Loudspeaker and loudspeaker enclosure design aren’t haphazard processes if done right. Good design encompasses a combination of science, art, engineering and craftsmanship. Simply constructing an arbitrary wooden box and mounting some drivers somewhere on the front panel will most likely not produce optimal results.

As would be expected, the theory and practice of loudspeaker and enclosure design have been continually developing and evolving over the decades since the first loudspeaker was produced and this research is carried on to this day. But a real turning point in describing enclosure design mathematically and predicting performance before screws and glue were put to wood came about in the early 1960s with a paper by A. Neville Thiele, first published in Australia in 1961 and republished in the Journal of the Audio Engineering Society (JAES) 10 years later.

Thiele investigated equivalent circuits of loudspeakers in vented boxes and discovered, according to his paper “Loudspeakers in Vented Boxes,” that “it is possible to make the low-frequency acoustic response equivalent to an ideal high-pass filter, or as close an approximation as is desired.” His method, he wrote, “provided a reasonably precise method of design that was previously lacking.”

A second researcher, Richard H. Small, took Thiele’s ideas (among others) and expanded upon them producing multi-part seminal papers in the JAES in late 1972 and throughout 1973, “Direct-Radiator Loudspeaker System Analysis” and “Vented Box Loudspeaker Systems.”

THIELE-SMALL PARAMETERS
In the first part of the latter paper, Small credited Thiele’s paper as “the first to provide an essentially complete, comprehensive, and practical understanding of vented-box systems on a quantitative level.”

Fig. 1: From “Vented-Box Loudspeaker Systems Part 1: Small-Signal Analysis” by Richard H. Small (JAES June 1973) The Thiele and Small approach began with a certain group of parameters for a given low-frequency (LF) driver. (Their analyses applied to low-frequency response.) These parameters came to be known as, not surprisingly, the Thiele-Small parameters, or T-S parameters for short (see Fig. 1).

The T-S parameters can be measured in the lab—some more easily than others—with both Thiele’s and Small’s papers presenting measurement protocols. Or they can be provided by loudspeaker driver manufacturers who have already done the lab work.

If certain parameters of a LF driver are known, one can then proceed to work through the equations Thiele and Small provided to develop the enclosure design. Or a loudspeaker system manufacturer can work at this from another direction.

If, for example, they have a certain box size in mind and specific acoustical, electrical and physical design goals, they can then work through the formulas to help develop drivers with the appropriate T-S parameters.

According to Thiele, a system that has a good flat low-frequency response down to a predictable cut-off frequency can be designed if the following three parameters are known about a LF driver.

The first is the free-air resonant frequency of the driver. The next, according to Thiele, is “the ratio of electrical resistance to motional reactance at the resonant frequency,” also called the total Q. And the third is the volume of air that has the same acoustic compliance as the driver suspension.

These parameters result from the physical and electrical construction of the driver.

The free-air resonance frequency is measured with the driver not mounted in any enclosure and is a peak in the driver’s LF response. It is dependent on the weight of the driver components that move, like the diaphragm (cone), dust cap and voice coil, and how much that movement is restrained by the driver’s suspension elements like the spider and surround.

The restraining effect of the mechanical suspension on the driver’s motion also factors into the total Q number along with the opposing effect of the electromagnetic (EM) part (the magnet and voice coil) that propels the cone back and forth with an applied signal.

The volume of air parameter corresponds to the stiffness of the driver’s spider and surround.

With these three parameters in hand, the box design can start. We’re not going to go into the formulas, as they are well-documented in the literature, but rather give a general idea of the design process, which is much more detailed than presented here.

THE SIZE OF THE BOX
One of the first things we’d like to know is how big a box is needed. The volume is calculated using the volume of air and total Q parameters. This is the actual volume of air needed in the box to create a maximally flat high-pass filter. The actual size of the box will be larger than this volume to allow for the space taken up by such items as the drivers, support bracing and the vented port. The volume of these items should be added to the calculated volume to give the total volume of the enclosure.

It’s up to the designer to determine the length, width and height of the box that provides the calculated volume. That’s where a lot of the art of design comes in. In his paper, Thiele suggested using ratios that would be used in good acoustical room design.

Another point that Thiele made is that “the box volume is closely proportional to the inverse square of cut-off frequency, which can be varied over a wide range.”

Since the box will act as a maximally flat high-pass filter, it will have a LF 3 dB down point. That value is calculated using the free-air resonance of the driver and the total Q. These two parameters are used in a different equation to obtain the resonant frequency of the box.

As might be expected, a larger LF driver will require a larger enclosure and have a lower cut-off (3 dB down) frequency.

Now we turn our attention to the port. Small’s paper on vented box design provides guidelines for the area of the vent (port) and the diameter of a circular vent so that peak air velocity through the vent will be limited to avoid air noise. We don’t want any air whistling or rustling through the tuned port.

Knowing the inside radius of the vent, as well as the box resonant frequency and box volume, the duct length can be calculated. Settling on a port size and duct length can be an iterative process as can be the design of the box itself.

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.

For any kind of loudspeaker system to be practical, the drivers need to be mounted in some kind of enclosure. But not just any old box will do, at least in a well-designed system.

The enclosure is an integral part of the proper acoustical functioning and overall performance of the entire system. A poorly designed or constructed enclosure can degrade even the best engineered and manufactured driver. One of the functions of an enclosure is to control and enhance low frequency response.

As noted in my last column, a loudspeaker driver moves back and forth in response to an input audio signal, producing sound waves from both its front and rear. The sound emanating from the rear is out of polarity (180 degrees out of phase) with that coming from the front.

If the driver was just hanging in free space, unenclosed, the sound from the front and rear would mix at the listener’s position. Since they are out of polarity with each other, they would add or null at different frequencies forming irregularities called comb filtering in the frequency response. (The term comb filtering presumably came about because of the shape of the peaks and dips as plotted on a frequency response chart, level vs. frequency.)

With an enclosure, the rear sound energy could be made to do something constructive. While there are a different ways to design an enclosure, we’ll give a general overall of two common types found in control room monitoring.

SEALED BOX DESIGN
The first is a sealed box design, sometimes called acoustic suspension. As might be expected from the name, the box is sealed top, bottom, and sides, with the drivers mounted on the front. (Actually a tiny bit of leakage is necessary and allowed to normalize air pressure.)

JBL Professional’s compact Control 1 Pro loudspeakers feature a professional crossover network coupled with professional drivers, housed in a rugged molded enclosure. Since the box is sealed, the sound coming from the rear of the drivers can’t escape into the outside world causing the dreadful comb filtering. Often this type of construction includes sound absorbing material inside the box, which tends to be more effective at mid to higher frequencies. Air pressure inside the box changes as the driver moves in and out. This is more pronounced at lower frequencies due to the longer wavelengths. That cushion of inside air, described as a “spring” in the literature, helps support the driver, effectively stiffening its suspension and altering its resonant frequency.

A sealed box is not that efficient compared to other designs, since a lot of the energy the drivers produce is contained or absorbed inside the box. Frequency response may not be able to go as low as some other box designs. On the plus side, this type of design can come in compact sizes, as its other common moniker, bookshelf speaker, can attest, and can provide good transient response even at lower frequencies.

VENTED BOX
Another and more common type of box is the ported enclosure, also called “vented box” or “bass reflex.” This type has a deliberate pathway and opening for the rear sound energy to leave the box in a controlled and beneficial way, with the port operating within a defined low frequency (LF) band. You can feel the air moving out of the port when the loudspeaker system is operating.

The pathway inside the box and the port are designed and built in such a way that the energy emanating from the port is in polarity with the front sound wave from the LF driver. The port is tuned to a specific range of low frequencies where this occurs.

Within this LF range, since the two wave fronts are in polarity, they reinforce each other, increasing the bass output. This has the benefit of increasing efficiency, compared to the sealed box design, since more acoustic watts are produced per electrical watt input. The driver doesn’t need to be driven (so to speak) as hard to produce a reasonable LF output. This allows it to operate in its more linear range, keeping distortion down.

This design may not be as tight as the sealed box for low-frequency transient response. While the ported output arrives in polarity with the direct output from the LF driver, it has to travel a little longer distance to get there. So it would be expected that there would be some sound spreading in the time domain. This lingering persistence of sound, often referred to as time smear, tends to be well-controlled in well-designed systems.

It’s important to remember to keep the port free and clear of obstructions when installing the loudspeakers. Or in other words, don’t block the port, or the intended performance will be degraded.

The loudspeaker enclosure has to accommodate more than just the drivers. There’s the connector panel, internal wiring, and depending on the design, passive cross over networks, and in the case of “powered” loudspeakers, power amps.

While the descriptions of these two enclosure types have been simplified, the actual design of the boxes is anything but, a topic for another time.

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.

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.