In technological terms, a trend can have several different interpretations or even stages. It can be applied to an emerging audio technology that is not yet fully mature or adopted. Once uptake is underway a trend becomes current, as an increasing number of broadcasters begin to follow the initial adopters until that particular technology is firmly established in the market.

This does not happen immediately, which means an innovation can keep appearing at NAB Show for several years and seem like it is not that new at all anymore.

The reality is that implementation often takes a long time, as broadcasters’ replacement cycles or plans for new facilities come around and they get ready to install what are still the latest systems. This point is illustrated by Costa Nikols, strategy adviser for media and entertainment at Telos Alliance, who quotes Devoncroft Associates research showing that upgrading the audio infrastructure, which had been well outside the “Top 10” of broadcast priorities, was now in the “Top 5.”

Changing Priorities
“There’s a lot of investment going into improving audio quality, incorporating Next Generation Audio [NGA],” he says, referring to an umbrella term to define immersive and personalized audio experiences. “That also brings all these immersive experiences and customization and personalization features to the table, but how long is that going to take?

“I think we’re just at the beginning of something that’s going to be exciting and wonderful,” Nikols adds. “But if you look at the early adopters, it’s generally the national broadcasters that have adopted SMPTE ST 2110 and 2022-6 initially. Slowly, it will make its way out to the station groups and Tier 2 and Tier 3 broadcasters.”

The most visible NGA system is Dolby Atmos immersive audio. Atmos is specified by many of the leading streaming platforms, including Netflix and Amazon Prime Video, but it’s not yet a standard part of linear broadcast transmissions. But, as Nikols points out, there is a growing market for enhanced sound that goes beyond TV and into mobile and other areas.

“Around the time of last year’s NAB Show, the number of devices that supported Dolby Atmos was 1.5 billion,” he says. “The most recent figure is over 3 billion, so it pretty much doubled by the beginning of this year. But the market, as far as the content providers are concerned, is not really delivering on the capabilities of the consumer electronics that are already deployed.”

Forward-Looking Feature
Professional equipment is also ready for the wider rollout of NGA and immersive systems, with all leading mixing console manufacturers accommodating technologies such as Dolby Atmos on their new and current desks.

“Global events often incorporate NGA formats like Atmos as they tend to push the technological boundaries,” Christian Struck, senior product manager for audio infrastructure at Lawo, says. “NGA has been a topic of interest for over a decade and we have integrated these capabilities into our systems, both physical and virtual. There is a gradual increase in demand for immersive audio and it remains a forward-looking feature for people exploring advanced audio workflows. Real-time metadata transport as defined in SMPTE 2110-41 will also help push this.”

Henry Goodman, director of product management at Calrec Audio, agrees most audio consoles now on the market enable immersive mixing, but does not see it as a priority for many viewers.

“Whether the public is taking advantage [of what immersive programming is available] is still quite a big question,” he says. “And the vast majority of distributed content is still not immersive.”

A parallel situation exists with the adoption of Audio Over IP (AoIP), Goodman adds, noting that Calrec is still selling consoles that are not SMPTE ST 2110-compliant.

“AoIP is getting towards being the norm, but it’s not across the board yet,” he says. “And you’ve also got different types of AoIP. On smaller systems, people are using Dante for connectivity and while it’s IP, it’s not a 2110 infrastructure.

“For larger, major investments people are certainly building AoIP infrastructures, but there’s still some way to go [because] if you’ve got a lot of investment in baseband audio and video, it’s not a slam dunk that when you upgrade one of those studios you would necessarily choose to go IP because of the impact it has on the rest of the system,” Goodman says.

The Importance of Having an IP Backbone
Phil Owens, senior sales engineer for Wheatstone, agrees that although old technologies continue to play a part in audio installations, there is now more of a swing towards newer formats.

“While legacy equipment still needs to be incorporated, the backbone of pretty much all new systems will be IP,” he says. “The flexibility of an IP network is the primary driver for this. As full 2110 networks become more widespread, stations are adapting the IP approach even if they don’t implement 2110 right away. Having a basic IP infrastructure keeps that door open.”

IP also plays a key role in remote and distributed production, which, again, have been a consideration for over five years and, after getting a boost during the pandemic, are growing in scope and reach.

“Distributed production is not just about saving costs, it’s about utilizing the facilities that you’ve got more effectively and more efficiently,” Calrec’s Goodman observes. “And we’ve had quite a lot of discussions with broadcasters about them widening their content offering with more niche-type sports because, by looking at doing things with remote production in a more distributed way, they can then justify covering sports that they would never have been able to justify covering before.”

Lawo’s Struck agrees that the rise of remote production—or REMI (remote integration)—models are another significant trend. “These centralize resources while enabling distributed production across multiple locations,” he says. “This approach offers flexibility in how equipment is bundled and deployed. Automated productions also remain a consistent focus. They are particularly well-suited for news programs and scripted productions, though they are not applicable to more complex, dynamic productions.”

Of all current trends, artificial intelligence (AI) will have a high profile at NAB Show, with initiatives such as the PropelME startup hub, showcasing companies using the technology to change how the broadcast, media and entertainment market operates. While most on the audio side of broadcasting agree AI will make an impact on sound, the view is that—like other technologies before it—this is not happening quite yet.

“When we talk about AI, sometimes it’s just confused with automation,” Nikols says. “Automation technologies have been around for a while and they allow you to have complex workflows with lots of ‘if-then’ decisions.

“And for the most part AI is that,” he added. “But Generative AI coming into the game allows some really cool stuff to happen with auto-mixing and can determine what content really needs to come up and how to adjust for all the other different channels that don’t necessarily need to be part of the main audio all the time.”

As familiar as some—or all—of this year’s audio trends are, they are playing an important role in shaping the future of broadcasting, even if their full implementation is not happening just yet.

SAN FRANCISCO—Last month's article discussed the theory behind high-end audio coding. The second half of this series will define broadcast quality and tandem coding losses, and look at their effect on perceived quality. (To read part one of this article series on understanding the theory behind high-end audio coding, go to Next-gen audio coding - Part 1.)

BROADCAST QUALITY
Being familiar with standardized test methodologies and knowing how to interpret their results will significantly aid in understanding broadcast quality. Unknown to many, there is an ITU recommendation that defines the requirements for audio coding systems in digital broadcasting. ITU-R BS.1548-1 (User Requirements for Audio Coding Systems for Digital Broadcasting, Annex 2) states that an audio codec (and the bit rate chosen) requires mean values consistently higher than 4.0 on the BS.1116-1 five-grade scale at the reference listening position. (See Table 1.)

Remember, a score of 4.0 on the BS.1116-1 scale is also equivalent to a diffgrade score of -1.0. Hence, looking at the results of the two audio coding systems discussed in the revious article (at the data rates tested), only the first system met the ITU-R criteria for broadcast quality. (See Figures 1 and 2.)

For a familiar example of what broadcast quality sounds like, consider that most Region 1 SD Hollywood DVD movies provide a decent benchmark for a codec being operated at a data rate that yields broadcast quality. However, high-definition DVDs typically use audio data rates at least two times higher than the rate of standard-definiton DVDs, and some even use a lossless audio codec. Therefore, with many broadcasters and next-generation service providers under increasing pressure to lower audio bit rates, the perceived quality between some next-generation broadcast systems, services and disc-based media (such as blue-laser DVD, for example) may be quite different in the near future.

Here are a few further items to look for with a properly administered and documented listening test:

• A graphical presentation of the test results
  
• General information about the audio coding system used to process the test material
  
• A specification for selecting test subjects as well as test materials
  
• Physical specifications of the listening environment, equipment, room dimensions, acoustic properties of the listening environment and transducer types/placement
  
• Detail regarding the analysis of the processed data
  
• A detailed basis for all conclusions
  
• Details of the test design, as well as the training process (instruction to test subjects)

Be wary if the test administrator, test facility, research facility or codec manufacturer cannot provide a complete set of supporting documentation regarding the details of its test and the basis for its results.

WHAT TO LISTEN FOR
Here is a high-level overview of what to listen for when evaluating a new coding system at a number of data rates.

The first thing to listen for is pre-echo, which is a type of impairment that affects (dampens) the sharpness and clarity of signals that are transient in nature. (As a side note, castanets are typically used to determine a codec's ability to handle transient signals across a number of data rates.)

Another type of common coding artifact is related to changes in timbre at higher frequencies and can sound similar to birds chirping (sometimes called birdies). This is most often caused by running a codec at too low of a data rate for spectrally demanding content.

Listen for a gritty or grainy sound quality, loss of bandwidth (typically in the high frequency region) — since many coding systems limit the coded audio bandwidth at aggressive (low) data rates — and image shifts with stereo or multichannel material. Many modern audio coding systems have a mechanism for synthesizing high frequency energy in the decoder from information that was generated and carried in the bit stream from the encoder.

TANDEM CODING LOSSES
Tandem coding losses, which affect perceived quality, occur when the coding errors in each system (used in tandem) combine to generate larger errors — that is, new errors created in addition to the old ones. These types of errors occur for several reasons, including:

• Quantization levels in one audio coding system do not map to the same levels in another.
  
• Different filter banks are used in the systems.
  
• There are time delays between the systems.
  
• There are changes in signal amplitude between the systems.
  
• Perceptual models are used.

To demonstrate the effect tandem coding losses have on perceived quality, consider Figures 3 and 4, which are both based on analysis performed in the lab with critical material. The results provide an approximation of the magnitude tandem coding losses have on perceived quality. Figure 3 shows a comparison between data rate and audio quality/relative coding error for two next-generation audio codecs available today. The x-axis indicates the data rate as a percentage of the data rate required for codec A to be at broadcast quality (as per ITU-R BS.1548-1). Codec B is more efficient where broadcast quality is about 80 percent of the data rate of codec A. If you were to operate codec B at a data rate 50 percent below the data rate required for broadcast quality with codec A, the quality would drop significantly to between “poor” and “fair.”

Many new codecs are designed for emission applications requiring the highest quality at the lowest data rate for only a single generation of encoding and decoding (that is, from the emission point to a viewer's home). These codecs are not designed for applications where different coding systems are operating in tandem (cascade) with each other (which is becoming commonplace throughout today's broadcast chain).

Figure 4 shows the effect of cascading the two different codecs versus single-generation performance. By placing the more efficient codec B in tandem with codec A (where A is being operated at a rate that is considered to be broadcast quality), the decrease in quality is significant (as shown by the curve with open circles). This is true even when both codecs are independently operating at data rates that yield broadcast quality.

As a real-world application example, consider an IPTV operator that is required to decode a broadcaster's signal that was originally encoded at a data rate that yields broadcast quality (codec A). The operator then needs to re-encode it into a more efficient format (codec B) for carriage to subscribers. Also assume that the IPTV operator chooses a bit rate (codec B) that yields broadcast quality (when codec B is in a standalone application (i.e. the source audio has never been through an audio coding system).

This data rate offers a 20 percent increase in efficiency over the bit rate that codec A needs to achieve broadcast quality. Note where each codec intersects the broadcast-quality threshold in Figure 4; codec B intersects broadcast quality at just below 80 percent of data rate required for codec A.

However, the result of both of these codecs operating in tandem with each other is described by the open circle curve in Figure 4. When codec B is operated as just described (20 percent lower bit rate than A), the net quality of both systems in tandem drops to between “good” and “fair.” Hence, true broadcast quality is no longer achievable in an application like this. Furthermore, many next-generation systems are looking to take the more efficient codecs' bit rates down to 50 percent of the data rate of codec A. Figure 4 shows that the tandem net quality when the bit rate of the more efficient codec (codec B) has been dropped to operate around 50 percent below the rate of the other codec (codec A). In this combination, yet a further drop in quality takes place to between “poor” and “fair.”

Before deciding on a target bit rate for a next-generation audio coder, consider that the realized efficiency gains of any new audio codec may be reduced in practice and will vary widely based on the application. This stems from the fact that in most, if not all cases, the new codec will be used in some portion of the distribution path to at least a portion of (or even all) of the viewers and will be in tandem with one or several different audio codecs. The advertised efficiency gains and quality some of these next-generation audio coding systems promise on paper will be different from what happens in real-world applications.

CONCLUSION
Before implementing any new coding system, audit the signal paths to quantify the number of codecs in tandem (don't forget that some broadcast servers use an audio codec too). Ask the experts lots of questions about steps to minimize tandem coding loss, and ask for suggestions on bit rates to minimize quality loss. The key: There is no substitute to the human ear. Listen carefully, and think about the level of quality your service requires. Also, does the new system cause any compatibility issues for the viewers?

Jeffrey C. Riedmiller is senior broadcast product manager for Dolby Laboratories.
Impairment Grade Imperceptible 5.0 Perceptible, but not annoying 4.0 Slightly annoying 3.0 Annoying 2.0 Very annoying 1.0
REFERENCES

• ITU-R BS.1116-1, Methods for the Subjective Assessment of Small Impairments in Audio Systems Including Multichannel Sound Systems
  
• ITU-R BS.1534-1, Methods for the Subjective Assessment of Intermediate Quality Level of Coding Systems (MUSHRA)
  
• ITU-R BS.1548-1, User Requirements for Audio Coding Systems for Digital Broadcasting
  
• ITU-R BS.1284-1, General Methods for the Subjective Assessment of Sound Quality
  
• Grant, Davidson and Fielder, “Subjective Evaluation of an Audio Distribution Coding System,” Audio Engineering Society Convention Paper 5443, September 2001
  
• “Perceptual Audio Coders: What to Listen For,” CD-ROM, Audio Engineering Society
  
• Tech 3253 - Sound Quality Assessment Material, European Broadcasting Union; www.ebu.ch/en/technical/publications/tech3000_series/tech3253/index.php