LAS VEGAS—The main attraction at the TSL Products booth, located in the South Lower Hall of the Las Vegas Convention Center, is going to be the latest updates to the company’s audio monitoring units, headlined by the new features of the PAM-IP range.

The PAM1-IP and PAM2-IP audio monitors now come with full support for ST-2110. Both units can be deployed to monitor audio and video content carried as ST-2022-6 and/or ST-2110 flows. The monitors also have two 10Gig/E SFP+ ports that allow users to monitor audio carried within ST-2110 and ST-2022-6 multicast streams; video content is displayed on the front panel. Audio only content, such as Dante or AES67, can be monitored when connected to a low-cost IP network infrastructure. Ember+ protocol is also supported for control with IP networks.

Another TSL product expected to be displayed at the booth is the SAM-Q audio monitoring platform.

TSL will showcase these products at booth SL4621.

To register for the NAB Show, visit www.nabshow.com.

I’m in the middle of a monumental project, which requires transferring audio and video content from the original media to a NAS. About 75 percent of the content is analog but the remaining portion is already digitized in some form. Getting files to the NAS is merely a matter of copying them directly to the server, but some of the digital content lives on media that is not file-based so must be ingested digitally into my workstation for cleanup and file creation. During this process I’ve periodically encountered content with digital levels over zero, making the file unusable, forcing a different method of ingest.

While frustrating, this phenomenon is not entirely surprising for the following reasons: the recording devices use lossy compression, the content was originally recorded as hot as possible without going into the red, and the audio meters were sample peak meters that are prone to miss peaks that occur between the samples. As it happens, these very reasons were among those that drove the call for “True Peak” meters.

ANSWERING A NEED

Examples of True Peak meters By now, every television audio engineer should be familiar with LKFS-based loudness meters and should be using them for loudness management, but the True Peak part of the meter seems to get left out of the discussion quite often. When I finally grasped what True Peak meters actually do I couldn’t figure out why everyone didn’t immediately start using them for recording and mixing. We finally have a meter that will tell us when we’re actually approaching 0 dB FS to keep us from clipping digital and people just shrug it off. Anyone who works on gear with built-in digital metering has experienced meters with really poor resolution along with that nagging feeling in the back of their brain questioning whether they can really trust them. Even the built-in meters on high-end digital audio workstations sometimes leave me wondering exactly what scale the meter is based on and whether it can be trusted. True Peak meters seem to finally answer the need for rock solid meters we can rely on.

An effort to find out more about the origins and ideas behind True Peak meters turned up several papers of interest: ITU-R BS 1770-3, AES R-7-2006 and AES Convention Paper 9041, which give us some of the background, the reasoning and points to consider as we use these tools in our day to day business.

In 2006, the AES and ITU recommended the creation of oversampling True Peak meters to resolve issues inherent in sample peak meters: to help engineers avoid digital audio clipping and to give more accurate, reliable readings. The specific problems cited as reasons for the new meter were inconsistent peak readings, unexpected overloads, under-reads and beating of metered tones.

‘NOT ENTIRELY ACCURATE’

True Peak meters in use in Pro Tools In the intervening nine years, a slew of manufacturers have answered the call and brought the meters to market. These meters are specifically designed to read peaks that occur between samples by interpolating additional measurement points, essentially upsampling the signal. An oversampling rate of four times was determined to be sufficient for the meter to properly read all peaks, so True Peak meters sample at 192 kHz (with a base sample rate of 48 kHz). All of this should mean that we finally have a digital peak meter we can trust since they pick up the peaks that sample peak meters miss. While that is true, it turns out that the value the meter displays may not be entirely accurate.

Signal flow in a True Peak meter is generally as follows: the signal is attenuated 12.04 dB, four times oversampling is applied, a low pass filter is applied, an absolute value is determined. While this seems to be a relatively simple process it turns out that oversampling may increase peaks by up to 3 dB, and filters in the chain could contribute phase shifts and overshoot, all of which can skew the accuracy of the meter. If the content being metered was created with a lossy codec, then additional overshoots or odd meter behavior may also be experienced. In other words, the complexity of the True Peak meter makes its displayed value less accurate than simpler peak meters.

Ian Dash, author of AES Convention Paper 9041, explains the situation rather starkly when he states “All True Peak meters will therefore have some error in their readings, unlike most other forms of digital level meter.” Fortunately the meters are not too far off. As long as readings are made at the oversampled rate, True Peak meters are no more than .5 to 1 dB inaccurate, but it makes a rock-solid case for setting True Peak limits to −2 or −3 dB TP rather than at 0. At this time there does not appear to be any standardized test of True Peak meter accuracy but Dash does list a couple of options in his paper.

At the end of the day a minor inaccuracy in the display of True Peak meters should mean very little to those of us using them. My recent experiences point out the sound logic behind their creation, especially when working with archival and lossy content. The meters are likely to become more accurate as technology, filter design and processing power improves but the fact is that they already give us a better idea of the actual peak values of our content. However, it is apparent that enough headroom should be left to make up for the minor inaccuracy inherent in the device and perhaps it’s worth considering lowering True Peak limits from −2 dB TP to −3 dB TP as a precaution against overloading the signal in the broadcast path. True Peak meters are one of my favorite audio tools of recent years so a 1 dB inaccuracy isn’t going to stop me from using them and it shouldn’t stop anyone else either. Let’s leave a little extra headroom and get back to work. I’ve got a lot more media to transfer.

Jay Yeary is a broadcast engineer specializing in audio. He is an AES Fellow and is a member of both SBE and SMPTE. He can be contacted via TV Technology.

RTW has delivered 145 TM3 TouchMonitor units to German broadcaster WDR. The units, which WDR has installed in its regional studios and editing suites, enable the broadcaster to comply with EBU R128 loudness metering.

With the development of EBU R128, the audio metering reference has shifted from PPM to loudness, with a peak level (QPPM) of -9 dBFS to a loudness target of -23 LUFS (Loudness Unit Full Scale). Since August 2012, when all German broadcasters agreed to implement EBU R128-compliant loudness metering, WDR has employed a large number of TM7 and TM9 units. Installing the TM3 for EBU R128-compliant loudness metering in smaller editing suites was a natural next step.

WDR selected the TM3, the smallest member of the TouchMonitor family, because it offers not only PPM and true-peak meters, also it supports all current loudness metering functions compliant with the major international standards (including EBU R128, ITU-R BS.1770-3/1771, ATSC A/85 and ARIB) for up to six channels. The broadcaster also liked that the TM3 could display measurements by single channels, summing bar graphs, loudness range or numerically, with the capability of separating displays as needed for easy viewing by multiple users. The TM3’s flexibility and easy-to-use setup software, which offers customizable administrative support, was also a factor in its purchase.

Mix Engineer Drew Vogelman, founder of The Garden, a renowned mix and dubbing studio in Brooklyn, New York, believes in using elements of both worlds to get the best sound. Vogelman has incorporated a TM7 TouchMonitor from RTW into his studio. The 7in, touch-sensitive, 16:9-screen TM7 enables him to monitor both the analog and digital aspects of his clients’ mixes, ensuring that a particular sound, whatever its source, truly contributes to the perfect track.

Named after the outdoor garden lounge area of the Brooklyn brownstone where it is located, The Garden comprises a large control room and a dubbing room with a Solid State Logic (SSL) AWS Hybrid Console/Controller at the center of its operations. Vogelman has outfitted the studio with a large selection of analog outboard gear and instruments — everything from vintage drum kits, guitars, synthesizers and amps, to plug-ins that replicate classic hardware and sound from storied music production studios and manufacturers — to give his clients a plethora of options for finessing their mixes. The TM7 takes things a step further by providing Vogelman a way to assess whether the sound of a particular analog element will complement a client’s particular mix.

Vogelman considers himself an old-school engineer, so he is a stickler for setting up what will be used on every mix. Because analog and digital are very different from each other, it’s important to go into a mix with an idea of what elements will work best for it. In a sense, the studio is going back and forth between the digital and analog realms, and the TM7 helps determine and maintain the right sound, regardless of its source.

The TM7’s intuitive graphical user interface, a feature of all models of the RTW TouchMonitor range, also helps Vogelman set up his tools for monitoring a mix. Users can control the interface with their finger, scaling, positioning and combining instruments in virtually any manner for the best use of available screen space. Multiple instruments of the same type, assigned to different input channels and configurations, can be displayed, along with other elements, such as meters — a feature Vogelman finds especially handy.

Once he sets up the GUI for a particular job, Vogelman employs the RTW “to bring everything out analog, through the desk, through the outboard." He sums everything back in through a Burl Audio ADC and takes the digital feed from the ADC to the RTW, so he can literally monitor the digital mix box and the analog mix box. This allows him to manage the analog mix head room and at the same time monitor and manage the digital head room, and the particular loudness factor.

As for loudness, Vogelman says it is as much of a concern in music mixing as it is in broadcast, so the TM7’s ability to monitor for all major audio loudness standards —including ATSC, EBU, ITU, ARIB and SPL, as well as custom standards — is a major benefit. Mix engineers have to be aware of and work with loudness issues, though with different sets of requirements. Artists are always aware of that loudness factor, and there are lots of opinions around what loudness is. We really hear a mid-range and, so, high fidelity is kind of about getting the lows and the highs in there, but modern mixes are very pointed on the mid-range. So being a mixer is about finding a way to almost satisfy both worlds in a way, whether you’re hearing it on a laptop or ear buds, and knowing that it’s going to cut through the clutter.

The TM7 TouchMonitor provides unparalleled flexibility and modularity combined with intuitive control. The software visualizes multiple sources simultaneously. It supports displaying the same signal on multiple instruments in parallel, each with dedicated defaults with both horizontal and vertical operation. The system visualizes up to 16 analog and/or digital sources at the same time. 

Broadcasters have monitoring rooms of varying sizes and shapes. To achieve consistent quality in recording, live productions and on-air transmission, complete control of the audio reproduction quality in all facilities is essential. This implies careful audio monitor installation, systems calibration and setup of the audio monitors in the production spaces.

Radiation space — a changing monitor behavior

Why would the sound of a monitor change if located in different places in a room? Are corners not a good place for monitors?

First, at low frequencies — below around 200Hz — monitors and subwoofers radiate very long wavelengths and generate a certain volume of air flow. This air flow spreads into all directions much like a pulsating sphere. Hence, if one limits the space — by means of a wall of floor — while keeping the monitor sound output identical, the low frequency will be boosted in the newly limited space. (See Figure 1.)

Figure 1. Decreasing the size of the radiation space boosts a monitor’s low-frequency sound level.

As a consequence, placing the monitor on the wall, or very close to it, will contribute to an increase of low-frequency sound level. The measured response of the monitor will no longer be flat. Hence, to avoid having monitors sounding boomy or bass-heavy, which will fool the subjective sound perception, it is important to correct the monitor’s response to maintain a flat and balanced frequency response in the listening area.

It should be noted that at high frequencies, the monitor no longer radiates in all directions. As the frequency increases and the wavelength becomes shorter, the monitor radiation becomes increasingly more directional. So, placing the monitor on the wall does not contribute to an increase of high-frequency sound level.

A source of cancellation — the wall behind the monitors

How is it possible that moving closer to the monitor makes little difference in low-frequency sound quality? Is it true that the farther away from the wall, the better the sound reproduction?

Here is a simple phenomenon: Two signals with the same level but in anti-phase (180 degrees out of phase) can cancel each other, resulting in silence. At low frequencies, if the monitor is placed a quarter-wavelength away from a sound-reflecting wall, the wave reflected off the wall will reach the monitor drivers in anti-phase. This will cancel totally, or partially, the initial signal radiated by the monitor at that particular frequency. How complete the cancellation is depends greatly on the ability of the wall to reflect sound in a specific frequency range. The sound level dips down at the frequencies where the reflected sound is in anti-phase. The depth and width of a cancellation dip varies, but in most cases such dips are audible and degrade the sound reproduction quality significantly. (See Figure 2.)

Figure 2. When a monitor is placed a quarter-wavelength distance from a wall, the reflected sound wave — now 180 degrees out of phase — can totally or partially cancel the initial wave.

No monitor equalization cures this problem; increasing the level of the monitor at the dip frequency also boosts the reflection, and thus their sum remains low and the dip is not removed. The most efficient solution to avoid such cancellations is to flush-mount the monitors in a hard (high-density) wall. For small monitors, placing them with their backs against a hard wall is also a good solution.

Negative interactions — walls and free-standing monitors

How is it possible that by adding a subwoofer, the satellite monitors are easier to place? How can the sound quality in small rooms benefit from an additional subwoofer?

Using a subwoofer with a crossover filter (typically set at 85Hz) between satellite monitors and the subwoofer can be useful. The satellite monitors do not reproduce low frequencies anymore; they are fed via high-pass filters. They can now be placed on the walls more freely, at distances where potential low-frequency notching does not occur in their limited low-frequency response. (See Figure 3.) The acceptable distance now extends out to 1.1m. (See Figure 4.) The monitors can also be placed farther away (1.1m to 2m) without seriously compromising the sound quality and tonal balance.

Figure 3. Using a subwoofer frees satellite monitors to be placed against walls.

Subwoofer — where to place it in the room

Does it matter where the subwoofer is located as it only radiates low frequencies? Is it not better to have the subwoofer as close to the listening position as possible? Why not put the subwoofers in the corner of the room?

One commonly observed location for a subwoofer is in the front center of the room, equidistant from the side walls. This position is often a compromise on the acoustical performance. In the case of a small room with parallel side walls, the subwoofer sits in the first pressure minimum of the lateral standing wave. The frequency response of a subwoofer in that location will most likely display serious frequency-response irregularities.

Figure 4. Each satellite monitor can be placed at least 1.1m from the wall.

A better position for the subwoofer(s) is on the floor close to the front wall and slightly offset from the middle axis of the listening room (avoiding then the first pressure minima), or in a corner close to both the front and side walls. The latter position maximizes subwoofer efficiency because of the limited radiation space, but it may also strongly excite various basic room resonances. Both solutions eliminate the most likely sources of cancellation dips in the subwoofer response. Another alternative is to use multiple subwoofers to achieve a more even excitation of the room resonances. However, the complexity of the low-frequency radiation increases with the number of subwoofers used.

One has to remember that the adjustments of gain and frequency response of the subwoofer are necessary during final in-situ calibration. The acoustical loading must be compensated for. The crossover phase adjustment must be set to maintain flat frequency response across the crossover region.

Correct vertical monitor placement

Why should monitors be physically oriented towards the listening position? Is it a problem to have monitors at different heights?

It is true that the vertical positioning of monitors is less critical than the horizontal one. However, inadequate monitor placement will affect the overall sound quality. In a typical stereo setup, it is essential that both monitors are positioned at the same height with their acoustical axis directed towards ear level. Similarly, in a multichannel configuration, the three front monitors should be positioned, ideally, with their acoustical axis at the same height.

If the brain has high capability to localize information on the horizontal plane, in the vertical plane the precision is (zenith angle) about 3 degrees above ear-level horizon and 3 to 10 degrees below ear-level horizon (azimuth angle). Because of this behavior of the ear/brain combination, human vertical localization tolerance is about 7 degrees. This allows two sources to be positioned at slightly different heights without the brain being disturbed by the difference.

This useful human hearing limitation can be exploited positively. In doing so, a center channel monitor can be placed above video screens or TV monitors. Then, one has to make sure that this center monitor does not suffer from a first-order ceiling or console reflection.

Sound coloration — the effect of early reflections

Why do large control surfaces affect sound reproduction quality? How can computer screens and racks affect sound imaging?

Early reflections are sounds that arrive at the listener after being reflected from elements in the listening space such as mixing consoles, tables, racks or computer screens. These reflections arrive later than the direct sound. If their levels are close to the monitor’s direct sound level, our brains will combine these reflections with the direct sound, and distinctive sound colorations will occur. Early reflections can also smear the coherence of sound images and compromise the localization of sound sources in the space between monitors.

So what is the solution? First, a symmetrical positioning of equipment is essential. Even with symmetry, reflections will remain, and everything possible should be done to remove all reflective surfaces between the monitors and the listening position. To improve the situation, monitors can be placed slightly above the typical listening height and tilted down towards the listener.

Calibration — the method to ensure quality and consistency

Why should a monitor be calibrated? Are the factory settings incorrect? Why would someone need to use advanced auto-calibration every time monitors are moved?

With all of the above explanations in mind, the reader should understand the importance of calibrating every monitoring system in its final installation to provide the best possible reproduction quality and consistency across production rooms. A monitor that is capable of automatically adapting to varying acoustical environments and correcting for levels, delays and room calibration is an indispensable tool for a broadcast sound professional.

Smart active monitors address all these demanding tasks and can be controlled with digital networking, allowing the implementation of highly flexible computer-controlled systems of monitors. Automatic calibration tools can measure and determine the system response and calculate all the correct acoustical compensations and correction parameter settings for each and every monitor and subwoofer. An automatic system can determine precise acoustical settings to give a flat frequency response at the listening position (or over an area via spatial averaging) using notch and shelving filters available in each monitor and subwoofer. Monitors can be time-aligned for equal delay from all monitors to the primary listening position, aligning monitor output levels and setting the subwoofer crossover phase. A complete calibration process typically takes less than five minutes for a full 5.1 system.

Conclusion

Today, most monitors are built with a flat response in anechoic conditions, but once they are placed into a listening room, their response changes because of room boundary loading, reflections, reverberation, time characteristics, etc. So, proper placement and precise adjustments of a monitor’s response are needed so that the complex monitor-room interaction stays optimal and allows for quality sound reproduction.

—Christophe Anet is education and training manager at Genelec Oy.