NEW YORK—Audio systems are being required to handle many more channels than 5.1 plus stereo. In the consumer world, Blu-ray with Dolby Atmos can support multiple channels for an immersive audio listening experience. For those who can’t add more loudspeakers to their listening rooms, solutions such as Yamaha Sound Bar, can provide a 3D-like sound field without the bulk. And major MSO Comcast will be supporting Dolby Atmos in its Xfinity X1 platform starting in 2016.

At the broadcast, production or origination end of the spectrum, we’ll need to rethink our audio systems to accommodate the growing requirements of next-generation audio, an umbrella term Tim Carroll, Linear Acoustic founder and Telos Alliance chief technology officer, prefers to use. This covers object-oriented audio, which can provide immersive audio, personalized audio, visual descriptions, multiple languages, headphone rendering and more.

MORE CHANNELS 

One of the big changes is the multitude of audio channels that will be required. Instead of BNC or XLR connections, multiple channels of audio will be carried on an RJ45 connector and interconnected through an AES67-based network using computer network switches. This will obviate the need for another audio system mainstay, the crosspoint-based routing switcher. No more determining matrix configurations or what sources need to be available to what destinations.

And maybe even more radical, now that most of us have gotten used to SDI where audio is carried along with the video, audio and video will need to once again go their separate—although synchronized—ways.

If you’re thinking of upgrading your plant or creating new, build in an AES67 network. Carroll said, “No matter what happens with audio, an [AES67 network] will be able to handle it. It will be more expensive to go SDI and SDI will probably not be able to handle it.”

The audio network shouldn’t be the same as that used for business computers and devices like printers, Carroll advised, but digital telephony could be carried on an audio network, and can have some advantages in tying in phone calls to an audio production system or server.

Certain off-the-shelf network switches can be used. Carroll said that Telos has tested and certified switches that they stand behind for broadcast-grade networks. The list can be found on the Telos Alliance website.

Reliability for an audio network is key. It must run 24/7 with no signal delays. This may not be something a business IT department is used to, and must be considered when determining who will maintain and monitor the system. Education on all sides can help the transition.

The audio network will need to be synchronized with video and other technical networks, but that has been made easier with the passage of two recent SMPTE standards, ST 2059-1 “Generation and Alignment of Interface Signals to the SMPTE Epoch,” and SMPTE ST 2059-2 “SMPTE Profile for Use of IEEE-1588 Precision Time Protocol in Professional Broadcast Applications.”

These standards are based on the IEEE 1588-2008 standard for Precision Time Protocol (PTP) and written to address sync requirements for professional audio and video systems.

Manufacturers of master sync generators are starting to make PTP available. An example is the Evertz 5601MSC, where PTP is an option.

While it looks like over-the-air broadcast of next-gen audio will need to wait until ATSC 3.0 is completed, streaming is an option today. Broadcasters don’t necessarily have to wait before upgrading their audio systems.

“AES67 can handle multiple audio channels in a single RJ45 connection using network cable,” said Ken Tankel, platform manager, TV Processing at Linear Acoustic. “This is much easier, more space-efficient and likely less expensive than placing multiple BNC connectors on the rear panel of a device. A user should consider what AES67 can offer. This includes audio contribution, audio monitoring, metering, and the ability to control source/destination connections anywhere an IP network can reach. All of these things can be done without needing a dedicated hardware audio router. We think that there are many places to start.”

One of them is the on-air chain. For example: “Linear Acoustic has the SDI x.Node that interfaces two HD/SD SDI I/O to 16 pairs of AES67 I/O. We also have separate audio processing and loudness control with AES67 I/O,” Tankel said.

Fig. 1 shows an example of how AES67-networked equipment can be incorporated in an on-air signal chain.

Fig. 1: An example of how AES67-networked equipment can be incorporated in an on-air signal chain

THE ADVANTAGES OF AES67

Production is a big area that can take advantage of AES67, not only for a single control room/studio complex, but especially across multiple studios and control rooms, as different control surfaces, stage boxes, IP servers and audio processing can all be easily linked, as well as the on-air devices. The same could be said of mobile trucks.

While audio over IP has been in use in radio facilities for over a decade, now that AES67 has been standardized, it should see more use for audio for TV or video. Production audio console systems from such manufacturers as Lawo, Studer, and Wheatstone offer AES67 capability, and Calrec has announced AES67 support with products out in the future.

“It is an ideal time to look at the equipment and manufacturers that you want to use and find out if AES67 interfaces are already available or planned for,” Tankel said. “The industry is in a transition and designing with AES67 in mind will become easier, and more common, as more manufacturers choose to include it in their products. Audio consoles, intercom, audio DAWs video editors, telephone, ISDN and IP-based audio transport, audio processors, loudness controllers and playout systems are just a few of the places that AES67 can find immediate application in the television facility.”

There could be an argument to wait a bit until AES67 becomes a feature in equipment you are interested in. And there are some things in flux. Still to be standardized are such functions as audio naming conventions, advertising and discovery, as well as carrying GPI control, and there is ongoing work in that direction.

No matter what one chooses to do now, it’s important to start becoming familiar with AES67 and networks in general.

Mary C. Gruszka is a systems design engineer, project manager, consultant and writer based in the New York metro area. She can be reached via TV Technology.

SAN ANTONIO, TEXAS— The importance of bass managing control room speakers was made crystal clear to me in May 2004, during the first game of the NBA Western Conference Finals. The audio control room for the event was the first 5.1 room my team had ever built and, in addition to mixing the studio show and wraparound elements in surround, we were receiving a full 5.1 mix from the mobile unit parked at the arena.

Audio Design Labs’ Audio Management Controller
Everything was going well right up to the moment we mixed in audio from the site, which was when we discovered that effects microphones inside the venue were picking up low-frequency energy that was not there before the game started. Unfortunately, this low-frequency content was overwhelming the mix and driving our dynamics processing nuts.

The mix engineer in the truck could not hear the problem and it took us a little while to figure out that the main speakers in the truck were not reproducing those low frequencies because they were each fed directly from a console monitor output. This meant that none of the low-frequency content from the main channels was being sent to the subwoofer where it could have been reproduced.

In a large audio mix room with good acoustics, thick walls and full-range speakers, this would be no problem, but mixing audio in a remote truck is essentially mixing in the polar opposite of that environment. Had the truck speakers been bass-managed, the mix engineer in the truck would have corrected the problem immediately without getting a call from the studio control room.

REPRODUCING ALL FREQUENCIES
Bass management ensures the speaker system can reproduce all frequencies of content in the mix so the engineer can actually hear what is going on. Oddly, making this happen seems to have become more difficult as the need for it has increased. Audio control rooms used to have large speakers that could reproduce frequencies below 80 Hz, even though NTSC television had a limited frequency response of around 50 Hz–15 kHz and all of about 40 dB of dynamic range on a good day.

Nuendo Bass Manager plug-in
Mixing for analog television meant rolling off and filtering high and low frequencies, carefully controlling levels with limiters and compressors, and trying not to trigger any processing downstream that would destroy the mix. NTSC became ATSC and we suddenly had five audio channels with 20 Hz–20 kHz frequency response, 100+ dB of dynamic range and a band-limited low frequency effects (LFE) channel that requires its own speaker.

Unfortunately, the installation of subwoofers did not make up for the fact that mix rooms and their main speakers were shrinking. It seems that as soon as we started creating content with significant low-frequency energy, many facilities stopped installing control room speakers capable of letting us hear it, or building rooms with enough space in them to handle that energy.

The smaller near-field speakers found in control rooms now are much less likely to be able to adequately reproduce low frequencies below 80 Hz properly, if at all, and it is difficult for their frequency response to remain linear at lower frequencies. This creates a real problem for the mix engineer since audio that cannot be heard in the control room will still make it to the home. Worse still is the fact that practically all home surround systems bass-manage automatically, sometimes to make up for the fact that their main speakers are small.

At the creation end of the chain, bass managing means redirecting frequencies that cannot be reproduced by the five main speakers into the subwoofer. Subs get a bad rap as just being there to make things shake, but a properly placed and set-up subwoofer can actually smooth out low-frequency problems in small rooms and, when used for bass management, allows smaller speakers to operate more efficiently because they aren’t wasting power trying to reproduce frequencies they weren’t designed for.

Subwoofer speakers are also necessary for proper reproduction of the point one (.1) LFE channel, which was designed specifically to make sure the low-frequency portions of audio effects are adequately represented. Bass management never adds content to the LFE channel, but it does redirect low-frequency content from the main channels into the subwoofer along with it.

Genelec PD7070A15 subwoofer connectors
IMPLEMENTING BASS MANAGEMENT
Actually implementing bass management is generally not difficult, but can be expensive. The speaker system we used for Western Conference Finals had bass-management features built into the subwoofer, so changing our control room and mobile unit setups meant running the monitor channels to the subwoofer first, then to their respective speakers.

Speaker systems without this feature require a monitor controller or audio console monitoring section with built-in bass management that then feeds the speakers. There are also a number of software options available, either included in digital audio workstation monitoring sections or as plug-ins.

No matter the method, the principle is the same. A low-frequency crossover point is selected—usually 80 Hz—and audio below that frequency is redirected from the main monitor channels to the subwoofer. This ensures that low-frequency content makes its way to a speaker that is capable of reproducing it. Since sound becomes omnidirectional as we reduce frequency it is more difficult (but not necessarily impossible) to pinpoint the source, so using one speaker for low-frequency content works fairly well.

Anyone mixing on speakers that are not gigantic, full-range monsters should give serious consideration to bass managing their speaker system if they really want to hear everything in the mix, especially if a subwoofer is already a part of that system. Those mixing in small rooms would also do well to assess whether bass managing their system along with proper subwoofer placement would help sort out some of the acoustic problems inherent in those spaces.

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

So you are “hearing” about new audio codecs, ones that promise to be more bit efficient as well as high quality, and will be able to deliver the “true” cinema sound experience in the home environment. You probably are thinking, “Didn’t AC-3 5.1 audio do all that?” Certainly it’s been in practice for over 15 years in ATSC 1.0, why add a new audio codec in ATSC 3.0? Before we can answer this question, I thought I’d begin by explaining what is behind this new audio experience.

IMMERSIVE AUDIO

While 5.1 audio envelops the listener, there is only the horizontal dimension to the sound field (Fig. 1). While this can be compelling, the limitations of 5.1 have meant only a “feeling” of being there at the event or in the movie. Due to stereo compatibility issues (including matrix approaches), the surround channels are not used to their full extent. While 5.1 can deliver a richer sound experience than stereo, it has not been able to deliver a sound program with immersive spatial dimensionality.

Fig. 1 (Courtesy, Fraunhofer IIS)

Immersive audio goes beyond 5.1 surround sound and uses more channels to create the sensation of height (sound above). Where this gets interesting is—how many more channels? Even more interesting is an approach called “Higher Order Ambisonics” (HOA), which is a format that does not rely on channel assignments, but rather represents a sound field as a set of audio vectors that can be combined into whatever speaker configuration is desirable. 

But before we leap ahead to HOA, let’s get some definitions out of the way. First, you may have heard about audio objects. Maybe you have experienced this in a theater showing a movie mixed for Dolby Atmos or Auro-3D sound. Similar to stereoscopic 3D movies, 3D movie sound is reproduced so that sound effects can be located in very specific and localized positions relative to each theater patron. Audio objects may be static or dynamic and thus panned around the theater space. One can imagine the creative effects possible with such a system. 

AUDIO OBJECTS FOR TV

Audio objects can be much more than special effects, however. Audio objects can be elements of the sound program that the listener has control over, such as alternate language tracks, coaches’ microphones, race-car radio traffic, etc. These audio objects can be static or dynamic. The interesting feature, from a viewer perspective, is the ability for the viewer to choose as well as modify the base audio program, adding or deleting objects, controlling their level or spatial position, thus enabling interactive audio features of a television program. 

You may be thinking, “Doesn’t our current 5.1/AC-3 system provide for mulitple languages as well as descriptive video tracks?” Yes, but with AC-3, these alternative objects have to be embedded within a separate audio program, so the viewer can choose to listen to the Spanish version, but only with a pre-mixed stereo program, not full 5.1. Likewise, the same problem occurs if a sight-impaired person wants to listen to the descriptive track but has to settle for a mono mix-down of the 5.1 program. 

Audio objects are coded separately from the primary audio program mix and can be rendered (yes, rendered, much as on-screen graphics are rendered on your TV’s on-screen menu/guide) along with the full program mix and placed in the appropriate speaker channels, at a level that can be adjusted by the viewer.

SPEAKER CONFIGURATIONS

I mentioned above that immersive audio expands the number of speaker channels to provide for a true immersive experience. How many? Well the NHK in their UHDTV standard (which is 8K pixels) define their sound program as 22.2 channels. 

Fig. 2 (Courtesy, Fraunhofer IIS)

 The speakers are placed at three levels—high, mid, and low—and includes two subwoofers (the 0.2 designation) (Fig. 2). This can provide theatrical realism, but is certainly not a living room or kitchen type of experience for TV viewers. More practical are the variations on 5.1 and 7.1, which provide four overhead speakers, for immersive presentations of 11.1 or 9.1 speaker configurations. The decoder/renderer as part of these new audio codecs, can create the downmix versions of the full immersive program to fit conventional 5.1 or stereo (and yes mono, too) speaker arrangements. The renderer, if programmed with the actual speaker locations, can also compensate for non-ideal speaker placements. And there are other novel solutions such as the 3D soundbar (an advanced soundbar that can create the effect of surround + height speakers) and “up-firing” speakers that use the ceiling as an acoustic reflector. 

HOA 

High-order ambisonics is a technique used to capture a unique sound field without concern for the final speaker configuration. The degree of precision to which the sound field can be captured is dependent on the order of the HOA process. For each order N, the number of audio signals required to convey the sound field is equal to (N+1)^2. Typically the order is 4, 5 or 6, resulting in 25, 36 or 49 HOA signals. Within the MPEG-H codec, these audio signals are further reduced (usually to six or eight signals) using a linear spatial compression system with very low latency prior to the core MPEG-H audio codec (USAC-Uniform Speech and Audio Coding). The decoder re-expands the spatially compressed signals and formats them for the desired speaker channel configuration. 

ADVANCED AUDIO CODECS

AC-3 has been around since the early ’90s and a lot has happened in audio since then. MP3 caused a revolution in the music world, not because of better quality, but because it reduced audio file sizes where portable players—at first with hard drives and then solid-state memory—could hold thousands of songs. The follow-up to MP3 was the Advanced Audio Codec family (AAC). There are many variations, some with very high quality, some low latency, but all very efficient in reducing the bit rate of the sound signal. 

The latest codecs being proposed for use by the ATSC are MPEG-H and AC-4. MPEG-H was developed by three companies— Fraunhofer IIS, Technicolor and Qualcomm—and has become an ISO standard. AC-4, developed by Dolby Labs, has been standardized by ETSI. Both codecs have similar features, such as audio objects, personalization, rendering at the player, with high compression efficiency (lower bit rates for high quality audio). MPEG-H also has a voice codec as part of the USAC core codec, the ability to transmit a sound program in HOA and a layered coding option. AC-4 has the ability to encode Atmos movie soundtracks into a broadcast program by grouping sets of objects for efficient encoding. Both systems have rich program metadata that provide for the channel definitions, loudness parameters, advance dynamic range control settings and the steering of dynamic objects. 

FINAL WORDS

It’s an exciting time for television and immersive audio can provide a richer fuller viewer experience, regardless of how they choose to listen. Immersive audio can be rendered to the speakers available and provide interactive control by the viewer to tailor the program to their taste. With full immersive sound and 4K (or 8K) displays that cover the full field of view, TV viewers will not only be immersed but will enjoy features unavailable today.

Editor's Note: In an early version of the story we had the equation for the sound field as (N-1)^2 instead of (N+1)^2. The equation and subsequent results have been changed.

Jim DeFilippis is CEO of TMS Consulting, Inc., in Los Angeles. He can be reached at JimD@TechnologyMadeSimple.pro.

As the shift to 5.1 and beyond surround sound mixing rooms continues—with the potential for immersive audio on the horizon—there have been some new thoughts on the internal acoustical design. The ultimate goal of providing a neutral listening environment where critical judgments can be made on the quality of sound sources and the overall mix doesn’t change. Where the rethinking comes in is how to handle the internal sound field to accomplish this goal. So, all the usual suspects of good design—solid construction, adequate room volume, low-frequency modal control, symmetry, noise and isolation control, loudspeaker choice and placement and mix position placement—still apply. I’ve covered these topics in the past, and they have recently been revisited by my TV Technology colleague, Jay Yeary (Exploring Audio Control Room Acoustics).

THE RFZ CONCEPT
Since this new approach builds on current best practices, a brief review may be helpful. Ever since the concept of Live End- Dead End control room design emerged in the late 1970s, the idea was to keep early reflections away from the mixing position by making the front end of a control room absorptive and the rear diffusive. This concept was soon expanded and enhanced by thinking of a reflection-free zone (RFZ) around the mix position.

Absorption, while effective in controlling reflections, could end up creating a “dead” sounding environment, especially if too much was used. The RFZ was achieved not only by properly specified and situated absorption, but in the room geometry with splayed walls re-directing sound energy away from the mix position to the diffusive rear wall or walls. If the angles were correct, the splayed walls could actually be reflective, thus keeping more sound energy in the room.

At the same time diffusor science and technology took a giant leap forward in the early 1980s with the introduction of the Quadratic Residue Diffusor (QRD) by Dr. Peter D’Antonio and the company he cofounded, RPG Diffusor Systems Inc.

Studio C control room at Blackbird Studio Using diffusor concepts based on number theory as advanced by Manfred R. Schroeder, it was possible to predict how sound would scatter upon hitting the diffusor. This allowed for the design of units that provided more uniform scattering of sound in space and time within the designed frequency range than other forms of diffusion, such as poly-cylinders or pyramid shapes common in use at the time. (Since the QRD, there have been further diffusion improvements by RPG.)

Timing and the control of the diffuse sound level factored into the RFZ concept. Measurements, experimentation and the latest research on human sound perception and localization indicated that about a 20-millisecond time difference from the arrival of the direct sound from the loudspeakers to the first reflections would be needed to create an RFZ. This meant that the zone was effectively anechoic (without echoes or reflections) during this time period.

To keep the room from being a truly anechoic space—which is very unnatural for listening—broadband diffuse, rather than specular, reflections would be allowed to arrive at the mix position after about 20 milliseconds. The level of this diffuse field would be quite a bit lower than the direct sound, around 30 dB, and decay fairly evenly over time. The QRDs, new at that time, provided this type of diffusion in a small space.

This level and spacing of diffusion didn’t interfere with the ability to critically evaluate the direct sound. Properly implemented, a control room with an RFZ provided good stereo imaging, and not necessarily in just one “sweet spot,” but across the width (or a good part) of the mixing console.

LEDR TEST SIGNALS
Imaging was not necessarily relegated to just the lateral plane. A test tape called LEDR, a trademarked acronym for Listening Environment Diagnostic Recording developed by Doug Jones, contained a series of specially designed synthesized and filtered sounds. When played back in a well-designed RFZ room over a pair of loudspeakers, these sounds would appear to travel not just left to right, but up, over and behind the listener in the RFZ.

These synthesized sounds simulate reflections off the pinna of the ear (the outer ear) combined with direct sound. This results in comb filtering in the frequency response. As sound moves around a head, the comb filters change, providing auditory cues as to location. This was brought to the audio community’s attention by Dr. Carolyn “Puddie” Rodgers in the late 1970s and early 1980s.

(There are other auditory localization cues that include level, phase and time of arrival differences between the two ears.)

Front and ceiling diffusive surfaces at Blackbird Studio The LEDR test signals are now online at www.audiocheck.net/audiotests_ledr.php. The effects can be heard on headphones and downloaded to check out a room. In a control room situation, if early reflections interfere with the direct sound, especially within the first 20 milliseconds of arrival, then the resultant comb filters can conflict with pinna localization cues, and the effects on the LEDR recordings won’t be heard.

The RFZ concept (and the LEDR test sounds) was developed in the age of stereo recording and monitoring. Could there be improvements for multichannel audio control rooms?

There are proponents of eliminating the reflection-free zone and instead create an overall diffuse sound field throughout the room, including the mixing position. There are different opinions as to the level and characteristics of this sound field so that the ability to critically perceive the direct sound isn’t hampered.

One recording studio put this concept into practice: Studio C of Blackbird Studio in Nashville, Tenn., which was designed by noted recording engineer, producer and audio equipment designer George Massenburg who collaborated with RPG on this project.

Studio C, based around a Digidesign ICON, has wide-bandwidth diffusive surfaces on all walls and ceiling. But these aren’t garden-variety off-the-shelf diffusors.

The large-scale, wide-bandwidth, two-dimensional diffusors were custom-designed and fabricated from a series of one-inch square (in cross-section) block pegs—each one a different height—that stick out from the wall in a specific arrangement. Over 130,000 different peg heights, with no two the same, were used to create the diffusive surface. The room treatment also included diffractals on the ceiling and low-frequency resonators (dampened metal plates).

The diffusion this treatment creates is very dense, with a level around 30 dB below that of the direct sound, and an even decay rate over time.

RPG is calling this concept “Ambechoic,” a trademarked name for ambient anechoic. The room as a whole has ambient energy due to the extensive wide-band diffusion, but at the mixing position it’s reported that there’s precise imaging and an extended “sweet” zone. In addition, monitoring can be done at lower levels.

Why does this work? It seems more research needs to be done, but the ear/brain integration time may play a pivotal role.

Studio C is a fairly large mixing room at more than 24,000 cubic feet, so the space was available to accommodate the diffusive surfaces with some of the elements sticking out more than three feet from the wall.

According to D’Antonio, “The Ambechoic design is really intended for larger rooms than broadcast because you have to develop low-level diffuse reflections, which is difficult when close to a boundary. 2D diffusors are also necessary to omnidirectionally distribute the energy. The idea is to provide a uniformly diffuse environment.”

Could this concept be adapted for smaller control rooms? It’s something to think about; and it will be interesting to see how this develops further as designers explore the possibilities.

Mary C. Gruszka is a systems design engineer, project manager, consultant and writer based in the New York metro area. She can be reached via TV Technology.