What is the .1 / LFE Channel and why does it exist?

The .1 or LFE is a channel which is exclusively used for frequencies below 120Hz (below 80Hz for DTS). It is therefore a bandwidth limited channel, but it is not the same as the subwoofer, which is the speaker that typically reproduces the LFE channel in a home audio or studio monitoring system (more information is provided below). The purpose of this channel is to provide an additional 10dB of electrical headroom, for low frequencies, above the main channels. This is done by calibrating the acoustic playback level, so that it has 10dB of in-band gain above the other channels (fronts and surrounds).

Blue Sky | SUB 15 Universal

Click this link to view an image showing how the LFE is calibrated relative to the main channels.

The LFE channel was officially launched as part of the film version of Dolby Digital, although its lineage actually dates back to six track 70mm soundtracks, and was mainly designed to enhance low frequency sound effects, such as large explosions and other low frequency content, which exceeded the headroom of the main channels. Hence the name LFE, or Low Frequency Effects, and not low frequency bass channel. It is important to note that all of the five main channels in a 5.1 delivery format, such as Dolby Digital or DTS, are capable of handling full-range audio (20Hz to 20kHz) on their own, without using the LFE channel. Because of this, the LFE channel isn’t always used, as it is not required for full-range audio.

Typically, unless the recording is something like Tchaikovsky’s 1812 Overture and is recorded with REAL cannons, the LFE doesn’t need to be used in most music applications, but can be useful in film and multimedia applications where dramatic low frequency effects are often important (explosions etc.)

AN IMPORTANT NOTE: Depending on the playback system, such as when a two channel system is being used instead of a full 5.1 system, the Dolby Digital decoder may drop the LFE channel completely (poof!).  Therefore persons creating a Dolby Digital mix should not put relevant audio content exclusively in the LFE channel.

OK, so if the LFE is a channel and the subwoofer reproduces this channel, what is bass-management and how does it fit in to the equation?

Bass management uses filters to extract low frequency information from the five 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).

A ‘typical’ bass-management signal flow diagram can be seen by clicking this link.

Bass-management is a function of the playback or monitoring system, in the same way that a crossover inside a speaker is part of the playback or monitoring system. It is a somewhat confusing name, for something that is basically a summation and crossover network – no more, no less.

The reasons for using bass-management are pretty simple:  It extends the frequency response of speakers that are not capable of reproducing full-range audio.  Since most 2-way speakers and studio monitors aren’t full-range and most modern recordings are, we feel bass-management, in combination with a subwoofer, is an extremely useful technology.

Click this link to again view the calibration diagram, which was also linked in the previous section. This time take note of how the bass-managed subwoofer signal is calibrated to splice in and extend the response of the each of the main channel speakers.

So what is the correct bass-management crossover frequency for my surround system?

Because bass-management is a function of the speaker system, determining the best bass-management crossover frequency for your audio system is completely dependent on the type of speakers and subwoofer you are using.   If you are using a THX Certified home theater speaker system, or a THX pm3 approved studio monitoring system, then an 80Hz setting is typically desirable.  This is also the default setting for THX receivers, THX controllers and Blue Sky’s BMC MK II.

If you are not using a THX Certified or pm3 approved system, then you will need to check with the manufacturer of the speakers you are using to see which crossover frequency setting is correct. 

Additional notes:

• Certain receivers and controllers offer expanded options with regard to filter settings.  These settings may include individual controls for high pass and low pass filters.  For a THX Certified speaker system, or THX pm3 approved monitoring system, these should be set to; 80Hz, 12dB per octave High Pass and 24dB per octave Low Pass.   Again, if you don’t have THX certified or pm3 approved speakers, check with the manufacturer of your speakers to see which settings you should use.
• For additional information about the benefits of using bass-management and why it is incorporated into every Blue Sky monitoring system, please follow this link.
• For more information about Dolby, please visit www.dolby.com
• For more information about DTS, please visit www.dts.com
• For more information about THX, please visit www.thx.com

If you have comments, questions or suggestions, please don’t hesitate to contact ADL, or post them below in the comments section.

The reasons for the lack of improvement are complex and often vary depending on the content creator, broadcast network and even differences between national and local broadcast networks. To help understand what is causing these ‘issues’, it is probably best to first review one of the key technologies that is available today, to help mitigate this problem.  Dolby Digital, which is currently used as the preferred audio delivery format for many HDTV broadcasts, includes several unique features which are designed to help improve consistency between various broadcasts and other audio sources.

Dolby Laboratories realized a long time ago, that there can be radical differences between different types of sources, channels and content, and that it would be extremely inconvenient for a TV viewer to continually have to adjust the audio level while watching TV.  Because of this, Dolby Labs decided to implement a feature called “Dialogue Normalization”.  Dolby includes this feature in what is known as the “Metadata” of any Dolby Digital bitstream. Metadata is non audio information, which helps to describe the audio content in the bitstream to the Dolby Digital decoder in your HDTV, home audio receiver, etc.

Dialogue Normalization works by assigning a relative numerical value to the average dialogue level of a given program, commercial, movie etc.  This numerical value should be calculated using hardware or software tools offered by Dolby and is then encoded into the Metadata by the content creator or broadcaster.   The graph below (credit www.tvtechnology.com) shows some of the typical relative levels between different types of source material.  The black line and the numerical number next to that line, would be the dialog normalization level.

This numerical value is used by the Dolby Digital decoder to automatically adjust the level of the audio, so that during playback all these different types of program material have the same average dialog level.  The graph below (credit www.tvtechnology.com) shows the resulting adjusted level, which is automatically implemented by the the decoder.  Theoretically, and in practice when done correctly, the consumer would now not need to adjust the level between these different types of program material and they would perceive a very consistent level across all of these different types of program material.

So if Dialogue Normalization is such a powerful tool, why hasn’t the situation gotten any better, especially with the advent of HDTV and Digital Broadcast Television? The answer is complicated and there are many different possible causes.  Below are some of my theories:

• Certain content is purposely mixed to be perceived by the listener as being louder: Commercials have for a long time been mixed so that their average level is much higher.  Although the peaks are technically not louder, the content is compressed and the dynamic range of the content is very low, so it is perceived as being much louder by the listener.  This is often done by the creators of commercials so that their commercials ‘stand out’. Although dialogue normalization, when properly implemented, would address this problem, it is easy to manually assign a dialogue normalization value which would make the decoder play it at any relative level the content creator, or broadcaster would want.
• Lack of consistent production and delivery standards between TV broadcasters: Unlike the film industry, which has very rigorous calibration, production and playback standards & practices, many broadcasters have very unique delivery specifications, which can impact the accuracy of the Dialog Normalization levels and audio levels in general.
• Local broadcast affiliates: Although the major national broadcasters are improving their implementation of dialog normalization, during a national broadcast the local affiliates and cables systems play their own local commercials, which are inserted during the broadcast.  Often, and particularly in smaller markets, there is a lack of expertise and more often than not, a lack of hardware to properly support dialogue normalization.  If this is the case, the local feed may not be broadcast in Dolby Digital and there can be huge differences in level between local and national broadcasts.  This also applies to locally produced TV shows and news broadcasts, which in these smaller markets may not take advantage of Dolby Digital and its features.  With the transition to DTV in 2009, this particular issue should improve over time as local affiliates continue to upgrade their hardware and gain experience with these new tools.

So, how can we consumers, the content creators & broadcasters improve this situation:

• Complain: The first thing we can do as consumers is to complain to the local affiliates, cable systems, broadcasters, etc. Broadcasters won’t know about, or deal with a problem, unless they are receiving complaints. Many broadcasters, especially at the national level, want your feedback and this will add to the internal pressure to resolve these types of issues.
• Education: The industry as a whole needs to do a better job of educating themselves on the proper setup of audio systems, using recognized standards, and also learn how to properly implement features such as Dialog Normalization.
• Industry Wide Production, Delivery and Broadcast Standards: The film and movie theater industries used to have a tremendous problem with the sound level of movie previews, in much the same way as consumers do with TV commercials.  The typical way this was handled by the movie theater operators was to turn down the level of the previews, when someone inevitably complained.  Unfortunately, the level would rarely be turned back up for the feature and this often left the viewers with a less than engaging audio experience.  Producers & directors, upset about their movies not being presented at the proper playback level, along with the National Organization of Theater Owners, began to complain about this problem and the industry as a whole took steps to address the issue. The industry implemented a set of standards, through an organization called TASA, and today all previews are tested to conform to a certain playback level standard.  Previews that don’t conform to this standard are typically not shown in a movie theater. The lesson from the film industry is clear; broadcasters and content creators need to work together, with manufacturers and industry organizations, to develop standards and practices to address this problem.  Doing so will result in a much more engaging experience for consumers and will reduce their frustration with this annoying problem.

For more information about Dialog Normalization, please follow these links (updated 7/23/2008):

• A Closer Look At Audio Metadata, by Tim Carrol | www.tvtechnology.com
• Estimating Dialogue Loudness with Leq(A) or ITU-R Rec. BS.1770 | www.dolby.com
• Managing DTV LOUDNESS with dialnorm, by Jim Starzynski | www.broadcastengineering.com
• Dialogue Normalization: Friend or Foe, by Brian Florian | www.hometheaterhifi.com
  

This article was also published in the October 2001 edition of Pro Sound News. You can download a PDF copy by following this link to the Blue Sky website (130KB Adobe Acrobat PDF).

Through the ages, science and art have always been closely linked.  These days people have forgotten how much individual artistic talent goes into truly magnificently engineered products.  An example of a product that requires a considerable amount of technical and artistic talent is an accurate studio monitor.  This article will show how a new technology, known as BOO®, can bring artistic vision and ideal acoustic performance to the marketplace more quickly and take out much of the guesswork.

Designing a modern loudspeaker involves many factors, including:  Output requirements, room acoustics, directivity requirements, psychoacoustics (how people perceive sound), and design philosophy.  Design philosophy can also be interpreted as the artistic vision for a product.  Artistic vision / design philosophy is typically made up of the factors, that a company or engineer has discovered through research and experience, which make a speaker sound “good”.   For Blue Sky International these factors usually include such things as smooth on & off axis frequency response, flat on axis response, a wide listening window, and meeting a certain output requirement.

Unfortunately the process of getting from a set of components and the necessary cabinet, to a “good” sounding speaker, is hit or miss at best.  Prior to starting the design of a loudspeaker, a manufacturer will typically take the performance data of the drivers, design a box and then start the process of modeling a crossover & EQ.  Typically a software program is used that takes into account some of the driver parameters and the box parameters and then creates a “rough” equalization and crossover design.  The problem with this simulated crossover network is that the software rarely considers on & off axis frequency response, electrical response and the many ways to get to the same point.  Additionally, the real world measurement data rarely correlates with the predicted acoustic response and always requires many tweaks.  These tweaks can take a huge amount of time and are akin to a painter having to try hundreds of mixtures of paint, then paint the entire canvas and then start over again because it doesn’t look right.  This process is wasteful and time consuming, plus it distracts the engineer from getting to the “artistic vision” and acoustic performance criteria that are important to him or her.

To deal with this problem, Audio Design Labs created a software program called BOO™ which is an acronym for Binary Organic Optimization. BOO™ combines a high-powered electro-acoustic simulator with an optimizer based on a genetic mathematical algorithm called differential evolution. Genetic algorithms are simplified mathematical models of the biological processes that organisms like bacteria use to mutate and evolve. You can apply the same strategy to optimize electro/acoustical circuits.  In this case, the speaker is the organism, the crossover the DNA, and the components are the genes. The program is given a target response curve, which is the goal of the evolution process. The program then creates a population of organisms (speakers) with genes (component values). Then parents are selected out of the gene pool and combined with a randomized vector (mutation) to create children. The children that match the target better than their parents survive (evolve). The ones that are inferior to their parents die (don’t survive).  Thereby, a new population of organisms is created and the process repeats until the process can evolve no more (the crossover is the best it will be).

BOO™ has now created an optimal crossover design, which takes into account the many complex characteristics of the drivers and the enclosure.  Multiple parameters can now be evaluated, including the electrical signal that will be sent to each driver, the on & off axis frequency response and how close the system performance will get to the design goal.  If the system doesn’t perform as desired, changes can be made (new tweeter, woofer, different box, new crossover topology, etc.) and BOO™ can re-run the evolutionary process until the system is performing as desired.

At this point it is time to build a real world crossover and speaker system, measure the performance of the product, compare it to the predicted results and move on to the next step of the design process, listening tests.

It must be said, that it is extremely rare for BOO’s predicted performance to be off by any great degree (if any).  In fact, real world measurement errors have been discovered using BOO’s predictions.

BOO™ is a proprietary technology first used in the development of Blue Sky International’s Sky System One, made up of the SAT 6.5 and SUB 12.  As a side note, no embryos were destroyed in the development of BOO™ but we may have lost a mouse or two.

click to download (Adobe / PDF)

To find out more about Blue Sky’s complete line of studio monitoring systems, follow these links to Product Reviews and Blue Sky News on the Blue Sky website.  If you have any questions, or would like to find out what ADL can do for you, please don’t hesitate to contact us.

This article was also published in the October 2001 edition of Pro Sound News. You can download a PDF copy by following this link to the Blue Sky website (130KB Adobe Acrobat PDF).

This article was also published in the October 2006 edition of Resolution Magazine. You can download a PDF copy by following this link to the Blue Sky website (143KB Adobe Acrobat PDF).

Intro

About 8 years ago, my business partners at Blue Sky and I had a number of meetings to discuss designing a new series of studio monitors. After much debate, we decided there was little point in building another set of “me too” monitors, in the typical 5”, 6” and 8” 2-way configurations. What we felt was needed was a different approach that overcame some of the limitations of current monitors and provided tangible real world benefits to the recording engineer.

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 effects 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 the effect of 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.

click to download (Adobe / PDF)

To find out more about Blue Sky’s complete line of studio monitoring systems, follow these links to Product Reviews and Blue Sky News on the Blue Sky website.  To learn more about how ADL designs speaker and studio monitors, please follow this link. If you have any questions, or would like to find out what ADL can do for you, please don’t hesitate to contact us.

This article was also published in the October 2006 edition of Resolution Magazine. You can download a PDF copy by following this link to the Blue Sky website (143KB Adobe Acrobat PDF).

• Bass-Management / subwoofers:
  
  Bass management (which is used in 90% + of all home theatre / audio 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 single mono subwoofer (sometimes multiple subs are used for greater SPL or to improve in-room low frequency performance). 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).
  
  As a general rule, bass-management should be used in any professional recording studio that is creating content for direct consumer consumption (stereo, surround, music, broadcast, radio, DVD etc.) and that is why it is incorporated into all Blue Sky monitoring systems (design and manufactured by ADL).  As mentioned above, most home theater systems, all home THX certified audio systems and many premium car audio systems already use subwoofers and the requisite bass-management electronics.
  
  For a more in depth overview of this concept, please follow this link.
  
  
• Broadband absorption and “bass traps”:
  
  One of the simplest ways to address low frequency standing wave problems is to install broadband absorbers.  These are most commonly made of home insulation products, such as rigid fiberglass or Rockwool and have a material density of around three pounds per cubic foot.  These types of acoustic treatments are usually framed or placed in a ‘fabric bag” and then mounted onto the walls of the room / studio.  For the greatest effectiveness at low frequencies, it is typically desirable to mount them across a corner, or with an air gap between the wall and the broadband absorption panel. 
  
  The second, more narrowly focused “bass traps”, are those that use a helmoltz resonator or a membrane type of absorber.  Because of the way these are constructed they tend to be more expensive, when compared to standard broadband absorption, but they can be more effective at lower frequencies. 
  
  Below is a short list of links to companies which offer complete broadband absorbers and narrow band ‘bass traps’ for purchase and / or kits to help end users fabricate their own broadband absorbers:
  
  Auralex Acoustics: http://www.auralex.com/
  GIK Acoustics: http://www.gikacoustics.com
  MSR Inc: http://www.msr-inc.com
  Ready Acoustics: http://www.readyacoustics.com/
  RealTraps: http://www.realtraps.com
  RPG Diffuser Systems: http://www.rpginc.com/
  
  
• Automated Room Correction / Room Equalization:
  
  Room EQ has become increasingly popular, with regard to possibly addressing room related acoustic problems.  Especially within the latest high-end consumer receivers, automated room correction has become a relatively prevalent feature.  Unfortunately the effectiveness of these products is still heavily debated.  Our general view is that they can be helpful, but that they are most effective when used in rooms with proper acoustic treatment and they certainly are not a substitute for proper acoustic treatment.  The reason for this is usually related to the inability of automated systems to accurately measure the room and speaker interaction when there is not enough acoustic treatment in the room.  The microphone and processing systems get overwhelmed by the reflections and the result is less than optimal equalization.  Additionally, because many of these systems can only optimize for a relatively small listening area, they are often not ideal for optimizing or resolving general room acoustic problems.  Lastly, there is only a limited amount of EQ available and it is often not sufficient alone at addressing the peaks and dips in most rooms. 
  
  Below are a few related links on this topic:
  
  Audyssey: http://www.audyssey.com/technology/index.html
  Meridian: http://www.meridian-audio.com/w_paper/Room_Correction_scr.pdf
  RealTraps: http://www.realtraps.com/art_audyssey.htm

Mid to High Frequency | Flutter Echoes & Discrete Reflections:

Reflections building up between two parallel surfaces, such as side walls, are called “Slap Echoes” or “Flutter Echoes”. If the side walls are too reflective a slap echo will develop. Slap echoes have a bright metallic character and can severely affect the sound quality. Discrete reflections are a dominant effect in small rooms / studios which typically exhibit low reverberation times. They give a distant and “smeary” character to the sound, which severely degrades the ability of a pair of speakers to convey a proper stereo image and can also greatly reduce dialog intelligibility. Sound reflections can occur from side wall, rear walls, furniture, video monitors, or any other large surface within the path of the speakers’ radiation  coverage area.

You can help limit the impact of flutter echoes & discrete reflections in three basic ways; absorption, diffusion and speaker placement / aiming:

• Absorption:
  
  For most home audio and studio applications, absorption should probably be your primary method for addressing this type of acoustic problem, especially for the first reflections off of the side walls, ceilings and floor.  As explained above, broadband absorption can also help to improve the low frequency characteristics of the room and is also a relatively inexpensive method of acoustic treatment. 
  
  Below is a short list of links to companies which offer complete broadband absorbers for purchase and / or kits to help end users fabricate their own broadband absorbers:
  
  Auralex Acoustics: http://www.auralex.com/
  GIK Acoustics: http://www.gikacoustics.com
  MSR Inc: http://www.msr-inc.com/
  Ready Acoustics: http://www.readyacoustics.com/
  RealTraps: http://www.realtraps.com
  RPG Diffuser Systems: http://www.rpginc.com/
  
  
• Diffusion:
  
  The second method which can be employed to address flutter echoes and reflections is diffusion.  Diffusers work by scattering reflections around the room or studio randomly, rather than combining with direct sound and causing spectral or time based distortion. Diffusion is especially useful in rooms where acoustic absorption is already used to a great extent and additional absorption is either not practical, or is making the room sound unnaturally “dead”. Although you can use items such as bookshelves filled with books to help defuse the reflections, it is typically much more effective to use a dedicated product specifically manufactured for this purpose.
  
  Below is a short list of links to companies which offer acoustic diffusers:
  
  Auralex Acoustics: http://www.auralex.com/
  GIK Acoustics: http://www.gikacoustics.com
  MSR Inc: http://www.msr-inc.com/
  RealTraps: http://www.realtraps.com
  RPG Diffuser Systems: http://www.rpginc.com/
  

• Speaker Placement:
  
  Although speaker placement will not solve all mid to high frequency acoustic reflection problems, it can help to limit their severity.  In recording studios one particularly useful solution, is to put your front monitors on speaker stands, instead of placing them on the recording console bridge.  This will greatly reduce the reflections off of the console, which are typically the source of the first reflection in a recording studio.  Also, aiming the speakers at the listening area and away from reflective surfaces, can help reduce the severity of the reflections.
  
  For more information about Blue Sky’s “The Stand”, please follow this link.

Reverberation Time (RT):

In the room sizes we are talking about, reverberation times are rarely too long to impact sound quality. However, tonal smoothness of the reverberation time can affect sonic performance. The ideal room should exhibit a completely smooth decay of sound, in which no special sonic event or peculiarity stands out. Also, rooms with too little reverberation because of excessive absorption, can have sonic problems of their own (see the diffusion section above). In most cases, living rooms, home theatres and smaller control rooms, with a regular amount of furniture don’t suffer from reverberation time problems.

Additional Educational Resources:

Blue Sky International: http:www.abluesky.com
GIK Acoustics: http://www.gikacoustics.com/education.html
RealTraps: http://www.realtraps.com/articles.htm

Design Services and Acoustical Consultants:

Charles Salter Associates: http://www.cmsalter.com/
Keith Yates Design Group: http://www.keithyates.com
Munro Acoustics: http://www.munro.co.uk/
Performance Media Industries Ltd: http://www.pmiltd.com/
Russ Berger Design Group: http://www.rbdg.com/

If you have any comments, questions or suggestions regarding this topic, please leave them below, or contact ADL directly.

Instructions for electro-acoustic calibration of a professional stereo, 2.1 and 5.1 studio monitoring systems (speakers) using a SPL meter and ADL’s test files.  If you are interested in calibrating a home theater system, we recommend using the test signals and procedures included with your home theater receiver.

Before starting, download the ADLTestFiles.zip (15 MB). To download the zip file, “Right Click” and select “Save Target As” and the file will begin downloading.

Once downloaded, either burn the test files on to a CD or import them directly into your DAW and follow the instructions below.

Additional Required Items

• Stereo, 2.1 or 5.1 Monitoring System
• SPL Meter – such as the SPL meter sold by RadioShack in the U.S.

ADLTestFiles.zip Includes 4 files:

• 1000Hz SINEWAVE -20dBFS.wav – a 1kHz file recorded at -20dBFS for electrical calibration
• 40-80Hz PINK NOISE -20dBFS.wav – a 40Hz to 80Hz bandwidth limited pink-noise file recorded at -20dBFS
• 500-2.5kHz PINK NOISE -20dBFS.wav – a 500Hz to 2500Hz bandwidth limited pink-noise file recorded at -20dBFS
• Pink Noise full bw -20dBFS.wav – a full-bandwidth pink-noise file recorded at – 20dBFS

All the test signals were created and measured on the Prism Sound dScope III and conform to the AES 17 guidelines for digital audio measurements. Because pink noise has a high crest factor, the levels of the pink noise signals were made using a time averaging technique, where multiple time samples are captured and averaged over a ten second interval to calculate the RMS level. These test files are all mono files. Please make sure you hard assign them to the left and then the right, not both channels at the same time. If you are using a CD / CD player, use only one channel of the CD player.

Theory
The purpose of calibration is to adjust the overall electro-acoustic system gain so that 0dBVU of electrical signal level equals a certain acoustic level at the listening position. Since most recording media is now digital, the reference electrical signal level is usually –20dBFS with 20dB of headroom. The reference SPL  however can vary based on the delivery media and speaker type.

Please note that the bandwidth limited signals that have been provided, limit many of the room interaction affects often associated with measuring SPL and broadband pink noise.

*Also note that the LFE channel gain in 5.1 formats varies from 0 to +10 dB depending on the encoding format used.  This level is referenced to the bass-managed subwoofer level, which is listed as “SUB” in the chart below. Since the LFE channel is not calibrated as a separate entity, the LFE gain will not affect system calibration.  It is important not to confuse the bass-managed subwoofer calibration level with the LFE channel.  For more information about bass-management and the LFE channel, please follow this link.

The common calibration levels are listed below:

**IMPORTANT: When compared to movie soundtracks, music tends to be much more compressed, with reduced dynamics and greater overall level.  Because of this, music mixes may seem too loud when played back on a system calibrated for an 85dBc reference level. If this is the case, we recommend calibrating for a lower reference level, such as 78dBc.

All test signals are recorded at –20dBFS including the 1 kHz sine wave tone. The sine wave tone is used to set the electrical output level throughout the signal path, right up to the point you get to the speakers. While the various pink noise signals are used for acoustic measurements and calibration.

The following procedure assumes you are calibrating the system to 85dBc SPL. If you are calibrating to TV, etc. substitute the appropriate level from the above chart.

1. TURN OFF THE MONITORING SYSTEM (until step 4)
2. Remove all eq and dynamics from the signal path and set all controls to zero / unity gain. Play the 1kHz Sine Wave, hard assign it to the left channel only, and adjust the output fader so the so the output meter reads -20dBFS. If you are using an analog console or measuring the output of your D to A, set the output level to 0dB VU (typically 0dB VU is equal to +4dBu / 185 nanoWebers per meter / 1.23 Volts RMS – using a true RMS volt meter). Then hard pan the signal to the right channel output and repeat for the right channel. For a 5.1 system assign the channel to each output and repeat the adjustment procedure. Once calibrated do not move the output faders.
3. Mute everything and make sure the 1kHz tone is OFF.
4. Now that the system has been electrically calibrated, turn ON the SUB and SATs.
5. Assign the 500-2.5kHz pink noise signal to the left channel only. Make sure there is nothing coming from the right channel (or any other channels). Because this signal is bandwidth limited, you don’t have to worry about turning the sub off. There are two methods of setting the levels.
   1. If you have a master monitor controller, with individual channel calibration adjustments (such as Blue Sky’s BMC), you can set the monitor gain controls on each speaker at the reference / max gain position and then adjust the master level control for 85 dBc. Then mark that master monitor level as your ‘reference position’ and use the individual channel calibration adjustment on your controller, to calibrate the rest of the channels, as outlined below. This method may not always be practical and if your monitors have too much gain, may result in a in poor signal to noise ratio.
   2. The other method is to set master monitor level to the position you want as a reference level and then use the individual monitor gain controls, until you measure 85 dBc. If you use this method you should mark the knob position, on each monitor, with a grease pencil so you can always go back to the reference level if the knob gets moved.
      For either method:
      SPL should be measured at the mix position, with the SPL meter at arms length, with the microphone at seated ear height, angled at approximately 45 degrees, and pointed at the center point between the left and right speakers.Once the left channel is set to 85dBc, repeat this step for the right channel (and C, LS & RS in a 5.1 system).
6. Feed 40-80Hz pink noise signal to the left channel only. Adjust the subwoofer level control until the subwoofer reads 85dBc (slow) at the mix position. The meter will bounce around a little, so you will need to do a mental average (I tend to filter out the peaks in my mind, so I don’t set the sub too hot). The right channel should measure about the same and no additional adjustments need to be made.
7. You can play the full-bandwidth pink noise, assigning it to the left and then the right channel (not at the same time). You should measure about 85dBc. It may be a little higher, because below 30Hz the room may have a little extra gain.No adjustments should be made with Full Bandwidth pink noise, unless you have an RTA (real time analyzer).
8. You are finished and the calibration process has been completed – enjoy!

If you have any questions, please don’t hesitate to contact us, or leave a comment to this topic.

Blue Sky | SUB 15 Universal

Today, many consumer and professional monitoring systems incorporate a subwoofer as an integral part of the system. Making the subwoofer an integral part of the monitoring system design relieves the satellite speakers of the requirement to reproduce low frequencies. Smaller satellite speakers are more easily placed for the best imaging and the sub can be positioned for the best bass response.

But what is the best location for your subwoofer?

Overview
The low frequency response and efficiency of a subwoofer are heavily influenced by the acoustics of the playback environment. More specifically, the response is influenced by the room’s dimensional ratios, types of construction and location of the subwoofer within that environment. You can significantly improve the subwoofer’s in-room response and efficiency by experimenting with various room placements until you find an optimum location.

When placing the subwoofer there are a few general guidelines that should be kept in mind. These include:

• Every acoustic space is unique and experimentation is an important key in finding the best possible location in your particular environment.
• A subwoofer becomes more acoustically efficient (has greater output) as you move it closer to a room surface (e.g. wall or floor).
• A subwoofer will give maximum output and maximum acoustic excitement when it is located in a corner.
• Under certain acoustic conditions corner locations are optimum; in others they can excite multiple “room modes”, producing “muddy” or “boomy” sound.

The following methods have been found to work successfully under most conditions.  These methods assume that you are using a single subwoofer in your room / studio.  For information on placing multiple subwoofers, please follow this link to the “Sound and Vision” website.

The first method described below doesn’t require any special test equipment. It does require a pair of good ears and familiar broad-spectrum music material – recordings with lots of energy across a wide frequency range (from low to high). The recordings should be highly dynamic and be the highest possible quality.

The second subwoofer placement method requires using a real time analyzer – such as those made by Gold Line™ or The Audio Toolbox(TM) by TerraSonde(TM), or a free software tool such as the “Room EQ Wizard“. Although many home audio enthusiasts may not have access to this equipment, if you are having a sound system or home theatre system professionally installed, this information may be useful to the installer.

Method 1
Place the subwoofer at the main listening, play a CD or other music source and make sure that the level of the subwoofer has been raised high enough so that low frequencies are not masked by the background noise in the room. Once you have roughly balanced the level, between the sub and main speakers, move around the room and pay careful attention to where the spectral response is smoothest and has the greatest low frequency extension, pay special attention to the corners and along the walls. Also, make sure to pay attention to where the system has its greatest impact and definition. You are not just listening for the most boom, but rather where the bass is most accurate and natural sounding. Remember, because the subwoofer is basically omni-directional, the best spot for the subwoofer may be next to, or even behind, the main monitoring area.

After finding the spot where the subwoofer has the best response in the room, place the subwoofer in that location. Now, listen from the main position and confirm that the subwoofers response is similar to when the positions were reversed. If it is, then leave the subwoofer in that location. If not, continue to experiment with the subwoofer location until the most accurate and best response has been achieved.

Method 2
Subwoofer placement using a real time analyzer – Such as those made by Gold Line™ or The Audio Toolbox™ by TerraSonde™, or a free software tool such as the “Room EQ Wizard“.

Place the subwoofer at the main listening position and connect the subwoofer to your pink noise generator. Turn on your pink noise generator and make sure that the level of the subwoofer has been raised high enough so that low frequencies are not masked by the background noise in the room. Now set the analyzer to 1/12 octave resolution (or whichever setting provides the highest resolution on your particular analyzer), real time mode and begin to take measurements around the room. If the analyzer you are using has the ability to do real time averaging, then use this function to better analyze the spectral response (for audio analyzers which works in the time domain, such as the “Room EQ Wizard“, these analyzer settings do not apply)

As you move the microphone around the room, be sure to pay careful attention to where the spectral response is smoothest and has the greatest low frequency extension, pay special attention to the corners and along the walls. Again, it is important to remember that because the subwoofer is basically omni-directional, the best spot for the subwoofer can be next to, or even behind, the main monitoring area.

After analyzing the data and finding the spot where the subwoofer has the best response in the room, place the subwoofer in that location. Now, take some additional measurements from the listening position and confirm that the subwoofers response is similar to when the positions were reversed. If it is, then leave the subwoofer in that location. If not, continue to experiment with the subwoofer location until the smoothest and best response has been achieved.

Additional Notes
It has been found that a subwoofers’ in-room response can sometimes be improved by facing the drivers toward a wall. Again, experimentation is the key to finding the best possible location.

Properly designed subwoofers generate tremendous energy, so they may vibrate objects close to them. If you hear buzzing or vibrating objects, make sure to try and dampen those objects. Rattling, buzzing and other sympathetic resonances can make the subwoofer localizable and therefore should be avoided. Using a sine wave generator can be helpful in locating these acoustic anomalies.

Blue Sky | "The Stand'

Proper placement and aiming of the speakers in your studio or home audio system, can greatly improve the audio experience.  Below are suggestions for placement of speakers for three general applications; stereo, 5.1 music and 5.1 film.  If at all possible, speaker placement should be symmetrical relative to the room boundaries and the listening position.

Suggested Guidelines For Stereo Monitoring Placement

The recommended position for the left and right speakers, in a stereo configuration, is at a 60 degree angle relative to the listener, forming an equilateral triangle (a triangle with equal sides) as shown in the Reference Monitoring Position diagram (click to view). Fortunately, this setup eliminates most of the math and is easily simplified to the following guidelines: If you want to sit 1 meter (39.37 inches) from the speakers, place the speakers 1 meter apart. If you want to sit 6 ft from the speakers, place the speakers 6 ft apart. Etc.

Ideally the speakers should be at seated ear height as show in the Monitoring Height Recommendation (click to view). If this is not possible, tilting the speaker cabinet at the listening area can improve high-frequency coverage.

Speaker Placement 5.1 Music Optimized System

As described above, the recommended listening angle for proper stereo imaging with music is 60 degrees between the left and right speakers and this doesn’t change when expanding to 5.1 channels of audio. The center channel speaker should be located on axis with the reference listening position, and both the left and right surround channel speakers should be at an angle of 110 degrees from the centerline (click to view).

In most applications, surrounds used for music are placed at the same height as the front speakers for a direct sound field (click to view). If this is not possible, tilting the cabinet at the listening area can improve high-frequency coverage.

Speaker Placement 5.1 Film/Video Optimized System

Although the recommendation for music applications should work equally well for film, there may be situations where a more “film” optimized setup is desirable. To correctly relate audio to picture, the recommended angle between left and right speakers is 45 degrees. This narrower listening angle should still yield a very satisfactory stereo image. As with the “music” setup, the center channel speaker, should still be located on axis with the reference monitoring position. Unlike music surrounds, which tend to be direct in nature, film surrounds are usually positioned for a more diffused sound field to simulate the effect of an array of surround speakers used in a theater. For a single pair of surrounds, this can be accomplished by placing them two feet above seated ear height, to the side and slightly by behind the main listening position.

For more information on this subject, along with 6.1 and 7.1 surround system configurations, please visit Dolby Laboratories’ website. For information about subwoofer placement, please follow this link.