So where did we begin?

When creating the requirements for Blue Sky’s nearfield and mid-field studio monitors, we decided to start at the end of the chain and work backwards to the beginning.

Over the last 25 years audio technology has changed and improved dramatically. 25 years ago, people in their homes were listening to scratchy vinyl records and watching movies on analog cable or VHS, in mono. The speakers they used were what came built into their TV, or at best, stereo 2-way ported bookshelf speakers. In their cars, they had a choice of AM radio, FM radio, cassette tapes, or eight track tapes, which were played through 6 x 9 speakers with whizzer cones. Back then the typical 2 way studio monitor was much better than what the typical consumer had available to them for playback.

However, things have changed greatly since then and today’s consumer now has a dizzying array of choices, such as HDTV, digital cable, satellite TV, DVDs, MP3, video games, etc. All these sources are capable of high bandwidth, wide dynamic range stereo or multi-channel audio. Likewise, speaker technology has also improved with the utilization of stereo and 5.1+ speaker systems, with integrated subwoofers. Similar technology advances have occurred in the car audio world. Drivers now have access to CDs, DVDs, digital satellite and HD radio. And like the home, many autos now have complete, factory installed, stereo and 5.1 bi-amplified speakers systems, also with integrated subwoofers.

Following a parallel course, most studio equipment has improved much over the last two decades. With the introduction of digital audio workstations, hard disk recorders, digital mixers, all of which include high performance 24 bit ADCs and DACs, studios today can essentially record DC to daylight. Unfortunately, studio monitors in many studios have not kept up with this trend and in many cases seem to be stuck an infinite time loop. Like 25 years ago, many professionals are still mixing content for today’s consumer systems, on a set of conventional 2-way ported monitors, which tend to exhibit poor low frequency extension. Looking at this situation, we felt that there had to be a better way.

The ‘typical’ studio monitor, in a typical studio

So next we moved one step up chain, to examine the precise reasons why a typical 2-way nearfield monitor is no longer suited to the task of creating mixes destined for today’s consumer.

Almost all 2–way nearfield monitors are ported, use a 5”, 6”, or 8” woofer and have a low frequency cutoff between 38 and 65 Hz. Since ported designs roll off at 24 dB per octave, or greater, these monitors are incapable of reproducing much of anything from 20 Hz up to their lower cutoff frequency. This kind of performance was perfectly acceptable when consumer playback systems had similar performance limitations. However, since many of today’s consumers have full-range speaker systems with subwoofers, the typical 2-way ported design, just doesn’t cut it anymore.

The other problem is what happens when you place these monitors in a typical recording studio. Although major movies have their final mix completed on a large dubbing stage, typically with a volume of 20,000 cubic feet or greater, a lot more material is mixed in studios with an internal volume closer to 3000 cubic feet, especially for music, radio and TV applications. Unfortunately, as the physical dimensions of the studio get smaller, the acoustic conditions change as well. The biggest change occurs at low frequencies, which in a large space is an issue relating to low frequency reverberation time. When you move into a smaller studio, the main acoustic factor at low frequencies is room modes, or standing waves. Room modes occur in all rooms / studios at frequencies where the wavelength of sound is an integer fraction (i.e. 1/1, 1/2, 1/3, 1/4, etc.) of the distance between two walls, or the distance between the ceiling and floor.

Whenever you place a speaker in a small room or studio, its measured low frequency response will be altered by the boundary affects and room modes that form in the studio. This means that invariably some frequencies are reinforced and some frequencies are canceled, resulting in peaks and dips in the frequency response at the listening position. These frequency variations change depending on the location of the speaker and where the listener is located in the studio relative to the speaker and boundaries of the studio. Because of this, bass reproduction from multiple speakers in a studio can be very inconsistent. To add a further complication, the speaker location that is best for imaging is almost never the best place for bass reproduction. So given little choice, recording engineers choose imaging over low frequency performance.

Granted, the use of broadband absorption, which we consider very important, can reduce to the affect of the studio / monitor interaction to a degree and absorption can definitely be utilized to improve low frequency performance. But, in a typical small studio, broadband absorption will not a fully address these problems.

So what is the solution?

True full-range monitoring, is the phrase which best describes our goal. Not just full-range monitoring for those willing to spend huge amounts of money on large in-wall monitoring systems, but true full-range monitoring for all applications, from the desktop on up. The technologies we employed to achieve this goal are based on well understood principles of physics, are relatively simple to implement and deliver superior real-world results.

First, to provide real low frequency reproduction, reduce intermodulation distortion and to reduce the influence of the studio on low frequency reproduction, we decided to incorporate a subwoofer as an integral part of the monitoring system. This is in sharp contrast to an optional subwoofer added on to an existing “quasi full range” 2- way monitor.

The second improvement was to eliminate ports or passive radiators and go with sealed box designs. There were three reasons for doing so: One, sealed box speakers have superior transient response when compared to ported or passive radiator designs. Two, satellite speakers using the correct sealed box design integrate much better with a subwoofer than typical ported speakers. Third and last, the 12 dB per octave roll off of a sealed box subwoofer provides a better match to the rising low frequency characteristics of small rooms / studios. This ‘room gain phenomenon’, which was documented in an AES paper by Louis D. Fielder of Dolby Labs, shows that smaller sealed rooms, such as the typical music studio, exhibit a 12 dB per octave gain below 30 to 35 Hz. This ‘room gain’ response perfectly matches the sealed box response of our subwoofers, allowing for incredible in-room low frequency extension, down to below 20 Hz. Compare this to a typical ported or passive radiator roll-off of 24 dB per octave, or greater, and you can see why the sealed box response is a much better choice for accurate full-range monitoring in a typical recording studio.

The third improvement was to tie this all together with a technique called bass management or bass redirection. Bass management uses filters to extract low frequency information from two or more main channels and redirects that bass to one or more mono subwoofers. This is the same technique that is used in virtually all consumer home theater systems and many high-end car audio systems. Bass management when used in conjunction with satellite and subwoofer speakers provides a number of advantages. First since the satellite speakers do not have to reproduce low frequencies they can be smaller, which make them easier to place in the environment, and they can be placed for best imaging without worrying about how that would affect their low frequency performance. Second, since bass reproduction is coming from a mono subwoofer, the subwoofer can be placed in the optimum position in the studio so as to offer the best overall low frequency response. Thirdly, because low frequencies from multiple channels are now summed electronically, instead of acoustically in the studio, low frequency phase issues between channels are resolved in the most absolute and accurate way possible – electrically.

The critical final step

Now that we have a monitoring system which has the bass being reproduced by a separate source, we have to find a way to ensure repeatable and accurate setup of the system. We have found that many listeners can actually do this very effectively by ear, when using familiar broadband source material. However, to make this process a little more foolproof and repeatable, we provide a free set of wav test files which allow the end user to quickly adjust the electro-acoustic level of the system. These test files can be downloaded from our website. To use them, you just need an inexpensive SPL meter.

The purpose of calibration is to adjust the relative level of the SUB and SAT, along with the overall electro-acoustic system gain, so that 0 dB VU equals a certain acoustic level at the listening position. Since most recording media is now digital, the reference electrical signal level is typically around –20 dBfs with 20 dB of headroom. The acoustic calibration level may vary, depending on the application and standards being used. For film applications this level is typically 85 dBc, but because music is typically more compressed, a lower level, often around 78 or 79 dBc, may be more appropriate.

Once the calibration procedure is completed, the end user has a system which provides extended bandwidth, seamless summation between SAT and SUB, along with an overall more accurate and repeatable system response. We believe this new methodology, which is based on simple and proven technology, makes for a clearly superior monitoring system. This true full-range monitoring system design allows the engineer to create more compelling full-range mixes that translate exceptionally well to the wide variety of modern consumer playback systems currently available on the market.

In small acoustics spaces (generally defined as rooms of less than 12,000 cubic feet in volume, with a more common room volume being approximately 3000 Cubic Feet, such as 16’H X 9’W X 21’L) the frequency response of a speaker system in the region below 100Hz is dominated by the modal response of the room. Room modes, also known as standing waves, occur in all rooms at frequencies where the wavelength of sound is an integer fraction (i.e. 1/1, 1/2, 1/3, 1/4, etc.) of the distance between two walls, or the distance between the ceiling and floor (this is a slightly over simplified explanation). This means that invariably, some frequencies are reinforced and some frequencies are canceled, resulting in peaks and dips in the frequency response at the listening position. These peaks and dips are affected by the relative position of the speakers to the boundaries in that room. Because of this, it is virtually impossible to get consistent bass response from multiple full-range speakers located around a “small” room (such as in a Stereo or 5.1 monitoring setup).

One solution to this problem is to employ a method called Bass Management, also referred to as bass redirection. Bass Management uses electronic filters to extract the low frequency information (typically below 80Hz) from the main channels and then reroutes that information to a single subwoofer channel (reproduced by one or more subwoofers). Since the low frequencies will now originate from a single source (a subwoofer) this source can be placed in the optimum location for bass reproduction in that room. And, because the main speakers are not required to handle frequencies below 80Hz, they can be reduced in size and easily placed for best imaging and coverage. The end result is that the overall frequency response of the entire audio system is considerable improved, without any sacrifice in performance or imaging.

One might ask, won’t I perceive a difference in imaging if the sound of one channel originates from two sources (subwoofer and satellite)? The answer is actually no. Bass Management works by taking advantage of the ear’s inability to determine the direction of frequencies below approximately 150Hz. Provided there is no audible distortion or sonic artifacts at higher frequencies (port noise etc.), and the sound emanating from the subwoofer is limited to below 100Hz, it will be impossible for the listener to identify the location of the subwoofer in the room.

This lack of auditory acuity is based on the fact that the wavelengths of frequencies below 100Hz are much greater in length than the distance between the listener’s ears. However, our ears can easily identify the source of high frequency information as coming from the main speakers. Because the sound of the main speakers is the listener’s primary audio location cue, the listener’s brain believes that the bass is actually emanating from the main speakers and not from a separate subwoofer (even when it is behind the listener).

With a properly designed satellite/subwoofer speaker system, using Bass Management, the response and overall accuracy of a monitoring system can be greatly improved. These benefits apply to any type of monitoring, whether two channels, 5.1 and beyond.

Bass management (which is used in 90%+ of all home theatre systems and most premium car audio systems) uses filters to extract low frequency information from the main channels and then reroutes that information, along with the LFE Channel in a 5.1 system, to a mono subwoofer (multiple SUBs can be used for higher SPL, or to improve frequency response over a wider area). The advantages are overwhelming and include smaller main speakers that are easier to place, better LF response, reduced inter-modulation distortion and more repeatable LF response from room to room (small or large).

For additional information about the benefits of bass-management, please follow this this link.

To understand why this is the case, we must first understand how our brains process location cues from our ears. Above approximately 700 Hz (depends on the size of your head), your brain uses Interaural Level Difference (ILD) as the primary factor in determining the directional location of a sound (slightly over simplified explanation). ILD is the difference in level of a sound, between your two ears.

Below approximately 700 Hz your brain begins to rely on the Interaural Time Difference (ITD) between your ears, also known as – phase shift, to determine the directional location of a sound. This works very well until the wavelengths get very long, the source becomes omni-directional, such as a subwoofer, which radiates energy spherically in its pass band, and you are in an enclosed space. In an enclosed space, such as a studio, with a source that is radiating spherically (again, such as a subwoofer), the ITD will be close to zero. This is because energy from the source is arriving at the listener from many paths, with many overlapping time differences and your brain will not be able to derive the primary location cues from your ears. Therefore your directional acuity at these low frequencies will be near zero. However, you will have very high directional acuity at higher frequencies and because your directional cues are coming from the SATs, which typically are playing the harmonics of the LF fundamentals, this is where your brain believes the sound is coming from. Provided there is no audible distortion or sonic artifacts at higher frequencies (port noise etc.), and the sound emanating from the subwoofer is limited to below approximately 100 Hz, it will be impossible for the listener to identify the location of the subwoofer in a studio.

Since Blue Sky’s design goal was accurate reproduction of the source material and not raw output per se; Blue Sky choose a sealed box design for the subwoofer and the satellite.

So why is there all this talk about acoustics? For one thing, acoustics are audible! We’ve all had the experience of walking into a room that is completely devoid of furnishings and playing around with the long, reverberating echoes. Stuffing that room full of furniture and personal items kills Little Sir Echo, but there are lots of other acoustic phenomena that are totally unaffected by our knick-knacks. While these other acoustic phenomena are not as readily identifiable as a long echo, they still exist and wreak havoc on an audio system.

The plain truth is that acoustics largely determine the perceived sound quality of an audio monitoring system in a project studio – by 50% or more in most cases. On the surface, the contribution of acoustics may not be as easy to understand as the contribution of some new piece of electronic wizardry, but the fact remains that the typical audio monitoring system in a project studio can be improved more by the implementation of acoustic treatments than by the addition of any piece of electronic equipment.

Knowing that room acoustics is important wouldn’t do us much good if there were nothing we could do about it. Fortunately for us, acoustic problems in studios are fixable! Purveyors of expensive electronics would have dealers, studio designers, and engineers believe otherwise, but there is no argument that can stand up against the evidence. The trick – the thing that confuses most people to the point where they cry “uncle” – is how to fix room acoustics.

Before we can learn how to fix room acoustics, we must first understand a little bit about what makes acoustics tick. Acoustics can be thought of as the interface between a speaker and a listener in the same way that an audio interconnect is the interface between two electronic components, and a speaker cable is the interface between an amplifier and a speaker. The elements of a cable interface are input/output impedance, connectors, wire, resistance, inductance, and capacitance; the elements of an acoustic interface are speakers, air, reflections, and listeners’ ears. Just as we would not want a poorly-shielded cable interface made from wire with high resistance, we would not want an acoustic interface with lots of destructive reflections and strong echoes. There are many good cables on the market today that provide great interface among electronic components. The picture is not so rosy for the acoustic interface. There are relatively few products available to control the acoustic interface, and many that are available can actually be detrimental to it rather than improving it.

What, then, should we expect from an acoustic treatment product, so we will know whether or not it is actually working for the greater good of an audio system? Acoustic treatments, as we will discover in greater detail, interact with things called reflections, flutter echoes, reverberation, and standing waves, which, if untreated, reduce clarity and articulation, confuse sound localization, collapse soundstages, and shift tonal balance. Therefore, a good acoustic treatment product will enhance clarity, articulation, and localization; open soundstages; and restore an even tonal balance.

Reflections

Up to this point, we have only introduced the concept of reflections. Now, we will take a more detailed look at the nature of reflections, the ways they degrade our audio systems, and the means we have at our disposal for treating them.

Reflections occur in every room, whether it is large or small. So, unless an audio system is outdoors, its sound will be affected by reflections. In an acoustic interface, they are comparable to distortion in an electrical interface. (The video guys in the crowd will recognize reflections as the things that cause ghosting in pictures.) If an electrical signal is distorted, we cannot hear the signal in its entirety. The same is true for reflections. If an acoustic signal is riddled with reflections, we cannot hear the original, pure sound.

A reflection is a sound that has bounced off one or more surfaces in its path from a speaker to a listener. We all know that speakers, even directional ones, do not radiate sound on a laser line directly to our ears. Speakers fire sound out in an infinite number of directions. True, a little of the sound radiated by a speaker does go straight from the speaker to our ears, but a lot of it bounces off of some surface (wall, ceiling, floor, console) first. When our ears combine all this reflected sound with the small amount of sound that comes straight from a speaker, the result is severe acoustic distortion!

Absorbers

One way to minimize the detrimental effects of reflections is to absorb them using treatments that are, remarkably, called absorbers. Absorbers, such as the StudioPanel Absorber, are like acoustic vacuum cleaners that suck in sound energy and convert it into heat energy through a resistive process. Little, if any, sound is reflected off of an absorber. The effectiveness of an absorber is detirmined by its thickness, which, contrary to popular opinion, mainly affects the range of sound absorbed, not how much sound is absorbed! For example, a 1″ think absorber absorbs sound over a range from 1,000 Hz to 20,000 Hz, a 2″ thick absorber from 500 Hz to 20,000 Hz, and a 4″ thick absorber from 250 Hz to 20,000 Hz. Naturally, if we want to absorb as many reflections as possible, the thicker an absorber is, the better. Unfortunately, to absorb reflections over the entire range of audible sound, an absorber would have to be 64″ thick! In the world of studios, 4″ thick absorbers provide the best compromise between range of absorption and practicality.

Diffusers

Diffusion is another method for treating reflections. Diffusers, like the StudioPanel Diffuser, control reflections by breaking them up into many “little” reflections that bounce around a room randomly rather than combining with direct sound at our ears and causing acoustic distortion. Like absorbers, diffusers only perform their magic over a certain range of frequencies which is – you guessed it – determined by the depth of the diffuser (among other things).

There is a rhyme and reason to using a blend of absorption and diffusion in a project studio. A general rule of thumb is to implement a blend of 50% absorption and 50% diffusion. If too much absorption is applied, the resulting sonic character of a room is too “dry” and “dead”. On the other hand, too much diffusion can spray an overabundance of little reflections around a room and confuse soundstaging.

Bazorbers

We’ve shown that we can’t use traditional Absorbers or Diffusers to control reflections over the entire range of audible sound. Does this mean that we have no way to treat reflections that are not absorbed by the StudioPanel Absorber or diffused by the StudioPanel Diffuser? Absolutely not! We may not be able to use the same type of absorber for reflections below 250 Hz that we used for reflections above 250 Hz, but there are absorbers designed specifically for the range of sound below 250 Hz. In the past, most of these absorbers have utilized one of two different approaches: Helmholtz or diaphragmatic. The StudioPanel Bazorber is a combination of both methods, taking advantage of the best of each! It does its work from 100 Hz to 250 Hz.

StudioPanel Absorbers, Diffusers, and Bazorbers are normally placed on the walls of a room in locations where reflections occur. Diffusers are usually placed directly across a room from Absorbers to insure that no reflections or flutters echoes (which we will discuss next) that are outside the Diffuser’s range of operation can develop between Diffusers. In most cases, Bazorbers are placed on the front wall of a room to reduce reflections in the upper bass. Reflections across the entire range of audible frequencies are now controlled, with the exception of very low frequencies. However, single point reflections at very low frequencies are not as problematic as another acoustic phenomenon known as standing waves. We will save the discussion of these mysterious standing waves for a later section.

Flutter Echoes

In addition to low, mid, and high frequency single point reflections, we must control pesky things called slap or flutter echoes. Flutter echoes occur when sound bounces back and forth between two large, flat, parallel surfaces. In rooms, we call these surfaces walls. Like reflections, which are close relatives, flutter echoes reduce clarity and articulation, confuse sound localization, collapse soundstages, shift tonal balance, and lead to bright sound with a characteristic “zingy” quality. Fortunately, StudioPanel Absorbers and Diffusers are very efficient over the range of sound where flutter echoes develop, so the Absorbers and Diffusers can be effectively employed to control flutters echoes.

Reflection Decay Time

Reflection decay time is another acoustic phenomenon that we must control in a project studio. After a period of time, the reflections in a room that are not absorbed combine to create an ambiguous wash of decaying sound. The time that is required for this wash of sound to decay to a certain level is called the reflection decay time of a room. Reflection decay time is very important. If the time window is too long, clarity and articulation will be reduced, sound localization will be confused, and stereo separation will suffer. Extensive research has been done to determine the proper level and time window for reflection decay time. This research shows that most people prefer a time window of about 0.2 to 0.4 seconds in a room the size of a typical project studio.

In large rooms, reflection decay time is called reverberation time, which is a statistically random soundfield with no particular time or direction component. Rooms the sizes of project studios are not big enough to exhibit true reverberation because the reflections die out before they reach fully random character.

Reflection decay time is largely determined by the percentage of surface area in a room that is covered with absorptive material. Rooms with little or no absorption will have time windows that are too long. For those of us who are not intimidated by numbers and math, there are equations that predict the reflection decay time of a room. The latest and most accurate equation is known as Arau-Puchades:

RT = {0.161V/[-S ln (1-ax)]}x/s x {0.161V/[-S ln (1-ay)]}y/s x {0.161V/[-S ln (1-az)]}z/s

We can use the data from this equation to prescribe the proper amount of absorption for a room…after researching absorption coefficients, calculating surface area, and simplifying complicated math problems. For those of us who are more afraid of math than we are of acoustics, StudioPanel is a true blessing. The engineers who created StudioPanel did the calculating for us, so all we have to do is pick the proper kit for our room based on square footage!

Standing Waves

We have now covered all the topics of acoustic reflections but one – perhaps the most intriguing, exciting, and complex one of all: standing waves! So what, exactly, are standing waves? We know they mess up the bass in our project studios, but what causes them and how do we get rid of them?

In order to understand what a standing wave is, we have to know something about sound waves. (Don’t be alarmed, because we’re not going to dive into advanced physics.) Sound waves of various pitches happen to be different lengths. Sound waves that we associate with bass are very long; sound waves that we associate with treble are really short. The rest of the sound waves we hear lie somewhere in between. Now, it happens that, when a sound wave is exactly as long as the dimension of a room, that wave will resonate in that room. A resonating wave is louder than waves that are not resonating, and also takes longer to decay. (Reference the above discussion of reflection decay time.) Such a wave is called a standing wave. In addition to the original standingwave, whose length matches that of a room dimension, other standing waves will develop when sound waves are one-half, one-and-a-half, two, two and-a-half, three, etc., times the length of a room dimension. If we consider that standing waves occur between all three pairs of wall surfaces in our project studios, we can understand why standing waves are so detrimental to sound quality!

For most project studios, the sound waves that resonate in the length, width, and height dimensions are all bass sound waves from 30 Hz to 150 Hz. The increased volume and longer decay times of the resonating bass sound waves totally destroys any chance of bass sounding clean, tight, and chestpounding like it does in large venues, commercial cinemas, and outdoor concerts. Furthermore, standing waves are not uniform across a room. Certain places in a room will experience much louder bass than others. We can only hope that our mixing position has the same level of bass as the producer’s couch!

It is easy to see that we must do something to eliminate these bass standing waves. How do we go about it? Fortunately, we have a whole arsenal of ways to treat standing waves. Some ways are acoustic, some are electrical, and some are structural. For example, during the design phase of a project studio, we can adjust the dimensions of the room so that the bass sound waves that resonate are all different. (If two room dimensions are or are almost the same, the resonating bass sound waves that correspond to those dimensions will aggravate each other, creating even greater sound pressure level variations and longer decay times.) In addition, we can place loudspeakers, subwoofers, and listening positions so that their interaction with standing waves is reasonably limited. We can also use electronic equalization to reduce the volume of the resonating waves. The StudioPanel SpringTrap is another way to eliminate standing waves. Like the Bazorber, the SpringTrap is designed to absorb bass sound waves that are below the range of the StudioPanel Absorbers. However, SpringTraps operate over an even lower range of sound than Bazorbers. SpringTraps are effective from 30 Hz to 100 Hz.

Unlike Absorbers, Diffusers, and Bazorbers, SpringTraps do not need to be placed on walls at reflection points in order to function properly. Due to the nature of standing waves, SpringTraps work most effectively when they are positioned in the corners of a room, either sitting on the floor, or hanging just below the ceiling. The SpringTrap’s unique shape accommodates easy and aesthetically-pleasing corner placement.

Room Acoustics Summary

Reflections, flutter echoes, reflection decay time, and standing waves are all acoustic phenomena that ruin the sound in our project studios. A blend of StudioPanel Absorbers and Diffusers can be used to control reflections, kill flutter echoes, and lower the reflection decay time so that it lies within acceptable tolerances. At and below frequencies where the Absorbers and Diffusers cease to operate, Bazorbers take over to control boundary reflections. Finally, SpringTraps can be used to eliminate bass standing waves that mess up low-end kick. Together, the whole StudioPanel package works really well, and will pay for itself after you’ve mixed a couple of records that sound great when they leave your studio!