CLEVELAND—Telos Alliance announced a software update for its Axia Quasar XR and SR line of audio-over-IP (AoIP) mixing consoles and will show the offering at the 2025 NAB Show, April 5-9, in Las Vegas.

Quasar V3.2 offers several improvements for Quasar Soft, the optional HTML5 browser-based remote-control solution for XR and SR consoles. Prior to the update, control was available only with hardware surfaces, the company said.

Quasar Soft users now can use Call Control to operate their Telos VX broadcast telephone systems and Intercom Control to manage a Telos Infinity intercom system. The update also adds 32 user-programmable hot keys in the full monitor section, as well as four user-programmable hot keys on each channel strip to control external devices. The benefits also extend to Quasar Engine RPS, which uses Quasar Soft to control connected surfaces or to create a standalone mixer without the need for a hardware surface, it said.

“Our goal is to provide the same features and user experience to all Axia Quasar users whether they are mixing on a physical XR or SR console, employing Quasar Soft to control a hardware surface, or using Quasar Soft as a standalone mixer,” Telos Alliance senior product manager Luca LaRosa said.

“The Quasar V3.2 update bridges the gap between the hardware and soft surface experience, allowing customers to choose the platform that works best for them, knowing that the features and capabilities they expect will be there.”

Quasar Soft also offers new slider-type faders and dynamic scaling of the browser window, providing an improved interface and user experience with less scrolling, particularly on smaller monitors, tablets and mobile devices, it said.

The company said it expects to become available March 31 from Axia channel partners.

See Telos Alliance at 2025 NAB Show booth N721.

More information is available on the company’s website.

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.”

LAS VEGAS—Allied Esports has multiple esports venues and content production facilities across the world, with HyperX Esports Arena in Las Vegas being one of the primary locations. Part of what we do is host people coming together to play friendly matches against one another. It’s exciting to oversee the many different kinds of technology required.

At the Las Vegas arena we have three floors housing our audio equipment. We use three Lawo Compact I/O stageboxes on these floors, which are tied to the main core of our Lawo system. This setup lets us control everything from two separate services across our compound: one in the back in the production area and one in the front of the house.

GOOD BANG FOR THE BUCK

Creating good sound in this fastmoving environment is only possible with very powerful tools that are easy to use. We chose Lawo mc²36 consoles and Compact I/O stageboxes because of their power and price point. The power inside the system enables us to shift through multiple floors in our venue using fiber that connects everything to one core and two mc²36 consoles.

We get a lot of functionality out of what we feel was a very good investment. The mixing desk is an important element for creating efficiencies: We have up to 12 players competing simultaneously in a broadcast scenario. The mix engineer must be able to work through multiple pages and have backup scenarios for every single pod on stage in the event of failure in the middle of a competition.

MIX AND MATCHING SIGNALS

One of the biggest challenges is mixing player intercoms. Through different mix zones, players can hear both the intercom and game playing. Players can control those items independently—hence the challenge.

I spend a lot of time using the mc²36’s auxes and sums to control and shift audio streams around our arenas. The summing function is a quick way for us to achieve multiple zones of action because we can route different feeds by simply touching different features.

Being able to move signals via digital patch field is a major benefit. Almost anyone can intuitively navigate through the system’s signal functions. We use the Lawo core as our primary audio router. It’s really important for us to be able to navigate through it directly and effectively from the mc²36 console.

With a specific bank of knobs, I can control and select anything, and it comes up on the screen. For instance, when I select a given PC, I immediately see all my gains displayed in a pop-up window. Basically, I just grab something, and all relevant information pops up right in front of me.

When we expand our facilities to global operation, we’ll need the ability to connect all locations together and have one central point of control for multiple simultaneous competitions. That’s where I can see the true power of wide-area-network IP and a native IP console. The ability is there, which is why it makes sense for us to keep investing in Lawo.

I can even log into the mc²36 console remotely and mix a sequence or route signals, which is pretty cool. It’s also exciting that when someone needs help, I can reach out and mix from home. I can’t wait for the ability to grab resources from one location and make them available elsewhere.

The Lawo mc²36 console is powerful at a good price, easy to use, and feature-rich. And that’s what I need.

Alex Markley currently serves as chief engineer at Allied Esports International, and is responsible for the technical activities at the HyperX Esports Arena in Las Vegas. He can be reached atinfo@alliedsports.com.

For more info, visitwww.lawo.com.

Petaluma, Calif. — Yvonne Ho and James Lamb, longtime partners in Point Source Audio, have announced the formation of Wider Pro, LLC, a new company to support their distributed lines that serve the broadcast industry. Based in Petaluma, California, Wider Pro, LLC will be the company to represent North American distribution of Digigram, AuviTran and other brands to be announced.

Point Source Audio will continue to handle all facets—manufacturing, distribution, sales, marketing as well as service—of its microphone and headset products for all markets including the broadcast sector. The timing is right for the transition of its current distributed brands from Point Source Audio to Wider Pro according to Point Source President James Lamb: “I have worked with the teams at both Digigram and AuviTran for almost two decades, and with the pending announcements of new products from both companies, we will be equipped to move into the next phase of growth under the Wider Pro company.”

Yvonne Ho, vice president of sales and marketing at Point Source Audio, and a principal at Wider underscores this point by saying, “Taking care of our customers will always be our first priority, and so we have been and will continue to add highly customer-service minded sales and marketing members to both the Point Source and Wider companies.” The principals equally stress that this pivotal change is intended to accelerate new products to market while maintaining hyper focus on customer service and support for all the brands.

Digigram, based in France, develops digital encoding and streaming links. The company manufactures sound cards, audio processing software and IP audio codecs that are used by broadcast professionals worldwide. Their IQOYA series of IP audio delivery devices are single rack units that can be ordered in a wide variety of configurations. The company also makes the Audioway Bridge line of products for facilitating the transfer of audio streams.

AuviTran, also in France, manufactures several innovative products for broadcast including the Audio ToolBox, a flexible and modular system that greatly simplifies IP networking when dealing with multiple protocols. Their AVDT-BOB is a compact DANTE breakout box with embedded digital signal processing.

Point Source Audio and Wider Pro will be exhibiting at upcoming broadcast industry events including the NRB Convention in Anaheim, March 26-29, booth 722; and the NAB Convention in Las Vegas, April 6-11, booth C2357. To learn more about Point Source Audio and Wider Pro products, please visit www.point-sourceaudio.com and www.widerpro.com

About Point Source Audio Point Source Audio (@PSA_audio) manufactures and distributes worldwide their SERIES8, EMBRACE, and CONFIDENCE collection of miniature microphones—a unique line of headset, earworn and earmounted microphones known for their robust bendable boom and waterproof features. The company also holds two patents for the EMBRACE concealable microphone as well as the patent for the world’s first modular in-ear comms headset that is supporting the hearing health for audio, lighting and camera techs using headsets everywhere from sports to space. Founded in 2004, Point Source Audio is headquartered in Petaluma, Calif. For more information call (415) 226-1122 or visit www.point-sourceaudio.com. Follow the company on Twitter at www.twitter.com/PSA_audio.

About Wider Pro Wider Pro is a distribution company led by a highly experienced and connected team of professionals who are dedicated to delivering imaginative technology solutions to the broadcast industry. Manufacturers can rely on Wider for world-class marketing and sales support throughout the Americas. For more information call (707) 728-8180 or visit www.widerpro.com. Follow the company on Twitter at www.twitter.com/WiderPro.

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.”

LOS ANGELES—Whether over standardized network protocols such as SMPTE ST2110 and AES67 or formats like Wheatstone’s WheatNet, Calrec’s Hydra2, RAVENNA, Audinate’s Dante, and others, audio mixing consoles are supporting remote television broadcast audio productions as they move in new directions. In sports broadcasts, especially, consoles are leveraging networks to implement a variety of applications, from at-home production to ensuring consistent audio quality at every connected mixer, to enabling remote control of, and set-up through, distant desks.

Glen Stilwell, audio operations manager for the PAC-12 Networks, which handles more than 450 at-home productions in a nine-month period each season, observed that IP audio networking is introducing efficiencies for remote broadcast operations. He hopes that new technology from console manufacturers will help him lower costs as the number of productions inevitably increases.

“With our development partners like Calrec, and their new RP1, we’re going to start eliminating the need to carry around big, heavy copper,” said Stilwell, who noted that all the schools in the conference have installed 10-Gig networks. The RP1 unit offers local DSP and an IP connection back to the plant where the mixer has full control of the remote inputs.

“We’re still precabling venues with copper mic cable, per season,” said Stilwell. “As we become more efficient and reduce personnel, we can’t spend the time to do that.” With devices such as the RP1, “We can put down either fiber or, over the long-term, build network drops in the field and just leave them forever,” he said.

PLAYING TOGETHER

But even with networking infrastructures in place, setting up AES67 streams between devices from different manufacturers is not yet as easy as it could be. Even using a common discovery protocol like mDNS/Bonjour, it’s currently necessary to log into a device using one brand’s web UI and create an AES67 stream, then switch to another brand’s web UI, log in, find that stream, and register to receive it.

According to Pete Walker, senior product manager for Calrec several companies are working on the problem—indeed, there is already a solution in the works. “NMOS [Networked Media Open Specifications] is going to take it to the next level,” Walker said. “You can use a central piece of software to control all your devices.” The initial NMOS tool will be IS-04, managing device discovery.

Walker says the protocol is still developing and Calrec doesn’t want to have to wait for it. “Calrec and others are looking at implementing cross-platform control systems, software that discovers all the devices on the network,” he said.

Calrec’s solution, “Connect,” will identify media streams and devices on the network regardless of their registration or discovery protocol. “You can filter, search, then drag and drop devices and interconnect them,” he said. “It visually represents all your streams and workflows.”

At this year’s three-part U.S. Open (Women’s, Senior’s, and Men’s) for Fox Sports, Senior Audio Engineer Dana Kirkpatrick leveraged Calrec’s Hydra2 network to QC the very large number of audio channels coming from the Shinnecock Hills Golf Club course in Southampton, New York. “We were feeding the router and six Calrec Brio consoles in the iso, two digital feature groups, and feeding another Brio, as well as a submix console, a Calrec Apollo,” Kirkpatrick said. “And I was creating two MADI streams for the world.”

The huge network enabled Kirkpatrick to achieve consistency across the attached consoles, he said. “The idea was that I would take everything from the golf course into the Calrec Artemis [in Game Creek’s Maverick remote truck] and EQ and distribute it. When the wind kicked up, I could adjust EQ and it would globally pass to all the mixers, so that the final product had the same EQ and the same gain structure.”

DOLBY ATMOS

Although some features need to be beefed up or even added to meet new production demands, the current generation of mixing console technology has enabled broadcasters in Europe to regularly generate programming, specifically for premium customers, in the Dolby Atmos immersive audio format for the past year or so. In the U.K., for example, all live Premier League soccer matches on Sky Sports, including a double-or even tripleheader on Sundays, have been produced in Dolby Atmos, as well as 5.1, since the start of the season in August 2017.

At a recent month-long competition hosted at multiple sports venues, U.K.-based freelance sound supervisor Pete Mercer produced both a Dolby Atmos mix and a 5.1 mix in a control room at the International Broadcast Center. For each event, Mercer received discrete feeds from the mixer in the remote stadium, including the outputs from a Schoeps 3D microphone suspended above the field of play, created a Dolby Atmos 5.1.4 mix plus six objects on a Lawo mc²96 console, and returned that mix to the competition location for encoding. He and his colleague, Felix Krückels, Lawo’s director of business development, in a second control room on a second mc²96, each mixed one or two games per competition day, Mercer reported.

The two mc²96 consoles were on the same network as the Lawo mc²56 (for stereo production) and mc²36 (for multifeeds) consoles at the competition’s 12 venues. “We could connect via a PC and Lawo MXGui software to check settings, re-patch signals, and even line up by ourselves,” he said. All audio was sent between the venues in each city and the IBC via Lawo’s Commentary system using Ravenna, according to Mercer.

Any console that supports multiple 5.1 buses may be used for Dolby Atmos immersive mixing. In Mercer’s case, the height component had no LFE or center channel information, so was, in effect, a 4.0 bus.

Mercer could monitor the 5.1.4 mix for the UHD feed via Dolby’s DP590 unit. But loudness metering of immersive mixes is still a work in progress and not yet available in consoles. However, said Mercer, “Having a separate value for the 5.1 and the 4.0 height was extremely helpful to make sure the height detail wasn’t too prominent or distant in the mix, especially when mixing extra elements into the 5.1 that didn’t appear in the 4.0 height.”

By now, you’ve heard that AES67 is part of the SMPTE 2110-30 standard and that all the major IP audio vendors offer this audio transport standard as part of their system.

The AES67 format will be useful for streaming audio between the control room and the master control and there’s good reason to believe that it will effectively eliminates the practice of HD/SDI audio embedding/de-embedding with video, and all the hardware that goes along with HD/SDI workflows.

There’s been a great deal of talk about AES67, but that is as far as it’s gone for most broadcasters – essentially a new standard still sitting on the dealer lot waiting for a test drive.

How easy will it be to commission AES67 in your plant?

We decided to take AES67 out for a spin to find out. Earlier this summer we did a trial run of AES67 through a large WheatNet-IP system staged at the Wheatstone factory in New Bern, North Carolina, during what we call a BLADEFest. (BLADEs are the I/O access units that make up the WheatNet-IP audio network). We do BLADEFests periodically to test our system under real-world conditions, and for this one, we added in a few AES67 devices while we were at it.

We added AES67 devices from Genelec, Ward-Beck, Dante, and Axia into the WheatNet-IP system of 12 mixing consoles, 62 hardware BLADEs (or I/O access units), 100 software BLADEs, talent stations, SideBoards, Smart Switch panels, and software including three different vendors’ automation systems. It was all tied together through Cisco and Dell switches.

We ran the system through a series of automated torture tests that included completely rebooting the system and verifying proper operation afterward. We’re happy to say that after more than 160 reboots, not a single connection failure or loss of audio occurred. We also learned a great deal about commissioning AES67 in a plant. Here are a few major findings.

Finding #1. AES67 specifies version 2 of the IEEE-1588 Precision Time Protocol, or PTP. For an AoIP system to maintain timing and stay synchronized with other AES67 devices, the system timing must be controlled by PTPv2. For that to occur there must be some device in the system that serves the role as PTPv2 timing generator to which all other devices slave their timing. Standardizing timing makes it possible to re-synchronize audio to video since every packet in the AES67 audio stream carries a time stamp. Once the PTPv2 clock is running, it’s possible to begin connecting AES67 devices to the network. 

Once the PTPv2 clock is running, the system is licensed for AES67 and it’s possible to begin connecting AES67 devices to the network.

Finding #2. Before connecting AES67 devices, map out an IP and stream multicast address plan with all devices on the same IP subnet. Each AoIP vendor has their own way of allocating addresses and a plan will assure there’s no overlap and that AES67 devices will be on the same IP subnet since multicasting does not normally cross subnet boundaries. Start with the AES67 devices that are least common or least flexible in specifying or changing multicast addresses.

Finding #3. When adding an AES67 device to the network, set the system sample rate at 48kHz unless you know the device sample rate. AES67 does not require devices to support 44.1kHz and many do not. You’ll most likely find this setting option and others in the admin software that comes with the network system. For example, the WheatNet-IP audio network uses Navigator, an interface screen of which is shown below. 

Finding #4. When adding an AES67 device to the network, pay attention to packet timing incompatibilities. WheatNet-IP uses 1/4 ms packet timing for minimum latency. Most AES67 devices also support 1/4 ms packet timing but some, such as Dante, do not. For those devices that do not use 1/4 ms packet timing, we enabled the AES67 1 ms Support option in WheatNet-IP Navigator, as shown below.

Finding #5. Some AES67 devices do not offer an easy way to manually manage streaming details, although these devices often can ingest these details in the form of an SDP file. In our case, we created SDP files by simply right-clicking on the desired source stream’s name in the Navigator crosspoint grid and opening a window that let us create the file. 

Above are a few sample SDP files from WheatNet-IP and Dante showing multicast address, packet timing, sample rate and stream formats.

Overall, commissioning AES67 in most broadcast plants should be a nonevent as broadcasters begin adopting the SMPTE 2110 suite of standards. 

Andy Calvenese is vice president of engineering for Wheatstone.

Editor's note, Finding #1 was updated July 30, per author's request. 

With the AES67, SMPTE ST 2110, AMWA NMOS and ATSC 3.0 standards and guidelines now largely in place, the inexorable march toward all-IP workflows continued at the 2018 NAB Show, with precious few new product introductions not offering some form of networking. New IP-enabled mixing console, interface and intercom products were in particular abundance, no doubt spurred by this new era of digital audio interoperability.

AUDIO MIXERS & INTERFACES

CALREC AUDIO debuted its Brio 12 compact 12-fader mixer, offering 12 physical dual-layer faders with additional virtual faders, 48 input channels and the same bus count as the Brio36, in less than 18 inches wide. New RP1 remote production unit features include a 2-band filter, 4-band EQ, expander and gate, compressor and sidechain EQ facilities, direct output for all remote faders, up to 5.1 wide, and remote auxes via Calrec Assist.

FAIRLIGHT, part of Blackmagic Design’s DaVinci Resolve 15 software suite, debuted an ADR toolset, 3D panners, shared sound libraries, support for legacy Fairlight projects, and built-in cross-platform plug-ins such as reverb, hum removal, vocal channel, and de-esser.

FERROFISH rolled out the PULSE 16 DX, a 16-channel, AD/DA converter with Dante compatibility.

LAWO unveiled its third-gen mc²56 console, incorporating features drawn from Lawo’s mc²96, in a compact, flexible and versatile form factor that address applications ranging from broadcast trucks and studios to live performance and recording.

NTP TECHNOLOGY introduced its new Dante/AES67 Penta 720 2RU modular audio interface with a variety of analog and digital interface options. The new Penta 721 is a 1U digital-only unit, with dual power supplies, that can be configured with various built-in options.

RME introduced its new Digiface Dante mobile audio interface, providing transmission of up to 64 Dante channels and 64 MADI channels via a USB 3.0 connection. The Digiface AVB mobile audio interface enables transmission of up to 256 network audio channels between a Windows computer and other AVB devices within an audio Gigabit network via a single USB 3.0 cable.

SOLID STATE LOGIC’S System T-S300 comes in two versions, 16+1 faders and 32+1 faders, supports Dante, AES67 and SMPTE 2110-30, and can connect to either a Tempest T25 (256 paths @ 48kHz) or a T80 (800 paths @ 48kHz) processor engine.

SONIFEX’S new AVN-PXH12 12x2-channel mixer monitor can simultaneously monitor/mix up to 24 RAVENNA, AES67 or AES67-enabled Dante sources in a compact 1U rack to a speaker/headphone combination.

STAGE TEC unveiled a preview of its Avatus IP mixing console, which features control and display via touch screens with color-coded faders and rotary encoders. IP-connected frame bays allow distributed placement for remote productions. Stage Tec’s new software release 4.5 adds audio processing highlights for Aurus and Crescendo desks. The company also announced at the show that it had joined the RAVENNA partnership.

STUDIO TECHNOLOGIES expanded its Dante-compatible offerings with three new compact, software-configurable announcer’s consoles—the Model 205, Model 206, and Model 208—that offer unique combinations of analog and digital audio resources to create on-air talent hubs for both AoIP and traditional analog audio environments.

WHEATSTONE debuted Screen-Builder 2.0, a new version of the development platform for its WheatNet-IP audio network that includes scripting tools and GUI for developing virtual news desks, producer workstations, control panels, and other virtual interfaces and workflows for the WheatNet-IP networked studio.

AUDIO RECORDING & PLAYOUT

AVID’S new Pro Tools|Ultimate includes the Avid Complete Plug-in Bundle and Pro Tools|MachineControl with cloud collaboration enhancements and Dolby Atmos mixing with supporting of up to 256 audio tracks. The Avid Connect App, available to Avid’s 1.4 million account holders this summer, offers a personalized connection into the Artist Community and Marketplace from computers, mobile devices and within Avid creative tools.

ZAXCOM is now shipping the fourth iteration of its Deva 24-track hard disc recorder, Deva 24 and Mix-16, a control surface for the unit featuring 16 motorized faders in five banks.

AUDIO MONITORING & METERING

RTW showcased its TM3 Smart and TM3-3G SmartTouchMonitors as well as TM7/TM9 TouchMonitors, MM3 MusicMeter and TM3-Primus. It also highlighted its software and plug-in solutions along with Nixer Pro Audio’s PD-Dante, a small portable monitoring and diagnostic system that is now available through RTW’s worldwide distribution.

TSL PRODUCTS unveiled the MPA1-MIXSDI, MPA1-SOLO-8 and MPA1-MIX-8, extending the range of connectivity to its MPA1 audio monitor line.

HEADPHONES/HEADSETS

AUDIO-TECHNICA announced the new BPHS2 broadcast stereo headset for news and sports broadcasting, available in the standard BPHS2 stereo and the BPHS2S single-ear versions, both with a hypercardioid dynamic mic, and the BPHS2C stereo with a condenser mic.

SENNHEISER launched its 300 PRO series, available in August 2018, comprising the HD 300 PRO and HD 300 PROtect monitoring headphones and HMD 300 PRO and single-sided HMD 301 PRO communications headsets.

AUDIO DISTRIBUTION, ROUTING & CODECS

AUDINATE released Dante firmware v4.1 for the Ultimo UXT family of audio networking chipsets, bringing AES67 networking to 2-and 4-channel count products. The company also launched Dante Domain Manager, which allows AV and IT professionals to define specific AV device groupings by room, building and site, allowing for the creation of independent Dante Domains and enabling a single Dante Domain to encompass multiple IP subnets.

ORBAN demonstrated OPTICLOUD Any Content | Any Device, a patented technology addressing the problem of voice intelligibility in the home as well as stereo/multichannel imaging for users who view content on mobile devices.

MICROPHONES

DPA MICROPHONES introduced its d:vote 4099 Instrument Mics with CORE, designed for use with every woodwind and acoustical instrument and featuring versatile mounting/clip options. DPA also unveiled CORE, a new amplifier technology designed to minimize distortion and increase the dynamic range of its d:screet, d:fine and d:vote lines of miniature lavalier, headset and instrument microphones.

SANKEN MICROPHONES’ new compact CS-M1 super cardioid shotgun microphone, for mounting on DSLR cameras, is intended for professional boom pole drama and film location recording applications.

SENNHEISER unveiled the Essential range of headset and lavalier microphones, available from July 2018, including the HSP Essential Omni omni-directional capsule and MKE Essential Omni omnidirectional lavalier microphone

WIRELESS

ALTEROS’ GTX Series L.A.W.N. ultra-wideband (UWB) 6.5 GHz wireless microphone system supports up to 24 channels via an antenna array (up to 32) and enables multiple systems to be used in close proximity with no interference and with no frequency coordination.

AUDIO-TECHNICA introduced its new ATW-T6002x handheld transmitter for the 6000 Series high density wireless system, plus a half-dozen new interchangeable microphone capsules.

PROFESSIONAL WIRELESS SYSTEMS announced the new PWS 4X4 quad antenna distribution for the Shure Axient Digital Quadversity wireless mic system. The new V-IC, 4-zone intercom multicoupler for the VHF range is intended for use with Radio Active Designs UV-1G wireless intercom systems. The new P-812S houses eight independent, passive two-way RF splitters, enabling two-zone transmission for up to four UV-1G base stations.

SENNHEISER’S new 500-p series professional packages for ENG and film work include the ew 512-p G4 Pro portable lavalier mic set, ew 500 BOOM G4 Pro portable boom set and ew 500 FILM G4 combo set variously ship with a camera receiver, bodypack transmitter, MKE 2 lavalier mic or plug-on transmitter and various accessories.

WISYCOM debuted its LFA ultra-wideband active antenna, which operates in the 410-1300 MHz frequency range and features three filters. The new BFA-B/BFA-N ultra-wideband booster can supplement existing antennas, ampliying the antenna signal up to +27 dB over the same frequency range.

INTERCOM & REMOTE CONTRIBUTION

CLEAR-COM’S new E-IPA IP card enables users to build out high density audio distribution and intercom systems by adding 16, 32, 48 or 64-port frame cards to existing Clear-Com Eclipse HX-Delta, Median, and Omega systems, and is AES67-compliant and SMPTE 2110-30-ready. Clear-Com also presented the BroaMan-brand of solutions including the MUX22-IVT/IC422, which integrates Clear-Com matrix-panel intercom connections including audio and control data transport.

COMREX unveiled EarShot, which delivers telephone-based IFB and live studio program audio to field-based remote broadcasts. Up to 30 users can listen to program or IFB feeds by calling into EarShot with a mobile phone. It can handle up to four program feeds or two IFB feeds, but for higher fidelity, smartphone apps can be used to pull the audio in studio quality.

EARTEC introduced the UltraLITE HUB series, which weighs 9 ounces and allows seven users to talk hands-free, in full duplex, simultaneous talk. Also new is Eartec’s UltraPAK, an ergonomic mini belt pack that allows the addition of lightweight headsets to the full duplex, simultaneous talk intercom.

GLENSOUND’S Beatrice R8 8-channel (expandable to 16) rackmount intercom system offers Dante/AES67 networking with 32 digital audio channels, two local analog I/O with two four-wire circuits plus redundant copper and fiber network interfaces.

RIEDEL COMMUNICATIONS introduced its new 1200 series SmartPanel, the RSP-1232HL, a 32-key user interface designed for workflow flexibility, power, and connectivity, including AES3 and SMPTE 2110-30 (AES67). Riedel’s new Bolero Standalone Application is a license-enabled upgrade that allows antennas to be daisy-chained to each other in a line or a redundant ring via a low-latency, synchronized TDM network, among other features.

RTS launched its Odin (OMNEO digital intercom matrix) IP comms system, scalable from 16 ports to 128, with the potential to gang up to eight panels to achieve 1,024 ports; it ships in June 2018.

STUDIO TECHNOLOGIES’ new Model 391 Dante Alerting Unit offers both visual and audible indications that attract operator attention and can be used in conjunction with Dante intercom beltpacks and user stations that support call light functionality.

TIELINE TECHNOLOGY unveiled a new 6-channel IFB and push-to-talk (PTT) intercom solution called ChatterBox. The ChatterBox smartphone app delivers up to six dedicated IFB and push-to-talk communications circuits over cellular networks (3G, 4G LTE, 5G) and Wi-Fi.

TELOS’ Infinity is a distributed IP intercom system based on VoIP and Livewire+ AES67 AoIP transports that integrates with existing analog, AES, SDI, and MADI systems using Telos Alliance xNode baseband-to-IP interfaces and other AES67 partner devices.

AUDIO PROCESSING

JÜNGER AUDIO launched AIXpressor, a 1RU x86 processor-based platform, equipped with a comprehensive range of built-in interfaces, supporting software-defined products that are created using Jünger’s new Flexible Audio Infrastructure flexAI.

NUGEN AUDIO showed the newly released Dolby E Module for the AMB Audio Management Batch Processor, offering improved processing speeds and greater scalability.

THE TELOS ALLIANCE announced a partnership with FiDef to deliver new audio-enhancing technology with a roundtable discussion featuring world-renowned producers, mixers and engineers Gary Katz, Al Schmitt, Frank Filipetti and Ed Cherney.

THX and QUALCOMM TECHNOLOGIES demonstrated an end-to-end THX Spatial Audio workflow using MPEG-H for the delivery of next-generation immersive audio experiences, including features to optimize audio playback over a wide range of consumer devices.

ACCESSORIES

BITTREE unveiled the limited edition 969A-Series 2x48 audio patchbay N96DC, featuring patented front-programmability and offering 96 TT (bantam) connections in a 2RU rackmount chassis.

This is the fourth installment in a series of articles about the newly-published SMPTE standard covering elementary media flows over managed IP networks. This month, the focus is on audio transport, specifically on uncompressed, studio quality audio for broadcast applications.

UNCOMPRESSED AUDIO

The full title of SMPTE ST 2110-21 is “Professional Media Over Managed IP Networks — PCM Digital Audio.” This standard is closely related to, and heavily based on AES67, which is titled “AES standard for audio applications of networks — High-performance streaming audio-over-IP interoperability.” Although the document titles may not be totally self-explanatory, both standards are all about transmitting raw, uncompressed samples of audio signals directly within RTP/UDP datagrams using an IP network.

To understand how audio signals are packed into these datagrams, it helps to remember that each individual channel of uncompressed digital audio signal is created using a fixed sampling frequency and a fixed number of bits per sample. In the case of ST 2110-30, all senders and receivers are required to support 48 kHz sampling, at a minimum. In broadcast applications, 24 bits (3 bytes) are generally used for every sample. So, for a 48 kHz, 24-bit stereo audio pair, the raw audio data would consume 48,000 x 3 x 2=288,000 bytes/sec which equals 2.304 Mbps without any packet headers.

PACKETIZATION

An example of how audio samples are placed into packets is shown in Figure 1, where an HD-SDI signal is separated into individual IP packet streams for each media type. Video is encapsulated using ST 2110-20 and ancillary data is done using 2110-40. Two audio groups are shown — one stereo pair and one set of 7.1 surround sound (which is eight channels uncompressed). Each of these signals is packetized into a separate stream, as shown in the two rows that make up the table. By keeping the number of audio channels at eight or below for each of the two packet streams, maximum flexibility in choosing receivers is achieved, since the minimum (Level A) conformance level for an ST 2110-30 receiver is eight channels.

[Read: What SMPTE-2110 Means for Broadcasters]

Four factors control the way that audio samples are packed into the RTP packets that make up a stream, and are listed in the table within Figure 1 for each audio stream. The four factors are:

· Audio sampling rate. Most broadcast applications today use 48 kHz sampling, so all ST 2110-30 senders and receivers are required to support it. Some applications use 96 kHz sampling, and 44.1 kHz can also be found in practice, so the standards recommends that both additional rates should be supported. Other sampling rates are out of scope for the standard.

· Audio sampling depth. Because IP packets are formatted in bytes, the audio data payload must be an integer number of bytes. Therefore, AES67 and ST 2110-30 only allow 16-bit and 24-bit audio sampling.

· Packet time. This parameter indicates the timespan covered by the audio samples contained in each packet. For example, when 48 kHz sampling is used with a packet time of 1 millisecond, there will be 48 audio samples from each audio channel in each packet. Note that longer packet times increase the end-to-end latency of the audio stream (because it takes longer to fill each packet) and shorter times increase the number of packets in a stream.

· Number of channels. Normally, all of the parts of a multichannel audio signal, such as stereo or surround sound, are transported in the same IP packet stream. Thus, a 5.1 surround sound signal would have samples from six different audio channels interleaved within each packet.

A receiver relies on information contained in the SDP (Session Description Protocol from RFC 4566) in order to properly interpret the packet contents. The SDP data, which typically consists of a few lines of text, can be transported in multiple ways from a sender to a receiver. SMPTE ST 2110-30 does not define a specific way to do this. Instead, methods are being developed by the AMWA (Advanced Media Workflow Association) for use by media production facilities.

Along with the four parameters described in the preceding paragraphs, the SDP values defined in ST 2110-30 provide standard order for the individual channels within the IP packets. Two examples of this are shown in the table in Figure 1, including both the symbols that are used in the SDP file (“ST” and “71”) and their associate channel order in the last column in the table. Symbols and channel orders are defined within ST 2110-30 for other audio systems such as matrix stereo, 5.1 surround and 22.2 surround (symbols “LtRt,” “51,” and “222,” respectively).

SMPTE ST 2110-30 also defines a set of six compliance levels for audio receivers, as shown in Figure 2. To achieve compliance at a particularly level, a receiver must be able to accept any quantity of audio channels in a single stream within the range shown in the table for each combination of packet time and sampling rate. Note that only the “–X” receivers are required to support 96 kHz sampling.

COMPATIBILITY & DIFFERENCES WITH AES67

One question that might be asked about AES67 and ST 2110-30 is: “Can they be made to work together?” The answer is: “Absolutely.” Because ST 2110-30 is based on AES67 and includes multiple “normative” (i.e. required) references, it is very easy to achieve interoperability. That being said, there are a few areas of difference.

First of all, ST 2110-30 receivers are not required to support SIP connection management for unicast audio signals. This is likely not an issue, since large audio networks frequently use IP multicasting to allow signals to be sent to multiple destinations simultaneously. This does mean, however, that ST 2110-30 receivers won’t be able to send or receive VoIP (Voice over IP) calls that use SIP for connection setup. Also note that RTCP (RTP Control Protocol) is recommended for use in AES67 but only needs to be “tolerated” in ST 2110-30 devices.

There are some differences with respect to how PTP (Precision Time Protocol, as defined in IEEE-1588) is implemented between the two standards. One important difference is that RTP clock offsets are not permitted in ST 2110-30. This means that AES67 receivers can work fine with ST 2110-30 senders, but that AES67 senders must not use an RTP clock offset when sending signals to ST 2110-30 receivers. There are also some differences in the specific profiles of PTP that are used in the two standards, however, the permitted ranges overlap so the two systems can be set up to work together.

One other slight difference: ST 2110 requires that every device has an option that allows it to be set in a PTP slave-only mode. When this mode is enabled, the device will never attempt to become a PTP master. This is important in large networks in order to prevent chaos when every device becomes available to take over as PTP master when an interruption in the PTP distribution system occurs. This option is not required in AES67, but should be a useful feature in many products.

HARMONIOUS SOUND

SMPTE ST 2110-30 was developed specifically to make audio as compatible as possible with video. By using the widely-accepted AES67 standard as a base, this new standard allows a wide range of existing audio equipment to harmonize with the rest of the 2110 suite.

Other entries in this series:

SMPTE ST 2110-21: Taming the Torrents

SMPTE ST 2110-20: Pass the Pixels, Please

SMPTE ST 2110-10: A Base to Build On

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Craig Newbury, vice president of sales for Wohler The one constant in the broadcast industry, despite its reputation for being staid and conservative, increasingly seems to be change these days. The growing momentum behind the adoption of IP networking and transports is accelerating the rate of that change, and manufacturers of certain products are beginning to feel the pressure.

Craig Newbury, vice president of sales for Wohler, the San Francisco-based provider of audio and video monitoring gear, noted that, with a previous employer, it would take 18 months to plan and launch a new TV channel. “Now I go back to the same building in London where I used to work and they’re putting channels on-air in two days,” he said. “Customers see an opportunity to monetize content in a different way or in a new territory and they can’t wait six months to be able to do that; they need to do it immediately.

Newbury adds that broadcasters have to recognize these changes and be prepared for them. “The companies that survive will be the companies that can react to customers’ demands,” he said.

THE MOVING FORMAT TARGET

TSL’s MPA1-MixSolo MADI audio monitor The migration to IP is in full swing, agreed Matthew Colman, audio product manager for U.K-based audio monitoring provider TSL Products. With that adoption has come a proliferation of new standards.

“TSL is going to follow the standards,” he said. “But we also support some of the proprietary audio-over-IP formats.” TSL’s PAM audio monitor will add support for AES67-compliant Dante and Ravenna transports at the 2016 IBC Show, according to Colman, who adds that the company’s entry-level MPA products include a Hydra2 option for interoperability with Calrec AoIP signals.

“We remain impartial,” he said. “If a customer comes to us and says they want to do this or that, if the volume is there then we’ll support that. We see a lot more demand for Dante than we have for Ravenna, yet we still offer Ravenna,” Colman offered as an example.

Through aggressive marketing, Audinate has surged ahead of most other AoIP mechanisms, in terms of licensees and the number of products now supporting Dante. Bel, one recent example, introduced its BM-A1-64DANTE rack-mounting confidence monitor last year, and has recently started shipping the unit. Capable of monitoring up to 64 channels from Dante’s 512 maximum, either individually or simultaneously in any combination of inputs, the unit enables both broadcast and live production sound engineers to check the presence of audio on their systems, with at-a-glance front panel visual feedback.

WHO WILL WIN THE RACE?

RTW’s 31900 audio monitor Not every audio monitoring product manufacturer is ready to jump on the IP bandwagon. As Mike Kahsnitz, technical director at RTW, a U.K.-based pro audio monitoring provider, observed in response to the question of whether Dante might ultimately win out over the competition: “It might be AES67; I have no idea.”

RTW is well known for its monitoring products, which principally take the form of very sophisticated metering instruments. The company makes just one audio monitor, the 31900. It’s not a new system, said Kahsnitz, but it handles 16 digital and 16 analog inputs with a variety of Dolby decoding, and incorporates all of the company’s metering capabilities.

Any new monitoring product designs would demand a great deal of thought, said Kahsnitz. Never mind the IP options, just look at the multichannel formats that have been introduced since the launch of the 31900, he noted. “There’s the NHK-driven 22.2 system, Dolby Atmos, Auro-3D,” he said. “It doesn’t make it easy for the manufacturers, because who will make the race?”

Wohler has addressed this environment of constant change with a new hardware platform, iAM, which enables the company to be iterative in its product development through software updates. ���A two-year development cycle just doesn’t cut it anymore,” Newbury pointed out. “You’ve got to be able to talk to customers, react to their requirements, to new standards, to new methodologies and technologies. We can add those to the product over time, and help our customers get a return on investment.”

OUTSIDE THE BOX

With the packetization of audio and video comes its attendant data and metadata, which monitoring manufacturers must also address in their products. TSL, which also makes control systems, is looking forward to the standardization of the VSF Technical Recommendations TR-03 and TR-04 in SMPTE 2110, said Colman. “Perhaps the industry will look to standardize network discovery, communication, and the control aspects of IP. We’re going to make sure that our products are totally aligned, so that we’re at the forefront of what’s going on.”

Wohler, too, is adding third-party control, said Newbury. “We can’t be isolationist. Customers want to be able to use our devices to drive other systems within their workflows. So we’re developing capabilities in our products that allow us to integrate more fully into a much more automated way of working.”

The company’s monitoring boxes and the new iAM platform also enable remote monitoring. “Customers can see the metering through a web browser interface,” Newbury said. “They can consume that data wherever they are, on any device, as long as they can network to the monitoring unit.”

With IP infrastructures, said Newbury, “you’ve got to have a full router inside the box. Our platform will enable those types of applications that allow our customers to not just monitor signals but to be able to utilize those signals within their workflows, which is becoming increasingly important.”

As to the future, Wohler plans to move into software monitoring, according to Newbury. “I can imagine moving towards a scenario where we have a range of different form factors that customers can enjoy, whether they want rack-mounting, handheld, or software devices,” he said.

Wherever the future takes the industry, said Colman, “I think you’ll always have a need for a humble audio monitor. But I think you’re going to see virtualization of television stations, so some kind of software, cloud-based audio monitor is inevitable.”

IP infrastructures are enabling the transport of thousands of channels of audio, observed Newbury. “If you layer in Dolby and audio description the number of audio tracks becomes incredibly large. The management of those audio channels is a real headache.”

Technologies such as MADI could handle that in the baseband domain, he continued. “The IP realm is marketed as being a lot simpler, but in my experience it’s simply not the case,” Newbury said. “Customers are running into a bunch of new problems that haven’t yet got a solution.”

LOS ANGELES—The ability to pass audio over IP between different brands of mixing consoles and other equipment potentially became somewhat easier with the publication of the AES67 standard in late 2013. AES67 may not be as comprehensively functional as other currently available audio networking protocols. Nevertheless, it offers a standardized method by which equipment can share signals, alongside those other networking protocols as well as transports such as AES3 and AES10, otherwise known as MADI.

Television broadcast appeared to reach something of a tipping point in 2015 with regard to the acceptance and adoption of IP-based audio and video workflows and infrastructures. But the industry’s wider AoIP adoption has lagged slightly behind other sectors such as live sound and radio broadcast, where audio networking has been in use for quite some time via various protocols, including AVB, CobraNet, Dante, EtherSound, Livewire, RAVENNA, WheatNet-IP and others.

INTEROPERABILITY TRANSPORT STANDARD
AES67 was not developed to supplant existing AoIP networking platforms; indeed, it lacks certain features found in other AoIP protocols, such as a specified method for channel or device discovery, signal routing and labelling, and system control. But AES67 is essentially a subset of RAVENNA and is supported by Dante and other audio networking protocols, enabling broadcast consoles of various brands to interoperate with a very wide variety of IP-enabled products and, potentially, with each other.

SSL’s System T broadcast audio production system environment, incorporates networked audio, control, and processing based around Dante networking technology. Paul MacDonald, spokesman for U.K.-based Solid State Logic, a Dante licensee, said “AES67 is an interoperability transport standard—control and discovery are intentionally not defined—and as such we don’t view it as a competing technology to Dante, but a complementary standard. However, we think there are developing standards, within the AES and beyond, that have the potential to provide future extensions to standards-based interoperability alongside AES67.”

SSL’s latest broadcast audio production environment, System T, incorporates networked audio, control, and processing based around Dante networking technology, according to MacDonald. “Dante is by far the most widely adopted and, we believe, the most complete and practical audio networking solution,” he said.

Wheatstone, whose Series Four, Dimension Three and other IP consoles route and control audio through the company’s proprietary WheatNet-IP Gigabit Ethernet network, values AES67 for its ability to bridge between its network and others. Phil Owens, head of Eastern U.S. Sales for the New Bern, N.C.-based company noted that AES67 “provides the common synchronization, clock identification and session description we can all share in order to transport audio between our various systems without jitter, delay and data dropout.”

Wheatstone was one of the first to add AES67 to its WheatNet-IP system as another way to “get in and out of our system,” alongside analog, AES, MADI and HD-SDI, Owens said, adding that AES67 is complimentary to WheatNet-IP’s full system approach.

A European plugfest held in October 2014—the first for AES67—led to revisions in the standard, published as AES67-2015, that included some minor updates. At the 2015 AES Convention in New York the Media Networking Alliance (MNA), founded by a group of manufacturers to promote awareness and uptake of AES67, held its first public display of audio networking using the interoperability standard. That demonstration included 22 products from 18 different companies.

Wheatstone products interconnect through the company’s proprietary WheatNet-IP Gigabit Ethernet network. Wheatstone Engineers Kelly Parker and Dave Breithaupt participated in the second AES67 plugfest, held recently in Washington, D.C., and reported on the event on the company’s website. AES67 requires support for both unicast and multicast streaming, the latter requiring SIP, the Session Initiation Protocol that is also behind Voice-over- IP (VoIP) communications, for connection management. According to the report, WheatNet-IP supports SIP, but SIP interoperability was not achieved by every product at the plugfest. It notes that a subsequent AES report “suggests that an SIP technical overview and recommendation be published prior to subsequent AES67 plugfests.” The next plugfest is currently planned for the U.K. during 2016.

Calrec, which has been refining its Hydra2 networking scheme for the past 15 years, is the only broadcast console manufacturer to support all current protocols, including RAVENNA, AVB, Dante, and AES67, according to Dave Letson, vice president of sales for the U.K.-based company. Hydra2 provides a layer of management over all those signals, he noted, providing a multitude of additional functionality like port protection, alias files, virtual patchbays, and access rights. “Uniquely, Hydra2 provides the ability to apply the same management functionality to any supported protocol, providing control and functionality irrespective of which protocol our customers choose for their facility,” Letson said.

A NEW ALLIANCE
An AES standards project, designated X210, is currently addressing an interoperable control protocol. Open Control Architecture (OCA), a technical descendent of AES24, first published in 1999, appears to be on track for ratification as a control and monitoring standard providing interoperability across audio—and, eventually, video— transports. A group of manufacturers has formed the OCA Alliance to actively promote the adoption and standardization of the architecture.

“The lack of an accepted standard is frustrating for both broadcasters and manufacturers alike, and so the main value of these alliances is to provide information and support so that we can all make informed choices for the future,” said Letson. “We are all coalescing protocols into a unified standard and there is still some cautiousness amongst broadcasters and manufacturers in committing to a single standard.”

Lawo has always been aligned to open standards, be it SMPTE 2022-6/-7 and TR-03 in video, AES3, MADI, RAVENNA, or AES67 in audio, or Ember+ in control, according to Andreas Hilmer, director marketing and communications with the German-based company. “We strongly believe in these standardized, open approaches in order to come to best possible solutions with a solid and secure long-term perspective for our clients,” said.

Lawo recently announced that it had joined AIMS, the Alliance for IP Media Solutions, “an independent trade association founded to ensure that all IP solutions brought to market offer complete interoperability and are based on open standards for seamless integration into media workflow environments,” according to an AIMS statement.

“AIMS’s support of open standards and technical recommendations such as TR-03, TR-04, and AES67 afford us an opportunity to eliminate the fragmentation of implementations that our industry has endured over the last 20 years. It’s our big chance to avoid repeating expensive and time-consuming mistakes of the past,” said Hilmer. “In order to support the transition to IP-based infrastructures, we encourage all manufacturers to come together and embrace interoperability and open standards as the best solution for our mutual customers.”

NEW YORK—Television broadcasting has gone through numerous significant changes since the first station went on-air more than 80 years ago, but nothing may prove to have such an impact on the medium as IP technology.

At the upcoming 139th AES International Convention, Oct. 29-Nov. 1 at the Jacob Javits Center in New York, a variety of panel discussions, tutorials and workshops will address the future of broadcast at this crucial point in its progress. The Broadcast and Streaming Media Track, once again chaired by David Bialik, will address a variety of IP-related issues, going to the heart of the matter and asking: “What is a broadcaster?” “Do you have to have a transmitter to be a broadcaster?” and perhaps the most provocative question, “Consider that next year, one company claims they will be the largest broadcaster in the world, and that company is Netflix.”

AUDIO OVER IP
“Streaming Audio from the Cloud,” a panel moderated by consultant John Kean on Thursday afternoon Oct. 29, will reveal some of the ways that streaming, podcasts and progressive file transfer are being used to deliver content, and the system architectures behind them. The program notes also promise discussion of loudness issues in the “Wild West” of streaming.

Several of the members of the Technical Committee on Loudness in Streaming and Network Playback, recently established by the AES, will also discuss regulations during the “Loudness for Streaming” panel later in the afternoon. The CALM Act in this country and PLOUD on the opposite side of the Atlantic have legally codified loudness controls for broadcast TV, but no such restrictions exist for the Internet yet. With the loudness war now moving to the web, AES has produced a new set of recommendations for streaming entities.

In early 2010, Reed Hundt, Federal Communications Committee chairman during the 1990s, publicly revealed that he led efforts to make the Internet the dominant communications medium during his tenure. “We decided in 1994 that the internet should be the common medium in the United States and broadcast should not be,” said Hundt. The reasoning, he said, was that “the United States had 100 percent, essentially, penetration of the telephone networks, and the world’s largest installed base of personal computers—I think it was about 50 percent in that time period.” Allowing that some of the regulations pushed through were “a little naughty,” he added, “We delayed the transition to HDTV, and fought a big battle against the whole idea but we lost.”

Twenty-one years after Hundt’s speech, Bialik has scheduled a panel that asks, “Audio and IP: Are We There Yet?” A panel of experts will discuss the many and various protocols available for audio networking and Internet delivery while also pondering the definition of broadcasting in the IP age. Attempts to invite Hundt to participate came to naught, according to the program notes.

Additional sessions covering audio over IP, with a particular focus on the AES67 standard, Andreas Hildebrand of ALC NetworX will lead a group of manufacturing reps on the possibilities of the standard in “Benefits of AES67 to the End User on Saturday afternoon, Oct. 31. The day will end with the panel, “How Will AES67 Affect the Industry,” and on Sunday, Kevin Gross of AVA Networks will chair a panel featuring Hildebrand and Greg Shay of the Telos Alliance on the equipment-oriented session, “AES67 Interoperability Testing.”

For those interested in how the coming 4K/UHD television standard (and beyond) will affect audio, Fred Willard from Univision will lead a panel on Oct. 31, to discuss “Audio for Broadcast—Immersive, Personalized, 4K and 8K.”

Ed Greene, mixing engineer for “The Price is Right” and “Whose Line is it Anyway?” will deliver the luncheon keynote on Friday, Oct. 30.

IMPACT OF MOBILE AND STREAMING
The theme of this year’s DTV Audio Group forum is “The Accelerating Pace of Change in Television Audio.” Tom Sahara, vice president, Operations and Technology for Turner Sports and chairman of the Sports Video Group, will get the four-hour forum started with a keynote considering the impact of mobile and fixed streaming services, which has been greater and more far-reaching than previously predicted. Among the topics on the agenda is an update on the audio standard for next-generation television and a debate on whether or not the proposed ATSC 3.0 standard currently in development will provide a template for other future audio services.

With viewers of wireless devices now reportedly outpacing traditional cable TVwatchers, “Streaming Facilities—Broadcast Scaled to Internet Feeds” will ponder the particular audio post production demands of streamed versus traditional broadcast content. Chaired by architect John Storyk, principal of WSDG, the panel will compare and contrast audio post requirements for the two delivery formats, including facility designs, acoustic requirements and other issues.

“Audio for Over the Top Television (OTT)” will dig deeper into the topic with a panel of technology providers, content producers and service operators chaired by the Skip Pizzi, director of digital strategies for NAB.

On Friday, Oct. 30, Ed Greene, mixing engineer for “The Price is Right” and “Whose Line is it Anyway?” will deliver the luncheon keynote.

For more information on the 139th AES, visit www.aes.org/events/139.

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.