate both as fully functional intercom keypanels and as multichannel audio gateways with no requirement for additional hardware or software licenses. This allowed rally crews to send and receive three audio channels from each V –Series panel at remote positions, including two of the V-Panels that were based in Paris, where the company’s integration partners Audiopole and Astrium are based. This feature extends the Eclipse HX’s existing capability to provide remote matrix intercom connectivity over diverse IP infrastructure at no additional cost.

The rally entourage travelled with two Eclipse systems, and for each stage they were set up either in a building or in a vehicle. The systems were used by numerous staff, including race management, medical and safety regulators and the media, including approximately 40 TV stations.

In this episode of “Inside Audio,” we’ll look at the most important topics from Avid Connect, the DTV Audio Group meeting, and the 2014 NAB Show, the three events that drew me to Las Vegas this year for what I’m now calling “NAB Show Week.”

The NAB Show itself has become so busy with meetings that it is almost impossible to see everything in all the exhibit halls, not to mention the offsite vendor locations. Fortunately, all three of these events turned out to be worth the time invested and valuable information was learned in each of them.

AVID CONNECT
Avid Connect started off with meetings of the new Avid Customer Association (ACA) Advisory Councils, followed by an introduction of the Avid Everywhere platform. Then it moved to breakout sessions for Audio Everywhere, Broadcast Everywhere and Video Everywhere.

Fig. 1: Avid Everywhere stack diagram

The Avid MediaCentral platform is comprised of a lot of component pieces including the Artist Suite, which contains Pro Tools, Media Composer and Sibelius, the Media Suite for asset management and distribution and the Storage Suite.

Supporting these suites is MediaCentral, which provides collaboration and administration. Connectivity comes from MediaCentral UX, which is designed to distribute to pretty much every computing platform and over practically any Internet connection. Finally there are the public and private marketplaces and service and support.

The Audio Everywhere breakout sessions were primarily centered on Pro Tools and the new collaboration features in the application. Pro Tools may be the component of this platform most ready for rollout because it contained collaborative features in a past life. Avid introduced DigiDelivery in Pro Tools 7 to send files between users, but after just a few years the technology was sold and the features were removed from Pro Tools. Collaboration has returned to the program and thanks to faster Internet speeds it may finally be a useful part of session workflow.

The demos, all running in Pro Tools 11 were only pseudo-remote since both participants were in the same room. Collaboration can take place in non-real-time, near real-time or real-time and will partly depend on network speed and bandwidth. So to minimize bandwidth hogging, users can share only the tracks or stems needed for the session. Any plug-ins needed for the shared session can be purchased or rented from the built-in store and then downloaded, installed and activated while the session is open.

To dismiss concerns about plug-in or system compatibility, audio routed to aux tracks can be committed or frozen, then shared. Other real-time collaboration features include text and video chat, with review and approval done by streaming high-quality audio from the session to whatever device the approver is using. Once the session is complete it can be posted to the private or public Avid marketplace for distribution.

The new platform also comes with the announcement that, starting with Media Composer, new pricing models will be available, including subscriptions, floating licenses and full license purchases. In addition, the company is opening up the platform to third-party developers through publication of an API/SDK and they promise to be more open to customers and the industry as well.

For instance, the new PXF session format shuns company branding and is instead named the “Production Exchange Format.” The advisory councils are made up of customers that represent a cross-section of the user base and are charged with helping the company set future direction. Avid appears to be taking steps to address the changing marketplace as well as how it relates to customers, but only time will reveal whether the company is committed to making these changes work.

DTV AUDIO GROUP
The DTV Audio Group meeting is always an interesting event with lively discussions from people working on the bleeding edge of audio for television.

This year’s meeting included presentations on IP audio infrastructure for the 2014 World Cup, AES67 adoption followed by an audio-over-IP interoperability panel, audio for the Sochi Olympics, a 3D audio case study and an overview of advanced music delivery formats.

The discussion I found most interesting was on object-oriented audio. While there was some discussion of immersion, virtually everyone on the panel agreed that when object-oriented audio does arrive it will be to deliver personalization and solve regulation-related issues first.

For instance, rather than riding the coattails of the primary language broadcast, alternate language voice or descriptive audio services could simply replace the primary language while maintaining the full surround image. Value-added services such as director’s commentary could accompany a movie or television program.

Driving all of these new capabilities and services will be more robust metadata than we currently have and it seems unlikely that any system with so many options and channels could function without it. There were some concerns expressed about this new technology including whether broadcasters will be willing to spend money on these new services and whether production of content containing them can be done affordably.

NAB SHOW
When I finally made it to the NAB Show I found an encouraging trend taking shape through the widespread adoption of the AES67 Audio-Over-IP interoperability standard. It seemed to be everywhere and it was a pleasure to see.

Yamaha Nuage accelerator card Dante, Ravenna and LiveWire, although competing technologies, have all announced AES67-compatibility and manufacturers are hopping on board as a result.

One of the most surprising developments regarding AES67 was Calrec’s announcement that they are partnering with Ravenna after holding off to see if audio-over-IP was going to catch on. Joining the Ravenna camp along with Calrec are Digigram, with their PCIe card; and Telos, with a compatibility mode for LiveWire. We also saw AES-67 compatible audio gear from Wheatstone with the Dimension Three and Series Four consoles.

New additions to the Dante partners include Studio Technologies, who introduced the new Model 214 announcer box; and Yamaha, with their Nuage accelerator card.

There are still some audio-over-IP technologies that are not AES67-compatible, but it was nice to see so many of the major manufacturers are now supporting the standard.

It will be interesting to see how these technologies and issues fare between now and the 2015 NAB Show Week.

Jay Yeary is a member of the Avid Products and Solutions Council, the DTV Audio Group, and AES working group SC-02-12. He can be reached through TV Technology or via Twitter at @TVTechJay

One day, soon perhaps, the miles of cabling required to shuttle audio and video from point to point will be relegated to the dustbin. The AES67-2013 standard for streaming audio has now been published, and the upcoming AES Convention (Thursday, Oct.17, through Sunday, Oct. 20, at the Javits Center in New York City) will have a number of seminars dealing with networked audio.

If you’re interested, check out the Networked Audio Track seminar (http://www.aes.org/events/135/networkaudio/) that Tim Shuttleworh, Engineering Manager at Renkus-Heinz, will be chairing. “Networking will influence every aspect of professional audio within the next several years, and it’s already well-entrenched in areas such as installed sound and live sound,” Shuttleworth says. “Networked audio is very rapidly integrating itself into the very fabric of pro audio, in a wide range of applications. As a result, learning as much as you can about networked audio is critical to furthering a career in any corner of the industry. This track was developed with that in mind, to offer a wide range of information to the widest audience possible.”